From 140828bc7a084b0492f2d1c01520927461bb2e86 Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Mon, 28 Apr 2025 12:08:55 +0000 Subject: [PATCH 001/292] Allow to provide prebuild Environment to construct PeerConnectionFactory That would allow to share the Environment across PeerConnectionFactory and its injected dependencies, e.g. AudioDeviceModuel Bug: webrtc:413413572 Change-Id: Ic99a2e39664e224a3d63e73ba8e584cbee838bfd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/388140 Commit-Queue: Danil Chapovalov Owners-Override: Harald Alvestrand Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44473} --- api/BUILD.gn | 1 + api/peer_connection_interface.h | 13 ++++++ pc/peer_connection_factory.cc | 32 ++++++++++---- pc/peer_connection_factory_unittest.cc | 42 ++++++++++++++++++- .../RTCPeerConnectionFactory+Native.h | 2 +- .../RTCPeerConnectionFactory.mm | 18 ++++---- .../RTCPeerConnectionFactoryBuilder.mm | 2 +- .../RTCPeerConnectionFactoryBuilderTest.mm | 8 ++-- .../iwyu/iwyu-verifier-filter_list.json | 1 + 9 files changed, 94 insertions(+), 25 deletions(-) diff --git a/api/BUILD.gn b/api/BUILD.gn index 7a3591881f..6340d2bed4 100644 --- a/api/BUILD.gn +++ b/api/BUILD.gn @@ -366,6 +366,7 @@ rtc_library("libjingle_peerconnection_api") { "crypto:frame_decryptor_interface", "crypto:frame_encryptor_interface", "crypto:options", + "environment", "metronome", "neteq:neteq_api", "rtc_event_log:rtc_event_log_factory_interface", diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h index de071d5b04..bffbf070d1 100644 --- a/api/peer_connection_interface.h +++ b/api/peer_connection_interface.h @@ -92,6 +92,7 @@ #include "api/data_channel_event_observer_interface.h" #include "api/data_channel_interface.h" #include "api/dtls_transport_interface.h" +#include "api/environment/environment.h" #include "api/fec_controller.h" #include "api/field_trials_view.h" #include "api/ice_transport_interface.h" @@ -1430,9 +1431,18 @@ struct RTC_EXPORT PeerConnectionFactoryDependencies final { Thread* worker_thread = nullptr; Thread* signaling_thread = nullptr; SocketFactory* socket_factory = nullptr; + + // Provides common widely used dependencies for webrtc subcomponents. + // `task_queue_factory` and `field_trials` members below override values in + // `env` when set. + std::optional env; + // The `packet_socket_factory` will only be used if CreatePeerConnection is // called without a `port_allocator`. std::unique_ptr packet_socket_factory; + // Deprecated. Instead provide custom task queue factory through the `env`. + // TODO: bugs.webrtc.org/42220378 - Mark [[deprecated]] once unused in WebRTC + // and chromium std::unique_ptr task_queue_factory; std::unique_ptr event_log_factory; std::unique_ptr fec_controller_factory; @@ -1448,6 +1458,9 @@ struct RTC_EXPORT PeerConnectionFactoryDependencies final { std::unique_ptr network_monitor_factory; std::unique_ptr neteq_factory; std::unique_ptr sctp_factory; + // Deprecated. Instead provide custom field trials through the `env`. + // TODO: bugs.webrtc.org/42220378 - Mark [[deprecated]] once unused in WebRTC + // and chromium std::unique_ptr trials; std::unique_ptr transport_controller_send_factory; diff --git a/pc/peer_connection_factory.cc b/pc/peer_connection_factory.cc index e62d25af0b..d23aec1363 100644 --- a/pc/peer_connection_factory.cc +++ b/pc/peer_connection_factory.cc @@ -13,6 +13,7 @@ #include #include #include +#include #include #include #include @@ -67,6 +68,25 @@ #include "rtc_base/system/file_wrapper.h" namespace webrtc { +namespace { + +Environment AssembleEnvironment(PeerConnectionFactoryDependencies& deps) { + // Assemble Environment here rather than in ConnectionContext::Create + // to avoid dependency on EnvironmentFactory by ConnectionContext and thus its + // users. + EnvironmentFactory env_factory = deps.env.has_value() + ? EnvironmentFactory(*deps.env) + : EnvironmentFactory(); + env_factory.Set(std::move(deps.trials)); + env_factory.Set(std::move(deps.task_queue_factory)); + + // Clear Environment from `deps` to avoid accidental usage of the wrong + // Environment. + deps.env = std::nullopt; + return env_factory.Create(); +} + +} // namespace scoped_refptr CreateModularPeerConnectionFactory( @@ -93,10 +113,8 @@ CreateModularPeerConnectionFactory( // Static scoped_refptr PeerConnectionFactory::Create( PeerConnectionFactoryDependencies dependencies) { - auto context = ConnectionContext::Create( - CreateEnvironment(std::move(dependencies.trials), - std::move(dependencies.task_queue_factory)), - &dependencies); + auto context = ConnectionContext::Create(AssembleEnvironment(dependencies), + &dependencies); if (!context) { return nullptr; } @@ -128,10 +146,8 @@ PeerConnectionFactory::PeerConnectionFactory( PeerConnectionFactory::PeerConnectionFactory( PeerConnectionFactoryDependencies dependencies) : PeerConnectionFactory( - ConnectionContext::Create( - CreateEnvironment(std::move(dependencies.trials), - std::move(dependencies.task_queue_factory)), - &dependencies), + ConnectionContext::Create(AssembleEnvironment(dependencies), + &dependencies), &dependencies) {} PeerConnectionFactory::~PeerConnectionFactory() { diff --git a/pc/peer_connection_factory_unittest.cc b/pc/peer_connection_factory_unittest.cc index f895496257..aeb79875f3 100644 --- a/pc/peer_connection_factory_unittest.cc +++ b/pc/peer_connection_factory_unittest.cc @@ -25,6 +25,8 @@ #include "api/enable_media.h" #include "api/enable_media_with_defaults.h" #include "api/environment/environment_factory.h" +#include "api/field_trials.h" +#include "api/field_trials_view.h" #include "api/jsep.h" #include "api/make_ref_counted.h" #include "api/media_stream_interface.h" @@ -32,7 +34,6 @@ #include "api/peer_connection_interface.h" #include "api/rtp_parameters.h" #include "api/scoped_refptr.h" -#include "api/task_queue/default_task_queue_factory.h" #include "api/test/mock_packet_socket_factory.h" #include "api/units/time_delta.h" #include "api/video_codecs/scalability_mode.h" @@ -80,6 +81,7 @@ using ::testing::A; using ::testing::AtLeast; using ::testing::InvokeWithoutArgs; using ::testing::NiceMock; +using ::testing::NotNull; using ::testing::Return; using ::testing::UnorderedElementsAre; using ::webrtc::test::MockAudioProcessing; @@ -275,7 +277,6 @@ CreatePeerConnectionFactoryWithRtxDisabled() { pcf_dependencies.signaling_thread = Thread::Current(); pcf_dependencies.worker_thread = Thread::Current(); pcf_dependencies.network_thread = Thread::Current(); - pcf_dependencies.task_queue_factory = CreateDefaultTaskQueueFactory(); pcf_dependencies.adm = FakeAudioCaptureModule::Create(); pcf_dependencies.audio_encoder_factory = CreateBuiltinAudioEncoderFactory(); @@ -662,6 +663,43 @@ TEST_F(PeerConnectionFactoryTest, LocalRendering) { EXPECT_FALSE(local_renderer.black_frame()); } +TEST(PeerConnectionFactoryDependenciesTest, + CanInjectFieldTrialsWithEnvironment) { + std::unique_ptr field_trials = + FieldTrials::CreateNoGlobal(""); + ASSERT_THAT(field_trials, NotNull()); + FieldTrialsView* raw_field_trials = field_trials.get(); + + PeerConnectionFactoryDependencies pcf_dependencies; + pcf_dependencies.env = CreateEnvironment(std::move(field_trials)); + pcf_dependencies.adm = FakeAudioCaptureModule::Create(); + EnableMediaWithDefaults(pcf_dependencies); + + scoped_refptr pcf = + PeerConnectionFactory::Create(std::move(pcf_dependencies)); + EXPECT_EQ(&pcf->field_trials(), raw_field_trials); +} + +TEST(PeerConnectionFactoryDependenciesTest, + PreferFieldTrialsInjectedExplicetly) { + std::unique_ptr env_field_trials = + FieldTrials::CreateNoGlobal(""); + std::unique_ptr explicit_field_trials = + FieldTrials::CreateNoGlobal(""); + ASSERT_FALSE(env_field_trials.get() == explicit_field_trials.get()); + FieldTrialsView* raw_explicit_field_trials = explicit_field_trials.get(); + + PeerConnectionFactoryDependencies pcf_dependencies; + pcf_dependencies.env = CreateEnvironment(std::move(env_field_trials)); + pcf_dependencies.trials = std::move(explicit_field_trials); + pcf_dependencies.adm = FakeAudioCaptureModule::Create(); + EnableMediaWithDefaults(pcf_dependencies); + + scoped_refptr pcf = + PeerConnectionFactory::Create(std::move(pcf_dependencies)); + EXPECT_EQ(&pcf->field_trials(), raw_explicit_field_trials); +} + TEST(PeerConnectionFactoryDependenciesTest, UsesNetworkManager) { constexpr TimeDelta kWaitTimeout = TimeDelta::Seconds(10); auto mock_network_manager = std::make_unique>(); diff --git a/sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h b/sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h index 08734083cf..b11aadb586 100644 --- a/sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h +++ b/sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h @@ -34,7 +34,7 @@ NS_ASSUME_NONNULL_BEGIN /* Initialize object with provided dependencies and with media support. */ - (instancetype)initWithMediaAndDependencies: - (webrtc::PeerConnectionFactoryDependencies)dependencies; + (webrtc::PeerConnectionFactoryDependencies &)dependencies; /* Initialize object with injectable native audio/video encoder/decoder * factories */ diff --git a/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm b/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm index 79afe271da..81eafe68df 100644 --- a/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm +++ b/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm @@ -81,7 +81,7 @@ - (instancetype)init { dependencies.video_decoder_factory = webrtc::ObjCToNativeVideoDecoderFactory( [[RTC_OBJC_TYPE(RTCVideoDecoderFactoryH264) alloc] init]); dependencies.adm = [self audioDeviceModule]; - return [self initWithMediaAndDependencies:std::move(dependencies)]; + return [self initWithMediaAndDependencies:dependencies]; } - (instancetype) @@ -122,12 +122,12 @@ - (instancetype)init { } else { dependencies.adm = [self audioDeviceModule]; } - return [self initWithMediaAndDependencies:std::move(dependencies)]; + return [self initWithMediaAndDependencies:dependencies]; #endif } - (instancetype)initWithNativeDependencies: - (webrtc::PeerConnectionFactoryDependencies)dependencies { + (webrtc::PeerConnectionFactoryDependencies &)dependencies { self = [super init]; if (self) { _networkThread = webrtc::Thread::CreateWithSocketServer(); @@ -167,8 +167,8 @@ - (instancetype)initWithNativeDependencies: } - (instancetype)initWithNoMedia { - return [self - initWithNativeDependencies:webrtc::PeerConnectionFactoryDependencies()]; + webrtc::PeerConnectionFactoryDependencies default_deps; + return [self initWithNativeDependencies:default_deps]; } - (instancetype) @@ -198,7 +198,7 @@ - (instancetype)initWithNoMedia { dependencies.audio_processing_builder = CustomAudioProcessing(std::move(audioProcessingModule)); } - return [self initWithMediaAndDependencies:std::move(dependencies)]; + return [self initWithMediaAndDependencies:dependencies]; } - (instancetype) @@ -232,11 +232,11 @@ - (instancetype)initWithNoMedia { CustomAudioProcessing(std::move(audioProcessingModule)); } dependencies.network_controller_factory = std::move(networkControllerFactory); - return [self initWithMediaAndDependencies:std::move(dependencies)]; + return [self initWithMediaAndDependencies:dependencies]; } - (instancetype)initWithMediaAndDependencies: - (webrtc::PeerConnectionFactoryDependencies)dependencies { + (webrtc::PeerConnectionFactoryDependencies &)dependencies { #pragma clang diagnostic push #pragma clang diagnostic ignored "-Wdeprecated-declarations" // audio_processing_builder should be used instead in new code. @@ -254,7 +254,7 @@ - (instancetype)initWithMediaAndDependencies: std::make_unique(); } webrtc::EnableMedia(dependencies); - return [self initWithNativeDependencies:std::move(dependencies)]; + return [self initWithNativeDependencies:dependencies]; } - (RTC_OBJC_TYPE(RTCRtpCapabilities) *)rtpSenderCapabilitiesForKind: diff --git a/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm b/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm index 8fe5565111..3485dc8d1e 100644 --- a/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm +++ b/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm @@ -28,7 +28,7 @@ + (RTCPeerConnectionFactoryBuilder *)builder { - (RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)createPeerConnectionFactory { return [[RTC_OBJC_TYPE(RTCPeerConnectionFactory) alloc] - initWithMediaAndDependencies:std::move(_dependencies)]; + initWithMediaAndDependencies:_dependencies]; } - (void)setFieldTrials:(std::unique_ptr)fieldTrials { diff --git a/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm b/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm index 95d8b058f9..929490b0ba 100644 --- a/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm +++ b/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm @@ -40,9 +40,9 @@ - (void)testBuilder { id factoryMock = OCMStrictClassMock([RTC_OBJC_TYPE(RTCPeerConnectionFactory) class]); OCMExpect([factoryMock alloc]).andReturn(factoryMock); + webrtc::PeerConnectionFactoryDependencies default_deps; RTC_UNUSED([[[[factoryMock expect] andReturn:factoryMock] - ignoringNonObjectArgs] initWithMediaAndDependencies: - webrtc::PeerConnectionFactoryDependencies()]); + ignoringNonObjectArgs] initWithMediaAndDependencies:default_deps]); RTCPeerConnectionFactoryBuilder* builder = [[RTCPeerConnectionFactoryBuilder alloc] init]; RTC_OBJC_TYPE(RTCPeerConnectionFactory)* peerConnectionFactory = @@ -55,9 +55,9 @@ - (void)testDefaultComponentsBuilder { id factoryMock = OCMStrictClassMock([RTC_OBJC_TYPE(RTCPeerConnectionFactory) class]); OCMExpect([factoryMock alloc]).andReturn(factoryMock); + webrtc::PeerConnectionFactoryDependencies default_deps; RTC_UNUSED([[[[factoryMock expect] andReturn:factoryMock] - ignoringNonObjectArgs] initWithMediaAndDependencies: - webrtc::PeerConnectionFactoryDependencies()]); + ignoringNonObjectArgs] initWithMediaAndDependencies:default_deps]); RTCPeerConnectionFactoryBuilder* builder = [RTCPeerConnectionFactoryBuilder defaultBuilder]; RTC_OBJC_TYPE(RTCPeerConnectionFactory)* peerConnectionFactory = diff --git a/tools_webrtc/iwyu/iwyu-verifier-filter_list.json b/tools_webrtc/iwyu/iwyu-verifier-filter_list.json index 9cd95be5d8..364d2c895a 100644 --- a/tools_webrtc/iwyu/iwyu-verifier-filter_list.json +++ b/tools_webrtc/iwyu/iwyu-verifier-filter_list.json @@ -9,6 +9,7 @@ "modules/desktop_capture/win/", "rtc_base/win", "sdk/android/src/jni/", + "sdk/objc/", "test/ios/", "test/mac/" ] From ff2fddb198d11751c4697b51fc102a650000d345 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Mon, 28 Apr 2025 09:03:01 -0700 Subject: [PATCH 002/292] Roll chromium_revision 92c0179e11..ae6ec80ac0 (1452499:1452664) Change log: https://chromium.googlesource.com/chromium/src/+log/92c0179e11..ae6ec80ac0 Full diff: https://chromium.googlesource.com/chromium/src/+/92c0179e11..ae6ec80ac0 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/86c814633c..32f65d20af * src/build: https://chromium.googlesource.com/chromium/src/build/+log/88030b3203..0345a25a86 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/058aa981a6..8d3354e85c * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/a89c37d36b..42f6eefe76 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/8062e0e102..7e53f288d9 * src/third_party/android_deps/autorolled/cipd: yZGuDQ9pDcy816oWVPUC2zMwLtFUfGt4W2PmIYEWTcAC.._rzZP_RJ0Fx7bXb_R5WFX8yvYsF5YffSZ0p9pbI_Z64C * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/fa8fc854e1..1fcc527019 * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/957c9f9522..97de5579d0 * src/third_party/llvm-build/Release+Asserts_version: Linux_x64/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clang-win-runtime-library-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/llvmobjdump-llvmorg-21-init-6681-g5b36835d-1.tar.xz..Linux_x64/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clang-win-runtime-library-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/llvmobjdump-llvmorg-21-init-9266-g09006611-1.tar.xz * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/a54dd38d60..220d71b5a8 * src/third_party/rust-toolchain_version: Linux_x64/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Mac/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Mac_arm64/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Win/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz..Linux_x64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Mac/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Mac_arm64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Win/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/ffcbc837bb..48eea1f5a0 DEPS diff: https://chromium.googlesource.com/chromium/src/+/92c0179e11..ae6ec80ac0/DEPS No update to Clang. BUG=None Change-Id: I6a7392e04206f17d3f7462f6098994be471859bf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/388900 Commit-Queue: Autoroller Bot-Commit: Autoroller Cr-Commit-Position: refs/heads/main@{#44474} --- DEPS | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/DEPS b/DEPS index 61027df618..fb05d990fd 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '92c0179e11451c30349504abb941d3615938ca29', + 'chromium_revision': 'ae6ec80ac0f466a1f5d1960fa19aca853e944139', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@86c814633cf284bc8057a539bc722e2a672afe2f', + 'https://chromium.googlesource.com/chromium/src/base@32f65d20af167d22990abb0aa28dbe4f3e65aa94', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@88030b320338e0706b6b93336c4b35e6bbaf467e', + 'https://chromium.googlesource.com/chromium/src/build@0345a25a8624b38c14f600f86a73d9a44a9cd772', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@0f32cb9025766951122d4ed19aba87a94ded3f43', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@058aa981a69171da048bdcf82d89c64fcd43d16b', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@8d3354e85c432a482a21167eb5de0872e46b32f7', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@a89c37d36bf80c05963727e28b9916835ae88d3a', + 'https://chromium.googlesource.com/chromium/src/testing@42f6eefe763bd5bf601f8caf87f98c9b0bdce10f', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@8062e0e102496ff14a8c58b586f014527424953d', + 'https://chromium.googlesource.com/chromium/src/third_party@7e53f288d97da8a861506cacb8ed30444459e88c', 'src/buildtools/linux64': { 'packages': [ @@ -497,13 +497,13 @@ deps = { 'src/third_party/crc32c/src': 'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558', 'src/third_party/depot_tools': - 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@fa8fc854e1766b86f10c9a15902cf3cc23adaac2', + 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@1fcc527019d786502b02f71b8b764ee674a40953', 'src/third_party/ffmpeg': 'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@01f23648c6b84de6c0f717fa4e1816f53b9ee72e', 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@957c9f95224b1e1318c0ecb98d0e7584ea5ccff2', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@97de5579d06e7ded32eee4a622501dd01145a064', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'a54dd38d60593129ae56d400f1a72860670abea4', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '220d71b5a8e54a558cb10e70518c8b22b2c57b46', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@ffcbc837bbb14d80d09147c2af5302ff6bd4bd69', + 'https://chromium.googlesource.com/chromium/src/tools@48eea1f5a02d31e4712c13cc349d953314180d77', 'src/third_party/espresso': { 'packages': [ @@ -827,7 +827,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_deps/autorolled', - 'version': 'yZGuDQ9pDcy816oWVPUC2zMwLtFUfGt4W2PmIYEWTcAC', + 'version': '_rzZP_RJ0Fx7bXb_R5WFX8yvYsF5YffSZ0p9pbI_Z64C', }, ], 'condition': 'checkout_android and non_git_source', From aaf8f8b89241508585ba4fed256e77fafb465844 Mon Sep 17 00:00:00 2001 From: Sunggook Chue Date: Thu, 24 Apr 2025 15:50:38 -0700 Subject: [PATCH 003/292] Utilize WGC improvement on 24H2 version. The Windows 11 WGC (Windows Graphics Capture) API has been improved starting from version 24H2 (build 26100). This enhancement allows the API to avoid capturing content when there are no changes to the capture target, a capability that was previously unavailable. The CL addresses unnecessary calling OS API for failed capture frame and skips frame comparison for change detection. Bug: chromium:347991512 Change-Id: I2fc03238cb698ad2d21fcad8660e4e16229cc4d2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/388220 Reviewed-by: Ilya Nikolaevskiy Reviewed-by: Alexander Cooper Reviewed-by: Harald Alvestrand Commit-Queue: Alexander Cooper Cr-Commit-Position: refs/heads/main@{#44475} --- modules/desktop_capture/win/wgc_capture_session.cc | 13 ++++++++++++- rtc_base/win/windows_version.cc | 9 ++++++++- rtc_base/win/windows_version.h | 3 +++ 3 files changed, 23 insertions(+), 2 deletions(-) diff --git a/modules/desktop_capture/win/wgc_capture_session.cc b/modules/desktop_capture/win/wgc_capture_session.cc index b4f9425699..100f28a0df 100644 --- a/modules/desktop_capture/win/wgc_capture_session.cc +++ b/modules/desktop_capture/win/wgc_capture_session.cc @@ -29,6 +29,7 @@ #include "rtc_base/time_utils.h" #include "rtc_base/win/create_direct3d_device.h" #include "rtc_base/win/get_activation_factory.h" +#include "rtc_base/win/windows_version.h" #include "system_wrappers/include/metrics.h" #include "system_wrappers/include/sleep.h" @@ -98,6 +99,11 @@ bool SizeHasChanged(ABI::Windows::Graphics::SizeInt32 size_new, size_new.Width != size_old.Width); } +bool DoesWgcSkipStaticFrames() { + return (webrtc::rtc_win::GetVersion() >= + webrtc::rtc_win::Version::VERSION_WIN11_24H2); +} + } // namespace WgcCaptureSession::WgcCaptureSession(intptr_t source_id, @@ -473,6 +479,7 @@ HRESULT WgcCaptureSession::ProcessFrame() { } else { if (!GetHmonitorFromDeviceIndex(source_id_, &monitor)) { RTC_LOG(LS_ERROR) << "Failed to get HMONITOR from device index."; + d3d_context->Unmap(mapped_texture_.Get(), 0); return E_FAIL; } } @@ -496,7 +503,11 @@ HRESULT WgcCaptureSession::ProcessFrame() { // previous. The idea is to get a low-complexity indication of if the content // is static or not without performing a full/deep memory comparison when // updating the damaged region. - bool frame_content_has_changed = false; + // `DoesWgcSkipStaticFrames()`: `TryGetNextFrame()` returns a frame + // successfully only if there is a region that has changed. This means that + // we can skip the full memory comparison if the running OS is Windows 11 + // 24H2 or later. + bool frame_content_has_changed = DoesWgcSkipStaticFrames(); // Check if the queue contains two frames whose content can be compared. const bool frame_content_can_be_compared = FrameContentCanBeCompared(); diff --git a/rtc_base/win/windows_version.cc b/rtc_base/win/windows_version.cc index 871be7fc37..6ea62aca36 100644 --- a/rtc_base/win/windows_version.cc +++ b/rtc_base/win/windows_version.cc @@ -221,10 +221,17 @@ Version MajorMinorBuildToVersion(int major, int minor, int build) { return VERSION_WIN10_21H2; } else if (build < 22000) { return VERSION_SERVER_2022; - } else { + } else if (build < 22621) { return VERSION_WIN11; + } else if (build < 22631) { + return VERSION_WIN11_22H2; + } else if (build < 26100) { + return VERSION_WIN11_23H2; + } else { + return VERSION_WIN11_24H2; } } else if (major == 11) { + RTC_DCHECK_NOTREACHED(); return VERSION_WIN11; } else if (major > 6) { RTC_DCHECK_NOTREACHED(); diff --git a/rtc_base/win/windows_version.h b/rtc_base/win/windows_version.h index cbe7cfd88b..cb75ae220c 100644 --- a/rtc_base/win/windows_version.h +++ b/rtc_base/win/windows_version.h @@ -50,6 +50,9 @@ enum Version { VERSION_WIN10_21H2 = 19, // 21H2: Build 19044. VERSION_SERVER_2022 = 20, // Server 2022: Build 20348. VERSION_WIN11 = 21, // Windows 11: Build 22000. + VERSION_WIN11_22H2 = 22, // Windows 11 22H2: Build 22621. + VERSION_WIN11_23H2 = 23, // Windows 11 23H2: Build 22631. + VERSION_WIN11_24H2 = 24, // Windows 11 24H2: Build 26100. VERSION_WIN_LAST, // Indicates error condition. }; From e3104f929bc3402b3e1320c8bab547ba6a7dae6e Mon Sep 17 00:00:00 2001 From: webrtc-version-updater Date: Mon, 28 Apr 2025 21:09:16 -0700 Subject: [PATCH 004/292] Update WebRTC code version (2025-04-29T04:08:55). Bug: None Change-Id: I1010e301c8e4c30ed8d84ecdfbc6be5376e9dc7a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/388864 Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Cr-Commit-Position: refs/heads/main@{#44476} --- call/version.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/call/version.cc b/call/version.cc index 7684820d5d..8a5bed828e 100644 --- a/call/version.cc +++ b/call/version.cc @@ -13,7 +13,7 @@ namespace webrtc { // The timestamp is always in UTC. -const char* const kSourceTimestamp = "WebRTC source stamp 2025-04-28T04:05:58"; +const char* const kSourceTimestamp = "WebRTC source stamp 2025-04-29T04:08:55"; void LoadWebRTCVersionInRegister() { // Using volatile to instruct the compiler to not optimize `p` away even From 84e40071f96f198867d4dfba827d23d9a15eb5a0 Mon Sep 17 00:00:00 2001 From: Tom Van Goethem Date: Mon, 28 Apr 2025 22:14:11 +0000 Subject: [PATCH 005/292] Restrict certain addresses to be used when SDP munging is enabled Gated behind the default-empty list of restricted addresses tracked by the field trial WebRTC-NoSdpMangleUfragRestrictedAddresses. Bug: b/409713509 Change-Id: Iabb5c8b5307c66215635336bee55f7bfbd040f3b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/388340 Auto-Submit: Tom Van Goethem Reviewed-by: Harald Alvestrand Commit-Queue: Tomas Gunnarsson Reviewed-by: Tomas Gunnarsson Cr-Commit-Position: refs/heads/main@{#44477} --- experiments/field_trials.py | 3 ++ pc/DEPS | 2 +- pc/sdp_munging_detector.cc | 34 ++++++++++++++ pc/sdp_munging_detector.h | 4 ++ pc/sdp_offer_answer.cc | 58 ++++++++++++++++++++---- pc/sdp_offer_answer.h | 4 ++ pc/sdp_offer_answer_unittest.cc | 78 ++++++++++++++++++++++++++++++++- 7 files changed, 172 insertions(+), 11 deletions(-) diff --git a/experiments/field_trials.py b/experiments/field_trials.py index 1c9f8c029e..b23b72457b 100755 --- a/experiments/field_trials.py +++ b/experiments/field_trials.py @@ -119,6 +119,9 @@ def bug_url(self) -> str: FieldTrial('WebRTC-NoSdpMangleUfrag', 375571816, date(2025, 10, 11)), + FieldTrial('WebRTC-NoSdpMangleUfragRestrictedAddresses', + 409713509, + date(2025, 10, 11)), FieldTrial('WebRTC-Pacer-FastRetransmissions', 40235589, date(2024, 4, 1)), diff --git a/pc/DEPS b/pc/DEPS index dd44ac1f77..53ef6e45ee 100644 --- a/pc/DEPS +++ b/pc/DEPS @@ -14,7 +14,7 @@ include_rules = [ "+net/dcsctp", "+p2p", "+system_wrappers", - "+absl/strings/str_cat.h", + "+absl/strings", ] specific_include_rules = { diff --git a/pc/sdp_munging_detector.cc b/pc/sdp_munging_detector.cc index baef265387..2497a3066d 100644 --- a/pc/sdp_munging_detector.cc +++ b/pc/sdp_munging_detector.cc @@ -10,6 +10,7 @@ #include "pc/sdp_munging_detector.h" +#include #include #include @@ -501,4 +502,37 @@ SdpMungingType DetermineSdpMungingType( return SdpMungingType::kUnknownModification; } +// Similar to DetermineSdpMungingType, but only checks whether the ICE ufrag or +// pwd of the SDP has been modified between createOffer and setLocalDescription. +bool HasUfragSdpMunging(const SessionDescriptionInterface* sdesc, + const SessionDescriptionInterface* last_created_desc) { + if (!sdesc || !sdesc->description()) { + RTC_LOG(LS_WARNING) << "SDP munging: Failed to parse session description."; + return false; + } + + if (!last_created_desc || !last_created_desc->description()) { + RTC_LOG(LS_WARNING) << "SDP munging: SetLocalDescription called without " + "CreateOffer or CreateAnswer."; + return false; + } + TransportInfos last_created_transport_infos = + last_created_desc->description()->transport_infos(); + TransportInfos transport_infos_to_set = + sdesc->description()->transport_infos(); + for (size_t i = 0; i < std::min(last_created_transport_infos.size(), + transport_infos_to_set.size()); + i++) { + if (last_created_transport_infos[i].description.ice_ufrag != + transport_infos_to_set[i].description.ice_ufrag) { + return true; + } + if (last_created_transport_infos[i].description.ice_pwd != + transport_infos_to_set[i].description.ice_pwd) { + return true; + } + } + return false; +} + } // namespace webrtc diff --git a/pc/sdp_munging_detector.h b/pc/sdp_munging_detector.h index 9b630a30dc..ec8af1a634 100644 --- a/pc/sdp_munging_detector.h +++ b/pc/sdp_munging_detector.h @@ -20,6 +20,10 @@ SdpMungingType DetermineSdpMungingType( const SessionDescriptionInterface* sdesc, const SessionDescriptionInterface* last_created_desc); +// Determines if the ICE ufrag or pwd of the SDP were modified. +bool HasUfragSdpMunging(const SessionDescriptionInterface* sdesc, + const SessionDescriptionInterface* last_created_desc); + } // namespace webrtc #endif // PC_SDP_MUNGING_DETECTOR_H_ diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc index fc7808d4d0..31808c506d 100644 --- a/pc/sdp_offer_answer.cc +++ b/pc/sdp_offer_answer.cc @@ -28,6 +28,7 @@ #include "absl/memory/memory.h" #include "absl/strings/match.h" #include "absl/strings/str_cat.h" +#include "absl/strings/str_split.h" #include "absl/strings/string_view.h" #include "api/array_view.h" #include "api/candidate.h" @@ -903,6 +904,21 @@ bool ContentHasHeaderExtension(const ContentInfo& content_info, return false; } +// Matches the given host with the wildcard host. A wildcard `*` matches the +// current character and anything that comes after it. For instance, 127.0.* +// matches any string that starts with `127.0.`. For IPv6 addresses, the +// wildcard host should also start with `[` and end with `]` or a wildcard. +bool WildcardHostPrefixMatch(absl::string_view host, + absl::string_view wildcard_host) { + for (size_t i = 0; i < std::min(host.size(), wildcard_host.size()); ++i) { + if (host[i] == wildcard_host[i]) { + continue; + } + return wildcard_host[i] == '*'; + } + return host.size() == wildcard_host.size(); +} + } // namespace void UpdateRtpHeaderExtensionPreferencesFromSdpMunging( @@ -2448,19 +2464,22 @@ void SdpOfferAnswerHandler::DoSetLocalDescription( return; } + SessionDescriptionInterface* last_created_desc = + desc->GetType() == SdpType::kOffer ? last_created_offer_.get() + : last_created_answer_.get(); + // Determine if SDP munging was done. This is not yet acted upon. bool had_local_description = !!local_description(); SdpMungingType sdp_munging_type = - DetermineSdpMungingType(desc.get(), desc->GetType() == SdpType::kOffer - ? last_created_offer_.get() - : last_created_answer_.get()); + DetermineSdpMungingType(desc.get(), last_created_desc); if (!disable_sdp_munging_checks_ && - pc_->trials().IsEnabled("WebRTC-NoSdpMangleUfrag")) { - if (sdp_munging_type == kIceUfrag || sdp_munging_type == kIcePwd) { + HasUfragSdpMunging(desc.get(), last_created_desc)) { + has_sdp_munged_ufrag_ = true; + if (pc_->trials().IsEnabled("WebRTC-NoSdpMangleUfrag")) { RTC_LOG(LS_ERROR) << "Rejecting SDP because of ufrag modification"; observer->OnSetLocalDescriptionComplete( - RTCError(RTCErrorType::INVALID_PARAMETER, + RTCError(RTCErrorType::INVALID_MODIFICATION, "SDP is modified in a non-acceptable way")); last_sdp_munging_type_ = sdp_munging_type; ReportInitialSdpMunging(had_local_description, desc->GetType()); @@ -2471,9 +2490,6 @@ void SdpOfferAnswerHandler::DoSetLocalDescription( // Workaround for isses.webrtc.org/412904801 - detect if packetization:raw // was added in SDP munging, and if so, push it back into the codec vendor. std::vector> codecs_mangled_to_raw; - auto last_created_desc = desc->GetType() == SdpType::kOffer - ? last_created_offer_.get() - : last_created_answer_.get(); // Ignore if other mungings have been applied. if (last_created_desc && last_created_desc->description()->contents().size() == @@ -5300,6 +5316,30 @@ bool SdpOfferAnswerHandler::ReadyToUseRemoteCandidate( return false; } + if (has_sdp_munged_ufrag_) { + // The format is + // restricted_addresses::|: + // The host can contain wildcards that match anything that comes after it. + // The port should either be a wildcard or a specific port. + const std::string restricted_addresses = + pc_->trials().Lookup("WebRTC-NoSdpMangleUfragRestrictedAddresses"); + const std::string port = candidate->candidate().address().PortAsString(); + const std::string host = candidate->candidate().address().HostAsURIString(); + const std::vector restricted_address_list = + absl::StrSplit(restricted_addresses, '|'); + for (const absl::string_view restricted_address : restricted_address_list) { + const std::pair address = + absl::StrSplit(restricted_address, ':'); + if ((address.second == port || address.second == "*") && + WildcardHostPrefixMatch(host, address.first)) { + RTC_LOG(LS_ERROR) << "ReadyToUseRemoteCandidate: Candidate not valid " + "because of SDP munging."; + *valid = false; + return false; + } + } + } + return true; } diff --git a/pc/sdp_offer_answer.h b/pc/sdp_offer_answer.h index 6bf16f681d..e5604bf37d 100644 --- a/pc/sdp_offer_answer.h +++ b/pc/sdp_offer_answer.h @@ -690,6 +690,10 @@ class SdpOfferAnswerHandler : public SdpStateProvider { // to disable. bool disable_sdp_munging_checks_ = false; CodecLookupHelper* codec_lookup_helper_ = nullptr; + + // Whether the username fragment or the password of the SDP was munged. + bool has_sdp_munged_ufrag_ = false; + WeakPtrFactory weak_ptr_factory_ RTC_GUARDED_BY(signaling_thread()); }; diff --git a/pc/sdp_offer_answer_unittest.cc b/pc/sdp_offer_answer_unittest.cc index a48df74fec..16c3c22de3 100644 --- a/pc/sdp_offer_answer_unittest.cc +++ b/pc/sdp_offer_answer_unittest.cc @@ -52,11 +52,11 @@ #include "media/base/stream_params.h" #include "p2p/base/transport_description.h" #include "pc/peer_connection_wrapper.h" -#include "pc/session_description.h" #include "pc/test/fake_audio_capture_module.h" #include "pc/test/fake_rtc_certificate_generator.h" #include "pc/test/integration_test_helpers.h" #include "pc/test/mock_peer_connection_observers.h" +#include "rtc_base/strings/string_format.h" #include "rtc_base/thread.h" #include "system_wrappers/include/metrics.h" #include "test/gmock.h" @@ -1951,6 +1951,82 @@ TEST_F(SdpOfferAnswerMungingTest, IcePwd) { ElementsAre(Pair(SdpMungingType::kIcePwd, 1))); } +TEST_F(SdpOfferAnswerMungingTest, IceUfragRestrictedAddresses) { + RTCConfiguration config; + config.certificates.push_back( + FakeRTCCertificateGenerator::GenerateCertificate()); + auto caller = CreatePeerConnection( + config, + FieldTrials::CreateNoGlobal("WebRTC-NoSdpMangleUfragRestrictedAddresses/" + "127.0.0.1:12345|127.0.0.*:23456|*:34567/")); + auto callee = CreatePeerConnection(); + caller->AddAudioTrack("audio_track", {}); + auto offer = caller->CreateOffer(); + auto& transport_infos = offer->description()->transport_infos(); + ASSERT_EQ(transport_infos.size(), 1u); + transport_infos[0].description.ice_ufrag = "amungediceufrag"; + + EXPECT_TRUE(caller->SetLocalDescription(offer->Clone())); + EXPECT_TRUE(callee->SetRemoteDescription(std::move(offer))); + + auto answer = callee->CreateAnswer(); + EXPECT_TRUE(callee->SetLocalDescription(answer->Clone())); + EXPECT_TRUE(caller->SetRemoteDescription(std::move(answer))); + + static constexpr const char tmpl[] = + "candidate:a0+B/1 1 udp 2130706432 %s typ host"; + + // Addresses to test. First field is the address in string format, + // second field is the expected outcome (success or failure). + const std::vector> address_tests = { + {"127.0.0.1:12345", false}, {"127.0.0.2:23456", false}, + {"8.8.8.8:34567", false}, {"127.0.0.2:12345", true}, + {"127.0.1.1:23456", true}, {"8.8.8.8:3456", true}, + }; + + for (const auto& address_test : address_tests) { + std::optional result; + const std::string candidate = StringFormat( + tmpl, absl::StrReplaceAll(address_test.first, {{":", " "}}).c_str()); + caller->pc()->AddIceCandidate( + std::unique_ptr( + CreateIceCandidate("", 0, candidate, nullptr)), + [&result](RTCError error) { result = error; }); + + ASSERT_THAT( + WaitUntil([&] { return result.has_value(); }, ::testing::IsTrue()), + IsRtcOk()); + if (address_test.second == true) { + EXPECT_TRUE(result.value().ok()); + } else { + EXPECT_FALSE(result.value().ok()); + EXPECT_EQ(result.value().type(), RTCErrorType::UNSUPPORTED_OPERATION); + } + } +} + +TEST_F(SdpOfferAnswerMungingTest, IceUfragSdpRejectedAndRestrictedAddresses) { + RTCConfiguration config; + config.certificates.push_back( + FakeRTCCertificateGenerator::GenerateCertificate()); + auto caller = CreatePeerConnection( + config, + FieldTrials::CreateNoGlobal("WebRTC-NoSdpMangleUfragRestrictedAddresses/" + "127.0.0.1:12345|127.0.0.*:23456|*:34567/" + "WebRTC-NoSdpMangleUfrag/Enabled/")); + auto callee = CreatePeerConnection(); + caller->AddAudioTrack("audio_track", {}); + auto offer = caller->CreateOffer(); + auto& transport_infos = offer->description()->transport_infos(); + ASSERT_EQ(transport_infos.size(), 1u); + transport_infos[0].description.ice_ufrag = "amungediceufrag"; + + EXPECT_FALSE(caller->SetLocalDescription(offer->Clone())); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kIceUfrag, 1))); +} + TEST_F(SdpOfferAnswerMungingTest, IceMode) { auto pc = CreatePeerConnection(); pc->AddAudioTrack("audio_track", {}); From ce6aa1f7a00289aa0e46eaa5bca53bf634d203e3 Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Mon, 28 Apr 2025 11:59:22 -0700 Subject: [PATCH 006/292] Add PacketOptions to SendRtcp which will allow sending RTCP with ECN flags. BUG=webrtc:42225697 Change-Id: Ie32a0e4fc0ce34aaa0f960f85258c1365ba706bc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/387120 Reviewed-by: Danil Chapovalov Commit-Queue: Philipp Hancke Reviewed-by: Per Kjellander Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44478} --- api/call/transport.h | 17 ++++- audio/BUILD.gn | 12 ++++ audio/channel_receive.h | 2 +- audio/channel_receive_unittest.cc | 39 ++++++++---- audio/voip/test/BUILD.gn | 8 +++ audio/voip/test/audio_channel_unittest.cc | 25 ++++++-- call/fake_network_pipe.cc | 3 +- .../androidvoip/jni/android_voip_client.cc | 3 +- .../androidvoip/jni/android_voip_client.h | 3 +- media/base/media_channel_impl.cc | 62 +++++++++---------- media/base/media_channel_impl.h | 5 +- media/engine/webrtc_video_engine_unittest.cc | 3 +- media/engine/webrtc_voice_engine_unittest.cc | 5 +- modules/rtp_rtcp/source/nack_rtx_unittest.cc | 3 +- modules/rtp_rtcp/source/rtcp_sender.cc | 7 ++- .../rtp_rtcp/source/rtcp_sender_unittest.cc | 8 ++- .../source/rtp_rtcp_impl2_unittest.cc | 3 +- .../rtp_rtcp/source/rtp_rtcp_impl_unittest.cc | 3 +- .../source/rtp_sender_audio_unittest.cc | 5 +- .../source/rtp_sender_egress_unittest.cc | 5 +- .../source/rtp_sender_video_unittest.cc | 5 +- rtc_tools/BUILD.gn | 6 ++ rtc_tools/rtp_generator/rtp_generator.cc | 29 +++++++-- rtc_tools/rtp_generator/rtp_generator.h | 13 ++-- test/BUILD.gn | 6 +- test/direct_transport.cc | 3 +- test/direct_transport.h | 12 +++- test/mock_transport.h | 8 ++- test/null_transport.cc | 8 ++- test/null_transport.h | 6 +- test/rtp_rtcp_observer.h | 22 ++++--- test/scenario/network_node.cc | 3 +- test/scenario/network_node.h | 3 +- video/BUILD.gn | 4 ++ video/end_to_end_tests/bandwidth_tests.cc | 19 +++++- video/end_to_end_tests/network_state_tests.cc | 26 ++++++-- .../end_to_end_tests/retransmission_tests.cc | 37 ++++++++--- video/end_to_end_tests/rtp_rtcp_tests.cc | 32 +++++++++- video/rtp_video_stream_receiver2_unittest.cc | 4 +- video/transport_adapter.cc | 9 ++- video/transport_adapter.h | 4 +- video/video_analyzer.cc | 5 +- video/video_analyzer.h | 24 ++++++- video/video_receive_stream2_unittest.cc | 5 +- 44 files changed, 389 insertions(+), 125 deletions(-) diff --git a/api/call/transport.h b/api/call/transport.h index b2bc18ef10..a445ffe799 100644 --- a/api/call/transport.h +++ b/api/call/transport.h @@ -28,7 +28,8 @@ struct PacketOptions { // as packet_id not being set. int64_t packet_id = -1; // Whether this is an audio or video packet, excluding retransmissions. - bool is_media = true; + // Defaults to `false` which is the more common case. + bool is_media = false; bool included_in_feedback = false; bool included_in_allocation = false; bool send_as_ect1 = false; @@ -42,7 +43,19 @@ class Transport { public: virtual bool SendRtp(ArrayView packet, const PacketOptions& options) = 0; - virtual bool SendRtcp(ArrayView packet) = 0; + // TODO: bugs.webrtc.org/42225697 - Make virtual when downstream is updated. + virtual bool SendRtcp(ArrayView packet, + const PacketOptions& /*options*/) { +#pragma clang diagnostic push +#pragma clang diagnostic ignored "-Wdeprecated-declarations" + return SendRtcp(packet); +#pragma clang diagnostic pop + } + [[deprecated("Pass PacketOptions")]] + // TODO: bugs.webrtc.org/42225697 - Remove when downstream is updated. + virtual bool SendRtcp(ArrayView packet) { + return SendRtcp(packet, /*packet_options=*/{}); + } protected: virtual ~Transport() {} diff --git a/audio/BUILD.gn b/audio/BUILD.gn index 702a8d839d..0492dd24e5 100644 --- a/audio/BUILD.gn +++ b/audio/BUILD.gn @@ -260,19 +260,31 @@ if (rtc_include_tests) { sources = [ "channel_receive_unittest.cc" ] deps = [ ":audio", + "../api:array_view", + "../api:make_ref_counted", "../api:mock_frame_transformer", + "../api:scoped_refptr", + "../api:transport_api", "../api/audio:audio_device", + "../api/audio:audio_frame_api", + "../api/audio_codecs:audio_codecs_api", "../api/audio_codecs:builtin_audio_decoder_factory", "../api/crypto:frame_decryptor_interface", + "../api/crypto:options", "../api/environment:environment_factory", "../api/task_queue:default_task_queue_factory", + "../api/units:time_delta", + "../api/units:timestamp", "../logging:mocks", "../modules/audio_device:mock_audio_device", "../modules/rtp_rtcp", "../modules/rtp_rtcp:ntp_time_util", "../modules/rtp_rtcp:rtp_rtcp_format", "../rtc_base:logging", + "../rtc_base:stringutils", "../rtc_base:threading", + "../rtc_base:timeutils", + "../system_wrappers", "../test:audio_codec_mocks", "../test:mock_transport", "../test:test_support", diff --git a/audio/channel_receive.h b/audio/channel_receive.h index ee72187b7d..6179c4f8f0 100644 --- a/audio/channel_receive.h +++ b/audio/channel_receive.h @@ -27,6 +27,7 @@ #include "api/call/audio_sink.h" #include "api/call/transport.h" #include "api/crypto/crypto_options.h" +#include "api/crypto/frame_decryptor_interface.h" #include "api/environment/environment.h" #include "api/frame_transformer_interface.h" #include "api/neteq/neteq_factory.h" @@ -42,7 +43,6 @@ namespace webrtc { class AudioDeviceModule; -class FrameDecryptorInterface; class PacketRouter; class RateLimiter; class ReceiveStatistics; diff --git a/audio/channel_receive_unittest.cc b/audio/channel_receive_unittest.cc index 7cedba677a..e228b42c5e 100644 --- a/audio/channel_receive_unittest.cc +++ b/audio/channel_receive_unittest.cc @@ -10,25 +10,38 @@ #include "audio/channel_receive.h" -#include "absl/strings/escaping.h" -#include "api/audio/audio_device.h" +#include +#include +#include +#include +#include +#include + +#include "absl/strings/string_view.h" +#include "api/array_view.h" +#include "api/audio/audio_frame.h" +#include "api/audio_codecs/audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" -#include "api/crypto/frame_decryptor_interface.h" +#include "api/call/transport.h" +#include "api/crypto/crypto_options.h" #include "api/environment/environment_factory.h" +#include "api/make_ref_counted.h" +#include "api/scoped_refptr.h" #include "api/test/mock_frame_transformer.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" #include "modules/audio_device/include/mock_audio_device.h" -#include "modules/rtp_rtcp/source/byte_io.h" #include "modules/rtp_rtcp/source/ntp_time_util.h" #include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h" #include "modules/rtp_rtcp/source/rtcp_packet/report_block.h" -#include "modules/rtp_rtcp/source/rtcp_packet/sdes.h" #include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "rtc_base/logging.h" -#include "rtc_base/thread.h" +#include "rtc_base/string_encode.h" +#include "rtc_base/time_utils.h" +#include "system_wrappers/include/ntp_time.h" #include "test/gmock.h" #include "test/gtest.h" -#include "test/mock_audio_decoder_factory.h" #include "test/mock_transport.h" #include "test/time_controller/simulated_time_controller.h" @@ -171,7 +184,8 @@ TEST_F(ChannelReceiveTest, ReceiveReportGeneratedOnTime) { bool receiver_report_sent = false; EXPECT_CALL(transport_, SendRtcp) - .WillRepeatedly([&](ArrayView packet) { + .WillRepeatedly([&](ArrayView packet, + const PacketOptions& options) { if (packet.size() >= 2 && packet[1] == rtcp::ReceiverReport::kPacketType) { receiver_report_sent = true; @@ -189,10 +203,11 @@ TEST_F(ChannelReceiveTest, CaptureStartTimeBecomesValid) { auto channel = CreateTestChannelReceive(); EXPECT_CALL(transport_, SendRtcp) - .WillRepeatedly([&](ArrayView packet) { - HandleGeneratedRtcp(*channel, packet); - return true; - }); + .WillRepeatedly( + [&](ArrayView packet, const PacketOptions& options) { + HandleGeneratedRtcp(*channel, packet); + return true; + }); // Before any packets are sent, CaptureStartTime is invalid. EXPECT_EQ(ProbeCaptureStartNtpTime(*channel), -1); diff --git a/audio/voip/test/BUILD.gn b/audio/voip/test/BUILD.gn index 7f120065b5..2b80dc66c6 100644 --- a/audio/voip/test/BUILD.gn +++ b/audio/voip/test/BUILD.gn @@ -46,17 +46,25 @@ if (rtc_include_tests) { deps = [ ":mock_task_queue", "..:audio_channel", + "../../../api:array_view", + "../../../api:make_ref_counted", + "../../../api:scoped_refptr", "../../../api:transport_api", + "../../../api/audio:audio_frame_api", + "../../../api/audio:audio_mixer_api", + "../../../api/audio_codecs:audio_codecs_api", "../../../api/audio_codecs:builtin_audio_decoder_factory", "../../../api/audio_codecs:builtin_audio_encoder_factory", "../../../api/environment", "../../../api/environment:environment_factory", "../../../api/task_queue:task_queue", + "../../../api/voip:voip_api", "../../../modules/audio_mixer:audio_mixer_impl", "../../../modules/audio_mixer:audio_mixer_test_utils", "../../../modules/rtp_rtcp:rtp_rtcp", "../../../modules/rtp_rtcp:rtp_rtcp_format", "../../../rtc_base:logging", + "../../../system_wrappers", "../../../test:mock_transport", "../../../test:test_support", "//third_party/abseil-cpp/absl/functional:any_invocable", diff --git a/audio/voip/test/audio_channel_unittest.cc b/audio/voip/test/audio_channel_unittest.cc index 9a47e03710..ad71c67120 100644 --- a/audio/voip/test/audio_channel_unittest.cc +++ b/audio/voip/test/audio_channel_unittest.cc @@ -10,19 +10,32 @@ #include "audio/voip/audio_channel.h" +#include +#include +#include +#include +#include + #include "absl/functional/any_invocable.h" +#include "api/array_view.h" +#include "api/audio/audio_frame.h" +#include "api/audio/audio_mixer.h" +#include "api/audio_codecs/audio_decoder_factory.h" +#include "api/audio_codecs/audio_encoder_factory.h" +#include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" -#include "api/call/transport.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" -#include "api/task_queue/task_queue_base.h" -#include "api/task_queue/task_queue_factory.h" +#include "api/make_ref_counted.h" +#include "api/scoped_refptr.h" +#include "api/voip/voip_statistics.h" #include "audio/voip/test/mock_task_queue.h" #include "modules/audio_mixer/audio_mixer_impl.h" #include "modules/audio_mixer/sine_wave_generator.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" -#include "rtc_base/logging.h" +#include "system_wrappers/include/clock.h" #include "test/gmock.h" #include "test/gtest.h" #include "test/mock_transport.h" @@ -229,7 +242,7 @@ TEST_F(AudioChannelTest, TestChannelStatistics) { audio_channel_->ReceivedRTPPacket(packet); return true; }; - auto loop_rtcp = [&](ArrayView packet) { + auto loop_rtcp = [&](ArrayView packet, Unused) { audio_channel_->ReceivedRTCPPacket(packet); return true; }; @@ -316,7 +329,7 @@ TEST_F(AudioChannelTest, RttIsAvailableAfterChangeOfRemoteSsrc) { auto send_recv_rtcp = [&](scoped_refptr rtcp_sender, scoped_refptr rtcp_receiver) { // Setup routing logic via transport_. - auto route_rtcp = [&](ArrayView packet) { + auto route_rtcp = [&](ArrayView packet, Unused) { rtcp_receiver->ReceivedRTCPPacket(packet); return true; }; diff --git a/call/fake_network_pipe.cc b/call/fake_network_pipe.cc index a1150b30c1..f63bf11ef1 100644 --- a/call/fake_network_pipe.cc +++ b/call/fake_network_pipe.cc @@ -323,7 +323,8 @@ void FakeNetworkPipe::DeliverNetworkPacket(NetworkPacket* packet) { return; } if (packet->is_rtcp()) { - transport->SendRtcp(MakeArrayView(packet->data(), packet->data_length())); + transport->SendRtcp(MakeArrayView(packet->data(), packet->data_length()), + packet->packet_options()); } else { transport->SendRtp(MakeArrayView(packet->data(), packet->data_length()), packet->packet_options()); diff --git a/examples/androidvoip/jni/android_voip_client.cc b/examples/androidvoip/jni/android_voip_client.cc index f15355d748..cea3d48582 100644 --- a/examples/androidvoip/jni/android_voip_client.cc +++ b/examples/androidvoip/jni/android_voip_client.cc @@ -479,7 +479,8 @@ void AndroidVoipClient::SendRtcpPacket( } } -bool AndroidVoipClient::SendRtcp(webrtc::ArrayView packet) { +bool AndroidVoipClient::SendRtcp(webrtc::ArrayView packet, + const webrtc::PacketOptions& options) { std::vector packet_copy(packet.begin(), packet.end()); voip_thread_->PostTask([this, packet_copy = std::move(packet_copy)] { SendRtcpPacket(packet_copy); diff --git a/examples/androidvoip/jni/android_voip_client.h b/examples/androidvoip/jni/android_voip_client.h index 29a0fc82e5..cb8b927978 100644 --- a/examples/androidvoip/jni/android_voip_client.h +++ b/examples/androidvoip/jni/android_voip_client.h @@ -120,7 +120,8 @@ class AndroidVoipClient : public webrtc::Transport { // Implementation for Transport. bool SendRtp(webrtc::ArrayView packet, const webrtc::PacketOptions& options) override; - bool SendRtcp(webrtc::ArrayView packet) override; + bool SendRtcp(webrtc::ArrayView packet, + const webrtc::PacketOptions& options) override; void OnSignalReadRTPPacket(webrtc::AsyncPacketSocket* socket, const webrtc::ReceivedIpPacket& packet); diff --git a/media/base/media_channel_impl.cc b/media/base/media_channel_impl.cc index 40da627b4b..b88007703c 100644 --- a/media/base/media_channel_impl.cc +++ b/media/base/media_channel_impl.cc @@ -19,9 +19,6 @@ #include "api/array_view.h" #include "api/audio_options.h" #include "api/call/transport.h" -#include "api/crypto/frame_decryptor_interface.h" -#include "api/crypto/frame_encryptor_interface.h" -#include "api/frame_transformer_interface.h" #include "api/media_stream_interface.h" #include "api/rtc_error.h" #include "api/rtp_sender_interface.h" @@ -173,15 +170,32 @@ MediaChannelUtil::TransportForMediaChannels::~TransportForMediaChannels() { RTC_DCHECK(!network_interface_); } +AsyncSocketPacketOptions +MediaChannelUtil::TransportForMediaChannels::TranslatePacketOptions( + const PacketOptions& options) { + AsyncSocketPacketOptions rtc_options; + rtc_options.packet_id = options.packet_id; + if (DscpEnabled()) { + rtc_options.dscp = PreferredDscp(); + } + rtc_options.info_signaled_after_sent.included_in_feedback = + options.included_in_feedback; + rtc_options.info_signaled_after_sent.included_in_allocation = + options.included_in_allocation; + rtc_options.info_signaled_after_sent.is_media = options.is_media; + rtc_options.ecn_1 = options.send_as_ect1; + rtc_options.batchable = options.batchable; + rtc_options.last_packet_in_batch = options.last_packet_in_batch; + return rtc_options; +} + bool MediaChannelUtil::TransportForMediaChannels::SendRtcp( - ArrayView packet) { - auto send = [this, packet = CopyOnWriteBuffer( - packet, webrtc::kMaxRtpPacketLen)]() mutable { - AsyncSocketPacketOptions rtc_options; - if (DscpEnabled()) { - rtc_options.dscp = PreferredDscp(); - } - DoSendPacket(&packet, true, rtc_options); + ArrayView packet, + const PacketOptions& options) { + auto send = [this, + packet = CopyOnWriteBuffer(packet, webrtc::kMaxRtpPacketLen), + options]() mutable { + DoSendPacket(&packet, true, TranslatePacketOptions(options)); }; if (network_thread_->IsCurrent()) { @@ -195,28 +209,10 @@ bool MediaChannelUtil::TransportForMediaChannels::SendRtcp( bool MediaChannelUtil::TransportForMediaChannels::SendRtp( ArrayView packet, const webrtc::PacketOptions& options) { - auto send = [this, packet_id = options.packet_id, - included_in_feedback = options.included_in_feedback, - included_in_allocation = options.included_in_allocation, - batchable = options.batchable, - last_packet_in_batch = options.last_packet_in_batch, - is_media = options.is_media, ect_1 = options.send_as_ect1, - packet = CopyOnWriteBuffer(packet, - webrtc::kMaxRtpPacketLen)]() mutable { - AsyncSocketPacketOptions rtc_options; - rtc_options.packet_id = packet_id; - if (DscpEnabled()) { - rtc_options.dscp = PreferredDscp(); - } - rtc_options.info_signaled_after_sent.included_in_feedback = - included_in_feedback; - rtc_options.info_signaled_after_sent.included_in_allocation = - included_in_allocation; - rtc_options.info_signaled_after_sent.is_media = is_media; - rtc_options.ecn_1 = ect_1; - rtc_options.batchable = batchable; - rtc_options.last_packet_in_batch = last_packet_in_batch; - DoSendPacket(&packet, false, rtc_options); + auto send = [this, + packet = CopyOnWriteBuffer(packet, webrtc::kMaxRtpPacketLen), + options]() mutable { + DoSendPacket(&packet, false, TranslatePacketOptions(options)); }; // TODO(bugs.webrtc.org/11993): ModuleRtpRtcpImpl2 and related classes (e.g. diff --git a/media/base/media_channel_impl.h b/media/base/media_channel_impl.h index 3c30e6bd1e..8c222b23b7 100644 --- a/media/base/media_channel_impl.h +++ b/media/base/media_channel_impl.h @@ -91,7 +91,8 @@ class MediaChannelUtil { // Implementation of webrtc::Transport bool SendRtp(ArrayView packet, const PacketOptions& options) override; - bool SendRtcp(ArrayView packet) override; + bool SendRtcp(ArrayView packet, + const PacketOptions& options) override; // Not implementation of webrtc::Transport void SetInterface(MediaChannelNetworkInterface* iface); @@ -99,6 +100,8 @@ class MediaChannelUtil { int SetOption(MediaChannelNetworkInterface::SocketType type, Socket::Option opt, int option); + AsyncSocketPacketOptions TranslatePacketOptions( + const PacketOptions& options); bool DoSendPacket(CopyOnWriteBuffer* packet, bool rtcp, diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc index 8e1029a2e1..0d2803ba3c 100644 --- a/media/engine/webrtc_video_engine_unittest.cc +++ b/media/engine/webrtc_video_engine_unittest.cc @@ -5788,7 +5788,8 @@ TEST_F(WebRtcVideoChannelTest, TestSetDscpOptions) { // Packets should also self-identify their dscp in PacketOptions. const uint8_t kData[10] = {0}; - EXPECT_TRUE(ChannelImplAsTransport(send_channel.get())->SendRtcp(kData)); + EXPECT_TRUE(ChannelImplAsTransport(send_channel.get()) + ->SendRtcp(kData, /*packet_options=*/{})); EXPECT_EQ(DSCP_CS1, network_interface->options().dscp); send_channel->SetInterface(nullptr); diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc index 4223f8878b..c4d1a1d02d 100644 --- a/media/engine/webrtc_voice_engine_unittest.cc +++ b/media/engine/webrtc_voice_engine_unittest.cc @@ -30,6 +30,7 @@ #include "api/audio_codecs/builtin_audio_encoder_factory.h" #include "api/audio_options.h" #include "api/call/audio_sink.h" +#include "api/call/transport.h" #include "api/crypto/crypto_options.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" @@ -3427,7 +3428,9 @@ TEST_P(WebRtcVoiceEngineTestFake, TestSetDscpOptions) { // Packets should also self-identify their dscp in PacketOptions. const uint8_t kData[10] = {0}; - EXPECT_TRUE(SendImplFromPointer(channel.get())->transport()->SendRtcp(kData)); + EXPECT_TRUE(SendImplFromPointer(channel.get()) + ->transport() + ->SendRtcp(kData, /*packet_options=*/{})); EXPECT_EQ(webrtc::DSCP_CS1, network_interface.options().dscp); channel->SetInterface(nullptr); diff --git a/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/modules/rtp_rtcp/source/nack_rtx_unittest.cc index 28b75a7a25..ea0a21a6b9 100644 --- a/modules/rtp_rtcp/source/nack_rtx_unittest.cc +++ b/modules/rtp_rtcp/source/nack_rtx_unittest.cc @@ -115,7 +115,8 @@ class RtxLoopBackTransport : public webrtc::Transport { return true; } - bool SendRtcp(ArrayView data) override { + bool SendRtcp(ArrayView data, + const PacketOptions& /* options */) override { module_->IncomingRtcpPacket(data); return true; } diff --git a/modules/rtp_rtcp/source/rtcp_sender.cc b/modules/rtp_rtcp/source/rtcp_sender.cc index c08fd08a73..e1cbe26e24 100644 --- a/modules/rtp_rtcp/source/rtcp_sender.cc +++ b/modules/rtp_rtcp/source/rtcp_sender.cc @@ -22,6 +22,7 @@ #include "absl/strings/string_view.h" #include "api/array_view.h" +#include "api/call/transport.h" #include "api/environment/environment.h" #include "api/rtc_event_log/rtc_event_log.h" #include "api/rtp_headers.h" @@ -238,7 +239,7 @@ int32_t RTCPSender::SendLossNotification(const FeedbackState& feedback_state, bool buffering_allowed) { int32_t error_code = -1; auto callback = [&](ArrayView packet) { - transport_->SendRtcp(packet); + transport_->SendRtcp(packet, /*packet_options=*/{}); error_code = 0; env_.event_log().Log(std::make_unique(packet)); }; @@ -596,7 +597,7 @@ int32_t RTCPSender::SendRTCP(const FeedbackState& feedback_state, const uint16_t* nack_list) { int32_t error_code = -1; auto callback = [&](ArrayView packet) { - if (transport_->SendRtcp(packet)) { + if (transport_->SendRtcp(packet, /*packet_options=*/{})) { error_code = 0; env_.event_log().Log( std::make_unique(packet)); @@ -894,7 +895,7 @@ void RTCPSender::SendCombinedRtcpPacket( } RTC_DCHECK_LE(max_packet_size, IP_PACKET_SIZE); auto callback = [&](ArrayView packet) { - if (transport_->SendRtcp(packet)) { + if (transport_->SendRtcp(packet, /*packet_options=*/{})) { env_.event_log().Log( std::make_unique(packet)); } diff --git a/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/modules/rtp_rtcp/source/rtcp_sender_unittest.cc index ffba6d76dd..e93b38e625 100644 --- a/modules/rtp_rtcp/source/rtcp_sender_unittest.cc +++ b/modules/rtp_rtcp/source/rtcp_sender_unittest.cc @@ -81,7 +81,9 @@ class TestTransport : public Transport { const PacketOptions& /* options */) override { return false; } - bool SendRtcp(ArrayView data) override { + bool SendRtcp(ArrayView data, + const PacketOptions& options) override { + EXPECT_FALSE(options.is_media); parser_.Parse(data); return true; } @@ -680,8 +682,8 @@ TEST_F(RtcpSenderTest, SendsTmmbnIfSetAndEmpty) { // of a RTCP compound packet. TEST_F(RtcpSenderTest, ByeMustBeLast) { MockTransport mock_transport; - EXPECT_CALL(mock_transport, SendRtcp(_)) - .WillOnce(Invoke([](ArrayView data) { + EXPECT_CALL(mock_transport, SendRtcp(_, _)) + .WillOnce(Invoke([](ArrayView data, ::testing::Unused) { const uint8_t* next_packet = data.data(); const uint8_t* const packet_end = data.data() + data.size(); rtcp::CommonHeader packet; diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc index 40bfcd3b4b..8fa658019a 100644 --- a/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc @@ -131,7 +131,8 @@ class SendTransport : public Transport, ++rtp_packets_sent_; return true; } - bool SendRtcp(ArrayView data) override { + bool SendRtcp(ArrayView data, + const PacketOptions& /* options */) override { test::RtcpPacketParser parser; parser.Parse(data); last_nack_list_ = parser.nack()->packet_ids(); diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc index 0fae24d0fb..153bc6ce3a 100644 --- a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc @@ -103,7 +103,8 @@ class SendTransport : public Transport { last_rtp_sequence_number_ = packet.SequenceNumber(); return true; } - bool SendRtcp(ArrayView data) override { + bool SendRtcp(ArrayView data, + const PacketOptions& /* options */) override { test::RtcpPacketParser parser; parser.Parse(data); last_nack_list_ = parser.nack()->packet_ids(); diff --git a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc index 6ce33aeb12..83e5a1ffcd 100644 --- a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc @@ -59,7 +59,10 @@ class LoopbackTransportTest : public webrtc::Transport { EXPECT_TRUE(sent_packets_.back().Parse(data)); return true; } - bool SendRtcp(ArrayView /* data */) override { return false; } + bool SendRtcp(ArrayView /* data */, + const PacketOptions& /* options */) override { + return false; + } const RtpPacketReceived& last_sent_packet() { return sent_packets_.back(); } int packets_sent() { return sent_packets_.size(); } diff --git a/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc index 833a93bfce..e86296f0ac 100644 --- a/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc @@ -115,7 +115,10 @@ class TestTransport : public Transport { return true; } - bool SendRtcp(ArrayView) override { RTC_CHECK_NOTREACHED(); } + bool SendRtcp(ArrayView /* packet */, + const PacketOptions& /* options */) override { + RTC_CHECK_NOTREACHED(); + } std::optional last_packet() { return last_packet_; } diff --git a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc index b5045aee12..a67ebfd35c 100644 --- a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc @@ -147,7 +147,10 @@ class LoopbackTransportTest : public webrtc::Transport { EXPECT_TRUE(sent_packets_.back().Parse(data)); return true; } - bool SendRtcp(ArrayView /* data */) override { return false; } + bool SendRtcp(ArrayView /* data */, + const PacketOptions& /* options */) override { + return false; + } const RtpPacketReceived& last_sent_packet() { return sent_packets_.back(); } int packets_sent() { return sent_packets_.size(); } const std::vector& sent_packets() const { diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn index bac6167a60..052ffd9e39 100644 --- a/rtc_tools/BUILD.gn +++ b/rtc_tools/BUILD.gn @@ -183,12 +183,16 @@ if (!is_component_build) { "..//api/video_codecs:video_encoder_factory_template_libvpx_vp9_adapter", "..//api/video_codecs:video_encoder_factory_template_open_h264_adapter", "..//test/network:simulated_network", + "../api:array_view", "../api:create_frame_generator", + "../api:make_ref_counted", + "../api:rtp_headers", "../api:rtp_parameters", "../api:transport_api", "../api/environment", "../api/environment:environment_factory", "../api/video:builtin_video_bitrate_allocator_factory", + "../api/video:video_bitrate_allocator_factory", "../api/video_codecs:video_codecs_api", "../call", "../call:call_interfaces", @@ -199,9 +203,11 @@ if (!is_component_build) { "../call:video_send_stream_api", "../media:media_constants", "../media:rtc_audio_video", + "../rtc_base:logging", "../rtc_base:rtc_json", "../rtc_base:threading", "../rtc_base/system:file_wrapper", + "../system_wrappers", "../test:fileutils", "../test:frame_generator_capturer", "../test:rtp_test_utils", diff --git a/rtc_tools/rtp_generator/rtp_generator.cc b/rtc_tools/rtp_generator/rtp_generator.cc index cb4a14cba4..51228a4b65 100644 --- a/rtc_tools/rtp_generator/rtp_generator.cc +++ b/rtc_tools/rtp_generator/rtp_generator.cc @@ -11,27 +11,47 @@ #include "rtc_tools/rtp_generator/rtp_generator.h" #include +#include +#include +#include #include +#include +#include #include +#include +#include "api/array_view.h" +#include "api/call/transport.h" #include "api/environment/environment_factory.h" +#include "api/make_ref_counted.h" +#include "api/media_types.h" +#include "api/rtp_headers.h" +#include "api/rtp_parameters.h" #include "api/test/create_frame_generator.h" +#include "api/video/builtin_video_bitrate_allocator_factory.h" +#include "api/video_codecs/video_codec.h" #include "api/video_codecs/video_decoder_factory_template.h" #include "api/video_codecs/video_decoder_factory_template_dav1d_adapter.h" #include "api/video_codecs/video_decoder_factory_template_libvpx_vp8_adapter.h" #include "api/video_codecs/video_decoder_factory_template_libvpx_vp9_adapter.h" #include "api/video_codecs/video_encoder.h" -#include "api/video_codecs/video_encoder_factory.h" #include "api/video_codecs/video_encoder_factory_template.h" #include "api/video_codecs/video_encoder_factory_template_libaom_av1_adapter.h" #include "api/video_codecs/video_encoder_factory_template_libvpx_vp8_adapter.h" #include "api/video_codecs/video_encoder_factory_template_libvpx_vp9_adapter.h" +#include "call/call.h" +#include "call/call_config.h" +#include "call/video_send_stream.h" #include "media/base/media_constants.h" +#include "rtc_base/logging.h" #include "rtc_base/strings/json.h" #include "rtc_base/system/file_wrapper.h" #include "rtc_base/thread.h" +#include "system_wrappers/include/clock.h" +#include "test/frame_generator_capturer.h" +#include "test/rtp_file_reader.h" +#include "test/rtp_file_writer.h" #include "test/testsupport/file_utils.h" -#include "video/config/encoder_stream_factory.h" #include "video/config/video_encoder_config.h" namespace webrtc { @@ -281,13 +301,14 @@ void RtpGenerator::GenerateRtpDump(const std::string& rtp_dump_path) { } bool RtpGenerator::SendRtp(ArrayView packet, - const webrtc::PacketOptions& options) { + const PacketOptions& /* options */) { test::RtpPacket rtp_packet = DataToRtpPacket(packet.data(), packet.size()); rtp_dump_writer_->WritePacket(&rtp_packet); return true; } -bool RtpGenerator::SendRtcp(ArrayView packet) { +bool RtpGenerator::SendRtcp(ArrayView packet, + const PacketOptions& /* options */) { test::RtpPacket rtcp_packet = DataToRtpPacket(packet.data(), packet.size()); rtp_dump_writer_->WritePacket(&rtcp_packet); return true; diff --git a/rtc_tools/rtp_generator/rtp_generator.h b/rtc_tools/rtp_generator/rtp_generator.h index bb5f141f5a..cd7ad29848 100644 --- a/rtc_tools/rtp_generator/rtp_generator.h +++ b/rtc_tools/rtp_generator/rtp_generator.h @@ -11,14 +11,17 @@ #ifndef RTC_TOOLS_RTP_GENERATOR_RTP_GENERATOR_H_ #define RTC_TOOLS_RTP_GENERATOR_RTP_GENERATOR_H_ +#include +#include #include +#include #include #include +#include "api/array_view.h" #include "api/call/transport.h" #include "api/environment/environment.h" -#include "api/media_types.h" -#include "api/video/builtin_video_bitrate_allocator_factory.h" +#include "api/video/video_bitrate_allocator_factory.h" #include "api/video_codecs/video_decoder_factory.h" #include "api/video_codecs/video_encoder_factory.h" #include "call/call.h" @@ -28,7 +31,6 @@ #include "test/frame_generator_capturer.h" #include "test/rtp_file_reader.h" #include "test/rtp_file_writer.h" -#include "video/config/video_encoder_config.h" namespace webrtc { @@ -92,9 +94,10 @@ class RtpGenerator final : public webrtc::Transport { // Captured RTP packets are written to the RTPDump file instead of over the // network. bool SendRtp(ArrayView packet, - const webrtc::PacketOptions& options) override; + const PacketOptions& options) override; // RTCP packets are ignored for now. - bool SendRtcp(ArrayView packet) override; + bool SendRtcp(ArrayView packet, + const PacketOptions& options) override; // Returns the maximum duration int GetMaxDuration() const; // Waits until all video streams have finished. diff --git a/test/BUILD.gn b/test/BUILD.gn index a17bbcdb13..292a4c6662 100644 --- a/test/BUILD.gn +++ b/test/BUILD.gn @@ -1031,7 +1031,10 @@ rtc_library("null_transport") { "null_transport.cc", "null_transport.h", ] - deps = [ "../api:transport_api" ] + deps = [ + "../api:array_view", + "../api:transport_api", + ] } rtc_library("encoder_settings") { @@ -1086,6 +1089,7 @@ rtc_library("mock_transport") { ] deps = [ ":test_support", + "../api:array_view", "../api:transport_api", ] } diff --git a/test/direct_transport.cc b/test/direct_transport.cc index fe68dd7048..7d96b81e5d 100644 --- a/test/direct_transport.cc +++ b/test/direct_transport.cc @@ -120,7 +120,8 @@ bool DirectTransport::SendRtp(ArrayView data, return true; } -bool DirectTransport::SendRtcp(ArrayView data) { +bool DirectTransport::SendRtcp(ArrayView data, + const PacketOptions& /* options */) { fake_network_->DeliverRtcpPacket(CopyOnWriteBuffer(data)); MutexLock lock(&process_lock_); if (!next_process_task_.Running()) diff --git a/test/direct_transport.h b/test/direct_transport.h index 4cf3f262b6..3b79957a3b 100644 --- a/test/direct_transport.h +++ b/test/direct_transport.h @@ -10,14 +10,19 @@ #ifndef TEST_DIRECT_TRANSPORT_H_ #define TEST_DIRECT_TRANSPORT_H_ +#include +#include +#include #include +#include "api/array_view.h" #include "api/call/transport.h" -#include "api/sequence_checker.h" +#include "api/media_types.h" +#include "api/rtp_parameters.h" #include "api/task_queue/task_queue_base.h" -#include "api/test/simulated_network.h" #include "call/call.h" #include "call/simulated_packet_receiver.h" +#include "modules/rtp_rtcp/include/rtp_header_extension_map.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/task_utils/repeating_task.h" #include "rtc_base/thread_annotations.h" @@ -63,7 +68,8 @@ class DirectTransport : public Transport { bool SendRtp(ArrayView data, const PacketOptions& options) override; - bool SendRtcp(ArrayView data) override; + bool SendRtcp(ArrayView data, + const PacketOptions& options) override; int GetAverageDelayMs(); diff --git a/test/mock_transport.h b/test/mock_transport.h index 87b60b3068..9cd6113ce7 100644 --- a/test/mock_transport.h +++ b/test/mock_transport.h @@ -11,6 +11,9 @@ #ifndef TEST_MOCK_TRANSPORT_H_ #define TEST_MOCK_TRANSPORT_H_ +#include + +#include "api/array_view.h" #include "api/call/transport.h" #include "test/gmock.h" @@ -25,7 +28,10 @@ class MockTransport : public Transport { SendRtp, (webrtc::ArrayView, const PacketOptions&), (override)); - MOCK_METHOD(bool, SendRtcp, (webrtc::ArrayView), (override)); + MOCK_METHOD(bool, + SendRtcp, + (webrtc::ArrayView, const PacketOptions&), + (override)); }; } // namespace webrtc diff --git a/test/null_transport.cc b/test/null_transport.cc index 884f294f29..ae6e19ba9f 100644 --- a/test/null_transport.cc +++ b/test/null_transport.cc @@ -9,6 +9,11 @@ */ #include "test/null_transport.h" +#include + +#include "api/array_view.h" +#include "api/call/transport.h" + namespace webrtc { namespace test { @@ -17,7 +22,8 @@ bool NullTransport::SendRtp(ArrayView packet, return true; } -bool NullTransport::SendRtcp(ArrayView packet) { +bool NullTransport::SendRtcp(ArrayView packet, + const PacketOptions& options) { return true; } diff --git a/test/null_transport.h b/test/null_transport.h index 16496225af..a65c6d77d4 100644 --- a/test/null_transport.h +++ b/test/null_transport.h @@ -10,6 +10,9 @@ #ifndef TEST_NULL_TRANSPORT_H_ #define TEST_NULL_TRANSPORT_H_ +#include + +#include "api/array_view.h" #include "api/call/transport.h" namespace webrtc { @@ -21,7 +24,8 @@ class NullTransport : public Transport { public: bool SendRtp(ArrayView packet, const PacketOptions& options) override; - bool SendRtcp(ArrayView packet) override; + bool SendRtcp(ArrayView packet, + const PacketOptions& options) override; }; } // namespace test } // namespace webrtc diff --git a/test/rtp_rtcp_observer.h b/test/rtp_rtcp_observer.h index d4b8ec4352..f392674bb8 100644 --- a/test/rtp_rtcp_observer.h +++ b/test/rtp_rtcp_observer.h @@ -10,20 +10,23 @@ #ifndef TEST_RTP_RTCP_OBSERVER_H_ #define TEST_RTP_RTCP_OBSERVER_H_ +#include #include #include #include -#include #include "absl/flags/flag.h" #include "api/array_view.h" -#include "api/test/simulated_network.h" +#include "api/call/transport.h" +#include "api/media_types.h" +#include "api/rtp_parameters.h" +#include "api/task_queue/task_queue_base.h" #include "api/units/time_delta.h" +#include "call/call.h" #include "call/simulated_packet_receiver.h" -#include "call/video_send_stream.h" #include "modules/rtp_rtcp/source/rtp_util.h" +#include "rtc_base/checks.h" #include "rtc_base/event.h" -#include "system_wrappers/include/field_trial.h" #include "test/direct_transport.h" #include "test/gtest.h" #include "test/test_flags.h" @@ -116,10 +119,12 @@ class PacketTransport : public test::DirectTransport { case RtpRtcpObserver::SEND_PACKET: return test::DirectTransport::SendRtp(packet, options); } - return true; // Will never happen, makes compiler happy. + RTC_DCHECK_NOTREACHED(); + return true; } - bool SendRtcp(ArrayView packet) override { + bool SendRtcp(ArrayView packet, + const PacketOptions& options) override { EXPECT_TRUE(IsRtcpPacket(packet)); RtpRtcpObserver::Action action = RtpRtcpObserver::SEND_PACKET; if (observer_) { @@ -134,9 +139,10 @@ class PacketTransport : public test::DirectTransport { // Drop packet silently. return true; case RtpRtcpObserver::SEND_PACKET: - return test::DirectTransport::SendRtcp(packet); + return test::DirectTransport::SendRtcp(packet, options); } - return true; // Will never happen, makes compiler happy. + RTC_DCHECK_NOTREACHED(); + return true; } RtpRtcpObserver* const observer_; diff --git a/test/scenario/network_node.cc b/test/scenario/network_node.cc index e64cf62cba..364f2e6da9 100644 --- a/test/scenario/network_node.cc +++ b/test/scenario/network_node.cc @@ -122,7 +122,8 @@ bool NetworkNodeTransport::SendRtp(ArrayView packet, return true; } -bool NetworkNodeTransport::SendRtcp(ArrayView packet) { +bool NetworkNodeTransport::SendRtcp(ArrayView packet, + const PacketOptions& options) { CopyOnWriteBuffer buffer(packet); MutexLock lock(&mutex_); if (!endpoint_) diff --git a/test/scenario/network_node.h b/test/scenario/network_node.h index e2f6f59c39..e0086fa6b4 100644 --- a/test/scenario/network_node.h +++ b/test/scenario/network_node.h @@ -62,7 +62,8 @@ class NetworkNodeTransport : public Transport { bool SendRtp(ArrayView packet, const PacketOptions& options) override; - bool SendRtcp(ArrayView packet) override; + bool SendRtcp(ArrayView packet, + const PacketOptions& options) override; void Connect(EmulatedEndpoint* endpoint, const SocketAddress& receiver_address, diff --git a/video/BUILD.gn b/video/BUILD.gn index 3e7af5a1e9..68bc3cfca4 100644 --- a/video/BUILD.gn +++ b/video/BUILD.gn @@ -555,6 +555,7 @@ if (rtc_include_tests) { "../api/test/metrics:global_metrics_logger_and_exporter", "../api/test/metrics:metric", "../api/units:time_delta", + "../api/units:timestamp", "../api/video:builtin_video_bitrate_allocator_factory", "../api/video:encoded_image", "../api/video:video_bitrate_allocation", @@ -887,6 +888,7 @@ if (rtc_include_tests) { "../api:sequence_checker", "../api:simulated_network_api", "../api:time_controller", + "../api:transport_api", "../api:video_track_source_constraints", "../api/adaptation:resource_adaptation_api", "../api/crypto:options", @@ -974,6 +976,7 @@ if (rtc_include_tests) { "../modules/video_coding/svc:scalability_structures", "../modules/video_coding/svc:scalable_video_controller", "../modules/video_coding/timing:timing_module", + "../rtc_base:buffer", "../rtc_base:byte_buffer", "../rtc_base:checks", "../rtc_base:copy_on_write_buffer", @@ -999,6 +1002,7 @@ if (rtc_include_tests) { "../rtc_base/experiments:rate_control_settings", "../rtc_base/synchronization:mutex", "../system_wrappers", + "../system_wrappers:field_trial", "../system_wrappers:metrics", "../test:direct_transport", "../test:encoder_settings", diff --git a/video/end_to_end_tests/bandwidth_tests.cc b/video/end_to_end_tests/bandwidth_tests.cc index 82643ef59d..d327a7670d 100644 --- a/video/end_to_end_tests/bandwidth_tests.cc +++ b/video/end_to_end_tests/bandwidth_tests.cc @@ -8,30 +8,45 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include #include +#include +#include +#include "api/array_view.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" +#include "api/rtp_headers.h" +#include "api/rtp_parameters.h" #include "api/task_queue/task_queue_base.h" #include "api/test/simulated_network.h" +#include "api/transport/bitrate_settings.h" #include "api/units/time_delta.h" #include "api/video/builtin_video_bitrate_allocator_factory.h" #include "api/video/video_bitrate_allocation.h" -#include "call/fake_network_pipe.h" +#include "api/video/video_bitrate_allocator_factory.h" +#include "call/call.h" +#include "call/video_receive_stream.h" +#include "call/video_send_stream.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h" +#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h" +#include "rtc_base/checks.h" #include "rtc_base/rate_limiter.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/task_queue_for_test.h" +#include "rtc_base/thread_annotations.h" +#include "system_wrappers/include/field_trial.h" #include "system_wrappers/include/sleep.h" #include "test/call_test.h" #include "test/fake_encoder.h" #include "test/field_trial.h" #include "test/gtest.h" -#include "test/network/simulated_network.h" #include "test/rtcp_packet_parser.h" #include "test/rtp_rtcp_observer.h" #include "test/video_encoder_proxy_factory.h" #include "test/video_test_constants.h" +#include "video/config/video_encoder_config.h" namespace webrtc { namespace { diff --git a/video/end_to_end_tests/network_state_tests.cc b/video/end_to_end_tests/network_state_tests.cc index 4def7097af..1f9b33664a 100644 --- a/video/end_to_end_tests/network_state_tests.cc +++ b/video/end_to_end_tests/network_state_tests.cc @@ -8,24 +8,40 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include #include +#include +#include "api/array_view.h" +#include "api/call/transport.h" +#include "api/environment/environment.h" #include "api/media_types.h" -#include "api/task_queue/default_task_queue_factory.h" +#include "api/rtp_headers.h" #include "api/task_queue/task_queue_base.h" #include "api/task_queue/task_queue_factory.h" #include "api/test/simulated_network.h" +#include "api/units/time_delta.h" +#include "api/video/video_frame.h" +#include "api/video/video_frame_type.h" +#include "api/video_codecs/video_codec.h" #include "api/video_codecs/video_encoder.h" +#include "call/call.h" #include "call/fake_network_pipe.h" +#include "call/video_receive_stream.h" +#include "call/video_send_stream.h" #include "modules/rtp_rtcp/source/rtp_packet.h" +#include "rtc_base/event.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/task_queue_for_test.h" +#include "rtc_base/thread_annotations.h" #include "system_wrappers/include/sleep.h" #include "test/call_test.h" #include "test/fake_encoder.h" #include "test/gtest.h" -#include "test/network/simulated_network.h" +#include "test/rtp_rtcp_observer.h" #include "test/video_encoder_proxy_factory.h" +#include "test/video_test_constants.h" +#include "video/config/video_encoder_config.h" namespace webrtc { namespace { @@ -42,7 +58,8 @@ class NetworkStateEndToEndTest : public test::CallTest { return false; } - bool SendRtcp(ArrayView packet) override { + bool SendRtcp(ArrayView packet, + const PacketOptions& /* options */) override { ADD_FAILURE() << "Unexpected RTCP sent."; return false; } @@ -68,7 +85,8 @@ class NetworkStateEndToEndTest : public test::CallTest { return true; } - bool SendRtcp(ArrayView packet) override { + bool SendRtcp(ArrayView packet, + const PacketOptions& /* options */) override { MutexLock lock(&mutex_); need_rtcp_ = false; return true; diff --git a/video/end_to_end_tests/retransmission_tests.cc b/video/end_to_end_tests/retransmission_tests.cc index 7c50a7c034..836a4a4598 100644 --- a/video/end_to_end_tests/retransmission_tests.cc +++ b/video/end_to_end_tests/retransmission_tests.cc @@ -8,25 +8,45 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include - -#include "absl/algorithm/container.h" +#include +#include +#include +#include +#include +#include + +#include "api/array_view.h" +#include "api/call/transport.h" +#include "api/environment/environment.h" +#include "api/rtp_parameters.h" #include "api/task_queue/task_queue_base.h" -#include "api/test/simulated_network.h" #include "api/test/video/function_video_encoder_factory.h" #include "api/units/time_delta.h" +#include "api/video/video_codec_type.h" +#include "api/video/video_frame.h" +#include "api/video/video_rotation.h" +#include "api/video/video_sink_interface.h" +#include "api/video_codecs/sdp_video_format.h" +#include "call/audio_receive_stream.h" +#include "call/audio_send_stream.h" #include "call/fake_network_pipe.h" +#include "call/video_receive_stream.h" +#include "call/video_send_stream.h" +#include "modules/rtp_rtcp/source/rtcp_packet/nack.h" #include "modules/rtp_rtcp/source/rtp_packet.h" #include "modules/video_coding/codecs/vp8/include/vp8.h" +#include "rtc_base/buffer.h" +#include "rtc_base/checks.h" #include "rtc_base/event.h" #include "rtc_base/synchronization/mutex.h" -#include "rtc_base/task_queue_for_test.h" +#include "rtc_base/thread_annotations.h" #include "test/call_test.h" -#include "test/field_trial.h" +#include "test/frame_generator_capturer.h" #include "test/gtest.h" -#include "test/network/simulated_network.h" #include "test/rtcp_packet_parser.h" +#include "test/rtp_rtcp_observer.h" #include "test/video_test_constants.h" +#include "video/config/video_encoder_config.h" namespace webrtc { namespace { @@ -166,7 +186,8 @@ TEST_F(RetransmissionEndToEndTest, ReceivesNackAndRetransmitsAudio) { nack.SetPacketIds(nack_list, 1); Buffer buffer = nack.Build(); - EXPECT_TRUE(receive_transport_->SendRtcp(buffer)); + EXPECT_TRUE( + receive_transport_->SendRtcp(buffer, /*packet_options=*/{})); } return SEND_PACKET; diff --git a/video/end_to_end_tests/rtp_rtcp_tests.cc b/video/end_to_end_tests/rtp_rtcp_tests.cc index e372a9408c..712d4a82bc 100644 --- a/video/end_to_end_tests/rtp_rtcp_tests.cc +++ b/video/end_to_end_tests/rtp_rtcp_tests.cc @@ -8,21 +8,47 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include +#include +#include +#include +#include #include - +#include +#include + +#include "api/array_view.h" +#include "api/call/transport.h" +#include "api/environment/environment.h" +#include "api/field_trials_view.h" +#include "api/make_ref_counted.h" +#include "api/rtp_headers.h" #include "api/test/simulated_network.h" +#include "api/test/video/function_video_encoder_factory.h" +#include "api/video/video_codec_type.h" +#include "api/video_codecs/sdp_video_format.h" #include "call/fake_network_pipe.h" +#include "call/flexfec_receive_stream.h" +#include "call/video_receive_stream.h" +#include "call/video_send_stream.h" #include "modules/include/module_common_types_public.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h" #include "modules/rtp_rtcp/source/rtp_packet.h" #include "modules/video_coding/codecs/vp8/include/vp8.h" +#include "rtc_base/buffer.h" +#include "rtc_base/checks.h" #include "rtc_base/numerics/sequence_number_unwrapper.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/task_queue_for_test.h" +#include "rtc_base/thread_annotations.h" #include "test/call_test.h" +#include "test/encoder_settings.h" #include "test/gtest.h" -#include "test/network/simulated_network.h" #include "test/rtcp_packet_parser.h" +#include "test/rtp_rtcp_observer.h" #include "test/video_test_constants.h" +#include "video/config/video_encoder_config.h" namespace webrtc { namespace { @@ -337,7 +363,7 @@ void RtpRtcpEndToEndTest::TestRtpStatePreservation( rtcp::RapidResyncRequest force_send_sr_back_request; Buffer packet = force_send_sr_back_request.Build(); static_cast(receive_transport_.get()) - ->SendRtcp(packet); + ->SendRtcp(packet, /*packet_options=*/{}); } CreateFrameGeneratorCapturer(30, 1280, 720); StartVideoSources(); diff --git a/video/rtp_video_stream_receiver2_unittest.cc b/video/rtp_video_stream_receiver2_unittest.cc index 068f476196..6a8f96368c 100644 --- a/video/rtp_video_stream_receiver2_unittest.cc +++ b/video/rtp_video_stream_receiver2_unittest.cc @@ -207,7 +207,9 @@ class RtpVideoStreamReceiver2Test : public ::testing::Test, /*raw_payload=*/false); ON_CALL(mock_transport_, SendRtcp) .WillByDefault( - Invoke(&rtcp_packet_parser_, &test::RtcpPacketParser::Parse)); + Invoke([this](ArrayView packet, ::testing::Unused) { + return rtcp_packet_parser_.Parse(packet); + })); } RTPVideoHeader GetDefaultH264VideoHeader() { diff --git a/video/transport_adapter.cc b/video/transport_adapter.cc index b3116ea528..95e430f00e 100644 --- a/video/transport_adapter.cc +++ b/video/transport_adapter.cc @@ -10,6 +10,10 @@ #include "video/transport_adapter.h" +#include + +#include "api/array_view.h" +#include "api/call/transport.h" #include "rtc_base/checks.h" namespace webrtc { @@ -30,11 +34,12 @@ bool TransportAdapter::SendRtp(ArrayView packet, return transport_->SendRtp(packet, options); } -bool TransportAdapter::SendRtcp(ArrayView packet) { +bool TransportAdapter::SendRtcp(ArrayView packet, + const PacketOptions& options) { if (!enabled_.load()) return false; - return transport_->SendRtcp(packet); + return transport_->SendRtcp(packet, options); } void TransportAdapter::Enable() { diff --git a/video/transport_adapter.h b/video/transport_adapter.h index 28c113e6ee..1752851500 100644 --- a/video/transport_adapter.h +++ b/video/transport_adapter.h @@ -15,6 +15,7 @@ #include +#include "api/array_view.h" #include "api/call/transport.h" namespace webrtc { @@ -27,7 +28,8 @@ class TransportAdapter : public Transport { bool SendRtp(ArrayView packet, const PacketOptions& options) override; - bool SendRtcp(ArrayView packet) override; + bool SendRtcp(ArrayView packet, + const PacketOptions& options) override; void Enable(); void Disable(); diff --git a/video/video_analyzer.cc b/video/video_analyzer.cc index 0cd9babf18..94f06ee465 100644 --- a/video/video_analyzer.cc +++ b/video/video_analyzer.cc @@ -337,8 +337,9 @@ bool VideoAnalyzer::SendRtp(ArrayView packet, return result; } -bool VideoAnalyzer::SendRtcp(ArrayView packet) { - return transport_->SendRtcp(packet); +bool VideoAnalyzer::SendRtcp(ArrayView packet, + const PacketOptions& options) { + return transport_->SendRtcp(packet, options); } void VideoAnalyzer::OnFrame(const VideoFrame& video_frame) { diff --git a/video/video_analyzer.h b/video/video_analyzer.h index 6d68a5daf8..8ccd353fc0 100644 --- a/video/video_analyzer.h +++ b/video/video_analyzer.h @@ -10,24 +10,43 @@ #ifndef VIDEO_VIDEO_ANALYZER_H_ #define VIDEO_VIDEO_ANALYZER_H_ +#include +#include +#include #include #include #include +#include #include #include #include "absl/strings/string_view.h" +#include "api/array_view.h" +#include "api/call/transport.h" +#include "api/media_types.h" #include "api/numerics/samples_stats_counter.h" #include "api/task_queue/task_queue_base.h" #include "api/test/metrics/metric.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" +#include "api/video/video_frame.h" +#include "api/video/video_sink_interface.h" #include "api/video/video_source_interface.h" +#include "call/audio_receive_stream.h" +#include "call/call.h" +#include "call/packet_receiver.h" +#include "call/video_receive_stream.h" +#include "call/video_send_stream.h" #include "modules/rtp_rtcp/source/rtp_packet.h" +#include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "modules/rtp_rtcp/source/video_rtp_depacketizer.h" +#include "rtc_base/copy_on_write_buffer.h" #include "rtc_base/event.h" -#include "rtc_base/numerics/running_statistics.h" #include "rtc_base/numerics/sequence_number_unwrapper.h" #include "rtc_base/platform_thread.h" #include "rtc_base/synchronization/mutex.h" +#include "rtc_base/thread_annotations.h" +#include "system_wrappers/include/clock.h" #include "test/layer_filtering_transport.h" #include "test/rtp_file_writer.h" @@ -79,7 +98,8 @@ class VideoAnalyzer : public PacketReceiver, bool SendRtp(ArrayView packet, const PacketOptions& options) override; - bool SendRtcp(ArrayView packet) override; + bool SendRtcp(ArrayView packet, + const PacketOptions& options) override; void OnFrame(const VideoFrame& video_frame) override; void Wait(); diff --git a/video/video_receive_stream2_unittest.cc b/video/video_receive_stream2_unittest.cc index 2feb51bb20..61e18724b6 100644 --- a/video/video_receive_stream2_unittest.cc +++ b/video/video_receive_stream2_unittest.cc @@ -22,6 +22,7 @@ #include #include "absl/memory/memory.h" +#include "api/array_view.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" #include "api/metronome/test/fake_metronome.h" @@ -214,7 +215,9 @@ class VideoReceiveStream2Test : public ::testing::TestWithParam { .WillByDefault(Invoke(&fake_decoder_, &test::FakeDecoder::Release)); ON_CALL(mock_transport_, SendRtcp) .WillByDefault( - Invoke(&rtcp_packet_parser_, &test::RtcpPacketParser::Parse)); + Invoke([this](ArrayView packet, ::testing::Unused) { + return rtcp_packet_parser_.Parse(packet); + })); } ~VideoReceiveStream2Test() override { From 2abf8de0bc5ab83b6abbb9dd9a03556680586f95 Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Fri, 25 Apr 2025 13:13:31 +0000 Subject: [PATCH 007/292] Update freshness of style-guide Remove recommendation to use ABSL_DEPRECATED - absl documentation explicitly recommends deprecated attribute directly for c++14 and above and since absl require minimum c++17, there is high chance the macro will be removed. Drop namespace for the webrtc::ArrayView No-Try: True Bug: b/413607319 Change-Id: I88b7ebf3694dd6a52cf8885952434e4b7416b9c9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/388402 Reviewed-by: Artem Titov Commit-Queue: Artem Titov Auto-Submit: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#44479} --- g3doc/style-guide.md | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/g3doc/style-guide.md b/g3doc/style-guide.md index dd4e2cbcac..6af842dee8 100644 --- a/g3doc/style-guide.md +++ b/g3doc/style-guide.md @@ -1,5 +1,5 @@ - + # WebRTC coding style guide @@ -123,18 +123,12 @@ In other words, rename the existing function, and provide an inline wrapper using the original name that calls it. That way, callers who are willing to call it using the `DEPRECATED_`-prefixed name don't get the warning. -NOTE 3: Occasionally, with long descriptions, `git cl format` will do the wrong -thing with the attribute. In that case, you can use the -[`ABSL_DEPRECATED` macro][ABSL_DEPRECATED], which is formatted in a more -readable way. - [DEPRECATED]: https://en.cppreference.com/w/cpp/language/attributes/deprecated -[ABSL_DEPRECATED]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/abseil-cpp/absl/base/attributes.h?q=ABSL_DEPRECATED [ABSL_DEPRECATE_AND_INLINE]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/abseil-cpp/absl/base/macros.h?q=ABSL_DEPRECATE_AND_INLINE ### ArrayView -When passing an array of values to a function, use `webrtc::ArrayView` +When passing an array of values to a function, use `ArrayView` whenever possible—that is, whenever you're not passing ownership of the array, and don't allow the callee to change the array size. @@ -146,7 +140,7 @@ For example, | `const T* ptr, size_t num_elements` | `ArrayView` | | `T* ptr, size_t num_elements` | `ArrayView` | -See the [source code for `webrtc::ArrayView`](api/array_view.h) for more detailed +See the [source code for `ArrayView`](api/array_view.h) for more detailed docs. ### Strings From f111207b8e9a998e4dfd8431230104a8697e71e8 Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Mon, 28 Apr 2025 19:35:30 +0000 Subject: [PATCH 008/292] Remove AudioDeviceModule::CreateForTest as unused Remove internal TestAudioDeviceModuleImpl as providing no value. Run iwyu (remove dependency on arraysize that is used on win only as it confuses iwyu and is not needed anyway) This is noop cleanup in preparation to provide Environment to AudioDeviceModule at construction Bug: webrtc:413413572 Change-Id: I6c13fd8e12b88bd281031126059e0a622dd4d7b9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/388940 Reviewed-by: Henrik Andreassson Commit-Queue: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#44480} --- api/audio/audio_device.h | 4 -- modules/audio_device/BUILD.gn | 1 + modules/audio_device/audio_device_impl.cc | 11 +++++- modules/audio_device/audio_device_impl.h | 12 ++++-- modules/audio_device/audio_device_unittest.cc | 21 +++++++---- .../audio_device/include/test_audio_device.cc | 37 +++++-------------- webrtc_lib_link_test.cc | 9 +++-- 7 files changed, 49 insertions(+), 46 deletions(-) diff --git a/api/audio/audio_device.h b/api/audio/audio_device.h index b22b021ec4..92eee1c0fa 100644 --- a/api/audio/audio_device.h +++ b/api/audio/audio_device.h @@ -63,10 +63,6 @@ class AudioDeviceModule : public webrtc::RefCountInterface { // Creates a default ADM for usage in production code. static scoped_refptr Create( AudioLayer audio_layer, TaskQueueFactory* task_queue_factory); - // Creates an ADM with support for extra test methods. Don't use this factory - // in production code. - static scoped_refptr CreateForTest( - AudioLayer audio_layer, TaskQueueFactory* task_queue_factory); // Retrieve the currently utilized audio layer virtual int32_t ActiveAudioLayer(AudioLayer* audioLayer) const = 0; diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn index aa4d216b28..2800fcd562 100644 --- a/modules/audio_device/BUILD.gn +++ b/modules/audio_device/BUILD.gn @@ -200,6 +200,7 @@ if (!build_with_chromium) { "../../rtc_base:safe_conversions", "../../rtc_base:timeutils", "../../rtc_base/synchronization:mutex", + "../../rtc_base/system:file_wrapper", "../../rtc_base/task_utils:repeating_task", "//third_party/abseil-cpp/absl/strings:string_view", ] diff --git a/modules/audio_device/audio_device_impl.cc b/modules/audio_device/audio_device_impl.cc index 44cfabeddd..6ba8b0efe6 100644 --- a/modules/audio_device/audio_device_impl.cc +++ b/modules/audio_device/audio_device_impl.cc @@ -12,8 +12,15 @@ #include +#include +#include +#include + +#include "api/audio/audio_device.h" +#include "api/audio/audio_device_defines.h" #include "api/make_ref_counted.h" #include "api/scoped_refptr.h" +#include "api/task_queue/task_queue_factory.h" #include "modules/audio_device/audio_device_config.h" // IWYU pragma: keep #include "modules/audio_device/audio_device_generic.h" #include "rtc_base/checks.h" @@ -66,11 +73,11 @@ scoped_refptr AudioDeviceModule::Create( AudioLayer audio_layer, TaskQueueFactory* task_queue_factory) { RTC_DLOG(LS_INFO) << __FUNCTION__; - return AudioDeviceModule::CreateForTest(audio_layer, task_queue_factory); + return AudioDeviceModuleImpl::Create(audio_layer, task_queue_factory); } // static -scoped_refptr AudioDeviceModule::CreateForTest( +scoped_refptr AudioDeviceModuleImpl::Create( AudioLayer audio_layer, TaskQueueFactory* task_queue_factory) { RTC_DLOG(LS_INFO) << __FUNCTION__; diff --git a/modules/audio_device/audio_device_impl.h b/modules/audio_device/audio_device_impl.h index 41d68209e8..98b11e94c6 100644 --- a/modules/audio_device/audio_device_impl.h +++ b/modules/audio_device/audio_device_impl.h @@ -18,6 +18,8 @@ #include #include "api/audio/audio_device.h" +#include "api/audio/audio_device_defines.h" +#include "api/scoped_refptr.h" #include "api/task_queue/task_queue_factory.h" #include "modules/audio_device/audio_device_buffer.h" @@ -42,9 +44,9 @@ class AudioDeviceModuleImpl : public AudioDeviceModuleForTest { kPlatformFuchsia = 7, }; - int32_t CheckPlatform(); - int32_t CreatePlatformSpecificObjects(); - int32_t AttachAudioBuffer(); + static scoped_refptr Create( + AudioLayer audio_layer, + TaskQueueFactory* task_queue_factory); AudioDeviceModuleImpl(AudioLayer audio_layer, TaskQueueFactory* task_queue_factory); @@ -56,6 +58,10 @@ class AudioDeviceModuleImpl : public AudioDeviceModuleForTest { bool create_detached); ~AudioDeviceModuleImpl() override; + int32_t CheckPlatform(); + int32_t CreatePlatformSpecificObjects(); + int32_t AttachAudioBuffer(); + // Retrieve the currently utilized audio layer int32_t ActiveAudioLayer(AudioLayer* audioLayer) const override; diff --git a/modules/audio_device/audio_device_unittest.cc b/modules/audio_device/audio_device_unittest.cc index 8c97b5f93f..728673ac12 100644 --- a/modules/audio_device/audio_device_unittest.cc +++ b/modules/audio_device/audio_device_unittest.cc @@ -11,21 +11,27 @@ #include "api/audio/audio_device.h" #include +#include +#include #include +#include #include #include #include #include +#include #include "api/array_view.h" +#include "api/audio/audio_device_defines.h" #include "api/scoped_refptr.h" #include "api/sequence_checker.h" #include "api/task_queue/default_task_queue_factory.h" #include "api/task_queue/task_queue_factory.h" +#include "api/units/time_delta.h" #include "modules/audio_device/audio_device_impl.h" #include "modules/audio_device/include/mock_audio_transport.h" -#include "rtc_base/arraysize.h" #include "rtc_base/buffer.h" +#include "rtc_base/checks.h" #include "rtc_base/event.h" #include "rtc_base/logging.h" #include "rtc_base/numerics/safe_conversions.h" @@ -35,6 +41,7 @@ #include "rtc_base/time_utils.h" #include "test/gmock.h" #include "test/gtest.h" + #ifdef WEBRTC_WIN #include "modules/audio_device/include/audio_device_factory.h" #include "modules/audio_device/win/core_audio_utility_win.h" @@ -589,8 +596,8 @@ class MAYBE_AudioDeviceTest // The value of `audio_layer_` is set at construction by GetParam() and two // different layers are tested on Windows only. if (audio_layer_ == AudioDeviceModule::kPlatformDefaultAudio) { - return AudioDeviceModule::CreateForTest(audio_layer_, - task_queue_factory_.get()); + return AudioDeviceModuleImpl::Create(audio_layer_, + task_queue_factory_.get()); } else if (audio_layer_ == AudioDeviceModule::kWindowsCoreAudio2) { #ifdef WEBRTC_WIN // We must initialize the COM library on a thread before we calling any of @@ -827,8 +834,8 @@ TEST_P(MAYBE_AudioDeviceTest, StartStopPlayoutWithRealDevice) { AudioDeviceModule::WindowsDeviceType device_role[] = { AudioDeviceModule::kDefaultDevice, AudioDeviceModule::kDefaultCommunicationDevice}; - for (size_t i = 0; i < arraysize(device_role); ++i) { - EXPECT_EQ(0, audio_device()->SetPlayoutDevice(device_role[i])); + for (AudioDeviceModule::WindowsDeviceType device_type : device_role) { + EXPECT_EQ(0, audio_device()->SetPlayoutDevice(device_type)); StartPlayout(); StopPlayout(); } @@ -854,8 +861,8 @@ TEST_P(MAYBE_AudioDeviceTest, StartStopRecordingWithRealDevice) { AudioDeviceModule::WindowsDeviceType device_role[] = { AudioDeviceModule::kDefaultDevice, AudioDeviceModule::kDefaultCommunicationDevice}; - for (size_t i = 0; i < arraysize(device_role); ++i) { - EXPECT_EQ(0, audio_device()->SetRecordingDevice(device_role[i])); + for (AudioDeviceModule::WindowsDeviceType device_type : device_role) { + EXPECT_EQ(0, audio_device()->SetRecordingDevice(device_type)); StartRecording(); StopRecording(); } diff --git a/modules/audio_device/include/test_audio_device.cc b/modules/audio_device/include/test_audio_device.cc index c945b016ce..a0e0ef87d6 100644 --- a/modules/audio_device/include/test_audio_device.cc +++ b/modules/audio_device/include/test_audio_device.cc @@ -12,29 +12,27 @@ #include #include #include +#include #include #include -#include #include #include #include "absl/strings/string_view.h" #include "api/array_view.h" +#include "api/audio/audio_device.h" #include "api/make_ref_counted.h" +#include "api/scoped_refptr.h" #include "api/task_queue/task_queue_factory.h" #include "common_audio/wav_file.h" #include "modules/audio_device/audio_device_impl.h" -#include "modules/audio_device/include/audio_device_default.h" #include "modules/audio_device/test_audio_device_impl.h" #include "rtc_base/buffer.h" #include "rtc_base/checks.h" -#include "rtc_base/event.h" -#include "rtc_base/logging.h" #include "rtc_base/numerics/safe_conversions.h" -#include "rtc_base/platform_thread.h" #include "rtc_base/random.h" #include "rtc_base/synchronization/mutex.h" -#include "rtc_base/task_utils/repeating_task.h" +#include "rtc_base/system/file_wrapper.h" #include "rtc_base/thread_annotations.h" #include "rtc_base/time_utils.h" @@ -45,25 +43,6 @@ namespace { constexpr int kFrameLengthUs = 10000; constexpr int kFramesPerSecond = kNumMicrosecsPerSec / kFrameLengthUs; -class TestAudioDeviceModuleImpl : public AudioDeviceModuleImpl { - public: - TestAudioDeviceModuleImpl( - TaskQueueFactory* task_queue_factory, - std::unique_ptr capturer, - std::unique_ptr renderer, - float speed = 1) - : AudioDeviceModuleImpl( - AudioLayer::kDummyAudio, - std::make_unique(task_queue_factory, - std::move(capturer), - std::move(renderer), - speed), - task_queue_factory, - /*create_detached=*/true) {} - - ~TestAudioDeviceModuleImpl() override = default; -}; - // A fake capturer that generates pulses with random samples between // -max_amplitude and +max_amplitude. class PulsedNoiseCapturerImpl final @@ -449,8 +428,12 @@ scoped_refptr TestAudioDeviceModule::Create( std::unique_ptr capturer, std::unique_ptr renderer, float speed) { - auto audio_device = make_ref_counted( - task_queue_factory, std::move(capturer), std::move(renderer), speed); + auto audio_device = make_ref_counted( + AudioDeviceModule::AudioLayer::kDummyAudio, + std::make_unique(task_queue_factory, std::move(capturer), + std::move(renderer), speed), + task_queue_factory, + /*create_detached=*/true); // Ensure that the current platform is supported. if (audio_device->CheckPlatform() == -1) { diff --git a/webrtc_lib_link_test.cc b/webrtc_lib_link_test.cc index 4eeb59a57a..97813ca1a8 100644 --- a/webrtc_lib_link_test.cc +++ b/webrtc_lib_link_test.cc @@ -8,8 +8,11 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include +#include +#include + #include "api/audio/audio_device.h" -#include "api/audio/audio_processing.h" #include "api/audio/builtin_audio_processing_builder.h" #include "api/audio_codecs/audio_decoder_factory_template.h" #include "api/audio_codecs/audio_encoder_factory_template.h" @@ -19,7 +22,6 @@ #include "api/enable_media.h" #include "api/peer_connection_interface.h" #include "api/rtc_event_log/rtc_event_log_factory.h" -#include "api/stats/rtcstats_objects.h" #include "api/task_queue/default_task_queue_factory.h" #include "api/video_codecs/video_decoder_factory_template.h" #include "api/video_codecs/video_decoder_factory_template_dav1d_adapter.h" @@ -31,11 +33,12 @@ #include "api/video_codecs/video_encoder_factory_template_libvpx_vp8_adapter.h" #include "api/video_codecs/video_encoder_factory_template_libvpx_vp9_adapter.h" #include "api/video_codecs/video_encoder_factory_template_open_h264_adapter.h" +#include "rtc_base/thread.h" namespace webrtc { void CreateSomeMediaDeps(PeerConnectionFactoryDependencies& media_deps) { - media_deps.adm = AudioDeviceModule::CreateForTest( + media_deps.adm = AudioDeviceModule::Create( AudioDeviceModule::kDummyAudio, media_deps.task_queue_factory.get()); media_deps.audio_encoder_factory = webrtc::CreateAudioEncoderFactory(); From 2ba08292e0f267cb490514fcc7d107000c9d3171 Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Mon, 28 Apr 2025 22:46:53 +0000 Subject: [PATCH 009/292] Cleanup VideoQualityTest helper Prepare Environment to create AudioDeviceModule with in a follow up change. Run include what you use Bug: webrtc:413413572 Change-Id: I56dcace2222b7de898614cea7a3bd5674c643313 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/388980 Commit-Queue: Danil Chapovalov Reviewed-by: Philip Eliasson Cr-Commit-Position: refs/heads/main@{#44481} --- video/BUILD.gn | 24 ++++++++---------------- video/video_quality_test.cc | 31 ++++++++++++++++--------------- video/video_quality_test.h | 29 ++++++++++++++++++++++------- 3 files changed, 46 insertions(+), 38 deletions(-) diff --git a/video/BUILD.gn b/video/BUILD.gn index 68bc3cfca4..c2772aa8a6 100644 --- a/video/BUILD.gn +++ b/video/BUILD.gn @@ -529,31 +529,29 @@ if (rtc_include_tests) { ] deps = [ ":frame_dumping_decoder", - "..//test/network:simulated_network", "../api:array_view", "../api:create_frame_generator", "../api:fec_controller_api", "../api:field_trials_view", "../api:frame_generator_api", - "../api:libjingle_peerconnection_api", "../api:make_ref_counted", "../api:rtc_event_log_output_file", "../api:rtp_parameters", "../api:scoped_refptr", "../api:simulated_network_api", - "../api:test_dependency_factory", "../api:transport_api", "../api:video_quality_test_fixture_api", "../api/audio:audio_device", "../api/audio:builtin_audio_processing_builder", "../api/environment", + "../api/environment:environment_factory", "../api/numerics", "../api/rtc_event_log", "../api/rtc_event_log:rtc_event_log_factory", "../api/task_queue", - "../api/task_queue:default_task_queue_factory", "../api/test/metrics:global_metrics_logger_and_exporter", "../api/test/metrics:metric", + "../api/test/video:function_video_factory", "../api/units:time_delta", "../api/units:timestamp", "../api/video:builtin_video_bitrate_allocator_factory", @@ -563,7 +561,6 @@ if (rtc_include_tests) { "../api/video:video_codec_constants", "../api/video:video_frame", "../api/video:video_frame_type", - "../api/video:video_rtp_headers", "../api/video_codecs:video_codecs_api", "../call:call_interfaces", "../call:fake_network", @@ -574,22 +571,15 @@ if (rtc_include_tests) { "../media:rtc_audio_video", "../media:rtc_internal_video_codecs", "../media:rtc_simulcast_encoder_adapter", - "../modules/audio_device:audio_device_module_from_input_and_output", "../modules/audio_device:test_audio_device_module", - "../modules/audio_device:windows_core_audio_utility", "../modules/audio_mixer:audio_mixer_impl", "../modules/rtp_rtcp", "../modules/rtp_rtcp:rtp_rtcp_format", - "../modules/video_coding", "../modules/video_coding:codec_globals_headers", "../modules/video_coding:video_coding_utility", - "../modules/video_coding:webrtc_h264", - "../modules/video_coding:webrtc_vp8", - "../modules/video_coding:webrtc_vp9", "../rtc_base:checks", "../rtc_base:copy_on_write_buffer", "../rtc_base:logging", - "../rtc_base:macromagic", "../rtc_base:platform_thread", "../rtc_base:rtc_base_tests_utils", "../rtc_base:rtc_event", @@ -603,7 +593,6 @@ if (rtc_include_tests) { "../rtc_base/task_utils:repeating_task", "../system_wrappers", "../test:direct_transport", - "../test:fake_video_codecs", "../test:fileutils", "../test:frame_generator_capturer", "../test:platform_video_capturer", @@ -614,13 +603,12 @@ if (rtc_include_tests) { "../test:test_support", "../test:test_support_test_artifacts", "../test:video_frame_writer", - "../test:video_test_common", "../test:video_test_constants", + "../test/network:simulated_network", "config:encoder_config", "config:streams_config", "//third_party/abseil-cpp/absl/algorithm:container", "//third_party/abseil-cpp/absl/flags:flag", - "//third_party/abseil-cpp/absl/flags:parse", "//third_party/abseil-cpp/absl/strings:string_view", ] @@ -628,7 +616,11 @@ if (rtc_include_tests) { deps += [ "../test:video_test_mac" ] } if (is_win) { - deps += [ "../rtc_base/win:scoped_com_initializer" ] + deps += [ + "../modules/audio_device:audio_device_module_from_input_and_output", + "../modules/audio_device:windows_core_audio_utility", + "../rtc_base/win:scoped_com_initializer", + ] } } diff --git a/video/video_quality_test.cc b/video/video_quality_test.cc index 39494cbe85..a4faa23776 100644 --- a/video/video_quality_test.cc +++ b/video/video_quality_test.cc @@ -19,6 +19,7 @@ #include "absl/flags/flag.h" #include "api/call/transport.h" #include "api/environment/environment.h" +#include "api/environment/environment_factory.h" #include "api/field_trials_view.h" #include "api/make_ref_counted.h" #include "api/rtc_event_log/rtc_event_log.h" @@ -70,7 +71,6 @@ #include "api/audio/builtin_audio_processing_builder.h" #include "api/fec_controller_override.h" #include "api/rtc_event_log_output_file.h" -#include "api/task_queue/default_task_queue_factory.h" #include "api/task_queue/task_queue_base.h" #include "api/test/create_frame_generator.h" #include "api/video/builtin_video_bitrate_allocator_factory.h" @@ -408,8 +408,7 @@ std::unique_ptr VideoQualityTest::CreateVideoEncoder( VideoQualityTest::VideoQualityTest( std::unique_ptr injection_components) - : clock_(Clock::GetRealTimeClock()), - task_queue_factory_(CreateDefaultTaskQueueFactory()), + : env_(CreateEnvironment()), video_decoder_factory_( [this](const Environment& env, const SdpVideoFormat& format) { return this->CreateVideoDecoder(env, format); @@ -1058,11 +1057,11 @@ void VideoQualityTest::SetupThumbnailCapturers(size_t num_thumbnail_streams) { for (size_t i = 0; i < num_thumbnail_streams; ++i) { auto frame_generator_capturer = std::make_unique( - clock_, + &env_.clock(), test::CreateSquareFrameGenerator(static_cast(thumbnail.width), static_cast(thumbnail.height), std::nullopt, std::nullopt), - thumbnail.max_framerate, *task_queue_factory_); + thumbnail.max_framerate, env_.task_queue_factory()); EXPECT_TRUE(frame_generator_capturer->Init()); thumbnail_capturers_.push_back(std::move(frame_generator_capturer)); } @@ -1109,8 +1108,8 @@ VideoQualityTest::CreateFrameGenerator(size_t video_idx) { params_.screenshare[video_idx].slide_change_interval); frame_generator = test::CreateScrollingInputFromYuvFilesFrameGenerator( - clock_, slides, kWidth, kHeight, params_.video[video_idx].width, - params_.video[video_idx].height, + &env_.clock(), slides, kWidth, kHeight, + params_.video[video_idx].width, params_.video[video_idx].height, params_.screenshare[video_idx].scroll_duration * 1000, kPauseDurationMs); } @@ -1170,8 +1169,8 @@ void VideoQualityTest::CreateCapturers() { ASSERT_TRUE(frame_generator); auto frame_generator_capturer = std::make_unique( - clock_, std::move(frame_generator), params_.video[video_idx].fps, - *task_queue_factory_); + &env_.clock(), std::move(frame_generator), + params_.video[video_idx].fps, env_.task_queue_factory()); EXPECT_TRUE(frame_generator_capturer->Init()); video_sources_[video_idx] = std::move(frame_generator_capturer); } @@ -1207,7 +1206,8 @@ VideoQualityTest::CreateSendTransport() { } return std::make_unique( task_queue(), - std::make_unique(clock_, std::move(network_behavior)), + std::make_unique(&env_.clock(), + std::move(network_behavior)), sender_call_.get(), test::VideoTestConstants::kPayloadTypeVP8, test::VideoTestConstants::kPayloadTypeVP9, params_.video[0].selected_tl, params_.ss[0].selected_sl, payload_type_map_, @@ -1227,7 +1227,8 @@ VideoQualityTest::CreateReceiveTransport() { } return std::make_unique( task_queue(), - std::make_unique(clock_, std::move(network_behavior)), + std::make_unique(&env_.clock(), + std::move(network_behavior)), receiver_call_.get(), payload_type_map_, GetRegisteredExtensions(), GetRegisteredExtensions()); } @@ -1306,7 +1307,7 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) { test::VideoTestConstants::kSendRtxSsrcs[params_.ss[0].selected_stream], static_cast(params_.ss[0].selected_stream), params.ss[0].selected_sl, params_.video[0].selected_tl, - is_quick_test_enabled, clock_, params_.logging.rtp_dump_name, + is_quick_test_enabled, &env_.clock(), params_.logging.rtp_dump_name, task_queue()); SendTask(task_queue(), [&]() { @@ -1386,11 +1387,11 @@ scoped_refptr VideoQualityTest::CreateAudioDevice() { RTC_CHECK(com_initializer_->Succeeded()); RTC_CHECK(webrtc_win::core_audio_utility::IsSupported()); RTC_CHECK(webrtc_win::core_audio_utility::IsMMCSSSupported()); - return CreateWindowsCoreAudioAudioDeviceModule(task_queue_factory_.get()); + return CreateWindowsCoreAudioAudioDeviceModule(&env_.task_queue_factory()); #else // Use legacy factory method on all platforms except Windows. return AudioDeviceModule::Create(AudioDeviceModule::kPlatformDefaultAudio, - task_queue_factory_.get()); + &env_.task_queue_factory()); #endif } @@ -1405,7 +1406,7 @@ void VideoQualityTest::InitializeAudioDevice(CallConfig* send_call_config, } else { // By default, create a test ADM which fakes audio. audio_device = TestAudioDeviceModule::Create( - task_queue_factory_.get(), + &env_.task_queue_factory(), TestAudioDeviceModule::CreatePulsedNoiseCapturer(32000, 48000), TestAudioDeviceModule::CreateDiscardRenderer(48000), 1.f); } diff --git a/video/video_quality_test.h b/video/video_quality_test.h index 4afe3977dc..ce9260cd0b 100644 --- a/video/video_quality_test.h +++ b/video/video_quality_test.h @@ -10,25 +10,41 @@ #ifndef VIDEO_VIDEO_QUALITY_TEST_H_ #define VIDEO_VIDEO_QUALITY_TEST_H_ +#include +#include #include #include #include #include +#include "api/audio/audio_device.h" +#include "api/call/transport.h" #include "api/environment/environment.h" -#include "api/fec_controller.h" #include "api/rtc_event_log/rtc_event_log_factory.h" -#include "api/task_queue/task_queue_base.h" -#include "api/task_queue/task_queue_factory.h" +#include "api/scoped_refptr.h" #include "api/test/frame_generator_interface.h" +#include "api/test/video/function_video_decoder_factory.h" +#include "api/test/video/function_video_encoder_factory.h" #include "api/test/video_quality_test_fixture.h" #include "api/video/video_bitrate_allocator_factory.h" +#include "api/video/video_frame.h" +#include "api/video/video_source_interface.h" +#include "api/video_codecs/sdp_video_format.h" +#include "api/video_codecs/video_codec.h" +#include "api/video_codecs/video_decoder.h" +#include "api/video_codecs/video_decoder_factory.h" +#include "api/video_codecs/video_encoder.h" +#include "api/video_codecs/video_encoder_factory.h" +#include "call/call_config.h" #include "call/fake_network_pipe.h" -#include "media/engine/internal_decoder_factory.h" -#include "media/engine/internal_encoder_factory.h" +#include "call/video_receive_stream.h" +#include "call/video_send_stream.h" #include "test/call_test.h" +#include "test/direct_transport.h" #include "test/layer_filtering_transport.h" +#include "video/config/video_encoder_config.h" #include "video/video_analyzer.h" + #ifdef WEBRTC_WIN #include "modules/audio_device/win/core_audio_utility_win.h" #include "rtc_base/win/scoped_com_initializer.h" @@ -103,10 +119,9 @@ class VideoQualityTest : public test::CallTest, virtual std::unique_ptr CreateSendTransport(); virtual std::unique_ptr CreateReceiveTransport(); + const Environment env_; std::vector>> thumbnail_capturers_; - Clock* const clock_; - const std::unique_ptr task_queue_factory_; RtcEventLogFactory rtc_event_log_factory_; test::FunctionVideoDecoderFactory video_decoder_factory_; From 1c178c07a2bf0d4f65144712a14ce448c6a4b002 Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Tue, 29 Apr 2025 11:59:27 +0000 Subject: [PATCH 010/292] Restore missing build dependency It was introduced due to CL race https://webrtc-review.googlesource.com/c/src/+/388980 removed build dependency that https://webrtc-review.googlesource.com/c/src/+/387120 added dependency on No-Try: True Bug: b/414557926 Change-Id: I061bc084d69e93aa6d4aa7168aa2b8d4315b18f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389020 Reviewed-by: Sameer Vijaykar Auto-Submit: Danil Chapovalov Reviewed-by: Harald Alvestrand Commit-Queue: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44482} --- video/BUILD.gn | 1 + 1 file changed, 1 insertion(+) diff --git a/video/BUILD.gn b/video/BUILD.gn index c2772aa8a6..7bf330143f 100644 --- a/video/BUILD.gn +++ b/video/BUILD.gn @@ -580,6 +580,7 @@ if (rtc_include_tests) { "../rtc_base:checks", "../rtc_base:copy_on_write_buffer", "../rtc_base:logging", + "../rtc_base:macromagic", "../rtc_base:platform_thread", "../rtc_base:rtc_base_tests_utils", "../rtc_base:rtc_event", From 3d7cbac0fe7f23d41f57361d66af4f36b85de1fa Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Wed, 23 Apr 2025 09:01:20 -0700 Subject: [PATCH 011/292] audio: remove neon shadowing variables MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit also add a IWYU keep pragma to prevent incorrect removal a include file (when IWYU is run in an Android checkout) BUG=webrtc:42223409 Change-Id: I513dfa3cd3ef71362c759ab1c003ee9c12529f6d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/387922 Reviewed-by: Gustaf Ullberg Reviewed-by: Per Åhgren Commit-Queue: Philipp Hancke Cr-Commit-Position: refs/heads/main@{#44483} --- .../audio_processing/aec3/matched_filter.cc | 19 +++++++++---------- 1 file changed, 9 insertions(+), 10 deletions(-) diff --git a/modules/audio_processing/aec3/matched_filter.cc b/modules/audio_processing/aec3/matched_filter.cc index 90d550a255..866f941697 100644 --- a/modules/audio_processing/aec3/matched_filter.cc +++ b/modules/audio_processing/aec3/matched_filter.cc @@ -13,7 +13,7 @@ #include // Defines WEBRTC_ARCH_X86_FAMILY, used below. -#include "rtc_base/system/arch.h" +#include "rtc_base/system/arch.h" // IWYU pragma: keep #if defined(WEBRTC_HAS_NEON) #include @@ -118,7 +118,7 @@ void MatchedFilterCoreWithAccumulatedError_NEON( } const float* x_p = chunk1 != h_size ? scratch_memory.data() : &x[x_start_index]; - const float* h_p = &h[0]; + const float* h_cp = &h[0]; float* accumulated_error_p = &accumulated_error[0]; // Initialize values for the accumulation. float32x4_t x2_sum_128 = vdupq_n_f32(0); @@ -127,10 +127,10 @@ void MatchedFilterCoreWithAccumulatedError_NEON( // Perform 128 bit vector operations. const int limit_by_4 = h_size >> 2; for (int k = limit_by_4; k > 0; - --k, h_p += 4, x_p += 4, accumulated_error_p++) { + --k, h_cp += 4, x_p += 4, accumulated_error_p++) { // Load the data into 128 bit vectors. const float32x4_t x_k = vld1q_f32(x_p); - const float32x4_t h_k = vld1q_f32(h_p); + const float32x4_t h_k = vld1q_f32(h_cp); // Compute and accumulate x * x. x2_sum_128 = vmlaq_f32(x2_sum_128, x_k, x_k); // Compute x * h @@ -154,7 +154,6 @@ void MatchedFilterCoreWithAccumulatedError_NEON( float* h_p = &h[0]; x_p = chunk1 != h_size ? scratch_memory.data() : &x[x_start_index]; // Perform 128 bit vector operations. - const int limit_by_4 = h_size >> 2; for (int k = limit_by_4; k > 0; --k, h_p += 4, x_p += 4) { // Load the data into 128 bit vectors. float32x4_t h_k = vld1q_f32(h_p); @@ -197,7 +196,7 @@ void MatchedFilterCore_NEON(size_t x_start_index, RTC_DCHECK_GT(x_size, x_start_index); const float* x_p = &x[x_start_index]; - const float* h_p = &h[0]; + const float* h_cp = &h[0]; // Initialize values for the accumulation. float32x4_t s_128 = vdupq_n_f32(0); @@ -215,20 +214,20 @@ void MatchedFilterCore_NEON(size_t x_start_index, for (int limit : {chunk1, chunk2}) { // Perform 128 bit vector operations. const int limit_by_4 = limit >> 2; - for (int k = limit_by_4; k > 0; --k, h_p += 4, x_p += 4) { + for (int k = limit_by_4; k > 0; --k, h_cp += 4, x_p += 4) { // Load the data into 128 bit vectors. const float32x4_t x_k = vld1q_f32(x_p); - const float32x4_t h_k = vld1q_f32(h_p); + const float32x4_t h_k = vld1q_f32(h_cp); // Compute and accumulate x * x and h * x. x2_sum_128 = vmlaq_f32(x2_sum_128, x_k, x_k); s_128 = vmlaq_f32(s_128, h_k, x_k); } // Perform non-vector operations for any remaining items. - for (int k = limit - limit_by_4 * 4; k > 0; --k, ++h_p, ++x_p) { + for (int k = limit - limit_by_4 * 4; k > 0; --k, ++h_cp, ++x_p) { const float x_k = *x_p; x2_sum += x_k * x_k; - s += *h_p * x_k; + s += *h_cp * x_k; } x_p = &x[0]; From 7912c7b8b02580340f56bbcf9b659b6c4546a5fc Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Tue, 29 Apr 2025 19:02:40 -0700 Subject: [PATCH 012/292] Roll chromium_revision ae6ec80ac0..c5d386d663 (1452664:1453677) Change log: https://chromium.googlesource.com/chromium/src/+log/ae6ec80ac0..c5d386d663 Full diff: https://chromium.googlesource.com/chromium/src/+/ae6ec80ac0..c5d386d663 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/32f65d20af..21c29fc96c * src/build: https://chromium.googlesource.com/chromium/src/build/+log/0345a25a86..54a242f2d5 * src/buildtools: https://chromium.googlesource.com/chromium/src/buildtools/+log/0f32cb9025..2bf618c359 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/8d3354e85c..8ff1667439 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/42f6eefe76..3291a34ffb * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/7e53f288d9..fdb5d002fe * src/third_party/android_build_tools/error_prone/cipd: -IrkxgAkj3RxGNBPp6b4r9OWZC6_dTbI-jC3c3UPgq8C..2t_u1XEjof609BQE4czlCQzEqRliKS_qY35FOu_VckgC * src/third_party/android_build_tools/lint/cipd: gx4NrAApWUVG5HG3WJRIIzZrTssD6H5uGbKe0g77mucC..TEDRB0F7VgCgyepd8FsIxPRqvKjvWQ_YkLV-ybx-GQYC * src/third_party/android_build_tools/manifest_merger/cipd: HhXknpcPosMUDHZQD-8Ogve-HaG4HmJ4p082zKKAcnMC..fPGWiT0LVG2_ifxOpHb5KNwTL-_7lb7tYOsB_ZoIiskC * src/third_party/androidx/cipd: cxEyTzVT7uuNifSPlUiuKlb4tSo35_YQjBbMVmgZeCwC..JBCFmzXl0Hn5K4GTl49NmYclGryCKT35FQAYn8HHbGIC * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/34492c89a8..a70ec8dd2c * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/000f47cfa3..083971e6b0 * src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/57213f125d..cf8cac9f64 * src/third_party/freetype/src: https://chromium.googlesource.com/chromium/src/third_party/freetype2.git/+log/1da283b8ae..79912716e7 * src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/b10387fdbb..9c1ffdc7fb * src/third_party/googletest/src: https://chromium.googlesource.com/external/github.com/google/googletest.git/+log/cd430b47a5..59c924bc47 * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/97de5579d0..5106a2a7c6 * src/third_party/kotlin_stdlib/cipd: WkqHVVoDtiOfLxiPT-eO3zyieJRoIRrvs7XqgcIXBYIC..KFgAvgfE4Ssv7PMRYGJVo89rhJCPGQitmhEh39RETFgC * src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/a23a4799ec..719f60edc5 * src/third_party/libc++/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git/+log/917609c669..62724eb55f * src/third_party/libvpx/source/libvpx: https://chromium.googlesource.com/webm/libvpx.git/+log/ff1d193f4b..b32e98f52f * src/third_party/libyuv: https://chromium.googlesource.com/libyuv/libyuv.git/+log/1e40e34573..9f9b5cf660 * src/third_party/llvm-build/Release+Asserts_version: Linux_x64/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clang-win-runtime-library-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/llvmobjdump-llvmorg-21-init-6681-g5b36835d-1.tar.xz..Linux_x64/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clang-win-runtime-library-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/llvmobjdump-llvmorg-21-init-9266-g09006611-1.tar.xz * src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/912274164f..c43b02cfdb * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/220d71b5a8..f308c40f9c * src/third_party/r8/d8/cipd: 6qLey8EBp9ivhThnqVPWy2ZDGpsf5Y29EsTbi_rZ1pMC..bA3htCoEd_EArHekDGQSNpmBzQrcby2ioG6SFyl3AtwC * src/third_party/rust-toolchain_version: Linux_x64/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Mac/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Mac_arm64/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Win/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz..Linux_x64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Mac/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Mac_arm64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Win/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/48eea1f5a0..4eb583ccf2 * src/tools/luci-go: git_revision:fd48b8efe637ea5f431fc371f1617b215b7ba1a2..git_revision:94e82b5e2ea91e24ae4f237a5b9e1f365ae80d68 * src/tools/luci-go: git_revision:fd48b8efe637ea5f431fc371f1617b215b7ba1a2..git_revision:94e82b5e2ea91e24ae4f237a5b9e1f365ae80d68 * src/tools/luci-go: git_revision:fd48b8efe637ea5f431fc371f1617b215b7ba1a2..git_revision:94e82b5e2ea91e24ae4f237a5b9e1f365ae80d68 DEPS diff: https://chromium.googlesource.com/chromium/src/+/ae6ec80ac0..c5d386d663/DEPS No update to Clang. BUG=None Change-Id: I560e2f41c96c1faa8038845e066aebe8528f93ad Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/388905 Commit-Queue: Autoroller Bot-Commit: Autoroller Cr-Commit-Position: refs/heads/main@{#44484} --- DEPS | 60 ++++++++++++++++++++++++++++++------------------------------ 1 file changed, 30 insertions(+), 30 deletions(-) diff --git a/DEPS b/DEPS index fb05d990fd..7fff663bb6 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': 'ae6ec80ac0f466a1f5d1960fa19aca853e944139', + 'chromium_revision': 'c5d386d663279bd27fde52ab88d54a3cb577fd95', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,24 +73,24 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@32f65d20af167d22990abb0aa28dbe4f3e65aa94', + 'https://chromium.googlesource.com/chromium/src/base@21c29fc96cd4f11cef7215683362cf812e464917', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@0345a25a8624b38c14f600f86a73d9a44a9cd772', + 'https://chromium.googlesource.com/chromium/src/build@54a242f2d53a68c2002e5b9e9f8ded597804c3bb', 'src/buildtools': - 'https://chromium.googlesource.com/chromium/src/buildtools@0f32cb9025766951122d4ed19aba87a94ded3f43', + 'https://chromium.googlesource.com/chromium/src/buildtools@2bf618c359b513ad97a286e09c5089944e698ed1', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. 'src/examples/androidtests/third_party/gradle': { 'url': 'https://chromium.googlesource.com/external/github.com/gradle/gradle.git@f2d1fb54a951d8b11d25748e4711bec8d128d7e3', 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@8d3354e85c432a482a21167eb5de0872e46b32f7', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@8ff16674396ba753ffb0712caccd66c4e1d14533', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@42f6eefe763bd5bf601f8caf87f98c9b0bdce10f', + 'https://chromium.googlesource.com/chromium/src/testing@3291a34ffb7203c906217cce3c04a5737e61e903', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@7e53f288d97da8a861506cacb8ed30444459e88c', + 'https://chromium.googlesource.com/chromium/src/third_party@fdb5d002fe914834a1f3a21daf91bf5f8e5a4d74', 'src/buildtools/linux64': { 'packages': [ @@ -323,13 +323,13 @@ deps = { 'src/third_party/clang-format/script': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@37f6e68a107df43b7d7e044fd36a13cbae3413f2', 'src/third_party/compiler-rt/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@57213f125d03209892fed26189feb3b736e96735', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@cf8cac9f64ff4ac6fd2041a699dd86f717c63586', 'src/third_party/libc++/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@917609c669e43edc850eeb192a342434a54e1dfd', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@62724eb55f2ad7389088265952b1bb62cb89c73b', 'src/third_party/libc++abi/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@f2a7f2987f9dcdf8b04c2d8cd4dcb186641a7c3e', 'src/third_party/llvm-libc/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@912274164f0877ca917c06e8484ad3be1784833a', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@c43b02cfdbd4a75e9ba233f3b5bfd8322e3bf1d8', 'src/third_party/libunwind/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@81e2cb40a70de2b6978e6d8658891ded9a77f7e3', @@ -427,7 +427,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_build_tools/error_prone', - 'version': '-IrkxgAkj3RxGNBPp6b4r9OWZC6_dTbI-jC3c3UPgq8C', + 'version': '2t_u1XEjof609BQE4czlCQzEqRliKS_qY35FOu_VckgC', }, ], 'condition': 'checkout_android', @@ -449,7 +449,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_build_tools/lint', - 'version': 'gx4NrAApWUVG5HG3WJRIIzZrTssD6H5uGbKe0g77mucC', + 'version': 'TEDRB0F7VgCgyepd8FsIxPRqvKjvWQ_YkLV-ybx-GQYC', }, ], 'condition': 'checkout_android and non_git_source', @@ -480,11 +480,11 @@ deps = { }, 'src/third_party/boringssl/src': - 'https://boringssl.googlesource.com/boringssl.git@34492c89a8e381e0e856a686cc71b1eb5bd728db', + 'https://boringssl.googlesource.com/boringssl.git@a70ec8dd2cc1816e9deadda2f6e536640380e20c', 'src/third_party/breakpad/breakpad': 'https://chromium.googlesource.com/breakpad/breakpad.git@232a723f5096ab02d53d87931efa485fa77d3b03', 'src/third_party/catapult': - 'https://chromium.googlesource.com/catapult.git@000f47cfa393d7f9557025a252862e2a61a60d44', + 'https://chromium.googlesource.com/catapult.git@083971e6b02dd794b3cb84d01033455889be7976', 'src/third_party/ced/src': { 'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5', }, @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@97de5579d06e7ded32eee4a622501dd01145a064', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@5106a2a7c6d26b94288b8230b53c8e92008dd259', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -511,7 +511,7 @@ deps = { 'condition': 'checkout_linux', }, 'src/third_party/freetype/src': - 'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@1da283b8ae6d6b94f34a5c4b8c1227adc9dbb1d8', + 'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@79912716e783b4c9e47c35907ae080e2ec02fbc6', 'src/third_party/harfbuzz-ng/src': 'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@9f83bbbe64654b45ba5bb06927ff36c2e7588495', 'src/third_party/google_benchmark/src': { @@ -525,7 +525,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/googletest/src': - 'https://chromium.googlesource.com/external/github.com/google/googletest.git@cd430b47a54841ec45d64d2377d7cabaf0eba610', + 'https://chromium.googlesource.com/external/github.com/google/googletest.git@59c924bc471cefea25a7cf7eb40b6c101b170b12', 'src/third_party/icu': { 'url': 'https://chromium.googlesource.com/chromium/deps/icu.git@4c8cc4b365a505ce35be1e0bd488476c5f79805d', }, @@ -563,7 +563,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/kotlin_stdlib', - 'version': 'WkqHVVoDtiOfLxiPT-eO3zyieJRoIRrvs7XqgcIXBYIC', + 'version': 'KFgAvgfE4Ssv7PMRYGJVo89rhJCPGQitmhEh39RETFgC', }, ], 'condition': 'checkout_android', @@ -583,7 +583,7 @@ deps = { 'src/third_party/libFuzzer/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@e31b99917861f891308269c36a32363b120126bb', 'src/third_party/fuzztest/src': - 'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@b10387fdbbca18192f85eaa5323a59f44bf9c468', + 'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@9c1ffdc7fb3d8db64e42c7a9662d8b9906625b77', 'src/third_party/libjpeg_turbo': 'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@e14cbfaa85529d47f9f55b0f104a579c1061f9ad', 'src/third_party/libsrtp': @@ -591,19 +591,19 @@ deps = { 'src/third_party/dav1d/libdav1d': 'https://chromium.googlesource.com/external/github.com/videolan/dav1d.git@8d956180934f16244bdb58b39175824775125e55', 'src/third_party/libaom/source/libaom': - 'https://aomedia.googlesource.com/aom.git@a23a4799ec2d7dd6e436c7b64a34553773014ed7', + 'https://aomedia.googlesource.com/aom.git@719f60edc51b6141a2434bf1b5110c2fb075b246', 'src/third_party/libunwindstack': { 'url': 'https://chromium.googlesource.com/chromium/src/third_party/libunwindstack.git@0d758dd57f42564acecdd7a1e7ac5c8521c1b01a', 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '220d71b5a8e54a558cb10e70518c8b22b2c57b46', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'f308c40f9c8b1905f1dacfff857d175a0b77060a', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': - 'https://chromium.googlesource.com/webm/libvpx.git@ff1d193f4b9dfa9b2ced51efbb6ec7a69e58e88c', + 'https://chromium.googlesource.com/webm/libvpx.git@b32e98f52fb99e00bc99ea8f2b57a8015b71aa39', 'src/third_party/libyuv': - 'https://chromium.googlesource.com/libyuv/libyuv.git@1e40e34573c3861480d107cd4a4ce290df79951f', + 'https://chromium.googlesource.com/libyuv/libyuv.git@9f9b5cf660dcfa0d3fdee41cf4ffbe4bb6e95114', 'src/third_party/lss': { 'url': 'https://chromium.googlesource.com/linux-syscall-support.git@ed31caa60f20a4f6569883b2d752ef7522de51e0', 'condition': 'checkout_android or checkout_linux', @@ -645,7 +645,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/r8', - 'version': '6qLey8EBp9ivhThnqVPWy2ZDGpsf5Y29EsTbi_rZ1pMC', + 'version': 'bA3htCoEd_EArHekDGQSNpmBzQrcby2ioG6SFyl3AtwC', }, ], 'condition': 'checkout_android', @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@48eea1f5a02d31e4712c13cc349d953314180d77', + 'https://chromium.googlesource.com/chromium/src/tools@4eb583ccf263251865f5b7bf1c6ce3f86c52a384', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'cxEyTzVT7uuNifSPlUiuKlb4tSo35_YQjBbMVmgZeCwC', + 'version': 'JBCFmzXl0Hn5K4GTl49NmYclGryCKT35FQAYn8HHbGIC', }, ], 'condition': 'checkout_android and non_git_source', @@ -706,7 +706,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_build_tools/manifest_merger', - 'version': 'HhXknpcPosMUDHZQD-8Ogve-HaG4HmJ4p082zKKAcnMC', + 'version': 'fPGWiT0LVG2_ifxOpHb5KNwTL-_7lb7tYOsB_ZoIiskC', }, ], 'condition': 'checkout_android', @@ -793,15 +793,15 @@ deps = { 'packages': [ { 'package': 'infra/tools/luci/cas/${{platform}}', - 'version': 'git_revision:fd48b8efe637ea5f431fc371f1617b215b7ba1a2', + 'version': 'git_revision:94e82b5e2ea91e24ae4f237a5b9e1f365ae80d68', }, { 'package': 'infra/tools/luci/isolate/${{platform}}', - 'version': 'git_revision:fd48b8efe637ea5f431fc371f1617b215b7ba1a2', + 'version': 'git_revision:94e82b5e2ea91e24ae4f237a5b9e1f365ae80d68', }, { 'package': 'infra/tools/luci/swarming/${{platform}}', - 'version': 'git_revision:fd48b8efe637ea5f431fc371f1617b215b7ba1a2', + 'version': 'git_revision:94e82b5e2ea91e24ae4f237a5b9e1f365ae80d68', } ], 'dep_type': 'cipd', From 8c0213aae7c699f7e118bf3a8608d2465e0fd6a1 Mon Sep 17 00:00:00 2001 From: webrtc-version-updater Date: Tue, 29 Apr 2025 21:09:14 -0700 Subject: [PATCH 013/292] Update WebRTC code version (2025-04-30T04:08:55). Bug: None Change-Id: If8809ee65dcacb8c4c6fb38214334ba7166975d5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/388871 Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Cr-Commit-Position: refs/heads/main@{#44485} --- call/version.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/call/version.cc b/call/version.cc index 8a5bed828e..e1d5ae0a44 100644 --- a/call/version.cc +++ b/call/version.cc @@ -13,7 +13,7 @@ namespace webrtc { // The timestamp is always in UTC. -const char* const kSourceTimestamp = "WebRTC source stamp 2025-04-29T04:08:55"; +const char* const kSourceTimestamp = "WebRTC source stamp 2025-04-30T04:08:55"; void LoadWebRTCVersionInRegister() { // Using volatile to instruct the compiler to not optimize `p` away even From 179a6207008d9d114e525c42c9dcb4c98b0c8735 Mon Sep 17 00:00:00 2001 From: Victor Boivie Date: Tue, 1 Apr 2025 12:31:53 +0200 Subject: [PATCH 014/292] dcsctp: Add pull mode to consume received messages The current API delivers received message by a callback API. The caller is expected to process the received message directly, and the socket will immediately restore the advertised receiver window as there is nothing buffered in the socket any longer. But it may be buffered in the application, if it can't process messages in the same rate as they are received. How can you as a receiver tell the sender to stop sending for a while, because it can't keep up with the pace? It's not possible - the callback is triggered unconditionally. In other words: The advertised receiver window is of little use for implementing back-pressure, which it's intending to do. This CL adds a new option, that might become the default, that replaces the delivery callback with a new "signal" callback, just indicating to the caller that there are received messages that can be consumed by calling a new API on the socket. If not, the message statys buffered and it's reflected in the advertised receiver window. Note that as a back-pressure algorithm, it's quite simple. The sender must make sure that it doesn't send more than the receiver can fit within its receiver window, but it's also allowed to always have a single MTU inflight. So the sender will never stop sending completely, but will definitely go down in send rate. Depending on application, it might be a good idea to ALSO implement a more sophisticated back-pressure algorithm by e.g. sending backpressure messages on a separate stream that has a higher priority. But this CL provides the fallback basics that the SCTP protocol provides. No-Iwyu: Base CL not clean. Needs separate cleaning. Co-authored-by: Nathan Lewis Bug: webrtc:407622662 Change-Id: I9a63ee117a0fcf0ccb8409287b6a0548f9bcb7b1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/383780 Reviewed-by: Harald Alvestrand Commit-Queue: Victor Boivie Cr-Commit-Position: refs/heads/main@{#44486} --- net/dcsctp/public/dcsctp_options.h | 6 ++ net/dcsctp/public/dcsctp_socket.h | 24 +++++- net/dcsctp/public/mock_dcsctp_socket.h | 3 + net/dcsctp/rx/reassembly_queue.cc | 11 ++- net/dcsctp/rx/reassembly_queue.h | 18 +++-- net/dcsctp/rx/reassembly_queue_test.cc | 62 +++++++------- net/dcsctp/socket/callback_deferrer.cc | 9 +++ net/dcsctp/socket/callback_deferrer.h | 1 + net/dcsctp/socket/dcsctp_socket.cc | 57 ++++++++++--- net/dcsctp/socket/dcsctp_socket.h | 2 + net/dcsctp/socket/dcsctp_socket_test.cc | 29 +++++++ .../socket/mock_dcsctp_socket_callbacks.h | 1 + .../socket/stream_reset_handler_test.cc | 81 +++++++++++-------- 13 files changed, 219 insertions(+), 85 deletions(-) diff --git a/net/dcsctp/public/dcsctp_options.h b/net/dcsctp/public/dcsctp_options.h index 6e1181f7e2..f5a9aac907 100644 --- a/net/dcsctp/public/dcsctp_options.h +++ b/net/dcsctp/public/dcsctp_options.h @@ -86,6 +86,12 @@ struct DcSctpOptions { // buffer is fully utilized. size_t max_receiver_window_buffer_size = 5 * 1024 * 1024; + // Enables receive pull mode - `DcSctpCallbacks::OnMessageReady` will be + // called when there are messages ready to be read instead of + // `DcSctpCallbacks::OnMessageReceived`. It is up to the + // caller to call `DcSctpSocket::GetNextMessage()` to receive the messages. + bool enable_receive_pull_mode = false; + // Send queue total size limit. It will not be possible to queue more data if // the queue size is larger than this number. size_t max_send_buffer_size = 2'000'000; diff --git a/net/dcsctp/public/dcsctp_socket.h b/net/dcsctp/public/dcsctp_socket.h index bce886fbbf..803bf8efe5 100644 --- a/net/dcsctp/public/dcsctp_socket.h +++ b/net/dcsctp/public/dcsctp_socket.h @@ -360,11 +360,19 @@ class DcSctpSocketCallbacks { virtual void NotifyOutgoingMessageBufferEmpty() {} // Called when the library has received an SCTP message in full and delivers - // it to the upper layer. + // it to the upper layer, given that `DcSctpOptions::enable_receive_pull_mode` + // isn't enabled. // // It is allowed to call into this library from within this callback. virtual void OnMessageReceived(DcSctpMessage message) = 0; + // Called when `DcSctpOptions::enable_receive_pull_mode` is enabled and the + // library has one or more SCTP messages ready to be received with + // `DcSctpSocket::GetNextMessage()`. + // + // It is allowed to call into this library from within this callback. + virtual void OnMessageReady() {} + // Triggered when an non-fatal error is reported by either this library or // from the other peer (by sending an ERROR command). These should be logged, // but no other action need to be taken as the association is still viable. @@ -525,6 +533,20 @@ class DcSctpSocketInterface { // To be called when an incoming SCTP packet is to be processed. virtual void ReceivePacket(webrtc::ArrayView data) = 0; + // Returns the number of received messages that can be retrieved by calling + // calling `::GetNextMessage`. + // + // Only used when `DcSctpOptions::enable_receive_pull_mode` is true (will + // always return zero if not enabled). + virtual size_t MessagesReady() const = 0; + + // To be called after `DcSctpSocketCallbacks::OnMessagesReady` has triggered + // to retrieve the next message, if any. + // + // Only used when `DcSctpOptions::enable_receive_pull_mode` is true (will + // always return empty if not enabled). + virtual std::optional GetNextMessage() = 0; + // To be called when a timeout has expired. The `timeout_id` is provided // when the timeout was initiated. virtual void HandleTimeout(TimeoutID timeout_id) = 0; diff --git a/net/dcsctp/public/mock_dcsctp_socket.h b/net/dcsctp/public/mock_dcsctp_socket.h index c1c59fea32..399a193464 100644 --- a/net/dcsctp/public/mock_dcsctp_socket.h +++ b/net/dcsctp/public/mock_dcsctp_socket.h @@ -24,6 +24,9 @@ class MockDcSctpSocket : public DcSctpSocketInterface { (webrtc::ArrayView data), (override)); + MOCK_METHOD(size_t, MessagesReady, (), (const, override)); + MOCK_METHOD(std::optional, GetNextMessage, (), (override)); + MOCK_METHOD(void, HandleTimeout, (TimeoutID timeout_id), (override)); MOCK_METHOD(void, Connect, (), (override)); diff --git a/net/dcsctp/rx/reassembly_queue.cc b/net/dcsctp/rx/reassembly_queue.cc index 439863a6cb..3d3c5a4205 100644 --- a/net/dcsctp/rx/reassembly_queue.cc +++ b/net/dcsctp/rx/reassembly_queue.cc @@ -158,9 +158,13 @@ void ReassemblyQueue::EnterDeferredReset( RTC_DCHECK(IsConsistent()); } -std::vector ReassemblyQueue::FlushMessages() { - std::vector ret; - reassembled_messages_.swap(ret); +std::optional ReassemblyQueue::GetNextMessage() { + if (reassembled_messages_.empty()) { + return std::nullopt; + } + DcSctpMessage ret = std::move(reassembled_messages_.front()); + reassembled_messages_.pop_front(); + queued_bytes_ -= ret.payload().size(); return ret; } @@ -177,6 +181,7 @@ void ReassemblyQueue::AddReassembledMessage( << ", ppid=" << *message.ppid() << ", payload=" << message.payload().size() << " bytes"; + queued_bytes_ += message.payload().size(); reassembled_messages_.emplace_back(std::move(message)); } diff --git a/net/dcsctp/rx/reassembly_queue.h b/net/dcsctp/rx/reassembly_queue.h index 7308fdd227..1cf4cb90e9 100644 --- a/net/dcsctp/rx/reassembly_queue.h +++ b/net/dcsctp/rx/reassembly_queue.h @@ -13,6 +13,7 @@ #include #include +#include #include #include #include @@ -79,11 +80,16 @@ class ReassemblyQueue { void Add(TSN tsn, Data data); // Indicates if the reassembly queue has any reassembled messages that can be - // retrieved by calling `FlushMessages`. + // retrieved by calling `GetNextMessage`. bool HasMessages() const { return !reassembled_messages_.empty(); } - // Returns any reassembled messages. - std::vector FlushMessages(); + // Returns the number of reassembled messages that are ready to be retrieved + // by calling `GetNextMessage`. + size_t MessagesReady() const { return reassembled_messages_.size(); } + + // Returns the next reassembled message or nullopt if there are no messages + // ready. + std::optional GetNextMessage(); // Handle a ForwardTSN chunk, when the sender has indicated that the received // (this class) should forget about some chunks. This is used to implement @@ -147,9 +153,9 @@ class ReassemblyQueue { const size_t watermark_bytes_; UnwrappedTSN::Unwrapper tsn_unwrapper_; - // Messages that have been reassembled, and will be returned by - // `FlushMessages`. - std::vector reassembled_messages_; + // Messages that have been reassembled, and will be consumed from by + // `GetNextMessage`. + std::deque reassembled_messages_; // If present, "deferred reset processing" mode is active. std::optional deferred_reset_streams_; diff --git a/net/dcsctp/rx/reassembly_queue_test.cc b/net/dcsctp/rx/reassembly_queue_test.cc index dc15aceb08..e619f403c8 100644 --- a/net/dcsctp/rx/reassembly_queue_test.cc +++ b/net/dcsctp/rx/reassembly_queue_test.cc @@ -74,6 +74,15 @@ MATCHER_P3(SctpMessageIs, stream_id, ppid, expected_payload, "") { return true; } +std::vector FlushMessages(ReassemblyQueue& reasm) { + std::vector messages; + while (reasm.HasMessages()) { + messages.emplace_back(reasm.GetNextMessage().value()); + } + EXPECT_EQ(reasm.queued_bytes(), 0u); + return messages; +} + class ReassemblyQueueTest : public testing::Test { protected: ReassemblyQueueTest() {} @@ -90,9 +99,8 @@ TEST_F(ReassemblyQueueTest, SingleUnorderedChunkMessage) { ReassemblyQueue reasm("log: ", kBufferSize); reasm.Add(TSN(10), gen_.Unordered({1, 2, 3, 4}, "BE")); EXPECT_TRUE(reasm.HasMessages()); - EXPECT_THAT(reasm.FlushMessages(), + EXPECT_THAT(FlushMessages(reasm), ElementsAre(SctpMessageIs(kStreamID, kPPID, kShortPayload))); - EXPECT_EQ(reasm.queued_bytes(), 0u); } TEST_F(ReassemblyQueueTest, LargeUnorderedChunkAllPermutations) { @@ -114,9 +122,8 @@ TEST_F(ReassemblyQueueTest, LargeUnorderedChunkAllPermutations) { EXPECT_FALSE(reasm.HasMessages()); } else { EXPECT_TRUE(reasm.HasMessages()); - EXPECT_THAT(reasm.FlushMessages(), + EXPECT_THAT(FlushMessages(reasm), ElementsAre(SctpMessageIs(kStreamID, kPPID, kLongPayload))); - EXPECT_EQ(reasm.queued_bytes(), 0u); } } } while (std::next_permutation(std::begin(tsns), std::end(tsns))); @@ -125,9 +132,9 @@ TEST_F(ReassemblyQueueTest, LargeUnorderedChunkAllPermutations) { TEST_F(ReassemblyQueueTest, SingleOrderedChunkMessage) { ReassemblyQueue reasm("log: ", kBufferSize); reasm.Add(TSN(10), gen_.Ordered({1, 2, 3, 4}, "BE")); - EXPECT_EQ(reasm.queued_bytes(), 0u); + EXPECT_EQ(reasm.queued_bytes(), 4u); EXPECT_TRUE(reasm.HasMessages()); - EXPECT_THAT(reasm.FlushMessages(), + EXPECT_THAT(FlushMessages(reasm), ElementsAre(SctpMessageIs(kStreamID, kPPID, kShortPayload))); } @@ -148,12 +155,11 @@ TEST_F(ReassemblyQueueTest, ManySmallOrderedMessages) { is_beginning, is_end, IsUnordered(false))); } EXPECT_THAT( - reasm.FlushMessages(), + FlushMessages(reasm), ElementsAre(SctpMessageIs(kStreamID, kPPID, payload.subview(0, 4)), SctpMessageIs(kStreamID, kPPID, payload.subview(4, 4)), SctpMessageIs(kStreamID, kPPID, payload.subview(8, 4)), SctpMessageIs(kStreamID, kPPID, payload.subview(12, 4)))); - EXPECT_EQ(reasm.queued_bytes(), 0u); } while (std::next_permutation(std::begin(tsns), std::end(tsns))); } @@ -177,9 +183,8 @@ TEST_F(ReassemblyQueueTest, RetransmissionInLargeOrdered) { reasm.Add(TSN(20), gen_.Ordered({11, 12, 13, 14, 15, 16}, "E")); EXPECT_TRUE(reasm.HasMessages()); - EXPECT_THAT(reasm.FlushMessages(), + EXPECT_THAT(FlushMessages(reasm), ElementsAre(SctpMessageIs(kStreamID, kPPID, kLongPayload))); - EXPECT_EQ(reasm.queued_bytes(), 0u); } TEST_F(ReassemblyQueueTest, ForwardTSNRemoveUnordered) { @@ -201,7 +206,7 @@ TEST_F(ReassemblyQueueTest, ForwardTSNRemoveUnordered) { // The second lost chunk comes, message is assembled. reasm.Add(TSN(16), gen_.Unordered({7})); EXPECT_TRUE(reasm.HasMessages()); - EXPECT_EQ(reasm.queued_bytes(), 0u); + EXPECT_EQ(reasm.queued_bytes(), 4u); } TEST_F(ReassemblyQueueTest, ForwardTSNRemoveOrdered) { @@ -220,11 +225,11 @@ TEST_F(ReassemblyQueueTest, ForwardTSNRemoveOrdered) { reasm.HandleForwardTsn( TSN(13), std::vector({SkippedStream(kStreamID, kSSN)})); - EXPECT_EQ(reasm.queued_bytes(), 0u); + EXPECT_EQ(reasm.queued_bytes(), 4u); // The lost chunk comes, but too late. EXPECT_TRUE(reasm.HasMessages()); - EXPECT_THAT(reasm.FlushMessages(), + EXPECT_THAT(FlushMessages(reasm), ElementsAre(SctpMessageIs(kStreamID, kPPID, kMessage2Payload))); } @@ -244,11 +249,11 @@ TEST_F(ReassemblyQueueTest, ForwardTSNRemoveALotOrdered) { reasm.HandleForwardTsn( TSN(13), std::vector({SkippedStream(kStreamID, kSSN)})); - EXPECT_EQ(reasm.queued_bytes(), 0u); + EXPECT_EQ(reasm.queued_bytes(), 4u); // The lost chunk comes, but too late. EXPECT_TRUE(reasm.HasMessages()); - EXPECT_THAT(reasm.FlushMessages(), + EXPECT_THAT(FlushMessages(reasm), ElementsAre(SctpMessageIs(kStreamID, kPPID, kMessage2Payload))); } @@ -256,7 +261,7 @@ TEST_F(ReassemblyQueueTest, NotReadyForHandoverWhenResetStreamIsDeferred) { ReassemblyQueue reasm("log: ", kBufferSize); reasm.Add(TSN(10), gen_.Ordered({1, 2, 3, 4}, "BE", {.mid = MID(0)})); reasm.Add(TSN(11), gen_.Ordered({1, 2, 3, 4}, "BE", {.mid = MID(1)})); - EXPECT_THAT(reasm.FlushMessages(), SizeIs(2)); + EXPECT_THAT(FlushMessages(reasm), SizeIs(2)); reasm.EnterDeferredReset(TSN(12), std::vector({StreamID(1)})); EXPECT_EQ(reasm.GetHandoverReadiness(), @@ -281,13 +286,14 @@ TEST_F(ReassemblyQueueTest, HandoverInInitialState) { reasm2.RestoreFromState(state); reasm2.Add(TSN(10), gen_.Ordered({1, 2, 3, 4}, "BE")); - EXPECT_THAT(reasm2.FlushMessages(), SizeIs(1)); + EXPECT_THAT(FlushMessages(reasm2), SizeIs(1)); } TEST_F(ReassemblyQueueTest, HandoverAfterHavingAssembedOneMessage) { ReassemblyQueue reasm1("log: ", kBufferSize); reasm1.Add(TSN(10), gen_.Ordered({1, 2, 3, 4}, "BE")); - EXPECT_THAT(reasm1.FlushMessages(), SizeIs(1)); + EXPECT_TRUE(reasm1.GetNextMessage().has_value()); + EXPECT_FALSE(reasm1.HasMessages()); EXPECT_EQ(reasm1.GetHandoverReadiness(), HandoverReadinessStatus()); DcSctpSocketHandoverState state; @@ -298,7 +304,8 @@ TEST_F(ReassemblyQueueTest, HandoverAfterHavingAssembedOneMessage) { reasm2.RestoreFromState(state); reasm2.Add(TSN(11), gen_.Ordered({1, 2, 3, 4}, "BE")); - EXPECT_THAT(reasm2.FlushMessages(), SizeIs(1)); + EXPECT_TRUE(reasm2.GetNextMessage().has_value()); + EXPECT_FALSE(reasm2.HasMessages()); } TEST_F(ReassemblyQueueTest, SingleUnorderedChunkMessageInRfc8260) { @@ -307,9 +314,9 @@ TEST_F(ReassemblyQueueTest, SingleUnorderedChunkMessageInRfc8260) { reasm.Add(TSN(10), Data(StreamID(1), SSN(0), MID(0), FSN(0), kPPID, {1, 2, 3, 4}, Data::IsBeginning(true), Data::IsEnd(true), IsUnordered(true))); - EXPECT_EQ(reasm.queued_bytes(), 0u); + EXPECT_EQ(reasm.queued_bytes(), 4u); EXPECT_TRUE(reasm.HasMessages()); - EXPECT_THAT(reasm.FlushMessages(), + EXPECT_THAT(FlushMessages(reasm), ElementsAre(SctpMessageIs(kStreamID, kPPID, kShortPayload))); } @@ -326,13 +333,13 @@ TEST_F(ReassemblyQueueTest, TwoInterleavedChunks) { reasm.Add(TSN(12), Data(StreamID(1), SSN(0), MID(0), FSN(1), kPPID, {5, 6, 7, 8}, Data::IsBeginning(false), Data::IsEnd(true), IsUnordered(true))); - EXPECT_EQ(reasm.queued_bytes(), 4u); + EXPECT_EQ(reasm.queued_bytes(), 12u); reasm.Add(TSN(13), Data(StreamID(2), SSN(0), MID(0), FSN(1), kPPID, {13, 14, 15, 16}, Data::IsBeginning(false), Data::IsEnd(true), IsUnordered(true))); - EXPECT_EQ(reasm.queued_bytes(), 0u); + EXPECT_EQ(reasm.queued_bytes(), 16u); EXPECT_TRUE(reasm.HasMessages()); - EXPECT_THAT(reasm.FlushMessages(), + EXPECT_THAT(FlushMessages(reasm), ElementsAre(SctpMessageIs(StreamID(1), kPPID, kMediumPayload1), SctpMessageIs(StreamID(2), kPPID, kMediumPayload2))); } @@ -356,11 +363,10 @@ TEST_F(ReassemblyQueueTest, UnorderedInterleavedMessagesAllPermutations) { is_beginning, is_end, IsUnordered(true))); } EXPECT_TRUE(reasm.HasMessages()); - EXPECT_THAT(reasm.FlushMessages(), + EXPECT_THAT(FlushMessages(reasm), UnorderedElementsAre( SctpMessageIs(StreamID(1), kPPID, kSixBytePayload), SctpMessageIs(StreamID(2), kPPID, kSixBytePayload))); - EXPECT_EQ(reasm.queued_bytes(), 0u); } while (std::next_permutation(std::begin(indexes), std::end(indexes))); } @@ -381,11 +387,11 @@ TEST_F(ReassemblyQueueTest, IForwardTSNRemoveALotOrdered) { reasm.HandleForwardTsn(TSN(13), std::vector({SkippedStream( IsUnordered(false), kStreamID, MID(0))})); - EXPECT_EQ(reasm.queued_bytes(), 0u); + EXPECT_EQ(reasm.queued_bytes(), 4u); // The lost chunk comes, but too late. ASSERT_TRUE(reasm.HasMessages()); - EXPECT_THAT(reasm.FlushMessages(), + EXPECT_THAT(FlushMessages(reasm), ElementsAre(SctpMessageIs(kStreamID, kPPID, kMessage2Payload))); } diff --git a/net/dcsctp/socket/callback_deferrer.cc b/net/dcsctp/socket/callback_deferrer.cc index a17ba75d9f..dcd54a9a91 100644 --- a/net/dcsctp/socket/callback_deferrer.cc +++ b/net/dcsctp/socket/callback_deferrer.cc @@ -69,6 +69,15 @@ void CallbackDeferrer::OnMessageReceived(DcSctpMessage message) { std::move(message)); } +void CallbackDeferrer::OnMessageReady() { + RTC_DCHECK(prepared_); + deferred_.emplace_back( + +[](CallbackData data, DcSctpSocketCallbacks& cb) { + return cb.OnMessageReady(); + }, + std::monostate{}); +} + void CallbackDeferrer::OnError(ErrorKind error, absl::string_view message) { RTC_DCHECK(prepared_); deferred_.emplace_back( diff --git a/net/dcsctp/socket/callback_deferrer.h b/net/dcsctp/socket/callback_deferrer.h index d7cb0db2a3..1b0d23d394 100644 --- a/net/dcsctp/socket/callback_deferrer.h +++ b/net/dcsctp/socket/callback_deferrer.h @@ -69,6 +69,7 @@ class CallbackDeferrer : public DcSctpSocketCallbacks { webrtc::Timestamp Now() override { return underlying_.Now(); } uint32_t GetRandomInt(uint32_t low, uint32_t high) override; void OnMessageReceived(DcSctpMessage message) override; + void OnMessageReady() override; void OnError(ErrorKind error, absl::string_view message) override; void OnAborted(ErrorKind error, absl::string_view message) override; void OnConnected() override; diff --git a/net/dcsctp/socket/dcsctp_socket.cc b/net/dcsctp/socket/dcsctp_socket.cc index 6c889cb482..eac23dfeb2 100644 --- a/net/dcsctp/socket/dcsctp_socket.cc +++ b/net/dcsctp/socket/dcsctp_socket.cc @@ -224,7 +224,8 @@ std::string DcSctpSocket::log_prefix() const { } bool DcSctpSocket::IsConsistent() const { - if (tcb_ != nullptr && tcb_->reassembly_queue().HasMessages()) { + if (tcb_ != nullptr && (!options_.enable_receive_pull_mode && + tcb_->reassembly_queue().HasMessages())) { return false; } switch (state_) { @@ -1082,15 +1083,24 @@ void DcSctpSocket::HandleDataCommon(AnyDataChunk& chunk) { << tcb_->reassembly_queue().is_above_watermark(); if (tcb_->reassembly_queue().is_full()) { - // If the reassembly queue is full, there is nothing that can be done. The - // specification only allows dropping gap-ack-blocks, and that's not - // likely to help as the socket has been trying to fill gaps since the - // watermark was reached. - packet_sender_.Send(tcb_->PacketBuilder().Add(AbortChunk( - true, Parameters::Builder().Add(OutOfResourceErrorCause()).Build()))); - InternalClose(ErrorKind::kResourceExhaustion, - "Reassembly Queue is exhausted"); - return; + if (tcb_->reassembly_queue().HasMessages()) { + // If the reassembly queue is full but there are assembled messages + // waiting to be pulled, we can't do anything with this data except drop + // it, and hope the upper layer drains the accumulated messages soon. + RTC_DLOG(LS_VERBOSE) << log_prefix() + << "Rejected data because of full reassembly queue"; + return; + } else { + // If the reassembly queue is full and there's no messages waiting, there + // is nothing that can be done. The specification only allows dropping + // gap-ack-blocks, and that's not likely to help as the socket has been + // trying to fill gaps since the watermark was reached. + packet_sender_.Send(tcb_->PacketBuilder().Add(AbortChunk( + true, Parameters::Builder().Add(OutOfResourceErrorCause()).Build()))); + InternalClose(ErrorKind::kResourceExhaustion, + "Reassembly Queue is exhausted"); + return; + } } if (tcb_->reassembly_queue().is_above_watermark()) { @@ -1472,10 +1482,33 @@ void DcSctpSocket::HandleCookieAck( } void DcSctpSocket::MaybeDeliverMessages() { - for (auto& message : tcb_->reassembly_queue().FlushMessages()) { + if (options_.enable_receive_pull_mode) { + if (tcb_->reassembly_queue().HasMessages()) { + callbacks_.OnMessageReady(); + } + return; + } + + while (std::optional message = + tcb_->reassembly_queue().GetNextMessage()) { + ++metrics_.rx_messages_count; + callbacks_.OnMessageReceived(*std::move(message)); + } +} + +size_t DcSctpSocket::MessagesReady() const { + return tcb_ != nullptr ? tcb_->reassembly_queue().MessagesReady() : 0; +} + +std::optional DcSctpSocket::GetNextMessage() { + if (tcb_ == nullptr) { + return std::nullopt; + } + std::optional ret = tcb_->reassembly_queue().GetNextMessage(); + if (ret.has_value()) { ++metrics_.rx_messages_count; - callbacks_.OnMessageReceived(std::move(message)); } + return ret; } void DcSctpSocket::HandleSack(const CommonHeader& /* header */, diff --git a/net/dcsctp/socket/dcsctp_socket.h b/net/dcsctp/socket/dcsctp_socket.h index d68711ea1f..18435e4aa6 100644 --- a/net/dcsctp/socket/dcsctp_socket.h +++ b/net/dcsctp/socket/dcsctp_socket.h @@ -84,6 +84,8 @@ class DcSctpSocket : public DcSctpSocketInterface { // Implementation of `DcSctpSocketInterface`. void ReceivePacket(webrtc::ArrayView data) override; + size_t MessagesReady() const override; + std::optional GetNextMessage() override; void HandleTimeout(TimeoutID timeout_id) override; void Connect() override; void RestoreFromState(const DcSctpSocketHandoverState& state) override; diff --git a/net/dcsctp/socket/dcsctp_socket_test.cc b/net/dcsctp/socket/dcsctp_socket_test.cc index d9c1c5768a..9e0ea0f8c5 100644 --- a/net/dcsctp/socket/dcsctp_socket_test.cc +++ b/net/dcsctp/socket/dcsctp_socket_test.cc @@ -810,6 +810,35 @@ TEST(DcSctpSocketTest, SendMessageAfterEstablished) { std::optional msg = z.cb.ConsumeReceivedMessage(); ASSERT_TRUE(msg.has_value()); EXPECT_EQ(msg->stream_id(), StreamID(1)); + + // Calling the pull-mode API with it not enabled just returns false. + EXPECT_EQ(z.socket.MessagesReady(), 0u); + EXPECT_FALSE(z.socket.GetNextMessage().has_value()); +} + +TEST(DcSctpSocketTest, SendMessageAfterEstablishedInPullMode) { + SocketUnderTest a("A"); + SocketUnderTest z("Z", {.enable_receive_pull_mode = true}); + + EXPECT_CALL(z.cb, OnMessageReceived).Times(0); + EXPECT_CALL(z.cb, OnMessageReady).Times(1); + + ConnectSockets(a, z); + + EXPECT_EQ(z.socket.MessagesReady(), 0u); + EXPECT_FALSE(z.socket.GetNextMessage().has_value()); + + a.socket.Send(DcSctpMessage(StreamID(1), PPID(53), {1, 2}), kSendOptions); + z.socket.ReceivePacket(a.cb.ConsumeSentPacket()); + + // Not delivered by callback. + ASSERT_FALSE(z.cb.ConsumeReceivedMessage().has_value()); + + // But available by polling. + EXPECT_EQ(z.socket.MessagesReady(), 1u); + std::optional msg = z.socket.GetNextMessage(); + EXPECT_TRUE(msg.has_value()); + EXPECT_EQ(msg->stream_id(), StreamID(1)); } TEST_P(DcSctpSocketParametrizedTest, TimeoutResendsPacket) { diff --git a/net/dcsctp/socket/mock_dcsctp_socket_callbacks.h b/net/dcsctp/socket/mock_dcsctp_socket_callbacks.h index 5dddf08d9c..9094c73814 100644 --- a/net/dcsctp/socket/mock_dcsctp_socket_callbacks.h +++ b/net/dcsctp/socket/mock_dcsctp_socket_callbacks.h @@ -100,6 +100,7 @@ class MockDcSctpSocketCallbacks : public DcSctpSocketCallbacks { } MOCK_METHOD(void, OnMessageReceived, (DcSctpMessage message), (override)); + MOCK_METHOD(void, OnMessageReady, (), (override)); MOCK_METHOD(void, OnError, (ErrorKind error, absl::string_view message), diff --git a/net/dcsctp/socket/stream_reset_handler_test.cc b/net/dcsctp/socket/stream_reset_handler_test.cc index 7b624a88d3..716eb9a649 100644 --- a/net/dcsctp/socket/stream_reset_handler_test.cc +++ b/net/dcsctp/socket/stream_reset_handler_test.cc @@ -44,6 +44,7 @@ namespace dcsctp { namespace { using ::testing::IsEmpty; using ::testing::NiceMock; +using ::testing::Optional; using ::testing::Property; using ::testing::Return; using ::testing::SizeIs; @@ -252,14 +253,15 @@ TEST_F(StreamResetHandlerTest, FailToDeliverWithoutResettingStream) { data_tracker_->Observe(kPeerInitialTsn); data_tracker_->Observe(AddTo(kPeerInitialTsn, 1)); - EXPECT_THAT(reasm_->FlushMessages(), - UnorderedElementsAre( - SctpMessageIs(StreamID(1), PPID(53), kShortPayload), - SctpMessageIs(StreamID(1), PPID(53), kShortPayload))); + EXPECT_THAT(reasm_->GetNextMessage(), + Optional(SctpMessageIs(StreamID(1), PPID(53), kShortPayload))); + EXPECT_THAT(reasm_->GetNextMessage(), + Optional(SctpMessageIs(StreamID(1), PPID(53), kShortPayload))); + EXPECT_FALSE(reasm_->HasMessages()); gen_.ResetStream(); reasm_->Add(AddTo(kPeerInitialTsn, 2), gen_.Ordered({1, 2, 3, 4}, "BE")); - EXPECT_THAT(reasm_->FlushMessages(), IsEmpty()); + EXPECT_FALSE(reasm_->HasMessages()); } TEST_F(StreamResetHandlerTest, ResetStreamsNotDeferred) { @@ -268,10 +270,11 @@ TEST_F(StreamResetHandlerTest, ResetStreamsNotDeferred) { data_tracker_->Observe(kPeerInitialTsn); data_tracker_->Observe(AddTo(kPeerInitialTsn, 1)); - EXPECT_THAT(reasm_->FlushMessages(), - UnorderedElementsAre( - SctpMessageIs(StreamID(1), PPID(53), kShortPayload), - SctpMessageIs(StreamID(1), PPID(53), kShortPayload))); + EXPECT_THAT(reasm_->GetNextMessage(), + Optional(SctpMessageIs(StreamID(1), PPID(53), kShortPayload))); + EXPECT_THAT(reasm_->GetNextMessage(), + Optional(SctpMessageIs(StreamID(1), PPID(53), kShortPayload))); + EXPECT_FALSE(reasm_->HasMessages()); Parameters::Builder builder; builder.Add(OutgoingSSNResetRequestParameter( @@ -285,9 +288,9 @@ TEST_F(StreamResetHandlerTest, ResetStreamsNotDeferred) { gen_.ResetStream(); reasm_->Add(AddTo(kPeerInitialTsn, 2), gen_.Ordered({1, 2, 3, 4}, "BE")); - EXPECT_THAT(reasm_->FlushMessages(), - UnorderedElementsAre( - SctpMessageIs(StreamID(1), PPID(53), kShortPayload))); + EXPECT_THAT(reasm_->GetNextMessage(), + Optional(SctpMessageIs(StreamID(1), PPID(53), kShortPayload))); + EXPECT_FALSE(reasm_->HasMessages()); } TEST_F(StreamResetHandlerTest, ResetStreamsDeferred) { @@ -298,10 +301,11 @@ TEST_F(StreamResetHandlerTest, ResetStreamsDeferred) { data_tracker_->Observe(TSN(11)); reasm_->Add(TSN(11), gen_.Ordered({1, 2, 3, 4}, "BE", {.mid = MID(1)})); - EXPECT_THAT( - reasm_->FlushMessages(), - UnorderedElementsAre(SctpMessageIs(kStreamId, PPID(53), kShortPayload), - SctpMessageIs(kStreamId, PPID(53), kShortPayload))); + EXPECT_THAT(reasm_->GetNextMessage(), + Optional(SctpMessageIs(kStreamId, PPID(53), kShortPayload))); + EXPECT_THAT(reasm_->GetNextMessage(), + Optional(SctpMessageIs(kStreamId, PPID(53), kShortPayload))); + EXPECT_FALSE(reasm_->HasMessages()); Parameters::Builder builder; builder.Add(OutgoingSSNResetRequestParameter( @@ -332,12 +336,15 @@ TEST_F(StreamResetHandlerTest, ResetStreamsDeferred) { ElementsAre(Property(&ReconfigurationResponseParameter::result, ResponseResult::kSuccessPerformed))); - EXPECT_THAT( - reasm_->FlushMessages(), - UnorderedElementsAre(SctpMessageIs(kStreamId, PPID(2), kShortPayload), - SctpMessageIs(kStreamId, PPID(3), kShortPayload), - SctpMessageIs(kStreamId, PPID(4), kShortPayload), - SctpMessageIs(kStreamId, PPID(5), kShortPayload))); + EXPECT_THAT(reasm_->GetNextMessage(), + Optional(SctpMessageIs(kStreamId, PPID(2), kShortPayload))); + EXPECT_THAT(reasm_->GetNextMessage(), + Optional(SctpMessageIs(kStreamId, PPID(3), kShortPayload))); + EXPECT_THAT(reasm_->GetNextMessage(), + Optional(SctpMessageIs(kStreamId, PPID(4), kShortPayload))); + EXPECT_THAT(reasm_->GetNextMessage(), + Optional(SctpMessageIs(kStreamId, PPID(5), kShortPayload))); + EXPECT_FALSE(reasm_->HasMessages()); } TEST_F(StreamResetHandlerTest, ResetStreamsDeferredOnlySelectedStreams) { @@ -396,12 +403,15 @@ TEST_F(StreamResetHandlerTest, ResetStreamsDeferredOnlySelectedStreams) { .mid = MID(1), .ppid = PPID(1006)})); - EXPECT_THAT(reasm_->FlushMessages(), - UnorderedElementsAre( - SctpMessageIs(StreamID(1), PPID(1001), kShortPayload), - SctpMessageIs(StreamID(2), PPID(1002), kShortPayload), - SctpMessageIs(StreamID(3), PPID(1003), kShortPayload), - SctpMessageIs(StreamID(3), PPID(1006), kShortPayload))); + EXPECT_THAT(reasm_->GetNextMessage(), + Optional(SctpMessageIs(StreamID(1), PPID(1001), kShortPayload))); + EXPECT_THAT(reasm_->GetNextMessage(), + Optional(SctpMessageIs(StreamID(2), PPID(1002), kShortPayload))); + EXPECT_THAT(reasm_->GetNextMessage(), + Optional(SctpMessageIs(StreamID(3), PPID(1003), kShortPayload))); + EXPECT_THAT(reasm_->GetNextMessage(), + Optional(SctpMessageIs(StreamID(3), PPID(1006), kShortPayload))); + EXPECT_FALSE(reasm_->HasMessages()); builder.Add(OutgoingSSNResetRequestParameter(ReconfigRequestSN(11), ReconfigRequestSN(3), TSN(13), @@ -410,10 +420,11 @@ TEST_F(StreamResetHandlerTest, ResetStreamsDeferredOnlySelectedStreams) { ElementsAre(Property(&ReconfigurationResponseParameter::result, ResponseResult::kSuccessPerformed))); - EXPECT_THAT(reasm_->FlushMessages(), - UnorderedElementsAre( - SctpMessageIs(StreamID(1), PPID(1004), kShortPayload), - SctpMessageIs(StreamID(2), PPID(1005), kShortPayload))); + EXPECT_THAT(reasm_->GetNextMessage(), + Optional(SctpMessageIs(StreamID(1), PPID(1004), kShortPayload))); + EXPECT_THAT(reasm_->GetNextMessage(), + Optional(SctpMessageIs(StreamID(2), PPID(1005), kShortPayload))); + EXPECT_FALSE(reasm_->HasMessages()); } TEST_F(StreamResetHandlerTest, ResetStreamsDefersForwardTsn) { @@ -469,9 +480,9 @@ TEST_F(StreamResetHandlerTest, ResetStreamsDefersForwardTsn) { ElementsAre(Property(&ReconfigurationResponseParameter::result, ResponseResult::kSuccessPerformed))); - EXPECT_THAT(reasm_->FlushMessages(), - UnorderedElementsAre( - SctpMessageIs(kStreamId, PPID(1005), kShortPayload))); + EXPECT_THAT(reasm_->GetNextMessage(), + Optional(SctpMessageIs(kStreamId, PPID(1005), kShortPayload))); + EXPECT_FALSE(reasm_->HasMessages()); } TEST_F(StreamResetHandlerTest, SendOutgoingRequestDirectly) { From 753b02bc45155b1c7418bab5732b73c44e6edded Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Tue, 29 Apr 2025 15:34:16 +0000 Subject: [PATCH 015/292] Introduce CreateAudioDeviceModule function This factory take Environment instead of a task queue factory and thus can provide field trials to AudioDeviceModule at construction in addition to the task queue factory Additionally, the new factory function keep declaration and definition of the factory function in the same build rule and thus avoid subtle linking errors Bug: webrtc:413413572 Change-Id: I2e9dd342e85dfd07347dbbb615a07265b5774024 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/388880 Commit-Queue: Danil Chapovalov Reviewed-by: Henrik Andreassson Cr-Commit-Position: refs/heads/main@{#44487} --- BUILD.gn | 1 + api/audio/BUILD.gn | 17 ++++++++++ api/audio/audio_device.h | 1 + api/audio/create_audio_device_module.cc | 27 ++++++++++++++++ api/audio/create_audio_device_module.h | 27 ++++++++++++++++ media/BUILD.gn | 5 +++ media/engine/webrtc_voice_engine.cc | 8 +++-- modules/audio_device/BUILD.gn | 5 +++ modules/audio_device/audio_device_impl.cc | 13 ++++++-- modules/audio_device/audio_device_impl.h | 10 +++++- modules/audio_device/audio_device_unittest.cc | 31 +++++++++---------- video/BUILD.gn | 1 + video/video_quality_test.cc | 5 +-- webrtc_lib_link_test.cc | 11 ++++--- 14 files changed, 132 insertions(+), 30 deletions(-) create mode 100644 api/audio/create_audio_device_module.cc create mode 100644 api/audio/create_audio_device_module.h diff --git a/BUILD.gn b/BUILD.gn index ca8d8faa61..bb56514e98 100644 --- a/BUILD.gn +++ b/BUILD.gn @@ -537,6 +537,7 @@ if (!build_with_chromium) { "api:libjingle_peerconnection_api", "api:rtc_error", "api:transport_api", + "api/audio:create_audio_device_module", "api/audio_codecs:opus_audio_decoder_factory", "api/crypto", "api/rtc_event_log:rtc_event_log_factory", diff --git a/api/audio/BUILD.gn b/api/audio/BUILD.gn index 13e8d1f82c..5683e4a372 100644 --- a/api/audio/BUILD.gn +++ b/api/audio/BUILD.gn @@ -23,6 +23,23 @@ rtc_source_set("audio_device") { ] } +if (rtc_include_internal_audio_device) { + rtc_library("create_audio_device_module") { + visibility = [ "*" ] + sources = [ + "create_audio_device_module.cc", + "create_audio_device_module.h", + ] + deps = [ + ":audio_device", + "..:scoped_refptr", + "../../modules/audio_device:audio_device_impl", + "../environment", + "//third_party/abseil-cpp/absl/base:nullability", + ] + } +} + rtc_library("audio_frame_api") { visibility = [ "*" ] sources = [ diff --git a/api/audio/audio_device.h b/api/audio/audio_device.h index 92eee1c0fa..e6df56dc9a 100644 --- a/api/audio/audio_device.h +++ b/api/audio/audio_device.h @@ -61,6 +61,7 @@ class AudioDeviceModule : public webrtc::RefCountInterface { public: // Creates a default ADM for usage in production code. + [[deprecated("Use CreateAudioDeviceModule instead")]] static scoped_refptr Create( AudioLayer audio_layer, TaskQueueFactory* task_queue_factory); diff --git a/api/audio/create_audio_device_module.cc b/api/audio/create_audio_device_module.cc new file mode 100644 index 0000000000..26cb6c12d8 --- /dev/null +++ b/api/audio/create_audio_device_module.cc @@ -0,0 +1,27 @@ +/* + * Copyright (c) 2025 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "api/audio/create_audio_device_module.h" + +#include "absl/base/nullability.h" +#include "api/audio/audio_device.h" +#include "api/environment/environment.h" +#include "api/scoped_refptr.h" +#include "modules/audio_device/audio_device_impl.h" + +namespace webrtc { + +absl_nullable scoped_refptr CreateAudioDeviceModule( + const Environment& env, + AudioDeviceModule::AudioLayer audio_layer) { + return AudioDeviceModuleImpl::Create(env, audio_layer); +} + +} // namespace webrtc diff --git a/api/audio/create_audio_device_module.h b/api/audio/create_audio_device_module.h new file mode 100644 index 0000000000..b86873d16a --- /dev/null +++ b/api/audio/create_audio_device_module.h @@ -0,0 +1,27 @@ +/* + * Copyright (c) 2025 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef API_AUDIO_CREATE_AUDIO_DEVICE_MODULE_H_ +#define API_AUDIO_CREATE_AUDIO_DEVICE_MODULE_H_ + +#include "absl/base/nullability.h" +#include "api/audio/audio_device.h" +#include "api/environment/environment.h" +#include "api/scoped_refptr.h" + +namespace webrtc { + +absl_nullable scoped_refptr CreateAudioDeviceModule( + const Environment& env, + AudioDeviceModule::AudioLayer audio_layer); + +} // namespace webrtc + +#endif // API_AUDIO_CREATE_AUDIO_DEVICE_MODULE_H_ diff --git a/media/BUILD.gn b/media/BUILD.gn index 8042a7ce34..3386c58c22 100644 --- a/media/BUILD.gn +++ b/media/BUILD.gn @@ -681,6 +681,11 @@ rtc_library("rtc_audio_video") { public_configs += [ ":rtc_media_defines_config" ] deps += [ "../modules/video_capture:video_capture_internal_impl" ] } + + if (rtc_include_internal_audio_device) { + deps += [ "../api/audio:create_audio_device_module" ] + } + if (rtc_enable_protobuf) { deps += [ "../modules/audio_coding:ana_config_proto", diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc index cb48226931..2d397c6008 100644 --- a/media/engine/webrtc_voice_engine.cc +++ b/media/engine/webrtc_voice_engine.cc @@ -112,6 +112,10 @@ #endif +#if defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE) +#include "api/audio/create_audio_device_module.h" +#endif + namespace webrtc { namespace { @@ -520,8 +524,8 @@ void WebRtcVoiceEngine::Init() { #if defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE) // No ADM supplied? Create a default one. if (!adm_) { - adm_ = AudioDeviceModule::Create(AudioDeviceModule::kPlatformDefaultAudio, - &env_.task_queue_factory()); + adm_ = + CreateAudioDeviceModule(env_, AudioDeviceModule::kPlatformDefaultAudio); } #endif // WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE RTC_CHECK(adm()); diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn index 2800fcd562..d527ef6b24 100644 --- a/modules/audio_device/BUILD.gn +++ b/modules/audio_device/BUILD.gn @@ -264,6 +264,7 @@ rtc_library("audio_device_impl") { "../../api:scoped_refptr", "../../api:sequence_checker", "../../api/audio:audio_device", + "../../api/environment", "../../api/task_queue", "../../api/units:time_delta", "../../common_audio", @@ -287,6 +288,7 @@ rtc_library("audio_device_impl") { "../../system_wrappers:metrics", "../utility", "//third_party/abseil-cpp/absl/base:core_headers", + "//third_party/abseil-cpp/absl/base:nullability", "//third_party/abseil-cpp/absl/strings:string_view", ] if (rtc_include_internal_audio_device && is_ios) { @@ -458,6 +460,9 @@ if (rtc_include_tests && !build_with_chromium) { "../../api:scoped_refptr", "../../api:sequence_checker", "../../api/audio:audio_device", + "../../api/audio:create_audio_device_module", + "../../api/environment", + "../../api/environment:environment_factory", "../../api/task_queue", "../../api/task_queue:default_task_queue_factory", "../../api/units:time_delta", diff --git a/modules/audio_device/audio_device_impl.cc b/modules/audio_device/audio_device_impl.cc index 6ba8b0efe6..eaeb89c420 100644 --- a/modules/audio_device/audio_device_impl.cc +++ b/modules/audio_device/audio_device_impl.cc @@ -16,8 +16,10 @@ #include #include +#include "absl/base/nullability.h" #include "api/audio/audio_device.h" #include "api/audio/audio_device_defines.h" +#include "api/environment/environment.h" #include "api/make_ref_counted.h" #include "api/scoped_refptr.h" #include "api/task_queue/task_queue_factory.h" @@ -69,6 +71,11 @@ namespace webrtc { +absl_nullable scoped_refptr +AudioDeviceModuleImpl::Create(const Environment& env, AudioLayer audio_layer) { + return Create(audio_layer, &env.task_queue_factory()); +} + scoped_refptr AudioDeviceModule::Create( AudioLayer audio_layer, TaskQueueFactory* task_queue_factory) { @@ -77,9 +84,9 @@ scoped_refptr AudioDeviceModule::Create( } // static -scoped_refptr AudioDeviceModuleImpl::Create( - AudioLayer audio_layer, - TaskQueueFactory* task_queue_factory) { +absl_nullable scoped_refptr +AudioDeviceModuleImpl::Create(AudioLayer audio_layer, + TaskQueueFactory* task_queue_factory) { RTC_DLOG(LS_INFO) << __FUNCTION__; // The "AudioDeviceModule::kWindowsCoreAudio2" audio layer has its own diff --git a/modules/audio_device/audio_device_impl.h b/modules/audio_device/audio_device_impl.h index 98b11e94c6..93707f21a9 100644 --- a/modules/audio_device/audio_device_impl.h +++ b/modules/audio_device/audio_device_impl.h @@ -17,8 +17,10 @@ #include +#include "absl/base/nullability.h" #include "api/audio/audio_device.h" #include "api/audio/audio_device_defines.h" +#include "api/environment/environment.h" #include "api/scoped_refptr.h" #include "api/task_queue/task_queue_factory.h" #include "modules/audio_device/audio_device_buffer.h" @@ -44,7 +46,13 @@ class AudioDeviceModuleImpl : public AudioDeviceModuleForTest { kPlatformFuchsia = 7, }; - static scoped_refptr Create( + static absl_nullable scoped_refptr Create( + const Environment& env, + AudioLayer audio_layer); + + // TODO: bugs.webrtc.org/413413572 - Remove this variant when top level + // `AudioDeviceModule::Create` is no longer used. + static absl_nullable scoped_refptr Create( AudioLayer audio_layer, TaskQueueFactory* task_queue_factory); diff --git a/modules/audio_device/audio_device_unittest.cc b/modules/audio_device/audio_device_unittest.cc index 728673ac12..e7c1386f81 100644 --- a/modules/audio_device/audio_device_unittest.cc +++ b/modules/audio_device/audio_device_unittest.cc @@ -16,17 +16,17 @@ #include #include #include -#include #include #include #include #include "api/array_view.h" #include "api/audio/audio_device_defines.h" +#include "api/audio/create_audio_device_module.h" +#include "api/environment/environment.h" +#include "api/environment/environment_factory.h" #include "api/scoped_refptr.h" #include "api/sequence_checker.h" -#include "api/task_queue/default_task_queue_factory.h" -#include "api/task_queue/task_queue_factory.h" #include "api/units/time_delta.h" #include "modules/audio_device/audio_device_impl.h" #include "modules/audio_device/include/mock_audio_transport.h" @@ -525,8 +525,7 @@ class MAYBE_AudioDeviceTest : public ::testing::TestWithParam { protected: MAYBE_AudioDeviceTest() - : audio_layer_(GetParam()), - task_queue_factory_(CreateDefaultTaskQueueFactory()) { + : audio_layer_(GetParam()), env_(CreateEnvironment()) { LogMessage::LogToDebug(LS_INFO); // Add extra logging fields here if needed for debugging. LogMessage::LogTimestamps(); @@ -596,8 +595,7 @@ class MAYBE_AudioDeviceTest // The value of `audio_layer_` is set at construction by GetParam() and two // different layers are tested on Windows only. if (audio_layer_ == AudioDeviceModule::kPlatformDefaultAudio) { - return AudioDeviceModuleImpl::Create(audio_layer_, - task_queue_factory_.get()); + return AudioDeviceModuleImpl::Create(env_, audio_layer_); } else if (audio_layer_ == AudioDeviceModule::kWindowsCoreAudio2) { #ifdef WEBRTC_WIN // We must initialize the COM library on a thread before we calling any of @@ -609,7 +607,7 @@ class MAYBE_AudioDeviceTest EXPECT_TRUE(webrtc_win::core_audio_utility::IsSupported()); EXPECT_TRUE(webrtc_win::core_audio_utility::IsMMCSSSupported()); return CreateWindowsCoreAudioAudioDeviceModuleForTest( - task_queue_factory_.get(), true); + &env_.task_queue_factory(), true); #else return nullptr; #endif @@ -666,7 +664,7 @@ class MAYBE_AudioDeviceTest std::unique_ptr com_initializer_; #endif AudioDeviceModule::AudioLayer audio_layer_; - std::unique_ptr task_queue_factory_; + const Environment env_; bool requirements_satisfied_ = true; Event event_; scoped_refptr audio_device_; @@ -676,13 +674,12 @@ class MAYBE_AudioDeviceTest // Instead of using the test fixture, verify that the different factory methods // work as intended. TEST(MAYBE_AudioDeviceTestWin, ConstructDestructWithFactory) { - std::unique_ptr task_queue_factory = - CreateDefaultTaskQueueFactory(); + const Environment env = CreateEnvironment(); scoped_refptr audio_device; - // The default factory should work for all platforms when a default ADM is + // The default environment should work for all platforms when a default ADM is // requested. - audio_device = AudioDeviceModule::Create( - AudioDeviceModule::kPlatformDefaultAudio, task_queue_factory.get()); + audio_device = + CreateAudioDeviceModule(env, AudioDeviceModule::kPlatformDefaultAudio); EXPECT_TRUE(audio_device); audio_device = nullptr; #ifdef WEBRTC_WIN @@ -690,8 +687,8 @@ TEST(MAYBE_AudioDeviceTestWin, ConstructDestructWithFactory) { // specific parts are implemented by an AudioDeviceGeneric object. Verify // that the old factory can't be used in combination with the latest audio // layer AudioDeviceModule::kWindowsCoreAudio2. - audio_device = AudioDeviceModule::Create( - AudioDeviceModule::kWindowsCoreAudio2, task_queue_factory.get()); + audio_device = + CreateAudioDeviceModule(env, AudioDeviceModule::kWindowsCoreAudio2); EXPECT_FALSE(audio_device); audio_device = nullptr; // Instead, ensure that the new dedicated factory method called @@ -701,7 +698,7 @@ TEST(MAYBE_AudioDeviceTestWin, ConstructDestructWithFactory) { ScopedCOMInitializer com_initializer(ScopedCOMInitializer::kMTA); EXPECT_TRUE(com_initializer.Succeeded()); audio_device = - CreateWindowsCoreAudioAudioDeviceModule(task_queue_factory.get()); + CreateWindowsCoreAudioAudioDeviceModule(&env.task_queue_factory()); EXPECT_TRUE(audio_device); AudioDeviceModule::AudioLayer audio_layer; EXPECT_EQ(0, audio_device->ActiveAudioLayer(&audio_layer)); diff --git a/video/BUILD.gn b/video/BUILD.gn index 7bf330143f..548442e193 100644 --- a/video/BUILD.gn +++ b/video/BUILD.gn @@ -543,6 +543,7 @@ if (rtc_include_tests) { "../api:video_quality_test_fixture_api", "../api/audio:audio_device", "../api/audio:builtin_audio_processing_builder", + "../api/audio:create_audio_device_module", "../api/environment", "../api/environment:environment_factory", "../api/numerics", diff --git a/video/video_quality_test.cc b/video/video_quality_test.cc index a4faa23776..f7a1239c25 100644 --- a/video/video_quality_test.cc +++ b/video/video_quality_test.cc @@ -17,6 +17,7 @@ #include #include "absl/flags/flag.h" +#include "api/audio/create_audio_device_module.h" #include "api/call/transport.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" @@ -1390,8 +1391,8 @@ scoped_refptr VideoQualityTest::CreateAudioDevice() { return CreateWindowsCoreAudioAudioDeviceModule(&env_.task_queue_factory()); #else // Use legacy factory method on all platforms except Windows. - return AudioDeviceModule::Create(AudioDeviceModule::kPlatformDefaultAudio, - &env_.task_queue_factory()); + return CreateAudioDeviceModule(env_, + AudioDeviceModule::kPlatformDefaultAudio); #endif } diff --git a/webrtc_lib_link_test.cc b/webrtc_lib_link_test.cc index 97813ca1a8..b5ba086f55 100644 --- a/webrtc_lib_link_test.cc +++ b/webrtc_lib_link_test.cc @@ -14,15 +14,16 @@ #include "api/audio/audio_device.h" #include "api/audio/builtin_audio_processing_builder.h" +#include "api/audio/create_audio_device_module.h" #include "api/audio_codecs/audio_decoder_factory_template.h" #include "api/audio_codecs/audio_encoder_factory_template.h" #include "api/audio_codecs/opus/audio_decoder_opus.h" #include "api/audio_codecs/opus/audio_encoder_opus.h" #include "api/create_peerconnection_factory.h" #include "api/enable_media.h" +#include "api/environment/environment_factory.h" #include "api/peer_connection_interface.h" #include "api/rtc_event_log/rtc_event_log_factory.h" -#include "api/task_queue/default_task_queue_factory.h" #include "api/video_codecs/video_decoder_factory_template.h" #include "api/video_codecs/video_decoder_factory_template_dav1d_adapter.h" #include "api/video_codecs/video_decoder_factory_template_libvpx_vp8_adapter.h" @@ -38,8 +39,8 @@ namespace webrtc { void CreateSomeMediaDeps(PeerConnectionFactoryDependencies& media_deps) { - media_deps.adm = AudioDeviceModule::Create( - AudioDeviceModule::kDummyAudio, media_deps.task_queue_factory.get()); + media_deps.adm = + CreateAudioDeviceModule(*media_deps.env, AudioDeviceModule::kDummyAudio); media_deps.audio_encoder_factory = webrtc::CreateAudioEncoderFactory(); media_deps.audio_decoder_factory = @@ -58,7 +59,7 @@ void CreateSomeMediaDeps(PeerConnectionFactoryDependencies& media_deps) { webrtc::PeerConnectionFactoryDependencies CreateSomePcfDeps() { webrtc::PeerConnectionFactoryDependencies pcf_deps; - pcf_deps.task_queue_factory = CreateDefaultTaskQueueFactory(); + pcf_deps.env = CreateEnvironment(); pcf_deps.signaling_thread = webrtc::Thread::Current(); pcf_deps.network_thread = webrtc::Thread::Current(); pcf_deps.worker_thread = webrtc::Thread::Current(); @@ -85,7 +86,7 @@ void TestCase1ModularFactory() { void TestCase2RegularFactory() { PeerConnectionFactoryDependencies media_deps; - media_deps.task_queue_factory = CreateDefaultTaskQueueFactory(); + media_deps.env = CreateEnvironment(); CreateSomeMediaDeps(media_deps); auto peer_connection_factory = webrtc::CreatePeerConnectionFactory( From 945d54db67ba88061ef29e354198261b776700ab Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Wed, 30 Apr 2025 05:02:10 -0700 Subject: [PATCH 016/292] Roll chromium_revision c5d386d663..85c17e9abc (1453677:1453829) Change log: https://chromium.googlesource.com/chromium/src/+log/c5d386d663..85c17e9abc Full diff: https://chromium.googlesource.com/chromium/src/+/c5d386d663..85c17e9abc Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/21c29fc96c..c522b129e6 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/54a242f2d5..a746c4fcb7 * src/buildtools: https://chromium.googlesource.com/chromium/src/buildtools/+log/2bf618c359..6d4e129ac6 * src/buildtools/linux64: git_revision:85cc21e94af590a267c1c7a47020d9b420f8a033..git_revision:487f8353f15456474437df32bb186187b0940b45 * src/buildtools/mac: git_revision:85cc21e94af590a267c1c7a47020d9b420f8a033..git_revision:487f8353f15456474437df32bb186187b0940b45 * src/buildtools/win: git_revision:85cc21e94af590a267c1c7a47020d9b420f8a033..git_revision:487f8353f15456474437df32bb186187b0940b45 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/8ff1667439..c99a48d0f2 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/3291a34ffb..a81bb5be85 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/fdb5d002fe..9f103c627f * src/third_party/androidx/cipd: JBCFmzXl0Hn5K4GTl49NmYclGryCKT35FQAYn8HHbGIC..QbucUc-32_nUGnCBPOH-odoVUKwICqCUlH5V2Cup6VwC * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/1fcc527019..6bcdcd8a85 * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/5106a2a7c6..e68f172a18 * src/third_party/llvm-build/Release+Asserts_version: Linux_x64/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clang-win-runtime-library-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/llvmobjdump-llvmorg-21-init-6681-g5b36835d-1.tar.xz..Linux_x64/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clang-win-runtime-library-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/llvmobjdump-llvmorg-21-init-9266-g09006611-1.tar.xz * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/f308c40f9c..f2d2ed5a0b * src/third_party/rust-toolchain_version: Linux_x64/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Mac/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Mac_arm64/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Win/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz..Linux_x64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Mac/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Mac_arm64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Win/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/4eb583ccf2..f3d9d83814 * src/tools/luci-go: git_revision:94e82b5e2ea91e24ae4f237a5b9e1f365ae80d68..git_revision:74734688ffc9d2807cda0d1dc7752197480de29e * src/tools/luci-go: git_revision:94e82b5e2ea91e24ae4f237a5b9e1f365ae80d68..git_revision:74734688ffc9d2807cda0d1dc7752197480de29e * src/tools/luci-go: git_revision:94e82b5e2ea91e24ae4f237a5b9e1f365ae80d68..git_revision:74734688ffc9d2807cda0d1dc7752197480de29e DEPS diff: https://chromium.googlesource.com/chromium/src/+/c5d386d663..85c17e9abc/DEPS No update to Clang. BUG=None Change-Id: I80eb5788c7e1dbca186aff66a9e6daa5f0e6d64d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/388873 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44488} --- DEPS | 36 ++++++++++++++++++------------------ 1 file changed, 18 insertions(+), 18 deletions(-) diff --git a/DEPS b/DEPS index 7fff663bb6..0836fe15b6 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': 'c5d386d663279bd27fde52ab88d54a3cb577fd95', + 'chromium_revision': '85c17e9abc4ed443ef6794a715b5e9430d7ab21c', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,30 +73,30 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@21c29fc96cd4f11cef7215683362cf812e464917', + 'https://chromium.googlesource.com/chromium/src/base@c522b129e6c3081f53141c105f2824f2519fba18', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@54a242f2d53a68c2002e5b9e9f8ded597804c3bb', + 'https://chromium.googlesource.com/chromium/src/build@a746c4fcb78f0f3257036c7fc01f4df93991f22d', 'src/buildtools': - 'https://chromium.googlesource.com/chromium/src/buildtools@2bf618c359b513ad97a286e09c5089944e698ed1', + 'https://chromium.googlesource.com/chromium/src/buildtools@6d4e129ac68b4c81d98765068ceb53dbf09e99d2', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. 'src/examples/androidtests/third_party/gradle': { 'url': 'https://chromium.googlesource.com/external/github.com/gradle/gradle.git@f2d1fb54a951d8b11d25748e4711bec8d128d7e3', 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@8ff16674396ba753ffb0712caccd66c4e1d14533', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@c99a48d0f2893d0959a0d40ac2b63b575590fd89', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@3291a34ffb7203c906217cce3c04a5737e61e903', + 'https://chromium.googlesource.com/chromium/src/testing@a81bb5be85794a01d4bbc3a12c0d1b7e14c16295', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@fdb5d002fe914834a1f3a21daf91bf5f8e5a4d74', + 'https://chromium.googlesource.com/chromium/src/third_party@9f103c627ffedcff735f4264dedc07ab34ba0e89', 'src/buildtools/linux64': { 'packages': [ { 'package': 'gn/gn/linux-${{arch}}', - 'version': 'git_revision:85cc21e94af590a267c1c7a47020d9b420f8a033', + 'version': 'git_revision:487f8353f15456474437df32bb186187b0940b45', } ], 'dep_type': 'cipd', @@ -106,7 +106,7 @@ deps = { 'packages': [ { 'package': 'gn/gn/mac-${{arch}}', - 'version': 'git_revision:85cc21e94af590a267c1c7a47020d9b420f8a033', + 'version': 'git_revision:487f8353f15456474437df32bb186187b0940b45', } ], 'dep_type': 'cipd', @@ -116,7 +116,7 @@ deps = { 'packages': [ { 'package': 'gn/gn/windows-amd64', - 'version': 'git_revision:85cc21e94af590a267c1c7a47020d9b420f8a033', + 'version': 'git_revision:487f8353f15456474437df32bb186187b0940b45', } ], 'dep_type': 'cipd', @@ -497,13 +497,13 @@ deps = { 'src/third_party/crc32c/src': 'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558', 'src/third_party/depot_tools': - 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@1fcc527019d786502b02f71b8b764ee674a40953', + 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@6bcdcd8a8551b623636850917bf6bf295d310e62', 'src/third_party/ffmpeg': 'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@01f23648c6b84de6c0f717fa4e1816f53b9ee72e', 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@5106a2a7c6d26b94288b8230b53c8e92008dd259', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@e68f172a18d586c351d58eb20afd909b39118575', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'f308c40f9c8b1905f1dacfff857d175a0b77060a', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'f2d2ed5a0be08bb7ed40a661298b0ad5252fec10', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@4eb583ccf263251865f5b7bf1c6ce3f86c52a384', + 'https://chromium.googlesource.com/chromium/src/tools@f3d9d83814f98361b0145d3c11e73deef43d51b7', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'JBCFmzXl0Hn5K4GTl49NmYclGryCKT35FQAYn8HHbGIC', + 'version': 'QbucUc-32_nUGnCBPOH-odoVUKwICqCUlH5V2Cup6VwC', }, ], 'condition': 'checkout_android and non_git_source', @@ -793,15 +793,15 @@ deps = { 'packages': [ { 'package': 'infra/tools/luci/cas/${{platform}}', - 'version': 'git_revision:94e82b5e2ea91e24ae4f237a5b9e1f365ae80d68', + 'version': 'git_revision:74734688ffc9d2807cda0d1dc7752197480de29e', }, { 'package': 'infra/tools/luci/isolate/${{platform}}', - 'version': 'git_revision:94e82b5e2ea91e24ae4f237a5b9e1f365ae80d68', + 'version': 'git_revision:74734688ffc9d2807cda0d1dc7752197480de29e', }, { 'package': 'infra/tools/luci/swarming/${{platform}}', - 'version': 'git_revision:94e82b5e2ea91e24ae4f237a5b9e1f365ae80d68', + 'version': 'git_revision:74734688ffc9d2807cda0d1dc7752197480de29e', } ], 'dep_type': 'cipd', From 66bd178791affb96dcb5c1f2490274169df8fdaa Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Tue, 29 Apr 2025 17:17:05 +0000 Subject: [PATCH 017/292] Cleanup legacy Transport::SendRtcp variant Bug: webrtc:42225697 Change-Id: Ie61a11a82142a2e8704a026b9cb04c6d5f4d866d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389060 Reviewed-by: Per Kjellander Commit-Queue: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#44489} --- api/call/transport.h | 15 ++------------- 1 file changed, 2 insertions(+), 13 deletions(-) diff --git a/api/call/transport.h b/api/call/transport.h index a445ffe799..74d7addf6c 100644 --- a/api/call/transport.h +++ b/api/call/transport.h @@ -43,22 +43,11 @@ class Transport { public: virtual bool SendRtp(ArrayView packet, const PacketOptions& options) = 0; - // TODO: bugs.webrtc.org/42225697 - Make virtual when downstream is updated. virtual bool SendRtcp(ArrayView packet, - const PacketOptions& /*options*/) { -#pragma clang diagnostic push -#pragma clang diagnostic ignored "-Wdeprecated-declarations" - return SendRtcp(packet); -#pragma clang diagnostic pop - } - [[deprecated("Pass PacketOptions")]] - // TODO: bugs.webrtc.org/42225697 - Remove when downstream is updated. - virtual bool SendRtcp(ArrayView packet) { - return SendRtcp(packet, /*packet_options=*/{}); - } + const PacketOptions& options) = 0; protected: - virtual ~Transport() {} + virtual ~Transport() = default; }; } // namespace webrtc From e57d4d11eb7750c4e3767aa51fcdf99af83ac0f2 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Wed, 30 Apr 2025 09:04:03 -0700 Subject: [PATCH 018/292] Roll chromium_revision 85c17e9abc..5590bd5c51 (1453829:1453968) Change log: https://chromium.googlesource.com/chromium/src/+log/85c17e9abc..5590bd5c51 Full diff: https://chromium.googlesource.com/chromium/src/+/85c17e9abc..5590bd5c51 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/c522b129e6..e699625442 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/a746c4fcb7..bebf043242 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/c99a48d0f2..0ffcda7930 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/a81bb5be85..dade203d64 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/9f103c627f..284cb7680b * src/third_party/android_deps/autorolled/cipd: _rzZP_RJ0Fx7bXb_R5WFX8yvYsF5YffSZ0p9pbI_Z64C..I4KwXm_DsdqDHlwWP2Ym6Qecv-6fM6c1Kt5u277RCoAC * src/third_party/androidx/cipd: QbucUc-32_nUGnCBPOH-odoVUKwICqCUlH5V2Cup6VwC..GErMCyXvL9XZEIBvbpHhvb3xe-US91Y1AxYaxw6NhjgC * src/third_party/llvm-build/Release+Asserts_version: Linux_x64/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clang-win-runtime-library-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/llvmobjdump-llvmorg-21-init-6681-g5b36835d-1.tar.xz..Linux_x64/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clang-win-runtime-library-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/llvmobjdump-llvmorg-21-init-9266-g09006611-1.tar.xz * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/f2d2ed5a0b..3c471eb06c * src/third_party/rust-toolchain_version: Linux_x64/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Mac/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Mac_arm64/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Win/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz..Linux_x64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Mac/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Mac_arm64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Win/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/f3d9d83814..19eca95f9b DEPS diff: https://chromium.googlesource.com/chromium/src/+/85c17e9abc..5590bd5c51/DEPS No update to Clang. BUG=None Change-Id: I1bada099066c7d06b7707bc67305fd7919e9ce07 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389160 Commit-Queue: Autoroller Bot-Commit: Autoroller Cr-Commit-Position: refs/heads/main@{#44490} --- DEPS | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/DEPS b/DEPS index 0836fe15b6..0d2b64ddb0 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '85c17e9abc4ed443ef6794a715b5e9430d7ab21c', + 'chromium_revision': '5590bd5c51932aa6fb8cca62a449e38e427919c3', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@c522b129e6c3081f53141c105f2824f2519fba18', + 'https://chromium.googlesource.com/chromium/src/base@e6996254420dd76642ed2a45067001c41ce104a8', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@a746c4fcb78f0f3257036c7fc01f4df93991f22d', + 'https://chromium.googlesource.com/chromium/src/build@bebf04324258b17ff4125fc7e59b575167d3aea3', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@6d4e129ac68b4c81d98765068ceb53dbf09e99d2', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@c99a48d0f2893d0959a0d40ac2b63b575590fd89', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@0ffcda793051b7ac0ad02fb8c61cd83276dd5093', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@a81bb5be85794a01d4bbc3a12c0d1b7e14c16295', + 'https://chromium.googlesource.com/chromium/src/testing@dade203d6456e86a6c57ff677ed474c59dd9d1c9', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@9f103c627ffedcff735f4264dedc07ab34ba0e89', + 'https://chromium.googlesource.com/chromium/src/third_party@284cb7680b6639f1cc915ca3f9878d6d5d22640b', 'src/buildtools/linux64': { 'packages': [ @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'f2d2ed5a0be08bb7ed40a661298b0ad5252fec10', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '3c471eb06cd03414e9da8a237b7c81a2ddf390fd', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@f3d9d83814f98361b0145d3c11e73deef43d51b7', + 'https://chromium.googlesource.com/chromium/src/tools@19eca95f9b325a84abb132dece616bb550585cae', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'QbucUc-32_nUGnCBPOH-odoVUKwICqCUlH5V2Cup6VwC', + 'version': 'GErMCyXvL9XZEIBvbpHhvb3xe-US91Y1AxYaxw6NhjgC', }, ], 'condition': 'checkout_android and non_git_source', @@ -827,7 +827,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_deps/autorolled', - 'version': '_rzZP_RJ0Fx7bXb_R5WFX8yvYsF5YffSZ0p9pbI_Z64C', + 'version': 'I4KwXm_DsdqDHlwWP2Ym6Qecv-6fM6c1Kt5u277RCoAC', }, ], 'condition': 'checkout_android and non_git_source', From 39ec17797d96daeeaeb3aca383f3bbfc4ea836c2 Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Tue, 29 Apr 2025 21:51:15 +0000 Subject: [PATCH 019/292] Remove setting global field trial string from FieldTrials class WebRTC tools where FieldTrials is used to set global field trials string no longer need that feature as they rely on the propagated field trials instead. Most of the users of the FieldTrials already prefer this class not to set the global field string by calling FieldTrials::CreateNoGlobal. Bug: webrtc:42220378 Change-Id: Ic5f401d80c7138788213d142a15877327d94dc91 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389100 Commit-Queue: Danil Chapovalov Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44491} --- api/BUILD.gn | 2 - api/field_trials.cc | 61 ++----------------- api/field_trials.h | 23 +++---- api/field_trials_unittest.cc | 99 +++++++------------------------ test/BUILD.gn | 11 +--- test/explicit_key_value_config.cc | 56 ----------------- test/explicit_key_value_config.h | 20 +------ 7 files changed, 41 insertions(+), 231 deletions(-) delete mode 100644 test/explicit_key_value_config.cc diff --git a/api/BUILD.gn b/api/BUILD.gn index 6340d2bed4..b0e44062f4 100644 --- a/api/BUILD.gn +++ b/api/BUILD.gn @@ -1627,7 +1627,6 @@ if (rtc_include_tests) { "task_queue:task_queue_default_factory_unittests", "test/pclf:media_configuration", "test/video:video_frame_writer", - "transport:field_trial_based_config", "units:time_delta", "units:timestamp", "units:units_unittests", @@ -1726,7 +1725,6 @@ rtc_library("field_trials") { ":field_trials_registry", "../rtc_base:checks", "../rtc_base/containers:flat_map", - "../system_wrappers:field_trial", "//third_party/abseil-cpp/absl/strings:string_view", ] } diff --git a/api/field_trials.cc b/api/field_trials.cc index 4aca69d781..c39d04cf17 100644 --- a/api/field_trials.cc +++ b/api/field_trials.cc @@ -10,20 +10,16 @@ #include "api/field_trials.h" -#include -#include #include #include #include "absl/strings/string_view.h" #include "rtc_base/checks.h" #include "rtc_base/containers/flat_map.h" -#include "system_wrappers/include/field_trial.h" -namespace { +namespace webrtc { -webrtc::flat_map InsertIntoMap(absl::string_view s) { - webrtc::flat_map key_value_map; +FieldTrials::FieldTrials(absl::string_view s) { while (!s.empty()) { absl::string_view::size_type separator_pos = s.find('/'); RTC_CHECK_NE(separator_pos, absl::string_view::npos) @@ -42,62 +38,17 @@ webrtc::flat_map InsertIntoMap(absl::string_view s) { s.remove_prefix(separator_pos + 1); // If a key is specified multiple times, only the value linked to the first - // key is stored. note: This will crash in debug build when calling - // InitFieldTrialsFromString(). - key_value_map.emplace(key, value); - } - - return key_value_map; -} - -// Makes sure that only one instance is created, since the usage -// of global string makes behaviour unpredicatable otherwise. -// TODO(bugs.webrtc.org/10335): Remove once global string is gone. -std::atomic instance_created_{false}; - -} // namespace - -namespace webrtc { - -FieldTrials::FieldTrials(absl::string_view s) - : uses_global_(true), - field_trial_string_(s), - previous_field_trial_string_(webrtc::field_trial::GetFieldTrialString()), - key_value_map_(InsertIntoMap(s)) { - // TODO(bugs.webrtc.org/10335): Remove the global string! - field_trial::InitFieldTrialsFromString(field_trial_string_.c_str()); - RTC_CHECK(!instance_created_.exchange(true)) - << "Only one instance may be instanciated at any given time!"; -} - -std::unique_ptr FieldTrials::CreateNoGlobal(absl::string_view s) { - return std::unique_ptr(new FieldTrials(s, true)); -} - -FieldTrials::FieldTrials(absl::string_view s, bool) - : uses_global_(false), - previous_field_trial_string_(nullptr), - key_value_map_(InsertIntoMap(s)) {} - -FieldTrials::~FieldTrials() { - // TODO(bugs.webrtc.org/10335): Remove the global string! - if (uses_global_) { - field_trial::InitFieldTrialsFromString(previous_field_trial_string_); - RTC_CHECK(instance_created_.exchange(false)); + // key is stored. + key_value_map_.emplace(key, value); } } std::string FieldTrials::GetValue(absl::string_view key) const { auto it = key_value_map_.find(key); - if (it != key_value_map_.end()) + if (it != key_value_map_.end()) { return it->second; - - // Check the global string so that programs using - // a mix between FieldTrials and the global string continue to work - // TODO(bugs.webrtc.org/10335): Remove the global string! - if (uses_global_) { - return field_trial::FindFullName(key); } + return ""; } diff --git a/api/field_trials.h b/api/field_trials.h index 693c1afe8c..ff42c32078 100644 --- a/api/field_trials.h +++ b/api/field_trials.h @@ -30,28 +30,23 @@ namespace webrtc { // feature is on/off. // // The field trials are injected into objects that use them at creation time. -// -// NOTE: Creating multiple FieldTrials-object is currently prohibited -// until we remove the global string (TODO(bugs.webrtc.org/10335)) -// (unless using CreateNoGlobal): class FieldTrials : public FieldTrialsRegistry { public: explicit FieldTrials(absl::string_view s); - ~FieldTrials(); + FieldTrials(FieldTrials&&) = default; + FieldTrials& operator=(FieldTrials&&) = default; + ~FieldTrials() = default; - // Create a FieldTrials object that is not reading/writing from - // global variable (i.e can not be used for all parts of webrtc). - static std::unique_ptr CreateNoGlobal(absl::string_view s); + // TODO: bugs.webrtc.org/42220378 - Deprecate and inline once no longer used + // within webrtc. + static std::unique_ptr CreateNoGlobal(absl::string_view s) { + return std::make_unique(s); + } private: - explicit FieldTrials(absl::string_view s, bool); - std::string GetValue(absl::string_view key) const override; - const bool uses_global_; - const std::string field_trial_string_; - const char* const previous_field_trial_string_; - const flat_map key_value_map_; + flat_map key_value_map_; }; } // namespace webrtc diff --git a/api/field_trials_unittest.cc b/api/field_trials_unittest.cc index 8144f11c38..d7a02513c7 100644 --- a/api/field_trials_unittest.cc +++ b/api/field_trials_unittest.cc @@ -10,23 +10,15 @@ #include "api/field_trials.h" -#include -#include "api/transport/field_trial_based_config.h" #include "rtc_base/containers/flat_set.h" #include "system_wrappers/include/field_trial.h" #include "test/field_trial.h" -#include "test/gmock.h" #include "test/gtest.h" -#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) -#include "test/testsupport/rtc_expect_death.h" -#endif // GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) - namespace webrtc { namespace { -using ::testing::NotNull; using ::webrtc::field_trial::FieldTrialsAllowedInScopeForTesting; using ::webrtc::test::ScopedFieldTrials; @@ -61,89 +53,40 @@ TEST(FieldTrialsTest, FieldTrialsDoesNotReadGlobalString) { EXPECT_FALSE(f.IsDisabled("MyUncoolTrial")); } -TEST(FieldTrialsTest, FieldTrialsWritesGlobalString) { - FieldTrialsAllowedInScopeForTesting k({"MyCoolTrial", "MyUncoolTrial"}); - FieldTrials f("MyCoolTrial/Enabled/MyUncoolTrial/Disabled/"); - EXPECT_TRUE(webrtc::field_trial::IsEnabled("MyCoolTrial")); - EXPECT_TRUE(webrtc::field_trial::IsDisabled("MyUncoolTrial")); -} - -TEST(FieldTrialsTest, FieldTrialsRestoresGlobalStringAfterDestruction) { - static constexpr char s[] = "SomeString/Enabled/"; - ScopedFieldTrials g(s); - { - FieldTrials f("SomeOtherString/Enabled/"); - EXPECT_STREQ(webrtc::field_trial::GetFieldTrialString(), - "SomeOtherString/Enabled/"); - } - EXPECT_STREQ(webrtc::field_trial::GetFieldTrialString(), s); -} - -#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) -TEST(FieldTrialsTest, FieldTrialsDoesNotSupportSimultaneousInstances) { - FieldTrials f("SomeString/Enabled/"); - RTC_EXPECT_DEATH(FieldTrials("SomeOtherString/Enabled/").Lookup("Whatever"), - "Only one instance"); -} -#endif // GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) - -TEST(FieldTrialsTest, FieldTrialsSupportsSeparateInstances) { - { FieldTrials f("SomeString/Enabled/"); } - { FieldTrials f("SomeOtherString/Enabled/"); } -} - -TEST(FieldTrialsTest, NonGlobalFieldTrialsInstanceDoesNotModifyGlobalString) { +TEST(FieldTrialsTest, FieldTrialsInstanceDoesNotModifyGlobalString) { FieldTrialsAllowedInScopeForTesting k({"SomeString"}); - std::unique_ptr f = - FieldTrials::CreateNoGlobal("SomeString/Enabled/"); - ASSERT_THAT(f, NotNull()); - f->RegisterKeysForTesting({"SomeString"}); - - EXPECT_TRUE(f->IsEnabled("SomeString")); - EXPECT_FALSE(webrtc::field_trial::IsEnabled("SomeString")); -} + FieldTrials f("SomeString/Enabled/"); + f.RegisterKeysForTesting({"SomeString"}); -TEST(FieldTrialsTest, NonGlobalFieldTrialsSupportSimultaneousInstances) { - std::unique_ptr f1 = - FieldTrials::CreateNoGlobal("SomeString/Enabled/"); - std::unique_ptr f2 = - FieldTrials::CreateNoGlobal("SomeOtherString/Enabled/"); - ASSERT_THAT(f1, NotNull()); - ASSERT_THAT(f2, NotNull()); - f1->RegisterKeysForTesting({"SomeString", "SomeOtherString"}); - f2->RegisterKeysForTesting({"SomeString", "SomeOtherString"}); - - EXPECT_TRUE(f1->IsEnabled("SomeString")); - EXPECT_FALSE(f1->IsEnabled("SomeOtherString")); - - EXPECT_FALSE(f2->IsEnabled("SomeString")); - EXPECT_TRUE(f2->IsEnabled("SomeOtherString")); + EXPECT_TRUE(f.IsEnabled("SomeString")); + EXPECT_FALSE(field_trial::IsEnabled("SomeString")); } -TEST(FieldTrialsTest, GlobalAndNonGlobalFieldTrialsAreDisjoint) { - FieldTrialsAllowedInScopeForTesting k({"SomeString", "SomeOtherString"}); +TEST(FieldTrialsTest, FieldTrialsSupportSimultaneousInstances) { FieldTrials f1("SomeString/Enabled/"); - std::unique_ptr f2 = - FieldTrials::CreateNoGlobal("SomeOtherString/Enabled/"); - ASSERT_THAT(f2, NotNull()); + FieldTrials f2("SomeOtherString/Enabled/"); f1.RegisterKeysForTesting({"SomeString", "SomeOtherString"}); - f2->RegisterKeysForTesting({"SomeString", "SomeOtherString"}); + f2.RegisterKeysForTesting({"SomeString", "SomeOtherString"}); EXPECT_TRUE(f1.IsEnabled("SomeString")); EXPECT_FALSE(f1.IsEnabled("SomeOtherString")); - EXPECT_FALSE(f2->IsEnabled("SomeString")); - EXPECT_TRUE(f2->IsEnabled("SomeOtherString")); + EXPECT_FALSE(f2.IsEnabled("SomeString")); + EXPECT_TRUE(f2.IsEnabled("SomeOtherString")); } -TEST(FieldTrialsTest, FieldTrialBasedConfigReadsGlobalString) { - FieldTrialsAllowedInScopeForTesting k({"MyCoolTrial", "MyUncoolTrial"}); - ScopedFieldTrials g("MyCoolTrial/Enabled/MyUncoolTrial/Disabled/"); - FieldTrialBasedConfig f; - f.RegisterKeysForTesting({"MyCoolTrial", "MyUncoolTrial"}); +TEST(FieldTrialsTest, GlobalAndNonGlobalFieldTrialsAreDisjoint) { + FieldTrialsAllowedInScopeForTesting k({"SomeString", "SomeOtherString"}); + ScopedFieldTrials g("SomeString/Enabled/"); + FieldTrials f("SomeOtherString/Enabled/"); - EXPECT_TRUE(f.IsEnabled("MyCoolTrial")); - EXPECT_TRUE(f.IsDisabled("MyUncoolTrial")); + f.RegisterKeysForTesting({"SomeString", "SomeOtherString"}); + + EXPECT_TRUE(field_trial::IsEnabled("SomeString")); + EXPECT_FALSE(field_trial::IsEnabled("SomeOtherString")); + + EXPECT_FALSE(f.IsEnabled("SomeString")); + EXPECT_TRUE(f.IsEnabled("SomeOtherString")); } } // namespace diff --git a/test/BUILD.gn b/test/BUILD.gn index 292a4c6662..7d106854a3 100644 --- a/test/BUILD.gn +++ b/test/BUILD.gn @@ -329,16 +329,9 @@ rtc_library("field_trial") { } rtc_library("explicit_key_value_config") { - sources = [ - "explicit_key_value_config.cc", - "explicit_key_value_config.h", - ] + sources = [ "explicit_key_value_config.h" ] - deps = [ - "../api:field_trials_registry", - "../rtc_base:checks", - "//third_party/abseil-cpp/absl/strings:string_view", - ] + deps = [ "../api:field_trials" ] } rtc_library("scoped_key_value_config") { diff --git a/test/explicit_key_value_config.cc b/test/explicit_key_value_config.cc deleted file mode 100644 index 90690c0514..0000000000 --- a/test/explicit_key_value_config.cc +++ /dev/null @@ -1,56 +0,0 @@ -/* - * Copyright (c) 2020 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "test/explicit_key_value_config.h" - -#include "absl/strings/string_view.h" -#include "rtc_base/checks.h" - -namespace webrtc { -namespace test { - -ExplicitKeyValueConfig::ExplicitKeyValueConfig(absl::string_view s) { - std::string::size_type field_start = 0; - while (field_start < s.size()) { - std::string::size_type separator_pos = s.find('/', field_start); - RTC_CHECK_NE(separator_pos, std::string::npos) - << "Missing separator '/' after field trial key."; - RTC_CHECK_GT(separator_pos, field_start) - << "Field trial key cannot be empty."; - std::string key(s.substr(field_start, separator_pos - field_start)); - field_start = separator_pos + 1; - - RTC_CHECK_LT(field_start, s.size()) - << "Missing value after field trial key. String ended."; - separator_pos = s.find('/', field_start); - RTC_CHECK_NE(separator_pos, std::string::npos) - << "Missing terminating '/' in field trial string."; - RTC_CHECK_GT(separator_pos, field_start) - << "Field trial value cannot be empty."; - std::string value(s.substr(field_start, separator_pos - field_start)); - field_start = separator_pos + 1; - - key_value_map_[key] = value; - } - // This check is technically redundant due to earlier checks. - // We nevertheless keep the check to make it clear that the entire - // string has been processed, and without indexing past the end. - RTC_CHECK_EQ(field_start, s.size()); -} - -std::string ExplicitKeyValueConfig::GetValue(absl::string_view key) const { - auto it = key_value_map_.find(key); - if (it != key_value_map_.end()) - return it->second; - return ""; -} - -} // namespace test -} // namespace webrtc diff --git a/test/explicit_key_value_config.h b/test/explicit_key_value_config.h index f14a10432c..0766447853 100644 --- a/test/explicit_key_value_config.h +++ b/test/explicit_key_value_config.h @@ -11,27 +11,13 @@ #ifndef TEST_EXPLICIT_KEY_VALUE_CONFIG_H_ #define TEST_EXPLICIT_KEY_VALUE_CONFIG_H_ -#include -#include -#include - -#include "absl/strings/string_view.h" -#include "api/field_trials_registry.h" +#include "api/field_trials.h" namespace webrtc { namespace test { -class ExplicitKeyValueConfig : public FieldTrialsRegistry { - public: - explicit ExplicitKeyValueConfig(absl::string_view s); - - private: - std::string GetValue(absl::string_view key) const override; - - // Unlike std::less, std::less<> is transparent and allows - // heterogeneous lookup directly with absl::string_view. - std::map> key_value_map_; -}; +// TODO: bugs.webrtc.org/42220378 - Use `FieldTrials` directly. +using ExplicitKeyValueConfig = FieldTrials; } // namespace test } // namespace webrtc From b3f5f60fbe71b08597d3a09cea0acec3382ec087 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Wed, 30 Apr 2025 13:16:38 -0700 Subject: [PATCH 020/292] Roll chromium_revision 5590bd5c51..08b8f439f5 (1453968:1454125) Change log: https://chromium.googlesource.com/chromium/src/+log/5590bd5c51..08b8f439f5 Full diff: https://chromium.googlesource.com/chromium/src/+/5590bd5c51..08b8f439f5 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/e699625442..93ea358d64 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/bebf043242..92397106a7 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/0ffcda7930..c1e1e650a0 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/284cb7680b..086afef984 * src/third_party/androidx/cipd: GErMCyXvL9XZEIBvbpHhvb3xe-US91Y1AxYaxw6NhjgC..DGv8hAr3K2Qyh0okP6syAJbvPRKjP0EMD6DLlEl-NK4C * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/6bcdcd8a85..e6596746dc * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/e68f172a18..9a5847828d * src/third_party/llvm-build/Release+Asserts_version: Linux_x64/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clang-win-runtime-library-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/llvmobjdump-llvmorg-21-init-6681-g5b36835d-1.tar.xz..Linux_x64/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clang-win-runtime-library-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/llvmobjdump-llvmorg-21-init-9266-g09006611-1.tar.xz * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/3c471eb06c..ed0bcb0f5a * src/third_party/rust-toolchain_version: Linux_x64/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Mac/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Mac_arm64/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Win/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz..Linux_x64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Mac/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Mac_arm64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Win/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/19eca95f9b..e88623b81b DEPS diff: https://chromium.googlesource.com/chromium/src/+/5590bd5c51..08b8f439f5/DEPS No update to Clang. BUG=None Change-Id: I9dfd5444269eeb534ed5d54a4e9592473a819366 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389200 Commit-Queue: Autoroller Bot-Commit: Autoroller Cr-Commit-Position: refs/heads/main@{#44492} --- DEPS | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/DEPS b/DEPS index 0d2b64ddb0..07650a5263 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '5590bd5c51932aa6fb8cca62a449e38e427919c3', + 'chromium_revision': '08b8f439f5b410cfc985051a00294c1e1c1d652c', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@e6996254420dd76642ed2a45067001c41ce104a8', + 'https://chromium.googlesource.com/chromium/src/base@93ea358d6460b8c7106c1f73f4d7bdddc1f12a83', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@bebf04324258b17ff4125fc7e59b575167d3aea3', + 'https://chromium.googlesource.com/chromium/src/build@92397106a791460358249e5254ea8ba85ce04122', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@6d4e129ac68b4c81d98765068ceb53dbf09e99d2', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@0ffcda793051b7ac0ad02fb8c61cd83276dd5093', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@c1e1e650a0db09a48595d211fabc3a5c8ed8fd95', 'condition': 'checkout_ios', }, 'src/testing': 'https://chromium.googlesource.com/chromium/src/testing@dade203d6456e86a6c57ff677ed474c59dd9d1c9', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@284cb7680b6639f1cc915ca3f9878d6d5d22640b', + 'https://chromium.googlesource.com/chromium/src/third_party@086afef98449fb8719e5d7438071a2dda342c5c6', 'src/buildtools/linux64': { 'packages': [ @@ -497,13 +497,13 @@ deps = { 'src/third_party/crc32c/src': 'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558', 'src/third_party/depot_tools': - 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@6bcdcd8a8551b623636850917bf6bf295d310e62', + 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@e6596746dc95fe658b3a5f0924c10e322c7d8e22', 'src/third_party/ffmpeg': 'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@01f23648c6b84de6c0f717fa4e1816f53b9ee72e', 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@e68f172a18d586c351d58eb20afd909b39118575', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@9a5847828d9835e7a7320042d2fb81536434eb06', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '3c471eb06cd03414e9da8a237b7c81a2ddf390fd', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'ed0bcb0f5a956cb4aad00d150851f07496cb23b0', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@19eca95f9b325a84abb132dece616bb550585cae', + 'https://chromium.googlesource.com/chromium/src/tools@e88623b81b18ec4650b21cdf25890d411001d4cb', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'GErMCyXvL9XZEIBvbpHhvb3xe-US91Y1AxYaxw6NhjgC', + 'version': 'DGv8hAr3K2Qyh0okP6syAJbvPRKjP0EMD6DLlEl-NK4C', }, ], 'condition': 'checkout_android and non_git_source', From b2f96be6db2db267361ca43a8dcbc86ff0d21382 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Wed, 30 Apr 2025 17:05:22 -0700 Subject: [PATCH 021/292] Roll chromium_revision 08b8f439f5..45bb02a7df (1454125:1454241) Change log: https://chromium.googlesource.com/chromium/src/+log/08b8f439f5..45bb02a7df Full diff: https://chromium.googlesource.com/chromium/src/+/08b8f439f5..45bb02a7df Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/93ea358d64..b9d38213d9 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/92397106a7..d5793d1e36 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/c1e1e650a0..6e92444555 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/dade203d64..ed4dfb506a * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/086afef984..75c636c54e * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/a70ec8dd2c..ab301d01c2 * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/083971e6b0..0f4bac59a3 * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/9a5847828d..1991e6da4d * src/third_party/kotlin_stdlib/cipd: KFgAvgfE4Ssv7PMRYGJVo89rhJCPGQitmhEh39RETFgC..74gAxjfyJIVJGIA4z0V-jCmMW6A6dDJSoWMh7qhlzp4C * src/third_party/llvm-build/Release+Asserts_version: Linux_x64/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clang-win-runtime-library-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/llvmobjdump-llvmorg-21-init-6681-g5b36835d-1.tar.xz..Linux_x64/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clang-win-runtime-library-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/llvmobjdump-llvmorg-21-init-9266-g09006611-1.tar.xz * src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/c43b02cfdb..bf95cc5da6 * src/third_party/r8/cipd: bA3htCoEd_EArHekDGQSNpmBzQrcby2ioG6SFyl3AtwC..sxgLKZyJNZMVi8cot5yvRzqCFQxX_HbbR0psaeto5RcC * src/third_party/rust-toolchain_version: Linux_x64/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Mac/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Mac_arm64/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Win/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz..Linux_x64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Mac/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Mac_arm64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Win/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/e88623b81b..48fe65c657 DEPS diff: https://chromium.googlesource.com/chromium/src/+/08b8f439f5..45bb02a7df/DEPS No update to Clang. BUG=None Change-Id: I3ae339b5f6340b7d7add585d4cda65175b771277 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389260 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44493} --- DEPS | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) diff --git a/DEPS b/DEPS index 07650a5263..90bd872ed7 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '08b8f439f5b410cfc985051a00294c1e1c1d652c', + 'chromium_revision': '45bb02a7df727e9aed4e8208d9e3d41c083eb216', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@93ea358d6460b8c7106c1f73f4d7bdddc1f12a83', + 'https://chromium.googlesource.com/chromium/src/base@b9d38213d963c49b541e7d2e31dc0ea390b80f59', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@92397106a791460358249e5254ea8ba85ce04122', + 'https://chromium.googlesource.com/chromium/src/build@d5793d1e366655bfda43b2cce5e7cb00fd9b5404', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@6d4e129ac68b4c81d98765068ceb53dbf09e99d2', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@c1e1e650a0db09a48595d211fabc3a5c8ed8fd95', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@6e924445554ed56e7c2137a2808f07c065600fe9', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@dade203d6456e86a6c57ff677ed474c59dd9d1c9', + 'https://chromium.googlesource.com/chromium/src/testing@ed4dfb506ae0078335299f9bbec1b0b55aec6cb1', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@086afef98449fb8719e5d7438071a2dda342c5c6', + 'https://chromium.googlesource.com/chromium/src/third_party@75c636c54ea00117712af2b401cb09fbc71fad6c', 'src/buildtools/linux64': { 'packages': [ @@ -329,7 +329,7 @@ deps = { 'src/third_party/libc++abi/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@f2a7f2987f9dcdf8b04c2d8cd4dcb186641a7c3e', 'src/third_party/llvm-libc/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@c43b02cfdbd4a75e9ba233f3b5bfd8322e3bf1d8', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@bf95cc5da608540ccddbd22f1866d713eaba3ac8', 'src/third_party/libunwind/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@81e2cb40a70de2b6978e6d8658891ded9a77f7e3', @@ -480,11 +480,11 @@ deps = { }, 'src/third_party/boringssl/src': - 'https://boringssl.googlesource.com/boringssl.git@a70ec8dd2cc1816e9deadda2f6e536640380e20c', + 'https://boringssl.googlesource.com/boringssl.git@ab301d01c2bf15da3716500e518425c085c25503', 'src/third_party/breakpad/breakpad': 'https://chromium.googlesource.com/breakpad/breakpad.git@232a723f5096ab02d53d87931efa485fa77d3b03', 'src/third_party/catapult': - 'https://chromium.googlesource.com/catapult.git@083971e6b02dd794b3cb84d01033455889be7976', + 'https://chromium.googlesource.com/catapult.git@0f4bac59a38c6aa5ad9874778f822d76ea737d35', 'src/third_party/ced/src': { 'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5', }, @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@9a5847828d9835e7a7320042d2fb81536434eb06', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@1991e6da4d0f0b1b5de038e0948b2636c0bcd525', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -563,7 +563,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/kotlin_stdlib', - 'version': 'KFgAvgfE4Ssv7PMRYGJVo89rhJCPGQitmhEh39RETFgC', + 'version': '74gAxjfyJIVJGIA4z0V-jCmMW6A6dDJSoWMh7qhlzp4C', }, ], 'condition': 'checkout_android', @@ -632,7 +632,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/r8', - 'version': 'bA3htCoEd_EArHekDGQSNpmBzQrcby2ioG6SFyl3AtwC', + 'version': 'sxgLKZyJNZMVi8cot5yvRzqCFQxX_HbbR0psaeto5RcC', }, ], 'condition': 'checkout_android', @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@e88623b81b18ec4650b21cdf25890d411001d4cb', + 'https://chromium.googlesource.com/chromium/src/tools@48fe65c657a260da105a98ecd99fa26f10ae641f', 'src/third_party/espresso': { 'packages': [ From 10d2e5ebcaaf91f57bc336caf6524db0d7751a92 Mon Sep 17 00:00:00 2001 From: webrtc-version-updater Date: Wed, 30 Apr 2025 21:03:43 -0700 Subject: [PATCH 022/292] Update WebRTC code version (2025-05-01T04:03:12). Bug: None Change-Id: I23b37ea8dfe6b61c436178b863abac18e426a942 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389202 Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Cr-Commit-Position: refs/heads/main@{#44494} --- call/version.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/call/version.cc b/call/version.cc index e1d5ae0a44..aa15cd0f7d 100644 --- a/call/version.cc +++ b/call/version.cc @@ -13,7 +13,7 @@ namespace webrtc { // The timestamp is always in UTC. -const char* const kSourceTimestamp = "WebRTC source stamp 2025-04-30T04:08:55"; +const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-01T04:03:12"; void LoadWebRTCVersionInRegister() { // Using volatile to instruct the compiler to not optimize `p` away even From d0a295c96fb8b1d13d3600d840ce16482ade8ec1 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Wed, 30 Apr 2025 23:02:22 -0700 Subject: [PATCH 023/292] Roll chromium_revision 45bb02a7df..fb0dacbdf6 (1454241:1454341) Change log: https://chromium.googlesource.com/chromium/src/+log/45bb02a7df..fb0dacbdf6 Full diff: https://chromium.googlesource.com/chromium/src/+/45bb02a7df..fb0dacbdf6 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/b9d38213d9..802e7fa5a7 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/d5793d1e36..d2aa4eb134 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/ed4dfb506a..c96680035b * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/75c636c54e..60b1cfd539 * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/0f4bac59a3..1591e813b6 * src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/cf8cac9f64..6558e6a3ea * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/e6596746dc..354f6026f1 * src/third_party/googletest/src: https://chromium.googlesource.com/external/github.com/google/googletest.git/+log/59c924bc47..04ee1b4f2a * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/1991e6da4d..4d2eae608a * src/third_party/llvm-build/Release+Asserts_version: Linux_x64/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clang-win-runtime-library-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/llvmobjdump-llvmorg-21-init-6681-g5b36835d-1.tar.xz..Linux_x64/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clang-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clang-tidy-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clang-win-runtime-library-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/clangd-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/llvm-code-coverage-llvmorg-21-init-9266-g09006611-1.tar.xz,Win/llvmobjdump-llvmorg-21-init-9266-g09006611-1.tar.xz * src/third_party/rust-toolchain_version: Linux_x64/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Mac/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Mac_arm64/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Win/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz..Linux_x64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Mac/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Mac_arm64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Win/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/48fe65c657..b2ca46af79 * src/tools/luci-go: git_revision:74734688ffc9d2807cda0d1dc7752197480de29e..git_revision:a9a10995e2889126ace91faf4052949e38c69d33 * src/tools/luci-go: git_revision:74734688ffc9d2807cda0d1dc7752197480de29e..git_revision:a9a10995e2889126ace91faf4052949e38c69d33 * src/tools/luci-go: git_revision:74734688ffc9d2807cda0d1dc7752197480de29e..git_revision:a9a10995e2889126ace91faf4052949e38c69d33 DEPS diff: https://chromium.googlesource.com/chromium/src/+/45bb02a7df..fb0dacbdf6/DEPS No update to Clang. BUG=None Change-Id: I415ca1ab33e9c9387cb4073ba4b7da25838008f3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389203 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44495} --- DEPS | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/DEPS b/DEPS index 90bd872ed7..8a3dc39fb1 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '45bb02a7df727e9aed4e8208d9e3d41c083eb216', + 'chromium_revision': 'fb0dacbdf6d4b50101b1888a7c31b19f0c78dac3', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@b9d38213d963c49b541e7d2e31dc0ea390b80f59', + 'https://chromium.googlesource.com/chromium/src/base@802e7fa5a71086c4b520ee4ebb1030f5425643bd', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@d5793d1e366655bfda43b2cce5e7cb00fd9b5404', + 'https://chromium.googlesource.com/chromium/src/build@d2aa4eb1342a48627e3ea3a174915f34a73bf6e6', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@6d4e129ac68b4c81d98765068ceb53dbf09e99d2', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -88,9 +88,9 @@ deps = { 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@ed4dfb506ae0078335299f9bbec1b0b55aec6cb1', + 'https://chromium.googlesource.com/chromium/src/testing@c96680035b6e74d7f6414510187c6f27055b0838', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@75c636c54ea00117712af2b401cb09fbc71fad6c', + 'https://chromium.googlesource.com/chromium/src/third_party@60b1cfd5394d5b50d835cffadbdae4a88a3de5e5', 'src/buildtools/linux64': { 'packages': [ @@ -323,7 +323,7 @@ deps = { 'src/third_party/clang-format/script': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@37f6e68a107df43b7d7e044fd36a13cbae3413f2', 'src/third_party/compiler-rt/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@cf8cac9f64ff4ac6fd2041a699dd86f717c63586', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@6558e6a3eacafd269527cf2595e0e722f117c1e6', 'src/third_party/libc++/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@62724eb55f2ad7389088265952b1bb62cb89c73b', 'src/third_party/libc++abi/src': @@ -484,7 +484,7 @@ deps = { 'src/third_party/breakpad/breakpad': 'https://chromium.googlesource.com/breakpad/breakpad.git@232a723f5096ab02d53d87931efa485fa77d3b03', 'src/third_party/catapult': - 'https://chromium.googlesource.com/catapult.git@0f4bac59a38c6aa5ad9874778f822d76ea737d35', + 'https://chromium.googlesource.com/catapult.git@1591e813b66e04a5b9d9e8fc5d94e74ef4effaa1', 'src/third_party/ced/src': { 'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5', }, @@ -497,13 +497,13 @@ deps = { 'src/third_party/crc32c/src': 'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558', 'src/third_party/depot_tools': - 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@e6596746dc95fe658b3a5f0924c10e322c7d8e22', + 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@354f6026f14f87466040e3e1878c599138a23bce', 'src/third_party/ffmpeg': 'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@01f23648c6b84de6c0f717fa4e1816f53b9ee72e', 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@1991e6da4d0f0b1b5de038e0948b2636c0bcd525', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@4d2eae608a8d03e233e10f800b95ea28c507bcf4', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -525,7 +525,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/googletest/src': - 'https://chromium.googlesource.com/external/github.com/google/googletest.git@59c924bc471cefea25a7cf7eb40b6c101b170b12', + 'https://chromium.googlesource.com/external/github.com/google/googletest.git@04ee1b4f2aefdffb0135d7cf2a2c519fe50dabe4', 'src/third_party/icu': { 'url': 'https://chromium.googlesource.com/chromium/deps/icu.git@4c8cc4b365a505ce35be1e0bd488476c5f79805d', }, @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@48fe65c657a260da105a98ecd99fa26f10ae641f', + 'https://chromium.googlesource.com/chromium/src/tools@b2ca46af795d7ae3277c3dba9afde61cf5090f4e', 'src/third_party/espresso': { 'packages': [ @@ -793,15 +793,15 @@ deps = { 'packages': [ { 'package': 'infra/tools/luci/cas/${{platform}}', - 'version': 'git_revision:74734688ffc9d2807cda0d1dc7752197480de29e', + 'version': 'git_revision:a9a10995e2889126ace91faf4052949e38c69d33', }, { 'package': 'infra/tools/luci/isolate/${{platform}}', - 'version': 'git_revision:74734688ffc9d2807cda0d1dc7752197480de29e', + 'version': 'git_revision:a9a10995e2889126ace91faf4052949e38c69d33', }, { 'package': 'infra/tools/luci/swarming/${{platform}}', - 'version': 'git_revision:74734688ffc9d2807cda0d1dc7752197480de29e', + 'version': 'git_revision:a9a10995e2889126ace91faf4052949e38c69d33', } ], 'dep_type': 'cipd', From 9ff88506604e7566efca10f3d61a0b8e85ccfc5b Mon Sep 17 00:00:00 2001 From: webrtc-version-updater Date: Thu, 1 May 2025 21:07:49 -0700 Subject: [PATCH 024/292] Update WebRTC code version (2025-05-02T04:07:31). Bug: None Change-Id: I887fc12e55325ef41c6356ae25d147f3eae54634 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389442 Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Cr-Commit-Position: refs/heads/main@{#44496} --- call/version.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/call/version.cc b/call/version.cc index aa15cd0f7d..21c1b34ff7 100644 --- a/call/version.cc +++ b/call/version.cc @@ -13,7 +13,7 @@ namespace webrtc { // The timestamp is always in UTC. -const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-01T04:03:12"; +const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-02T04:07:31"; void LoadWebRTCVersionInRegister() { // Using volatile to instruct the compiler to not optimize `p` away even From 8786355b066096170e54de815d0c8072820a78df Mon Sep 17 00:00:00 2001 From: Junji Watanabe Date: Mon, 28 Apr 2025 21:53:48 +0900 Subject: [PATCH 025/292] build: Generate gn_logs.txt for Siso build support gn_logs.txt file is used to propagate computed GN variables to Siso. Bug: chromium:412968361 Change-Id: Iee451e6022a83a2be190e494f3a785307d337f49 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/388500 Commit-Queue: Christoffer Dewerin Reviewed-by: Mirko Bonadei Reviewed-by: Christoffer Dewerin Auto-Submit: Junji Watanabe Cr-Commit-Position: refs/heads/main@{#44497} --- BUILD.gn | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/BUILD.gn b/BUILD.gn index bb56514e98..2d142310c3 100644 --- a/BUILD.gn +++ b/BUILD.gn @@ -824,3 +824,14 @@ group("poison_environment_construction") { group("poison_software_video_codecs") { } + +if (!build_with_chromium) { + # Write debug logs to gn_logs.txt. + # This is also required for Siso builds. + import("//build/gn_logs.gni") + lines = [ + "Generated during 'gn gen' by //BUILD.gn.", + "", + ] + build_gn_logs + write_file("$root_build_dir/gn_logs.txt", lines) +} From e82968a32195d33a23ee29b3ae6fab82f551ed3d Mon Sep 17 00:00:00 2001 From: Gyuyoung Kim Date: Thu, 1 May 2025 18:09:26 +0900 Subject: [PATCH 026/292] [tvOS] Exclude RTCCameraPreviewView.m/h files in the build RTCCameraPreviewView.m/h files use UIDeviceOrientation to check the device orientation, which is not supported on tvOS. Therefore, this CL excludes those files from the build. Bug: chromium:391914246 Change-Id: Ib7b720dcffd8134273198e3004bc70b61ebe9db2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389320 Reviewed-by: Peter Hanspers Reviewed-by: Anders Lilienthal Commit-Queue: Gyuyoung Kim Cr-Commit-Position: refs/heads/main@{#44498} --- sdk/BUILD.gn | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn index e76b68c92e..b556063e8a 100644 --- a/sdk/BUILD.gn +++ b/sdk/BUILD.gn @@ -9,6 +9,7 @@ import("//third_party/libaom/options.gni") import("../webrtc.gni") if (is_ios) { + import("//build/config/apple/mobile_config.gni") import("//build/config/ios/ios_sdk.gni") import("//build/config/ios/rules.gni") } @@ -167,11 +168,17 @@ if (is_ios || is_mac) { if (is_ios) { sources += [ - "objc/helpers/RTCCameraPreviewView.h", - "objc/helpers/RTCCameraPreviewView.m", "objc/helpers/UIDevice+RTCDevice.h", "objc/helpers/UIDevice+RTCDevice.mm", ] + + if (target_platform != "tvos") { + sources += [ + "objc/helpers/RTCCameraPreviewView.h", + "objc/helpers/RTCCameraPreviewView.m", + ] + } + frameworks += [ "UIKit.framework" ] } } From 626c9f19118aac2e0faf45db73e9101d4502190c Mon Sep 17 00:00:00 2001 From: Harald Alvestrand Date: Fri, 2 May 2025 09:22:12 +0000 Subject: [PATCH 027/292] Don't reset send params when turning send off Bug: chromium:410960672 Change-Id: I0fe574b0ec1849521788ab51bb9bfdfbe7b304b7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389120 Reviewed-by: Evan Shrubsole Commit-Queue: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44499} --- pc/peer_connection_jsep_unittest.cc | 14 ++++++++++++++ pc/rtp_sender.cc | 26 ++++++++++++++++---------- 2 files changed, 30 insertions(+), 10 deletions(-) diff --git a/pc/peer_connection_jsep_unittest.cc b/pc/peer_connection_jsep_unittest.cc index 27aba158e4..b7f62a1729 100644 --- a/pc/peer_connection_jsep_unittest.cc +++ b/pc/peer_connection_jsep_unittest.cc @@ -33,6 +33,7 @@ #include "api/rtp_transceiver_interface.h" #include "api/scoped_refptr.h" #include "api/task_queue/default_task_queue_factory.h" +#include "api/test/rtc_error_matchers.h" #include "api/transport/field_trial_based_config.h" #include "media/base/stream_params.h" #include "p2p/base/p2p_constants.h" @@ -2400,4 +2401,17 @@ TEST_F(PeerConnectionJsepTest, BundleOnlySectionDoesNotNeedRtcpMux) { EXPECT_TRUE(callee->SetRemoteDescription(std::move(offer))); } +// This test is a regression test for crbug.com/410960672 +TEST_F(PeerConnectionJsepTest, OfferRollbackRemoveReoffer) { + auto caller = CreatePeerConnection(); + // Add first video track. + auto sender = caller->AddVideoTrack("foo"); + caller->SetLocalDescription(caller->CreateOffer()); + caller->SetRemoteDescription(caller->CreateRollback()); + RTCError error = caller->pc()->RemoveTrackOrError(sender); + EXPECT_THAT(error, IsRtcOk()); + auto offer = caller->CreateOffer(); + caller->SetLocalDescription(std::move(offer)); +} + } // namespace webrtc diff --git a/pc/rtp_sender.cc b/pc/rtp_sender.cc index 31bbba0493..ef956548b5 100644 --- a/pc/rtp_sender.cc +++ b/pc/rtp_sender.cc @@ -528,17 +528,23 @@ void RtpSenderBase::SetSsrc(uint32_t ssrc) { // we need to copy. RtpParameters current_parameters = media_channel_->GetRtpSendParameters(ssrc_); - RTC_CHECK_GE(current_parameters.encodings.size(), - init_parameters_.encodings.size()); - for (size_t i = 0; i < init_parameters_.encodings.size(); ++i) { - init_parameters_.encodings[i].ssrc = - current_parameters.encodings[i].ssrc; - init_parameters_.encodings[i].rid = current_parameters.encodings[i].rid; - current_parameters.encodings[i] = init_parameters_.encodings[i]; + // SSRC 0 has special meaning as "no stream". + // In this case, current_parameters may have size 0. + if (ssrc != 0) { + RTC_CHECK_GE(current_parameters.encodings.size(), + init_parameters_.encodings.size()); + for (size_t i = 0; i < init_parameters_.encodings.size(); ++i) { + init_parameters_.encodings[i].ssrc = + current_parameters.encodings[i].ssrc; + init_parameters_.encodings[i].rid = + current_parameters.encodings[i].rid; + current_parameters.encodings[i] = init_parameters_.encodings[i]; + } + current_parameters.degradation_preference = + init_parameters_.degradation_preference; + media_channel_->SetRtpSendParameters(ssrc_, current_parameters, + nullptr); } - current_parameters.degradation_preference = - init_parameters_.degradation_preference; - media_channel_->SetRtpSendParameters(ssrc_, current_parameters, nullptr); init_parameters_.encodings.clear(); init_parameters_.degradation_preference = std::nullopt; }); From e05231abb3790df5e6ccbc739c1c2952c8416f98 Mon Sep 17 00:00:00 2001 From: Mirko Bonadei Date: Fri, 2 May 2025 12:20:21 +0000 Subject: [PATCH 028/292] Support GCS dependencies in the DEPS Autoroller. Before this CL, GCS dependencies were parsed but they were not rolled because the script was generating the wrong command for `gclient setdep`. This CL fixes the issue and also adds a custom CL description for GCS dependencies. Bug: None Change-Id: I5515743993211abf7f3a0dcffdf8bb0db05cdf08 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389541 Reviewed-by: Jeremy Leconte Commit-Queue: Mirko Bonadei Cr-Commit-Position: refs/heads/main@{#44500} --- tools_webrtc/autoroller/roll_deps.py | 36 ++++++++++++++++++++++++---- 1 file changed, 31 insertions(+), 5 deletions(-) diff --git a/tools_webrtc/autoroller/roll_deps.py b/tools_webrtc/autoroller/roll_deps.py index 0ffc7b6562..e0af99a309 100755 --- a/tools_webrtc/autoroller/roll_deps.py +++ b/tools_webrtc/autoroller/roll_deps.py @@ -88,6 +88,10 @@ def FindRootPath(): NOTIFY_EMAIL = 'webrtc-trooper@grotations.appspotmail.com' +GCS_OBJECTS_ERROR = ( + 'The number of objects in %s is different between ' + 'Chromium\'s DEPS and WebRTC\'s DEPS. They must be the same.') + sys.path.append(os.path.join(CHECKOUT_ROOT_DIR, 'build')) import find_depot_tools @@ -107,6 +111,8 @@ def FindRootPath(): 'ChangedCipdPackage', 'path package current_version new_version') ChangedVersionEntry = collections.namedtuple( 'ChangedVersionEntry', 'path current_version new_version') +ChangedGcsPackage = collections.namedtuple( + 'ChangedGcsPackage', 'path setdep_arg current_version new_version') ChromiumRevisionUpdate = collections.namedtuple('ChromiumRevisionUpdate', ('current_chromium_rev ' @@ -323,6 +329,24 @@ def AddVersionEntry(vars_subdict): return result +def _FindChangedGcsPackage(path, old_pkg, new_pkg): + assert len(old_pkg.objects) == len(new_pkg.objects), (GCS_OBJECTS_ERROR % + path) + current_version_str = ','.join(x['object_name'] for x in old_pkg.objects) + new_version_str = ','.join(x['object_name'] for x in new_pkg.objects) + if current_version_str != new_version_str: + objects = [ + ','.join([ + o['object_name'], o['sha256sum'], + str(o['size_bytes']), + str(o['generation']) + ]) for o in new_pkg.objects + ] + setdep_arg = '%s@%s' % (path, '?'.join(objects)) + yield ChangedGcsPackage(path, setdep_arg, current_version_str, + new_version_str) + + def _FindChangedCipdPackages(path, old_pkgs, new_pkgs): old_pkgs_names = {p['package'] for p in old_pkgs} new_pkgs_names = {p['package'] for p in new_pkgs} @@ -464,11 +488,8 @@ def CalculateChangedDeps(webrtc_deps, new_cr_deps): if isinstance(cr_deps_entry, GcsDepsEntry): result.extend( - _FindChangedVars( - path, ','.join(x['object_name'] - for x in webrtc_deps_entry.objects), - ','.join(x['object_name'] - for x in cr_deps_entry.objects))) + _FindChangedGcsPackage(path, webrtc_deps_entry, + cr_deps_entry)) continue if isinstance(cr_deps_entry, VersionEntry): @@ -554,6 +575,9 @@ def Section(adjective, deps): if isinstance(c, ChangedCipdPackage): commit_msg.append('* %s: %s..%s' % (c.path, c.current_version, c.new_version)) + elif isinstance(c, ChangedGcsPackage): + commit_msg.append('* %s: %s..%s' % + (c.path, c.current_version, c.new_version)) elif isinstance(c, ChangedVersionEntry): commit_msg.append('* %s_version: %s..%s' % (c.path, c.current_version, c.new_version)) @@ -638,6 +662,8 @@ def UpdateDepsFile(deps_filename, rev_update, changed_deps, new_cr_content): if isinstance(dep, ChangedCipdPackage): package = dep.package.format() # Eliminate double curly brackets update = '%s:%s@%s' % (dep.path, package, dep.new_version) + elif isinstance(dep, ChangedGcsPackage): + update = dep.setdep_arg else: update = '%s@%s' % (dep.path, dep.new_rev) _RunCommand(['gclient', 'setdep', '--revision', update], From 663638e5a6cdd1aac2538c6d265c65848c2f0589 Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Wed, 30 Apr 2025 16:15:41 +0000 Subject: [PATCH 029/292] Deprecate EnableMediaWithDefaultsAndTimeController With PeerConnectionFactoryDependencies acepting Environment, clock can be injected through it, and thus extra function to inject Clock is not needed. Note that custom Clock still should be provided with care as some components rely on global rather than injected clock. Bug: webrtc:42220378 Change-Id: I82b66561cfd29ca0c39e78b1422de0fb636ab12a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389180 Reviewed-by: Per Kjellander Reviewed-by: Artem Titov Commit-Queue: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#44501} --- api/BUILD.gn | 1 - api/enable_media_with_defaults.cc | 4 ---- api/test/create_time_controller.h | 1 + test/pc/e2e/BUILD.gn | 5 ++-- test/pc/e2e/test_peer_factory.cc | 27 ++++++++++------------ test/peer_scenario/BUILD.gn | 5 ++-- test/peer_scenario/peer_scenario_client.cc | 21 ++++++++--------- test/peer_scenario/peer_scenario_client.h | 23 +++++++++++++++--- 8 files changed, 49 insertions(+), 38 deletions(-) diff --git a/api/BUILD.gn b/api/BUILD.gn index b0e44062f4..939a4b2d74 100644 --- a/api/BUILD.gn +++ b/api/BUILD.gn @@ -65,7 +65,6 @@ rtc_source_set("enable_media_with_defaults") { "audio:builtin_audio_processing_builder", "audio_codecs:builtin_audio_decoder_factory", "audio_codecs:builtin_audio_encoder_factory", - "task_queue:default_task_queue_factory", "video_codecs:builtin_video_decoder_factory", "video_codecs:builtin_video_encoder_factory", ] diff --git a/api/enable_media_with_defaults.cc b/api/enable_media_with_defaults.cc index 2ad88849d8..83d6f172f9 100644 --- a/api/enable_media_with_defaults.cc +++ b/api/enable_media_with_defaults.cc @@ -18,16 +18,12 @@ #include "api/enable_media.h" #include "api/peer_connection_interface.h" #include "api/scoped_refptr.h" -#include "api/task_queue/default_task_queue_factory.h" #include "api/video_codecs/builtin_video_decoder_factory.h" #include "api/video_codecs/builtin_video_encoder_factory.h" namespace webrtc { void EnableMediaWithDefaults(PeerConnectionFactoryDependencies& deps) { - if (deps.task_queue_factory == nullptr) { - deps.task_queue_factory = CreateDefaultTaskQueueFactory(); - } if (deps.audio_encoder_factory == nullptr) { deps.audio_encoder_factory = CreateBuiltinAudioEncoderFactory(); } diff --git a/api/test/create_time_controller.h b/api/test/create_time_controller.h index 7d1d9b3800..5572ef0989 100644 --- a/api/test/create_time_controller.h +++ b/api/test/create_time_controller.h @@ -23,6 +23,7 @@ std::unique_ptr CreateSimulatedTimeController(); // Adjusts media `deps` to use clock `time_controller` provides, fills media // related dependencies, and enables media support for a PeerConnectionFactory // created from `deps`. +[[deprecated("Inject Clock through the deps.env instead.")]] void EnableMediaWithDefaultsAndTimeController( TimeController& time_controller, PeerConnectionFactoryDependencies& deps); diff --git a/test/pc/e2e/BUILD.gn b/test/pc/e2e/BUILD.gn index ac87522574..578b9922a0 100644 --- a/test/pc/e2e/BUILD.gn +++ b/test/pc/e2e/BUILD.gn @@ -102,17 +102,18 @@ if (!build_with_chromium) { ":echo_emulation", ":test_peer", "../..:copy_to_file_audio_capturer", - "../../../api:create_time_controller", + "../../../api:enable_media_with_defaults", "../../../api:libjingle_peerconnection_api", "../../../api:scoped_refptr", "../../../api:time_controller", "../../../api/audio:audio_device", + "../../../api/environment", + "../../../api/environment:environment_factory", "../../../api/rtc_event_log:rtc_event_log_factory", "../../../api/task_queue", "../../../api/test/pclf:media_configuration", "../../../api/test/pclf:media_quality_test_params", "../../../api/test/pclf:peer_configurer", - "../../../api/transport:field_trial_based_config", "../../../api/video_codecs:builtin_video_decoder_factory", "../../../api/video_codecs:builtin_video_encoder_factory", "../../../api/video_codecs:video_codecs_api", diff --git a/test/pc/e2e/test_peer_factory.cc b/test/pc/e2e/test_peer_factory.cc index 07b980a084..a251cafdc8 100644 --- a/test/pc/e2e/test_peer_factory.cc +++ b/test/pc/e2e/test_peer_factory.cc @@ -18,16 +18,17 @@ #include "absl/memory/memory.h" #include "absl/strings/string_view.h" #include "api/audio/audio_device.h" +#include "api/enable_media_with_defaults.h" +#include "api/environment/environment.h" +#include "api/environment/environment_factory.h" #include "api/peer_connection_interface.h" #include "api/rtc_event_log/rtc_event_log_factory.h" #include "api/scoped_refptr.h" #include "api/task_queue/task_queue_factory.h" -#include "api/test/create_time_controller.h" #include "api/test/pclf/media_configuration.h" #include "api/test/pclf/media_quality_test_params.h" #include "api/test/pclf/peer_configurer.h" #include "api/test/time_controller.h" -#include "api/transport/field_trial_based_config.h" #include "api/video_codecs/builtin_video_decoder_factory.h" #include "api/video_codecs/builtin_video_encoder_factory.h" #include "api/video_codecs/video_decoder_factory.h" @@ -66,10 +67,6 @@ void SetMandatoryEntities(InjectableComponents* components) { components->pcf_dependencies->event_log_factory = std::make_unique(); } - if (!components->pcf_dependencies->trials) { - components->pcf_dependencies->trials = - std::make_unique(); - } } // Returns mapping from stream label to optional spatial index. @@ -193,8 +190,8 @@ void WrapVideoDecoderFactory( // Creates PeerConnectionFactoryDependencies objects, providing entities // from InjectableComponents::PeerConnectionFactoryComponents. PeerConnectionFactoryDependencies CreatePCFDependencies( + const Environment& env, std::unique_ptr pcf_dependencies, - TimeController& time_controller, scoped_refptr audio_device_module, Thread* signaling_thread, Thread* worker_thread, @@ -207,7 +204,7 @@ PeerConnectionFactoryDependencies CreatePCFDependencies( pcf_deps.network_manager = std::move(pcf_dependencies->network_manager); pcf_deps.event_log_factory = std::move(pcf_dependencies->event_log_factory); - pcf_deps.task_queue_factory = time_controller.CreateTaskQueueFactory(); + pcf_deps.env = env; if (pcf_dependencies->fec_controller_factory != nullptr) { pcf_deps.fec_controller_factory = @@ -220,9 +217,6 @@ PeerConnectionFactoryDependencies CreatePCFDependencies( if (pcf_dependencies->neteq_factory != nullptr) { pcf_deps.neteq_factory = std::move(pcf_dependencies->neteq_factory); } - if (pcf_dependencies->trials != nullptr) { - pcf_deps.trials = std::move(pcf_dependencies->trials); - } // Media dependencies pcf_deps.adm = std::move(audio_device_module); @@ -235,7 +229,7 @@ PeerConnectionFactoryDependencies CreatePCFDependencies( std::move(pcf_dependencies->video_decoder_factory); pcf_deps.audio_encoder_factory = pcf_dependencies->audio_encoder_factory; pcf_deps.audio_decoder_factory = pcf_dependencies->audio_decoder_factory; - EnableMediaWithDefaultsAndTimeController(time_controller, pcf_deps); + EnableMediaWithDefaults(pcf_deps); return pcf_deps; } @@ -296,11 +290,14 @@ std::unique_ptr TestPeerFactory::CreateTestPeer( SetMandatoryEntities(components.get()); params->rtc_configuration.sdp_semantics = SdpSemantics::kUnifiedPlan; + const Environment env = CreateEnvironment( + std::move(components->pcf_dependencies->trials), + time_controller_.GetClock(), time_controller_.GetTaskQueueFactory()); + // Create peer connection factory. scoped_refptr audio_device_module = CreateAudioDeviceModule(params->audio_config, remote_audio_config, - echo_emulation_config, - time_controller_.GetTaskQueueFactory()); + echo_emulation_config, &env.task_queue_factory()); WrapVideoEncoderFactory( params->name.value(), params->video_encoder_bitrate_multiplier, CalculateRequiredSpatialIndexPerStream( @@ -319,7 +316,7 @@ std::unique_ptr TestPeerFactory::CreateTestPeer( } PeerConnectionFactoryDependencies pcf_deps = CreatePCFDependencies( - std::move(components->pcf_dependencies), time_controller_, + env, std::move(components->pcf_dependencies), std::move(audio_device_module), signaling_thread_, components->worker_thread, components->network_thread); scoped_refptr peer_connection_factory = diff --git a/test/peer_scenario/BUILD.gn b/test/peer_scenario/BUILD.gn index 41bcd65f0b..b56607f80a 100644 --- a/test/peer_scenario/BUILD.gn +++ b/test/peer_scenario/BUILD.gn @@ -31,7 +31,7 @@ if (rtc_include_tests) { "../../api:array_view", "../../api:audio_options_api", "../../api:candidate", - "../../api:create_time_controller", + "../../api:enable_media_with_defaults", "../../api:libjingle_peerconnection_api", "../../api:make_ref_counted", "../../api:media_stream_interface", @@ -41,11 +41,11 @@ if (rtc_include_tests) { "../../api:sequence_checker", "../../api:time_controller", "../../api/environment", + "../../api/environment:environment_factory", "../../api/rtc_event_log:rtc_event_log_factory", "../../api/test/network_emulation", "../../api/transport:datagram_transport_interface", "../../api/transport:enums", - "../../api/transport:field_trial_based_config", "../../api/video:video_frame", "../../api/video_codecs:scalability_mode", "../../api/video_codecs:video_codecs_api", @@ -89,6 +89,7 @@ if (rtc_include_tests) { "../../rtc_base:task_queue_for_test", "../../rtc_base:threading", "../../rtc_base/third_party/sigslot", + "../../system_wrappers", "../logging:log_writer", "../network:emulated_network", "../scenario", diff --git a/test/peer_scenario/peer_scenario_client.cc b/test/peer_scenario/peer_scenario_client.cc index 391967f14e..52dedbd900 100644 --- a/test/peer_scenario/peer_scenario_client.cc +++ b/test/peer_scenario/peer_scenario_client.cc @@ -23,7 +23,9 @@ #include "api/audio_options.h" #include "api/candidate.h" #include "api/data_channel_interface.h" +#include "api/enable_media_with_defaults.h" #include "api/environment/environment.h" +#include "api/environment/environment_factory.h" #include "api/jsep.h" #include "api/make_ref_counted.h" #include "api/media_stream_interface.h" @@ -36,10 +38,8 @@ #include "api/sequence_checker.h" #include "api/set_local_description_observer_interface.h" #include "api/set_remote_description_observer_interface.h" -#include "api/test/create_time_controller.h" #include "api/test/network_emulation/network_emulation_interfaces.h" #include "api/test/network_emulation_manager.h" -#include "api/transport/field_trial_based_config.h" #include "api/video/video_frame.h" #include "api/video/video_sink_interface.h" #include "api/video/video_source_interface.h" @@ -241,8 +241,9 @@ PeerScenarioClient::PeerScenarioClient( Thread* signaling_thread, std::unique_ptr log_writer_factory, PeerScenarioClient::Config config) - : endpoints_(CreateEndpoints(net, config.endpoints)), - task_queue_factory_(net->time_controller()->GetTaskQueueFactory()), + : env_(CreateEnvironment(net->time_controller()->GetClock(), + net->time_controller()->GetTaskQueueFactory())), + endpoints_(CreateEndpoints(net, config.endpoints)), signaling_thread_(signaling_thread), log_writer_factory_(std::move(log_writer_factory)), worker_thread_(net->time_controller()->CreateThread("worker")), @@ -282,13 +283,11 @@ PeerScenarioClient::PeerScenarioClient( pcf_deps.worker_thread = worker_thread_.get(); pcf_deps.socket_factory = manager->socket_factory(); pcf_deps.network_manager = manager->ReleaseNetworkManager(); - pcf_deps.task_queue_factory = - net->time_controller()->CreateTaskQueueFactory(); pcf_deps.event_log_factory = std::make_unique(); - pcf_deps.trials = std::make_unique(); + pcf_deps.env = env_; pcf_deps.adm = TestAudioDeviceModule::Create( - task_queue_factory_, + &env_.task_queue_factory(), TestAudioDeviceModule::CreatePulsedNoiseCapturer( config.audio.pulsed_noise->amplitude * std::numeric_limits::max(), @@ -311,7 +310,7 @@ PeerScenarioClient::PeerScenarioClient( OpenH264DecoderTemplateAdapter, Dav1dDecoderTemplateAdapter>>(); } - EnableMediaWithDefaultsAndTimeController(*net->time_controller(), pcf_deps); + EnableMediaWithDefaults(pcf_deps); pcf_deps.fec_controller_factory = nullptr; pcf_deps.network_controller_factory = nullptr; @@ -356,8 +355,8 @@ PeerScenarioClient::VideoSendTrack PeerScenarioClient::CreateVideo( VideoSendTrackConfig config) { RTC_DCHECK_RUN_ON(signaling_thread_); VideoSendTrack res; - auto capturer = CreateFrameGeneratorCapturer(clock(), *task_queue_factory_, - config.generator); + auto capturer = CreateFrameGeneratorCapturer( + clock(), env_.task_queue_factory(), config.generator); res.capturer = capturer.get(); capturer->Init(); res.source = make_ref_counted( diff --git a/test/peer_scenario/peer_scenario_client.h b/test/peer_scenario/peer_scenario_client.h index dfe502ec95..6e2510fda1 100644 --- a/test/peer_scenario/peer_scenario_client.h +++ b/test/peer_scenario/peer_scenario_client.h @@ -14,15 +14,32 @@ #include #include #include +#include #include #include -#include "absl/memory/memory.h" +#include "api/audio_options.h" +#include "api/candidate.h" +#include "api/data_channel_interface.h" +#include "api/environment/environment.h" +#include "api/jsep.h" +#include "api/media_stream_interface.h" #include "api/peer_connection_interface.h" +#include "api/rtp_receiver_interface.h" +#include "api/rtp_sender_interface.h" +#include "api/rtp_transceiver_interface.h" +#include "api/scoped_refptr.h" +#include "api/sequence_checker.h" +#include "api/test/network_emulation/network_emulation_interfaces.h" #include "api/test/network_emulation_manager.h" -#include "api/test/time_controller.h" +#include "api/video/video_frame.h" +#include "api/video/video_sink_interface.h" #include "pc/test/frame_generator_capturer_video_track_source.h" +#include "rtc_base/thread.h" +#include "rtc_base/thread_annotations.h" +#include "system_wrappers/include/clock.h" #include "test/create_frame_generator_capturer.h" +#include "test/frame_generator_capturer.h" #include "test/logging/log_writer.h" namespace webrtc { @@ -159,8 +176,8 @@ class PeerScenarioClient { void AddIceCandidate(std::unique_ptr candidate); private: + const Environment env_; const std::map endpoints_; - TaskQueueFactory* const task_queue_factory_; Thread* const signaling_thread_; const std::unique_ptr log_writer_factory_; const std::unique_ptr worker_thread_; From 6f7701efa5aca7c2493d22e898673b87460ccea7 Mon Sep 17 00:00:00 2001 From: Mirko Bonadei Date: Fri, 2 May 2025 14:26:17 +0000 Subject: [PATCH 030/292] Roll chromium_revision fb0dacbdf6..c07fbad232 (1454341:1454965) This is a partial roll (only rolling clang and rust-toolchain). Without this CL, we cannot land [1], which removes the double roll of clang and Rust toolchains. [1] - https://webrtc-review.googlesource.com/c/src/+/389560 * src/third_party/libc++/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git/+log/62724eb55f..f034fd5662 * src/third_party/llvm-build/Release+Asserts: Linux_x64/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clang-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clang-tidy-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clang-win-runtime-library-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/clangd-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/llvm-code-coverage-llvmorg-21-init-6681-g5b36835d-1.tar.xz,Win/llvmobjdump-llvmorg-21-init-6681-g5b36835d-1.tar.xz..Linux_x64/clang-llvmorg-21-init-9266-g09006611-2.tar.xz,Linux_x64/clang-tidy-llvmorg-21-init-9266-g09006611-2.tar.xz,Linux_x64/clangd-llvmorg-21-init-9266-g09006611-2.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-2.tar.xz,Linux_x64/llvmobjdump-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac/clang-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac/clang-mac-runtime-library-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac/clang-tidy-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac/clangd-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac/llvm-code-coverage-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac_arm64/clang-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac_arm64/clang-tidy-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac_arm64/clangd-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-2.tar.xz,Win/clang-llvmorg-21-init-9266-g09006611-2.tar.xz,Win/clang-tidy-llvmorg-21-init-9266-g09006611-2.tar.xz,Win/clang-win-runtime-library-llvmorg-21-init-9266-g09006611-2.tar.xz,Win/clangd-llvmorg-21-init-9266-g09006611-2.tar.xz,Win/llvm-code-coverage-llvmorg-21-init-9266-g09006611-2.tar.xz,Win/llvmobjdump-llvmorg-21-init-9266-g09006611-2.tar.xz * src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/bf95cc5da6..fa2469dd9d * src/third_party/rust-toolchain: Linux_x64/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Mac/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Mac_arm64/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz,Win/rust-toolchain-9fcc9cf4a202aadfe1f44722b39c83536eba3dba-2-llvmorg-21-init-1655-g7b473dfe.tar.xz..Linux_x64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Mac/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Mac_arm64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Win/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz DEPS diff: https://chromium.googlesource.com/chromium/src/+/fb0dacbdf6..c07fbad232/DEPS BUG=None Change-Id: Id3d9abb9bc2a5a6f38ab5a570913a1f3b8dbf360 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389583 Commit-Queue: Artem Titov Commit-Queue: Mirko Bonadei Auto-Submit: Mirko Bonadei Reviewed-by: Artem Titov Cr-Commit-Position: refs/heads/main@{#44502} --- DEPS | 196 +++++++++++++++++++++++++++++------------------------------ 1 file changed, 98 insertions(+), 98 deletions(-) diff --git a/DEPS b/DEPS index 8a3dc39fb1..dd217306ea 100644 --- a/DEPS +++ b/DEPS @@ -142,143 +142,143 @@ deps = { 'objects': [ { # The Android libclang_rt.builtins libraries are currently only included in the Linux clang package. - 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'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@62724eb55f2ad7389088265952b1bb62cb89c73b', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@f034fd5662d033c281ab6a9b45164066ddd18809', 'src/third_party/libc++abi/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@f2a7f2987f9dcdf8b04c2d8cd4dcb186641a7c3e', 'src/third_party/llvm-libc/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@bf95cc5da608540ccddbd22f1866d713eaba3ac8', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@fa2469dd9d05382f04e927456ca3c482c3d705e9', 'src/third_party/libunwind/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@81e2cb40a70de2b6978e6d8658891ded9a77f7e3', From b8404180d8136c9fa19f32acdf7a4737505a819c Mon Sep 17 00:00:00 2001 From: SamingLin Date: Fri, 18 Apr 2025 18:48:38 +0800 Subject: [PATCH 031/292] Add SDP munging detection for transceiver direction Changing direction of transceiver is unexpected in most cases. Bug: webrtc:40567530 Change-Id: I7b8c4132ac04bf20649ce6152d4289de23a7a60c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/387180 Reviewed-by: Harald Alvestrand Reviewed-by: Evan Shrubsole Commit-Queue: Philipp Hancke Reviewed-by: Philipp Hancke Cr-Commit-Position: refs/heads/main@{#44503} --- AUTHORS | 1 + api/uma_metrics.h | 1 + pc/sdp_munging_detector.cc | 6 ++++++ pc/sdp_offer_answer_unittest.cc | 22 ++++++++++++++++++++++ 4 files changed, 30 insertions(+) diff --git a/AUTHORS b/AUTHORS index 0599f49da1..62ae7720b3 100644 --- a/AUTHORS +++ b/AUTHORS @@ -122,6 +122,7 @@ Robert Mader Robert Mader Robert Nagy Ryan Yoakum +Saming Lin Samuel Attard Sarah Thompson Satender Saroha diff --git a/api/uma_metrics.h b/api/uma_metrics.h index 1525f08526..980cb883fd 100644 --- a/api/uma_metrics.h +++ b/api/uma_metrics.h @@ -195,6 +195,7 @@ enum SdpMungingType { kPayloadTypes = 26, kSsrcs = 27, kIceOptionsRenomination = 28, + kDirection = 29, // RTP header extension munging. kRtpHeaderExtensionRemoved = 40, kRtpHeaderExtensionAdded = 41, diff --git a/pc/sdp_munging_detector.cc b/pc/sdp_munging_detector.cc index 2497a3066d..e54c228d5d 100644 --- a/pc/sdp_munging_detector.cc +++ b/pc/sdp_munging_detector.cc @@ -444,6 +444,12 @@ SdpMungingType DetermineSdpMungingType( } } + if (last_created_media_description->direction() != + media_description_to_set->direction()) { + RTC_LOG(LS_WARNING) << "SDP munging: transceiver direction modified."; + return SdpMungingType::kDirection; + } + // Validate media streams. if (last_created_media_description->streams().size() != media_description_to_set->streams().size()) { diff --git a/pc/sdp_offer_answer_unittest.cc b/pc/sdp_offer_answer_unittest.cc index 16c3c22de3..20567b5f81 100644 --- a/pc/sdp_offer_answer_unittest.cc +++ b/pc/sdp_offer_answer_unittest.cc @@ -2111,6 +2111,28 @@ TEST_F(SdpOfferAnswerMungingTest, RemoveContent) { ElementsAre(Pair(SdpMungingType::kNumberOfContents, 1))); } +TEST_F(SdpOfferAnswerMungingTest, TransceiverDirection) { + auto pc = CreatePeerConnection(); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + auto direction = media_description->direction(); + if (direction == RtpTransceiverDirection::kInactive) { + media_description->set_direction(RtpTransceiverDirection::kSendRecv); + } else { + media_description->set_direction(RtpTransceiverDirection::kInactive); + } + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kDirection, 1))); +} + TEST_F(SdpOfferAnswerMungingTest, Mid) { auto pc = CreatePeerConnection(); pc->AddAudioTrack("audio_track", {}); From cd107c00ae6598f55c636a77be83987607afa577 Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Wed, 30 Apr 2025 10:40:43 -0700 Subject: [PATCH 032/292] Remove DTLS record parsing utils since we no longer use the record numbers for acks BUG=webrtc:367395350 Change-Id: Iccee634225fe27dc23337519ab5c699b654d9672 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389161 Reviewed-by: Jonas Oreland Reviewed-by: Harald Alvestrand Commit-Queue: Philipp Hancke Cr-Commit-Position: refs/heads/main@{#44504} --- p2p/dtls/dtls_utils.cc | 91 --------------- p2p/dtls/dtls_utils_unittest.cc | 181 ------------------------------ test/fuzzers/BUILD.gn | 11 -- test/fuzzers/dtls_utils_fuzzer.cc | 21 ---- 4 files changed, 304 deletions(-) delete mode 100644 test/fuzzers/dtls_utils_fuzzer.cc diff --git a/p2p/dtls/dtls_utils.cc b/p2p/dtls/dtls_utils.cc index c1b3a0ab10..226369a968 100644 --- a/p2p/dtls/dtls_utils.cc +++ b/p2p/dtls/dtls_utils.cc @@ -13,13 +13,11 @@ #include #include #include -#include #include #include "absl/container/flat_hash_set.h" #include "api/array_view.h" #include "rtc_base/buffer.h" -#include "rtc_base/byte_buffer.h" #include "rtc_base/checks.h" #include "rtc_base/crc32.h" @@ -28,11 +26,6 @@ namespace { const uint8_t kDtlsChangeCipherSpecRecord = 20; const uint8_t kDtlsHandshakeRecord = 22; -// https://www.rfc-editor.org/rfc/rfc9147.html#section-4 -const uint8_t kFixedBitmask = 0b00100000; -const uint8_t kConnectionBitmask = 0b00010000; -const uint8_t kSequenceNumberBitmask = 0b00001000; -const uint8_t kLengthPresentBitmask = 0b00000100; } // namespace namespace webrtc { @@ -62,90 +55,6 @@ bool IsDtlsHandshakePacket(ArrayView payload) { payload[0] == kDtlsChangeCipherSpecRecord); } -// Returns a (unsorted) list of (msg_seq) received as part of the handshake. -std::optional> GetDtlsHandshakeAcks( - ArrayView dtls_packet) { - std::vector acks; - ByteBufferReader record_buf(dtls_packet); - // https://datatracker.ietf.org/doc/html/rfc6347#section-4.1 - while (record_buf.Length() >= kDtlsRecordHeaderLen) { - uint8_t content_type; - uint64_t epoch_and_seq; - uint16_t len; - // Read content_type(1). - if (!record_buf.ReadUInt8(&content_type)) { - return std::nullopt; - } - - // DTLS 1.3 rules: - // https://www.rfc-editor.org/rfc/rfc9147.html#section-4.1 - if ((content_type & kFixedBitmask) == kFixedBitmask) { - // Interpret as DTLSCipherText: - // https://www.rfc-editor.org/rfc/rfc9147.html#appendix-A.1 - // We assume no connection id is used so C must be 0. - if ((content_type & kConnectionBitmask) != 0) { - return std::nullopt; - } - // Skip sequence_number(1 or 2 bytes depending on S bit). - if (!record_buf.Consume((content_type & kSequenceNumberBitmask) == - kSequenceNumberBitmask - ? 2 - : 1)) { - return std::nullopt; - } - // If the L bit is set, consume the 16 bit length field. - if ((content_type & kLengthPresentBitmask) == kLengthPresentBitmask) { - if (!(record_buf.ReadUInt16(&len) && record_buf.Consume(len))) { - return std::nullopt; - } - } - // DTLSCipherText is encrypted so we can not read it. - continue; - } - // Skip version(2), read epoch+seq(2+6), read len(2) - if (!(record_buf.Consume(2) && record_buf.ReadUInt64(&epoch_and_seq) && - record_buf.ReadUInt16(&len) && record_buf.Length() >= len)) { - return std::nullopt; - } - if (content_type != kDtlsHandshakeRecord) { - record_buf.Consume(len); - continue; - } - // Epoch 1+ is encrypted so we can not parse it. - if (epoch_and_seq >> 6 != 0) { - record_buf.Consume(len); - continue; - } - - // https://www.rfc-editor.org/rfc/rfc6347.html#section-4.2.2 - ByteBufferReader handshake_buf(record_buf.DataView().subview(0, len)); - while (handshake_buf.Length() > 0) { - uint16_t msg_seq; - uint32_t fragment_len; - uint32_t fragment_offset; - // Skip msg_type(1) and length(3), read msg_seq(2), skip - // fragment_offset(3), read fragment_length(3) and consume it. - if (!(handshake_buf.Consume(1 + 3) && - handshake_buf.ReadUInt16(&msg_seq) && - handshake_buf.ReadUInt24(&fragment_offset) && - handshake_buf.ReadUInt24(&fragment_len) && - handshake_buf.Consume(fragment_len))) { - return std::nullopt; - } - acks.push_back(msg_seq); - // Advance outer buffer. - record_buf.Consume(12 + fragment_len); - } - RTC_DCHECK(handshake_buf.Length() == 0); - } - - // Should have consumed everything. - if (record_buf.Length() != 0) { - return std::nullopt; - } - return acks; -} - uint32_t ComputeDtlsPacketHash(ArrayView dtls_packet) { return webrtc::ComputeCrc32(dtls_packet.data(), dtls_packet.size()); } diff --git a/p2p/dtls/dtls_utils_unittest.cc b/p2p/dtls/dtls_utils_unittest.cc index 5dc15621c9..15d4739d4e 100644 --- a/p2p/dtls/dtls_utils_unittest.cc +++ b/p2p/dtls/dtls_utils_unittest.cc @@ -11,195 +11,14 @@ #include "p2p/dtls/dtls_utils.h" #include -#include #include #include "absl/container/flat_hash_set.h" #include "api/array_view.h" -#include "test/gmock.h" #include "test/gtest.h" namespace webrtc { -TEST(DtlsUtils, GetDtlsHandshakeAcksRejectsTooShort) { - std::vector packet = { - 0x16, 0xfe, 0xff, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0xde, 0xad // Length given but bytes not present. - }; - EXPECT_FALSE(GetDtlsHandshakeAcks(packet)); -} - -TEST(DtlsUtils, GetDtlsHandshakeAcksRejectsInvalidContent) { - std::vector packet = {0x16, 0xfe, 0xff, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, - // Correct length given but data is garbage. - 0x04, 0xde, 0xad, 0xbe, 0xef}; - EXPECT_FALSE(GetDtlsHandshakeAcks(packet)); -} - -TEST(DtlsUtils, GetDtlsHandshakeAcksRejectsTrailingData) { - std::vector packet = { - 0x16, 0xfe, 0xff, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x0c, 0x0e, 0x00, 0x00, 0x00, 0x00, - 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, // Server hello done. - 0xde, 0xad, 0xbe, 0xef // Trailing data. - }; - EXPECT_FALSE(GetDtlsHandshakeAcks(packet)); -} - -TEST(DtlsUtils, GetDtlsHandshakeAcksBasic) { - std::vector packet = { - 0x16, 0xfe, 0xff, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x0c, 0x0e, 0x00, 0x00, 0x00, 0x00, - 0xac, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, // Server hello done. - }; - std::optional> acks = GetDtlsHandshakeAcks(packet); - ASSERT_TRUE(acks); - EXPECT_EQ(acks->size(), 1u); - EXPECT_THAT(*acks, ::testing::ElementsAreArray({0xac})); -} - -TEST(DtlsUtils, GetDtlsHandshakeAcksPackedRecords) { - // Flight two from server to client but with fragment packing per - // https://boringssl.googlesource.com/boringssl/+/5245371a08528f7fb21ab20bd7a479d8e395b61c - std::vector packet = { - 0x16, 0xfe, 0xfd, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, - 0x43, 0x02, 0x00, 0x00, 0x60, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x60, 0xfe, 0xfd, 0x67, 0x2a, 0x87, 0x84, 0xf8, 0xe5, 0xbc, 0xe5, 0xd1, - 0x2b, 0xfe, 0x53, 0x20, 0xd2, 0xd4, 0x53, 0xa5, 0xbe, 0xd8, 0x38, 0x58, - 0x91, 0xdf, 0x76, 0x21, 0x81, 0x60, 0x7c, 0x6d, 0x8c, 0xdb, 0x93, 0x20, - 0x91, 0xc8, 0xf6, 0x9c, 0xaa, 0xbe, 0x79, 0xa3, 0x28, 0xa6, 0x84, 0xc9, - 0xfa, 0xee, 0x59, 0x22, 0x5d, 0xe2, 0x11, 0x28, 0xf4, 0x80, 0xd6, 0x1a, - 0x3a, 0xb5, 0x3d, 0xb6, 0x61, 0x74, 0xb6, 0x1d, 0xc0, 0x2b, 0x00, 0x00, - 0x18, 0x00, 0x17, 0x00, 0x00, 0xff, 0x01, 0x00, 0x01, 0x00, 0x00, 0x0b, - 0x00, 0x02, 0x01, 0x00, 0x00, 0x0e, 0x00, 0x05, 0x00, 0x02, 0x00, 0x01, - 0x00, 0x0b, 0x00, 0x01, 0x1f, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x01, - 0x1f, 0x00, 0x01, 0x1c, 0x00, 0x01, 0x19, 0x30, 0x82, 0x01, 0x15, 0x30, - 0x81, 0xbd, 0xa0, 0x03, 0x02, 0x01, 0x02, 0x02, 0x09, 0x00, 0x9d, 0x2a, - 0x69, 0x9b, 0x1d, 0x5a, 0x38, 0xe5, 0x30, 0x0a, 0x06, 0x08, 0x2a, 0x86, - 0x48, 0xce, 0x3d, 0x04, 0x03, 0x02, 0x30, 0x11, 0x31, 0x0f, 0x30, 0x0d, - 0x06, 0x03, 0x55, 0x04, 0x03, 0x0c, 0x06, 0x57, 0x65, 0x62, 0x52, 0x54, - 0x43, 0x30, 0x1e, 0x17, 0x0d, 0x32, 0x34, 0x31, 0x31, 0x30, 0x34, 0x32, - 0x31, 0x30, 0x30, 0x35, 0x31, 0x5a, 0x17, 0x0d, 0x32, 0x34, 0x31, 0x32, - 0x30, 0x35, 0x32, 0x31, 0x30, 0x30, 0x35, 0x31, 0x5a, 0x30, 0x11, 0x31, - 0x0f, 0x30, 0x0d, 0x06, 0x03, 0x55, 0x04, 0x03, 0x0c, 0x06, 0x57, 0x65, - 0x62, 0x52, 0x54, 0x43, 0x30, 0x59, 0x30, 0x13, 0x06, 0x07, 0x2a, 0x86, - 0x48, 0xce, 0x3d, 0x02, 0x01, 0x06, 0x08, 0x2a, 0x86, 0x48, 0xce, 0x3d, - 0x03, 0x01, 0x07, 0x03, 0x42, 0x00, 0x04, 0x1e, 0xd8, 0xad, 0x96, 0x82, - 0xd0, 0xfb, 0xc8, 0xaa, 0xff, 0x84, 0x40, 0x84, 0xfc, 0x1e, 0x4a, 0xfd, - 0x8b, 0xfc, 0x13, 0xbb, 0xee, 0x93, 0xea, 0x91, 0x55, 0x61, 0x7d, 0xc3, - 0x96, 0x66, 0x38, 0x6d, 0x51, 0x59, 0x57, 0xbd, 0xc3, 0xd2, 0x03, 0xf4, - 0xde, 0x48, 0x3f, 0x61, 0x5e, 0x59, 0x2b, 0xfa, 0xfe, 0x68, 0xc0, 0x98, - 0xa3, 0x33, 0xe7, 0xd6, 0xb4, 0x0e, 0xbe, 0x56, 0x48, 0x50, 0x4b, 0x30, - 0x0a, 0x06, 0x08, 0x2a, 0x86, 0x48, 0xce, 0x3d, 0x04, 0x03, 0x02, 0x03, - 0x47, 0x00, 0x30, 0x44, 0x02, 0x20, 0x4d, 0xff, 0x9f, 0xf3, 0xc4, 0x08, - 0x15, 0xe0, 0xdd, 0x76, 0x64, 0x0d, 0x50, 0x42, 0x30, 0xbb, 0xf7, 0xca, - 0x78, 0xff, 0xe7, 0x86, 0x05, 0x0f, 0x23, 0x6e, 0xd2, 0x69, 0xd3, 0xc5, - 0xbd, 0xaa, 0x02, 0x20, 0x43, 0x71, 0x52, 0x2f, 0x74, 0x25, 0x78, 0xcd, - 0x62, 0x62, 0x62, 0x0b, 0xbf, 0x76, 0x35, 0xe1, 0xfe, 0x8c, 0x03, 0x6b, - 0x56, 0xb8, 0x96, 0x1f, 0xb1, 0x3a, 0x9f, 0xd9, 0x78, 0x05, 0x66, 0xa7, - 0x0c, 0x00, 0x00, 0x6f, 0x00, 0x02, 0x00, 0x00, 0x00, 0x00, 0x00, 0x6f, - 0x03, 0x00, 0x1d, 0x20, 0xf5, 0x50, 0xad, 0x14, 0x55, 0xd1, 0xbc, 0x82, - 0xa8, 0xb0, 0x2b, 0x81, 0x3d, 0x18, 0xf4, 0xba, 0x11, 0x54, 0xbb, 0x24, - 0x8b, 0x07, 0xa7, 0x17, 0xf1, 0x33, 0xca, 0x45, 0xc0, 0x6a, 0x16, 0x0a, - 0x04, 0x03, 0x00, 0x47, 0x30, 0x45, 0x02, 0x21, 0x00, 0xad, 0xb6, 0x59, - 0x0c, 0xe0, 0x56, 0x42, 0xb8, 0x9f, 0x40, 0x43, 0xd3, 0x7f, 0x9f, 0xa0, - 0x1d, 0xbc, 0x78, 0xf5, 0xc3, 0x38, 0x99, 0x02, 0xde, 0x11, 0x85, 0x0f, - 0x50, 0xd6, 0x5b, 0x82, 0x7c, 0x02, 0x20, 0x57, 0x2e, 0x0a, 0x82, 0xf7, - 0x14, 0xb6, 0xd6, 0xb2, 0x4b, 0xd7, 0x1a, 0xd6, 0x1b, 0xc2, 0xf6, 0xc2, - 0x4f, 0x3f, 0xe2, 0x8a, 0x06, 0x97, 0xf3, 0x84, 0xc8, 0x60, 0xf1, 0xab, - 0x2d, 0x29, 0xaf, 0x0d, 0x00, 0x00, 0x19, 0x00, 0x03, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x19, 0x02, 0x01, 0x40, 0x00, 0x12, 0x04, 0x03, 0x08, 0x04, - 0x04, 0x01, 0x05, 0x03, 0x08, 0x05, 0x05, 0x01, 0x08, 0x06, 0x06, 0x01, - 0x02, 0x01, 0x00, 0x00, 0x0e, 0x00, 0x00, 0x00, 0x00, 0x04, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00}; - - std::optional> acks = GetDtlsHandshakeAcks(packet); - ASSERT_TRUE(acks); - EXPECT_EQ(acks->size(), 5u); - EXPECT_THAT(*acks, ::testing::ElementsAreArray({0, 1, 2, 3, 4})); - - std::vector packet2 = { - 0x16, 0xfe, 0xfd, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x6c, 0x02, 0x00, 0x00, 0x60, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x60, 0xfe, 0xfd, 0x00, 0x00, 0x00, 0x00, 0xb2, 0xeb, 0x05, 0xc9, 0xba, - 0x39, 0xd0, 0xf6, 0x4b, 0xc9, 0x7e, 0xee, 0x57, 0xfc, 0x2b, 0x90, 0x93, - 0x09, 0xfd, 0x05, 0x80, 0xb3, 0xf5, 0xfc, 0x06, 0x68, 0x38, 0xa8, 0x20, - 0x42, 0x1d, 0x78, 0xe1, 0x97, 0x73, 0x55, 0x0a, 0x16, 0x2d, 0xc1, 0x3e, - 0x4f, 0x71, 0x55, 0xb4, 0x9f, 0xf8, 0x61, 0xe1, 0xbd, 0xe3, 0xf2, 0x2e, - 0x40, 0x29, 0x30, 0x58, 0x37, 0x26, 0x0d, 0xe8, 0xc0, 0x2b, 0x00, 0x00, - 0x18, 0x00, 0x17, 0x00, 0x00, 0xff, 0x01, 0x00, 0x01, 0x00, 0x00, 0x0b, - 0x00, 0x02, 0x01, 0x00, 0x00, 0x0e, 0x00, 0x05, 0x00, 0x02, 0x00, 0x01, - 0x00, // - 0x16, 0xfe, 0xfd, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x01, - 0x28, 0x0b, 0x00, 0x01, 0x1c, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x01, - 0x1c, 0x00, 0x01, 0x19, 0x00, 0x01, 0x16, 0x30, 0x82, 0x01, 0x12, 0x30, - 0x81, 0xb8, 0xa0, 0x03, 0x02, 0x01, 0x02, 0x02, 0x08, 0x15, 0xc9, 0xcc, - 0xd0, 0x55, 0x57, 0xa2, 0x32, 0x30, 0x0a, 0x06, 0x08, 0x2a, 0x86, 0x48, - 0xce, 0x3d, 0x04, 0x03, 0x02, 0x30, 0x0f, 0x31, 0x0d, 0x30, 0x0b, 0x06, - 0x03, 0x55, 0x04, 0x03, 0x0c, 0x04, 0x74, 0x65, 0x73, 0x74, 0x30, 0x1e, - 0x17, 0x0d, 0x32, 0x34, 0x31, 0x31, 0x30, 0x35, 0x32, 0x30, 0x35, 0x33, - 0x34, 0x38, 0x5a, 0x17, 0x0d, 0x32, 0x34, 0x31, 0x32, 0x30, 0x36, 0x32, - 0x30, 0x35, 0x33, 0x34, 0x38, 0x5a, 0x30, 0x0f, 0x31, 0x0d, 0x30, 0x0b, - 0x06, 0x03, 0x55, 0x04, 0x03, 0x0c, 0x04, 0x74, 0x65, 0x73, 0x74, 0x30, - 0x59, 0x30, 0x13, 0x06, 0x07, 0x2a, 0x86, 0x48, 0xce, 0x3d, 0x02, 0x01, - 0x06, 0x08, 0x2a, 0x86, 0x48, 0xce, 0x3d, 0x03, 0x01, 0x07, 0x03, 0x42, - 0x00, 0x04, 0x8f, 0x43, 0xeb, 0x7b, 0x88, 0x73, 0x4f, 0xe7, 0x69, 0x06, - 0x81, 0xb6, 0xb9, 0xf3, 0xca, 0x73, 0x32, 0x69, 0xb2, 0xc5, 0xe6, 0x4e, - 0xf0, 0x8c, 0xf4, 0xdd, 0x4e, 0x5b, 0xea, 0x06, 0x52, 0x94, 0x9a, 0x12, - 0x77, 0x11, 0xde, 0xf9, 0x12, 0x9a, 0xeb, 0x3c, 0x7c, 0xe4, 0xcf, 0x58, - 0x4c, 0x74, 0x44, 0x84, 0x0a, 0x84, 0xeb, 0xe6, 0xa4, 0xd5, 0xd3, 0x06, - 0xca, 0x52, 0x15, 0x7e, 0xeb, 0x19, 0x30, 0x0a, 0x06, 0x08, 0x2a, 0x86, - 0x48, 0xce, 0x3d, 0x04, 0x03, 0x02, 0x03, 0x49, 0x00, 0x30, 0x46, 0x02, - 0x21, 0x00, 0xab, 0xc7, 0x06, 0x7e, 0x36, 0x9b, 0xad, 0xe0, 0x26, 0x61, - 0x6b, 0x59, 0xa0, 0x1c, 0x70, 0x0c, 0xa6, 0xd3, 0xff, 0x8a, 0xc7, 0xba, - 0xe4, 0x23, 0x0a, 0x8b, 0x22, 0x82, 0xcd, 0x5a, 0x5c, 0x56, 0x02, 0x21, - 0x00, 0xc7, 0xe8, 0x57, 0x04, 0xb5, 0x44, 0x69, 0x42, 0xa2, 0xa2, 0x1e, - 0xde, 0x7f, 0xc4, 0x44, 0x98, 0xa4, 0x5c, 0x84, 0x41, 0xa1, 0x31, 0x38, - 0x3c, 0xe5, 0x4f, 0xf5, 0xc0, 0xa9, 0xa8, 0xbc, 0x16, // - 0x16, 0xfe, 0xfd, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, - 0x7a, 0x0c, 0x00, 0x00, 0x6e, 0x00, 0x02, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x6e, 0x03, 0x00, 0x1d, 0x20, 0x89, 0xb9, 0xeb, 0x39, 0x29, 0xa0, 0x31, - 0x08, 0x9a, 0xbf, 0xc3, 0xc0, 0x20, 0x60, 0xbb, 0xea, 0x73, 0x19, 0xcf, - 0x63, 0xe4, 0x5a, 0xa8, 0xa9, 0x56, 0x77, 0xe8, 0x81, 0x48, 0xae, 0x9f, - 0x34, 0x04, 0x03, 0x00, 0x46, 0x30, 0x44, 0x02, 0x20, 0x23, 0x34, 0xc6, - 0x39, 0x94, 0x84, 0xcc, 0x67, 0xeb, 0x44, 0xf9, 0xc3, 0x5c, 0x52, 0xb3, - 0x99, 0x52, 0xf7, 0x4f, 0xff, 0x8b, 0xc5, 0xea, 0xb5, 0xd0, 0xf9, 0x36, - 0xb3, 0xe6, 0xfc, 0x37, 0x50, 0x02, 0x20, 0x4c, 0xe2, 0x29, 0xf5, 0x4a, - 0x4c, 0x7a, 0x01, 0x37, 0xce, 0xc1, 0xb0, 0x15, 0x23, 0xfd, 0xa5, 0xd9, - 0xac, 0x75, 0xcb, 0x55, 0x56, 0x99, 0x97, 0xe3, 0x13, 0xbd, 0x5b, 0xcc, - 0x5d, 0x0c, - 0xa8, // - 0x16, 0xfe, 0xfd, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x03, 0x00, - 0x25, 0x0d, 0x00, 0x00, 0x19, 0x00, 0x03, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x19, 0x02, 0x01, 0x40, 0x00, 0x12, 0x04, 0x03, 0x08, 0x04, 0x04, 0x01, - 0x05, 0x03, 0x08, 0x05, 0x05, 0x01, 0x08, 0x06, 0x06, 0x01, 0x02, 0x01, - 0x00, 0x00, // - 0x16, 0xfe, 0xfd, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x04, 0x00, - 0x0c, 0x0e, 0x00, 0x00, 0x00, 0x00, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00}; - - std::optional> acks2 = GetDtlsHandshakeAcks(packet2); - ASSERT_TRUE(acks2); - EXPECT_EQ(acks2->size(), 5u); - EXPECT_THAT(*acks2, ::testing::ElementsAreArray({0, 1, 2, 3, 4})); -} - -TEST(DtlsUtils, GetDtls13HandshakeAcks) { - // DTLS 1.3 encrypted data, captured with Wireshark. This is a single - // encrypted record which can not be parsed and should be skipped. - std::vector packet = { - 0x2f, 0x5b, 0x4c, 0x00, 0x23, 0x47, 0xab, 0xe7, 0x90, 0x96, - 0xc0, 0xac, 0x2f, 0x25, 0x40, 0x35, 0x35, 0xa3, 0x81, 0x50, - 0x0c, 0x38, 0x0a, 0xf6, 0xd4, 0xd5, 0x7d, 0xbe, 0x9a, 0xa3, - 0xcb, 0xcb, 0x67, 0xb0, 0x77, 0x79, 0x8b, 0x48, 0x60, 0xf8, - }; - - std::optional> acks = GetDtlsHandshakeAcks(packet); - ASSERT_TRUE(acks); - EXPECT_EQ(acks->size(), 0u); -} - std::vector ToVector(ArrayView array) { return std::vector(array.begin(), array.end()); } diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn index 60a9d18b99..c57a1409a3 100644 --- a/test/fuzzers/BUILD.gn +++ b/test/fuzzers/BUILD.gn @@ -105,17 +105,6 @@ webrtc_fuzzer_test("vp9_depacketizer_fuzzer") { ] } -webrtc_fuzzer_test("dtls_utils_fuzzer") { - sources = [ "dtls_utils_fuzzer.cc" ] - deps = [ - "../../api:array_view", - "../../p2p:dtls_utils", - ] - - # Seed from boringssl DTLS corpus. - seed_corpus = "//third_party/boringssl/src/fuzz/dtls_client_corpus" -} - webrtc_fuzzer_test("vp8_qp_parser_fuzzer") { sources = [ "vp8_qp_parser_fuzzer.cc" ] deps = [ diff --git a/test/fuzzers/dtls_utils_fuzzer.cc b/test/fuzzers/dtls_utils_fuzzer.cc deleted file mode 100644 index 622955864e..0000000000 --- a/test/fuzzers/dtls_utils_fuzzer.cc +++ /dev/null @@ -1,21 +0,0 @@ -/* - * Copyright (c) 2024 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ -#include -#include - -#include "api/array_view.h" -#include "p2p/dtls/dtls_utils.h" - -namespace webrtc { -void FuzzOneInput(const uint8_t* data, size_t size) { - webrtc::GetDtlsHandshakeAcks(webrtc::MakeArrayView(data, size)); -} - -} // namespace webrtc From 6dd6dce2ca04201280e3d9313b18591f11cbc2db Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Fri, 2 May 2025 14:03:24 +0000 Subject: [PATCH 033/292] Provide Environment when AudioDeviceModule created for testing purpose Bug: webrtc:413413572 Change-Id: Ic9425b0d787fc5ec03aa3c012483036a72ce89fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389140 Commit-Queue: Danil Chapovalov Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44505} --- call/call_perf_tests.cc | 3 +- modules/audio_device/BUILD.gn | 7 +++ .../audio_device/include/test_audio_device.cc | 16 ++++++- .../audio_device/include/test_audio_device.h | 10 +++- .../include/test_audio_device_unittest.cc | 30 ++++++++---- test/BUILD.gn | 5 ++ test/call_test.cc | 47 ++++++++++++++++--- test/network/BUILD.gn | 3 ++ test/network/network_emulation_pc_unittest.cc | 12 ++--- test/pc/e2e/test_peer_factory.cc | 11 ++--- test/peer_scenario/peer_scenario_client.cc | 2 +- test/scenario/BUILD.gn | 2 + test/scenario/call_client.cc | 30 ++++++++++-- video/video_quality_test.cc | 3 +- 14 files changed, 142 insertions(+), 39 deletions(-) diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc index b72558d978..17e988e76c 100644 --- a/call/call_perf_tests.cc +++ b/call/call_perf_tests.cc @@ -237,8 +237,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec, metrics::Reset(); scoped_refptr fake_audio_device = TestAudioDeviceModule::Create( - &env().task_queue_factory(), - TestAudioDeviceModule::CreatePulsedNoiseCapturer(256, 48000), + env(), TestAudioDeviceModule::CreatePulsedNoiseCapturer(256, 48000), TestAudioDeviceModule::CreateDiscardRenderer(48000), audio_rtp_speed); EXPECT_EQ(0, fake_audio_device->Init()); diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn index d527ef6b24..805a6ab6c3 100644 --- a/modules/audio_device/BUILD.gn +++ b/modules/audio_device/BUILD.gn @@ -178,6 +178,11 @@ if (!build_with_chromium) { "test_audio_device_impl.cc", "test_audio_device_impl.h", ] + + # TODO: bugs.webrtc.org/413413572 - Mark this build target as testonly or + # or update downstream not to use deprecated function. Explicit poison + # can be removed then. + allow_poison = [ "environment_construction" ] deps = [ ":audio_device_buffer", ":audio_device_default", @@ -187,6 +192,8 @@ if (!build_with_chromium) { "../../api:make_ref_counted", "../../api:scoped_refptr", "../../api/audio:audio_device", + "../../api/environment", + "../../api/environment:environment_factory", "../../api/task_queue", "../../api/units:time_delta", "../../common_audio", diff --git a/modules/audio_device/include/test_audio_device.cc b/modules/audio_device/include/test_audio_device.cc index a0e0ef87d6..f6d5386271 100644 --- a/modules/audio_device/include/test_audio_device.cc +++ b/modules/audio_device/include/test_audio_device.cc @@ -21,6 +21,8 @@ #include "absl/strings/string_view.h" #include "api/array_view.h" #include "api/audio/audio_device.h" +#include "api/environment/environment.h" +#include "api/environment/environment_factory.h" #include "api/make_ref_counted.h" #include "api/scoped_refptr.h" #include "api/task_queue/task_queue_factory.h" @@ -428,11 +430,21 @@ scoped_refptr TestAudioDeviceModule::Create( std::unique_ptr capturer, std::unique_ptr renderer, float speed) { + return Create(CreateEnvironment(task_queue_factory), std::move(capturer), + std::move(renderer), speed); +} + +scoped_refptr TestAudioDeviceModule::Create( + const Environment& env, + std::unique_ptr capturer, + std::unique_ptr renderer, + float speed) { auto audio_device = make_ref_counted( AudioDeviceModule::AudioLayer::kDummyAudio, - std::make_unique(task_queue_factory, std::move(capturer), + std::make_unique(&env.task_queue_factory(), + std::move(capturer), std::move(renderer), speed), - task_queue_factory, + &env.task_queue_factory(), /*create_detached=*/true); // Ensure that the current platform is supported. diff --git a/modules/audio_device/include/test_audio_device.h b/modules/audio_device/include/test_audio_device.h index 6a9834235d..d7c556861d 100644 --- a/modules/audio_device/include/test_audio_device.h +++ b/modules/audio_device/include/test_audio_device.h @@ -14,12 +14,11 @@ #include #include -#include #include "absl/strings/string_view.h" #include "api/array_view.h" #include "api/audio/audio_device.h" -#include "api/audio/audio_device_defines.h" +#include "api/environment/environment.h" #include "api/scoped_refptr.h" #include "api/task_queue/task_queue_factory.h" #include "rtc_base/buffer.h" @@ -81,6 +80,13 @@ class TestAudioDeviceModule { // `renderer` is an object that receives audio data that would have been // played out. Can be nullptr if this device is never used for playing. // Use one of the Create... functions to get these instances. + static scoped_refptr Create( + const Environment& env, + std::unique_ptr capturer, + std::unique_ptr renderer, + float speed = 1); + + [[deprecated("bugs.webrtc.org/413413572")]] static scoped_refptr Create( TaskQueueFactory* task_queue_factory, std::unique_ptr capturer, diff --git a/modules/audio_device/include/test_audio_device_unittest.cc b/modules/audio_device/include/test_audio_device_unittest.cc index 18abf00f84..5e4888b372 100644 --- a/modules/audio_device/include/test_audio_device_unittest.cc +++ b/modules/audio_device/include/test_audio_device_unittest.cc @@ -10,22 +10,32 @@ #include "modules/audio_device/include/test_audio_device.h" +#include + #include -#include +#include +#include +#include +#include #include #include +#include #include +#include #include "api/array_view.h" +#include "api/audio/audio_device.h" #include "api/audio/audio_device_defines.h" -#include "api/task_queue/task_queue_factory.h" +#include "api/environment/environment.h" +#include "api/environment/environment_factory.h" +#include "api/scoped_refptr.h" #include "api/units/time_delta.h" #include "api/units/timestamp.h" #include "common_audio/wav_file.h" -#include "common_audio/wav_header.h" +#include "rtc_base/buffer.h" #include "rtc_base/checks.h" -#include "rtc_base/logging.h" #include "rtc_base/synchronization/mutex.h" +#include "rtc_base/thread_annotations.h" #include "test/gmock.h" #include "test/gtest.h" #include "test/testsupport/file_utils.h" @@ -463,15 +473,17 @@ class TestAudioTransport : public AudioTransport { TEST(TestAudioDeviceModuleTest, CreatedADMCanRecord) { GlobalSimulatedTimeController time_controller(kStartTime); + const Environment env = CreateEnvironment( + time_controller.GetClock(), time_controller.GetTaskQueueFactory()); TestAudioTransport audio_transport(TestAudioTransport::Mode::kRecording); std::unique_ptr capturer = TestAudioDeviceModule::CreatePulsedNoiseCapturer( /*max_amplitude=*/1000, /*sampling_frequency_in_hz=*/48000, /*num_channels=*/2); - scoped_refptr adm = TestAudioDeviceModule::Create( - time_controller.GetTaskQueueFactory(), std::move(capturer), - /*renderer=*/nullptr); + scoped_refptr adm = + TestAudioDeviceModule::Create(env, std::move(capturer), + /*renderer=*/nullptr); ASSERT_EQ(adm->RegisterAudioCallback(&audio_transport), 0); ASSERT_EQ(adm->Init(), 0); @@ -495,13 +507,15 @@ TEST(TestAudioDeviceModuleTest, CreatedADMCanRecord) { TEST(TestAudioDeviceModuleTest, CreatedADMCanPlay) { GlobalSimulatedTimeController time_controller(kStartTime); + const Environment env = CreateEnvironment( + time_controller.GetClock(), time_controller.GetTaskQueueFactory()); TestAudioTransport audio_transport(TestAudioTransport::Mode::kPlaying); std::unique_ptr renderer = TestAudioDeviceModule::CreateDiscardRenderer( /*sampling_frequency_in_hz=*/48000, /*num_channels=*/2); scoped_refptr adm = - TestAudioDeviceModule::Create(time_controller.GetTaskQueueFactory(), + TestAudioDeviceModule::Create(env, /*capturer=*/nullptr, std::move(renderer)); ASSERT_EQ(adm->RegisterAudioCallback(&audio_transport), 0); diff --git a/test/BUILD.gn b/test/BUILD.gn index 7d106854a3..ad3012d1dc 100644 --- a/test/BUILD.gn +++ b/test/BUILD.gn @@ -1260,10 +1260,12 @@ if (!build_with_chromium) { "../api:frame_generator_api", "../api:rtp_headers", "../api:rtp_parameters", + "../api:scoped_refptr", "../api:simulated_network_api", "../api:transport_api", "../api/audio:audio_device", "../api/audio:builtin_audio_processing_builder", + "../api/audio_codecs:audio_codecs_api", "../api/audio_codecs:builtin_audio_decoder_factory", "../api/audio_codecs:builtin_audio_encoder_factory", "../api/environment", @@ -1271,16 +1273,19 @@ if (!build_with_chromium) { "../api/rtc_event_log", "../api/task_queue", "../api/test/video:function_video_factory", + "../api/transport:bitrate_settings", "../api/transport:field_trial_based_config", "../api/units:time_delta", "../api/video:builtin_video_bitrate_allocator_factory", "../api/video:video_bitrate_allocator_factory", "../api/video:video_frame", + "../api/video:video_rtp_headers", "../api/video_codecs:video_codecs_api", "../call", "../call:call_interfaces", "../call:fake_network", "../call:simulated_packet_receiver", + "../call:video_receive_stream_api", "../call:video_send_stream_api", "../modules/audio_device:test_audio_device_module", "../modules/audio_mixer:audio_mixer_impl", diff --git a/test/call_test.cc b/test/call_test.cc index 37e2ef9f09..4d55cde9ef 100644 --- a/test/call_test.cc +++ b/test/call_test.cc @@ -10,26 +10,59 @@ #include "test/call_test.h" -#include +#include +#include +#include +#include #include +#include +#include +#include +#include #include "api/audio/audio_device.h" #include "api/audio/builtin_audio_processing_builder.h" +#include "api/audio_codecs/audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" +#include "api/call/transport.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" -#include "api/task_queue/task_queue_base.h" +#include "api/rtc_event_log/rtc_event_log.h" +#include "api/rtp_headers.h" +#include "api/rtp_parameters.h" +#include "api/scoped_refptr.h" +#include "api/task_queue/task_queue_factory.h" #include "api/test/create_frame_generator.h" +#include "api/test/simulated_network.h" +#include "api/transport/bitrate_settings.h" +#include "api/units/time_delta.h" #include "api/video/builtin_video_bitrate_allocator_factory.h" +#include "api/video/video_codec_type.h" +#include "api/video/video_rotation.h" +#include "api/video_codecs/sdp_video_format.h" +#include "api/video_codecs/video_decoder_factory.h" +#include "call/audio_receive_stream.h" +#include "call/audio_send_stream.h" +#include "call/audio_state.h" +#include "call/call.h" +#include "call/call_config.h" #include "call/fake_network_pipe.h" +#include "call/flexfec_receive_stream.h" #include "call/packet_receiver.h" +#include "call/video_receive_stream.h" +#include "call/video_send_stream.h" #include "modules/audio_device/include/test_audio_device.h" #include "modules/audio_mixer/audio_mixer_impl.h" +#include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "rtc_base/checks.h" -#include "rtc_base/event.h" #include "rtc_base/task_queue_for_test.h" +#include "test/encoder_settings.h" +#include "test/fake_decoder.h" #include "test/fake_encoder.h" +#include "test/fake_vp8_encoder.h" +#include "test/frame_generator_capturer.h" +#include "test/gtest.h" #include "test/network/simulated_network.h" #include "test/rtp_rtcp_observer.h" #include "test/testsupport/file_utils.h" @@ -531,10 +564,10 @@ void CallTest::CreateFrameGeneratorCapturer(int framerate, void CallTest::CreateFakeAudioDevices( std::unique_ptr capturer, std::unique_ptr renderer) { - fake_send_audio_device_ = TestAudioDeviceModule::Create( - &env_.task_queue_factory(), std::move(capturer), nullptr, 1.f); - fake_recv_audio_device_ = TestAudioDeviceModule::Create( - &env_.task_queue_factory(), nullptr, std::move(renderer), 1.f); + fake_send_audio_device_ = + TestAudioDeviceModule::Create(env_, std::move(capturer), nullptr, 1.f); + fake_recv_audio_device_ = + TestAudioDeviceModule::Create(env_, nullptr, std::move(renderer), 1.f); } void CallTest::CreateVideoStreams() { diff --git a/test/network/BUILD.gn b/test/network/BUILD.gn index 6f2b030081..81ce6e3295 100644 --- a/test/network/BUILD.gn +++ b/test/network/BUILD.gn @@ -112,6 +112,8 @@ if (rtc_include_tests && !build_with_chromium) { "../../api:rtc_error_matchers", "../../api:scoped_refptr", "../../api:simulated_network_api", + "../../api/environment", + "../../api/environment:environment_factory", "../../api/rtc_event_log:rtc_event_log_factory", "../../api/task_queue:default_task_queue_factory", "../../api/test/network_emulation", @@ -122,6 +124,7 @@ if (rtc_include_tests && !build_with_chromium) { "../../pc:peerconnection_wrapper", "../../rtc_base:network", "../../rtc_base:task_queue_for_test", + "../../rtc_base:threading", ] } } diff --git a/test/network/network_emulation_pc_unittest.cc b/test/network/network_emulation_pc_unittest.cc index 2eca595ebe..5a248450ad 100644 --- a/test/network/network_emulation_pc_unittest.cc +++ b/test/network/network_emulation_pc_unittest.cc @@ -14,23 +14,23 @@ #include "api/audio_options.h" #include "api/enable_media_with_defaults.h" +#include "api/environment/environment.h" +#include "api/environment/environment_factory.h" #include "api/jsep.h" #include "api/media_stream_interface.h" #include "api/peer_connection_interface.h" #include "api/rtc_event_log/rtc_event_log_factory.h" #include "api/scoped_refptr.h" -#include "api/task_queue/default_task_queue_factory.h" #include "api/test/network_emulation/network_emulation_interfaces.h" #include "api/test/network_emulation_manager.h" #include "api/test/rtc_error_matchers.h" #include "api/test/simulated_network.h" -#include "api/transport/field_trial_based_config.h" #include "modules/audio_device/include/test_audio_device.h" #include "p2p/base/port_allocator.h" #include "pc/peer_connection_wrapper.h" #include "pc/test/mock_peer_connection_observers.h" -#include "rtc_base/network.h" #include "rtc_base/task_queue_for_test.h" +#include "rtc_base/thread.h" #include "test/gmock.h" #include "test/gtest.h" #include "test/network/network_emulation.h" @@ -63,17 +63,17 @@ bool AddIceCandidates(PeerConnectionWrapper* peer, scoped_refptr CreatePeerConnectionFactory( Thread* signaling_thread, EmulatedNetworkManagerInterface* network) { + const Environment env = CreateEnvironment(); PeerConnectionFactoryDependencies pcf_deps; - pcf_deps.task_queue_factory = CreateDefaultTaskQueueFactory(); + pcf_deps.env = env; pcf_deps.event_log_factory = std::make_unique(); pcf_deps.network_thread = network->network_thread(); pcf_deps.signaling_thread = signaling_thread; - pcf_deps.trials = std::make_unique(); pcf_deps.socket_factory = network->socket_factory(); pcf_deps.network_manager = network->ReleaseNetworkManager(); pcf_deps.adm = TestAudioDeviceModule::Create( - pcf_deps.task_queue_factory.get(), + env, TestAudioDeviceModule::CreatePulsedNoiseCapturer(kMaxAptitude, kSamplingFrequency), TestAudioDeviceModule::CreateDiscardRenderer(kSamplingFrequency), diff --git a/test/pc/e2e/test_peer_factory.cc b/test/pc/e2e/test_peer_factory.cc index a251cafdc8..4ff994ada1 100644 --- a/test/pc/e2e/test_peer_factory.cc +++ b/test/pc/e2e/test_peer_factory.cc @@ -24,7 +24,6 @@ #include "api/peer_connection_interface.h" #include "api/rtc_event_log/rtc_event_log_factory.h" #include "api/scoped_refptr.h" -#include "api/task_queue/task_queue_factory.h" #include "api/test/pclf/media_configuration.h" #include "api/test/pclf/media_quality_test_params.h" #include "api/test/pclf/peer_configurer.h" @@ -124,10 +123,10 @@ std::unique_ptr CreateAudioCapturer( } scoped_refptr CreateAudioDeviceModule( + const Environment& env, std::optional audio_config, std::optional remote_audio_config, - std::optional echo_emulation_config, - TaskQueueFactory* task_queue_factory) { + std::optional echo_emulation_config) { std::unique_ptr renderer = CreateAudioRenderer(remote_audio_config); std::unique_ptr capturer = @@ -150,7 +149,7 @@ scoped_refptr CreateAudioDeviceModule( std::move(capturer), audio_config->input_dump_file_name.value()); } - return TestAudioDeviceModule::Create(task_queue_factory, std::move(capturer), + return TestAudioDeviceModule::Create(env, std::move(capturer), std::move(renderer), /*speed=*/1.f); } @@ -296,8 +295,8 @@ std::unique_ptr TestPeerFactory::CreateTestPeer( // Create peer connection factory. scoped_refptr audio_device_module = - CreateAudioDeviceModule(params->audio_config, remote_audio_config, - echo_emulation_config, &env.task_queue_factory()); + CreateAudioDeviceModule(env, params->audio_config, remote_audio_config, + echo_emulation_config); WrapVideoEncoderFactory( params->name.value(), params->video_encoder_bitrate_multiplier, CalculateRequiredSpatialIndexPerStream( diff --git a/test/peer_scenario/peer_scenario_client.cc b/test/peer_scenario/peer_scenario_client.cc index 52dedbd900..2c688e0129 100644 --- a/test/peer_scenario/peer_scenario_client.cc +++ b/test/peer_scenario/peer_scenario_client.cc @@ -287,7 +287,7 @@ PeerScenarioClient::PeerScenarioClient( pcf_deps.env = env_; pcf_deps.adm = TestAudioDeviceModule::Create( - &env_.task_queue_factory(), + env_, TestAudioDeviceModule::CreatePulsedNoiseCapturer( config.audio.pulsed_noise->amplitude * std::numeric_limits::max(), diff --git a/test/scenario/BUILD.gn b/test/scenario/BUILD.gn index 2cae7625cd..b7fd6a3506 100644 --- a/test/scenario/BUILD.gn +++ b/test/scenario/BUILD.gn @@ -103,6 +103,7 @@ if (rtc_include_tests && !build_with_chromium) { "../../api/task_queue", "../../api/test/network_emulation", "../../api/test/video:function_video_factory", + "../../api/transport:bitrate_settings", "../../api/transport:network_control", "../../api/units:data_rate", "../../api/units:data_size", @@ -140,6 +141,7 @@ if (rtc_include_tests && !build_with_chromium) { "../../modules/video_coding/svc:scalability_mode_util", "../../rtc_base:checks", "../../rtc_base:copy_on_write_buffer", + "../../rtc_base:logging", "../../rtc_base:macromagic", "../../rtc_base:net_helper", "../../rtc_base:net_helpers", diff --git a/test/scenario/call_client.cc b/test/scenario/call_client.cc index b598c2878c..1f6719d88d 100644 --- a/test/scenario/call_client.cc +++ b/test/scenario/call_client.cc @@ -9,24 +9,48 @@ */ #include "test/scenario/call_client.h" -#include +#include +#include +#include #include +#include #include +#include "api/array_view.h" #include "api/audio/builtin_audio_processing_builder.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" #include "api/media_types.h" #include "api/rtc_event_log/rtc_event_log.h" #include "api/rtc_event_log/rtc_event_log_factory.h" +#include "api/rtc_event_log_output.h" +#include "api/rtp_parameters.h" +#include "api/scoped_refptr.h" +#include "api/task_queue/task_queue_factory.h" +#include "api/test/network_emulation/network_emulation_interfaces.h" +#include "api/test/time_controller.h" +#include "api/transport/bitrate_settings.h" +#include "api/transport/network_control.h" #include "api/transport/network_types.h" +#include "api/units/data_rate.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" +#include "call/audio_state.h" #include "call/call.h" -#include "call/rtp_transport_controller_send_factory.h" +#include "call/call_config.h" #include "modules/audio_device/include/test_audio_device.h" #include "modules/audio_mixer/audio_mixer_impl.h" #include "modules/rtp_rtcp/include/rtp_header_extension_map.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "modules/rtp_rtcp/source/rtp_util.h" +#include "rtc_base/checks.h" +#include "rtc_base/event.h" +#include "rtc_base/logging.h" +#include "rtc_base/strings/string_builder.h" +#include "test/logging/log_writer.h" +#include "test/scenario/column_printer.h" +#include "test/scenario/network_node.h" +#include "test/scenario/scenario_config.h" namespace webrtc { namespace test { @@ -48,7 +72,7 @@ CallClientFakeAudio InitAudio(const Environment& env) { auto capturer = TestAudioDeviceModule::CreatePulsedNoiseCapturer(256, 48000); auto renderer = TestAudioDeviceModule::CreateDiscardRenderer(48000); setup.fake_audio_device = TestAudioDeviceModule::Create( - &env.task_queue_factory(), std::move(capturer), std::move(renderer), 1.f); + env, std::move(capturer), std::move(renderer), 1.f); setup.apm = BuiltinAudioProcessingBuilder().Build(env); setup.fake_audio_device->Init(); AudioState::Config audio_state_config; diff --git a/video/video_quality_test.cc b/video/video_quality_test.cc index f7a1239c25..66a8a25f39 100644 --- a/video/video_quality_test.cc +++ b/video/video_quality_test.cc @@ -1407,8 +1407,7 @@ void VideoQualityTest::InitializeAudioDevice(CallConfig* send_call_config, } else { // By default, create a test ADM which fakes audio. audio_device = TestAudioDeviceModule::Create( - &env_.task_queue_factory(), - TestAudioDeviceModule::CreatePulsedNoiseCapturer(32000, 48000), + env_, TestAudioDeviceModule::CreatePulsedNoiseCapturer(32000, 48000), TestAudioDeviceModule::CreateDiscardRenderer(48000), 1.f); } RTC_CHECK(audio_device); From 82aab16a29782ee2b0b674aaef25fbfdddad98e2 Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Wed, 23 Apr 2025 14:56:40 -0700 Subject: [PATCH 034/292] sdp munging: detect "raw packetization" which is a nonstandard attribute used by some services to control packetization. BUG=chromium:40567530,webrtc:412904801 Change-Id: Ibd6399a8584ff8c9ff67dfa37f27a1ef2f534f8e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/388200 Reviewed-by: Harald Alvestrand Commit-Queue: Philipp Hancke Reviewed-by: Johannes Kron Cr-Commit-Position: refs/heads/main@{#44506} --- api/uma_metrics.h | 2 ++ pc/sdp_munging_detector.cc | 22 +++++++++++++++++ pc/sdp_offer_answer_unittest.cc | 44 +++++++++++++++++++++++++++++++++ 3 files changed, 68 insertions(+) diff --git a/api/uma_metrics.h b/api/uma_metrics.h index 980cb883fd..fdca7dcceb 100644 --- a/api/uma_metrics.h +++ b/api/uma_metrics.h @@ -222,6 +222,8 @@ enum SdpMungingType { kVideoCodecsFmtpH264SpsPpsIdrInKeyframe = 84, kVideoCodecsFmtp = 85, kVideoCodecsRtcpFb = 86, + kVideoCodecsAddedWithRawPacketization = 87, + kVideoCodecsModifiedWithRawPacketization = 88, kMaxValue, }; diff --git a/pc/sdp_munging_detector.cc b/pc/sdp_munging_detector.cc index e54c228d5d..af1ebd3be0 100644 --- a/pc/sdp_munging_detector.cc +++ b/pc/sdp_munging_detector.cc @@ -279,6 +279,22 @@ SdpMungingType DetermineVideoSdpMungingType( } if (last_created_media_description->codecs().size() < media_description_to_set->codecs().size()) { + // Nonstandard a=packetization:raw + bool created_raw_packetization = + absl::c_find_if(last_created_media_description->codecs(), + [](const Codec codec) { + return codec.packetization.has_value(); + }) != last_created_media_description->codecs().end(); + bool set_raw_packetization = + absl::c_find_if(media_description_to_set->codecs(), + [](const Codec codec) { + return codec.packetization.has_value(); + }) != media_description_to_set->codecs().end(); + if (!created_raw_packetization && set_raw_packetization) { + RTC_LOG(LS_WARNING) + << "SDP munging: video codecs with raw packetization added."; + return SdpMungingType::kVideoCodecsAddedWithRawPacketization; + } RTC_LOG(LS_WARNING) << "SDP munging: video codecs added."; return SdpMungingType::kVideoCodecsAdded; } @@ -439,6 +455,12 @@ SdpMungingType DetermineSdpMungingType( ? SdpMungingType::kAudioCodecsRtcpFb : SdpMungingType::kVideoCodecsRtcpFb; } + // Nonstandard a=packetization:raw added by munging + if (media_type == MediaType::VIDEO && + last_created_codecs[i].packetization != + codecs_to_set[i].packetization) { + return SdpMungingType::kVideoCodecsModifiedWithRawPacketization; + } // At this point clockrate or channels changed. This should already be // rejected later in the process so ignore for munging. } diff --git a/pc/sdp_offer_answer_unittest.cc b/pc/sdp_offer_answer_unittest.cc index 20567b5f81..c47f086599 100644 --- a/pc/sdp_offer_answer_unittest.cc +++ b/pc/sdp_offer_answer_unittest.cc @@ -2391,6 +2391,50 @@ TEST_F(SdpOfferAnswerMungingTest, VideoCodecsAdded) { ElementsAre(Pair(SdpMungingType::kVideoCodecsAdded, 1))); } +TEST_F(SdpOfferAnswerMungingTest, VideoCodecsAddedWithRawPacketization) { + auto pc = CreatePeerConnection(); + pc->AddVideoTrack("video_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + std::vector codecs = media_description->codecs(); + auto codec = CreateVideoCodec(SdpVideoFormat("VP8", {})); + codec.id = 19; // IANA reserved payload type, should not conflict. + codec.packetization = "raw"; + codecs.push_back(codec); + media_description->set_codecs(codecs); + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre( + Pair(SdpMungingType::kVideoCodecsAddedWithRawPacketization, 1))); +} + +TEST_F(SdpOfferAnswerMungingTest, VideoCodecsModifiedWithRawPacketization) { + auto pc = CreatePeerConnection(); + pc->AddVideoTrack("video_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + std::vector codecs = media_description->codecs(); + ASSERT_TRUE(!codecs.empty()); + codecs[0].packetization = "raw"; + media_description->set_codecs(codecs); + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre( + Pair(SdpMungingType::kVideoCodecsModifiedWithRawPacketization, 1))); +} + TEST_F(SdpOfferAnswerMungingTest, MultiOpus) { auto pc = CreatePeerConnection(); pc->AddAudioTrack("audio_track", {}); From 25b8384bb5698620f5e0b296ac035dffffc231b2 Mon Sep 17 00:00:00 2001 From: webrtc-version-updater Date: Fri, 2 May 2025 21:03:34 -0700 Subject: [PATCH 035/292] Update WebRTC code version (2025-05-03T04:03:10). Bug: None Change-Id: If0c8b94fe13e72410c215e341f6f0a51537641ba Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389448 Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Cr-Commit-Position: refs/heads/main@{#44507} --- call/version.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/call/version.cc b/call/version.cc index 21c1b34ff7..113c7887a1 100644 --- a/call/version.cc +++ b/call/version.cc @@ -13,7 +13,7 @@ namespace webrtc { // The timestamp is always in UTC. -const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-02T04:07:31"; +const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-03T04:03:10"; void LoadWebRTCVersionInRegister() { // Using volatile to instruct the compiler to not optimize `p` away even From 2abc8422c7a7c3dfb2ffca85a2b34f787b349c9a Mon Sep 17 00:00:00 2001 From: webrtc-version-updater Date: Sun, 4 May 2025 21:09:30 -0700 Subject: [PATCH 036/292] Update WebRTC code version (2025-05-05T04:09:29). Bug: None Change-Id: I7da320862ac3a29ad333ba4dea0569aa647547ba Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389742 Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Cr-Commit-Position: refs/heads/main@{#44508} --- call/version.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/call/version.cc b/call/version.cc index 113c7887a1..98a8ccea46 100644 --- a/call/version.cc +++ b/call/version.cc @@ -13,7 +13,7 @@ namespace webrtc { // The timestamp is always in UTC. -const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-03T04:03:10"; +const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-05T04:09:29"; void LoadWebRTCVersionInRegister() { // Using volatile to instruct the compiler to not optimize `p` away even From afc72d32d688b8f2364d16b475f43ffc59c5ddeb Mon Sep 17 00:00:00 2001 From: Sameer Vijaykar Date: Tue, 29 Apr 2025 15:42:32 +0200 Subject: [PATCH 037/292] Test order of ICE candidate and gathering state change events. The new test is based on the DataChannelOnly test, but focuses on the order of surfacing ICE candidates and changes in the ICE gathering state. Bug: webrtc:411020146 Change-Id: I6eac7c831e86c53280742180adba1abe54ae0f74 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389000 Reviewed-by: Harald Alvestrand Commit-Queue: Sameer Vijaykar Cr-Commit-Position: refs/heads/main@{#44509} --- .../webrtc/PeerConnectionEndToEndTest.java | 99 +++++++++++++++++++ 1 file changed, 99 insertions(+) diff --git a/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionEndToEndTest.java b/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionEndToEndTest.java index 5ec5aae2e3..33807af2f4 100644 --- a/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionEndToEndTest.java +++ b/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionEndToEndTest.java @@ -169,6 +169,11 @@ public synchronized void expectSignalingChange(SignalingState newState) { // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression. @SuppressWarnings("NoSynchronizedMethodCheck") public synchronized void onSignalingChange(SignalingState newState) { + if (expectedSignalingChanges.isEmpty()) { + Logging.d(TAG, name + "Got an unexpected signaling state change " + newState); + return; + } + assertEquals(newState, expectedSignalingChanges.remove()); } @@ -261,6 +266,7 @@ public synchronized void onIceGatheringChange(IceGatheringState newState) { } if (expectedIceGatheringChanges.isEmpty()) { Logging.d(TAG, name + "Got an unexpected ICE gathering change " + newState); + return; } assertEquals(newState, expectedIceGatheringChanges.remove()); } @@ -1123,6 +1129,99 @@ public void testDataChannelOnlySession() throws Exception { System.gc(); } + @Test + @MediumTest + public void testSurfaceIceCandidatesBeforeIceGatheringStateComplete() throws Exception { + // Allow loopback interfaces too since our Android devices often don't + // have those. + PeerConnectionFactory.Options options = new PeerConnectionFactory.Options(); + options.networkIgnoreMask = 0; + PeerConnectionFactory factory = + PeerConnectionFactory.builder().setOptions(options).createPeerConnectionFactory(); + + List iceServers = new ArrayList<>(); + iceServers.add( + PeerConnection.IceServer.builder("stun:stun.l.google.com:19302").createIceServer()); + + PeerConnection.RTCConfiguration rtcConfig = new PeerConnection.RTCConfiguration(iceServers); + rtcConfig.sdpSemantics = PeerConnection.SdpSemantics.UNIFIED_PLAN; + + ObserverExpectations offeringExpectations = new ObserverExpectations("PCTest:offerer"); + PeerConnection offeringPC = factory.createPeerConnection(rtcConfig, offeringExpectations); + assertNotNull(offeringPC); + + offeringExpectations.expectRenegotiationNeeded(); + DataChannel offeringDC = offeringPC.createDataChannel("offeringDC", new DataChannel.Init()); + + offeringExpectations.expectIceCandidates(2); + + SdpObserverLatch sdpLatch = new SdpObserverLatch(); + offeringPC.createOffer(sdpLatch, new MediaConstraints()); + assertTrue(sdpLatch.await()); + SessionDescription offerSdp = sdpLatch.getSdp(); + + sdpLatch = new SdpObserverLatch(); + offeringPC.setLocalDescription(sdpLatch, offerSdp); + + assertTrue(offeringExpectations.waitForAllExpectationsToBeSatisfied(DEFAULT_TIMEOUT_SECONDS)); + assertEquals(IceGatheringState.GATHERING, offeringPC.iceGatheringState()); + + ObserverExpectations answeringExpectations = new ObserverExpectations("PCTest:answerer"); + PeerConnection answeringPC = factory.createPeerConnection(rtcConfig, answeringExpectations); + assertNotNull(answeringPC); + + sdpLatch = new SdpObserverLatch(); + answeringPC.setRemoteDescription(sdpLatch, offerSdp); + assertTrue(sdpLatch.await()); + + sdpLatch = new SdpObserverLatch(); + answeringPC.createAnswer(sdpLatch, new MediaConstraints()); + assertTrue(sdpLatch.await()); + SessionDescription answerSdp = sdpLatch.getSdp(); + + answeringExpectations.expectIceCandidates(2); + + sdpLatch = new SdpObserverLatch(); + answeringPC.setLocalDescription(sdpLatch, answerSdp); + assertTrue(sdpLatch.await()); + + assertTrue(answeringExpectations.waitForAllExpectationsToBeSatisfied(DEFAULT_TIMEOUT_SECONDS)); + assertEquals(IceGatheringState.GATHERING, answeringPC.iceGatheringState()); + + sdpLatch = new SdpObserverLatch(); + offeringPC.setRemoteDescription(sdpLatch, answerSdp); + assertTrue(sdpLatch.await()); + + assertTrue(offeringExpectations.waitForAllExpectationsToBeSatisfied(DEFAULT_TIMEOUT_SECONDS)); + assertEquals(IceGatheringState.GATHERING, offeringPC.iceGatheringState()); + + // SCTP DataChannels are announced via OPEN messages over the established + // connection (not via SDP), so answeringExpectations can only register + // expecting the channel during ICE. + answeringExpectations.expectDataChannel("offeringDC"); + answeringExpectations.expectStateChange(DataChannel.State.OPEN); + + offeringExpectations.expectIceGatheringChange(IceGatheringState.COMPLETE); + answeringExpectations.expectIceGatheringChange(IceGatheringState.COMPLETE); + + // Wait for at least one ice candidate from the offering PC and forward them to + // the answering PC. + for (IceCandidate candidate : offeringExpectations.getAtLeastOneIceCandidate()) { + answeringPC.addIceCandidate(candidate); + } + // Wait for at least one ice candidate from the answering PC and forward them to + // the offering PC. + for (IceCandidate candidate : answeringExpectations.getAtLeastOneIceCandidate()) { + offeringPC.addIceCandidate(candidate); + } + + assertTrue(offeringExpectations.waitForAllExpectationsToBeSatisfied(DEFAULT_TIMEOUT_SECONDS)); + assertTrue(answeringExpectations.waitForAllExpectationsToBeSatisfied(DEFAULT_TIMEOUT_SECONDS)); + + assertEquals(IceGatheringState.COMPLETE, offeringPC.iceGatheringState()); + assertEquals(IceGatheringState.COMPLETE, answeringPC.iceGatheringState()); + } + // Tests that ICE candidates that are not allowed by an ICE transport type, thus not being // signaled to the gathering PeerConnection, can be surfaced via configuration if allowed by the // new ICE transport type, when RTCConfiguration.surfaceIceCandidatesOnIceTransportTypeChanged is From e83cf6587af12f7d36f09a40424340376b6a0e6b Mon Sep 17 00:00:00 2001 From: Fredrik Solenberg Date: Mon, 5 May 2025 10:17:16 +0200 Subject: [PATCH 038/292] - Use std::identity (pending C++20) - Remove webrtc::identity Bug: none Change-Id: Id7543bcdc23cdf61b95d25b8677bd66ffb9ad2da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/385922 Reviewed-by: Harald Alvestrand Commit-Queue: Fredrik Solenberg Cr-Commit-Position: refs/heads/main@{#44510} --- rtc_base/containers/BUILD.gn | 2 +- rtc_base/containers/flat_set.h | 3 +- rtc_base/containers/flat_set_unittest.cc | 1 - rtc_base/containers/flat_tree_unittest.cc | 51 +++++++++++++++-------- rtc_base/containers/identity.h | 36 ---------------- 5 files changed, 35 insertions(+), 58 deletions(-) delete mode 100644 rtc_base/containers/identity.h diff --git a/rtc_base/containers/BUILD.gn b/rtc_base/containers/BUILD.gn index a2a21157b7..c4d9f28f9f 100644 --- a/rtc_base/containers/BUILD.gn +++ b/rtc_base/containers/BUILD.gn @@ -12,7 +12,6 @@ rtc_library("flat_containers_internal") { sources = [ "flat_tree.cc", "flat_tree.h", - "identity.h", "invoke.h", "move_only_int.h", ] @@ -51,5 +50,6 @@ rtc_library("unittests") { "../../test:test_support", "//testing/gmock:gmock", "//testing/gtest:gtest", + "//third_party/abseil-cpp/absl/algorithm:container", ] } diff --git a/rtc_base/containers/flat_set.h b/rtc_base/containers/flat_set.h index 355690b09d..22001d3d5c 100644 --- a/rtc_base/containers/flat_set.h +++ b/rtc_base/containers/flat_set.h @@ -17,7 +17,6 @@ #include #include "rtc_base/containers/flat_tree.h" // IWYU pragma: export -#include "rtc_base/containers/identity.h" namespace webrtc { @@ -159,7 +158,7 @@ template , class Container = std::vector> using flat_set = typename ::webrtc::flat_containers_internal:: - flat_tree; + flat_tree; // ---------------------------------------------------------------------------- // General operations. diff --git a/rtc_base/containers/flat_set_unittest.cc b/rtc_base/containers/flat_set_unittest.cc index 617db92440..22c051a3c8 100644 --- a/rtc_base/containers/flat_set_unittest.cc +++ b/rtc_base/containers/flat_set_unittest.cc @@ -12,7 +12,6 @@ #include "rtc_base/containers/flat_set.h" -#include #include #include #include diff --git a/rtc_base/containers/flat_tree_unittest.cc b/rtc_base/containers/flat_tree_unittest.cc index 9bb803d16d..eb45cc7c2a 100644 --- a/rtc_base/containers/flat_tree_unittest.cc +++ b/rtc_base/containers/flat_tree_unittest.cc @@ -36,6 +36,7 @@ // * No tests with min_allocator and no tests counting allocations. // Flat sets currently don't support allocators. +#include #include #include #include @@ -43,9 +44,12 @@ #include #include #include +#include +#include +#include #include -#include "rtc_base/containers/identity.h" +#include "absl/algorithm/container.h" #include "rtc_base/containers/move_only_int.h" #include "test/gmock.h" #include "test/gtest.h" @@ -157,22 +161,30 @@ struct LessByFirst { // Common test trees. template using TypedTree = flat_tree, ContainerT>; using IntTree = TypedTree>; using IntPair = std::pair; -using IntPairTree = - flat_tree, std::vector>; -using MoveOnlyTree = - flat_tree, std::vector>; -using EmplaceableTree = - flat_tree, std::vector>; +using IntPairTree = flat_tree, + std::vector>; +using MoveOnlyTree = flat_tree, + std::vector>; +using EmplaceableTree = flat_tree, + std::vector>; using ReversedTree = - flat_tree, std::vector>; + flat_tree, std::vector>; -using TreeWithStrangeCompare = - flat_tree>; +using TreeWithStrangeCompare = flat_tree>; using ::testing::ElementsAre; using ::testing::IsEmpty; @@ -203,8 +215,8 @@ TEST(FlatTree, NoExcept) { MoveThrows& operator=(MoveThrows&&) noexcept(false) { return *this; } }; - using MoveThrowsTree = - flat_tree, std::array>; + using MoveThrowsTree = flat_tree, + std::array>; static_assert(std::is_nothrow_move_constructible::value, "Error: IntTree is not nothrow move constructible"); @@ -225,7 +237,7 @@ TEST(FlatTree, NoExcept) { TEST(FlatTree, IncompleteType) { struct A { - using Tree = flat_tree, std::vector>; + using Tree = flat_tree, std::vector>; int data; Tree set_with_incomplete_type; Tree::iterator it; @@ -240,7 +252,8 @@ TEST(FlatTree, IncompleteType) { TEST(FlatTree, Stability) { using Pair = std::pair; - using Tree = flat_tree, std::vector>; + using Tree = + flat_tree, std::vector>; // Constructors are stable. Tree cont({{0, 0}, {1, 0}, {0, 1}, {2, 0}, {0, 2}, {1, 1}}); @@ -388,7 +401,8 @@ TEST(FlatTree, ContainerMoveConstructor) { storage.push_back(Pair(1, MoveOnlyInt(0))); storage.push_back(Pair(2, MoveOnlyInt(1))); - using Tree = flat_tree, std::vector>; + using Tree = + flat_tree, std::vector>; Tree tree(std::move(storage)); // The list should be two items long, with only the first "2" saved. @@ -470,7 +484,8 @@ TEST(FlatTree, SortedUniqueVectorMoveConstructor) { storage.push_back(Pair(1, MoveOnlyInt(0))); storage.push_back(Pair(2, MoveOnlyInt(0))); - using Tree = flat_tree, std::vector>; + using Tree = + flat_tree, std::vector>; Tree tree(sorted_unique, std::move(storage)); ASSERT_EQ(2u, tree.size()); @@ -997,7 +1012,7 @@ TYPED_TEST_P(FlatTreeTest, ErasePosition) { { using T = TemplateConstructor; - flat_tree, std::vector> cont; + flat_tree, std::vector> cont; T v(0); auto it = cont.find(v); diff --git a/rtc_base/containers/identity.h b/rtc_base/containers/identity.h deleted file mode 100644 index 29592931bd..0000000000 --- a/rtc_base/containers/identity.h +++ /dev/null @@ -1,36 +0,0 @@ -/* - * Copyright (c) 2021 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -// This implementation is borrowed from Chromium. - -#ifndef RTC_BASE_CONTAINERS_IDENTITY_H_ -#define RTC_BASE_CONTAINERS_IDENTITY_H_ - -#include - -namespace webrtc { - -// Implementation of C++20's std::identity. -// -// Reference: -// - https://en.cppreference.com/w/cpp/utility/functional/identity -// - https://wg21.link/func.identity -struct identity { - template - constexpr T&& operator()(T&& t) const noexcept { - return std::forward(t); - } - - using is_transparent = void; -}; - -} // namespace webrtc - -#endif // RTC_BASE_CONTAINERS_IDENTITY_H_ From f222f40e3d7a3c662509c8dcc90306c063da14df Mon Sep 17 00:00:00 2001 From: Johannes Kron Date: Mon, 5 May 2025 07:36:41 +0000 Subject: [PATCH 039/292] Don't log ERROR for initial max bitrate == -1 An initial max bitrate of -1 is used to signal that the value is not set. Remove log spam by not logging an error for -1. Bug: None Change-Id: Ia57d7f8ef6c3798fa9d2f2f42a2f8537a6d38d9a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389580 Reviewed-by: Ilya Nikolaevskiy Auto-Submit: Johannes Kron Commit-Queue: Johannes Kron Commit-Queue: Ilya Nikolaevskiy Cr-Commit-Position: refs/heads/main@{#44511} --- video/video_send_stream_impl.cc | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/video/video_send_stream_impl.cc b/video/video_send_stream_impl.cc index dfcb9a30e9..9f093f5f82 100644 --- a/video/video_send_stream_impl.cc +++ b/video/video_send_stream_impl.cc @@ -53,7 +53,6 @@ #include "call/rtp_config.h" #include "call/rtp_transport_controller_send_interface.h" #include "call/video_send_stream.h" -#include "media/base/media_constants.h" #include "media/base/sdp_video_format_utils.h" #include "modules/pacing/pacing_controller.h" #include "modules/rtp_rtcp/include/rtp_header_extension_map.h" @@ -68,7 +67,6 @@ #include "rtc_base/experiments/rate_control_settings.h" #include "rtc_base/logging.h" #include "rtc_base/numerics/safe_conversions.h" -#include "rtc_base/strings/string_builder.h" #include "rtc_base/task_utils/repeating_task.h" #include "rtc_base/trace_event.h" #include "system_wrappers/include/clock.h" @@ -250,8 +248,12 @@ uint32_t GetInitialEncoderMaxBitrate(int initial_encoder_max_bitrate) { // reasonable use cases as it allows adding the max of multiple streams // without wrappping around. const int kFallbackMaxBitrateBps = 10000000; - RTC_DLOG(LS_ERROR) << "ERROR: Initial encoder max bitrate = " - << initial_encoder_max_bitrate << " which is <= 0!"; + // Don't log an error for -1 since this is the default value that is used to + // signal that the max bitrate is unset. + if (initial_encoder_max_bitrate != -1) { + RTC_DLOG(LS_ERROR) << "ERROR: Initial encoder max bitrate = " + << initial_encoder_max_bitrate << " which is <= 0!"; + } RTC_DLOG(LS_INFO) << "Using default encoder max bitrate = 10 Mbps"; return kFallbackMaxBitrateBps; } From 4d9dc6841e8339dd4cb0413019fb31adcf414838 Mon Sep 17 00:00:00 2001 From: Johannes Kron Date: Mon, 5 May 2025 08:21:37 +0000 Subject: [PATCH 040/292] Update error messages in apply-include-cleaner Improve the workflow for users that don't have a workdir named out/Default by adding some hints to the error messages. Bug: webrtc:15874 Change-Id: If214fee762b56cf4c82a72c8efb76626d0fbd238 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389760 Reviewed-by: Tomas Gunnarsson Reviewed-by: Harald Alvestrand Commit-Queue: Johannes Kron Cr-Commit-Position: refs/heads/main@{#44512} --- tools_webrtc/iwyu/apply-include-cleaner | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/tools_webrtc/iwyu/apply-include-cleaner b/tools_webrtc/iwyu/apply-include-cleaner index 02de99a4ca..cec4dac31e 100755 --- a/tools_webrtc/iwyu/apply-include-cleaner +++ b/tools_webrtc/iwyu/apply-include-cleaner @@ -64,6 +64,7 @@ usage() { echo " -d : Set debug level to " echo " -w : Specify the workdir (out/Default if not specified)" echo " -h : Print this help message" + echo "" } COMMAND=" --edit" @@ -93,11 +94,16 @@ if [[ -z "$COMPILE_COMMANDS" ]]; then fi COMPILE_COMMANDS="$WORKDIR/compile_commands.json" else - error "Could not generate $WORKDIR/compile_commands.json." + usage + error "Could not generate $WORKDIR/compile_commands.json. Specify the workdir with -w if other than out/Default." fi fi -# To get a list of files in a commit: git diff-tree --no-commit-id --name-only -r HEAD +if [ $# -eq 0 ]; then + usage + error "No files provided as arguments. To get a list of files in a commit: git diff-tree --no-commit-id --name-only -r HEAD" +fi + for FILE in "$@" do if [ -z $FILE ] || [ ! -f $FILE ]; then From 9b272274b850f9bd40fd1b95907ddaf5483391cc Mon Sep 17 00:00:00 2001 From: Sergey Silkin Date: Mon, 5 May 2025 11:07:11 +0200 Subject: [PATCH 041/292] Prevent encoder switch on unaligned resolution MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Return NO_OUTPUT instead of ERROR from Encode() if input resolution doesn't satisfy required alignment. Bug: webrtc:415329365 Change-Id: If2d4794b1ffd68819744157a3b15c2613b90a047 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389561 Reviewed-by: Ilya Nikolaevskiy Commit-Queue: Sergey Silkin Reviewed-by: Åsa Persson Cr-Commit-Position: refs/heads/main@{#44513} --- experiments/field_trials.py | 3 ++ media/engine/simulcast_encoder_adapter.cc | 8 ++- media/engine/simulcast_encoder_adapter.h | 1 + .../simulcast_encoder_adapter_unittest.cc | 53 +++++++++++++++++++ 4 files changed, 63 insertions(+), 2 deletions(-) diff --git a/experiments/field_trials.py b/experiments/field_trials.py index b23b72457b..dff2fe0ff4 100755 --- a/experiments/field_trials.py +++ b/experiments/field_trials.py @@ -161,6 +161,9 @@ def bug_url(self) -> str: FieldTrial('WebRTC-SetReadyToSendFalseIfSendFail', 361124449, date(2024, 12, 1)), + FieldTrial('WebRTC-SimulcastEncoderAdapter-DropUnalignedResolution', + 415329365, + date(2025, 11, 2)), FieldTrial('WebRTC-SrtpRemoveReceiveStream', 42225949, date(2024, 10, 1)), diff --git a/media/engine/simulcast_encoder_adapter.cc b/media/engine/simulcast_encoder_adapter.cc index 73040a485f..0a1e108c8a 100644 --- a/media/engine/simulcast_encoder_adapter.cc +++ b/media/engine/simulcast_encoder_adapter.cc @@ -270,6 +270,8 @@ SimulcastEncoderAdapter::SimulcastEncoderAdapter( prefer_temporal_support_on_base_layer_(env_.field_trials().IsEnabled( "WebRTC-Video-PreferTemporalSupportOnBaseLayer")), per_layer_pli_(SupportsPerLayerPictureLossIndication(format.parameters)), + drop_unaligned_resolution_(!env_.field_trials().IsDisabled( + "WebRTC-SimulcastEncoderAdapter-DropUnalignedResolution")), encoder_info_override_(env.field_trials()) { RTC_DCHECK(primary_factory); @@ -476,7 +478,8 @@ int SimulcastEncoderAdapter::Encode( RTC_LOG(LS_WARNING) << "Frame " << input_image.width() << "x" << input_image.height() << " not divisible by " << alignment; - return WEBRTC_VIDEO_CODEC_ERROR; + return drop_unaligned_resolution_ ? WEBRTC_VIDEO_CODEC_NO_OUTPUT + : WEBRTC_VIDEO_CODEC_ERROR; } if (encoder_info_override_.apply_alignment_to_all_simulcast_layers()) { for (const auto& layer : stream_contexts_) { @@ -484,7 +487,8 @@ int SimulcastEncoderAdapter::Encode( RTC_LOG(LS_WARNING) << "Codec " << layer.width() << "x" << layer.height() << " not divisible by " << alignment; - return WEBRTC_VIDEO_CODEC_ERROR; + return drop_unaligned_resolution_ ? WEBRTC_VIDEO_CODEC_NO_OUTPUT + : WEBRTC_VIDEO_CODEC_ERROR; } } } diff --git a/media/engine/simulcast_encoder_adapter.h b/media/engine/simulcast_encoder_adapter.h index 1461416c4c..7673683127 100644 --- a/media/engine/simulcast_encoder_adapter.h +++ b/media/engine/simulcast_encoder_adapter.h @@ -196,6 +196,7 @@ class RTC_EXPORT SimulcastEncoderAdapter : public VideoEncoder { const bool boost_base_layer_quality_; const bool prefer_temporal_support_on_base_layer_; const bool per_layer_pli_; + const bool drop_unaligned_resolution_; const SimulcastEncoderAdapterEncoderInfoSettings encoder_info_override_; }; diff --git a/media/engine/simulcast_encoder_adapter_unittest.cc b/media/engine/simulcast_encoder_adapter_unittest.cc index a332003054..c8e18057e8 100644 --- a/media/engine/simulcast_encoder_adapter_unittest.cc +++ b/media/engine/simulcast_encoder_adapter_unittest.cc @@ -2126,5 +2126,58 @@ TEST_F(TestSimulcastEncoderAdapterFake, PopulatesScalabilityModeOfSubcodecs) { ScalabilityMode::kL1T3); } +TEST_F(TestSimulcastEncoderAdapterFake, + EncodeDropsFrameIfResolutionIsNotAlignedByDefault) { + test::ScopedKeyValueConfig field_trials( + field_trials_, + "WebRTC-SimulcastEncoderAdapter-GetEncoderInfoOverride/" + "requested_resolution_alignment:8," + "apply_alignment_to_all_simulcast_layers/"); + SetUp(); + SimulcastTestFixtureImpl::DefaultSettings( + &codec_, static_cast(kTestTemporalLayerProfile), + kVideoCodecVP8); + SetupCodec(); + EXPECT_EQ(0, adapter_->InitEncode(&codec_, kSettings)); + scoped_refptr buffer(I420Buffer::Create(1280, 720)); + VideoFrame input_frame = VideoFrame::Builder() + .set_video_frame_buffer(buffer) + .set_rtp_timestamp(0) + .set_timestamp_ms(0) + .build(); + std::vector frame_types; + frame_types.resize(codec_.numberOfSimulcastStreams, + VideoFrameType::kVideoFrameKey); + EXPECT_EQ(WEBRTC_VIDEO_CODEC_NO_OUTPUT, + adapter_->Encode(input_frame, &frame_types)); +} + +TEST_F(TestSimulcastEncoderAdapterFake, + EncodeReturnsErrorIfResolutionIsNotAlignedAndDroppingIsDisabled) { + test::ScopedKeyValueConfig field_trials( + field_trials_, + "WebRTC-SimulcastEncoderAdapter-DropUnalignedResolution/Disabled/" + "WebRTC-SimulcastEncoderAdapter-GetEncoderInfoOverride/" + "requested_resolution_alignment:8," + "apply_alignment_to_all_simulcast_layers/"); + SetUp(); + SimulcastTestFixtureImpl::DefaultSettings( + &codec_, static_cast(kTestTemporalLayerProfile), + kVideoCodecVP8); + SetupCodec(); + EXPECT_EQ(0, adapter_->InitEncode(&codec_, kSettings)); + scoped_refptr buffer(I420Buffer::Create(1280, 720)); + VideoFrame input_frame = VideoFrame::Builder() + .set_video_frame_buffer(buffer) + .set_rtp_timestamp(0) + .set_timestamp_ms(0) + .build(); + std::vector frame_types; + frame_types.resize(codec_.numberOfSimulcastStreams, + VideoFrameType::kVideoFrameKey); + EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERROR, + adapter_->Encode(input_frame, &frame_types)); +} + } // namespace test } // namespace webrtc From 73535dcd1b4b689c6c25a1dc5324cdd82dabd0ed Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Fri, 2 May 2025 20:01:15 +0000 Subject: [PATCH 042/292] Deprecate trials and task_queue_factory in PeerConnectionFactoryDependencies Use Environment instead. Bug: webrtc:42220378 Change-Id: I852e29fa76ae0a5c5d838c5005327c579361f2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/388881 Reviewed-by: Harald Alvestrand Commit-Queue: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#44514} --- api/BUILD.gn | 5 ++- api/create_peerconnection_factory.cc | 3 +- api/peer_connection_interface.h | 8 ++--- examples/BUILD.gn | 3 +- examples/androidnativeapi/BUILD.gn | 7 ++++ .../jni/android_call_client.cc | 22 ++++++++++-- .../objcnativeapi/objc/objc_call_client.mm | 2 -- examples/peerconnection/client/conductor.cc | 15 ++++---- examples/peerconnection/client/conductor.h | 4 +-- pc/BUILD.gn | 2 +- pc/peer_connection_data_channel_unittest.cc | 2 -- pc/peer_connection_factory.cc | 3 ++ pc/peer_connection_factory_unittest.cc | 3 ++ pc/peer_connection_field_trial_tests.cc | 5 ++- ...er_connection_header_extension_unittest.cc | 2 -- pc/peer_connection_interface_unittest.cc | 4 --- pc/peer_connection_jsep_unittest.cc | 4 --- pc/peer_connection_media_unittest.cc | 2 -- pc/test/integration_test_helpers.cc | 7 ++-- sdk/BUILD.gn | 1 + sdk/android/BUILD.gn | 10 ++++++ .../peer_connection_factory_unittest.cc | 12 ++++--- .../src/jni/pc/peer_connection_factory.cc | 34 ++++++++++++++++--- .../RTCPeerConnectionFactory.mm | 9 ++--- .../RTCPeerConnectionFactoryBuilder.mm | 3 +- .../iwyu/iwyu-verifier-filter_list.json | 2 +- 26 files changed, 113 insertions(+), 61 deletions(-) diff --git a/api/BUILD.gn b/api/BUILD.gn index 939a4b2d74..70c3af22f5 100644 --- a/api/BUILD.gn +++ b/api/BUILD.gn @@ -83,10 +83,7 @@ if (!build_with_chromium) { ":field_trials_view", ":libjingle_peerconnection_api", ":scoped_refptr", - "../api/rtc_event_log:rtc_event_log_factory", "../pc:peer_connection_factory", - "../pc:webrtc_sdp", - "../rtc_base:socket_server", "../rtc_base:threading", "../rtc_base/system:rtc_export", "../stats:rtc_stats", @@ -95,6 +92,8 @@ if (!build_with_chromium) { "audio:audio_processing", "audio:builtin_audio_processing_builder", "audio_codecs:audio_codecs_api", + "environment:environment_factory", + "rtc_event_log:rtc_event_log_factory", "video_codecs:video_codecs_api", ] } diff --git a/api/create_peerconnection_factory.cc b/api/create_peerconnection_factory.cc index ededc00816..e0b50fc02e 100644 --- a/api/create_peerconnection_factory.cc +++ b/api/create_peerconnection_factory.cc @@ -20,6 +20,7 @@ #include "api/audio_codecs/audio_decoder_factory.h" #include "api/audio_codecs/audio_encoder_factory.h" #include "api/enable_media.h" +#include "api/environment/environment_factory.h" #include "api/field_trials_view.h" #include "api/peer_connection_interface.h" #include "api/rtc_event_log/rtc_event_log_factory.h" @@ -48,7 +49,7 @@ scoped_refptr CreatePeerConnectionFactory( dependencies.worker_thread = worker_thread; dependencies.signaling_thread = signaling_thread; dependencies.event_log_factory = std::make_unique(); - dependencies.trials = std::move(field_trials); + dependencies.env = webrtc::CreateEnvironment(std::move(field_trials)); if (network_thread) { // TODO(bugs.webrtc.org/13145): Add an webrtc::SocketFactory* argument. diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h index bffbf070d1..b02e055029 100644 --- a/api/peer_connection_interface.h +++ b/api/peer_connection_interface.h @@ -1440,9 +1440,7 @@ struct RTC_EXPORT PeerConnectionFactoryDependencies final { // The `packet_socket_factory` will only be used if CreatePeerConnection is // called without a `port_allocator`. std::unique_ptr packet_socket_factory; - // Deprecated. Instead provide custom task queue factory through the `env`. - // TODO: bugs.webrtc.org/42220378 - Mark [[deprecated]] once unused in WebRTC - // and chromium + [[deprecated("Pass custom task queue factory through the 'env'")]] std::unique_ptr task_queue_factory; std::unique_ptr event_log_factory; std::unique_ptr fec_controller_factory; @@ -1458,9 +1456,7 @@ struct RTC_EXPORT PeerConnectionFactoryDependencies final { std::unique_ptr network_monitor_factory; std::unique_ptr neteq_factory; std::unique_ptr sctp_factory; - // Deprecated. Instead provide custom field trials through the `env`. - // TODO: bugs.webrtc.org/42220378 - Mark [[deprecated]] once unused in WebRTC - // and chromium + [[deprecated("Pass custom field trials through the 'env'")]] std::unique_ptr trials; std::unique_ptr transport_controller_send_factory; diff --git a/examples/BUILD.gn b/examples/BUILD.gn index 69022724e4..e17d40ab0d 100644 --- a/examples/BUILD.gn +++ b/examples/BUILD.gn @@ -696,8 +696,9 @@ if (is_linux || is_chromeos || is_win) { "../api/audio:audio_mixer_api", "../api/audio:audio_processing", "../api/audio_codecs:audio_codecs_api", + "../api/environment", + "../api/environment:environment_factory", "../api/task_queue", - "../api/task_queue:default_task_queue_factory", "../api/task_queue:pending_task_safety_flag", "../api/units:time_delta", "../api/video:video_frame", diff --git a/examples/androidnativeapi/BUILD.gn b/examples/androidnativeapi/BUILD.gn index 65ab875893..7211192206 100644 --- a/examples/androidnativeapi/BUILD.gn +++ b/examples/androidnativeapi/BUILD.gn @@ -48,10 +48,17 @@ if (is_android) { deps = [ ":generated_jni", "../../api:enable_media_with_defaults", + "../../api:make_ref_counted", + "../../api:media_stream_interface", + "../../api:rtc_error", "../../api:scoped_refptr", "../../api:sequence_checker", + "../../api/video:video_frame", + "../../rtc_base:checks", + "../../rtc_base:logging", "../../rtc_base:ssl", "../../rtc_base:ssl_adapter", + "../../rtc_base:threading", "../../rtc_base/synchronization:mutex", "//api:libjingle_peerconnection_api", "//api/rtc_event_log:rtc_event_log_factory", diff --git a/examples/androidnativeapi/jni/android_call_client.cc b/examples/androidnativeapi/jni/android_call_client.cc index b61af374d4..33ccc2e5b1 100644 --- a/examples/androidnativeapi/jni/android_call_client.cc +++ b/examples/androidnativeapi/jni/android_call_client.cc @@ -10,18 +10,35 @@ #include "examples/androidnativeapi/jni/android_call_client.h" +#include + #include +#include #include +#include "api/data_channel_interface.h" #include "api/enable_media_with_defaults.h" +#include "api/jsep.h" +#include "api/make_ref_counted.h" +#include "api/media_stream_interface.h" #include "api/peer_connection_interface.h" +#include "api/rtc_error.h" #include "api/rtc_event_log/rtc_event_log_factory.h" -#include "api/task_queue/default_task_queue_factory.h" +#include "api/rtp_transceiver_interface.h" +#include "api/scoped_refptr.h" +#include "api/sequence_checker.h" +#include "api/set_remote_description_observer_interface.h" +#include "api/video/video_source_interface.h" #include "examples/androidnativeapi/generated_jni/CallClient_jni.h" #include "media/engine/internal_decoder_factory.h" #include "media/engine/internal_encoder_factory.h" -#include "media/engine/webrtc_media_engine.h" +#include "rtc_base/checks.h" +#include "rtc_base/logging.h" +#include "rtc_base/synchronization/mutex.h" +#include "rtc_base/thread.h" #include "sdk/android/native_api/jni/java_types.h" +#include "sdk/android/native_api/jni/scoped_java_ref.h" +#include "sdk/android/native_api/video/video_source.h" #include "sdk/android/native_api/video/wrapper.h" namespace webrtc_examples { @@ -153,7 +170,6 @@ void AndroidCallClient::CreatePeerConnectionFactory() { pcf_deps.network_thread = network_thread_.get(); pcf_deps.worker_thread = worker_thread_.get(); pcf_deps.signaling_thread = signaling_thread_.get(); - pcf_deps.task_queue_factory = webrtc::CreateDefaultTaskQueueFactory(); pcf_deps.event_log_factory = std::make_unique(); pcf_deps.video_encoder_factory = diff --git a/examples/objcnativeapi/objc/objc_call_client.mm b/examples/objcnativeapi/objc/objc_call_client.mm index 15406ad7d0..a0c9e4bfdb 100644 --- a/examples/objcnativeapi/objc/objc_call_client.mm +++ b/examples/objcnativeapi/objc/objc_call_client.mm @@ -25,7 +25,6 @@ #include "api/enable_media.h" #include "api/peer_connection_interface.h" #include "api/rtc_event_log/rtc_event_log_factory.h" -#include "api/task_queue/default_task_queue_factory.h" #include "sdk/objc/native/api/video_capturer.h" #include "sdk/objc/native/api/video_decoder_factory.h" #include "sdk/objc/native/api/video_encoder_factory.h" @@ -121,7 +120,6 @@ explicit CreateOfferObserver( dependencies.network_thread = network_thread_.get(); dependencies.worker_thread = worker_thread_.get(); dependencies.signaling_thread = signaling_thread_.get(); - dependencies.task_queue_factory = webrtc::CreateDefaultTaskQueueFactory(); dependencies.audio_encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory(); dependencies.audio_decoder_factory = diff --git a/examples/peerconnection/client/conductor.cc b/examples/peerconnection/client/conductor.cc index 8356484207..34aefe96f1 100644 --- a/examples/peerconnection/client/conductor.cc +++ b/examples/peerconnection/client/conductor.cc @@ -22,8 +22,8 @@ #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" #include "api/audio_options.h" -#include "api/create_peerconnection_factory.h" #include "api/enable_media.h" +#include "api/environment/environment_factory.h" #include "api/jsep.h" #include "api/make_ref_counted.h" #include "api/media_stream_interface.h" @@ -32,7 +32,6 @@ #include "api/rtp_receiver_interface.h" #include "api/rtp_sender_interface.h" #include "api/scoped_refptr.h" -#include "api/task_queue/default_task_queue_factory.h" #include "api/task_queue/task_queue_factory.h" #include "api/test/create_frame_generator.h" #include "api/video/video_frame.h" @@ -59,6 +58,7 @@ #include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/strings/json.h" +#include "rtc_base/thread.h" #include "system_wrappers/include/clock.h" #include "test/frame_generator_capturer.h" #include "test/platform_video_capturer.h" @@ -141,7 +141,11 @@ class CapturerTrackSource : public webrtc::VideoTrackSource { } // namespace Conductor::Conductor(PeerConnectionClient* client, MainWindow* main_wnd) - : peer_id_(-1), loopback_(false), client_(client), main_wnd_(main_wnd) { + : peer_id_(-1), + loopback_(false), + env_(webrtc::CreateEnvironment()), + client_(client), + main_wnd_(main_wnd) { client_->RegisterObserver(this); main_wnd->RegisterObserver(this); } @@ -170,7 +174,7 @@ bool Conductor::InitializePeerConnection() { webrtc::PeerConnectionFactoryDependencies deps; deps.signaling_thread = signaling_thread_.get(); - deps.task_queue_factory = webrtc::CreateDefaultTaskQueueFactory(), + deps.env = env_, deps.audio_encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory(); deps.audio_decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory(); deps.video_encoder_factory = @@ -186,7 +190,6 @@ bool Conductor::InitializePeerConnection() { webrtc::OpenH264DecoderTemplateAdapter, webrtc::Dav1dDecoderTemplateAdapter>>(); webrtc::EnableMedia(deps); - task_queue_factory_ = deps.task_queue_factory.get(); peer_connection_factory_ = webrtc::CreateModularPeerConnectionFactory(std::move(deps)); @@ -514,7 +517,7 @@ void Conductor::AddTracks() { } webrtc::scoped_refptr video_device = - CapturerTrackSource::Create(*task_queue_factory_); + CapturerTrackSource::Create(env_.task_queue_factory()); if (video_device) { webrtc::scoped_refptr video_track_( peer_connection_factory_->CreateVideoTrack(video_device, kVideoLabel)); diff --git a/examples/peerconnection/client/conductor.h b/examples/peerconnection/client/conductor.h index ffc5c27012..3970963f8e 100644 --- a/examples/peerconnection/client/conductor.h +++ b/examples/peerconnection/client/conductor.h @@ -17,13 +17,13 @@ #include #include "api/data_channel_interface.h" +#include "api/environment/environment.h" #include "api/jsep.h" #include "api/media_stream_interface.h" #include "api/peer_connection_interface.h" #include "api/rtc_error.h" #include "api/rtp_receiver_interface.h" #include "api/scoped_refptr.h" -#include "api/task_queue/task_queue_factory.h" #include "examples/peerconnection/client/main_wnd.h" #include "examples/peerconnection/client/peer_connection_client.h" #include "rtc_base/thread.h" @@ -124,8 +124,8 @@ class Conductor : public webrtc::PeerConnectionObserver, int peer_id_; bool loopback_; + webrtc::Environment env_; std::unique_ptr signaling_thread_; - webrtc::TaskQueueFactory* task_queue_factory_ = nullptr; webrtc::scoped_refptr peer_connection_; webrtc::scoped_refptr peer_connection_factory_; diff --git a/pc/BUILD.gn b/pc/BUILD.gn index d6b0acbe56..00bba5ec57 100644 --- a/pc/BUILD.gn +++ b/pc/BUILD.gn @@ -2699,10 +2699,10 @@ if (rtc_include_tests && !build_with_chromium) { "../api:sequence_checker", "../api/audio:builtin_audio_processing_builder", "../api/crypto:options", + "../api/environment:environment_factory", "../api/metronome", "../api/rtc_event_log:rtc_event_log_factory", "../api/task_queue", - "../api/task_queue:default_task_queue_factory", "../api/task_queue:pending_task_safety_flag", "../api/units:time_delta", "../api/video:video_rtp_headers", diff --git a/pc/peer_connection_data_channel_unittest.cc b/pc/peer_connection_data_channel_unittest.cc index 47a85a5e48..5cf750e456 100644 --- a/pc/peer_connection_data_channel_unittest.cc +++ b/pc/peer_connection_data_channel_unittest.cc @@ -19,7 +19,6 @@ #include "api/peer_connection_interface.h" #include "api/scoped_refptr.h" #include "api/sctp_transport_interface.h" -#include "api/task_queue/default_task_queue_factory.h" #include "p2p/base/p2p_constants.h" #include "pc/media_session.h" #include "pc/peer_connection.h" @@ -56,7 +55,6 @@ PeerConnectionFactoryDependencies CreatePeerConnectionFactoryDependencies() { deps.network_thread = Thread::Current(); deps.worker_thread = Thread::Current(); deps.signaling_thread = Thread::Current(); - deps.task_queue_factory = CreateDefaultTaskQueueFactory(); EnableFakeMedia(deps); deps.sctp_factory = std::make_unique(); return deps; diff --git a/pc/peer_connection_factory.cc b/pc/peer_connection_factory.cc index d23aec1363..60820a16d7 100644 --- a/pc/peer_connection_factory.cc +++ b/pc/peer_connection_factory.cc @@ -77,8 +77,11 @@ Environment AssembleEnvironment(PeerConnectionFactoryDependencies& deps) { EnvironmentFactory env_factory = deps.env.has_value() ? EnvironmentFactory(*deps.env) : EnvironmentFactory(); +#pragma clang diagnostic push +#pragma clang diagnostic ignored "-Wdeprecated-declarations" env_factory.Set(std::move(deps.trials)); env_factory.Set(std::move(deps.task_queue_factory)); +#pragma clang diagnostic pop // Clear Environment from `deps` to avoid accidental usage of the wrong // Environment. diff --git a/pc/peer_connection_factory_unittest.cc b/pc/peer_connection_factory_unittest.cc index aeb79875f3..7511471f3a 100644 --- a/pc/peer_connection_factory_unittest.cc +++ b/pc/peer_connection_factory_unittest.cc @@ -691,7 +691,10 @@ TEST(PeerConnectionFactoryDependenciesTest, PeerConnectionFactoryDependencies pcf_dependencies; pcf_dependencies.env = CreateEnvironment(std::move(env_field_trials)); +#pragma clang diagnostic push +#pragma clang diagnostic ignored "-Wdeprecated-declarations" pcf_dependencies.trials = std::move(explicit_field_trials); +#pragma clang diagnostic pop pcf_dependencies.adm = FakeAudioCaptureModule::Create(); EnableMediaWithDefaults(pcf_dependencies); diff --git a/pc/peer_connection_field_trial_tests.cc b/pc/peer_connection_field_trial_tests.cc index 9ce3f16415..a0b30b0e94 100644 --- a/pc/peer_connection_field_trial_tests.cc +++ b/pc/peer_connection_field_trial_tests.cc @@ -17,13 +17,13 @@ #include "absl/algorithm/container.h" #include "api/enable_media_with_defaults.h" +#include "api/environment/environment_factory.h" #include "api/field_trials.h" #include "api/field_trials_view.h" #include "api/media_types.h" #include "api/peer_connection_interface.h" #include "api/rtp_parameters.h" #include "api/scoped_refptr.h" -#include "api/task_queue/default_task_queue_factory.h" #include "pc/peer_connection_wrapper.h" #include "pc/session_description.h" #include "pc/test/fake_audio_capture_module.h" @@ -65,8 +65,7 @@ class PeerConnectionFieldTrialTest : public ::testing::Test { void CreatePCFactory(std::unique_ptr field_trials) { PeerConnectionFactoryDependencies pcf_deps; pcf_deps.signaling_thread = Thread::Current(); - pcf_deps.trials = std::move(field_trials); - pcf_deps.task_queue_factory = CreateDefaultTaskQueueFactory(); + pcf_deps.env = CreateEnvironment(std::move(field_trials)); pcf_deps.adm = FakeAudioCaptureModule::Create(); EnableMediaWithDefaults(pcf_deps); pc_factory_ = CreateModularPeerConnectionFactory(std::move(pcf_deps)); diff --git a/pc/peer_connection_header_extension_unittest.cc b/pc/peer_connection_header_extension_unittest.cc index b62d68b7aa..def77de11d 100644 --- a/pc/peer_connection_header_extension_unittest.cc +++ b/pc/peer_connection_header_extension_unittest.cc @@ -26,7 +26,6 @@ #include "api/rtp_transceiver_direction.h" #include "api/rtp_transceiver_interface.h" #include "api/scoped_refptr.h" -#include "api/task_queue/default_task_queue_factory.h" #include "media/base/fake_media_engine.h" #include "p2p/test/fake_port_allocator.h" #include "pc/peer_connection_wrapper.h" @@ -82,7 +81,6 @@ class PeerConnectionHeaderExtensionTest factory_dependencies.network_thread = Thread::Current(); factory_dependencies.worker_thread = Thread::Current(); factory_dependencies.signaling_thread = Thread::Current(); - factory_dependencies.task_queue_factory = CreateDefaultTaskQueueFactory(); EnableFakeMedia(factory_dependencies, std::move(media_engine)); factory_dependencies.event_log_factory = diff --git a/pc/peer_connection_interface_unittest.cc b/pc/peer_connection_interface_unittest.cc index d56492afde..b917484e89 100644 --- a/pc/peer_connection_interface_unittest.cc +++ b/pc/peer_connection_interface_unittest.cc @@ -42,11 +42,9 @@ #include "api/rtp_sender_interface.h" #include "api/rtp_transceiver_direction.h" #include "api/scoped_refptr.h" -#include "api/task_queue/default_task_queue_factory.h" #include "api/test/rtc_error_matchers.h" #include "api/transport/bitrate_settings.h" #include "api/transport/enums.h" -#include "api/transport/field_trial_based_config.h" #include "api/units/time_delta.h" #include "api/video_codecs/video_decoder_factory_template.h" #include "api/video_codecs/video_decoder_factory_template_dav1d_adapter.h" @@ -612,8 +610,6 @@ class PeerConnectionFactoryForTest : public PeerConnectionFactory { dependencies.worker_thread = Thread::Current(); dependencies.network_thread = Thread::Current(); dependencies.signaling_thread = Thread::Current(); - dependencies.task_queue_factory = CreateDefaultTaskQueueFactory(); - dependencies.trials = std::make_unique(); // Use fake audio device module since we're only testing the interface // level, and using a real one could make tests flaky when run in parallel. dependencies.adm = FakeAudioCaptureModule::Create(); diff --git a/pc/peer_connection_jsep_unittest.cc b/pc/peer_connection_jsep_unittest.cc index b7f62a1729..b9a3911ca4 100644 --- a/pc/peer_connection_jsep_unittest.cc +++ b/pc/peer_connection_jsep_unittest.cc @@ -32,9 +32,7 @@ #include "api/rtp_transceiver_direction.h" #include "api/rtp_transceiver_interface.h" #include "api/scoped_refptr.h" -#include "api/task_queue/default_task_queue_factory.h" #include "api/test/rtc_error_matchers.h" -#include "api/transport/field_trial_based_config.h" #include "media/base/stream_params.h" #include "p2p/base/p2p_constants.h" #include "p2p/base/transport_info.h" @@ -73,8 +71,6 @@ PeerConnectionFactoryDependencies CreatePeerConnectionFactoryDependencies() { dependencies.worker_thread = Thread::Current(); dependencies.network_thread = Thread::Current(); dependencies.signaling_thread = Thread::Current(); - dependencies.task_queue_factory = CreateDefaultTaskQueueFactory(); - dependencies.trials = std::make_unique(); dependencies.adm = FakeAudioCaptureModule::Create(); EnableMediaWithDefaults(dependencies); dependencies.sctp_factory = std::make_unique(); diff --git a/pc/peer_connection_media_unittest.cc b/pc/peer_connection_media_unittest.cc index 242984a675..7eab5c5851 100644 --- a/pc/peer_connection_media_unittest.cc +++ b/pc/peer_connection_media_unittest.cc @@ -35,7 +35,6 @@ #include "api/rtp_transceiver_direction.h" #include "api/rtp_transceiver_interface.h" #include "api/scoped_refptr.h" -#include "api/task_queue/default_task_queue_factory.h" #include "media/base/codec.h" #include "media/base/fake_media_engine.h" #include "media/base/media_channel.h" @@ -169,7 +168,6 @@ class PeerConnectionMediaBaseTest : public ::testing::Test { factory_dependencies.network_thread = Thread::Current(); factory_dependencies.worker_thread = Thread::Current(); factory_dependencies.signaling_thread = Thread::Current(); - factory_dependencies.task_queue_factory = CreateDefaultTaskQueueFactory(); EnableFakeMedia(factory_dependencies, std::move(media_engine)); factory_dependencies.event_log_factory = std::make_unique(); diff --git a/pc/test/integration_test_helpers.cc b/pc/test/integration_test_helpers.cc index 23a712a3c8..4fc57c3bf5 100644 --- a/pc/test/integration_test_helpers.cc +++ b/pc/test/integration_test_helpers.cc @@ -20,18 +20,20 @@ #include "absl/functional/any_invocable.h" #include "api/audio/builtin_audio_processing_builder.h" #include "api/enable_media_with_defaults.h" +#include "api/environment/environment_factory.h" #include "api/field_trials_view.h" #include "api/jsep.h" #include "api/peer_connection_interface.h" #include "api/rtc_event_log/rtc_event_log_factory.h" #include "api/sequence_checker.h" #include "api/stats/rtcstats_objects.h" -#include "api/task_queue/default_task_queue_factory.h" #include "api/task_queue/pending_task_safety_flag.h" #include "api/task_queue/task_queue_base.h" #include "api/units/time_delta.h" #include "logging/rtc_event_log/fake_rtc_event_log_factory.h" +#include "media/base/stream_params.h" #include "pc/peer_connection_factory.h" +#include "pc/session_description.h" #include "pc/test/fake_audio_capture_module.h" #include "rtc_base/checks.h" #include "rtc_base/fake_network.h" @@ -247,8 +249,7 @@ bool PeerConnectionIntegrationWrapper::Init( pc_factory_dependencies.signaling_thread = signaling_thread; pc_factory_dependencies.socket_factory = socket_server; pc_factory_dependencies.network_manager = std::move(network_manager); - pc_factory_dependencies.task_queue_factory = CreateDefaultTaskQueueFactory(); - pc_factory_dependencies.trials = std::move(field_trials); + pc_factory_dependencies.env = CreateEnvironment(std::move(field_trials)); pc_factory_dependencies.decode_metronome = std::make_unique(TimeDelta::Millis(8)); diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn index b556063e8a..c0c8eb4796 100644 --- a/sdk/BUILD.gn +++ b/sdk/BUILD.gn @@ -1101,6 +1101,7 @@ if (is_ios || is_mac) { "../api/audio_codecs:builtin_audio_encoder_factory", "../api/crypto:frame_decryptor_interface", "../api/crypto:frame_encryptor_interface", + "../api/environment:environment_factory", "../api/rtc_event_log:rtc_event_log_factory", "../api/task_queue:default_task_queue_factory", "../api/transport:field_trial_based_config", diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn index 6d8992f8b4..d9ca1e008b 100644 --- a/sdk/android/BUILD.gn +++ b/sdk/android/BUILD.gn @@ -782,25 +782,33 @@ if (current_os == "linux" || is_android) { ":generated_external_classes_jni", ":generated_peerconnection_jni", ":generated_rtcerror_jni", + ":internal_jni", ":logging_jni", ":native_api_jni", ":native_api_stacktrace", "..:media_constraints", + "../../api:audio_options_api", "../../api:dtmf_sender_interface", "../../api:enable_media", + "../../api:fec_controller_api", "../../api:libjingle_peerconnection_api", "../../api:media_stream_interface", + "../../api:network_state_predictor_api", "../../api:rtc_event_log_output_file", "../../api:rtc_stats_api", "../../api:rtp_parameters", "../../api:rtp_sender_interface", + "../../api:scoped_refptr", "../../api:turn_customizer", "../../api/audio:audio_device", "../../api/audio:audio_processing", "../../api/audio:builtin_audio_processing_builder", + "../../api/audio_codecs:audio_codecs_api", "../../api/crypto:options", + "../../api/neteq:neteq_api", "../../api/rtc_event_log:rtc_event_log_factory", "../../api/task_queue:default_task_queue_factory", + "../../api/transport:network_control", "../../api/video_codecs:video_codecs_api", "../../call:call_interfaces", "../../media:rtc_media_base", @@ -816,6 +824,7 @@ if (current_os == "linux" || is_android) { "../../rtc_base:refcount", "../../rtc_base:rtc_certificate_generator", "../../rtc_base:safe_conversions", + "../../rtc_base:socket_factory", "../../rtc_base:ssl", "../../rtc_base:stringutils", "../../rtc_base:threading", @@ -1671,6 +1680,7 @@ if (is_android) { ":video_jni", "../../api:enable_media_with_defaults", "../../api:field_trials_view", + "../../api:libjingle_peerconnection_api", "../../api:scoped_refptr", "../../api/audio:audio_device", "../../api/environment", diff --git a/sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc b/sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc index 110fbb1b17..0e243c527e 100644 --- a/sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc +++ b/sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc @@ -9,15 +9,19 @@ */ #include "sdk/android/native_api/peerconnection/peer_connection_factory.h" +#include + +#include #include +#include #include "api/enable_media_with_defaults.h" +#include "api/peer_connection_interface.h" #include "api/rtc_event_log/rtc_event_log_factory.h" -#include "api/task_queue/default_task_queue_factory.h" -#include "media/base/media_engine.h" +#include "api/scoped_refptr.h" #include "media/engine/internal_decoder_factory.h" #include "media/engine/internal_encoder_factory.h" -#include "media/engine/webrtc_media_engine.h" +#include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/physical_socket_server.h" #include "rtc_base/thread.h" @@ -25,7 +29,6 @@ #include "sdk/android/native_api/audio_device_module/audio_device_android.h" #include "sdk/android/native_api/jni/application_context_provider.h" #include "sdk/android/native_api/jni/jvm.h" -#include "sdk/android/src/jni/jni_helpers.h" #include "test/gtest.h" namespace webrtc { @@ -49,7 +52,6 @@ webrtc::scoped_refptr CreateTestPCF( pcf_deps.network_thread = network_thread; pcf_deps.worker_thread = worker_thread; pcf_deps.signaling_thread = signaling_thread; - pcf_deps.task_queue_factory = CreateDefaultTaskQueueFactory(); pcf_deps.event_log_factory = std::make_unique(); pcf_deps.adm = CreateJavaAudioDeviceModule(jni, GetAppContext(jni).obj()); diff --git a/sdk/android/src/jni/pc/peer_connection_factory.cc b/sdk/android/src/jni/pc/peer_connection_factory.cc index 7404a950f8..f05259b478 100644 --- a/sdk/android/src/jni/pc/peer_connection_factory.cc +++ b/sdk/android/src/jni/pc/peer_connection_factory.cc @@ -10,35 +10,60 @@ #include "sdk/android/src/jni/pc/peer_connection_factory.h" +#include +#include +#include + +#include +#include #include +#include +#include #include #include "absl/memory/memory.h" #include "api/audio/audio_device.h" #include "api/audio/audio_processing.h" #include "api/audio/builtin_audio_processing_builder.h" +#include "api/audio_codecs/audio_decoder_factory.h" +#include "api/audio_codecs/audio_encoder_factory.h" +#include "api/audio_options.h" #include "api/enable_media.h" +#include "api/fec_controller.h" +#include "api/media_stream_interface.h" +#include "api/neteq/neteq_factory.h" +#include "api/network_state_predictor.h" +#include "api/peer_connection_interface.h" #include "api/rtc_event_log/rtc_event_log_factory.h" -#include "api/task_queue/default_task_queue_factory.h" -#include "api/video_codecs/video_decoder_factory.h" -#include "api/video_codecs/video_encoder_factory.h" +#include "api/scoped_refptr.h" +#include "api/transport/network_control.h" #include "modules/utility/include/jvm_android.h" +#include "rtc_base/checks.h" #include "rtc_base/event_tracer.h" +#include "rtc_base/logging.h" #include "rtc_base/physical_socket_server.h" +#include "rtc_base/rtc_certificate.h" +#include "rtc_base/rtc_certificate_generator.h" +#include "rtc_base/socket_factory.h" +#include "rtc_base/ssl_identity.h" #include "rtc_base/thread.h" #include "sdk/android/generated_peerconnection_jni/PeerConnectionFactory_jni.h" #include "sdk/android/native_api/jni/java_types.h" +#include "sdk/android/native_api/jni/scoped_java_ref.h" #include "sdk/android/native_api/stacktrace/stacktrace.h" +#include "sdk/android/src/jni/android_network_monitor.h" #include "sdk/android/src/jni/jni_helpers.h" +#include "sdk/android/src/jni/jvm.h" #include "sdk/android/src/jni/logging/log_sink.h" #include "sdk/android/src/jni/pc/android_network_monitor.h" -#include "sdk/android/src/jni/pc/ice_candidate.h" +#include "sdk/android/src/jni/pc/media_constraints.h" #include "sdk/android/src/jni/pc/media_stream_track.h" #include "sdk/android/src/jni/pc/owned_factory_and_threads.h" #include "sdk/android/src/jni/pc/peer_connection.h" #include "sdk/android/src/jni/pc/rtp_capabilities.h" #include "sdk/android/src/jni/pc/ssl_certificate_verifier_wrapper.h" #include "sdk/android/src/jni/pc/video.h" +#include "sdk/media_constraints.h" #include "system_wrappers/include/field_trial.h" #include "third_party/jni_zero/jni_zero.h" @@ -272,7 +297,6 @@ ScopedJavaLocalRef CreatePeerConnectionFactoryForJava( dependencies.network_thread = network_thread.get(); dependencies.worker_thread = worker_thread.get(); dependencies.signaling_thread = signaling_thread.get(); - dependencies.task_queue_factory = CreateDefaultTaskQueueFactory(); dependencies.event_log_factory = std::make_unique(); dependencies.fec_controller_factory = std::move(fec_controller_factory); dependencies.network_controller_factory = diff --git a/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm b/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm index 81eafe68df..54618fba63 100644 --- a/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm +++ b/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm @@ -36,9 +36,9 @@ #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" #include "api/enable_media.h" +#include "api/environment/environment_factory.h" #include "api/rtc_event_log/rtc_event_log_factory.h" #include "api/task_queue/default_task_queue_factory.h" -#include "api/transport/field_trial_based_config.h" #import "components/video_codec/RTCVideoDecoderFactoryH264.h" #import "components/video_codec/RTCVideoEncoderFactoryH264.h" #include "media/base/media_constants.h" @@ -150,11 +150,12 @@ - (instancetype)initWithNativeDependencies: dependencies.network_thread = _networkThread.get(); dependencies.worker_thread = _workerThread.get(); dependencies.signaling_thread = _signalingThread.get(); - if (dependencies.trials == nullptr) { - dependencies.trials = std::make_unique(); + if (!dependencies.env.has_value()) { + dependencies.env = webrtc::CreateEnvironment(); } if (dependencies.network_monitor_factory == nullptr && - dependencies.trials->IsEnabled("WebRTC-Network-UseNWPathMonitor")) { + dependencies.env->field_trials().IsEnabled( + "WebRTC-Network-UseNWPathMonitor")) { dependencies.network_monitor_factory = webrtc::CreateNetworkMonitorFactory(); } diff --git a/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm b/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm index 3485dc8d1e..6e2f7c4ef2 100644 --- a/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm +++ b/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm @@ -15,6 +15,7 @@ #include "api/audio/audio_processing.h" #include "api/audio_codecs/audio_decoder_factory.h" #include "api/audio_codecs/audio_encoder_factory.h" +#include "api/environment/environment_factory.h" #include "api/video_codecs/video_decoder_factory.h" #include "api/video_codecs/video_encoder_factory.h" @@ -32,7 +33,7 @@ + (RTCPeerConnectionFactoryBuilder *)builder { } - (void)setFieldTrials:(std::unique_ptr)fieldTrials { - _dependencies.trials = std::move(fieldTrials); + _dependencies.env = webrtc::CreateEnvironment(std::move(fieldTrials)); } - (void)setVideoEncoderFactory: diff --git a/tools_webrtc/iwyu/iwyu-verifier-filter_list.json b/tools_webrtc/iwyu/iwyu-verifier-filter_list.json index 364d2c895a..fd5d8b3751 100644 --- a/tools_webrtc/iwyu/iwyu-verifier-filter_list.json +++ b/tools_webrtc/iwyu/iwyu-verifier-filter_list.json @@ -8,7 +8,7 @@ "modules/desktop_capture/", "modules/desktop_capture/win/", "rtc_base/win", - "sdk/android/src/jni/", + "sdk/android/", "sdk/objc/", "test/ios/", "test/mac/" From 7b17217e592aa4cc548007c1044a121938f9635f Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Fri, 2 May 2025 15:28:41 -0700 Subject: [PATCH 043/292] Move priority implementation to cc file follow-up from https://webrtc-review.googlesource.com/c/src/+/382120 where IWYU failed on that header-only target. BUG=webrtc:42226242 Change-Id: I8555e62e5324db8f3f0554879f61fdffdc053419 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/386061 Commit-Queue: Philipp Hancke Reviewed-by: Harald Alvestrand Reviewed-by: Evan Shrubsole Cr-Commit-Position: refs/heads/main@{#44515} --- api/BUILD.gn | 8 ++++++-- api/priority.cc | 36 ++++++++++++++++++++++++++++++++++++ api/priority.h | 24 +++--------------------- 3 files changed, 45 insertions(+), 23 deletions(-) create mode 100644 api/priority.cc diff --git a/api/BUILD.gn b/api/BUILD.gn index 70c3af22f5..3e4bd7762c 100644 --- a/api/BUILD.gn +++ b/api/BUILD.gn @@ -525,12 +525,16 @@ rtc_source_set("rtp_transceiver_direction") { sources = [ "rtp_transceiver_direction.h" ] } -rtc_source_set("priority") { +rtc_library("priority") { visibility = [ "*" ] - sources = [ "priority.h" ] + sources = [ + "priority.cc", + "priority.h", + ] deps = [ "../rtc_base:checks", "../rtc_base:strong_alias", + "../rtc_base/system:rtc_export", ] } diff --git a/api/priority.cc b/api/priority.cc new file mode 100644 index 0000000000..3ae4fd5c9c --- /dev/null +++ b/api/priority.cc @@ -0,0 +1,36 @@ +/* + * Copyright 2025 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "api/priority.h" + +#include "rtc_base/checks.h" + +namespace webrtc { + +PriorityValue::PriorityValue(Priority priority) { + switch (priority) { + case Priority::kVeryLow: + value_ = 128; + break; + case Priority::kLow: + value_ = 256; + break; + case Priority::kMedium: + value_ = 512; + break; + case Priority::kHigh: + value_ = 1024; + break; + default: + RTC_CHECK_NOTREACHED(); + } +} + +} // namespace webrtc diff --git a/api/priority.h b/api/priority.h index 2735c2c282..bb2f88187b 100644 --- a/api/priority.h +++ b/api/priority.h @@ -13,8 +13,8 @@ #include -#include "rtc_base/checks.h" #include "rtc_base/strong_alias.h" +#include "rtc_base/system/rtc_export.h" namespace webrtc { @@ -26,28 +26,10 @@ enum class Priority { kHigh, }; -class PriorityValue +class RTC_EXPORT PriorityValue : public webrtc::StrongAlias { public: - explicit PriorityValue(Priority priority) { - switch (priority) { - case Priority::kVeryLow: - value_ = 128; - break; - case Priority::kLow: - value_ = 256; - break; - case Priority::kMedium: - value_ = 512; - break; - case Priority::kHigh: - value_ = 1024; - break; - default: - RTC_CHECK_NOTREACHED(); - } - } - + explicit PriorityValue(Priority priority); explicit PriorityValue(uint16_t priority) : StrongAlias(priority) {} }; From 3d09240cd078eb2ab45dccc1553c84ad5ba59a7b Mon Sep 17 00:00:00 2001 From: Harald Alvestrand Date: Mon, 5 May 2025 13:32:53 +0000 Subject: [PATCH 044/292] Change some typedefs to "using" statements These are supposed to be equivalent, and "using" is preferred by the style guide. Bug: None Change-Id: I8a32fcdb600faf1c46fa3f3c283eb564e9f9d3a4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389820 Auto-Submit: Harald Alvestrand Reviewed-by: Evan Shrubsole Commit-Queue: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44516} --- pc/channel_unittest.cc | 30 +++++++++++++++--------------- pc/external_hmac.h | 4 ++-- pc/legacy_stats_collector.cc | 8 ++++---- pc/session_description.h | 11 ++++------- 4 files changed, 25 insertions(+), 28 deletions(-) diff --git a/pc/channel_unittest.cc b/pc/channel_unittest.cc index bdb4a05a9d..0f52078245 100644 --- a/pc/channel_unittest.cc +++ b/pc/channel_unittest.cc @@ -45,11 +45,11 @@ #include "p2p/dtls/fake_dtls_transport.h" #include "p2p/test/fake_packet_transport.h" #include "pc/dtls_srtp_transport.h" -#include "pc/jsep_transport.h" #include "pc/rtp_transport.h" #include "pc/rtp_transport_internal.h" #include "pc/session_description.h" #include "rtc_base/arraysize.h" +#include "rtc_base/async_packet_socket.h" #include "rtc_base/buffer.h" #include "rtc_base/byte_order.h" #include "rtc_base/checks.h" @@ -103,14 +103,14 @@ template class Traits { public: - typedef ChannelT Channel; - typedef MediaSendChannelT MediaSendChannel; - typedef MediaReceiveChannelT MediaReceiveChannel; - typedef MediaSendChannelInterfaceT MediaSendChannelInterface; - typedef MediaReceiveChannelInterfaceT MediaReceiveChannelInterface; - typedef ContentT Content; - typedef MediaInfoT MediaInfo; - typedef OptionsT Options; + using Channel = ChannelT; + using MediaSendChannel = MediaSendChannelT; + using MediaReceiveChannel = MediaReceiveChannelT; + using MediaSendChannelInterface = MediaSendChannelInterfaceT; + using MediaReceiveChannelInterface = MediaReceiveChannelInterfaceT; + using Content = ContentT; + using MediaInfo = MediaInfoT; + using Options = OptionsT; }; class VoiceTraits : public Traits::AddLegacyStreamInContent( class VoiceChannelSingleThreadTest : public ChannelTest { public: - typedef ChannelTest Base; + using Base = ChannelTest; VoiceChannelSingleThreadTest() : Base(true, kPcmuFrame, kRtcpReport, NetworkIsWorker::Yes) {} }; class VoiceChannelDoubleThreadTest : public ChannelTest { public: - typedef ChannelTest Base; + using Base = ChannelTest; VoiceChannelDoubleThreadTest() : Base(true, kPcmuFrame, kRtcpReport, NetworkIsWorker::No) {} }; @@ -1659,7 +1659,7 @@ class VoiceChannelDoubleThreadTest : public ChannelTest { class VoiceChannelWithEncryptedRtpHeaderExtensionsSingleThreadTest : public ChannelTest { public: - typedef ChannelTest Base; + using Base = ChannelTest; VoiceChannelWithEncryptedRtpHeaderExtensionsSingleThreadTest() : Base(true, kPcmuFrameWithExtensions, @@ -1670,7 +1670,7 @@ class VoiceChannelWithEncryptedRtpHeaderExtensionsSingleThreadTest class VoiceChannelWithEncryptedRtpHeaderExtensionsDoubleThreadTest : public ChannelTest { public: - typedef ChannelTest Base; + using Base = ChannelTest; VoiceChannelWithEncryptedRtpHeaderExtensionsDoubleThreadTest() : Base(true, kPcmuFrameWithExtensions, kRtcpReport, NetworkIsWorker::No) { } @@ -1724,14 +1724,14 @@ void ChannelTest::AddLegacyStreamInContent( class VideoChannelSingleThreadTest : public ChannelTest { public: - typedef ChannelTest Base; + using Base = ChannelTest; VideoChannelSingleThreadTest() : Base(false, kH264Packet, kRtcpReport, NetworkIsWorker::Yes) {} }; class VideoChannelDoubleThreadTest : public ChannelTest { public: - typedef ChannelTest Base; + using Base = ChannelTest; VideoChannelDoubleThreadTest() : Base(false, kH264Packet, kRtcpReport, NetworkIsWorker::No) {} }; diff --git a/pc/external_hmac.h b/pc/external_hmac.h index 3319beaed4..7dec6360df 100644 --- a/pc/external_hmac.h +++ b/pc/external_hmac.h @@ -40,10 +40,10 @@ // The HMAC context structure used to store authentication keys. // The pointer to the key will be allocated in the external_hmac_init function. // This pointer is owned by srtp_t in a template context. -typedef struct { +struct ExternalHmacContext { uint8_t key[HMAC_KEY_LENGTH]; int key_length; -} ExternalHmacContext; +}; srtp_err_status_t external_hmac_alloc(srtp_auth_t** a, int key_len, diff --git a/pc/legacy_stats_collector.cc b/pc/legacy_stats_collector.cc index 87cc3eb7f8..2529b88c89 100644 --- a/pc/legacy_stats_collector.cc +++ b/pc/legacy_stats_collector.cc @@ -97,10 +97,10 @@ struct TypeForAdd { const ValueType& value; }; -typedef TypeForAdd BoolForAdd; -typedef TypeForAdd FloatForAdd; -typedef TypeForAdd Int64ForAdd; -typedef TypeForAdd IntForAdd; +using BoolForAdd = TypeForAdd; +using FloatForAdd = TypeForAdd; +using Int64ForAdd = TypeForAdd; +using IntForAdd = TypeForAdd; StatsReport* AddTrackReport(StatsCollection* reports, const std::string& track_id) { diff --git a/pc/session_description.h b/pc/session_description.h index b60acdd3b0..3c9575f1a7 100644 --- a/pc/session_description.h +++ b/pc/session_description.h @@ -14,21 +14,18 @@ #include #include -#include #include #include -#include #include #include +#include "absl/algorithm/container.h" #include "absl/memory/memory.h" #include "absl/strings/string_view.h" #include "api/media_types.h" #include "api/rtp_parameters.h" #include "api/rtp_transceiver_direction.h" -#include "api/rtp_transceiver_interface.h" #include "media/base/codec.h" -#include "media/base/media_channel.h" #include "media/base/media_constants.h" #include "media/base/rid_description.h" #include "media/base/stream_params.h" @@ -438,7 +435,7 @@ class RTC_EXPORT ContentInfo { std::unique_ptr description_; }; -typedef std::vector ContentNames; +using ContentNames = std::vector; // This class provides a mechanism to aggregate different media contents into a // group. This group can also be shared with the peers in a pre-defined format. @@ -468,8 +465,8 @@ class ContentGroup { ContentNames content_names_; }; -typedef std::vector ContentInfos; -typedef std::vector ContentGroups; +using ContentInfos = std::vector; +using ContentGroups = std::vector; // Determines how the MSID will be signaled in the SDP. // These can be used as bit flags to indicate both or the special value none. From 5edfd244fed9a339f6279cdccb4d57d0c6749b21 Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Sun, 4 May 2025 12:12:31 -0700 Subject: [PATCH 045/292] sdp munging: forbid removing a content with SDP munging This is now disabled by default, guarded by WebRTC-NoSdpMangleNumberOfContents which can be disabled if necessary. BUG=webrtc:40567530 Change-Id: I02a3d4d21678f41e0910144be1aa159828c40757 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/387560 Reviewed-by: Evan Shrubsole Reviewed-by: Harald Alvestrand Commit-Queue: Philipp Hancke Cr-Commit-Position: refs/heads/main@{#44517} --- experiments/field_trials.py | 3 +++ pc/sdp_offer_answer.cc | 19 ++++++++++++++----- pc/sdp_offer_answer_unittest.cc | 25 ++++++++++++++++++++++++- 3 files changed, 41 insertions(+), 6 deletions(-) diff --git a/experiments/field_trials.py b/experiments/field_trials.py index dff2fe0ff4..bf6b833970 100755 --- a/experiments/field_trials.py +++ b/experiments/field_trials.py @@ -122,6 +122,9 @@ def bug_url(self) -> str: FieldTrial('WebRTC-NoSdpMangleUfragRestrictedAddresses', 409713509, date(2025, 10, 11)), + FieldTrial('WebRTC-NoSdpMangleNumberOfContents', + 40567530, + date(2025, 10, 11)), FieldTrial('WebRTC-Pacer-FastRetransmissions', 40235589, date(2024, 4, 1)), diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc index 31808c506d..6f19d1bee6 100644 --- a/pc/sdp_offer_answer.cc +++ b/pc/sdp_offer_answer.cc @@ -2473,11 +2473,20 @@ void SdpOfferAnswerHandler::DoSetLocalDescription( SdpMungingType sdp_munging_type = DetermineSdpMungingType(desc.get(), last_created_desc); - if (!disable_sdp_munging_checks_ && - HasUfragSdpMunging(desc.get(), last_created_desc)) { - has_sdp_munged_ufrag_ = true; - if (pc_->trials().IsEnabled("WebRTC-NoSdpMangleUfrag")) { - RTC_LOG(LS_ERROR) << "Rejecting SDP because of ufrag modification"; + if (!disable_sdp_munging_checks_) { + bool reject_error = false; + if (HasUfragSdpMunging(desc.get(), last_created_desc)) { + has_sdp_munged_ufrag_ = true; + if (pc_->trials().IsEnabled("WebRTC-NoSdpMangleUfrag")) { + RTC_LOG(LS_ERROR) << "Rejecting SDP because of ufrag modification"; + reject_error = true; + } + } else if (sdp_munging_type == kNumberOfContents && + !pc_->trials().IsDisabled( + "WebRTC-NoSdpMangleNumberOfContents")) { + reject_error = true; + } + if (reject_error) { observer->OnSetLocalDescriptionComplete( RTCError(RTCErrorType::INVALID_MODIFICATION, "SDP is modified in a non-acceptable way")); diff --git a/pc/sdp_offer_answer_unittest.cc b/pc/sdp_offer_answer_unittest.cc index c47f086599..3fe6e19799 100644 --- a/pc/sdp_offer_answer_unittest.cc +++ b/pc/sdp_offer_answer_unittest.cc @@ -2089,10 +2089,33 @@ TEST_F(SdpOfferAnswerMungingTest, DtlsRole) { ElementsAre(Pair(SdpMungingType::kDtlsSetup, 1))); } -TEST_F(SdpOfferAnswerMungingTest, RemoveContent) { +TEST_F(SdpOfferAnswerMungingTest, RemoveContentDefault) { auto pc = CreatePeerConnection(); pc->AddAudioTrack("audio_track", {}); + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto name = contents[0].mid(); + EXPECT_TRUE(offer->description()->RemoveContentByName(contents[0].mid())); + std::string sdp; + offer->ToString(&sdp); + auto modified_offer = CreateSessionDescription( + SdpType::kOffer, + absl::StrReplaceAll(sdp, {{"a=group:BUNDLE " + name, "a=group:BUNDLE"}})); + + RTCError error; + EXPECT_FALSE(pc->SetLocalDescription(std::move(modified_offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kNumberOfContents, 1))); +} + +TEST_F(SdpOfferAnswerMungingTest, RemoveContentKillswitch) { + auto pc = CreatePeerConnection(FieldTrials::CreateNoGlobal( + "WebRTC-NoSdpMangleNumberOfContents/Disabled/")); + pc->AddAudioTrack("audio_track", {}); + auto offer = pc->CreateOffer(); auto& contents = offer->description()->contents(); ASSERT_EQ(contents.size(), 1u); From 09cd40fb3aa7afaa5741deba07f5827c4bfe2206 Mon Sep 17 00:00:00 2001 From: webrtc-version-updater Date: Mon, 5 May 2025 21:06:55 -0700 Subject: [PATCH 046/292] Update WebRTC code version (2025-05-06T04:06:28). Bug: None Change-Id: I3f00795f8be10b9dd7fb987590a370faad596e83 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389885 Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Cr-Commit-Position: refs/heads/main@{#44518} --- call/version.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/call/version.cc b/call/version.cc index 98a8ccea46..789b9410c6 100644 --- a/call/version.cc +++ b/call/version.cc @@ -13,7 +13,7 @@ namespace webrtc { // The timestamp is always in UTC. -const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-05T04:09:29"; +const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-06T04:06:28"; void LoadWebRTCVersionInRegister() { // Using volatile to instruct the compiler to not optimize `p` away even From b4a8567cc8966273e1e51ec1ebfa8ec441ab53a5 Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Thu, 1 May 2025 22:10:05 +0000 Subject: [PATCH 047/292] Cleanup depercated function in test helper PeerConfigurer Bug: None Change-Id: I22076877c4d79c3a6f5896d6565e30a05a891fdc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389520 Auto-Submit: Danil Chapovalov Reviewed-by: Artem Titov Commit-Queue: Artem Titov Cr-Commit-Position: refs/heads/main@{#44519} --- api/test/pclf/BUILD.gn | 2 -- api/test/pclf/peer_configurer.h | 8 -------- 2 files changed, 10 deletions(-) diff --git a/api/test/pclf/BUILD.gn b/api/test/pclf/BUILD.gn index fd0a4a10e4..ff8109be57 100644 --- a/api/test/pclf/BUILD.gn +++ b/api/test/pclf/BUILD.gn @@ -81,7 +81,6 @@ rtc_library("peer_configurer") { "../..:frame_generator_api", "../..:ice_transport_interface", "../..:libjingle_peerconnection_api", - "../..:network_emulation_manager_api", "../..:peer_network_dependencies", "../..:scoped_refptr", "../../../p2p:port_allocator", @@ -96,7 +95,6 @@ rtc_library("peer_configurer") { "../../transport:bitrate_settings", "../../transport:network_control", "../../video_codecs:video_codecs_api", - "//third_party/abseil-cpp/absl/base:core_headers", "//third_party/abseil-cpp/absl/strings:string_view", ] } diff --git a/api/test/pclf/peer_configurer.h b/api/test/pclf/peer_configurer.h index 2eb525ed9d..9fab752fe7 100644 --- a/api/test/pclf/peer_configurer.h +++ b/api/test/pclf/peer_configurer.h @@ -13,11 +13,9 @@ #include #include #include -#include #include #include -#include "absl/base/macros.h" #include "absl/strings/string_view.h" #include "api/async_dns_resolver.h" #include "api/audio/audio_mixer.h" @@ -81,12 +79,6 @@ class PeerConfigurer { PeerConfigurer* SetNetEqFactory(std::unique_ptr neteq_factory); PeerConfigurer* SetAudioProcessing( std::unique_ptr audio_processing); - ABSL_DEPRECATE_AND_INLINE() - PeerConfigurer* SetAudioProcessing( - scoped_refptr audio_processing) { - return SetAudioProcessing( - CustomAudioProcessing(std::move(audio_processing))); - } PeerConfigurer* SetAudioMixer(scoped_refptr audio_mixer); // Forces the Peerconnection to use the network thread as the worker thread. From b4f1cf2e12dde3961ba7bf6afefe4981571269ff Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Tue, 6 May 2025 10:34:57 +0000 Subject: [PATCH 048/292] [android] Allow to configure field trials per PeerConnectionFactory MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Provide webrtc::Environment to create AudioProcessing and AudioDeviceModule through java layer Bug: webrtc:369904700, webrtc:413413572 Change-Id: I79454ceb9706fb9c79c995dc7ce8d65924ff84d0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/388620 Reviewed-by: Zoé Lepaul Commit-Queue: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#44520} --- sdk/android/BUILD.gn | 30 +++++++++++ .../org/webrtc/AudioProcessingFactory.java | 11 +++- sdk/android/api/org/webrtc/Environment.java | 53 +++++++++++++++++++ .../api/org/webrtc/PeerConnectionFactory.java | 51 ++++++++++-------- .../org/webrtc/audio/AudioDeviceModule.java | 9 +++- .../org/webrtc/PeerConnectionFactoryTest.java | 27 ++++++++++ sdk/android/src/jni/environment.cc | 40 ++++++++++++++ .../src/jni/pc/peer_connection_factory.cc | 11 ++-- 8 files changed, 206 insertions(+), 26 deletions(-) create mode 100644 sdk/android/api/org/webrtc/Environment.java create mode 100644 sdk/android/src/jni/environment.cc diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn index d9ca1e008b..0d0f973218 100644 --- a/sdk/android/BUILD.gn +++ b/sdk/android/BUILD.gn @@ -117,6 +117,7 @@ if (is_android) { ":base_jni", ":builtin_audio_codecs_jni", ":default_video_codec_factory_jni", + ":environment_jni", ":java_audio_device_module_jni", ":peerconnection_jni", ":video_jni", @@ -182,6 +183,15 @@ if (is_android) { ] } + rtc_android_library("environment_java") { + visibility = [ "*" ] + sources = [ "api/org/webrtc/Environment.java" ] + deps = [ + ":generated_environment_jni_java", + "//third_party/androidx:androidx_annotation_annotation_java", + ] + } + rtc_android_library("audio_api_java") { visibility = [ "*" ] sources = [ @@ -330,6 +340,7 @@ if (is_android) { ":base_java", ":builtin_audio_codecs_java", ":default_video_codec_factory_java", + ":environment_java", ":generated_base_jni_java", ":generated_peerconnection_jni_java", @@ -616,6 +627,18 @@ if (current_os == "linux" || is_android) { ] } + rtc_library("environment_jni") { + sources = [ "src/jni/environment.cc" ] + deps = [ + ":generated_environment_jni", + ":native_api_jni", + "../../api:field_trials", + "../../api/environment", + "../../api/environment:environment_factory", + "//third_party/jni_zero", + ] + } + rtc_library("builtin_audio_codecs_jni") { visibility = [ "*" ] allow_poison = [ "audio_codecs" ] @@ -805,6 +828,7 @@ if (current_os == "linux" || is_android) { "../../api/audio:builtin_audio_processing_builder", "../../api/audio_codecs:audio_codecs_api", "../../api/crypto:options", + "../../api/environment", "../../api/neteq:neteq_api", "../../api/rtc_event_log:rtc_event_log_factory", "../../api/task_queue:default_task_queue_factory", @@ -1319,6 +1343,12 @@ if (current_os == "linux" || is_android) { jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h" } + generate_jni("generated_environment_jni") { + sources = [ "api/org/webrtc/Environment.java" ] + namespace = "webrtc::jni" + jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h" + } + generate_jni("generated_metrics_jni") { sources = [ "api/org/webrtc/Metrics.java" ] namespace = "webrtc::jni" diff --git a/sdk/android/api/org/webrtc/AudioProcessingFactory.java b/sdk/android/api/org/webrtc/AudioProcessingFactory.java index bd8fdb8989..e0aa25b87f 100644 --- a/sdk/android/api/org/webrtc/AudioProcessingFactory.java +++ b/sdk/android/api/org/webrtc/AudioProcessingFactory.java @@ -16,5 +16,14 @@ public interface AudioProcessingFactory { * Dynamically allocates a webrtc::AudioProcessing instance and returns a pointer to it. * The caller takes ownership of the object. */ - public long createNative(); + // TODO: bugs.webrtc.org/369904700 - Remove when implementations switch to + // another variant. + @Deprecated + public default long createNative() { + return 0; + } + + public default long createNative(long webrtcEnvRef) { + return createNative(); + } } diff --git a/sdk/android/api/org/webrtc/Environment.java b/sdk/android/api/org/webrtc/Environment.java new file mode 100644 index 0000000000..88cef4f6da --- /dev/null +++ b/sdk/android/api/org/webrtc/Environment.java @@ -0,0 +1,53 @@ +/* + * Copyright 2025 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.webrtc; + +import androidx.annotation.Nullable; + +/** Wrapper for the webrtc::Environment native class. */ +public final class Environment implements AutoCloseable { + private final long webrtcEnv; + + public static class Builder { + Builder setFieldTrials(String fieldTrials) { + this.fieldTrials = fieldTrials; + return this; + } + + Environment build() { + return new Environment(this.fieldTrials); + } + + private @Nullable String fieldTrials; + } + + public static Builder builder() { + return new Builder(); + } + + /** Returns non-owning non-null native pointer to the webrtc::Environment */ + public long ref() { + return webrtcEnv; + } + + @Override + public void close() { + nativeFree(webrtcEnv); + } + + private Environment(@Nullable String fieldTrials) { + this.webrtcEnv = nativeCreate(fieldTrials); + } + + private static native long nativeCreate(@Nullable String fieldTrials); + + private static native void nativeFree(long webrtcEnv); +} diff --git a/sdk/android/api/org/webrtc/PeerConnectionFactory.java b/sdk/android/api/org/webrtc/PeerConnectionFactory.java index 2b2ea155c1..eac3bb02a0 100644 --- a/sdk/android/api/org/webrtc/PeerConnectionFactory.java +++ b/sdk/android/api/org/webrtc/PeerConnectionFactory.java @@ -15,9 +15,6 @@ import androidx.annotation.Nullable; import java.util.List; import org.webrtc.Logging.Severity; -import org.webrtc.MediaStreamTrack; -import org.webrtc.PeerConnection; -import org.webrtc.RtpCapabilities; import org.webrtc.audio.AudioDeviceModule; import org.webrtc.audio.JavaAudioDeviceModule; @@ -166,6 +163,7 @@ boolean getDisableNetworkMonitor() { public static class Builder { @Nullable private Options options; + private Environment.Builder envBuilder = Environment.builder(); @Nullable private AudioDeviceModule audioDeviceModule; private AudioEncoderFactoryFactory audioEncoderFactoryFactory = new BuiltinAudioEncoderFactoryFactory(); @@ -186,6 +184,11 @@ public Builder setOptions(Options options) { return this; } + public Builder setFieldTrials(String fieldTrials) { + envBuilder.setFieldTrials(fieldTrials); + return this; + } + public Builder setAudioDeviceModule(AudioDeviceModule audioDeviceModule) { this.audioDeviceModule = audioDeviceModule; return this; @@ -261,24 +264,30 @@ public Builder setNetEqFactoryFactory(NetEqFactoryFactory neteqFactoryFactory) { public PeerConnectionFactory createPeerConnectionFactory() { checkInitializeHasBeenCalled(); - if (audioDeviceModule == null) { - audioDeviceModule = JavaAudioDeviceModule.builder(ContextUtils.getApplicationContext()) - .createAudioDeviceModule(); + try (Environment env = envBuilder.build()) { + if (audioDeviceModule == null) { + audioDeviceModule = JavaAudioDeviceModule.builder(ContextUtils.getApplicationContext()) + .createAudioDeviceModule(); + } + return nativeCreatePeerConnectionFactory( + ContextUtils.getApplicationContext(), + options, + env.ref(), + audioDeviceModule.getNative(env.ref()), + audioEncoderFactoryFactory.createNativeAudioEncoderFactory(), + audioDecoderFactoryFactory.createNativeAudioDecoderFactory(), + videoEncoderFactory, + videoDecoderFactory, + audioProcessingFactory == null ? 0 : audioProcessingFactory.createNative(env.ref()), + fecControllerFactoryFactory == null ? 0 : fecControllerFactoryFactory.createNative(), + networkControllerFactoryFactory == null + ? 0 + : networkControllerFactoryFactory.createNativeNetworkControllerFactory(), + networkStatePredictorFactoryFactory == null + ? 0 + : networkStatePredictorFactoryFactory.createNativeNetworkStatePredictorFactory(), + neteqFactoryFactory == null ? 0 : neteqFactoryFactory.createNativeNetEqFactory()); } - return nativeCreatePeerConnectionFactory(ContextUtils.getApplicationContext(), options, - audioDeviceModule.getNativeAudioDeviceModulePointer(), - audioEncoderFactoryFactory.createNativeAudioEncoderFactory(), - audioDecoderFactoryFactory.createNativeAudioDecoderFactory(), videoEncoderFactory, - videoDecoderFactory, - audioProcessingFactory == null ? 0 : audioProcessingFactory.createNative(), - fecControllerFactoryFactory == null ? 0 : fecControllerFactoryFactory.createNative(), - networkControllerFactoryFactory == null - ? 0 - : networkControllerFactoryFactory.createNativeNetworkControllerFactory(), - networkStatePredictorFactoryFactory == null - ? 0 - : networkStatePredictorFactoryFactory.createNativeNetworkStatePredictorFactory(), - neteqFactoryFactory == null ? 0 : neteqFactoryFactory.createNativeNetEqFactory()); } } @@ -604,7 +613,7 @@ private void onSignalingThreadReady() { private static native void nativeStopInternalTracingCapture(); private static native PeerConnectionFactory nativeCreatePeerConnectionFactory(Context context, - Options options, long nativeAudioDeviceModule, long audioEncoderFactory, + Options options, long webrtcEnvRef, long nativeAudioDeviceModule, long audioEncoderFactory, long audioDecoderFactory, VideoEncoderFactory encoderFactory, VideoDecoderFactory decoderFactory, long nativeAudioProcessor, long nativeFecControllerFactory, long nativeNetworkControllerFactory, diff --git a/sdk/android/api/org/webrtc/audio/AudioDeviceModule.java b/sdk/android/api/org/webrtc/audio/AudioDeviceModule.java index 5a0bf5c74d..ba5097d6c8 100644 --- a/sdk/android/api/org/webrtc/audio/AudioDeviceModule.java +++ b/sdk/android/api/org/webrtc/audio/AudioDeviceModule.java @@ -22,7 +22,14 @@ public interface AudioDeviceModule { * Returns a C++ pointer to a webrtc::AudioDeviceModule. Caller does _not_ take ownership and * lifetime is handled through the release() call. */ - long getNativeAudioDeviceModulePointer(); + @Deprecated + default long getNativeAudioDeviceModulePointer() { + return 0; + } + + public default long getNative(long webrtcEnvRef) { + return getNativeAudioDeviceModulePointer(); + } /** * Release resources for this AudioDeviceModule, including native resources. The object should not diff --git a/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionFactoryTest.java b/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionFactoryTest.java index a28b9c25ff..1e36fef760 100644 --- a/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionFactoryTest.java +++ b/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionFactoryTest.java @@ -62,4 +62,31 @@ public void testInitializeWithTracerAndShutdown() { .createInitializationOptions()); PeerConnectionFactory.shutdownInternalTracer(); } + + // Tests that the JNI glue between Java and C++ does not crash when creating a + // PeerConnectionFactory. + @Test + @SmallTest + public void testCreation() throws Exception { + PeerConnectionFactory.initialize( + PeerConnectionFactory.InitializationOptions.builder( + InstrumentationRegistry.getTargetContext()) + .setNativeLibraryName(TestConstants.NATIVE_LIBRARY) + .createInitializationOptions()); + + PeerConnectionFactory factory = PeerConnectionFactory.builder().createPeerConnectionFactory(); + } + + @Test + @SmallTest + public void testCreationWithFieldTrials() throws Exception { + PeerConnectionFactory.initialize( + PeerConnectionFactory.InitializationOptions.builder( + InstrumentationRegistry.getTargetContext()) + .setNativeLibraryName(TestConstants.NATIVE_LIBRARY) + .createInitializationOptions()); + + PeerConnectionFactory factory = + PeerConnectionFactory.builder().setFieldTrials("").createPeerConnectionFactory(); + } } diff --git a/sdk/android/src/jni/environment.cc b/sdk/android/src/jni/environment.cc new file mode 100644 index 0000000000..adf5fbca20 --- /dev/null +++ b/sdk/android/src/jni/environment.cc @@ -0,0 +1,40 @@ +/* + * Copyright 2025 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ +#include "api/environment/environment.h" + +#include + +#include "api/environment/environment_factory.h" +#include "api/field_trials.h" +#include "sdk/android/generated_environment_jni/Environment_jni.h" +#include "sdk/android/native_api/jni/java_types.h" +#include "third_party/jni_zero/jni_zero.h" + +namespace webrtc { +namespace jni { + +void JNI_Environment_Free(JNIEnv* /*env*/, jlong webrtc_env) { + delete reinterpret_cast(webrtc_env); +} + +jlong JNI_Environment_Create( + JNIEnv* jni, + const jni_zero::JavaParamRef& j_field_trials) { + std::unique_ptr field_trials; + if (!j_field_trials.is_null()) { + field_trials = + FieldTrials::CreateNoGlobal(JavaToNativeString(jni, j_field_trials)); + } + return NativeToJavaPointer( + new Environment(CreateEnvironment(std::move(field_trials)))); +} + +} // namespace jni +} // namespace webrtc diff --git a/sdk/android/src/jni/pc/peer_connection_factory.cc b/sdk/android/src/jni/pc/peer_connection_factory.cc index f05259b478..80b92046c4 100644 --- a/sdk/android/src/jni/pc/peer_connection_factory.cc +++ b/sdk/android/src/jni/pc/peer_connection_factory.cc @@ -29,6 +29,7 @@ #include "api/audio_codecs/audio_encoder_factory.h" #include "api/audio_options.h" #include "api/enable_media.h" +#include "api/environment/environment.h" #include "api/fec_controller.h" #include "api/media_stream_interface.h" #include "api/neteq/neteq_factory.h" @@ -256,6 +257,7 @@ ScopedJavaLocalRef CreatePeerConnectionFactoryForJava( JNIEnv* jni, const jni_zero::JavaParamRef& jcontext, const jni_zero::JavaParamRef& joptions, + const Environment& env, scoped_refptr audio_device_module, scoped_refptr audio_encoder_factory, scoped_refptr audio_decoder_factory, @@ -292,8 +294,8 @@ ScopedJavaLocalRef CreatePeerConnectionFactoryForJava( JavaToNativePeerConnectionFactoryOptions(jni, joptions); PeerConnectionFactoryDependencies dependencies; - // TODO(bugs.webrtc.org/13145): Also add socket_server.get() to the - // dependencies. + dependencies.env = env; + dependencies.socket_factory = socket_server.get(); dependencies.network_thread = network_thread.get(); dependencies.worker_thread = worker_thread.get(); dependencies.signaling_thread = signaling_thread.get(); @@ -347,6 +349,7 @@ JNI_PeerConnectionFactory_CreatePeerConnectionFactory( JNIEnv* jni, const jni_zero::JavaParamRef& jcontext, const jni_zero::JavaParamRef& joptions, + jlong webrtc_env_ref, jlong native_audio_device_module, jlong native_audio_encoder_factory, jlong native_audio_decoder_factory, @@ -357,10 +360,12 @@ JNI_PeerConnectionFactory_CreatePeerConnectionFactory( jlong native_network_controller_factory, jlong native_network_state_predictor_factory, jlong native_neteq_factory) { + const Environment* env = reinterpret_cast(webrtc_env_ref); + RTC_CHECK(env != nullptr); scoped_refptr audio_processor( reinterpret_cast(native_audio_processor)); return CreatePeerConnectionFactoryForJava( - jni, jcontext, joptions, + jni, jcontext, joptions, *env, scoped_refptr( reinterpret_cast(native_audio_device_module)), TakeOwnershipOfRefPtr(native_audio_encoder_factory), From d4bdaed57bbce5bac8e7c6d53d8d277865b27332 Mon Sep 17 00:00:00 2001 From: Mirko Bonadei Date: Tue, 6 May 2025 12:25:46 +0000 Subject: [PATCH 049/292] Disable JS protobuf. This prevents these imports [1] in the //third_party/protobuf/BUILD.gn. They are problematic because WebRTC doesn't checkout //ui. While a solution gets prepared, this CL unblocks the roll. [1] = https://source.chromium.org/chromium/chromium/src/+/main:third_party/protobuf/BUILD.gn;l=8-13;drc=a87812bfda7d359447bfa383eeee2820926bd634 Bug: chromium:415076048 Change-Id: Idbaa5a7d9755f6e343d1d1c39d54e7943f1f8bfe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389981 Commit-Queue: Mirko Bonadei Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44521} --- .gn | 3 +++ 1 file changed, 3 insertions(+) diff --git a/.gn b/.gn index e628c3abba..5f2f67fbef 100644 --- a/.gn +++ b/.gn @@ -90,4 +90,7 @@ default_args = { enable_rust = true enable_rust_cxx = true enable_chromium_prelude = true + + # See context on https://issues.chromium.org/issues/415076048. + enable_js_protobuf = false } From f844699005fefc6844bef184bcc101389951caa7 Mon Sep 17 00:00:00 2001 From: Andrew Grieve Date: Mon, 5 May 2025 20:42:18 -0400 Subject: [PATCH 050/292] Fix license name in README.chromium for fft4g Bug: chromium:414606466 Change-Id: I627dd5b01e157125c9811ec36086efd7e16855b1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389920 Reviewed-by: Sam Zackrisson Reviewed-by: Henrik Lundin Commit-Queue: Sam Zackrisson Auto-Submit: Andrew Grieve Cr-Commit-Position: refs/heads/main@{#44522} --- common_audio/third_party/ooura/README.chromium | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/common_audio/third_party/ooura/README.chromium b/common_audio/third_party/ooura/README.chromium index 3b105fe050..f414ac28d7 100644 --- a/common_audio/third_party/ooura/README.chromium +++ b/common_audio/third_party/ooura/README.chromium @@ -3,7 +3,7 @@ Short Name: fft4g URL: http://www.kurims.kyoto-u.ac.jp/~ooura/fft.html Version: N/A Date: 2018-06-19 -License: Custome license +License: LicenseRef-takuya-ooura License File: LICENSE Security Critical: yes Shipped: yes From ce7668b1e7e43150412067e642f5f2ce09627f6c Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Tue, 6 May 2025 07:04:16 -0700 Subject: [PATCH 051/292] Roll chromium_revision fb0dacbdf6..b377c950c8 (1454341:1456285) Change log: https://chromium.googlesource.com/chromium/src/+log/fb0dacbdf6..b377c950c8 Full diff: https://chromium.googlesource.com/chromium/src/+/fb0dacbdf6..b377c950c8 Changed dependencies * fuchsia_version: version:27.20250424.2.1..version:27.20250501.3.1 * src/base: https://chromium.googlesource.com/chromium/src/base/+log/802e7fa5a7..135ccf38f8 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/d2aa4eb134..2021df387e * src/buildtools: https://chromium.googlesource.com/chromium/src/buildtools/+log/6d4e129ac6..b841ab22bc * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/6e92444555..00e20b50b2 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/c96680035b..c2ffcbf942 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/60b1cfd539..f2a66cfb2b * src/third_party/android_build_tools/lint/cipd: TEDRB0F7VgCgyepd8FsIxPRqvKjvWQ_YkLV-ybx-GQYC..0xqmMJ1qdP4NPN5s9VmWiNhchvNz_RLBrjXN4jUKN3gC * src/third_party/android_build_tools/manifest_merger/cipd: fPGWiT0LVG2_ifxOpHb5KNwTL-_7lb7tYOsB_ZoIiskC..7lstFdOjh1bFmfSZ8le-aZ1a1COQtwirGzuEWa43tnUC * src/third_party/android_build_tools/nullaway/cipd: F6la8NsEkr27_Sm2MswVDBDB7UXVqRtaCMS5YWEQ9dQC..mFkUqdlIqmMQ3rAaTHFmOBYjNFgFUQOmMlPG6PGgmHYC * src/third_party/android_deps/autorolled/cipd: I4KwXm_DsdqDHlwWP2Ym6Qecv-6fM6c1Kt5u277RCoAC..nb19wuIgamdmcgMY_6xHCGH3ph_tLVHkHYH33tKbup4C * src/third_party/androidx/cipd: DGv8hAr3K2Qyh0okP6syAJbvPRKjP0EMD6DLlEl-NK4C..k1zV9FUHrRghPTZBejg81vzwei2vVTdjpM7GhDEXyxQC * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/ab301d01c2..0f1d0df618 * src/third_party/breakpad/breakpad: https://chromium.googlesource.com/breakpad/breakpad.git/+log/232a723f50..b319c12317 * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/1591e813b6..417522f618 * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/354f6026f1..30d9438e7b * src/third_party/ffmpeg: https://chromium.googlesource.com/chromium/third_party/ffmpeg.git/+log/01f23648c6..dcdd0fa51b * src/third_party/googletest/src: https://chromium.googlesource.com/external/github.com/google/googletest.git/+log/04ee1b4f2a..90a4152114 * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/4d2eae608a..9e195f61ff * src/third_party/icu: https://chromium.googlesource.com/chromium/deps/icu.git/+log/4c8cc4b365..b929596bae * src/third_party/kotlinc/current: YASKBvUSO-m2borC3V_hGLsRTyOSnHl2J0BaCmINZkUC..DMzAXJeHsmdOETLSEAYTfnPUX4VvhvmX2__q71V_Ia8C * src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/719f60edc5..378f15d1cb * src/third_party/libc++/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git/+log/f034fd5662..a01c02c9d4 * src/third_party/libc++abi/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git/+log/f2a7f2987f..7ff13dd8c6 * src/third_party/libvpx/source/libvpx: https://chromium.googlesource.com/webm/libvpx.git/+log/b32e98f52f..6bb2886336 * src/third_party/libyuv: https://chromium.googlesource.com/libyuv/libyuv.git/+log/9f9b5cf660..4db2af62da * src/third_party/llvm-build/Release+Asserts: Linux_x64/clang-llvmorg-21-init-9266-g09006611-2.tar.xz,Linux_x64/clang-tidy-llvmorg-21-init-9266-g09006611-2.tar.xz,Linux_x64/clangd-llvmorg-21-init-9266-g09006611-2.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-2.tar.xz,Linux_x64/llvmobjdump-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac/clang-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac/clang-mac-runtime-library-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac/clang-tidy-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac/clangd-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac/llvm-code-coverage-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac_arm64/clang-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac_arm64/clang-tidy-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac_arm64/clangd-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-2.tar.xz,Win/clang-llvmorg-21-init-9266-g09006611-2.tar.xz,Win/clang-tidy-llvmorg-21-init-9266-g09006611-2.tar.xz,Win/clang-win-runtime-library-llvmorg-21-init-9266-g09006611-2.tar.xz,Win/clangd-llvmorg-21-init-9266-g09006611-2.tar.xz,Win/llvm-code-coverage-llvmorg-21-init-9266-g09006611-2.tar.xz,Win/llvmobjdump-llvmorg-21-init-9266-g09006611-2.tar.xz..Linux_x64/clang-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Linux_x64/clang-tidy-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Linux_x64/clangd-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Mac/clang-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Mac/clang-tidy-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Mac/clangd-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Mac/llvm-code-coverage-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Mac_arm64/clang-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Mac_arm64/clangd-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Win/clang-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Win/clang-tidy-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Win/clang-win-runtime-library-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Win/clangd-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Win/llvm-code-coverage-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Win/llvmobjdump-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz * src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/fa2469dd9d..c8248a038f * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/ed0bcb0f5a..fc9aac5739 * src/third_party/rust-toolchain: Linux_x64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Mac/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Mac_arm64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Win/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz..Linux_x64/rust-toolchain-3350c1eb3fd8fe1bee1ed4c76944d707bd256876-1-llvmorg-21-init-10502-gb2e2ae87.tar.xz,Mac/rust-toolchain-3350c1eb3fd8fe1bee1ed4c76944d707bd256876-1-llvmorg-21-init-10502-gb2e2ae87.tar.xz,Mac_arm64/rust-toolchain-3350c1eb3fd8fe1bee1ed4c76944d707bd256876-1-llvmorg-21-init-10502-gb2e2ae87.tar.xz,Win/rust-toolchain-3350c1eb3fd8fe1bee1ed4c76944d707bd256876-1-llvmorg-21-init-10502-gb2e2ae87.tar.xz * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/b2ca46af79..725c5898d1 DEPS diff: https://chromium.googlesource.com/chromium/src/+/fb0dacbdf6..b377c950c8/DEPS Clang version changed llvmorg-21-init-9266-g09006611:llvmorg-21-init-10502-gb2e2ae87 Details: https://chromium.googlesource.com/chromium/src/+/fb0dacbdf6..b377c950c8/tools/clang/scripts/update.py BUG=None Change-Id: I69ce200df5edf8bcbbdc0ae6cd74aba026852bf3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390040 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44523} --- DEPS | 252 +++++++++++++++++++++++++++++------------------------------ 1 file changed, 126 insertions(+), 126 deletions(-) diff --git a/DEPS b/DEPS index dd217306ea..0cc5f71e49 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': 'fb0dacbdf6d4b50101b1888a7c31b19f0c78dac3', + 'chromium_revision': 'b377c950c8162b7bdce9a93e9265ff3229061383', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -36,7 +36,7 @@ vars = { # By default, download the fuchsia sdk from the public sdk directory. 'fuchsia_sdk_cipd_prefix': 'fuchsia/sdk/core/', - 'fuchsia_version': 'version:27.20250424.2.1', + 'fuchsia_version': 'version:27.20250501.3.1', # By default, download the fuchsia images from the fuchsia GCS bucket. 'fuchsia_images_bucket': 'fuchsia', 'checkout_fuchsia': False, @@ -73,24 +73,24 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@802e7fa5a71086c4b520ee4ebb1030f5425643bd', + 'https://chromium.googlesource.com/chromium/src/base@135ccf38f87dfaf4726c3e6cf5f759d1c67bff96', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@d2aa4eb1342a48627e3ea3a174915f34a73bf6e6', + 'https://chromium.googlesource.com/chromium/src/build@2021df387efef46ca51da8e3356402a706617368', 'src/buildtools': - 'https://chromium.googlesource.com/chromium/src/buildtools@6d4e129ac68b4c81d98765068ceb53dbf09e99d2', + 'https://chromium.googlesource.com/chromium/src/buildtools@b841ab22bc4522fe4e72df46ca57abbe2fad8640', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. 'src/examples/androidtests/third_party/gradle': { 'url': 'https://chromium.googlesource.com/external/github.com/gradle/gradle.git@f2d1fb54a951d8b11d25748e4711bec8d128d7e3', 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@6e924445554ed56e7c2137a2808f07c065600fe9', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@00e20b50b20c5c33e6013c112dffb1bee277523c', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@c96680035b6e74d7f6414510187c6f27055b0838', + 'https://chromium.googlesource.com/chromium/src/testing@c2ffcbf94259f1b0cccdfc674a2a100ddd068f0c', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@60b1cfd5394d5b50d835cffadbdae4a88a3de5e5', + 'https://chromium.googlesource.com/chromium/src/third_party@f2a66cfb2b7dc8b68cf55c6cc2cce12b6fb77bc2', 'src/buildtools/linux64': { 'packages': [ @@ -142,143 +142,143 @@ deps = { 'objects': [ { # The Android libclang_rt.builtins libraries are currently only included in the Linux clang package. - 'object_name': 'Linux_x64/clang-llvmorg-21-init-9266-g09006611-2.tar.xz', - 'sha256sum': 'dee83e0a8b35564c277a7902a7872a77f4170b4380f9053acd3e1cb9a502a69c', - 'size_bytes': 55555048, - 'generation': 1746105739973602, + 'object_name': 'Linux_x64/clang-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz', + 'sha256sum': '557ae6a59be1eb465d0e726766b757132f0ff5f1a13bda0a871887421a59497d', + 'size_bytes': 55589384, + 'generation': 1746143871500001, 'condition': '(host_os == "linux" or checkout_android) and non_git_source', }, { - 'object_name': 'Linux_x64/clang-tidy-llvmorg-21-init-9266-g09006611-2.tar.xz', - 'sha256sum': 'bb8684fb985a35cc680bda9161735c409e6409106497f4ec0253ff9e07dd6912', - 'size_bytes': 13555996, - 'generation': 1746105739988254, + 'object_name': 'Linux_x64/clang-tidy-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz', + 'sha256sum': 'd940c80b7c1358e99759b90fd3ec3bee4099ee5bdc7858132dfb9474504b5baf', + 'size_bytes': 13553540, + 'generation': 1746143871510594, 'condition': 'host_os == "linux" and checkout_clang_tidy and non_git_source', }, { - 'object_name': 'Linux_x64/clangd-llvmorg-21-init-9266-g09006611-2.tar.xz', - 'sha256sum': '5074d634984fa526b868126ead41c013de337edf36b369422005cb4a0edb95dc', - 'size_bytes': 13762680, - 'generation': 1746105740081588, + 'object_name': 'Linux_x64/clangd-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz', + 'sha256sum': 'b138ec884298dd0601232dd98f92d2697e64eefe2ac5b4eee383eefa8909ed15', + 'size_bytes': 13783716, + 'generation': 1746143871533160, 'condition': 'host_os == "linux" and checkout_clangd and non_git_source', }, { - 'object_name': 'Linux_x64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-2.tar.xz', - 'sha256sum': 'fa46a9d107f515d0a16c41ae5751a2a4be7be34a550f315a1506fcfb17327936', - 'size_bytes': 2294512, - 'generation': 1746105740273331, + 'object_name': 'Linux_x64/llvm-code-coverage-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz', + 'sha256sum': '98c252d7459a0645fb2daacb95b996fab9325072b29f5ef752e5864d7cea44d2', + 'size_bytes': 2292976, + 'generation': 1746143871601054, 'condition': 'host_os == "linux" and checkout_clang_coverage_tools and non_git_source', }, { - 'object_name': 'Linux_x64/llvmobjdump-llvmorg-21-init-9266-g09006611-2.tar.xz', - 'sha256sum': '4041ca56834653e78338f50aa6590c8160bc6b522bdcddbd533027d9116e4994', - 'size_bytes': 5694452, - 'generation': 1746105740224915, + 'object_name': 'Linux_x64/llvmobjdump-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz', + 'sha256sum': '0bf6b769345d7234e16ce57e839732d3be94ab1043a55d0aae2cb3f20399f7bd', + 'size_bytes': 5697144, + 'generation': 1746143871548416, 'condition': '((checkout_linux or checkout_mac or checkout_android) and host_os == "linux") and non_git_source', }, { - 'object_name': 'Mac/clang-llvmorg-21-init-9266-g09006611-2.tar.xz', - 'sha256sum': '5ee059eddd55064d3e638b83db4c7843f5448bb74c62c474b5f5652a5631cb85', - 'size_bytes': 51962684, - 'generation': 1746105741728605, + 'object_name': 'Mac/clang-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz', + 'sha256sum': '29482bbf5baff85fe278d12a7456d517e10497a888f3082b49043a1f73c0def4', + 'size_bytes': 52061888, + 'generation': 1746143873509655, 'condition': 'host_os == "mac" and host_cpu == "x64"', }, { - 'object_name': 'Mac/clang-mac-runtime-library-llvmorg-21-init-9266-g09006611-2.tar.xz', - 'sha256sum': '8482f9c688466057de438a15ff01d62ea1f0e2b709f3e7a470cdd452c81da731', - 'size_bytes': 989704, - 'generation': 1746105749642530, + 'object_name': 'Mac/clang-mac-runtime-library-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz', + 'sha256sum': '167dcfa307f1a6e0d5ac26701be84b6a0acc2dc9ebf48d2601c37f89579fac90', + 'size_bytes': 992296, + 'generation': 1746143880809968, 'condition': 'checkout_mac and not host_os == "mac"', }, { - 'object_name': 'Mac/clang-tidy-llvmorg-21-init-9266-g09006611-2.tar.xz', - 'sha256sum': 'e23d1a41409e2637f662cb9306d3269ea1c4f41ae1aa70f14369b6463de95c03', - 'size_bytes': 13610744, - 'generation': 1746105741733748, + 'object_name': 'Mac/clang-tidy-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz', + 'sha256sum': '2503c1f5c1d304aaf30da2ccc6ffb0834a409429d5754780f1b80afbffc9bd5d', + 'size_bytes': 13636716, + 'generation': 1746143873535744, 'condition': 'host_os == "mac" and host_cpu == "x64" and checkout_clang_tidy', }, { - 'object_name': 'Mac/clangd-llvmorg-21-init-9266-g09006611-2.tar.xz', - 'sha256sum': '6921b4b53cdf8bc80fcad184e229c4f38a49463390016eed65f12fa547f24950', - 'size_bytes': 14998636, - 'generation': 1746105741782145, + 'object_name': 'Mac/clangd-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz', + 'sha256sum': '04a7e47676d58e10fc5ca28f87035a452c5b6d959c5f965a2d7efae7d72d2424', + 'size_bytes': 15070916, + 'generation': 1746143873533731, 'condition': 'host_os == "mac" and host_cpu == "x64" and checkout_clangd', }, { - 'object_name': 'Mac/llvm-code-coverage-llvmorg-21-init-9266-g09006611-2.tar.xz', - 'sha256sum': '0b41a82139b4a107f1a2a149d438b600685cbab5c0f81c8d0459f2711503460b', - 'size_bytes': 2263096, - 'generation': 1746105741945685, + 'object_name': 'Mac/llvm-code-coverage-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz', + 'sha256sum': '97680b6f03a575050b941ead3ee816bf697b159c0b3304d29af7d0fee6cbfdd3', + 'size_bytes': 2266332, + 'generation': 1746143873609231, 'condition': 'host_os == "mac" and host_cpu == "x64" and checkout_clang_coverage_tools', }, { - 'object_name': 'Mac_arm64/clang-llvmorg-21-init-9266-g09006611-2.tar.xz', - 'sha256sum': '99c31f2e700274ade6b25205597510a5c665f17dd9ddaa03b89c9659ec387fd4', - 'size_bytes': 43999184, - 'generation': 1746105751053626, + 'object_name': 'Mac_arm64/clang-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz', + 'sha256sum': '8ab032725c51a542f7e54b8795952039b0b910f3f5315bca13ae669febaa425f', + 'size_bytes': 44082720, + 'generation': 1746143882532819, 'condition': 'host_os == "mac" and host_cpu == "arm64"', }, { - 'object_name': 'Mac_arm64/clang-tidy-llvmorg-21-init-9266-g09006611-2.tar.xz', - 'sha256sum': '78280eb8dceeaa330fb53ec8249c740db0f369f3b6d6857c65337dc5a21b3160', - 'size_bytes': 11773472, - 'generation': 1746105751166820, + 'object_name': 'Mac_arm64/clang-tidy-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz', + 'sha256sum': '35679319e006d4fdc27aa5e62e1637db33026ef06c2cfa36abe14aa4113f34ab', + 'size_bytes': 11794244, + 'generation': 1746143882559932, 'condition': 'host_os == "mac" and host_cpu == "arm64" and checkout_clang_tidy', }, { - 'object_name': 'Mac_arm64/clangd-llvmorg-21-init-9266-g09006611-2.tar.xz', - 'sha256sum': 'f2de67c31e6ce01485b4cc0a76567809319021c4523609b93ab1804252f95ddc', - 'size_bytes': 12044048, - 'generation': 1746105751239361, + 'object_name': 'Mac_arm64/clangd-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz', + 'sha256sum': '6ecfc7eee127d298d55757d405bc29d3197d60b8316d51355eca218442c55b90', + 'size_bytes': 12067876, + 'generation': 1746143882556967, 'condition': 'host_os == "mac" and host_cpu == "arm64" and checkout_clangd', }, { - 'object_name': 'Mac_arm64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-2.tar.xz', - 'sha256sum': 'a2c44380d13276f1c3393abff82e8d397c893a912b157bea44b0006bd35fe618', - 'size_bytes': 1974668, - 'generation': 1746105751515503, + 'object_name': 'Mac_arm64/llvm-code-coverage-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz', + 'sha256sum': '09f89d979a09cad96b4d81090bec5cc3e7c12c7b5431ab21842d47b67b93122a', + 'size_bytes': 1975672, + 'generation': 1746143882686932, 'condition': 'host_os == "mac" and host_cpu == "arm64" and checkout_clang_coverage_tools', }, { - 'object_name': 'Win/clang-llvmorg-21-init-9266-g09006611-2.tar.xz', - 'sha256sum': '9362a0993354be586748f4148493e2b9cc63a898aa85fea301f2de0e60ecbd93', - 'size_bytes': 47049060, - 'generation': 1746105761436952, + 'object_name': 'Win/clang-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz', + 'sha256sum': 'e9b3c67428ee5fff61ab2e229da2ecd807cceb5e48983baa2aa31ae675059931', + 'size_bytes': 47070352, + 'generation': 1746143892525539, 'condition': 'host_os == "win"', }, { - 'object_name': 'Win/clang-tidy-llvmorg-21-init-9266-g09006611-2.tar.xz', - 'sha256sum': 'c55a73aa688dfa2e6195b1dc233079504f1ed9a95c8e8a92582add133bb8a240', - 'size_bytes': 13414380, - 'generation': 1746105761440911, + 'object_name': 'Win/clang-tidy-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz', + 'sha256sum': 'b7ff1c51e8f91ffe5bab9db99529f802114ce62ec588d0953d3a548424520836', + 'size_bytes': 13437572, + 'generation': 1746143892543580, 'condition': 'host_os == "win" and checkout_clang_tidy', }, { - 'object_name': 'Win/clang-win-runtime-library-llvmorg-21-init-9266-g09006611-2.tar.xz', - 'sha256sum': 'c42cfa237dbd6841d61013051de739775fe0ae3187ff5fecc7ed243de2941da6', - 'size_bytes': 2484424, - 'generation': 1746105769629959, + 'object_name': 'Win/clang-win-runtime-library-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz', + 'sha256sum': '1ed7bf852f8255bf35335c14110c959743e17903e0020c776be3b8d80a564ae9', + 'size_bytes': 2484344, + 'generation': 1746143899795872, 'condition': 'checkout_win and not host_os == "win"', }, { - 'object_name': 'Win/clangd-llvmorg-21-init-9266-g09006611-2.tar.xz', - 'sha256sum': 'fcf5bfeccbca8a2445d579093c9d7da87dff142b223f24f25d87f6e2e6e15c50', - 'size_bytes': 13840260, - 'generation': 1746105761501287, + 'object_name': 'Win/clangd-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz', + 'sha256sum': '90f9035116a665e8a620b259d299c845be42b68c7dce8a6439ba189d550bd6ca', + 'size_bytes': 13855648, + 'generation': 1746143892528921, 'condition': 'host_os == "win" and checkout_clangd', }, { - 'object_name': 'Win/llvm-code-coverage-llvmorg-21-init-9266-g09006611-2.tar.xz', - 'sha256sum': '1a5e31c5e69ea7b84fdfbff27024670cf2eec538c68fc4213a5bcf576263b0ca', - 'size_bytes': 2370540, - 'generation': 1746105761636379, + 'object_name': 'Win/llvm-code-coverage-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz', + 'sha256sum': '100dc0773b81d1a65b40356038d2c85e5aabd613b2e5830846d60c9fa2ca04bf', + 'size_bytes': 2374764, + 'generation': 1746143892621367, 'condition': 'host_os == "win" and checkout_clang_coverage_tools', }, { - 'object_name': 'Win/llvmobjdump-llvmorg-21-init-9266-g09006611-2.tar.xz', - 'sha256sum': '3ac42a8e816d1fa89594bfe882d033b6c351551b055b9d78867a2b3d24ac2451', - 'size_bytes': 5698696, - 'generation': 1746105761558030, + 'object_name': 'Win/llvmobjdump-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz', + 'sha256sum': '104ef589d1d1d9a7f1acd38e25c418b12de8b685c4f55f45f626cc64f3f1cc98', + 'size_bytes': 5685804, + 'generation': 1746143892549487, 'condition': '(checkout_linux or checkout_mac or checkout_android) and host_os == "win"', }, ] @@ -290,31 +290,31 @@ deps = { 'bucket': 'chromium-browser-clang', 'objects': [ { - 'object_name': 'Linux_x64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz', - 'sha256sum': '378c432f7739bb5da11aad7b3a2687f8252565eae5f0dcfc55c39a15382c519c', - 'size_bytes': 118598336, - 'generation': 1745271335898717, + 'object_name': 'Linux_x64/rust-toolchain-3350c1eb3fd8fe1bee1ed4c76944d707bd256876-1-llvmorg-21-init-10502-gb2e2ae87.tar.xz', + 'sha256sum': 'b7b13f2fbd22c5415abec739570acd74066361ce3c82313660be65ab67f1a075', + 'size_bytes': 118912744, + 'generation': 1746143864066683, 'condition': 'host_os == "linux" and non_git_source', }, { - 'object_name': 'Mac/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz', - 'sha256sum': 'bf05c8b5e90d6904de02dca9b3e4cb5e45a1a56207e7af1fbb3a10707704a26a', - 'size_bytes': 111932536, - 'generation': 1745271337336068, + 'object_name': 'Mac/rust-toolchain-3350c1eb3fd8fe1bee1ed4c76944d707bd256876-1-llvmorg-21-init-10502-gb2e2ae87.tar.xz', + 'sha256sum': '322dc36f74871d746545a1bb75470968315bce5ed4b385f43033e376f4c12a9b', + 'size_bytes': 112193460, + 'generation': 1746143865768180, 'condition': 'host_os == "mac" and host_cpu == "x64"', }, { - 'object_name': 'Mac_arm64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz', - 'sha256sum': '1aec99f479ff28cefe44ed739844833e016a1da255cf3c17d79e59a273246615', - 'size_bytes': 101605468, - 'generation': 1745271339727037, + 'object_name': 'Mac_arm64/rust-toolchain-3350c1eb3fd8fe1bee1ed4c76944d707bd256876-1-llvmorg-21-init-10502-gb2e2ae87.tar.xz', + 'sha256sum': '2a95ec6fe7623c43e87f06829bae17c45f1c94c19db4c8f085d77cb948c0ee3f', + 'size_bytes': 101896012, + 'generation': 1746143867490688, 'condition': 'host_os == "mac" and host_cpu == "arm64"', }, { - 'object_name': 'Win/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz', - 'sha256sum': 'b291520613a3ebc415e4576a7fa31d840a5ebf4ab9be6e9dc5d90062dc001c1e', - 'size_bytes': 193280372, - 'generation': 1745271341223097, + 'object_name': 'Win/rust-toolchain-3350c1eb3fd8fe1bee1ed4c76944d707bd256876-1-llvmorg-21-init-10502-gb2e2ae87.tar.xz', + 'sha256sum': 'da814703ae38af456eeeb63985198cb7ec48740b0a143249830fc880d633f80b', + 'size_bytes': 193736940, + 'generation': 1746143869370201, 'condition': 'host_os == "win"', }, ], @@ -325,11 +325,11 @@ deps = { 'src/third_party/compiler-rt/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@6558e6a3eacafd269527cf2595e0e722f117c1e6', 'src/third_party/libc++/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@f034fd5662d033c281ab6a9b45164066ddd18809', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@a01c02c9d4acbdae3b7e8a2f3ee58579a9c29f96', 'src/third_party/libc++abi/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@f2a7f2987f9dcdf8b04c2d8cd4dcb186641a7c3e', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@7ff13dd8c66d5047a3ee1e1ba26ddeb70d3d8740', 'src/third_party/llvm-libc/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@fa2469dd9d05382f04e927456ca3c482c3d705e9', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@c8248a038fe70b7495d1d52fde500cf2dd6fae82', 'src/third_party/libunwind/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@81e2cb40a70de2b6978e6d8658891ded9a77f7e3', @@ -449,7 +449,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_build_tools/lint', - 'version': 'TEDRB0F7VgCgyepd8FsIxPRqvKjvWQ_YkLV-ybx-GQYC', + 'version': '0xqmMJ1qdP4NPN5s9VmWiNhchvNz_RLBrjXN4jUKN3gC', }, ], 'condition': 'checkout_android and non_git_source', @@ -461,7 +461,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_build_tools/nullaway', - 'version': 'F6la8NsEkr27_Sm2MswVDBDB7UXVqRtaCMS5YWEQ9dQC', + 'version': 'mFkUqdlIqmMQ3rAaTHFmOBYjNFgFUQOmMlPG6PGgmHYC', }, ], 'condition': 'checkout_android and non_git_source', @@ -480,11 +480,11 @@ deps = { }, 'src/third_party/boringssl/src': - 'https://boringssl.googlesource.com/boringssl.git@ab301d01c2bf15da3716500e518425c085c25503', + 'https://boringssl.googlesource.com/boringssl.git@0f1d0df6183d6ddf0b4d7a10bf80122c7ec260e6', 'src/third_party/breakpad/breakpad': - 'https://chromium.googlesource.com/breakpad/breakpad.git@232a723f5096ab02d53d87931efa485fa77d3b03', + 'https://chromium.googlesource.com/breakpad/breakpad.git@b319c12317824bfd0e92828db20bbfe0ec6e31f8', 'src/third_party/catapult': - 'https://chromium.googlesource.com/catapult.git@1591e813b66e04a5b9d9e8fc5d94e74ef4effaa1', + 'https://chromium.googlesource.com/catapult.git@417522f6185fde0e4e0812bce00bca1277f286a5', 'src/third_party/ced/src': { 'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5', }, @@ -497,13 +497,13 @@ deps = { 'src/third_party/crc32c/src': 'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558', 'src/third_party/depot_tools': - 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@354f6026f14f87466040e3e1878c599138a23bce', + 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@30d9438e7baf0c045f973de2f517b51554d14d7d', 'src/third_party/ffmpeg': - 'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@01f23648c6b84de6c0f717fa4e1816f53b9ee72e', + 'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@dcdd0fa51b65a0b1688ff6b8f0cc81908f09ded2', 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@4d2eae608a8d03e233e10f800b95ea28c507bcf4', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@9e195f61ff47384d4014fb7dcf9808f15e3ec711', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -525,9 +525,9 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/googletest/src': - 'https://chromium.googlesource.com/external/github.com/google/googletest.git@04ee1b4f2aefdffb0135d7cf2a2c519fe50dabe4', + 'https://chromium.googlesource.com/external/github.com/google/googletest.git@90a41521142c978131f38c6da07b4eb96a9f1ff6', 'src/third_party/icu': { - 'url': 'https://chromium.googlesource.com/chromium/deps/icu.git@4c8cc4b365a505ce35be1e0bd488476c5f79805d', + 'url': 'https://chromium.googlesource.com/chromium/deps/icu.git@b929596baebf0ab4ac7ec07f38365db4c50a559d', }, 'src/third_party/jdk/current': { 'packages': [ @@ -574,7 +574,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/kotlinc', - 'version': 'YASKBvUSO-m2borC3V_hGLsRTyOSnHl2J0BaCmINZkUC', + 'version': 'DMzAXJeHsmdOETLSEAYTfnPUX4VvhvmX2__q71V_Ia8C', }, ], 'condition': 'checkout_android', @@ -591,19 +591,19 @@ deps = { 'src/third_party/dav1d/libdav1d': 'https://chromium.googlesource.com/external/github.com/videolan/dav1d.git@8d956180934f16244bdb58b39175824775125e55', 'src/third_party/libaom/source/libaom': - 'https://aomedia.googlesource.com/aom.git@719f60edc51b6141a2434bf1b5110c2fb075b246', + 'https://aomedia.googlesource.com/aom.git@378f15d1cbd5102f89a52881bcdcf2c010f3c5c8', 'src/third_party/libunwindstack': { 'url': 'https://chromium.googlesource.com/chromium/src/third_party/libunwindstack.git@0d758dd57f42564acecdd7a1e7ac5c8521c1b01a', 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'ed0bcb0f5a956cb4aad00d150851f07496cb23b0', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'fc9aac573945f2da6d05bf1904d500e6d5cdf18d', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': - 'https://chromium.googlesource.com/webm/libvpx.git@b32e98f52fb99e00bc99ea8f2b57a8015b71aa39', + 'https://chromium.googlesource.com/webm/libvpx.git@6bb288633613c613c87d23b32e5f1d23102b1a43', 'src/third_party/libyuv': - 'https://chromium.googlesource.com/libyuv/libyuv.git@9f9b5cf660dcfa0d3fdee41cf4ffbe4bb6e95114', + 'https://chromium.googlesource.com/libyuv/libyuv.git@4db2af62dab48895226be6b52737247e898ebe36', 'src/third_party/lss': { 'url': 'https://chromium.googlesource.com/linux-syscall-support.git@ed31caa60f20a4f6569883b2d752ef7522de51e0', 'condition': 'checkout_android or checkout_linux', @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@b2ca46af795d7ae3277c3dba9afde61cf5090f4e', + 'https://chromium.googlesource.com/chromium/src/tools@725c5898d1c0775d1468c60252df9709a48fd1c8', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'DGv8hAr3K2Qyh0okP6syAJbvPRKjP0EMD6DLlEl-NK4C', + 'version': 'k1zV9FUHrRghPTZBejg81vzwei2vVTdjpM7GhDEXyxQC', }, ], 'condition': 'checkout_android and non_git_source', @@ -706,7 +706,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_build_tools/manifest_merger', - 'version': 'fPGWiT0LVG2_ifxOpHb5KNwTL-_7lb7tYOsB_ZoIiskC', + 'version': '7lstFdOjh1bFmfSZ8le-aZ1a1COQtwirGzuEWa43tnUC', }, ], 'condition': 'checkout_android', @@ -827,7 +827,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_deps/autorolled', - 'version': 'I4KwXm_DsdqDHlwWP2Ym6Qecv-6fM6c1Kt5u277RCoAC', + 'version': 'nb19wuIgamdmcgMY_6xHCGH3ph_tLVHkHYH33tKbup4C', }, ], 'condition': 'checkout_android and non_git_source', From 252193ca6e197375432f90b3b3b9b37545d4f16f Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Mon, 5 May 2025 14:43:50 -0700 Subject: [PATCH 052/292] Move SDP munging detector unit tests to separate file Bug: chromium:40567530 Change-Id: I880b419d64c138df6faf4c60362feb54b97fd22f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389420 Commit-Queue: Philipp Hancke Reviewed-by: Harald Alvestrand Reviewed-by: Evan Shrubsole Cr-Commit-Position: refs/heads/main@{#44524} --- pc/BUILD.gn | 1 + pc/sdp_munging_detector_unittest.cc | 1168 +++++++++++++++++++++++++++ pc/sdp_offer_answer_unittest.cc | 1043 +----------------------- 3 files changed, 1172 insertions(+), 1040 deletions(-) create mode 100644 pc/sdp_munging_detector_unittest.cc diff --git a/pc/BUILD.gn b/pc/BUILD.gn index 00bba5ec57..14e96b7541 100644 --- a/pc/BUILD.gn +++ b/pc/BUILD.gn @@ -2375,6 +2375,7 @@ if (rtc_include_tests && !build_with_chromium) { "rtp_sender_receiver_unittest.cc", "rtp_transceiver_unittest.cc", "sctp_utils_unittest.cc", + "sdp_munging_detector_unittest.cc", "sdp_offer_answer_unittest.cc", "simulcast_sdp_serializer_unittest.cc", "test/fake_audio_capture_module_unittest.cc", diff --git a/pc/sdp_munging_detector_unittest.cc b/pc/sdp_munging_detector_unittest.cc new file mode 100644 index 0000000000..a06735b6f8 --- /dev/null +++ b/pc/sdp_munging_detector_unittest.cc @@ -0,0 +1,1168 @@ +/* + * Copyright 2025 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include "absl/strings/str_cat.h" +#include "absl/strings/str_replace.h" +#include "api/audio_codecs/audio_format.h" +#include "api/audio_codecs/builtin_audio_decoder_factory.h" +#include "api/audio_codecs/builtin_audio_encoder_factory.h" +#include "api/create_peerconnection_factory.h" +#include "api/field_trials.h" +#include "api/field_trials_view.h" +#include "api/jsep.h" +#include "api/media_types.h" +#include "api/peer_connection_interface.h" +#include "api/rtc_error.h" +#include "api/rtp_parameters.h" +#include "api/rtp_transceiver_direction.h" +#include "api/scoped_refptr.h" +#include "api/test/rtc_error_matchers.h" +#include "api/uma_metrics.h" +#include "api/video_codecs/sdp_video_format.h" +#include "api/video_codecs/video_decoder_factory_template.h" +#include "api/video_codecs/video_decoder_factory_template_dav1d_adapter.h" +#include "api/video_codecs/video_decoder_factory_template_libvpx_vp8_adapter.h" +#include "api/video_codecs/video_decoder_factory_template_libvpx_vp9_adapter.h" +#include "api/video_codecs/video_decoder_factory_template_open_h264_adapter.h" +#include "api/video_codecs/video_encoder_factory_template.h" +#include "api/video_codecs/video_encoder_factory_template_libaom_av1_adapter.h" +#include "api/video_codecs/video_encoder_factory_template_libvpx_vp8_adapter.h" +#include "api/video_codecs/video_encoder_factory_template_libvpx_vp9_adapter.h" +#include "api/video_codecs/video_encoder_factory_template_open_h264_adapter.h" +#include "media/base/codec.h" +#include "media/base/media_constants.h" +#include "media/base/stream_params.h" +#include "p2p/base/transport_description.h" +#include "pc/peer_connection_wrapper.h" +#include "pc/test/fake_audio_capture_module.h" +#include "pc/test/fake_rtc_certificate_generator.h" +#include "pc/test/integration_test_helpers.h" +#include "pc/test/mock_peer_connection_observers.h" +#include "rtc_base/strings/string_format.h" +#include "rtc_base/thread.h" +#include "system_wrappers/include/metrics.h" +#include "test/gmock.h" +#include "test/gtest.h" +#include "test/wait_until.h" + +// This file contains unit tests that relate to the behavior of the +// SDP munging detector module. +// Tests are written as integration tests with PeerConnection, since the +// behaviors are still linked so closely that it is hard to test them in +// isolation. + +namespace webrtc { + +using ::testing::Eq; +using ::testing::IsTrue; +using ::testing::Pair; + +namespace { + +std::unique_ptr CreateAndStartThread() { + auto thread = Thread::Create(); + thread->Start(); + return thread; +} + +} // namespace + +class SdpMungingTest : public ::testing::Test { + public: + SdpMungingTest() + // Note: We use a PeerConnectionFactory with a distinct + // signaling thread, so that thread handling can be tested. + : signaling_thread_(CreateAndStartThread()), + pc_factory_(CreatePeerConnectionFactory( + nullptr, + nullptr, + signaling_thread_.get(), + FakeAudioCaptureModule::Create(), + CreateBuiltinAudioEncoderFactory(), + CreateBuiltinAudioDecoderFactory(), + std::make_unique< + VideoEncoderFactoryTemplate>(), + std::make_unique< + VideoDecoderFactoryTemplate>(), + nullptr /* audio_mixer */, + nullptr /* audio_processing */, + nullptr /* audio_frame_processor */)) { + metrics::Reset(); + } + + std::unique_ptr CreatePeerConnection( + std::unique_ptr field_trials = nullptr) { + RTCConfiguration config; + config.sdp_semantics = SdpSemantics::kUnifiedPlan; + return CreatePeerConnection(config, std::move(field_trials)); + } + + std::unique_ptr CreatePeerConnection( + const RTCConfiguration& config, + std::unique_ptr field_trials) { + auto observer = std::make_unique(); + PeerConnectionDependencies pc_deps(observer.get()); + pc_deps.trials = std::move(field_trials); + auto result = + pc_factory_->CreatePeerConnectionOrError(config, std::move(pc_deps)); + EXPECT_TRUE(result.ok()); + observer->SetPeerConnectionInterface(result.value().get()); + return std::make_unique( + pc_factory_, result.MoveValue(), std::move(observer)); + } + + protected: + std::unique_ptr signaling_thread_; + scoped_refptr pc_factory_; + + private: + AutoThread main_thread_; +}; + +TEST_F(SdpMungingTest, DISABLED_ReportUMAMetricsWithNoMunging) { + auto caller = CreatePeerConnection(); + auto callee = CreatePeerConnection(); + + caller->AddTransceiver(webrtc::MediaType::AUDIO); + caller->AddTransceiver(webrtc::MediaType::VIDEO); + + // Negotiate, gather candidates, then exchange ICE candidates. + ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get())); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kNoModification, 1))); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Answer.Initial"), + ElementsAre(Pair(SdpMungingType::kNoModification, 1))); + + EXPECT_THAT(WaitUntil([&] { return caller->IsIceGatheringDone(); }, IsTrue(), + {.timeout = kDefaultTimeout}), + IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return callee->IsIceGatheringDone(); }, IsTrue(), + {.timeout = kDefaultTimeout}), + IsRtcOk()); + for (const auto& candidate : caller->observer()->GetAllCandidates()) { + callee->pc()->AddIceCandidate(candidate); + } + for (const auto& candidate : callee->observer()->GetAllCandidates()) { + caller->pc()->AddIceCandidate(candidate); + } + EXPECT_THAT( + WaitUntil([&] { return caller->pc()->peer_connection_state(); }, + Eq(PeerConnectionInterface::PeerConnectionState::kConnected), + {.timeout = kDefaultTimeout}), + IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return callee->pc()->peer_connection_state(); }, + Eq(PeerConnectionInterface::PeerConnectionState::kConnected), + {.timeout = kDefaultTimeout}), + IsRtcOk()); + + caller->pc()->Close(); + callee->pc()->Close(); + + EXPECT_THAT( + metrics::Samples( + "WebRTC.PeerConnection.SdpMunging.Offer.ConnectionEstablished"), + ElementsAre(Pair(SdpMungingType::kNoModification, 1))); + EXPECT_THAT( + metrics::Samples( + "WebRTC.PeerConnection.SdpMunging.Answer.ConnectionEstablished"), + ElementsAre(Pair(SdpMungingType::kNoModification, 1))); + + EXPECT_THAT(metrics::Samples( + "WebRTC.PeerConnection.SdpMunging.Offer.ConnectionClosed"), + ElementsAre(Pair(SdpMungingType::kNoModification, 1))); + EXPECT_THAT(metrics::Samples( + "WebRTC.PeerConnection.SdpMunging.Answer.ConnectionClosed"), + ElementsAre(Pair(SdpMungingType::kNoModification, 1))); +} + +TEST_F(SdpMungingTest, InitialSetLocalDescriptionWithoutCreateOffer) { + RTCConfiguration config; + config.certificates.push_back( + FakeRTCCertificateGenerator::GenerateCertificate()); + auto pc = CreatePeerConnection(config, nullptr); + std::string sdp = + "v=0\r\n" + "o=- 0 3 IN IP4 127.0.0.1\r\n" + "s=-\r\n" + "t=0 0\r\n" + "a=fingerprint:sha-1 " + "D9:AB:00:AA:12:7B:62:54:CF:AD:3B:55:F7:60:BC:F3:40:A7:0B:5B\r\n" + "a=setup:actpass\r\n" + "a=ice-ufrag:ETEn\r\n" + "a=ice-pwd:OtSK0WpNtpUjkY4+86js7Z/l\r\n"; + auto offer = CreateSessionDescription(SdpType::kOffer, sdp); + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kWithoutCreateOffer, 1))); +} + +TEST_F(SdpMungingTest, InitialSetLocalDescriptionWithoutCreateAnswer) { + RTCConfiguration config; + config.certificates.push_back( + FakeRTCCertificateGenerator::GenerateCertificate()); + auto pc = CreatePeerConnection(config, nullptr); + std::string sdp = + "v=0\r\n" + "o=- 0 3 IN IP4 127.0.0.1\r\n" + "s=-\r\n" + "t=0 0\r\n" + "a=fingerprint:sha-1 " + "D9:AB:00:AA:12:7B:62:54:CF:AD:3B:55:F7:60:BC:F3:40:A7:0B:5B\r\n" + "a=setup:actpass\r\n" + "a=ice-ufrag:ETEn\r\n" + "a=ice-pwd:OtSK0WpNtpUjkY4+86js7Z/l\r\n" + "m=audio 9 UDP/TLS/RTP/SAVPF 111\r\n" + "c=IN IP4 0.0.0.0\r\n" + "a=rtcp-mux\r\n" + "a=sendrecv\r\n" + "a=mid:0\r\n" + "a=rtpmap:111 opus/48000/2\r\n"; + auto offer = CreateSessionDescription(SdpType::kOffer, sdp); + EXPECT_TRUE(pc->SetRemoteDescription(std::move(offer))); + + RTCError error; + auto answer = CreateSessionDescription(SdpType::kAnswer, sdp); + answer->description()->transport_infos()[0].description.connection_role = + CONNECTIONROLE_ACTIVE; + EXPECT_TRUE(pc->SetLocalDescription(std::move(answer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Answer.Initial"), + ElementsAre(Pair(SdpMungingType::kWithoutCreateAnswer, 1))); +} + +TEST_F(SdpMungingTest, IceUfrag) { + auto pc = CreatePeerConnection( + FieldTrials::CreateNoGlobal("WebRTC-NoSdpMangleUfrag/Enabled/")); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& transport_infos = offer->description()->transport_infos(); + ASSERT_EQ(transport_infos.size(), 1u); + transport_infos[0].description.ice_ufrag = + "amungediceufragthisshouldberejected"; + RTCError error; + // Ufrag is rejected. + EXPECT_FALSE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kIceUfrag, 1))); +} + +TEST_F(SdpMungingTest, IceUfragCheckDisabledByFieldTrial) { + auto pc = CreatePeerConnection( + FieldTrials::CreateNoGlobal("WebRTC-NoSdpMangleUfrag/Disabled/")); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& transport_infos = offer->description()->transport_infos(); + ASSERT_EQ(transport_infos.size(), 1u); + transport_infos[0].description.ice_ufrag = + "amungediceufragthisshouldberejected"; + RTCError error; + // Ufrag is not rejected. + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kIceUfrag, 1))); +} + +TEST_F(SdpMungingTest, IceUfragWithCheckDisabledForTesting) { + auto pc = CreatePeerConnection(); + pc->GetInternalPeerConnection()->DisableSdpMungingChecksForTesting(); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& transport_infos = offer->description()->transport_infos(); + ASSERT_EQ(transport_infos.size(), 1u); + transport_infos[0].description.ice_ufrag = + "amungediceufragthisshouldberejected"; + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kIceUfrag, 1))); +} + +TEST_F(SdpMungingTest, IcePwdCheckDisabledByFieldTrial) { + auto pc = CreatePeerConnection( + FieldTrials::CreateNoGlobal("WebRTC-NoSdpMangleUfrag/Disabled/")); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& transport_infos = offer->description()->transport_infos(); + ASSERT_EQ(transport_infos.size(), 1u); + transport_infos[0].description.ice_pwd = "amungedicepwdthisshouldberejected"; + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kIcePwd, 1))); +} + +TEST_F(SdpMungingTest, IcePwd) { + auto pc = CreatePeerConnection( + FieldTrials::CreateNoGlobal("WebRTC-NoSdpMangleUfrag/Enabled/")); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& transport_infos = offer->description()->transport_infos(); + ASSERT_EQ(transport_infos.size(), 1u); + transport_infos[0].description.ice_pwd = "amungedicepwdthisshouldberejected"; + RTCError error; + EXPECT_FALSE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kIcePwd, 1))); +} + +TEST_F(SdpMungingTest, IceUfragRestrictedAddresses) { + RTCConfiguration config; + config.certificates.push_back( + FakeRTCCertificateGenerator::GenerateCertificate()); + auto caller = CreatePeerConnection( + config, + FieldTrials::CreateNoGlobal("WebRTC-NoSdpMangleUfragRestrictedAddresses/" + "127.0.0.1:12345|127.0.0.*:23456|*:34567/")); + auto callee = CreatePeerConnection(); + caller->AddAudioTrack("audio_track", {}); + auto offer = caller->CreateOffer(); + auto& transport_infos = offer->description()->transport_infos(); + ASSERT_EQ(transport_infos.size(), 1u); + transport_infos[0].description.ice_ufrag = "amungediceufrag"; + + EXPECT_TRUE(caller->SetLocalDescription(offer->Clone())); + EXPECT_TRUE(callee->SetRemoteDescription(std::move(offer))); + + auto answer = callee->CreateAnswer(); + EXPECT_TRUE(callee->SetLocalDescription(answer->Clone())); + EXPECT_TRUE(caller->SetRemoteDescription(std::move(answer))); + + static constexpr const char tmpl[] = + "candidate:a0+B/1 1 udp 2130706432 %s typ host"; + + // Addresses to test. First field is the address in string format, + // second field is the expected outcome (success or failure). + const std::vector> address_tests = { + {"127.0.0.1:12345", false}, {"127.0.0.2:23456", false}, + {"8.8.8.8:34567", false}, {"127.0.0.2:12345", true}, + {"127.0.1.1:23456", true}, {"8.8.8.8:3456", true}, + }; + + for (const auto& address_test : address_tests) { + std::optional result; + const std::string candidate = StringFormat( + tmpl, absl::StrReplaceAll(address_test.first, {{":", " "}}).c_str()); + caller->pc()->AddIceCandidate( + std::unique_ptr( + CreateIceCandidate("", 0, candidate, nullptr)), + [&result](RTCError error) { result = error; }); + + ASSERT_THAT( + WaitUntil([&] { return result.has_value(); }, ::testing::IsTrue()), + IsRtcOk()); + if (address_test.second == true) { + EXPECT_TRUE(result.value().ok()); + } else { + EXPECT_FALSE(result.value().ok()); + EXPECT_EQ(result.value().type(), RTCErrorType::UNSUPPORTED_OPERATION); + } + } +} + +TEST_F(SdpMungingTest, IceUfragSdpRejectedAndRestrictedAddresses) { + RTCConfiguration config; + config.certificates.push_back( + FakeRTCCertificateGenerator::GenerateCertificate()); + auto caller = CreatePeerConnection( + config, + FieldTrials::CreateNoGlobal("WebRTC-NoSdpMangleUfragRestrictedAddresses/" + "127.0.0.1:12345|127.0.0.*:23456|*:34567/" + "WebRTC-NoSdpMangleUfrag/Enabled/")); + auto callee = CreatePeerConnection(); + caller->AddAudioTrack("audio_track", {}); + auto offer = caller->CreateOffer(); + auto& transport_infos = offer->description()->transport_infos(); + ASSERT_EQ(transport_infos.size(), 1u); + transport_infos[0].description.ice_ufrag = "amungediceufrag"; + + EXPECT_FALSE(caller->SetLocalDescription(offer->Clone())); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kIceUfrag, 1))); +} + +TEST_F(SdpMungingTest, IceMode) { + auto pc = CreatePeerConnection(); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& transport_infos = offer->description()->transport_infos(); + ASSERT_EQ(transport_infos.size(), 1u); + transport_infos[0].description.ice_mode = ICEMODE_LITE; + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kIceMode, 1))); +} + +TEST_F(SdpMungingTest, IceOptions) { + auto pc = CreatePeerConnection(); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& transport_infos = offer->description()->transport_infos(); + ASSERT_EQ(transport_infos.size(), 1u); + transport_infos[0].description.transport_options.push_back( + "something-unsupported"); + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kIceOptions, 1))); +} + +TEST_F(SdpMungingTest, IceOptionsRenomination) { + auto pc = CreatePeerConnection(); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& transport_infos = offer->description()->transport_infos(); + ASSERT_EQ(transport_infos.size(), 1u); + transport_infos[0].description.transport_options.push_back( + ICE_OPTION_RENOMINATION); + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kIceOptionsRenomination, 1))); +} + +TEST_F(SdpMungingTest, DtlsRole) { + auto pc = CreatePeerConnection(); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& transport_infos = offer->description()->transport_infos(); + ASSERT_EQ(transport_infos.size(), 1u); + transport_infos[0].description.connection_role = CONNECTIONROLE_PASSIVE; + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kDtlsSetup, 1))); +} + +TEST_F(SdpMungingTest, RemoveContentDefault) { + auto pc = CreatePeerConnection(); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto name = contents[0].mid(); + EXPECT_TRUE(offer->description()->RemoveContentByName(contents[0].mid())); + std::string sdp; + offer->ToString(&sdp); + auto modified_offer = CreateSessionDescription( + SdpType::kOffer, + absl::StrReplaceAll(sdp, {{"a=group:BUNDLE " + name, "a=group:BUNDLE"}})); + + RTCError error; + EXPECT_FALSE(pc->SetLocalDescription(std::move(modified_offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kNumberOfContents, 1))); +} + +TEST_F(SdpMungingTest, RemoveContentKillswitch) { + auto pc = CreatePeerConnection(FieldTrials::CreateNoGlobal( + "WebRTC-NoSdpMangleNumberOfContents/Disabled/")); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto name = contents[0].mid(); + EXPECT_TRUE(offer->description()->RemoveContentByName(contents[0].mid())); + std::string sdp; + offer->ToString(&sdp); + auto modified_offer = CreateSessionDescription( + SdpType::kOffer, + absl::StrReplaceAll(sdp, {{"a=group:BUNDLE " + name, "a=group:BUNDLE"}})); + + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(modified_offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kNumberOfContents, 1))); +} + +TEST_F(SdpMungingTest, TransceiverDirection) { + auto pc = CreatePeerConnection(); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + auto direction = media_description->direction(); + if (direction == RtpTransceiverDirection::kInactive) { + media_description->set_direction(RtpTransceiverDirection::kSendRecv); + } else { + media_description->set_direction(RtpTransceiverDirection::kInactive); + } + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kDirection, 1))); +} + +TEST_F(SdpMungingTest, Mid) { + auto pc = CreatePeerConnection(); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + std::string name(contents[0].mid()); + contents[0].set_mid("amungedmid"); + + auto& transport_infos = offer->description()->transport_infos(); + ASSERT_EQ(transport_infos.size(), 1u); + transport_infos[0].content_name = "amungedmid"; + std::string sdp; + offer->ToString(&sdp); + auto modified_offer = CreateSessionDescription( + SdpType::kOffer, + absl::StrReplaceAll( + sdp, {{"a=group:BUNDLE " + name, "a=group:BUNDLE amungedmid"}})); + + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(modified_offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kMid, 1))); +} + +TEST_F(SdpMungingTest, LegacySimulcast) { + auto pc = CreatePeerConnection(); + pc->AddVideoTrack("video_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + uint32_t ssrc = media_description->first_ssrc(); + ASSERT_EQ(media_description->streams().size(), 1u); + const std::string& cname = media_description->streams()[0].cname; + + std::string sdp; + offer->ToString(&sdp); + sdp += "a=ssrc-group:SIM " + absl::StrCat(ssrc) + " " + + absl::StrCat(ssrc + 1) + "\r\n" + // + "a=ssrc-group:FID " + absl::StrCat(ssrc + 1) + " " + + absl::StrCat(ssrc + 2) + "\r\n" + // + "a=ssrc:" + absl::StrCat(ssrc + 1) + " msid:- video_track\r\n" + // + "a=ssrc:" + absl::StrCat(ssrc + 1) + " cname:" + cname + "\r\n" + // + "a=ssrc:" + absl::StrCat(ssrc + 2) + " msid:- video_track\r\n" + // + "a=ssrc:" + absl::StrCat(ssrc + 2) + " cname:" + cname + "\r\n"; + auto modified_offer = CreateSessionDescription(SdpType::kOffer, sdp); + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(modified_offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kVideoCodecsLegacySimulcast, 1))); +} + +#ifdef WEBRTC_USE_H264 +TEST_F(SdpMungingTest, H264SpsPpsIdrInKeyFrame) { + auto pc = CreatePeerConnection(); + pc->AddVideoTrack("video_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + std::vector codecs = media_description->codecs(); + for (auto& codec : codecs) { + if (codec.name == webrtc::kH264CodecName) { + codec.SetParam(webrtc::kH264FmtpSpsPpsIdrInKeyframe, + webrtc::kParamValueTrue); + } + } + media_description->set_codecs(codecs); + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre( + Pair(SdpMungingType::kVideoCodecsFmtpH264SpsPpsIdrInKeyframe, 1))); +} +#endif // WEBRTC_USE_H264 + +TEST_F(SdpMungingTest, OpusStereo) { + auto pc = CreatePeerConnection(); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + std::vector codecs = media_description->codecs(); + for (auto& codec : codecs) { + if (codec.name == kOpusCodecName) { + codec.SetParam(kCodecParamStereo, kParamValueTrue); + } + } + media_description->set_codecs(codecs); + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kAudioCodecsFmtpOpusStereo, 1))); +} + +TEST_F(SdpMungingTest, OpusFec) { + auto pc = CreatePeerConnection(); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + std::vector codecs = media_description->codecs(); + for (auto& codec : codecs) { + if (codec.name == kOpusCodecName) { + // Enabled by default so we need to remove the parameter. + EXPECT_TRUE(codec.RemoveParam(kCodecParamUseInbandFec)); + } + } + media_description->set_codecs(codecs); + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kAudioCodecsFmtpOpusFec, 1))); +} + +TEST_F(SdpMungingTest, OpusDtx) { + auto pc = CreatePeerConnection(); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + std::vector codecs = media_description->codecs(); + for (auto& codec : codecs) { + if (codec.name == kOpusCodecName) { + codec.SetParam(kCodecParamUseDtx, kParamValueTrue); + } + } + media_description->set_codecs(codecs); + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kAudioCodecsFmtpOpusDtx, 1))); +} + +TEST_F(SdpMungingTest, OpusCbr) { + auto pc = CreatePeerConnection(); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + std::vector codecs = media_description->codecs(); + for (auto& codec : codecs) { + if (codec.name == kOpusCodecName) { + codec.SetParam(kCodecParamCbr, kParamValueTrue); + } + } + media_description->set_codecs(codecs); + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kAudioCodecsFmtpOpusCbr, 1))); +} + +TEST_F(SdpMungingTest, AudioCodecsRemoved) { + auto pc = CreatePeerConnection(); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + std::vector codecs = media_description->codecs(); + codecs.pop_back(); + media_description->set_codecs(codecs); + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kAudioCodecsRemoved, 1))); +} + +TEST_F(SdpMungingTest, AudioCodecsAdded) { + auto pc = CreatePeerConnection(); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + std::vector codecs = media_description->codecs(); + auto codec = CreateAudioCodec(SdpAudioFormat("pcmu", 8000, 1, {})); + codec.id = 19; // IANA reserved payload type, should not conflict. + codecs.push_back(codec); + media_description->set_codecs(codecs); + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kAudioCodecsAdded, 1))); +} + +TEST_F(SdpMungingTest, VideoCodecsRemoved) { + auto pc = CreatePeerConnection(); + pc->AddVideoTrack("video_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + std::vector codecs = media_description->codecs(); + codecs.pop_back(); + media_description->set_codecs(codecs); + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kVideoCodecsRemoved, 1))); +} + +TEST_F(SdpMungingTest, VideoCodecsAdded) { + auto pc = CreatePeerConnection(); + pc->AddVideoTrack("video_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + std::vector codecs = media_description->codecs(); + auto codec = CreateVideoCodec(SdpVideoFormat("VP8", {})); + codec.id = 19; // IANA reserved payload type, should not conflict. + codecs.push_back(codec); + media_description->set_codecs(codecs); + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kVideoCodecsAdded, 1))); +} + +TEST_F(SdpMungingTest, VideoCodecsAddedWithRawPacketization) { + auto pc = CreatePeerConnection(); + pc->AddVideoTrack("video_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + std::vector codecs = media_description->codecs(); + auto codec = CreateVideoCodec(SdpVideoFormat("VP8", {})); + codec.id = 19; // IANA reserved payload type, should not conflict. + codec.packetization = "raw"; + codecs.push_back(codec); + media_description->set_codecs(codecs); + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre( + Pair(SdpMungingType::kVideoCodecsAddedWithRawPacketization, 1))); +} + +TEST_F(SdpMungingTest, VideoCodecsModifiedWithRawPacketization) { + auto pc = CreatePeerConnection(); + pc->AddVideoTrack("video_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + std::vector codecs = media_description->codecs(); + ASSERT_TRUE(!codecs.empty()); + codecs[0].packetization = "raw"; + media_description->set_codecs(codecs); + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre( + Pair(SdpMungingType::kVideoCodecsModifiedWithRawPacketization, 1))); +} + +TEST_F(SdpMungingTest, MultiOpus) { + auto pc = CreatePeerConnection(); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + std::vector codecs = media_description->codecs(); + auto multiopus = + CreateAudioCodec(SdpAudioFormat("multiopus", 48000, 4, + {{"channel_mapping", "0,1,2,3"}, + {"coupled_streams", "2"}, + {"num_streams", "2"}})); + multiopus.id = 19; // IANA reserved payload type, should not conflict. + codecs.push_back(multiopus); + media_description->set_codecs(codecs); + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kAudioCodecsAddedMultiOpus, 1))); +} + +TEST_F(SdpMungingTest, L16) { + auto pc = CreatePeerConnection(); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + std::vector codecs = media_description->codecs(); + auto l16 = CreateAudioCodec(SdpAudioFormat("L16", 48000, 2, {})); + l16.id = 19; // IANA reserved payload type, should not conflict. + codecs.push_back(l16); + media_description->set_codecs(codecs); + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kAudioCodecsAddedL16, 1))); +} + +TEST_F(SdpMungingTest, AudioSsrc) { + // Note: same applies to video but is harder to write since one needs to + // modify the ssrc-group too. + auto pc = CreatePeerConnection(); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + ASSERT_EQ(media_description->streams().size(), 1u); + media_description->mutable_streams()[0].ssrcs[0] = 4404; + + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kSsrcs, 1))); +} + +TEST_F(SdpMungingTest, HeaderExtensionAdded) { + auto pc = CreatePeerConnection(); + pc->AddVideoTrack("video_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + // VLA is off by default, id=42 should be unused. + media_description->AddRtpHeaderExtension( + {RtpExtension::kVideoLayersAllocationUri, 42}); + + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kRtpHeaderExtensionAdded, 1))); +} + +TEST_F(SdpMungingTest, HeaderExtensionRemoved) { + auto pc = CreatePeerConnection(); + pc->AddVideoTrack("video_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + media_description->ClearRtpHeaderExtensions(); + + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kRtpHeaderExtensionRemoved, 1))); +} + +TEST_F(SdpMungingTest, HeaderExtensionModified) { + auto pc = CreatePeerConnection(); + pc->AddVideoTrack("video_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + auto extensions = media_description->rtp_header_extensions(); + ASSERT_GT(extensions.size(), 0u); + extensions[0].id = 42; // id=42 should be unused. + media_description->set_rtp_header_extensions(extensions); + + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kRtpHeaderExtensionModified, 1))); +} + +TEST_F(SdpMungingTest, PayloadTypeChanged) { + auto pc = CreatePeerConnection(); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + auto codecs = media_description->codecs(); + ASSERT_GT(codecs.size(), 0u); + codecs[0].id = 19; // IANA reserved payload type, should not conflict. + media_description->set_codecs(codecs); + + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kPayloadTypes, 1))); +} + +TEST_F(SdpMungingTest, AudioCodecsReordered) { + auto pc = CreatePeerConnection(); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + auto codecs = media_description->codecs(); + ASSERT_GT(codecs.size(), 1u); + std::swap(codecs[0], codecs[1]); + media_description->set_codecs(codecs); + + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kAudioCodecsReordered, 1))); +} + +TEST_F(SdpMungingTest, VideoCodecsReordered) { + auto pc = CreatePeerConnection(); + pc->AddVideoTrack("video_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + auto codecs = media_description->codecs(); + ASSERT_GT(codecs.size(), 1u); + std::swap(codecs[0], codecs[1]); + media_description->set_codecs(codecs); + + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kVideoCodecsReordered, 1))); +} + +TEST_F(SdpMungingTest, AudioCodecsFmtp) { + auto pc = CreatePeerConnection(); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + auto codecs = media_description->codecs(); + ASSERT_GT(codecs.size(), 0u); + codecs[0].params["dont"] = "munge"; + media_description->set_codecs(codecs); + + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kAudioCodecsFmtp, 1))); +} + +TEST_F(SdpMungingTest, VideoCodecsFmtp) { + auto pc = CreatePeerConnection(); + pc->AddVideoTrack("video_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + auto codecs = media_description->codecs(); + ASSERT_GT(codecs.size(), 0u); + codecs[0].params["dont"] = "munge"; + media_description->set_codecs(codecs); + + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kVideoCodecsFmtp, 1))); +} + +TEST_F(SdpMungingTest, AudioCodecsRtcpFb) { + auto pc = CreatePeerConnection(); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + auto codecs = media_description->codecs(); + ASSERT_GT(codecs.size(), 0u); + codecs[0].feedback_params.Add({"dont", "munge"}); + media_description->set_codecs(codecs); + + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kAudioCodecsRtcpFb, 1))); +} + +TEST_F(SdpMungingTest, AudioCodecsRtcpFbNack) { + auto pc = CreatePeerConnection(); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + auto codecs = media_description->codecs(); + ASSERT_GT(codecs.size(), 0u); + codecs[0].feedback_params.Add(FeedbackParam("nack")); + media_description->set_codecs(codecs); + + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kAudioCodecsRtcpFbAudioNack, 1))); +} + +TEST_F(SdpMungingTest, AudioCodecsRtcpFbRrtr) { + auto pc = CreatePeerConnection(); + pc->AddAudioTrack("audio_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + auto codecs = media_description->codecs(); + ASSERT_GT(codecs.size(), 0u); + codecs[0].feedback_params.Add(FeedbackParam("rrtr")); + media_description->set_codecs(codecs); + + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kAudioCodecsRtcpFbRrtr, 1))); +} + +TEST_F(SdpMungingTest, VideoCodecsRtcpFb) { + auto pc = CreatePeerConnection(); + pc->AddVideoTrack("video_track", {}); + + auto offer = pc->CreateOffer(); + auto& contents = offer->description()->contents(); + ASSERT_EQ(contents.size(), 1u); + auto* media_description = contents[0].media_description(); + ASSERT_TRUE(media_description); + auto codecs = media_description->codecs(); + ASSERT_GT(codecs.size(), 0u); + codecs[0].feedback_params.Add({"dont", "munge"}); + media_description->set_codecs(codecs); + + RTCError error; + EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), + ElementsAre(Pair(SdpMungingType::kVideoCodecsRtcpFb, 1))); +} + +} // namespace webrtc diff --git a/pc/sdp_offer_answer_unittest.cc b/pc/sdp_offer_answer_unittest.cc index 3fe6e19799..3019c3cced 100644 --- a/pc/sdp_offer_answer_unittest.cc +++ b/pc/sdp_offer_answer_unittest.cc @@ -20,7 +20,6 @@ #include "absl/strings/match.h" #include "absl/strings/str_cat.h" #include "absl/strings/str_replace.h" -#include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" #include "api/create_peerconnection_factory.h" @@ -34,8 +33,6 @@ #include "api/rtp_transceiver_direction.h" #include "api/rtp_transceiver_interface.h" #include "api/scoped_refptr.h" -#include "api/test/rtc_error_matchers.h" -#include "api/uma_metrics.h" #include "api/video_codecs/sdp_video_format.h" #include "api/video_codecs/video_decoder_factory_template.h" #include "api/video_codecs/video_decoder_factory_template_dav1d_adapter.h" @@ -50,33 +47,29 @@ #include "media/base/codec.h" #include "media/base/media_constants.h" #include "media/base/stream_params.h" -#include "p2p/base/transport_description.h" #include "pc/peer_connection_wrapper.h" #include "pc/test/fake_audio_capture_module.h" -#include "pc/test/fake_rtc_certificate_generator.h" #include "pc/test/integration_test_helpers.h" #include "pc/test/mock_peer_connection_observers.h" -#include "rtc_base/strings/string_format.h" #include "rtc_base/thread.h" #include "system_wrappers/include/metrics.h" #include "test/gmock.h" #include "test/gtest.h" -#include "test/wait_until.h" // This file contains unit tests that relate to the behavior of the // SdpOfferAnswer module. -// Tests are writen as integration tests with PeerConnection, since the +// Tests are written as integration tests with PeerConnection, since the // behaviors are still linked so closely that it is hard to test them in // isolation. namespace webrtc { +using ::testing::ElementsAre; using ::testing::Eq; using ::testing::IsTrue; -using RTCConfiguration = PeerConnectionInterface::RTCConfiguration; -using ::testing::ElementsAre; using ::testing::Pair; using ::testing::SizeIs; +using RTCConfiguration = PeerConnectionInterface::RTCConfiguration; namespace { @@ -1743,1034 +1736,4 @@ TEST_F(SdpOfferAnswerTest, PayloadTypeMatchingWithSubsequentOfferAnswer) { EXPECT_EQ(codecs[1].id, av1.id); } -class SdpOfferAnswerMungingTest : public SdpOfferAnswerTest { - public: - SdpOfferAnswerMungingTest() : SdpOfferAnswerTest() { metrics::Reset(); } -}; - -TEST_F(SdpOfferAnswerMungingTest, DISABLED_ReportUMAMetricsWithNoMunging) { - auto caller = CreatePeerConnection(); - auto callee = CreatePeerConnection(); - - caller->AddTransceiver(webrtc::MediaType::AUDIO); - caller->AddTransceiver(webrtc::MediaType::VIDEO); - - // Negotiate, gather candidates, then exchange ICE candidates. - ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get())); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kNoModification, 1))); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Answer.Initial"), - ElementsAre(Pair(SdpMungingType::kNoModification, 1))); - - EXPECT_THAT(WaitUntil([&] { return caller->IsIceGatheringDone(); }, IsTrue(), - {.timeout = kDefaultTimeout}), - IsRtcOk()); - EXPECT_THAT(WaitUntil([&] { return callee->IsIceGatheringDone(); }, IsTrue(), - {.timeout = kDefaultTimeout}), - IsRtcOk()); - for (const auto& candidate : caller->observer()->GetAllCandidates()) { - callee->pc()->AddIceCandidate(candidate); - } - for (const auto& candidate : callee->observer()->GetAllCandidates()) { - caller->pc()->AddIceCandidate(candidate); - } - EXPECT_THAT( - WaitUntil([&] { return caller->pc()->peer_connection_state(); }, - Eq(PeerConnectionInterface::PeerConnectionState::kConnected), - {.timeout = kDefaultTimeout}), - IsRtcOk()); - EXPECT_THAT( - WaitUntil([&] { return callee->pc()->peer_connection_state(); }, - Eq(PeerConnectionInterface::PeerConnectionState::kConnected), - {.timeout = kDefaultTimeout}), - IsRtcOk()); - - caller->pc()->Close(); - callee->pc()->Close(); - - EXPECT_THAT( - metrics::Samples( - "WebRTC.PeerConnection.SdpMunging.Offer.ConnectionEstablished"), - ElementsAre(Pair(SdpMungingType::kNoModification, 1))); - EXPECT_THAT( - metrics::Samples( - "WebRTC.PeerConnection.SdpMunging.Answer.ConnectionEstablished"), - ElementsAre(Pair(SdpMungingType::kNoModification, 1))); - - EXPECT_THAT(metrics::Samples( - "WebRTC.PeerConnection.SdpMunging.Offer.ConnectionClosed"), - ElementsAre(Pair(SdpMungingType::kNoModification, 1))); - EXPECT_THAT(metrics::Samples( - "WebRTC.PeerConnection.SdpMunging.Answer.ConnectionClosed"), - ElementsAre(Pair(SdpMungingType::kNoModification, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, - InitialSetLocalDescriptionWithoutCreateOffer) { - RTCConfiguration config; - config.certificates.push_back( - FakeRTCCertificateGenerator::GenerateCertificate()); - auto pc = CreatePeerConnection(config, nullptr); - std::string sdp = - "v=0\r\n" - "o=- 0 3 IN IP4 127.0.0.1\r\n" - "s=-\r\n" - "t=0 0\r\n" - "a=fingerprint:sha-1 " - "D9:AB:00:AA:12:7B:62:54:CF:AD:3B:55:F7:60:BC:F3:40:A7:0B:5B\r\n" - "a=setup:actpass\r\n" - "a=ice-ufrag:ETEn\r\n" - "a=ice-pwd:OtSK0WpNtpUjkY4+86js7Z/l\r\n"; - auto offer = CreateSessionDescription(SdpType::kOffer, sdp); - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kWithoutCreateOffer, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, - InitialSetLocalDescriptionWithoutCreateAnswer) { - RTCConfiguration config; - config.certificates.push_back( - FakeRTCCertificateGenerator::GenerateCertificate()); - auto pc = CreatePeerConnection(config, nullptr); - std::string sdp = - "v=0\r\n" - "o=- 0 3 IN IP4 127.0.0.1\r\n" - "s=-\r\n" - "t=0 0\r\n" - "a=fingerprint:sha-1 " - "D9:AB:00:AA:12:7B:62:54:CF:AD:3B:55:F7:60:BC:F3:40:A7:0B:5B\r\n" - "a=setup:actpass\r\n" - "a=ice-ufrag:ETEn\r\n" - "a=ice-pwd:OtSK0WpNtpUjkY4+86js7Z/l\r\n" - "m=audio 9 UDP/TLS/RTP/SAVPF 111\r\n" - "c=IN IP4 0.0.0.0\r\n" - "a=rtcp-mux\r\n" - "a=sendrecv\r\n" - "a=mid:0\r\n" - "a=rtpmap:111 opus/48000/2\r\n"; - auto offer = CreateSessionDescription(SdpType::kOffer, sdp); - EXPECT_TRUE(pc->SetRemoteDescription(std::move(offer))); - - RTCError error; - auto answer = CreateSessionDescription(SdpType::kAnswer, sdp); - answer->description()->transport_infos()[0].description.connection_role = - CONNECTIONROLE_ACTIVE; - EXPECT_TRUE(pc->SetLocalDescription(std::move(answer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Answer.Initial"), - ElementsAre(Pair(SdpMungingType::kWithoutCreateAnswer, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, IceUfrag) { - auto pc = CreatePeerConnection( - FieldTrials::CreateNoGlobal("WebRTC-NoSdpMangleUfrag/Enabled/")); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& transport_infos = offer->description()->transport_infos(); - ASSERT_EQ(transport_infos.size(), 1u); - transport_infos[0].description.ice_ufrag = - "amungediceufragthisshouldberejected"; - RTCError error; - // Ufrag is rejected. - EXPECT_FALSE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kIceUfrag, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, IceUfragCheckDisabledByFieldTrial) { - auto pc = CreatePeerConnection( - FieldTrials::CreateNoGlobal("WebRTC-NoSdpMangleUfrag/Disabled/")); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& transport_infos = offer->description()->transport_infos(); - ASSERT_EQ(transport_infos.size(), 1u); - transport_infos[0].description.ice_ufrag = - "amungediceufragthisshouldberejected"; - RTCError error; - // Ufrag is not rejected. - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kIceUfrag, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, IceUfragWithCheckDisabledForTesting) { - auto pc = CreatePeerConnection(); - pc->GetInternalPeerConnection()->DisableSdpMungingChecksForTesting(); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& transport_infos = offer->description()->transport_infos(); - ASSERT_EQ(transport_infos.size(), 1u); - transport_infos[0].description.ice_ufrag = - "amungediceufragthisshouldberejected"; - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kIceUfrag, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, IcePwdCheckDisabledByFieldTrial) { - auto pc = CreatePeerConnection( - FieldTrials::CreateNoGlobal("WebRTC-NoSdpMangleUfrag/Disabled/")); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& transport_infos = offer->description()->transport_infos(); - ASSERT_EQ(transport_infos.size(), 1u); - transport_infos[0].description.ice_pwd = "amungedicepwdthisshouldberejected"; - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kIcePwd, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, IcePwd) { - auto pc = CreatePeerConnection( - FieldTrials::CreateNoGlobal("WebRTC-NoSdpMangleUfrag/Enabled/")); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& transport_infos = offer->description()->transport_infos(); - ASSERT_EQ(transport_infos.size(), 1u); - transport_infos[0].description.ice_pwd = "amungedicepwdthisshouldberejected"; - RTCError error; - EXPECT_FALSE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kIcePwd, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, IceUfragRestrictedAddresses) { - RTCConfiguration config; - config.certificates.push_back( - FakeRTCCertificateGenerator::GenerateCertificate()); - auto caller = CreatePeerConnection( - config, - FieldTrials::CreateNoGlobal("WebRTC-NoSdpMangleUfragRestrictedAddresses/" - "127.0.0.1:12345|127.0.0.*:23456|*:34567/")); - auto callee = CreatePeerConnection(); - caller->AddAudioTrack("audio_track", {}); - auto offer = caller->CreateOffer(); - auto& transport_infos = offer->description()->transport_infos(); - ASSERT_EQ(transport_infos.size(), 1u); - transport_infos[0].description.ice_ufrag = "amungediceufrag"; - - EXPECT_TRUE(caller->SetLocalDescription(offer->Clone())); - EXPECT_TRUE(callee->SetRemoteDescription(std::move(offer))); - - auto answer = callee->CreateAnswer(); - EXPECT_TRUE(callee->SetLocalDescription(answer->Clone())); - EXPECT_TRUE(caller->SetRemoteDescription(std::move(answer))); - - static constexpr const char tmpl[] = - "candidate:a0+B/1 1 udp 2130706432 %s typ host"; - - // Addresses to test. First field is the address in string format, - // second field is the expected outcome (success or failure). - const std::vector> address_tests = { - {"127.0.0.1:12345", false}, {"127.0.0.2:23456", false}, - {"8.8.8.8:34567", false}, {"127.0.0.2:12345", true}, - {"127.0.1.1:23456", true}, {"8.8.8.8:3456", true}, - }; - - for (const auto& address_test : address_tests) { - std::optional result; - const std::string candidate = StringFormat( - tmpl, absl::StrReplaceAll(address_test.first, {{":", " "}}).c_str()); - caller->pc()->AddIceCandidate( - std::unique_ptr( - CreateIceCandidate("", 0, candidate, nullptr)), - [&result](RTCError error) { result = error; }); - - ASSERT_THAT( - WaitUntil([&] { return result.has_value(); }, ::testing::IsTrue()), - IsRtcOk()); - if (address_test.second == true) { - EXPECT_TRUE(result.value().ok()); - } else { - EXPECT_FALSE(result.value().ok()); - EXPECT_EQ(result.value().type(), RTCErrorType::UNSUPPORTED_OPERATION); - } - } -} - -TEST_F(SdpOfferAnswerMungingTest, IceUfragSdpRejectedAndRestrictedAddresses) { - RTCConfiguration config; - config.certificates.push_back( - FakeRTCCertificateGenerator::GenerateCertificate()); - auto caller = CreatePeerConnection( - config, - FieldTrials::CreateNoGlobal("WebRTC-NoSdpMangleUfragRestrictedAddresses/" - "127.0.0.1:12345|127.0.0.*:23456|*:34567/" - "WebRTC-NoSdpMangleUfrag/Enabled/")); - auto callee = CreatePeerConnection(); - caller->AddAudioTrack("audio_track", {}); - auto offer = caller->CreateOffer(); - auto& transport_infos = offer->description()->transport_infos(); - ASSERT_EQ(transport_infos.size(), 1u); - transport_infos[0].description.ice_ufrag = "amungediceufrag"; - - EXPECT_FALSE(caller->SetLocalDescription(offer->Clone())); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kIceUfrag, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, IceMode) { - auto pc = CreatePeerConnection(); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& transport_infos = offer->description()->transport_infos(); - ASSERT_EQ(transport_infos.size(), 1u); - transport_infos[0].description.ice_mode = ICEMODE_LITE; - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kIceMode, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, IceOptions) { - auto pc = CreatePeerConnection(); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& transport_infos = offer->description()->transport_infos(); - ASSERT_EQ(transport_infos.size(), 1u); - transport_infos[0].description.transport_options.push_back( - "something-unsupported"); - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kIceOptions, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, IceOptionsRenomination) { - auto pc = CreatePeerConnection(); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& transport_infos = offer->description()->transport_infos(); - ASSERT_EQ(transport_infos.size(), 1u); - transport_infos[0].description.transport_options.push_back( - ICE_OPTION_RENOMINATION); - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kIceOptionsRenomination, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, DtlsRole) { - auto pc = CreatePeerConnection(); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& transport_infos = offer->description()->transport_infos(); - ASSERT_EQ(transport_infos.size(), 1u); - transport_infos[0].description.connection_role = CONNECTIONROLE_PASSIVE; - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kDtlsSetup, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, RemoveContentDefault) { - auto pc = CreatePeerConnection(); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto name = contents[0].mid(); - EXPECT_TRUE(offer->description()->RemoveContentByName(contents[0].mid())); - std::string sdp; - offer->ToString(&sdp); - auto modified_offer = CreateSessionDescription( - SdpType::kOffer, - absl::StrReplaceAll(sdp, {{"a=group:BUNDLE " + name, "a=group:BUNDLE"}})); - - RTCError error; - EXPECT_FALSE(pc->SetLocalDescription(std::move(modified_offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kNumberOfContents, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, RemoveContentKillswitch) { - auto pc = CreatePeerConnection(FieldTrials::CreateNoGlobal( - "WebRTC-NoSdpMangleNumberOfContents/Disabled/")); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto name = contents[0].mid(); - EXPECT_TRUE(offer->description()->RemoveContentByName(contents[0].mid())); - std::string sdp; - offer->ToString(&sdp); - auto modified_offer = CreateSessionDescription( - SdpType::kOffer, - absl::StrReplaceAll(sdp, {{"a=group:BUNDLE " + name, "a=group:BUNDLE"}})); - - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(modified_offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kNumberOfContents, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, TransceiverDirection) { - auto pc = CreatePeerConnection(); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - auto direction = media_description->direction(); - if (direction == RtpTransceiverDirection::kInactive) { - media_description->set_direction(RtpTransceiverDirection::kSendRecv); - } else { - media_description->set_direction(RtpTransceiverDirection::kInactive); - } - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kDirection, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, Mid) { - auto pc = CreatePeerConnection(); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - std::string name(contents[0].mid()); - contents[0].set_mid("amungedmid"); - - auto& transport_infos = offer->description()->transport_infos(); - ASSERT_EQ(transport_infos.size(), 1u); - transport_infos[0].content_name = "amungedmid"; - std::string sdp; - offer->ToString(&sdp); - auto modified_offer = CreateSessionDescription( - SdpType::kOffer, - absl::StrReplaceAll( - sdp, {{"a=group:BUNDLE " + name, "a=group:BUNDLE amungedmid"}})); - - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(modified_offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kMid, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, LegacySimulcast) { - auto pc = CreatePeerConnection(); - pc->AddVideoTrack("video_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - uint32_t ssrc = media_description->first_ssrc(); - ASSERT_EQ(media_description->streams().size(), 1u); - const std::string& cname = media_description->streams()[0].cname; - - std::string sdp; - offer->ToString(&sdp); - sdp += "a=ssrc-group:SIM " + absl::StrCat(ssrc) + " " + - absl::StrCat(ssrc + 1) + "\r\n" + // - "a=ssrc-group:FID " + absl::StrCat(ssrc + 1) + " " + - absl::StrCat(ssrc + 2) + "\r\n" + // - "a=ssrc:" + absl::StrCat(ssrc + 1) + " msid:- video_track\r\n" + // - "a=ssrc:" + absl::StrCat(ssrc + 1) + " cname:" + cname + "\r\n" + // - "a=ssrc:" + absl::StrCat(ssrc + 2) + " msid:- video_track\r\n" + // - "a=ssrc:" + absl::StrCat(ssrc + 2) + " cname:" + cname + "\r\n"; - auto modified_offer = CreateSessionDescription(SdpType::kOffer, sdp); - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(modified_offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kVideoCodecsLegacySimulcast, 1))); -} - -#ifdef WEBRTC_USE_H264 -TEST_F(SdpOfferAnswerMungingTest, H264SpsPpsIdrInKeyFrame) { - auto pc = CreatePeerConnection(); - pc->AddVideoTrack("video_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - std::vector codecs = media_description->codecs(); - for (auto& codec : codecs) { - if (codec.name == webrtc::kH264CodecName) { - codec.SetParam(webrtc::kH264FmtpSpsPpsIdrInKeyframe, - webrtc::kParamValueTrue); - } - } - media_description->set_codecs(codecs); - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre( - Pair(SdpMungingType::kVideoCodecsFmtpH264SpsPpsIdrInKeyframe, 1))); -} -#endif // WEBRTC_USE_H264 - -TEST_F(SdpOfferAnswerMungingTest, OpusStereo) { - auto pc = CreatePeerConnection(); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - std::vector codecs = media_description->codecs(); - for (auto& codec : codecs) { - if (codec.name == kOpusCodecName) { - codec.SetParam(kCodecParamStereo, kParamValueTrue); - } - } - media_description->set_codecs(codecs); - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kAudioCodecsFmtpOpusStereo, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, OpusFec) { - auto pc = CreatePeerConnection(); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - std::vector codecs = media_description->codecs(); - for (auto& codec : codecs) { - if (codec.name == kOpusCodecName) { - // Enabled by default so we need to remove the parameter. - EXPECT_TRUE(codec.RemoveParam(kCodecParamUseInbandFec)); - } - } - media_description->set_codecs(codecs); - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kAudioCodecsFmtpOpusFec, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, OpusDtx) { - auto pc = CreatePeerConnection(); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - std::vector codecs = media_description->codecs(); - for (auto& codec : codecs) { - if (codec.name == kOpusCodecName) { - codec.SetParam(kCodecParamUseDtx, kParamValueTrue); - } - } - media_description->set_codecs(codecs); - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kAudioCodecsFmtpOpusDtx, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, OpusCbr) { - auto pc = CreatePeerConnection(); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - std::vector codecs = media_description->codecs(); - for (auto& codec : codecs) { - if (codec.name == kOpusCodecName) { - codec.SetParam(kCodecParamCbr, kParamValueTrue); - } - } - media_description->set_codecs(codecs); - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kAudioCodecsFmtpOpusCbr, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, AudioCodecsRemoved) { - auto pc = CreatePeerConnection(); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - std::vector codecs = media_description->codecs(); - codecs.pop_back(); - media_description->set_codecs(codecs); - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kAudioCodecsRemoved, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, AudioCodecsAdded) { - auto pc = CreatePeerConnection(); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - std::vector codecs = media_description->codecs(); - auto codec = CreateAudioCodec(SdpAudioFormat("pcmu", 8000, 1, {})); - codec.id = 19; // IANA reserved payload type, should not conflict. - codecs.push_back(codec); - media_description->set_codecs(codecs); - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kAudioCodecsAdded, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, VideoCodecsRemoved) { - auto pc = CreatePeerConnection(); - pc->AddVideoTrack("video_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - std::vector codecs = media_description->codecs(); - codecs.pop_back(); - media_description->set_codecs(codecs); - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kVideoCodecsRemoved, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, VideoCodecsAdded) { - auto pc = CreatePeerConnection(); - pc->AddVideoTrack("video_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - std::vector codecs = media_description->codecs(); - auto codec = CreateVideoCodec(SdpVideoFormat("VP8", {})); - codec.id = 19; // IANA reserved payload type, should not conflict. - codecs.push_back(codec); - media_description->set_codecs(codecs); - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kVideoCodecsAdded, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, VideoCodecsAddedWithRawPacketization) { - auto pc = CreatePeerConnection(); - pc->AddVideoTrack("video_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - std::vector codecs = media_description->codecs(); - auto codec = CreateVideoCodec(SdpVideoFormat("VP8", {})); - codec.id = 19; // IANA reserved payload type, should not conflict. - codec.packetization = "raw"; - codecs.push_back(codec); - media_description->set_codecs(codecs); - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre( - Pair(SdpMungingType::kVideoCodecsAddedWithRawPacketization, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, VideoCodecsModifiedWithRawPacketization) { - auto pc = CreatePeerConnection(); - pc->AddVideoTrack("video_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - std::vector codecs = media_description->codecs(); - ASSERT_TRUE(!codecs.empty()); - codecs[0].packetization = "raw"; - media_description->set_codecs(codecs); - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre( - Pair(SdpMungingType::kVideoCodecsModifiedWithRawPacketization, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, MultiOpus) { - auto pc = CreatePeerConnection(); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - std::vector codecs = media_description->codecs(); - auto multiopus = - CreateAudioCodec(SdpAudioFormat("multiopus", 48000, 4, - {{"channel_mapping", "0,1,2,3"}, - {"coupled_streams", "2"}, - {"num_streams", "2"}})); - multiopus.id = 19; // IANA reserved payload type, should not conflict. - codecs.push_back(multiopus); - media_description->set_codecs(codecs); - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kAudioCodecsAddedMultiOpus, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, L16) { - auto pc = CreatePeerConnection(); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - std::vector codecs = media_description->codecs(); - auto l16 = CreateAudioCodec(SdpAudioFormat("L16", 48000, 2, {})); - l16.id = 19; // IANA reserved payload type, should not conflict. - codecs.push_back(l16); - media_description->set_codecs(codecs); - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kAudioCodecsAddedL16, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, AudioSsrc) { - // Note: same applies to video but is harder to write since one needs to - // modify the ssrc-group too. - auto pc = CreatePeerConnection(); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - ASSERT_EQ(media_description->streams().size(), 1u); - media_description->mutable_streams()[0].ssrcs[0] = 4404; - - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kSsrcs, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, HeaderExtensionAdded) { - auto pc = CreatePeerConnection(); - pc->AddVideoTrack("video_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - // VLA is off by default, id=42 should be unused. - media_description->AddRtpHeaderExtension( - {RtpExtension::kVideoLayersAllocationUri, 42}); - - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kRtpHeaderExtensionAdded, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, HeaderExtensionRemoved) { - auto pc = CreatePeerConnection(); - pc->AddVideoTrack("video_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - media_description->ClearRtpHeaderExtensions(); - - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kRtpHeaderExtensionRemoved, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, HeaderExtensionModified) { - auto pc = CreatePeerConnection(); - pc->AddVideoTrack("video_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - auto extensions = media_description->rtp_header_extensions(); - ASSERT_GT(extensions.size(), 0u); - extensions[0].id = 42; // id=42 should be unused. - media_description->set_rtp_header_extensions(extensions); - - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kRtpHeaderExtensionModified, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, PayloadTypeChanged) { - auto pc = CreatePeerConnection(); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - auto codecs = media_description->codecs(); - ASSERT_GT(codecs.size(), 0u); - codecs[0].id = 19; // IANA reserved payload type, should not conflict. - media_description->set_codecs(codecs); - - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kPayloadTypes, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, AudioCodecsReordered) { - auto pc = CreatePeerConnection(); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - auto codecs = media_description->codecs(); - ASSERT_GT(codecs.size(), 1u); - std::swap(codecs[0], codecs[1]); - media_description->set_codecs(codecs); - - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kAudioCodecsReordered, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, VideoCodecsReordered) { - auto pc = CreatePeerConnection(); - pc->AddVideoTrack("video_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - auto codecs = media_description->codecs(); - ASSERT_GT(codecs.size(), 1u); - std::swap(codecs[0], codecs[1]); - media_description->set_codecs(codecs); - - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kVideoCodecsReordered, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, AudioCodecsFmtp) { - auto pc = CreatePeerConnection(); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - auto codecs = media_description->codecs(); - ASSERT_GT(codecs.size(), 0u); - codecs[0].params["dont"] = "munge"; - media_description->set_codecs(codecs); - - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kAudioCodecsFmtp, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, VideoCodecsFmtp) { - auto pc = CreatePeerConnection(); - pc->AddVideoTrack("video_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - auto codecs = media_description->codecs(); - ASSERT_GT(codecs.size(), 0u); - codecs[0].params["dont"] = "munge"; - media_description->set_codecs(codecs); - - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kVideoCodecsFmtp, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, AudioCodecsRtcpFb) { - auto pc = CreatePeerConnection(); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - auto codecs = media_description->codecs(); - ASSERT_GT(codecs.size(), 0u); - codecs[0].feedback_params.Add({"dont", "munge"}); - media_description->set_codecs(codecs); - - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kAudioCodecsRtcpFb, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, AudioCodecsRtcpFbNack) { - auto pc = CreatePeerConnection(); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - auto codecs = media_description->codecs(); - ASSERT_GT(codecs.size(), 0u); - codecs[0].feedback_params.Add(FeedbackParam("nack")); - media_description->set_codecs(codecs); - - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kAudioCodecsRtcpFbAudioNack, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, AudioCodecsRtcpFbRrtr) { - auto pc = CreatePeerConnection(); - pc->AddAudioTrack("audio_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - auto codecs = media_description->codecs(); - ASSERT_GT(codecs.size(), 0u); - codecs[0].feedback_params.Add(FeedbackParam("rrtr")); - media_description->set_codecs(codecs); - - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kAudioCodecsRtcpFbRrtr, 1))); -} - -TEST_F(SdpOfferAnswerMungingTest, VideoCodecsRtcpFb) { - auto pc = CreatePeerConnection(); - pc->AddVideoTrack("video_track", {}); - - auto offer = pc->CreateOffer(); - auto& contents = offer->description()->contents(); - ASSERT_EQ(contents.size(), 1u); - auto* media_description = contents[0].media_description(); - ASSERT_TRUE(media_description); - auto codecs = media_description->codecs(); - ASSERT_GT(codecs.size(), 0u); - codecs[0].feedback_params.Add({"dont", "munge"}); - media_description->set_codecs(codecs); - - RTCError error; - EXPECT_TRUE(pc->SetLocalDescription(std::move(offer), &error)); - EXPECT_THAT( - metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), - ElementsAre(Pair(SdpMungingType::kVideoCodecsRtcpFb, 1))); -} - } // namespace webrtc From c75b14092e4073d5bce4725afdc68424e861291b Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Tue, 6 May 2025 09:06:39 -0700 Subject: [PATCH 053/292] Revert "[android] Allow to configure field trials per PeerConnectionFactory" MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This reverts commit 799d64b003ef0e23814efb0f09d15e0fabdf89f6. Reason for revert: fallback to global field trials probably is incorrect, causing PeerConnectionFactory be misconfigured. Bug: webrtc:369904700, webrtc:413413572 Original change's description: > [android] Allow to configure field trials per PeerConnectionFactory > > Provide webrtc::Environment to create AudioProcessing and > AudioDeviceModule through java layer > > Bug: webrtc:369904700, webrtc:413413572 > Change-Id: I79454ceb9706fb9c79c995dc7ce8d65924ff84d0 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/388620 > Reviewed-by: Zoé Lepaul > Commit-Queue: Danil Chapovalov > Cr-Commit-Position: refs/heads/main@{#44520} Bug: webrtc:369904700, webrtc:413413572 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I69beefb2af090d026ad4e75fa025754d9a2cda81 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390100 Owners-Override: Danil Chapovalov Bot-Commit: rubber-stamper@appspot.gserviceaccount.com Commit-Queue: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#44525} --- sdk/android/BUILD.gn | 30 ----------- .../org/webrtc/AudioProcessingFactory.java | 11 +--- sdk/android/api/org/webrtc/Environment.java | 53 ------------------- .../api/org/webrtc/PeerConnectionFactory.java | 51 ++++++++---------- .../org/webrtc/audio/AudioDeviceModule.java | 9 +--- .../org/webrtc/PeerConnectionFactoryTest.java | 27 ---------- sdk/android/src/jni/environment.cc | 40 -------------- .../src/jni/pc/peer_connection_factory.cc | 11 ++-- 8 files changed, 26 insertions(+), 206 deletions(-) delete mode 100644 sdk/android/api/org/webrtc/Environment.java delete mode 100644 sdk/android/src/jni/environment.cc diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn index 0d0f973218..d9ca1e008b 100644 --- a/sdk/android/BUILD.gn +++ b/sdk/android/BUILD.gn @@ -117,7 +117,6 @@ if (is_android) { ":base_jni", ":builtin_audio_codecs_jni", ":default_video_codec_factory_jni", - ":environment_jni", ":java_audio_device_module_jni", ":peerconnection_jni", ":video_jni", @@ -183,15 +182,6 @@ if (is_android) { ] } - rtc_android_library("environment_java") { - visibility = [ "*" ] - sources = [ "api/org/webrtc/Environment.java" ] - deps = [ - ":generated_environment_jni_java", - "//third_party/androidx:androidx_annotation_annotation_java", - ] - } - rtc_android_library("audio_api_java") { visibility = [ "*" ] sources = [ @@ -340,7 +330,6 @@ if (is_android) { ":base_java", ":builtin_audio_codecs_java", ":default_video_codec_factory_java", - ":environment_java", ":generated_base_jni_java", ":generated_peerconnection_jni_java", @@ -627,18 +616,6 @@ if (current_os == "linux" || is_android) { ] } - rtc_library("environment_jni") { - sources = [ "src/jni/environment.cc" ] - deps = [ - ":generated_environment_jni", - ":native_api_jni", - "../../api:field_trials", - "../../api/environment", - "../../api/environment:environment_factory", - "//third_party/jni_zero", - ] - } - rtc_library("builtin_audio_codecs_jni") { visibility = [ "*" ] allow_poison = [ "audio_codecs" ] @@ -828,7 +805,6 @@ if (current_os == "linux" || is_android) { "../../api/audio:builtin_audio_processing_builder", "../../api/audio_codecs:audio_codecs_api", "../../api/crypto:options", - "../../api/environment", "../../api/neteq:neteq_api", "../../api/rtc_event_log:rtc_event_log_factory", "../../api/task_queue:default_task_queue_factory", @@ -1343,12 +1319,6 @@ if (current_os == "linux" || is_android) { jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h" } - generate_jni("generated_environment_jni") { - sources = [ "api/org/webrtc/Environment.java" ] - namespace = "webrtc::jni" - jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h" - } - generate_jni("generated_metrics_jni") { sources = [ "api/org/webrtc/Metrics.java" ] namespace = "webrtc::jni" diff --git a/sdk/android/api/org/webrtc/AudioProcessingFactory.java b/sdk/android/api/org/webrtc/AudioProcessingFactory.java index e0aa25b87f..bd8fdb8989 100644 --- a/sdk/android/api/org/webrtc/AudioProcessingFactory.java +++ b/sdk/android/api/org/webrtc/AudioProcessingFactory.java @@ -16,14 +16,5 @@ public interface AudioProcessingFactory { * Dynamically allocates a webrtc::AudioProcessing instance and returns a pointer to it. * The caller takes ownership of the object. */ - // TODO: bugs.webrtc.org/369904700 - Remove when implementations switch to - // another variant. - @Deprecated - public default long createNative() { - return 0; - } - - public default long createNative(long webrtcEnvRef) { - return createNative(); - } + public long createNative(); } diff --git a/sdk/android/api/org/webrtc/Environment.java b/sdk/android/api/org/webrtc/Environment.java deleted file mode 100644 index 88cef4f6da..0000000000 --- a/sdk/android/api/org/webrtc/Environment.java +++ /dev/null @@ -1,53 +0,0 @@ -/* - * Copyright 2025 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -package org.webrtc; - -import androidx.annotation.Nullable; - -/** Wrapper for the webrtc::Environment native class. */ -public final class Environment implements AutoCloseable { - private final long webrtcEnv; - - public static class Builder { - Builder setFieldTrials(String fieldTrials) { - this.fieldTrials = fieldTrials; - return this; - } - - Environment build() { - return new Environment(this.fieldTrials); - } - - private @Nullable String fieldTrials; - } - - public static Builder builder() { - return new Builder(); - } - - /** Returns non-owning non-null native pointer to the webrtc::Environment */ - public long ref() { - return webrtcEnv; - } - - @Override - public void close() { - nativeFree(webrtcEnv); - } - - private Environment(@Nullable String fieldTrials) { - this.webrtcEnv = nativeCreate(fieldTrials); - } - - private static native long nativeCreate(@Nullable String fieldTrials); - - private static native void nativeFree(long webrtcEnv); -} diff --git a/sdk/android/api/org/webrtc/PeerConnectionFactory.java b/sdk/android/api/org/webrtc/PeerConnectionFactory.java index eac3bb02a0..2b2ea155c1 100644 --- a/sdk/android/api/org/webrtc/PeerConnectionFactory.java +++ b/sdk/android/api/org/webrtc/PeerConnectionFactory.java @@ -15,6 +15,9 @@ import androidx.annotation.Nullable; import java.util.List; import org.webrtc.Logging.Severity; +import org.webrtc.MediaStreamTrack; +import org.webrtc.PeerConnection; +import org.webrtc.RtpCapabilities; import org.webrtc.audio.AudioDeviceModule; import org.webrtc.audio.JavaAudioDeviceModule; @@ -163,7 +166,6 @@ boolean getDisableNetworkMonitor() { public static class Builder { @Nullable private Options options; - private Environment.Builder envBuilder = Environment.builder(); @Nullable private AudioDeviceModule audioDeviceModule; private AudioEncoderFactoryFactory audioEncoderFactoryFactory = new BuiltinAudioEncoderFactoryFactory(); @@ -184,11 +186,6 @@ public Builder setOptions(Options options) { return this; } - public Builder setFieldTrials(String fieldTrials) { - envBuilder.setFieldTrials(fieldTrials); - return this; - } - public Builder setAudioDeviceModule(AudioDeviceModule audioDeviceModule) { this.audioDeviceModule = audioDeviceModule; return this; @@ -264,30 +261,24 @@ public Builder setNetEqFactoryFactory(NetEqFactoryFactory neteqFactoryFactory) { public PeerConnectionFactory createPeerConnectionFactory() { checkInitializeHasBeenCalled(); - try (Environment env = envBuilder.build()) { - if (audioDeviceModule == null) { - audioDeviceModule = JavaAudioDeviceModule.builder(ContextUtils.getApplicationContext()) - .createAudioDeviceModule(); - } - return nativeCreatePeerConnectionFactory( - ContextUtils.getApplicationContext(), - options, - env.ref(), - audioDeviceModule.getNative(env.ref()), - audioEncoderFactoryFactory.createNativeAudioEncoderFactory(), - audioDecoderFactoryFactory.createNativeAudioDecoderFactory(), - videoEncoderFactory, - videoDecoderFactory, - audioProcessingFactory == null ? 0 : audioProcessingFactory.createNative(env.ref()), - fecControllerFactoryFactory == null ? 0 : fecControllerFactoryFactory.createNative(), - networkControllerFactoryFactory == null - ? 0 - : networkControllerFactoryFactory.createNativeNetworkControllerFactory(), - networkStatePredictorFactoryFactory == null - ? 0 - : networkStatePredictorFactoryFactory.createNativeNetworkStatePredictorFactory(), - neteqFactoryFactory == null ? 0 : neteqFactoryFactory.createNativeNetEqFactory()); + if (audioDeviceModule == null) { + audioDeviceModule = JavaAudioDeviceModule.builder(ContextUtils.getApplicationContext()) + .createAudioDeviceModule(); } + return nativeCreatePeerConnectionFactory(ContextUtils.getApplicationContext(), options, + audioDeviceModule.getNativeAudioDeviceModulePointer(), + audioEncoderFactoryFactory.createNativeAudioEncoderFactory(), + audioDecoderFactoryFactory.createNativeAudioDecoderFactory(), videoEncoderFactory, + videoDecoderFactory, + audioProcessingFactory == null ? 0 : audioProcessingFactory.createNative(), + fecControllerFactoryFactory == null ? 0 : fecControllerFactoryFactory.createNative(), + networkControllerFactoryFactory == null + ? 0 + : networkControllerFactoryFactory.createNativeNetworkControllerFactory(), + networkStatePredictorFactoryFactory == null + ? 0 + : networkStatePredictorFactoryFactory.createNativeNetworkStatePredictorFactory(), + neteqFactoryFactory == null ? 0 : neteqFactoryFactory.createNativeNetEqFactory()); } } @@ -613,7 +604,7 @@ private void onSignalingThreadReady() { private static native void nativeStopInternalTracingCapture(); private static native PeerConnectionFactory nativeCreatePeerConnectionFactory(Context context, - Options options, long webrtcEnvRef, long nativeAudioDeviceModule, long audioEncoderFactory, + Options options, long nativeAudioDeviceModule, long audioEncoderFactory, long audioDecoderFactory, VideoEncoderFactory encoderFactory, VideoDecoderFactory decoderFactory, long nativeAudioProcessor, long nativeFecControllerFactory, long nativeNetworkControllerFactory, diff --git a/sdk/android/api/org/webrtc/audio/AudioDeviceModule.java b/sdk/android/api/org/webrtc/audio/AudioDeviceModule.java index ba5097d6c8..5a0bf5c74d 100644 --- a/sdk/android/api/org/webrtc/audio/AudioDeviceModule.java +++ b/sdk/android/api/org/webrtc/audio/AudioDeviceModule.java @@ -22,14 +22,7 @@ public interface AudioDeviceModule { * Returns a C++ pointer to a webrtc::AudioDeviceModule. Caller does _not_ take ownership and * lifetime is handled through the release() call. */ - @Deprecated - default long getNativeAudioDeviceModulePointer() { - return 0; - } - - public default long getNative(long webrtcEnvRef) { - return getNativeAudioDeviceModulePointer(); - } + long getNativeAudioDeviceModulePointer(); /** * Release resources for this AudioDeviceModule, including native resources. The object should not diff --git a/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionFactoryTest.java b/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionFactoryTest.java index 1e36fef760..a28b9c25ff 100644 --- a/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionFactoryTest.java +++ b/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionFactoryTest.java @@ -62,31 +62,4 @@ public void testInitializeWithTracerAndShutdown() { .createInitializationOptions()); PeerConnectionFactory.shutdownInternalTracer(); } - - // Tests that the JNI glue between Java and C++ does not crash when creating a - // PeerConnectionFactory. - @Test - @SmallTest - public void testCreation() throws Exception { - PeerConnectionFactory.initialize( - PeerConnectionFactory.InitializationOptions.builder( - InstrumentationRegistry.getTargetContext()) - .setNativeLibraryName(TestConstants.NATIVE_LIBRARY) - .createInitializationOptions()); - - PeerConnectionFactory factory = PeerConnectionFactory.builder().createPeerConnectionFactory(); - } - - @Test - @SmallTest - public void testCreationWithFieldTrials() throws Exception { - PeerConnectionFactory.initialize( - PeerConnectionFactory.InitializationOptions.builder( - InstrumentationRegistry.getTargetContext()) - .setNativeLibraryName(TestConstants.NATIVE_LIBRARY) - .createInitializationOptions()); - - PeerConnectionFactory factory = - PeerConnectionFactory.builder().setFieldTrials("").createPeerConnectionFactory(); - } } diff --git a/sdk/android/src/jni/environment.cc b/sdk/android/src/jni/environment.cc deleted file mode 100644 index adf5fbca20..0000000000 --- a/sdk/android/src/jni/environment.cc +++ /dev/null @@ -1,40 +0,0 @@ -/* - * Copyright 2025 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ -#include "api/environment/environment.h" - -#include - -#include "api/environment/environment_factory.h" -#include "api/field_trials.h" -#include "sdk/android/generated_environment_jni/Environment_jni.h" -#include "sdk/android/native_api/jni/java_types.h" -#include "third_party/jni_zero/jni_zero.h" - -namespace webrtc { -namespace jni { - -void JNI_Environment_Free(JNIEnv* /*env*/, jlong webrtc_env) { - delete reinterpret_cast(webrtc_env); -} - -jlong JNI_Environment_Create( - JNIEnv* jni, - const jni_zero::JavaParamRef& j_field_trials) { - std::unique_ptr field_trials; - if (!j_field_trials.is_null()) { - field_trials = - FieldTrials::CreateNoGlobal(JavaToNativeString(jni, j_field_trials)); - } - return NativeToJavaPointer( - new Environment(CreateEnvironment(std::move(field_trials)))); -} - -} // namespace jni -} // namespace webrtc diff --git a/sdk/android/src/jni/pc/peer_connection_factory.cc b/sdk/android/src/jni/pc/peer_connection_factory.cc index 80b92046c4..f05259b478 100644 --- a/sdk/android/src/jni/pc/peer_connection_factory.cc +++ b/sdk/android/src/jni/pc/peer_connection_factory.cc @@ -29,7 +29,6 @@ #include "api/audio_codecs/audio_encoder_factory.h" #include "api/audio_options.h" #include "api/enable_media.h" -#include "api/environment/environment.h" #include "api/fec_controller.h" #include "api/media_stream_interface.h" #include "api/neteq/neteq_factory.h" @@ -257,7 +256,6 @@ ScopedJavaLocalRef CreatePeerConnectionFactoryForJava( JNIEnv* jni, const jni_zero::JavaParamRef& jcontext, const jni_zero::JavaParamRef& joptions, - const Environment& env, scoped_refptr audio_device_module, scoped_refptr audio_encoder_factory, scoped_refptr audio_decoder_factory, @@ -294,8 +292,8 @@ ScopedJavaLocalRef CreatePeerConnectionFactoryForJava( JavaToNativePeerConnectionFactoryOptions(jni, joptions); PeerConnectionFactoryDependencies dependencies; - dependencies.env = env; - dependencies.socket_factory = socket_server.get(); + // TODO(bugs.webrtc.org/13145): Also add socket_server.get() to the + // dependencies. dependencies.network_thread = network_thread.get(); dependencies.worker_thread = worker_thread.get(); dependencies.signaling_thread = signaling_thread.get(); @@ -349,7 +347,6 @@ JNI_PeerConnectionFactory_CreatePeerConnectionFactory( JNIEnv* jni, const jni_zero::JavaParamRef& jcontext, const jni_zero::JavaParamRef& joptions, - jlong webrtc_env_ref, jlong native_audio_device_module, jlong native_audio_encoder_factory, jlong native_audio_decoder_factory, @@ -360,12 +357,10 @@ JNI_PeerConnectionFactory_CreatePeerConnectionFactory( jlong native_network_controller_factory, jlong native_network_state_predictor_factory, jlong native_neteq_factory) { - const Environment* env = reinterpret_cast(webrtc_env_ref); - RTC_CHECK(env != nullptr); scoped_refptr audio_processor( reinterpret_cast(native_audio_processor)); return CreatePeerConnectionFactoryForJava( - jni, jcontext, joptions, *env, + jni, jcontext, joptions, scoped_refptr( reinterpret_cast(native_audio_device_module)), TakeOwnershipOfRefPtr(native_audio_encoder_factory), From d66b2f9f60329e08046c0c9e021f0bc967e9a3bd Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Thu, 10 Apr 2025 13:13:27 -0700 Subject: [PATCH 054/292] IWYU modules/video_coding/codecs/h264 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit using find modules/video_coding/codecs/h264 -name "*.h" -o -name "*.cc" | | xargs tools_webrtc/iwyu/apply-include-cleaner followed by tools_webrtc/gn_check_autofix.py -C out/Default/ and git cl format Manual changes to move the includes inside the RTC_USE_H264 define and wrap libavcodec/libavformat inside extern "C". Some RTC_USE_H264 includes no longer need to be guarded as the files which are included are fully guarded by that define. BUG=webrtc:42226242 Change-Id: I3f49028be95e984526d96372acffa3ef21127333 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/385820 Reviewed-by: Sergey Silkin Commit-Queue: Philipp Hancke Reviewed-by: Erik Språng Cr-Commit-Position: refs/heads/main@{#44526} --- modules/video_coding/codecs/h264/h264.cc | 13 ++++++---- .../codecs/h264/h264_color_space.cc | 7 +++++ .../codecs/h264/h264_decoder_impl.cc | 23 ++++++++++++++++ .../codecs/h264/h264_decoder_impl.h | 4 +++ .../codecs/h264/h264_encoder_impl.cc | 26 ++++++++++++++++--- .../codecs/h264/h264_encoder_impl.h | 12 +++++++-- .../codecs/h264/h264_encoder_impl_unittest.cc | 3 +++ .../codecs/h264/h264_simulcast_unittest.cc | 5 ++++ .../codecs/h264/include/h264_globals.h | 3 +-- .../codecs/h264/test/h264_impl_unittest.cc | 4 +-- 10 files changed, 85 insertions(+), 15 deletions(-) diff --git a/modules/video_coding/codecs/h264/h264.cc b/modules/video_coding/codecs/h264/h264.cc index ffee2af7f5..396c6847f4 100644 --- a/modules/video_coding/codecs/h264/h264.cc +++ b/modules/video_coding/codecs/h264/h264.cc @@ -14,19 +14,22 @@ #include #include #include +#include +#include "absl/base/nullability.h" #include "absl/container/inlined_vector.h" +#include "api/environment/environment.h" +#include "api/video_codecs/h264_profile_level_id.h" +#include "api/video_codecs/scalability_mode.h" #include "api/video_codecs/sdp_video_format.h" +#include "api/video_codecs/video_encoder.h" #include "media/base/media_constants.h" -#include "rtc_base/trace_event.h" - -#if defined(WEBRTC_USE_H264) #include "modules/video_coding/codecs/h264/h264_decoder_impl.h" #include "modules/video_coding/codecs/h264/h264_encoder_impl.h" -#endif - +#include "modules/video_coding/codecs/h264/include/h264_globals.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" +#include "rtc_base/trace_event.h" namespace webrtc { diff --git a/modules/video_coding/codecs/h264/h264_color_space.cc b/modules/video_coding/codecs/h264/h264_color_space.cc index 59921263e3..a93e5e3800 100644 --- a/modules/video_coding/codecs/h264/h264_color_space.cc +++ b/modules/video_coding/codecs/h264/h264_color_space.cc @@ -15,6 +15,13 @@ #include "modules/video_coding/codecs/h264/h264_color_space.h" +extern "C" { +#include +#include +} + +#include "api/video/color_space.h" + namespace webrtc { ColorSpace ExtractH264ColorSpace(AVCodecContext* codec) { diff --git a/modules/video_coding/codecs/h264/h264_decoder_impl.cc b/modules/video_coding/codecs/h264/h264_decoder_impl.cc index f3737a1edd..b9f81e5a67 100644 --- a/modules/video_coding/codecs/h264/h264_decoder_impl.cc +++ b/modules/video_coding/codecs/h264/h264_decoder_impl.cc @@ -18,17 +18,40 @@ extern "C" { #include +#include +#include +#include #include +#include +#include +#include #include +#include } // extern "C" #include +#include +#include +#include #include #include +#include +#include "api/scoped_refptr.h" #include "api/video/color_space.h" +#include "api/video/encoded_image.h" #include "api/video/i010_buffer.h" +#include "api/video/i210_buffer.h" +#include "api/video/i410_buffer.h" #include "api/video/i420_buffer.h" +#include "api/video/i422_buffer.h" +#include "api/video/i444_buffer.h" +#include "api/video/render_resolution.h" +#include "api/video/video_codec_type.h" +#include "api/video/video_frame.h" +#include "api/video/video_frame_buffer.h" +#include "api/video/video_rotation.h" +#include "api/video_codecs/video_decoder.h" #include "common_video/include/video_frame_buffer.h" #include "modules/video_coding/codecs/h264/h264_color_space.h" #include "modules/video_coding/include/video_error_codes.h" diff --git a/modules/video_coding/codecs/h264/h264_decoder_impl.h b/modules/video_coding/codecs/h264/h264_decoder_impl.h index e9da6a7a0d..7446d8fe00 100644 --- a/modules/video_coding/codecs/h264/h264_decoder_impl.h +++ b/modules/video_coding/codecs/h264/h264_decoder_impl.h @@ -37,10 +37,14 @@ extern "C" { #include +#include } // extern "C" +#include #include +#include "api/video/encoded_image.h" +#include "api/video_codecs/video_decoder.h" #include "common_video/h264/h264_bitstream_parser.h" #include "common_video/include/video_frame_buffer_pool.h" #include "modules/video_coding/codecs/h264/include/h264.h" diff --git a/modules/video_coding/codecs/h264/h264_encoder_impl.cc b/modules/video_coding/codecs/h264/h264_encoder_impl.cc index 7792d53e32..19d00a3372 100644 --- a/modules/video_coding/codecs/h264/h264_encoder_impl.cc +++ b/modules/video_coding/codecs/h264/h264_encoder_impl.cc @@ -17,24 +17,42 @@ #include "modules/video_coding/codecs/h264/h264_encoder_impl.h" #include +#include +#include +#include #include #include #include - -#include "absl/strings/match.h" +#include + +#include "api/environment/environment.h" +#include "api/scoped_refptr.h" +#include "api/units/data_rate.h" +#include "api/video/encoded_image.h" +#include "api/video/i420_buffer.h" +#include "api/video/video_bitrate_allocation.h" +#include "api/video/video_bitrate_allocator.h" #include "api/video/video_codec_constants.h" +#include "api/video/video_codec_type.h" +#include "api/video/video_frame.h" +#include "api/video/video_frame_buffer.h" +#include "api/video/video_frame_type.h" #include "api/video_codecs/scalability_mode.h" +#include "api/video_codecs/video_codec.h" +#include "api/video_codecs/video_encoder.h" #include "common_video/libyuv/include/webrtc_libyuv.h" +#include "modules/video_coding/codecs/h264/include/h264.h" +#include "modules/video_coding/codecs/h264/include/h264_globals.h" +#include "modules/video_coding/codecs/interface/common_constants.h" #include "modules/video_coding/include/video_codec_interface.h" #include "modules/video_coding/include/video_error_codes.h" #include "modules/video_coding/svc/create_scalability_structure.h" +#include "modules/video_coding/svc/scalable_video_controller.h" #include "modules/video_coding/utility/simulcast_rate_allocator.h" #include "modules/video_coding/utility/simulcast_utility.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" -#include "rtc_base/time_utils.h" #include "system_wrappers/include/metrics.h" -#include "third_party/libyuv/include/libyuv/convert.h" #include "third_party/libyuv/include/libyuv/scale.h" #include "third_party/openh264/src/codec/api/wels/codec_api.h" #include "third_party/openh264/src/codec/api/wels/codec_app_def.h" diff --git a/modules/video_coding/codecs/h264/h264_encoder_impl.h b/modules/video_coding/codecs/h264/h264_encoder_impl.h index 2d4a95311d..ea5feb75e1 100644 --- a/modules/video_coding/codecs/h264/h264_encoder_impl.h +++ b/modules/video_coding/codecs/h264/h264_encoder_impl.h @@ -21,19 +21,27 @@ #error "See: bugs.webrtc.org/9213#c13." #endif +#include +#include #include +#include #include #include "absl/container/inlined_vector.h" -#include "api/transport/rtp/dependency_descriptor.h" +#include "api/environment/environment.h" +#include "api/scoped_refptr.h" +#include "api/video/encoded_image.h" #include "api/video/i420_buffer.h" #include "api/video/video_codec_constants.h" +#include "api/video/video_frame.h" +#include "api/video/video_frame_type.h" #include "api/video_codecs/scalability_mode.h" +#include "api/video_codecs/video_codec.h" #include "api/video_codecs/video_encoder.h" #include "common_video/h264/h264_bitstream_parser.h" #include "modules/video_coding/codecs/h264/include/h264.h" +#include "modules/video_coding/codecs/h264/include/h264_globals.h" #include "modules/video_coding/svc/scalable_video_controller.h" -#include "modules/video_coding/utility/quality_scaler.h" #include "third_party/openh264/src/codec/api/wels/codec_app_def.h" class ISVCEncoder; diff --git a/modules/video_coding/codecs/h264/h264_encoder_impl_unittest.cc b/modules/video_coding/codecs/h264/h264_encoder_impl_unittest.cc index e714e277e8..2f3dfaf1a6 100644 --- a/modules/video_coding/codecs/h264/h264_encoder_impl_unittest.cc +++ b/modules/video_coding/codecs/h264/h264_encoder_impl_unittest.cc @@ -12,7 +12,10 @@ #include "modules/video_coding/codecs/h264/h264_encoder_impl.h" #include "api/environment/environment_factory.h" +#include "api/video/video_codec_type.h" +#include "api/video_codecs/video_codec.h" #include "api/video_codecs/video_encoder.h" +#include "modules/video_coding/codecs/h264/include/h264_globals.h" #include "modules/video_coding/include/video_error_codes.h" #include "test/gtest.h" diff --git a/modules/video_coding/codecs/h264/h264_simulcast_unittest.cc b/modules/video_coding/codecs/h264/h264_simulcast_unittest.cc index b9ea324ca2..39443fcdcc 100644 --- a/modules/video_coding/codecs/h264/h264_simulcast_unittest.cc +++ b/modules/video_coding/codecs/h264/h264_simulcast_unittest.cc @@ -9,11 +9,16 @@ */ #include +#include +#include "api/environment/environment.h" #include "api/test/create_simulcast_test_fixture.h" #include "api/test/simulcast_test_fixture.h" #include "api/test/video/function_video_decoder_factory.h" #include "api/test/video/function_video_encoder_factory.h" +#include "api/video_codecs/sdp_video_format.h" +#include "api/video_codecs/video_decoder_factory.h" +#include "api/video_codecs/video_encoder_factory.h" #include "modules/video_coding/codecs/h264/include/h264.h" #include "test/gtest.h" diff --git a/modules/video_coding/codecs/h264/include/h264_globals.h b/modules/video_coding/codecs/h264/include/h264_globals.h index dac36dd734..9b0b7b6b9e 100644 --- a/modules/video_coding/codecs/h264/include/h264_globals.h +++ b/modules/video_coding/codecs/h264/include/h264_globals.h @@ -14,11 +14,10 @@ #ifndef MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_GLOBALS_H_ #define MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_GLOBALS_H_ -#include +#include #include #include -#include "modules/video_coding/codecs/interface/common_constants.h" #include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc b/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc index 985224d07e..239aae6548 100644 --- a/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc +++ b/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc @@ -15,13 +15,13 @@ #include "api/video/color_space.h" #include "api/video/encoded_image.h" +#include "api/video/video_codec_type.h" #include "api/video/video_frame.h" +#include "api/video/video_frame_type.h" #include "api/video_codecs/video_codec.h" #include "api/video_codecs/video_decoder.h" #include "api/video_codecs/video_encoder.h" #include "common_video/libyuv/include/webrtc_libyuv.h" -#include "media/base/codec.h" -#include "media/base/media_constants.h" #include "modules/video_coding/codecs/h264/include/h264.h" #include "modules/video_coding/codecs/test/video_codec_unittest.h" #include "modules/video_coding/include/video_codec_interface.h" From ba48850b66809edb30f105175575a8f39b14bf7e Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Tue, 6 May 2025 09:55:14 -0700 Subject: [PATCH 055/292] Tweak codec vendor logging to include the "inconsistent" codec. Bug: chromium:413300860 Change-Id: Ib826fdcc7567c36ff00dddc345cdecbc39cb4fca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389445 Reviewed-by: Harald Alvestrand Reviewed-by: Evan Shrubsole Commit-Queue: Philipp Hancke Cr-Commit-Position: refs/heads/main@{#44527} --- media/base/codec_list.cc | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/media/base/codec_list.cc b/media/base/codec_list.cc index 14f8fbde54..27183a0279 100644 --- a/media/base/codec_list.cc +++ b/media/base/codec_list.cc @@ -36,8 +36,11 @@ RTCError CheckInputConsistency(const std::vector& codecs) { for (size_t i = 0; i < codecs.size(); i++) { const Codec& codec = codecs[i]; if (codec.id != Codec::kIdNotSet) { - bool inserted = pt_to_index.insert({codec.id, i}).second; - if (!inserted) { + auto [it, success] = pt_to_index.insert({codec.id, i}); + if (!success) { + RTC_LOG(LS_ERROR) << "Duplicate payload type in codec list, " << codec + << " and " << codecs[it->second] + << " have the same ID"; LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "Duplicate payload type in codec list"); } From 6a2608ecab1f3d7d56f9fc0f7254dd1b4dd1c79d Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Tue, 6 May 2025 13:03:34 -0700 Subject: [PATCH 056/292] Roll chromium_revision b377c950c8..3bac98a091 (1456285:1456565) Change log: https://chromium.googlesource.com/chromium/src/+log/b377c950c8..3bac98a091 Full diff: https://chromium.googlesource.com/chromium/src/+/b377c950c8..3bac98a091 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/135ccf38f8..6de05cbf0c * src/build: https://chromium.googlesource.com/chromium/src/build/+log/2021df387e..37f833c4f7 * src/buildtools: https://chromium.googlesource.com/chromium/src/buildtools/+log/b841ab22bc..1df2adafaa * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/00e20b50b2..19cbcf4234 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/c2ffcbf942..ca26dad53b * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/f2a66cfb2b..c01eb5725e * src/third_party/android_deps/autorolled/cipd: nb19wuIgamdmcgMY_6xHCGH3ph_tLVHkHYH33tKbup4C..TZHdcfZjA4uMdOqeaQ-7MG_Pi3P1vgW-fN5TrDzqfVsC * src/third_party/androidx/cipd: k1zV9FUHrRghPTZBejg81vzwei2vVTdjpM7GhDEXyxQC..maUv18JmcvP3nwKYXghGhozFkEce3pFDH6JtbSR0SqQC * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/417522f618..4a839855c5 * src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/6558e6a3ea..cff6ae8a80 * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/9e195f61ff..abf2a0d268 * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/fc9aac5739..f4cde1b052 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/725c5898d1..678f57506d DEPS diff: https://chromium.googlesource.com/chromium/src/+/b377c950c8..3bac98a091/DEPS No update to Clang. BUG=None Change-Id: I40c86c1553a26bfe1a2fd734b62f28ca96977b3f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390120 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44528} --- DEPS | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/DEPS b/DEPS index 0cc5f71e49..e4660b355d 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': 'b377c950c8162b7bdce9a93e9265ff3229061383', + 'chromium_revision': '3bac98a09145b27fec236e1b528e883f3fcea876', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,24 +73,24 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@135ccf38f87dfaf4726c3e6cf5f759d1c67bff96', + 'https://chromium.googlesource.com/chromium/src/base@6de05cbf0c0cf7af590bc79123cca396f28d017d', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@2021df387efef46ca51da8e3356402a706617368', + 'https://chromium.googlesource.com/chromium/src/build@37f833c4f7b05a39f9cff7029769a4861e7d0fb3', 'src/buildtools': - 'https://chromium.googlesource.com/chromium/src/buildtools@b841ab22bc4522fe4e72df46ca57abbe2fad8640', + 'https://chromium.googlesource.com/chromium/src/buildtools@1df2adafaa36e2bcff31e28cd814505382e04df6', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. 'src/examples/androidtests/third_party/gradle': { 'url': 'https://chromium.googlesource.com/external/github.com/gradle/gradle.git@f2d1fb54a951d8b11d25748e4711bec8d128d7e3', 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@00e20b50b20c5c33e6013c112dffb1bee277523c', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@19cbcf42345c6d240df1c758328c3484d2bd5e0b', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@c2ffcbf94259f1b0cccdfc674a2a100ddd068f0c', + 'https://chromium.googlesource.com/chromium/src/testing@ca26dad53be50e879e65286a00bfed4ef0852ce5', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@f2a66cfb2b7dc8b68cf55c6cc2cce12b6fb77bc2', + 'https://chromium.googlesource.com/chromium/src/third_party@c01eb5725e85c8bd4aaac98762e210d0328090ee', 'src/buildtools/linux64': { 'packages': [ @@ -323,7 +323,7 @@ deps = { 'src/third_party/clang-format/script': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@37f6e68a107df43b7d7e044fd36a13cbae3413f2', 'src/third_party/compiler-rt/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@6558e6a3eacafd269527cf2595e0e722f117c1e6', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@cff6ae8a80960ceca4d83cf6efa271222d2e7d85', 'src/third_party/libc++/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@a01c02c9d4acbdae3b7e8a2f3ee58579a9c29f96', 'src/third_party/libc++abi/src': @@ -484,7 +484,7 @@ deps = { 'src/third_party/breakpad/breakpad': 'https://chromium.googlesource.com/breakpad/breakpad.git@b319c12317824bfd0e92828db20bbfe0ec6e31f8', 'src/third_party/catapult': - 'https://chromium.googlesource.com/catapult.git@417522f6185fde0e4e0812bce00bca1277f286a5', + 'https://chromium.googlesource.com/catapult.git@4a839855c5e79d91689784e2c7284f3f6288123b', 'src/third_party/ced/src': { 'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5', }, @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@9e195f61ff47384d4014fb7dcf9808f15e3ec711', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@abf2a0d268dc589f991abd9208f73523e2e14123', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'fc9aac573945f2da6d05bf1904d500e6d5cdf18d', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'f4cde1b052fd591a5c50fe11dbcda61c51c4cccf', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@725c5898d1c0775d1468c60252df9709a48fd1c8', + 'https://chromium.googlesource.com/chromium/src/tools@678f57506d545d668ad4c34e89dfd616c6e6d4c3', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'k1zV9FUHrRghPTZBejg81vzwei2vVTdjpM7GhDEXyxQC', + 'version': 'maUv18JmcvP3nwKYXghGhozFkEce3pFDH6JtbSR0SqQC', }, ], 'condition': 'checkout_android and non_git_source', @@ -827,7 +827,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_deps/autorolled', - 'version': 'nb19wuIgamdmcgMY_6xHCGH3ph_tLVHkHYH33tKbup4C', + 'version': 'TZHdcfZjA4uMdOqeaQ-7MG_Pi3P1vgW-fN5TrDzqfVsC', }, ], 'condition': 'checkout_android and non_git_source', From a02f9fcedaf3ec73b05cc79c9e645a0ed0cceab0 Mon Sep 17 00:00:00 2001 From: webrtc-version-updater Date: Tue, 6 May 2025 21:11:12 -0700 Subject: [PATCH 057/292] Update WebRTC code version (2025-05-07T04:10:34). Bug: None Change-Id: Iaa8ba66bf5953f4d418b034d238aedada80a1820 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390182 Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Cr-Commit-Position: refs/heads/main@{#44529} --- call/version.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/call/version.cc b/call/version.cc index 789b9410c6..93b1799f2e 100644 --- a/call/version.cc +++ b/call/version.cc @@ -13,7 +13,7 @@ namespace webrtc { // The timestamp is always in UTC. -const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-06T04:06:28"; +const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-07T04:10:34"; void LoadWebRTCVersionInRegister() { // Using volatile to instruct the compiler to not optimize `p` away even From d69d0808c37b4ea338e59a58837ef180023316a6 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Thu, 24 Apr 2025 08:00:29 +0000 Subject: [PATCH 058/292] Reland "Delete 3p base64 library" This is a reland of commit dc55fcf2828d1817336735dc8c07c446d1cb2dbf Safe for reland: Upstream fixed. Original change's description: > Delete 3p base64 library > > Bug: webrtc:42220265 > Change-Id: Ifc05d296013e55cb9d11e61f7cc18755e5f7fbf7 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/385461 > Auto-Submit: Evan Shrubsole > Commit-Queue: Mirko Bonadei > Reviewed-by: Mirko Bonadei > Cr-Commit-Position: refs/heads/main@{#44442} Bug: webrtc:42220265 Change-Id: Ife6c06e30d3e128f4e1ec6c7b29264fbe28d2b3e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390220 Commit-Queue: Evan Shrubsole Reviewed-by: Mirko Bonadei Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44530} --- rtc_base/third_party/base64/BUILD.gn | 38 - rtc_base/third_party/base64/LICENSE | 14 - rtc_base/third_party/base64/README.chromium | 12 - rtc_base/third_party/base64/base64.cc | 279 --- rtc_base/third_party/base64/base64.h | 135 -- .../third_party/base64/base64_unittest.cc | 1459 ------------ rtc_base/third_party/base64/test_base64.h | 2111 ----------------- 7 files changed, 4048 deletions(-) delete mode 100644 rtc_base/third_party/base64/BUILD.gn delete mode 100644 rtc_base/third_party/base64/LICENSE delete mode 100644 rtc_base/third_party/base64/README.chromium delete mode 100644 rtc_base/third_party/base64/base64.cc delete mode 100644 rtc_base/third_party/base64/base64.h delete mode 100644 rtc_base/third_party/base64/base64_unittest.cc delete mode 100644 rtc_base/third_party/base64/test_base64.h diff --git a/rtc_base/third_party/base64/BUILD.gn b/rtc_base/third_party/base64/BUILD.gn deleted file mode 100644 index fac16ff881..0000000000 --- a/rtc_base/third_party/base64/BUILD.gn +++ /dev/null @@ -1,38 +0,0 @@ -# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved. -# -# Use of this source code is governed by a BSD-style license -# that can be found in the ../../../LICENSE file in the root of the source -# tree. An additional intellectual property rights grant can be found -# in the file PATENTS. All contributing project authors may -# be found in the AUTHORS file in the root of the source tree. - -import("../../../webrtc.gni") - -rtc_library("base64") { - sources = [ - "base64.cc", - "base64.h", - ] - deps = [ - "../..:checks", - "../../system:rtc_export", - "//third_party/abseil-cpp/absl/strings:string_view", - ] -} - -if (rtc_include_tests) { - rtc_library("base64_unittest") { - testonly = true - sources = [ - "base64_unittest.cc", - "test_base64.h", - ] - deps = [ - ":base64", - "../..:logging", - "../../../test:test_main", - "../../../test:test_support", - "//third_party/abseil-cpp/absl/strings:string_view", - ] - } -} diff --git a/rtc_base/third_party/base64/LICENSE b/rtc_base/third_party/base64/LICENSE deleted file mode 100644 index 4ad21d1b71..0000000000 --- a/rtc_base/third_party/base64/LICENSE +++ /dev/null @@ -1,14 +0,0 @@ -//********************************************************************* -//* Base64 - a simple base64 encoder and decoder. -//* -//* Copyright (c) 1999, Bob Withers - bwit@pobox.com -//* -//* This code may be freely used for any purpose, either personal -//* or commercial, provided the authors copyright notice remains -//* intact. -//* -//* Enhancements by Stanley Yamane: -//* o reverse lookup table for the decode function -//* o reserve string buffer space in advance -//* -//********************************************************************* diff --git a/rtc_base/third_party/base64/README.chromium b/rtc_base/third_party/base64/README.chromium deleted file mode 100644 index 1f01511fee..0000000000 --- a/rtc_base/third_party/base64/README.chromium +++ /dev/null @@ -1,12 +0,0 @@ -Name: A simple base64 encoder and decoder -Short Name: base64 -URL: -Version: 0 -Date: 2018-06-20 -License: Custom license -License File: LICENSE -Security Critical: yes -Shipped: yes - -Description: -A simple base64 encoder and decoder diff --git a/rtc_base/third_party/base64/base64.cc b/rtc_base/third_party/base64/base64.cc deleted file mode 100644 index cc579ade91..0000000000 --- a/rtc_base/third_party/base64/base64.cc +++ /dev/null @@ -1,279 +0,0 @@ - -//********************************************************************* -//* Base64 - a simple base64 encoder and decoder. -//* -//* Copyright (c) 1999, Bob Withers - bwit@pobox.com -//* -//* This code may be freely used for any purpose, either personal -//* or commercial, provided the authors copyright notice remains -//* intact. -//* -//* Enhancements by Stanley Yamane: -//* o reverse lookup table for the decode function -//* o reserve string buffer space in advance -//* -//********************************************************************* - -#include "rtc_base/third_party/base64/base64.h" - -#include - -#include "absl/strings/string_view.h" -#include "rtc_base/checks.h" - -using std::vector; - -namespace webrtc { - -static const char kPad = '='; -static const unsigned char pd = 0xFD; // Padding -static const unsigned char sp = 0xFE; // Whitespace -static const unsigned char il = 0xFF; // Illegal base64 character - -const char Base64::Base64Table[] = - // 0000000000111111111122222222223333333333444444444455555555556666 - // 0123456789012345678901234567890123456789012345678901234567890123 - "ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789+/"; - -// Decode Table gives the index of any valid base64 character in the -// Base64 table -// 65 == A, 97 == a, 48 == 0, 43 == +, 47 == / - -const unsigned char Base64::DecodeTable[] = { - // 0 1 2 3 4 5 6 7 8 9 - il, il, il, il, il, il, il, il, il, sp, // 0 - 9 - sp, sp, sp, sp, il, il, il, il, il, il, // 10 - 19 - il, il, il, il, il, il, il, il, il, il, // 20 - 29 - il, il, sp, il, il, il, il, il, il, il, // 30 - 39 - il, il, il, 62, il, il, il, 63, 52, 53, // 40 - 49 - 54, 55, 56, 57, 58, 59, 60, 61, il, il, // 50 - 59 - il, pd, il, il, il, 0, 1, 2, 3, 4, // 60 - 69 - 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, // 70 - 79 - 15, 16, 17, 18, 19, 20, 21, 22, 23, 24, // 80 - 89 - 25, il, il, il, il, il, il, 26, 27, 28, // 90 - 99 - 29, 30, 31, 32, 33, 34, 35, 36, 37, 38, // 100 - 109 - 39, 40, 41, 42, 43, 44, 45, 46, 47, 48, // 110 - 119 - 49, 50, 51, il, il, il, il, il, il, il, // 120 - 129 - il, il, il, il, il, il, il, il, il, il, // 130 - 139 - il, il, il, il, il, il, il, il, il, il, // 140 - 149 - il, il, il, il, il, il, il, il, il, il, // 150 - 159 - il, il, il, il, il, il, il, il, il, il, // 160 - 169 - il, il, il, il, il, il, il, il, il, il, // 170 - 179 - il, il, il, il, il, il, il, il, il, il, // 180 - 189 - il, il, il, il, il, il, il, il, il, il, // 190 - 199 - il, il, il, il, il, il, il, il, il, il, // 200 - 209 - il, il, il, il, il, il, il, il, il, il, // 210 - 219 - il, il, il, il, il, il, il, il, il, il, // 220 - 229 - il, il, il, il, il, il, il, il, il, il, // 230 - 239 - il, il, il, il, il, il, il, il, il, il, // 240 - 249 - il, il, il, il, il, il // 250 - 255 -}; - -bool Base64::IsBase64Char(char ch) { - return (('A' <= ch) && (ch <= 'Z')) || (('a' <= ch) && (ch <= 'z')) || - (('0' <= ch) && (ch <= '9')) || (ch == '+') || (ch == '/'); -} - -bool Base64::GetNextBase64Char(char ch, char* next_ch) { - if (next_ch == nullptr) { - return false; - } - const char* p = strchr(Base64Table, ch); - if (!p) - return false; - ++p; - *next_ch = (*p) ? *p : Base64Table[0]; - return true; -} - -bool Base64::IsBase64Encoded(absl::string_view str) { - for (size_t i = 0; i < str.size(); ++i) { - if (!IsBase64Char(str.at(i))) - return false; - } - return true; -} - -void Base64::EncodeFromArray(const void* data, - size_t len, - std::string* result) { - RTC_DCHECK(result); - result->clear(); - result->resize(((len + 2) / 3) * 4); - const unsigned char* byte_data = static_cast(data); - - unsigned char c; - size_t i = 0; - size_t dest_ix = 0; - while (i < len) { - c = (byte_data[i] >> 2) & 0x3f; - (*result)[dest_ix++] = Base64Table[c]; - - c = (byte_data[i] << 4) & 0x3f; - if (++i < len) { - c |= (byte_data[i] >> 4) & 0x0f; - } - (*result)[dest_ix++] = Base64Table[c]; - - if (i < len) { - c = (byte_data[i] << 2) & 0x3f; - if (++i < len) { - c |= (byte_data[i] >> 6) & 0x03; - } - (*result)[dest_ix++] = Base64Table[c]; - } else { - (*result)[dest_ix++] = kPad; - } - - if (i < len) { - c = byte_data[i] & 0x3f; - (*result)[dest_ix++] = Base64Table[c]; - ++i; - } else { - (*result)[dest_ix++] = kPad; - } - } -} - -size_t Base64::GetNextQuantum(DecodeFlags parse_flags, - bool illegal_pads, - const char* data, - size_t len, - size_t* dpos, - unsigned char qbuf[4], - bool* padded) { - size_t byte_len = 0, pad_len = 0, pad_start = 0; - for (; (byte_len < 4) && (*dpos < len); ++*dpos) { - qbuf[byte_len] = DecodeTable[static_cast(data[*dpos])]; - if ((il == qbuf[byte_len]) || (illegal_pads && (pd == qbuf[byte_len]))) { - if (parse_flags != DO_PARSE_ANY) - break; - // Ignore illegal characters - } else if (sp == qbuf[byte_len]) { - if (parse_flags == DO_PARSE_STRICT) - break; - // Ignore spaces - } else if (pd == qbuf[byte_len]) { - if (byte_len < 2) { - if (parse_flags != DO_PARSE_ANY) - break; - // Ignore unexpected padding - } else if (byte_len + pad_len >= 4) { - if (parse_flags != DO_PARSE_ANY) - break; - // Ignore extra pads - } else { - if (1 == ++pad_len) { - pad_start = *dpos; - } - } - } else { - if (pad_len > 0) { - if (parse_flags != DO_PARSE_ANY) - break; - // Ignore pads which are followed by data - pad_len = 0; - } - ++byte_len; - } - } - for (size_t i = byte_len; i < 4; ++i) { - qbuf[i] = 0; - } - if (4 == byte_len + pad_len) { - *padded = true; - } else { - *padded = false; - if (pad_len) { - // Roll back illegal padding - *dpos = pad_start; - } - } - return byte_len; -} - -bool Base64::DecodeFromArray(const char* data, - size_t len, - DecodeFlags flags, - std::string* result, - size_t* data_used) { - return DecodeFromArrayTemplate(data, len, flags, result, - data_used); -} - -bool Base64::DecodeFromArray(const char* data, - size_t len, - DecodeFlags flags, - vector* result, - size_t* data_used) { - return DecodeFromArrayTemplate>(data, len, flags, result, - data_used); -} - -bool Base64::DecodeFromArray(const char* data, - size_t len, - DecodeFlags flags, - vector* result, - size_t* data_used) { - return DecodeFromArrayTemplate>(data, len, flags, result, - data_used); -} - -template -bool Base64::DecodeFromArrayTemplate(const char* data, - size_t len, - DecodeFlags flags, - T* result, - size_t* data_used) { - RTC_DCHECK(result); - RTC_DCHECK_LE(flags, (DO_PARSE_MASK | DO_PAD_MASK | DO_TERM_MASK)); - - const DecodeFlags parse_flags = flags & DO_PARSE_MASK; - const DecodeFlags pad_flags = flags & DO_PAD_MASK; - const DecodeFlags term_flags = flags & DO_TERM_MASK; - RTC_DCHECK_NE(0, parse_flags); - RTC_DCHECK_NE(0, pad_flags); - RTC_DCHECK_NE(0, term_flags); - - result->clear(); - result->reserve(len); - - size_t dpos = 0; - bool success = true, padded; - unsigned char c, qbuf[4]; - while (dpos < len) { - size_t qlen = GetNextQuantum(parse_flags, (DO_PAD_NO == pad_flags), data, - len, &dpos, qbuf, &padded); - c = (qbuf[0] << 2) | ((qbuf[1] >> 4) & 0x3); - if (qlen >= 2) { - result->push_back(c); - c = ((qbuf[1] << 4) & 0xf0) | ((qbuf[2] >> 2) & 0xf); - if (qlen >= 3) { - result->push_back(c); - c = ((qbuf[2] << 6) & 0xc0) | qbuf[3]; - if (qlen >= 4) { - result->push_back(c); - c = 0; - } - } - } - if (qlen < 4) { - if ((DO_TERM_ANY != term_flags) && (0 != c)) { - success = false; // unused bits - } - if ((DO_PAD_YES == pad_flags) && !padded) { - success = false; // expected padding - } - break; - } - } - if ((DO_TERM_BUFFER == term_flags) && (dpos != len)) { - success = false; // unused chars - } - if (data_used) { - *data_used = dpos; - } - return success; -} - -} // namespace webrtc diff --git a/rtc_base/third_party/base64/base64.h b/rtc_base/third_party/base64/base64.h deleted file mode 100644 index 7a4b1e6209..0000000000 --- a/rtc_base/third_party/base64/base64.h +++ /dev/null @@ -1,135 +0,0 @@ - -//********************************************************************* -//* C_Base64 - a simple base64 encoder and decoder. -//* -//* Copyright (c) 1999, Bob Withers - bwit@pobox.com -//* -//* This code may be freely used for any purpose, either personal -//* or commercial, provided the authors copyright notice remains -//* intact. -//********************************************************************* - -#ifndef RTC_BASE_THIRD_PARTY_BASE64_BASE64_H_ -#define RTC_BASE_THIRD_PARTY_BASE64_BASE64_H_ - -#include -#include -#include -#include - -#include "absl/strings/string_view.h" - -namespace webrtc { - -class Base64 { - public: - enum DecodeOption { - DO_PARSE_STRICT = 1, // Parse only base64 characters - DO_PARSE_WHITE = 2, // Parse only base64 and whitespace characters - DO_PARSE_ANY = 3, // Parse all characters - DO_PARSE_MASK = 3, - - DO_PAD_YES = 4, // Padding is required - DO_PAD_ANY = 8, // Padding is optional - DO_PAD_NO = 12, // Padding is disallowed - DO_PAD_MASK = 12, - - DO_TERM_BUFFER = 16, // Must termiante at end of buffer - DO_TERM_CHAR = 32, // May terminate at any character boundary - DO_TERM_ANY = 48, // May terminate at a sub-character bit offset - DO_TERM_MASK = 48, - - // Strictest interpretation - DO_STRICT = DO_PARSE_STRICT | DO_PAD_YES | DO_TERM_BUFFER, - - DO_LAX = DO_PARSE_ANY | DO_PAD_ANY | DO_TERM_CHAR, - }; - typedef int DecodeFlags; - - static bool IsBase64Char(char ch); - - // Get the char next to the `ch` from the Base64Table. - // If the `ch` is the last one in the Base64Table then returns - // the first one from the table. - // Expects the `ch` be a base64 char. - // The result will be saved in `next_ch`. - // Returns true on success. - static bool GetNextBase64Char(char ch, char* next_ch); - - // Determines whether the given string consists entirely of valid base64 - // encoded characters. - static bool IsBase64Encoded(absl::string_view str); - - static void EncodeFromArray(const void* data, - size_t len, - std::string* result); - static bool DecodeFromArray(const char* data, - size_t len, - DecodeFlags flags, - std::string* result, - size_t* data_used); - static bool DecodeFromArray(const char* data, - size_t len, - DecodeFlags flags, - std::vector* result, - size_t* data_used); - static bool DecodeFromArray(const char* data, - size_t len, - DecodeFlags flags, - std::vector* result, - size_t* data_used); - - // Convenience Methods - static inline std::string Encode(absl::string_view data) { - std::string result; - EncodeFromArray(data.data(), data.size(), &result); - return result; - } - static inline std::string Decode(absl::string_view data, DecodeFlags flags) { - std::string result; - DecodeFromArray(data.data(), data.size(), flags, &result, nullptr); - return result; - } - static inline bool Decode(absl::string_view data, - DecodeFlags flags, - std::string* result, - size_t* data_used) { - return DecodeFromArray(data.data(), data.size(), flags, result, data_used); - } - static inline bool Decode(absl::string_view data, - DecodeFlags flags, - std::vector* result, - size_t* data_used) { - return DecodeFromArray(data.data(), data.size(), flags, result, data_used); - } - - private: - static const char Base64Table[]; - static const unsigned char DecodeTable[]; - - static size_t GetNextQuantum(DecodeFlags parse_flags, - bool illegal_pads, - const char* data, - size_t len, - size_t* dpos, - unsigned char qbuf[4], - bool* padded); - template - static bool DecodeFromArrayTemplate(const char* data, - size_t len, - DecodeFlags flags, - T* result, - size_t* data_used); -}; - -} // namespace webrtc - -// Re-export symbols from the webrtc namespace for backwards compatibility. -// TODO(bugs.webrtc.org/4222596): Remove once all references are updated. -#ifdef WEBRTC_ALLOW_DEPRECATED_NAMESPACES -namespace rtc { -using ::webrtc::Base64; -} // namespace rtc -#endif // WEBRTC_ALLOW_DEPRECATED_NAMESPACES - -#endif /* RTC_BASE_THIRD_PARTY_BASE64_BASE64_H_ */ diff --git a/rtc_base/third_party/base64/base64_unittest.cc b/rtc_base/third_party/base64/base64_unittest.cc deleted file mode 100644 index f753bf689f..0000000000 --- a/rtc_base/third_party/base64/base64_unittest.cc +++ /dev/null @@ -1,1459 +0,0 @@ -/* - * Copyright 2011 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "rtc_base/third_party/base64/base64.h" - -#include -#include - -#include -#include - -#include "rtc_base/logging.h" -#include "rtc_base/third_party/base64/test_base64.h" -#include "test/gtest.h" - -#ifdef WEBRTC_ALLOW_DEPRECATED_NAMESPACES -namespace rtc { -namespace { - -static struct { - size_t plain_length; - const char* plaintext; - const char* cyphertext; -} base64_tests[] = { - - // Basic bit patterns; - // values obtained with "echo -n '...' | uuencode -m test" - - {1, "\000", "AA=="}, - {1, "\001", "AQ=="}, - {1, "\002", "Ag=="}, - {1, "\004", "BA=="}, - {1, "\010", "CA=="}, - {1, "\020", "EA=="}, - {1, "\040", "IA=="}, - {1, "\100", "QA=="}, - {1, "\200", "gA=="}, - - {1, "\377", "/w=="}, - {1, "\376", "/g=="}, - {1, "\375", "/Q=="}, - {1, "\373", "+w=="}, - {1, "\367", "9w=="}, - {1, "\357", "7w=="}, - {1, "\337", "3w=="}, - {1, "\277", "vw=="}, - {1, "\177", "fw=="}, - {2, "\000\000", "AAA="}, - {2, "\000\001", "AAE="}, - {2, "\000\002", "AAI="}, - {2, "\000\004", "AAQ="}, - {2, "\000\010", "AAg="}, - {2, "\000\020", "ABA="}, - {2, "\000\040", "ACA="}, - {2, "\000\100", "AEA="}, - {2, "\000\200", "AIA="}, - {2, "\001\000", "AQA="}, - {2, "\002\000", "AgA="}, - {2, "\004\000", "BAA="}, - {2, "\010\000", "CAA="}, - {2, "\020\000", "EAA="}, - {2, "\040\000", "IAA="}, - {2, "\100\000", "QAA="}, - {2, "\200\000", "gAA="}, - - {2, "\377\377", "//8="}, - {2, "\377\376", "//4="}, - {2, "\377\375", "//0="}, - {2, "\377\373", "//s="}, - {2, "\377\367", "//c="}, - {2, "\377\357", "/+8="}, - {2, "\377\337", "/98="}, - {2, "\377\277", "/78="}, - {2, "\377\177", "/38="}, - {2, "\376\377", "/v8="}, - {2, "\375\377", "/f8="}, - {2, "\373\377", "+/8="}, - {2, "\367\377", "9/8="}, - {2, "\357\377", "7/8="}, - {2, "\337\377", "3/8="}, - {2, "\277\377", "v/8="}, - {2, "\177\377", "f/8="}, - - {3, "\000\000\000", "AAAA"}, - {3, "\000\000\001", "AAAB"}, - {3, "\000\000\002", "AAAC"}, - {3, "\000\000\004", "AAAE"}, - {3, "\000\000\010", "AAAI"}, - {3, "\000\000\020", "AAAQ"}, - {3, "\000\000\040", "AAAg"}, - {3, "\000\000\100", "AABA"}, - {3, "\000\000\200", "AACA"}, - {3, "\000\001\000", "AAEA"}, - {3, "\000\002\000", "AAIA"}, - {3, "\000\004\000", "AAQA"}, - {3, "\000\010\000", "AAgA"}, - {3, "\000\020\000", "ABAA"}, - {3, "\000\040\000", "ACAA"}, - {3, "\000\100\000", "AEAA"}, - {3, "\000\200\000", "AIAA"}, - {3, "\001\000\000", "AQAA"}, - {3, "\002\000\000", "AgAA"}, - {3, "\004\000\000", "BAAA"}, - {3, "\010\000\000", "CAAA"}, - {3, "\020\000\000", "EAAA"}, - {3, "\040\000\000", "IAAA"}, - {3, "\100\000\000", "QAAA"}, - {3, "\200\000\000", "gAAA"}, - - {3, "\377\377\377", "////"}, - {3, "\377\377\376", "///+"}, - {3, "\377\377\375", "///9"}, - {3, "\377\377\373", "///7"}, - {3, "\377\377\367", "///3"}, - {3, "\377\377\357", "///v"}, - {3, "\377\377\337", "///f"}, - {3, "\377\377\277", "//+/"}, - {3, "\377\377\177", "//9/"}, - {3, "\377\376\377", "//7/"}, - {3, "\377\375\377", "//3/"}, - {3, "\377\373\377", "//v/"}, - {3, "\377\367\377", "//f/"}, - {3, "\377\357\377", "/+//"}, - {3, "\377\337\377", "/9//"}, - {3, "\377\277\377", "/7//"}, - {3, "\377\177\377", "/3//"}, - {3, "\376\377\377", "/v//"}, - {3, "\375\377\377", "/f//"}, - {3, "\373\377\377", "+///"}, - {3, "\367\377\377", "9///"}, - {3, "\357\377\377", "7///"}, - {3, "\337\377\377", "3///"}, - {3, "\277\377\377", "v///"}, - {3, "\177\377\377", "f///"}, - - // Random numbers: values obtained with - // - // #! /bin/bash - // dd bs=$1 count=1 if=/dev/random of=/tmp/bar.random - // od -N $1 -t o1 /tmp/bar.random - // uuencode -m test < /tmp/bar.random - // - // where $1 is the number of bytes (2, 3) - - {2, "\243\361", "o/E="}, - {2, "\024\167", "FHc="}, - {2, "\313\252", "y6o="}, - {2, "\046\041", "JiE="}, - {2, "\145\236", "ZZ4="}, - {2, "\254\325", "rNU="}, - {2, "\061\330", "Mdg="}, - {2, "\245\032", "pRo="}, - {2, "\006\000", "BgA="}, - {2, "\375\131", "/Vk="}, - {2, "\303\210", "w4g="}, - {2, "\040\037", "IB8="}, - {2, "\261\372", "sfo="}, - {2, "\335\014", "3Qw="}, - {2, "\233\217", "m48="}, - {2, "\373\056", "+y4="}, - {2, "\247\232", "p5o="}, - {2, "\107\053", "Rys="}, - {2, "\204\077", "hD8="}, - {2, "\276\211", "vok="}, - {2, "\313\110", "y0g="}, - {2, "\363\376", "8/4="}, - {2, "\251\234", "qZw="}, - {2, "\103\262", "Q7I="}, - {2, "\142\312", "Yso="}, - {2, "\067\211", "N4k="}, - {2, "\220\001", "kAE="}, - {2, "\152\240", "aqA="}, - {2, "\367\061", "9zE="}, - {2, "\133\255", "W60="}, - {2, "\176\035", "fh0="}, - {2, "\032\231", "Gpk="}, - - {3, "\013\007\144", "Cwdk"}, - {3, "\030\112\106", "GEpG"}, - {3, "\047\325\046", "J9Um"}, - {3, "\310\160\022", "yHAS"}, - {3, "\131\100\237", "WUCf"}, - {3, "\064\342\134", "NOJc"}, - {3, "\010\177\004", "CH8E"}, - {3, "\345\147\205", "5WeF"}, - {3, "\300\343\360", "wOPw"}, - {3, "\061\240\201", "MaCB"}, - {3, "\225\333\044", "ldsk"}, - {3, "\215\137\352", "jV/q"}, - {3, "\371\147\160", "+Wdw"}, - {3, "\030\320\051", "GNAp"}, - {3, "\044\174\241", "JHyh"}, - {3, "\260\127\037", "sFcf"}, - {3, "\111\045\033", "SSUb"}, - {3, "\202\114\107", "gkxH"}, - {3, "\057\371\042", "L/ki"}, - {3, "\223\247\244", "k6ek"}, - {3, "\047\216\144", "J45k"}, - {3, "\203\070\327", "gzjX"}, - {3, "\247\140\072", "p2A6"}, - {3, "\124\115\116", "VE1O"}, - {3, "\157\162\050", "b3Io"}, - {3, "\357\223\004", "75ME"}, - {3, "\052\117\156", "Kk9u"}, - {3, "\347\154\000", "52wA"}, - {3, "\303\012\142", "wwpi"}, - {3, "\060\035\362", "MB3y"}, - {3, "\130\226\361", "WJbx"}, - {3, "\173\013\071", "ews5"}, - {3, "\336\004\027", "3gQX"}, - {3, "\357\366\234", "7/ac"}, - {3, "\353\304\111", "68RJ"}, - {3, "\024\264\131", "FLRZ"}, - {3, "\075\114\251", "PUyp"}, - {3, "\315\031\225", "zRmV"}, - {3, "\154\201\276", "bIG+"}, - {3, "\200\066\072", "gDY6"}, - {3, "\142\350\267", "Yui3"}, - {3, "\033\000\166", "GwB2"}, - {3, "\210\055\077", "iC0/"}, - {3, "\341\037\124", "4R9U"}, - {3, "\161\103\152", "cUNq"}, - {3, "\270\142\131", "uGJZ"}, - {3, "\337\076\074", "3z48"}, - {3, "\375\106\362", "/Uby"}, - {3, "\227\301\127", "l8FX"}, - {3, "\340\002\234", "4AKc"}, - {3, "\121\064\033", "UTQb"}, - {3, "\157\134\143", "b1xj"}, - {3, "\247\055\327", "py3X"}, - {3, "\340\142\005", "4GIF"}, - {3, "\060\260\143", "MLBj"}, - {3, "\075\203\170", "PYN4"}, - {3, "\143\160\016", "Y3AO"}, - {3, "\313\013\063", "ywsz"}, - {3, "\174\236\135", "fJ5d"}, - {3, "\103\047\026", "QycW"}, - {3, "\365\005\343", "9QXj"}, - {3, "\271\160\223", "uXCT"}, - {3, "\362\255\172", "8q16"}, - {3, "\113\012\015", "SwoN"}, - - // various lengths, generated by this python script: - // - // from string import lowercase as lc - // for i in range(27): - // print '{ %2d, "%s",%s "%s" },' % (i, lc[:i], ' ' * (26-i), - // lc[:i].encode('base64').strip()) - - {0, "abcdefghijklmnopqrstuvwxyz", ""}, - {1, "abcdefghijklmnopqrstuvwxyz", "YQ=="}, - {2, "abcdefghijklmnopqrstuvwxyz", "YWI="}, - {3, "abcdefghijklmnopqrstuvwxyz", "YWJj"}, - {4, "abcdefghijklmnopqrstuvwxyz", "YWJjZA=="}, - {5, "abcdefghijklmnopqrstuvwxyz", "YWJjZGU="}, - {6, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVm"}, - {7, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZw=="}, - {8, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2g="}, - {9, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hp"}, - {10, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpag=="}, - {11, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpams="}, - {12, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamts"}, - {13, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbQ=="}, - {14, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW4="}, - {15, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5v"}, - {16, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcA=="}, - {17, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHE="}, - {18, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFy"}, - {19, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFycw=="}, - {20, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFyc3Q="}, - {21, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFyc3R1"}, - {22, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFyc3R1dg=="}, - {23, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFyc3R1dnc="}, - {24, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFyc3R1dnd4"}, - {25, "abcdefghijklmnopqrstuvwxy", "YWJjZGVmZ2hpamtsbW5vcHFyc3R1dnd4eQ=="}, - {26, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFyc3R1dnd4eXo="}, -}; -#if 0 -static struct { - const char* plaintext; - const char* cyphertext; -} base64_strings[] = { - - // The first few Google quotes - // Cyphertext created with "uuencode - GNU sharutils 4.2.1" - { - "Everyone! We're teetering on the brink of disaster." - " - Sergey Brin, 6/24/99, regarding the company's state " - "after the unleashing of Netscape/Google search", - - "RXZlcnlvbmUhICBXZSdyZSB0ZWV0ZXJpbmcgb24gdGhlIGJyaW5rIG9mIGRp" - "c2FzdGVyLiAtIFNlcmdleSBCcmluLCA2LzI0Lzk5LCByZWdhcmRpbmcgdGhl" - "IGNvbXBhbnkncyBzdGF0ZSBhZnRlciB0aGUgdW5sZWFzaGluZyBvZiBOZXRz" - "Y2FwZS9Hb29nbGUgc2VhcmNo" }, - - { - "I'm not sure why we're still alive, but we seem to be." - " - Larry Page, 6/24/99, while hiding in the kitchenette " - "during the Netscape traffic overflow", - - "SSdtIG5vdCBzdXJlIHdoeSB3ZSdyZSBzdGlsbCBhbGl2ZSwgYnV0IHdlIHNl" - "ZW0gdG8gYmUuIC0gTGFycnkgUGFnZSwgNi8yNC85OSwgd2hpbGUgaGlkaW5n" - "IGluIHRoZSBraXRjaGVuZXR0ZSBkdXJpbmcgdGhlIE5ldHNjYXBlIHRyYWZm" - "aWMgb3ZlcmZsb3c" }, - - { - "I think kids want porn." - " - Sergey Brin, 6/99, on why Google shouldn't prioritize a " - "filtered search for children and families", - - "SSB0aGluayBraWRzIHdhbnQgcG9ybi4gLSBTZXJnZXkgQnJpbiwgNi85OSwg" - "b24gd2h5IEdvb2dsZSBzaG91bGRuJ3QgcHJpb3JpdGl6ZSBhIGZpbHRlcmVk" - "IHNlYXJjaCBmb3IgY2hpbGRyZW4gYW5kIGZhbWlsaWVz" }, -}; -#endif -// Compare bytes 0..len-1 of x and y. If not equal, abort with verbose error -// message showing position and numeric value that differed. -// Handles embedded nulls just like any other byte. -// Only added because string.compare() in gcc-3.3.3 seems to misbehave with -// embedded nulls. -// TODO: switch back to string.compare() if/when gcc is fixed -#define EXPECT_EQ_ARRAY(len, x, y, msg) \ - for (size_t j = 0; j < len; ++j) { \ - if (x[j] != y[j]) { \ - RTC_LOG(LS_ERROR) << "" #x << " != " #y << " byte " << j \ - << " msg: " << msg; \ - } \ - } - -size_t Base64Escape(const unsigned char* src, - size_t szsrc, - char* dest, - size_t szdest) { - std::string escaped; - webrtc::Base64::EncodeFromArray((const char*)src, szsrc, &escaped); - memcpy(dest, escaped.data(), std::min(escaped.size(), szdest)); - return escaped.size(); -} - -size_t Base64Unescape(const char* src, - size_t szsrc, - char* dest, - size_t szdest) { - std::string unescaped; - EXPECT_TRUE(webrtc::Base64::DecodeFromArray( - src, szsrc, webrtc::Base64::DO_LAX, &unescaped, nullptr)); - memcpy(dest, unescaped.data(), std::min(unescaped.size(), szdest)); - return unescaped.size(); -} - -size_t Base64Unescape(const char* src, size_t szsrc, std::string* s) { - EXPECT_TRUE(webrtc::Base64::DecodeFromArray( - src, szsrc, webrtc::Base64::DO_LAX, s, nullptr)); - return s->size(); -} - -TEST(Base64, EncodeDecodeBattery) { - RTC_LOG(LS_VERBOSE) << "Testing base-64"; - - size_t i; - - // Check the short strings; this tests the math (and boundaries) - for (i = 0; i < sizeof(base64_tests) / sizeof(base64_tests[0]); ++i) { - char encode_buffer[100]; - size_t encode_length; - char decode_buffer[100]; - size_t decode_length; - size_t cypher_length; - - RTC_LOG(LS_VERBOSE) << "B64: " << base64_tests[i].cyphertext; - - const unsigned char* unsigned_plaintext = - reinterpret_cast(base64_tests[i].plaintext); - - cypher_length = strlen(base64_tests[i].cyphertext); - - // The basic escape function: - memset(encode_buffer, 0, sizeof(encode_buffer)); - encode_length = - Base64Escape(unsigned_plaintext, base64_tests[i].plain_length, - encode_buffer, sizeof(encode_buffer)); - // Is it of the expected length? - EXPECT_EQ(encode_length, cypher_length); - - // Is it the expected encoded value? - EXPECT_STREQ(encode_buffer, base64_tests[i].cyphertext); - - // If we encode it into a buffer of exactly the right length... - memset(encode_buffer, 0, sizeof(encode_buffer)); - encode_length = - Base64Escape(unsigned_plaintext, base64_tests[i].plain_length, - encode_buffer, cypher_length); - // Is it still of the expected length? - EXPECT_EQ(encode_length, cypher_length); - - // And is the value still correct? (i.e., not losing the last byte) - EXPECT_STREQ(encode_buffer, base64_tests[i].cyphertext); - - // If we decode it back: - memset(decode_buffer, 0, sizeof(decode_buffer)); - decode_length = Base64Unescape(encode_buffer, cypher_length, decode_buffer, - sizeof(decode_buffer)); - - // Is it of the expected length? - EXPECT_EQ(decode_length, base64_tests[i].plain_length); - - // Is it the expected decoded value? - EXPECT_EQ(0, - memcmp(decode_buffer, base64_tests[i].plaintext, decode_length)); - - // Our decoder treats the padding '=' characters at the end as - // optional. If encode_buffer has any, run some additional - // tests that fiddle with them. - char* first_equals = strchr(encode_buffer, '='); - if (first_equals) { - // How many equals signs does the string start with? - int equals = (*(first_equals + 1) == '=') ? 2 : 1; - - // Try chopping off the equals sign(s) entirely. The decoder - // should still be okay with this. - std::string decoded2("this junk should also be ignored"); - *first_equals = '\0'; - EXPECT_NE(0U, Base64Unescape(encode_buffer, first_equals - encode_buffer, - &decoded2)); - EXPECT_EQ(decoded2.size(), base64_tests[i].plain_length); - EXPECT_EQ_ARRAY(decoded2.size(), decoded2.data(), - base64_tests[i].plaintext, i); - - size_t len; - - // try putting some extra stuff after the equals signs, or in between them - if (equals == 2) { - snprintf(first_equals, 6, " = = "); - len = first_equals - encode_buffer + 5; - } else { - snprintf(first_equals, 6, " = "); - len = first_equals - encode_buffer + 3; - } - decoded2.assign("this junk should be ignored"); - EXPECT_NE(0U, Base64Unescape(encode_buffer, len, &decoded2)); - EXPECT_EQ(decoded2.size(), base64_tests[i].plain_length); - EXPECT_EQ_ARRAY(decoded2.size(), decoded2, base64_tests[i].plaintext, i); - } - } -} - -// here's a weird case: a giant base64 encoded stream which broke our base64 -// decoding. Let's test it explicitly. -const char SpecificTest[] = - "/9j/4AAQSkZJRgABAgEASABIAAD/" - "4Q0HRXhpZgAATU0AKgAAAAgADAEOAAIAAAAgAAAAngEPAAI\n" - "AAAAFAAAAvgEQAAIAAAAJAAAAwwESAAMAAAABAAEAAAEaAAUAAAABAAAAzAEbAAUAAAABAAAA1" - "A\n" - "EoAAMAAAABAAIAAAExAAIAAAAUAAAA3AEyAAIAAAAUAAAA8AE8AAIAAAAQAAABBAITAAMAAAAB" - "A\n" - "AIAAIdpAAQAAAABAAABFAAAAsQgICAgICAgICAgICAgICAgICAgICAgICAgICAgICAgAFNPTlk" - "A\n" - "RFNDLVAyMDAAAAAASAAAAAEAAABIAAAAAUFkb2JlIFBob3Rvc2hvcCA3LjAAMjAwNzowMTozMC" - "A\n" - "yMzoxMDowNABNYWMgT1MgWCAxMC40LjgAAByCmgAFAAAAAQAAAmqCnQAFAAAAAQAAAnKIIgADA" - "A\n" - "AAAQACAACIJwADAAAAAQBkAACQAAAHAAAABDAyMjCQAwACAAAAFAAAAnqQBAACAAAAFAAAAo6R" - "A\n" - "QAHAAAABAECAwCRAgAFAAAAAQAAAqKSBAAKAAAAAQAAAqqSBQAFAAAAAQAAArKSBwADAAAAAQA" - "F\n" - "AACSCAADAAAAAQAAAACSCQADAAAAAQAPAACSCgAFAAAAAQAAArqgAAAHAAAABDAxMDCgAQADAA" - "A\n" - "AAf//" - "AACgAgAEAAAAAQAAAGSgAwAEAAAAAQAAAGSjAAAHAAAAAQMAAACjAQAHAAAAAQEAAACkAQ\n" - "ADAAAAAQAAAACkAgADAAAAAQAAAACkAwADAAAAAQAAAACkBgADAAAAAQAAAACkCAADAAAAAQAA" - "A\n" - "ACkCQADAAAAAQAAAACkCgADAAAAAQAAAAAAAAAAAAAACgAAAZAAAAAcAAAACjIwMDc6MDE6MjA" - "g\n" - "MjM6MDU6NTIAMjAwNzowMToyMCAyMzowNTo1MgAAAAAIAAAAAQAAAAAAAAAKAAAAMAAAABAAAA" - "B\n" - "PAAAACgAAAAYBAwADAAAAAQAGAAABGgAFAAAAAQAAAxIBGwAFAAAAAQAAAxoBKAADAAAAAQACA" - "A\n" - "ACAQAEAAAAAQAAAyICAgAEAAAAAQAACd0AAAAAAAAASAAAAAEAAABIAAAAAf/Y/" - "+AAEEpGSUYAA\n" - "QIBAEgASAAA/+0ADEFkb2JlX0NNAAL/7gAOQWRvYmUAZIAAAAAB/" - "9sAhAAMCAgICQgMCQkMEQsK\n" - "CxEVDwwMDxUYExMVExMYEQwMDAwMDBEMDAwMDAwMDAwMDAwMDAwMDAwMDAwMDAwMDAwMAQ0LCw" - "0\n" - "ODRAODhAUDg4OFBQODg4OFBEMDAwMDBERDAwMDAwMEQwMDAwMDAwMDAwMDAwMDAwMDAwMDAwMD" - "A\n" - "wMDAz/wAARCABkAGQDASIAAhEBAxEB/90ABAAH/" - "8QBPwAAAQUBAQEBAQEAAAAAAAAAAwABAgQFB\n" - "gcICQoLAQABBQEBAQEBAQAAAAAAAAABAAIDBAUGBwgJCgsQAAEEAQMCBAIFBwYIBQMMMwEAAhE" - "D\n" - "BCESMQVBUWETInGBMgYUkaGxQiMkFVLBYjM0coLRQwclklPw4fFjczUWorKDJkSTVGRFwqN0Nh" - "f\n" - "SVeJl8rOEw9N14/" - "NGJ5SkhbSVxNTk9KW1xdXl9VZmdoaWprbG1ub2N0dXZ3eHl6e3x9fn9xEAAg\n" - "IBAgQEAwQFBgcHBgU1AQACEQMhMRIEQVFhcSITBTKBkRShsUIjwVLR8DMkYuFygpJDUxVjczTx" - "J\n" - "QYWorKDByY1wtJEk1SjF2RFVTZ0ZeLys4TD03Xj80aUpIW0lcTU5PSltcXV5fVWZnaGlqa2xtb" - "m\n" - "9ic3R1dnd4eXp7fH/" - "9oADAMBAAIRAxEAPwDy7bKNTUXNLz9EaJPDWMjxH4ozhtpYwaACT8ShaaW\n" - "bW0uEc9/JFfjj0Q4Hk/PRDxwX7y47W9z/" - "AN9Cv4+O3ILK2DcRqT2CaSvEbcl1Jbz37KG1dBldLo\n" - "qaS4l9xGjG9v6yoDAdYIaIjUk+AREgo4y5sapirb8Yl0NHHdKvBNm4yA1o5Pc+" - "SPEFvCWqB3HZF\n" - "Hj2SbWQ/" - "afGFP0bHP8ATY0uc4w1o1JPkkimGiS2KvqlnmBkOZQTyydzgPMM9v8A0lp4v1Nx9gF1\n" - "tpdqJaGtH/S3I0i3lISXW/8AMqnd/O2bfg2eUkqVYf/" - "Q8zuncO4Bj7lZ+n7f5Mj5KsJcY8NUZ4d\n" - "uEDVo1HkeU0rg3Om4H2rabCWUN7DQuK1n5FWKW4uCwG92gDRJBS6exhxmMboQI+" - "Cv4WFTQ42Bs2\n" - "fvnkkqEmy2YxoMMbpVzaz6jt+RbpHZs8lzkHqrasKkYOKP0jgDfZ4N/" - "wDM1tNrcWfSPmRyq9uNV\n" - "DnFg2s97i7UkjxKVrq0eVz3spZsja+ASDzwsh9jnOk/" - "JFzb3XZD3v1c4yT8UACTCniKDUnKz5Nj\n" - "G33XV1DV73BrT8dF23SejV4zg9g33cOsPb+SxVvqv9ViwNy8vS0iWs/" - "daf8A0Y5dpTi1sADGxCR\n" - "K1o0YBEmInlXWYbDBcDLdPJXa8f71Yrx2jnUoAqLnfZK5hJaW2vdwEk5a/wD/0fN6Ia/" - "e76IiVf\n" - "xavUL7CPpnT4LNbYXAVjuQt/AqDmNYO/" - 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"Gg5lQybQK+" - "82GBqEQUA1kOHPYf3LLsoyN36G5w8iUfHxepbXE2l0cApALgLHzBq9UxhTXU5hMC1\n" - "ktnSCup6S4Ctk+C5XqVGcaHPfuiuHkeTTuWz0+9zaKiH6CC0/yXBSQ2a/" - "MxojV57634rq+v2PLY\n" - "be1r2nsYG13/" - "AFKxbfCBMcr0brGAzrGEwCG31ncx0SfBzf7S4+zoHUWWsJq3hz9oLfcBH77R9H+\n" - "0pA13u/qPgDp/Q6ri39JlfpXkDx+h/" - "msWn1L6wdO6bSbcrIbU2Q0xLnSe21kuVejJspbVS5+4bd\n" - "ocBAkD/orG+tP1ar67Wy7GtZTm1SCXfRsb+a18fRe38x6SG3/44H1Z3f0y2I+l6DoSXD/" - "8xPrDs\n" - "3enVu3bdnqN3R+//USSVo//" - "1PLohhce+gRWS0Nsby3lRgFkKxQyW7SgUh3em5Tbq2uB9wWw1wey\n" - "J1XGV2XYdm5k7e4WzidXY9oMwo5RZ4T6Hd1ixwfp96PWbAJBVTHzK7O6Ky5oJB1HZMqmUEFlkG" - "y\n" - "xpa4zI1Hkq31dy7bMN9BAc3HeWAnnbyxEycmuup1jiAGglZ31PyrmZ9tQg1WtNj54EHR3/" - "S2qTH\n" - "1Yc5GgD1FFtzPdWGkd2AyflogZmRmsz6PSrbXbdo+" - "txOrP337f3fzVo15DK2uyrTtqpBOnBKx6b\n" - "7MjJsz7tHWOAYP3WD6LU6cqGjFCNl1MmvLcxv6YtDTLSAqP27LrdtYHXFnJZI+" - "Tp3MWg68OpDPv\n" - "UMUM2lkQBoouKQ6swjE9Nml+1sz1PW+z6xt27zuj+skrX2ZvqR5z8kkuOfdPt43/1fMm/" - "grFG6f\n" - "Lss9JA7JG7tnZs/SfJUrfS3foJ9TvHCopJsV8nWx/t24bJn8Fo/5TjWJXMJIS+i+G36TsZ/" - 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"aFT1ANOXemV5piR2b9NiljB4yyHy9CLOVI5GJhB+CvXY9R8xmINzs5HNZ+Z96BZpbxA\n" - "fVJo39UFefwopYgL4nMiMd2qZoIn/AJx00u3t/" - "Lt7qpp9Yv5GLf5MUTERqfbvmzBeezjd9H+VlL\n" - "wSQzBqsvOGQD7L12rXsemPNxmXQSxxIPU2nFV4HYqR1xEUWj4ZAxBryr2G+" - "J2VGDZlLrxUH6KZA\n" - "Fkqb15VFelfwy+2FP8A/" - "9Xxlf6AdA182Yk9eFeLxSjoVfcfSMo4uIOfkweFOnpvlWYrLEwNFAA+\n" - "nMOYdrhFvQLeSO7coBXiK8iKiv07Zj8Ac4QtNrW1njUcKcT+yAR/" - "xGmR4WcsStLpTuPU9IFaEsV\n" - "BP3k4m2AgBzSwyQNcIwNTE1aI3wnam9O2Ug7s5Ckk/" - "NDndeVXa2H78MqqV6jmeBp9+ZWKXqDjZ4\n" - "+gvVvy30qCy0qzsLRBCnBI2VdgUTqPvOZ7y+Q7pz+bn5q6d+VflZxZlJ/" - "NN4ypptk5qtB9qRwDX\n" - "gn/AAx2y2ItpfKFv+eH5qNeTajJ5ovVaVywSqvEtTUKqupAA6D2y0BNPtv/AJx//" - "M5PzL8mJeXT\n" - "L+ndPf6rqarSpkAqsnEAAeoN6DpkJRYci9lROSgSUUH9o9K5Tw0ztfSHnXkOtK9q+PHwydq//" - "9b\n" - "yxrVoZNBtNSA5zRMPXmH8j0CLXuBmHE+" - "qneamHpEuqYeV7pzFVTRgQK5XMNmnlb1vyyY5QA1OwJ\n" - "+eUF2seTOLu5s7azVIVAkpVn/" - "hhnIALG73Yz5jvb1dICqzpDNIqyFD8SxH7R28cxibZCiWOsdJs\n" - "PTM6XNstPhnkjIhcHuJBVfvOCiUSn0TfWrTTLjyw8guA/PifTO3xcxxA8a5ZAbimvJP0m3p/" - "kFF\n" - "WxhmpWQJ9NW3zZPHz5vlb/nIDVbrWfzO1RJhxGnpDaRL/" - "khA1T7ktmSOTAJhZaAUtLawsbayl8v\n" - "xWi3Gpay0cF3HPcFRJJHJMXVrcJ8UaAFG5LWjF8tAYW9H/wCcOo9bTzxrt/" - "owkTyksZW5gkIKvI\n" - "7k26nvyReRJHyyBWT7dWQyOWlbnK2526e1O1MqIUFE84uPLkOdK9RXI0E2/wD/1/DA1bURZLY/" - "W\n" - "ZDZqwb0eXw7dMgIi7bjllVXsz7yNcfWC0Vd3Ip92Y2UOz0cnsPlwyx8xQ/" - "u24sMxCadoJp9LOXk\n" - "VX/" - "uwRUE0BI8cokbLMyoKouHu2MaKGXw7fLDwgoGSkbHpaNZyLLHRSKcFFQQRvUdMlwUFOQyLzr\n" - "ztpCaba6fPau4ijv4OURY8AjVFKV7ZZiO+7Vnh6XvXkSWNbW2WTb92KDxIFMzwHlZc3zX+" - "fuizW\n" - "f5p3ty8XGDU4YLmCQiisyII3+4rvl8UB5ffEghRGvOm7AbnvWvjk1fen/" - "ONPldPKP5aWOpPCfr2\n" - "uE31y6q2wbaMEn+VAMDSdyzrzj+avlHyTp0l/r2rxWFuHWJuIeacu4qFCRgsajfBwsty89/" - "6Gr/\n" - "ACa9an+JL/hSnrfoubhXwpXpjwhaL//" - "Q8E1AqtcAZMs8l6i1nqMa1oSVP0VynKLDmaWdSfQXl69\n" - "jF1Jv8MhDb5rpB3AO7INRRLhhGp4R05FgaGvTMU8200xS70zVDMRp2pTIOvBmB3PgQP15kxIcn" - "D\n" - "LH/" - "EEz0rRvOJhldr9pQtCqyd6VrShGTqw5d4ARv9jHfOGl+ZJNMluLkyenaFbiRdqFYW5nrWuwO\n" - "MKB5MdSMRxnhlu9N8p6lLFpti63FUjCtFJTrDKvse2bEDZ4XJ9RZB+YPli2/" - "Mjy5bxoUi1a0YS2\n" - "85UOwIXiy9jRu+TBppfOF1+V3m22vrdpNPM8cs/oo0VJlUqQPjValR3+IZNNvtLS9Yu9Mi0/" - "TJr\n" - "kyp6QhWVVCIWRATsKBemwwFrDzT87fybs/" - "wA1bW21PRb+DTvNlgGSRp6iC8i3KJJx+y6n7D0Pwm\n" - "hxBZXT55/6Fi/Nf0PW+qWXq+t6X1X67F6vD/ftK04V/wBl344U8b//" - "0fBapxheVh9ocV+nviqY2\n" - "/qQJDew/" - "bioWHiuQ8m0bbvaPKGtQ6jaxSo9JloCK75gZI0Xb4sgkHo8MouoAvP94BsRmGY7uWJU\n" - "gzbypOQpNOvIdK4Nw2WCE2tXulTkjEEbdafgclxMhFBas93dwyQzsWDghlJFONKHJCZtjOFBJf" - "y\n" - "j1y9vPL9zpbIs0WkXL2sUjA8hDXlGCRXtt07ZuYvL5KJeo6bfajbkzWkcToR8dqshZ6in2fhNK" - "/\n" - "PDTUlXmHVvMdr5o0v9H2kdrqGpfu7m0nkY87Uf7tkKAU4/" - "s03ynLkEBbfihx7dGT6va67LbRMNR\n" - "aKOBuUTKgIBXoK1BOYR1M3aQ0mOt9yxUeZNdtJhFapLqMluSXkg5oxJrUMW5KevQ9MmNXXNqOi" - "H\n" - "Rr/Hmv8A1r9I/oj95w+r+j9Yf1+NP5+nXtTD+dF8tkfkOlv/0vC3ph7f0/" - "alcVTbS4A8QibuKb5\n" - "RI05EBYRFpdX3ly79a2qYCavH/" - "EY7TCYyMD5PSdD8+wXUSn1ArDqOhBzFlipz4ZwWbaV5htbsgF\n" - "qg9crMXKErGyYwajFGzxyHlGSePbbwyqg5UZlCaxrFpaWU95LIqrEjMAT4Dp9OShGy1ZslBhv/" - "A\n" - "Dj9rd/a+aL+xUK+m38L3d0HrxRo2HFtu5D8c27y8t30raarbWkU+u6g4gsNORn+EcUaSh2Pc0/" - "4\n" - "lgtAjezzbT9SutY1i782al8Nxdyotqh6xWybIg+jc5q8s+I27bFDgFPQp9RE+nrag70+" - "L6crrZu\n" - "4jajokdv6LW/Dii1Wo61PXKQN3KPK0L+h4/rnD/K5V78a5LhXxd3/0/" - "DMXXtwxVNtL9Xkaf3f7N\n" - "etfbKMjdjtkZ9D6ufrlK0+HpX8coF9HJ26sXvfqXrf7i/U+uften/d/" - "wCyrmQL6uOav0pvpP8Ai\n" - "b1F+rV59+vH6a5XLhcjH4nRmY/xpxHP0/UptWvT6Mx/RbmjxWK+aP8AFf1M/" - "pCv1Kvxen9inavf\n" - "MrFwXtzcLUeLXq5Mv/I3nz1b0v8AjofuKVry9KrUpTanOlf9jmQ68va/zH9b/COn/o7/" - "AI431mP\n" - "65SvLh+zWvbl9rMfNfC34K4kmj9T6lD6FKclp/DNYXZx5srsPrHor6nXvkgxTPS/" - "U+rv6dPU5mt\n" - "fngFN5ulv+l/pL/Lp/scerHo//2Q==\n"; - -static std::string gCommandLine; - -TEST(Base64, LargeSample) { - RTC_LOG(LS_VERBOSE) << "Testing specific base64 file"; - - char unescaped[64 * 1024]; - - // unescape that massive blob above - size_t size = Base64Unescape(SpecificTest, sizeof(SpecificTest), unescaped, - sizeof(unescaped)); - - EXPECT_EQ(size, sizeof(testbase64)); - EXPECT_EQ(0, memcmp(testbase64, unescaped, sizeof(testbase64))); -} - -bool DecodeTest(const char* encoded, - size_t expect_unparsed, - const char* decoded, - webrtc::Base64::DecodeFlags flags) { - std::string result; - size_t consumed = 0, encoded_len = strlen(encoded); - bool success = webrtc::Base64::DecodeFromArray(encoded, encoded_len, flags, - &result, &consumed); - size_t unparsed = encoded_len - consumed; - EXPECT_EQ(expect_unparsed, unparsed) - << "\"" << encoded << "\" -> \"" << decoded << "\""; - EXPECT_STREQ(decoded, result.c_str()); - return success; -} - -#define Flags(x, y, z) \ - webrtc::Base64::DO_PARSE_##x | webrtc::Base64::DO_PAD_##y | \ - webrtc::Base64::DO_TERM_##z - -TEST(Base64, DecodeParseOptions) { - // Trailing whitespace - EXPECT_TRUE(DecodeTest("YWJjZA== ", 1, "abcd", Flags(STRICT, YES, CHAR))); - EXPECT_TRUE(DecodeTest("YWJjZA== ", 0, "abcd", Flags(WHITE, YES, CHAR))); - EXPECT_TRUE(DecodeTest("YWJjZA== ", 0, "abcd", Flags(ANY, YES, CHAR))); - - // Embedded whitespace - EXPECT_FALSE(DecodeTest("YWJjZA= =", 3, "abcd", Flags(STRICT, YES, CHAR))); - EXPECT_TRUE(DecodeTest("YWJjZA= =", 0, "abcd", Flags(WHITE, YES, CHAR))); - EXPECT_TRUE(DecodeTest("YWJjZA= =", 0, "abcd", Flags(ANY, YES, CHAR))); - - // Embedded non-base64 characters - EXPECT_FALSE(DecodeTest("YWJjZA=*=", 3, "abcd", Flags(STRICT, YES, CHAR))); - EXPECT_FALSE(DecodeTest("YWJjZA=*=", 3, "abcd", Flags(WHITE, YES, CHAR))); - EXPECT_TRUE(DecodeTest("YWJjZA=*=", 0, "abcd", Flags(ANY, YES, CHAR))); - - // Unexpected padding characters - EXPECT_FALSE(DecodeTest("YW=JjZA==", 7, "a", Flags(STRICT, YES, CHAR))); - EXPECT_FALSE(DecodeTest("YW=JjZA==", 7, "a", Flags(WHITE, YES, CHAR))); - EXPECT_TRUE(DecodeTest("YW=JjZA==", 0, "abcd", Flags(ANY, YES, CHAR))); -} - -TEST(Base64, DecodePadOptions) { - // Padding - EXPECT_TRUE(DecodeTest("YWJjZA==", 0, "abcd", Flags(STRICT, YES, CHAR))); - EXPECT_TRUE(DecodeTest("YWJjZA==", 0, "abcd", Flags(STRICT, ANY, CHAR))); - EXPECT_TRUE(DecodeTest("YWJjZA==", 2, "abcd", Flags(STRICT, NO, CHAR))); - - // Incomplete padding - EXPECT_FALSE(DecodeTest("YWJjZA=", 1, "abcd", Flags(STRICT, YES, CHAR))); - EXPECT_TRUE(DecodeTest("YWJjZA=", 1, "abcd", Flags(STRICT, ANY, CHAR))); - EXPECT_TRUE(DecodeTest("YWJjZA=", 1, "abcd", Flags(STRICT, NO, CHAR))); - - // No padding - EXPECT_FALSE(DecodeTest("YWJjZA", 0, "abcd", Flags(STRICT, YES, CHAR))); - EXPECT_TRUE(DecodeTest("YWJjZA", 0, "abcd", Flags(STRICT, ANY, CHAR))); - EXPECT_TRUE(DecodeTest("YWJjZA", 0, "abcd", Flags(STRICT, NO, CHAR))); -} - -TEST(Base64, DecodeTerminateOptions) { - // Complete quantum - EXPECT_TRUE(DecodeTest("YWJj", 0, "abc", Flags(STRICT, NO, BUFFER))); - EXPECT_TRUE(DecodeTest("YWJj", 0, "abc", Flags(STRICT, NO, CHAR))); - EXPECT_TRUE(DecodeTest("YWJj", 0, "abc", Flags(STRICT, NO, ANY))); - - // Complete quantum with trailing data - EXPECT_FALSE(DecodeTest("YWJj*", 1, "abc", Flags(STRICT, NO, BUFFER))); - EXPECT_TRUE(DecodeTest("YWJj*", 1, "abc", Flags(STRICT, NO, CHAR))); - EXPECT_TRUE(DecodeTest("YWJj*", 1, "abc", Flags(STRICT, NO, ANY))); - - // Incomplete quantum - EXPECT_FALSE(DecodeTest("YWJ", 0, "ab", Flags(STRICT, NO, BUFFER))); - EXPECT_FALSE(DecodeTest("YWJ", 0, "ab", Flags(STRICT, NO, CHAR))); - EXPECT_TRUE(DecodeTest("YWJ", 0, "ab", Flags(STRICT, NO, ANY))); -} - -TEST(Base64, GetNextBase64Char) { - // The table looks like this: - // "ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789+/" - char next_char; - EXPECT_TRUE(webrtc::Base64::GetNextBase64Char('A', &next_char)); - EXPECT_EQ('B', next_char); - EXPECT_TRUE(webrtc::Base64::GetNextBase64Char('Z', &next_char)); - EXPECT_EQ('a', next_char); - EXPECT_TRUE(webrtc::Base64::GetNextBase64Char('/', &next_char)); - EXPECT_EQ('A', next_char); - EXPECT_FALSE(webrtc::Base64::GetNextBase64Char('&', &next_char)); - EXPECT_FALSE(webrtc::Base64::GetNextBase64Char('Z', nullptr)); -} - -} // namespace -} // namespace rtc -#endif // WEBRTC_ALLOW_DEPRECATED_NAMESPACES diff --git a/rtc_base/third_party/base64/test_base64.h b/rtc_base/third_party/base64/test_base64.h deleted file mode 100644 index 51d8c58113..0000000000 --- a/rtc_base/third_party/base64/test_base64.h +++ /dev/null @@ -1,2111 +0,0 @@ -/* - * Copyright 2004 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef RTC_BASE_TEST_BASE64_H_ -#define RTC_BASE_TEST_BASE64_H_ - -/* This file was generated by googleclient/talk/binary2header.sh */ - -static unsigned char testbase64[] = { - 0xff, 0xd8, 0xff, 0xe0, 0x00, 0x10, 0x4a, 0x46, 0x49, 0x46, 0x00, 0x01, - 0x02, 0x01, 0x00, 0x48, 0x00, 0x48, 0x00, 0x00, 0xff, 0xe1, 0x0d, 0x07, - 0x45, 0x78, 0x69, 0x66, 0x00, 0x00, 0x4d, 0x4d, 0x00, 0x2a, 0x00, 0x00, - 0x00, 0x08, 0x00, 0x0c, 0x01, 0x0e, 0x00, 0x02, 0x00, 0x00, 0x00, 0x20, - 0x00, 0x00, 0x00, 0x9e, 0x01, 0x0f, 0x00, 0x02, 0x00, 0x00, 0x00, 0x05, - 0x00, 0x00, 0x00, 0xbe, 0x01, 0x10, 0x00, 0x02, 0x00, 0x00, 0x00, 0x09, - 0x00, 0x00, 0x00, 0xc3, 0x01, 0x12, 0x00, 0x03, 0x00, 0x00, 0x00, 0x01, - 0x00, 0x01, 0x00, 0x00, 0x01, 0x1a, 0x00, 0x05, 0x00, 0x00, 0x00, 0x01, - 0x00, 0x00, 0x00, 0xcc, 0x01, 0x1b, 0x00, 0x05, 0x00, 0x00, 0x00, 0x01, - 0x00, 0x00, 0x00, 0xd4, 0x01, 0x28, 0x00, 0x03, 0x00, 0x00, 0x00, 0x01, - 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MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The implementation of VideoFrameBuffer might have a more efficient way to scale images. Therefore it is beneficial to first scale the image down to the top active layer resolution. Bug: chromium:397485312 Change-Id: I782bc12d78e5bcab75345dbb145bce68100ba570 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/379680 Reviewed-by: Erik Språng Commit-Queue: Ilya Nikolaevskiy Cr-Commit-Position: refs/heads/main@{#44531} --- .../codecs/av1/libaom_av1_encoder.cc | 132 ++++++++++++++---- .../codecs/av1/libaom_av1_encoder_unittest.cc | 95 +++++++++++++ 2 files changed, 201 insertions(+), 26 deletions(-) diff --git a/modules/video_coding/codecs/av1/libaom_av1_encoder.cc b/modules/video_coding/codecs/av1/libaom_av1_encoder.cc index d0c513e91b..339a508256 100644 --- a/modules/video_coding/codecs/av1/libaom_av1_encoder.cc +++ b/modules/video_coding/codecs/av1/libaom_av1_encoder.cc @@ -13,6 +13,7 @@ #include #include +#include #include #include #include @@ -124,10 +125,22 @@ class LibaomAv1Encoder final : public VideoEncoder { void SetSvcRefFrameConfig( const ScalableVideoController::LayerFrameConfig& layer_frame); // If pixel format doesn't match, then reallocate. - void MaybeRewrapImgWithFormat(const aom_img_fmt_t fmt); + void MaybeRewrapImgWithFormat(const aom_img_fmt_t fmt, + unsigned int width, + unsigned int height); + + // Adjust sclaing factors assuming that the top active SVC layer + // will be the input resolution. + void AdjustScalingFactorsForTopActiveLayer(); std::unique_ptr svc_controller_; std::optional scalability_mode_; + // Original scaling factors for all configured layers active and inactive. + // `svc_params_` stores factors ignoring top inactive layers. + std::vector scaling_factors_num_; + std::vector scaling_factors_den_; + int last_active_layer_ = 0; + bool inited_; bool rates_configured_; std::optional svc_params_; @@ -469,9 +482,19 @@ bool LibaomAv1Encoder::SetSvcParams( 1 << (svc_config.num_temporal_layers - tid - 1); } + scaling_factors_den_.resize(svc_config.num_spatial_layers); + scaling_factors_num_.resize(svc_config.num_spatial_layers); for (int sid = 0; sid < svc_config.num_spatial_layers; ++sid) { + scaling_factors_num_[sid] = svc_config.scaling_factor_num[sid]; svc_params.scaling_factor_num[sid] = svc_config.scaling_factor_num[sid]; + scaling_factors_den_[sid] = svc_config.scaling_factor_den[sid]; svc_params.scaling_factor_den[sid] = svc_config.scaling_factor_den[sid]; + encoder_settings_.spatialLayers[sid].width = encoder_settings_.width * + scaling_factors_num_[sid] / + scaling_factors_den_[sid]; + encoder_settings_.spatialLayers[sid].height = encoder_settings_.height * + scaling_factors_num_[sid] / + scaling_factors_den_[sid]; } // svc_params.layer_target_bitrate is set in SetRates() before svc_params is @@ -540,21 +563,61 @@ int32_t LibaomAv1Encoder::Release() { return WEBRTC_VIDEO_CODEC_OK; } -void LibaomAv1Encoder::MaybeRewrapImgWithFormat(const aom_img_fmt_t fmt) { +void LibaomAv1Encoder::MaybeRewrapImgWithFormat(const aom_img_fmt_t fmt, + unsigned int width, + unsigned int height) { if (!frame_for_encode_) { - frame_for_encode_ = - aom_img_wrap(nullptr, fmt, cfg_.g_w, cfg_.g_h, 1, nullptr); - - } else if (frame_for_encode_->fmt != fmt) { + RTC_LOG(LS_INFO) << "Configuring AV1 encoder pixel format to " + << (fmt == AOM_IMG_FMT_NV12 ? "NV12" : "I420") << " " + << width << "x" << height; + frame_for_encode_ = aom_img_wrap(nullptr, fmt, width, height, 1, nullptr); + } else if (frame_for_encode_->fmt != fmt || frame_for_encode_->d_w != width || + frame_for_encode_->d_h != height) { RTC_LOG(LS_INFO) << "Switching AV1 encoder pixel format to " - << (fmt == AOM_IMG_FMT_NV12 ? "NV12" : "I420"); + << (fmt == AOM_IMG_FMT_NV12 ? "NV12" : "I420") << " " + << width << "x" << height; aom_img_free(frame_for_encode_); - frame_for_encode_ = - aom_img_wrap(nullptr, fmt, cfg_.g_w, cfg_.g_h, 1, nullptr); + frame_for_encode_ = aom_img_wrap(nullptr, fmt, width, height, 1, nullptr); } // else no-op since the image is already in the right format. } +void LibaomAv1Encoder::AdjustScalingFactorsForTopActiveLayer() { + if (!SvcEnabled()) + return; + last_active_layer_ = svc_params_->number_spatial_layers - 1; + for (int sid = 0; sid < svc_params_->number_spatial_layers; ++sid) { + for (int tid = 0; tid < svc_params_->number_temporal_layers; ++tid) { + int layer_index = sid * svc_params_->number_temporal_layers + tid; + if (svc_params_->layer_target_bitrate[layer_index] > 0) { + last_active_layer_ = sid; + } + } + } + if (static_cast(cfg_.g_w) == + encoder_settings_.spatialLayers[last_active_layer_].width) { + return; + } + + cfg_.g_w = encoder_settings_.spatialLayers[last_active_layer_].width; + cfg_.g_h = encoder_settings_.spatialLayers[last_active_layer_].height; + + // Recalculate scaling factors ignoring top inactive layers. + // Divide all by scaling factor of the last active layer. + for (int i = 0; i <= last_active_layer_; ++i) { + int n = scaling_factors_num_[i] * scaling_factors_den_[last_active_layer_]; + int d = scaling_factors_den_[i] * scaling_factors_num_[last_active_layer_]; + int gcd = std::gcd(n, d); + svc_params_->scaling_factor_num[i] = n / gcd; + svc_params_->scaling_factor_den[i] = d / gcd; + } + for (int i = last_active_layer_ + 1; i < svc_params_->number_spatial_layers; + ++i) { + svc_params_->scaling_factor_num[i] = 1; + svc_params_->scaling_factor_den[i] = 1; + } +} + int32_t LibaomAv1Encoder::Encode( const VideoFrame& frame, const std::vector* frame_types) { @@ -578,13 +641,24 @@ int32_t LibaomAv1Encoder::Encode( absl::InlinedVector supported_formats = {VideoFrameBuffer::Type::kI420, VideoFrameBuffer::Type::kNV12}; + + scoped_refptr scaled_image; + if (!SvcEnabled() || + last_active_layer_ + 1 == svc_params_->number_spatial_layers) { + scaled_image = buffer; + } else { + scaled_image = buffer->Scale( + encoder_settings_.spatialLayers[last_active_layer_].width, + encoder_settings_.spatialLayers[last_active_layer_].height); + } + scoped_refptr mapped_buffer; - if (buffer->type() != VideoFrameBuffer::Type::kNative) { + if (scaled_image->type() != VideoFrameBuffer::Type::kNative) { // `buffer` is already mapped. - mapped_buffer = buffer; + mapped_buffer = scaled_image; } else { // Attempt to map to one of the supported formats. - mapped_buffer = buffer->GetMappedFrameBuffer(supported_formats); + mapped_buffer = scaled_image->GetMappedFrameBuffer(supported_formats); } // Convert input frame to I420, if needed. @@ -610,7 +684,8 @@ int32_t LibaomAv1Encoder::Encode( case VideoFrameBuffer::Type::kI420: case VideoFrameBuffer::Type::kI420A: { // Set frame_for_encode_ data pointers and strides. - MaybeRewrapImgWithFormat(AOM_IMG_FMT_I420); + MaybeRewrapImgWithFormat(AOM_IMG_FMT_I420, mapped_buffer->width(), + mapped_buffer->height()); auto i420_buffer = mapped_buffer->GetI420(); RTC_DCHECK(i420_buffer); RTC_CHECK_EQ(i420_buffer->width(), frame_for_encode_->d_w); @@ -627,7 +702,8 @@ int32_t LibaomAv1Encoder::Encode( break; } case VideoFrameBuffer::Type::kNV12: { - MaybeRewrapImgWithFormat(AOM_IMG_FMT_NV12); + MaybeRewrapImgWithFormat(AOM_IMG_FMT_NV12, mapped_buffer->width(), + mapped_buffer->height()); const NV12BufferInterface* nv12_buffer = mapped_buffer->GetNV12(); RTC_DCHECK(nv12_buffer); RTC_CHECK_EQ(nv12_buffer->width(), frame_for_encode_->d_w); @@ -725,10 +801,10 @@ int32_t LibaomAv1Encoder::Encode( // If encoded image width/height info are added to aom_codec_cx_pkt_t, // use those values in lieu of the values in frame. if (svc_params_) { - int n = svc_params_->scaling_factor_num[layer_frame->SpatialId()]; - int d = svc_params_->scaling_factor_den[layer_frame->SpatialId()]; - encoded_image._encodedWidth = cfg_.g_w * n / d; - encoded_image._encodedHeight = cfg_.g_h * n / d; + int n = scaling_factors_num_[layer_frame->SpatialId()]; + int d = scaling_factors_den_[layer_frame->SpatialId()]; + encoded_image._encodedWidth = encoder_settings_.width * n / d; + encoded_image._encodedHeight = encoder_settings_.height * n / d; encoded_image.SetSpatialIndex(layer_frame->SpatialId()); encoded_image.SetTemporalIndex(layer_frame->TemporalId()); } else { @@ -762,10 +838,11 @@ int32_t LibaomAv1Encoder::Encode( if (SvcEnabled()) { resolutions.resize(svc_params_->number_spatial_layers); for (int sid = 0; sid < svc_params_->number_spatial_layers; ++sid) { - int n = svc_params_->scaling_factor_num[sid]; - int d = svc_params_->scaling_factor_den[sid]; + int n = scaling_factors_num_[sid]; + int d = scaling_factors_den_[sid]; resolutions[sid] = - RenderResolution(cfg_.g_w * n / d, cfg_.g_h * n / d); + RenderResolution(encoder_settings_.width * n / d, + encoder_settings_.height * n / d); } } else { resolutions = {RenderResolution(cfg_.g_w, cfg_.g_h)}; @@ -807,11 +884,6 @@ void LibaomAv1Encoder::SetRates(const RateControlParameters& parameters) { // total target bitrate is not updated first a division by zero could happen. svc_controller_->OnRatesUpdated(parameters.bitrate); cfg_.rc_target_bitrate = parameters.bitrate.get_sum_kbps(); - aom_codec_err_t error_code = aom_codec_enc_config_set(&ctx_, &cfg_); - if (error_code != AOM_CODEC_OK) { - RTC_LOG(LS_WARNING) << "Error configuring encoder, error code: " - << error_code; - } if (SvcEnabled()) { for (int sid = 0; sid < svc_params_->number_spatial_layers; ++sid) { @@ -824,9 +896,17 @@ void LibaomAv1Encoder::SetRates(const RateControlParameters& parameters) { parameters.bitrate.GetTemporalLayerSum(sid, tid) / 1000; } } + AdjustScalingFactorsForTopActiveLayer(); SetEncoderControlParameters(AV1E_SET_SVC_PARAMS, &*svc_params_); } + // AdjustScalingFactorsForTopActiveLayer() may update `cfg_`. + aom_codec_err_t error_code = aom_codec_enc_config_set(&ctx_, &cfg_); + if (error_code != AOM_CODEC_OK) { + RTC_LOG(LS_WARNING) << "Error configuring encoder, error code: " + << error_code; + } + framerate_fps_ = parameters.framerate_fps; rates_configured_ = true; diff --git a/modules/video_coding/codecs/av1/libaom_av1_encoder_unittest.cc b/modules/video_coding/codecs/av1/libaom_av1_encoder_unittest.cc index 33b750cdc6..0bbf6a2814 100644 --- a/modules/video_coding/codecs/av1/libaom_av1_encoder_unittest.cc +++ b/modules/video_coding/codecs/av1/libaom_av1_encoder_unittest.cc @@ -37,6 +37,7 @@ #include "api/video_codecs/video_codec.h" #include "api/video_codecs/video_encoder.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "modules/video_coding/codecs/av1/av1_svc_config.h" #include "modules/video_coding/codecs/test/encoded_video_frame_producer.h" #include "modules/video_coding/include/video_error_codes.h" #include "rtc_base/checks.h" @@ -68,6 +69,18 @@ VideoCodec DefaultCodecSettings() { return codec_settings; } +VideoCodec HDCodecSettings() { + VideoCodec codec_settings; + codec_settings.codecType = kVideoCodecAV1; + codec_settings.width = 1280; + codec_settings.height = 720; + codec_settings.maxFramerate = 30; + codec_settings.startBitrate = 2048; + codec_settings.maxBitrate = 2048; + codec_settings.qpMax = 63; + return codec_settings; +} + VideoEncoder::Settings DefaultEncoderSettings() { return VideoEncoder::Settings( VideoEncoder::Capabilities(/*loss_notification=*/false), @@ -592,5 +605,87 @@ TEST(LibaomAv1EncoderTest, PostEncodeFrameDrop) { RTC_CHECK_EQ(callback.frames_encoded(), 1); } +TEST(LibaomAv1EncoderTest, EnableDisableSpatialLayersWithSvcController) { + constexpr int kNumSpatialLayers = 3; + constexpr int kNumTemporalLayers = 1; + constexpr size_t kWidth = 1280; + constexpr size_t kHeight = 720; + + // Configure encoder to produce 3 spatial layers. Encode frames of layer 0 + // then enable layer 1 and encode more frames and so on. + // Then disable layers one by one in the same way. + // Note: bit rate allocation is high to avoid frame dropping due to rate + // control, the encoder should always produce a frame. A dropped + // frame indicates a problem and the test will fail. + std::unique_ptr encoder = + CreateLibaomAv1Encoder(CreateEnvironment()); + VideoCodec codec_settings = HDCodecSettings(); + SetAv1SvcConfig(codec_settings, kNumTemporalLayers, kNumSpatialLayers); + codec_settings.SetFrameDropEnabled(true); + EXPECT_EQ(encoder->InitEncode(&codec_settings, DefaultEncoderSettings()), + WEBRTC_VIDEO_CODEC_OK); + + EncodedVideoFrameProducer producer(*encoder); + producer.SetResolution({kWidth, kHeight}); + + VideoBitrateAllocation bitrate_allocation; + + // Set all layers active for initial allocation. + for (size_t sl_idx = 0; sl_idx < kNumSpatialLayers; ++sl_idx) { + // Allocate high bit rate to avoid frame dropping due to rate control. + bitrate_allocation.SetBitrate( + sl_idx, 0, + codec_settings.spatialLayers[sl_idx].targetBitrate * 1000 * 2); + } + + encoder->SetRates(VideoEncoder::RateControlParameters( + bitrate_allocation, codec_settings.maxFramerate)); + + // Encode a key frame to validate all other frames are delta frames. + std::vector frames = + producer.SetNumInputFrames(1).Encode(); + ASSERT_THAT(frames, Not(IsEmpty())); + EXPECT_TRUE(frames[0].codec_specific_info.template_structure); + + constexpr size_t kNumFramesToEncode = 5; + + // Disable layers one by one. + for (int sl_idx = kNumSpatialLayers - 1; sl_idx > 0; --sl_idx) { + bitrate_allocation.SetBitrate(sl_idx, 0, 0); + encoder->SetRates(VideoEncoder::RateControlParameters( + bitrate_allocation, codec_settings.maxFramerate)); + + frames = producer.SetNumInputFrames(kNumFramesToEncode).Encode(); + // With `sl_idx` spatial layer disabled, there are `sl_idx` spatial layers + // left. + ASSERT_THAT(frames, SizeIs(kNumFramesToEncode * sl_idx)); + for (size_t i = 0; i < frames.size(); ++i) { + EXPECT_TRUE(frames[i].codec_specific_info.generic_frame_info); + EXPECT_FALSE(frames[i].codec_specific_info.template_structure); + } + } + + // Enable layers back one by one. + for (size_t sl_idx = 1; sl_idx < kNumSpatialLayers; ++sl_idx) { + // Allocate high bit rate to avoid frame dropping due to rate control. + bitrate_allocation.SetBitrate( + sl_idx, 0, + codec_settings.spatialLayers[sl_idx].targetBitrate * 1000 * 2); + encoder->SetRates(VideoEncoder::RateControlParameters( + bitrate_allocation, codec_settings.maxFramerate)); + + frames = producer.SetNumInputFrames(kNumFramesToEncode).Encode(); + // With (sl_idx+1) spatial layers expect (sl_idx+1) frames per input frame. + ASSERT_THAT(frames, SizeIs(kNumFramesToEncode * (sl_idx + 1))); + // Only the first frame after enabling the layer must be a keyframe. + EXPECT_TRUE(frames[0].codec_specific_info.generic_frame_info); + EXPECT_TRUE(frames[0].codec_specific_info.template_structure); + for (size_t i = 1; i < frames.size(); ++i) { + EXPECT_TRUE(frames[i].codec_specific_info.generic_frame_info); + EXPECT_FALSE(frames[i].codec_specific_info.template_structure); + } + } +} + } // namespace } // namespace webrtc From 50544d8dc470205fcce35f4cea6ac373e718a358 Mon Sep 17 00:00:00 2001 From: Jonas Oreland Date: Wed, 7 May 2025 10:44:59 +0200 Subject: [PATCH 060/292] WebRTC/PQC Add EphemeralKeyExchangeCipherGroups to CryptoOptions. BUG=webrtc:404763475 No-Iwyu: Doesn't seem to understand the #ifdef surrounding open/boring ssl Change-Id: I187e5d9b4da4f1494980a8fc257fabcfcc77b2ad Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390000 Reviewed-by: Harald Alvestrand Commit-Queue: Jonas Oreland Cr-Commit-Position: refs/heads/main@{#44532} --- BUILD.gn | 2 + api/crypto/BUILD.gn | 19 +++ api/crypto/crypto_options.cc | 55 ++++++++- api/crypto/crypto_options.h | 41 +++++++ api/crypto/crypto_options_unittest.cc | 89 ++++++++++++++ experiments/field_trials.py | 3 + p2p/dtls/dtls_ice_integrationtest.cc | 14 ++- p2p/dtls/dtls_transport.cc | 7 ++ p2p/dtls/dtls_transport.h | 2 + p2p/dtls/dtls_transport_unittest.cc | 28 +++-- pc/peer_connection.cc | 4 + rtc_base/BUILD.gn | 2 + rtc_base/openssl_stream_adapter.cc | 22 ++-- rtc_base/openssl_stream_adapter.h | 14 ++- rtc_base/openssl_stream_adapter_unittest.cc | 129 ++++++++++++++++++++ rtc_base/ssl_stream_adapter.cc | 73 +++++++++++ rtc_base/ssl_stream_adapter.h | 13 +- rtc_base/ssl_stream_adapter_unittest.cc | 62 ++++++---- 18 files changed, 522 insertions(+), 57 deletions(-) create mode 100644 api/crypto/crypto_options_unittest.cc create mode 100644 rtc_base/openssl_stream_adapter_unittest.cc diff --git a/BUILD.gn b/BUILD.gn index 2d142310c3..2d84f3cae7 100644 --- a/BUILD.gn +++ b/BUILD.gn @@ -633,6 +633,8 @@ if (rtc_include_tests && !build_with_chromium) { "api:rtc_api_unittests", "api/audio:audio_api_unittests", "api/audio_codecs/test:audio_codecs_api_unittests", + "api/crypto:crypto_options_unittest", + "api/crypto:options", "api/numerics:numerics_unittests", "api/task_queue:pending_task_safety_flag_unittests", "api/test/metrics:metrics_unittests", diff --git a/api/crypto/BUILD.gn b/api/crypto/BUILD.gn index 2970f341b1..7bfae89caf 100644 --- a/api/crypto/BUILD.gn +++ b/api/crypto/BUILD.gn @@ -23,6 +23,7 @@ rtc_library("options") { "crypto_options.h", ] deps = [ + "..:field_trials_view", "../../rtc_base:checks", "../../rtc_base:ssl_adapter", "../../rtc_base/system:rtc_export", @@ -50,3 +51,21 @@ rtc_source_set("frame_encryptor_interface") { "../../rtc_base:refcount", ] } + +if (rtc_include_tests) { + rtc_library("crypto_options_unittest") { + visibility = [ "*" ] + testonly = true + sources = [ "crypto_options_unittest.cc" ] + deps = [ + ":options", + "../../rtc_base:ssl_adapter", + "../../test:scoped_key_value_config", + "../../test:test_support", + "//testing/gtest", + ] + if (rtc_build_ssl) { + deps += [ "//third_party/boringssl" ] + } + } +} diff --git a/api/crypto/crypto_options.cc b/api/crypto/crypto_options.cc index 2923833c2f..53c1cf06ae 100644 --- a/api/crypto/crypto_options.cc +++ b/api/crypto/crypto_options.cc @@ -10,8 +10,12 @@ #include "api/crypto/crypto_options.h" +#include +#include +#include #include +#include "api/field_trials_view.h" #include "rtc_base/checks.h" #include "rtc_base/ssl_stream_adapter.h" @@ -69,6 +73,7 @@ bool CryptoOptions::operator==(const CryptoOptions& other) const { struct SFrame { bool require_frame_encryption; } sframe; + EphemeralKeyExchangeCipherGroups ephemeral_key_exchange_cipher_groups; }; static_assert(sizeof(data_being_tested_for_equality) == sizeof(*this), "Did you add something to CryptoOptions and forget to " @@ -82,11 +87,59 @@ bool CryptoOptions::operator==(const CryptoOptions& other) const { srtp.enable_encrypted_rtp_header_extensions == other.srtp.enable_encrypted_rtp_header_extensions && sframe.require_frame_encryption == - other.sframe.require_frame_encryption; + other.sframe.require_frame_encryption && + ephemeral_key_exchange_cipher_groups == + other.ephemeral_key_exchange_cipher_groups; } bool CryptoOptions::operator!=(const CryptoOptions& other) const { return !(*this == other); } +CryptoOptions::EphemeralKeyExchangeCipherGroups:: + EphemeralKeyExchangeCipherGroups() + : enabled_(SSLStreamAdapter::GetDefaultEphemeralKeyExchangeCipherGroups( + /* field_trials= */ nullptr)) {} + +bool CryptoOptions::EphemeralKeyExchangeCipherGroups::operator==( + const CryptoOptions::EphemeralKeyExchangeCipherGroups& other) const { + return enabled_ == other.enabled_; +} + +std::set +CryptoOptions::EphemeralKeyExchangeCipherGroups::GetSupported() { + return SSLStreamAdapter::GetSupportedEphemeralKeyExchangeCipherGroups(); +} + +void CryptoOptions::EphemeralKeyExchangeCipherGroups::AddFirst(uint16_t group) { + enabled_.erase(std::remove(enabled_.begin(), enabled_.end(), group)); + enabled_.insert(enabled_.begin(), group); +} + +void CryptoOptions::EphemeralKeyExchangeCipherGroups::Update( + const FieldTrialsView* field_trials, + const std::vector* disabled_groups) { + // Note: assumption is that these lists contains few elements...so converting + // to set<> is not worth it. + std::vector current; + enabled_ = SSLStreamAdapter::GetDefaultEphemeralKeyExchangeCipherGroups( + field_trials); + // Remove all disabled. + if (disabled_groups) { + enabled_.erase( + std::remove_if(enabled_.begin(), enabled_.end(), [&](uint16_t val) { + return std::find(disabled_groups->begin(), disabled_groups->end(), + val) != disabled_groups->end(); + })); + } + + // Add all current not already present to end of list. + auto end = enabled_.end(); + for (auto val : current) { + if (std::find(enabled_.begin(), end, val) == end) { + enabled_.push_back(val); + } + } +} + } // namespace webrtc diff --git a/api/crypto/crypto_options.h b/api/crypto/crypto_options.h index a937490787..0840d8d5df 100644 --- a/api/crypto/crypto_options.h +++ b/api/crypto/crypto_options.h @@ -11,8 +11,13 @@ #ifndef API_CRYPTO_CRYPTO_OPTIONS_H_ #define API_CRYPTO_CRYPTO_OPTIONS_H_ +#include +#include +#include +#include #include +#include "api/field_trials_view.h" #include "rtc_base/system/rtc_export.h" namespace webrtc { @@ -66,6 +71,42 @@ struct RTC_EXPORT CryptoOptions { // FrameDecryptor attached to them before they are able to receive packets. bool require_frame_encryption = false; } sframe; + + // Cipher groups used by DTLS when establishing an ephemeral key during + // handshake. + class EphemeralKeyExchangeCipherGroups { + public: + // Which cipher groups are supported by this binary, + // - ssl.h: SSL_GROUP_{} + // - https://www.rfc-editor.org/rfc/rfc8422#section-5.1.1 + // - https://datatracker.ietf.org/doc/draft-ietf-tls-mlkem + static constexpr uint16_t kSECP224R1 = 21; + static constexpr uint16_t kSECP256R1 = 23; + static constexpr uint16_t kSECP384R1 = 24; + static constexpr uint16_t kSECP521R1 = 25; + static constexpr uint16_t kX25519 = 29; + static constexpr uint16_t kX25519_MLKEM768 = 0x11ec; + + static std::set GetSupported(); + static std::optional GetName(uint16_t); + + EphemeralKeyExchangeCipherGroups(); + + // Which cipher groups are enabled in this crypto options. + std::vector GetEnabled() const { return enabled_; } + void SetEnabled(const std::vector& groups); + void AddFirst(uint16_t group); + + // Update list of enabled groups based on field_trials, + // optionally providing list of groups that should NOT be added. + void Update(const FieldTrialsView* field_trials, + const std::vector* disabled_groups = nullptr); + + bool operator==(const EphemeralKeyExchangeCipherGroups& other) const; + + private: + std::vector enabled_; + } ephemeral_key_exchange_cipher_groups; }; } // namespace webrtc diff --git a/api/crypto/crypto_options_unittest.cc b/api/crypto/crypto_options_unittest.cc new file mode 100644 index 0000000000..2b79118d63 --- /dev/null +++ b/api/crypto/crypto_options_unittest.cc @@ -0,0 +1,89 @@ +/* + * Copyright 2025 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "api/crypto/crypto_options.h" + +#include +#include +#include + +#include "rtc_base/openssl_stream_adapter.h" +#include "test/gtest.h" +#include "test/scoped_key_value_config.h" + +TEST(EphemeralKeyExchangeCipherGroupsTest, GetSupported) { + std::set expected = { +#ifdef SSL_GROUP_SECP224R1 + SSL_GROUP_SECP224R1, +#endif +#ifdef SSL_GROUP_SECP256R1 + SSL_GROUP_SECP256R1, +#endif +#ifdef SSL_GROUP_SECP384R1 + SSL_GROUP_SECP384R1, +#endif +#ifdef SSL_GROUP_SECP521R1 + SSL_GROUP_SECP521R1, +#endif +#ifdef SSL_GROUP_X25519 + SSL_GROUP_X25519, +#endif +#ifdef SSL_GROUP_X25519_MLKEM768 + SSL_GROUP_X25519_MLKEM768, +#endif + }; + auto supported = + webrtc::CryptoOptions::EphemeralKeyExchangeCipherGroups::GetSupported(); + for (auto group : expected) { + EXPECT_TRUE(supported.contains(group)); + } +} + +TEST(EphemeralKeyExchangeCipherGroupsTest, GetEnabled) { + std::vector expected = { +#ifdef SSL_GROUP_X25519 + SSL_GROUP_X25519, +#endif +#ifdef SSL_GROUP_SECP256R1 + SSL_GROUP_SECP256R1, +#endif +#ifdef SSL_GROUP_SECP384R1 + SSL_GROUP_SECP384R1, +#endif + }; + webrtc::CryptoOptions::EphemeralKeyExchangeCipherGroups groups; + EXPECT_EQ(groups.GetEnabled(), expected); +} + +TEST(EphemeralKeyExchangeCipherGroupsTest, Update) { + std::vector expected = { +#ifdef SSL_GROUP_X25519_MLKEM768 + SSL_GROUP_X25519_MLKEM768, +#endif +#ifdef SSL_GROUP_SECP256R1 + SSL_GROUP_SECP256R1, +#endif +#ifdef SSL_GROUP_SECP384R1 + SSL_GROUP_SECP384R1, +#endif + }; + + std::vector disable = { +#ifdef SSL_GROUP_X25519 + SSL_GROUP_X25519, +#endif + }; + + webrtc::CryptoOptions::EphemeralKeyExchangeCipherGroups groups; + webrtc::test::ScopedKeyValueConfig field_trials( + "WebRTC-EnableDtlsPqc/Enabled/"); + groups.Update(&field_trials, &disable); + EXPECT_EQ(groups.GetEnabled(), expected); +} diff --git a/experiments/field_trials.py b/experiments/field_trials.py index bf6b833970..ec7e88a673 100755 --- a/experiments/field_trials.py +++ b/experiments/field_trials.py @@ -80,6 +80,9 @@ def bug_url(self) -> str: FieldTrial('WebRTC-DisableRtxRateLimiter', 42225500, date(2024, 4, 1)), + FieldTrial('WebRTC-DisableSslGroupIds', + 404763475, + date(2025,9,1)), FieldTrial('WebRTC-ElasticBitrateAllocation', 350555527, date(2025, 3, 1)), diff --git a/p2p/dtls/dtls_ice_integrationtest.cc b/p2p/dtls/dtls_ice_integrationtest.cc index d45eeda2e0..01ebb8bd42 100644 --- a/p2p/dtls/dtls_ice_integrationtest.cc +++ b/p2p/dtls/dtls_ice_integrationtest.cc @@ -15,7 +15,6 @@ #include #include -#include "absl/strings/str_cat.h" #include "api/candidate.h" #include "api/crypto/crypto_options.h" #include "api/environment/environment.h" @@ -82,9 +81,8 @@ class DtlsIceIntegrationTest : public ::testing::TestWithParam( client ? "client_transport" : "server_transport", 0, ep.allocator.get(), &ep.env.field_trials()); + CryptoOptions crypto_options; + if (ep.pqc) { + FieldTrials field_trials("WebRTC-EnableDtlsPqc/Enabled/"); + crypto_options.ephemeral_key_exchange_cipher_groups.Update( + &field_trials); + } ep.dtls = std::make_unique( - ep.ice.get(), webrtc::CryptoOptions(), + ep.ice.get(), crypto_options, /*event_log=*/nullptr, std::get<2>(GetParam())); // Enable(or disable) the dtls_in_stun parameter before diff --git a/p2p/dtls/dtls_transport.cc b/p2p/dtls/dtls_transport.cc index f8d118a8ac..4eff3b1953 100644 --- a/p2p/dtls/dtls_transport.cc +++ b/p2p/dtls/dtls_transport.cc @@ -198,6 +198,8 @@ DtlsTransportInternalImpl::DtlsTransportInternalImpl( ice_transport_(ice_transport), downward_(nullptr), srtp_ciphers_(crypto_options.GetSupportedDtlsSrtpCryptoSuites()), + ephemeral_key_exchange_cipher_groups_( + crypto_options.ephemeral_key_exchange_cipher_groups.GetEnabled()), ssl_max_version_(max_version), event_log_(event_log), dtls_stun_piggyback_controller_( @@ -487,6 +489,11 @@ bool DtlsTransportInternalImpl::SetupDtls() { RTC_LOG(LS_INFO) << ToString() << ": Not using DTLS-SRTP."; } + if (!dtls_->SetSslGroupIds(ephemeral_key_exchange_cipher_groups_)) { + RTC_LOG(LS_ERROR) << ToString() << ": Couldn't set DTLS SSL Group Ids."; + return false; + } + RTC_LOG(LS_INFO) << ToString() << ": DTLS setup complete, dtls_in_stun: " << dtls_in_stun_; diff --git a/p2p/dtls/dtls_transport.h b/p2p/dtls/dtls_transport.h index 1f099ee757..cacd77f451 100644 --- a/p2p/dtls/dtls_transport.h +++ b/p2p/dtls/dtls_transport.h @@ -280,6 +280,8 @@ class DtlsTransportInternalImpl : public webrtc::DtlsTransportInternal { StreamInterfaceChannel* downward_; // Wrapper for ice_transport_, owned by dtls_. const std::vector srtp_ciphers_; // SRTP ciphers to use with DTLS. + // Cipher groups used for DTLS handshake to establish ephemeral key. + const std::vector ephemeral_key_exchange_cipher_groups_; bool dtls_active_ = false; scoped_refptr local_certificate_; std::optional dtls_role_; diff --git a/p2p/dtls/dtls_transport_unittest.cc b/p2p/dtls/dtls_transport_unittest.cc index 0440516458..b8cfd06074 100644 --- a/p2p/dtls/dtls_transport_unittest.cc +++ b/p2p/dtls/dtls_transport_unittest.cc @@ -28,6 +28,7 @@ #include "api/array_view.h" #include "api/crypto/crypto_options.h" #include "api/dtls_transport_interface.h" +#include "api/field_trials.h" #include "api/scoped_refptr.h" #include "api/test/rtc_error_matchers.h" #include "api/transport/stun.h" @@ -107,19 +108,19 @@ class DtlsTestClient : public sigslot::has_slots<> { void set_async_delay(int async_delay_ms) { async_delay_ms_ = async_delay_ms; } // Set up fake ICE transport and real DTLS transport under test. - void SetupTransports(IceRole role, - bool rtt_estimate = true, - absl::string_view field_trials_string = "") { + void SetupTransports(IceRole role, bool rtt_estimate = true) { dtls_transport_ = nullptr; fake_ice_transport_ = nullptr; - if (field_trials_string.empty() && pqc_) { - field_trials_string = "WebRTC-EnableDtlsPqc/Enabled/"; + webrtc::CryptoOptions crypto_options; + if (pqc_) { + FieldTrials field_trials("WebRTC-EnableDtlsPqc/Enabled/"); + crypto_options.ephemeral_key_exchange_cipher_groups.Update(&field_trials); } fake_ice_transport_.reset(new FakeIceTransport( absl::StrCat("fake-", name_), 0, - /* network_thread= */ nullptr, field_trials_string)); + /* network_thread= */ nullptr, /* field_trials_string= */ "")); if (rtt_estimate) { fake_ice_transport_->set_rtt_estimate( async_delay_ms_ ? std::optional(async_delay_ms_) : std::nullopt, @@ -137,7 +138,7 @@ class DtlsTestClient : public sigslot::has_slots<> { }); dtls_transport_ = std::make_unique( - fake_ice_transport_.get(), CryptoOptions(), + fake_ice_transport_.get(), crypto_options, /*event_log=*/nullptr, ssl_max_version_); // Note: Certificate may be null here if testing passthrough. dtls_transport_->SetLocalCertificate(certificate_); @@ -789,12 +790,13 @@ class DtlsTransportInternalImplVersionTest client1_.set_async_delay(50); client2_.set_async_delay(50); - client1_.SetupTransports( - config1.ice_role.value_or(ICEROLE_CONTROLLING), rtt_estimate, - config1.pqc ? "WebRTC-EnableDtlsPqc/Enabled/" : ""); - client2_.SetupTransports( - config2.ice_role.value_or(ICEROLE_CONTROLLED), rtt_estimate, - config2.pqc ? "WebRTC-EnableDtlsPqc/Enabled/" : ""); + client1_.SetPqc(config1.pqc); + client2_.SetPqc(config2.pqc); + + client1_.SetupTransports(config1.ice_role.value_or(ICEROLE_CONTROLLING), + rtt_estimate); + client2_.SetupTransports(config2.ice_role.value_or(ICEROLE_CONTROLLED), + rtt_estimate); client1_.dtls_transport()->SetDtlsRole( config1.ssl_role.value_or(SSL_CLIENT)); client2_.dtls_transport()->SetDtlsRole( diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc index 74c4c4531e..fcb2975270 100644 --- a/pc/peer_connection.cc +++ b/pc/peer_connection.cc @@ -691,6 +691,10 @@ JsepTransportController* PeerConnection::InitializeTransportController_n( config.crypto_options = configuration.crypto_options.has_value() ? *configuration.crypto_options : options_.crypto_options; + + // Maybe enable PQC from FieldTrials + config.crypto_options.ephemeral_key_exchange_cipher_groups.Update( + &context_->env().field_trials()); config.transport_observer = this; config.rtcp_handler = InitializeRtcpCallback(); config.un_demuxable_packet_handler = InitializeUnDemuxablePacketHandler(); diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn index 88164115c4..239adb82c7 100644 --- a/rtc_base/BUILD.gn +++ b/rtc_base/BUILD.gn @@ -2116,6 +2116,7 @@ if (rtc_include_tests) { "message_digest_unittest.cc", "network_route_unittest.cc", "network_unittest.cc", + "openssl_stream_adapter_unittest.cc", "rolling_accumulator_unittest.cc", "rtc_certificate_generator_unittest.cc", "rtc_certificate_unittest.cc", @@ -2171,6 +2172,7 @@ if (rtc_include_tests) { "../api:rtc_error_matchers", "../api:scoped_refptr", "../api:sequence_checker", + "../api/crypto:options", "../api/environment", "../api/environment:environment_factory", "../api/task_queue", diff --git a/rtc_base/openssl_stream_adapter.cc b/rtc_base/openssl_stream_adapter.cc index b46faff911..654fc46821 100644 --- a/rtc_base/openssl_stream_adapter.cc +++ b/rtc_base/openssl_stream_adapter.cc @@ -303,8 +303,8 @@ OpenSSLStreamAdapter::OpenSSLStreamAdapter( ssl_mode_(webrtc::SSL_MODE_DTLS), ssl_max_version_(webrtc::SSL_PROTOCOL_DTLS_12), force_dtls_13_(GetForceDtls13(field_trials)), - enable_dtls_pqc_(field_trials && - field_trials->IsEnabled("WebRTC-EnableDtlsPqc")) { + disable_ssl_group_ids_(field_trials && field_trials->IsEnabled( + "WebRTC-DisableSslGroupIds")) { stream_->SetEventCallback( [this](int events, int err) { OnEvent(events, err); }); } @@ -436,7 +436,7 @@ bool OpenSSLStreamAdapter::GetSslVersionBytes(int* version) const { return true; } -uint16_t OpenSSLStreamAdapter::GetSslGroupIdForTesting() const { +uint16_t OpenSSLStreamAdapter::GetSslGroupId() const { if (state_ != SSL_CONNECTED) { return 0; } @@ -903,11 +903,9 @@ int OpenSSLStreamAdapter::BeginSSL() { DTLSv1_set_initial_timeout_duration(ssl_, dtls_handshake_timeout_ms_); } - // Experimental code guarded by WebRTC-EnableDtlsPqc. - if (enable_dtls_pqc_) { - const uint16_t kGroups[] = {SSL_GROUP_X25519_MLKEM768, SSL_GROUP_X25519, - SSL_GROUP_SECP256R1, SSL_GROUP_SECP384R1}; - if (!SSL_set1_group_ids(ssl_, kGroups, std::size(kGroups))) { + if (!disable_ssl_group_ids_) { + if (!SSL_set1_group_ids(ssl_, ssl_cipher_groups_.data(), + ssl_cipher_groups_.size())) { RTC_LOG(LS_WARNING) << "Failed to call SSL_set1_group_ids."; return -1; } @@ -1356,4 +1354,12 @@ SSLProtocolVersion OpenSSLStreamAdapter::GetMaxSupportedDTLSProtocolVersion() { #endif } +bool OpenSSLStreamAdapter::SetSslGroupIds(const std::vector& groups) { + if (state_ != SSL_NONE) { + return false; + } + ssl_cipher_groups_ = groups; + return true; +} + } // namespace webrtc diff --git a/rtc_base/openssl_stream_adapter.h b/rtc_base/openssl_stream_adapter.h index 4259f8c1b5..257f100ef5 100644 --- a/rtc_base/openssl_stream_adapter.h +++ b/rtc_base/openssl_stream_adapter.h @@ -136,9 +136,13 @@ class OpenSSLStreamAdapter final : public SSLStreamAdapter { // Used for testing (and maybe put into stats?). int GetRetransmissionCount() const override { return retransmission_count_; } + // Set cipher group ids to use during DTLS handshake to establish ephemeral + // key, see CryptoOptions::EphemeralKeyExchangeCipherGroups. + bool SetSslGroupIds(const std::vector& group_ids) override; + // Return the the ID of the group used by the adapters most recently // completed handshake, or 0 if not applicable (e.g. before the handshake). - uint16_t GetSslGroupIdForTesting() const override; + uint16_t GetSslGroupId() const override; private: enum SSLState { @@ -241,6 +245,9 @@ class OpenSSLStreamAdapter final : public SSLStreamAdapter { // The DtlsSrtp ciphers std::string srtp_ciphers_; + // The ssl cipher groups to be used for DTLS handshake. + std::vector ssl_cipher_groups_; + // Do DTLS or not SSLMode ssl_mode_; @@ -261,8 +268,9 @@ class OpenSSLStreamAdapter final : public SSLStreamAdapter { int retransmission_count_ = 0; - // Experimental flag to enable Post-Quantum Cryptography TLS. - const bool enable_dtls_pqc_ = false; + // Kill switch (from field-trial) flag to disable the use of + // SSL_set_group_ids. + const bool disable_ssl_group_ids_ = false; }; ///////////////////////////////////////////////////////////////////////////// diff --git a/rtc_base/openssl_stream_adapter_unittest.cc b/rtc_base/openssl_stream_adapter_unittest.cc new file mode 100644 index 0000000000..9513ad60ab --- /dev/null +++ b/rtc_base/openssl_stream_adapter_unittest.cc @@ -0,0 +1,129 @@ +/* + * Copyright 2025 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "rtc_base/openssl_stream_adapter.h" + +#include + +#include +#include +#include + +#include "rtc_base/logging.h" +#include "rtc_base/ssl_stream_adapter.h" +#include "test/gtest.h" +#include "test/scoped_key_value_config.h" + +TEST(OpenSSLStreamAdapterTest, GetSupportedEphemeralKeyExchangeCipherGroups) { + RTC_LOG(LS_INFO) << "OpenSSLStreamAdapter::IsBoringSsl(): " + << webrtc::OpenSSLStreamAdapter::IsBoringSsl(); + std::set expected = { +#ifdef SSL_GROUP_SECP224R1 + SSL_GROUP_SECP224R1, +#endif +#ifdef SSL_GROUP_SECP256R1 + SSL_GROUP_SECP256R1, +#endif +#ifdef SSL_GROUP_SECP384R1 + SSL_GROUP_SECP384R1, +#endif +#ifdef SSL_GROUP_SECP521R1 + SSL_GROUP_SECP521R1, +#endif +#ifdef SSL_GROUP_X25519 + SSL_GROUP_X25519, +#endif +#ifdef SSL_GROUP_X25519_MLKEM768 + SSL_GROUP_X25519_MLKEM768, +#endif + }; + EXPECT_EQ( + webrtc::SSLStreamAdapter::GetSupportedEphemeralKeyExchangeCipherGroups(), + expected); +} + +TEST(OpenSSLStreamAdapterTest, GetEphemeralKeyExchangeCipherGroupName) { +#ifdef SSL_GROUP_SECP224R1 + EXPECT_EQ(*webrtc::SSLStreamAdapter::GetEphemeralKeyExchangeCipherGroupName( + SSL_GROUP_SECP224R1), + "P-224"); +#endif +#ifdef SSL_GROUP_SECP256R1 + EXPECT_EQ(*webrtc::SSLStreamAdapter::GetEphemeralKeyExchangeCipherGroupName( + SSL_GROUP_SECP256R1), + "P-256"); +#endif +#ifdef SSL_GROUP_SECP384R1 + EXPECT_EQ(*webrtc::SSLStreamAdapter::GetEphemeralKeyExchangeCipherGroupName( + SSL_GROUP_SECP384R1), + "P-384"); +#endif +#ifdef SSL_GROUP_SECP521R1 + EXPECT_EQ(*webrtc::SSLStreamAdapter::GetEphemeralKeyExchangeCipherGroupName( + SSL_GROUP_SECP521R1), + "P-521"); +#endif +#ifdef SSL_GROUP_X25519 + EXPECT_EQ(*webrtc::SSLStreamAdapter::GetEphemeralKeyExchangeCipherGroupName( + SSL_GROUP_X25519), + "X25519"); +#endif +#ifdef SSL_GROUP_X25519_MLKEM768 + EXPECT_EQ(*webrtc::SSLStreamAdapter::GetEphemeralKeyExchangeCipherGroupName( + SSL_GROUP_X25519_MLKEM768), + "X25519MLKEM768"); +#endif + + EXPECT_FALSE( + webrtc::SSLStreamAdapter::GetEphemeralKeyExchangeCipherGroupName(0) + .has_value()); +} + +TEST(OpenSSLStreamAdapterTest, GetDefaultEphemeralKeyExchangeCipherGroups) { + std::vector expected = { +#ifdef SSL_GROUP_X25519 + SSL_GROUP_X25519, +#endif +#ifdef SSL_GROUP_SECP256R1 + SSL_GROUP_SECP256R1, +#endif +#ifdef SSL_GROUP_SECP384R1 + SSL_GROUP_SECP384R1, +#endif + }; + EXPECT_EQ( + webrtc::SSLStreamAdapter::GetDefaultEphemeralKeyExchangeCipherGroups( + /* field_trials= */ nullptr), + expected); +} + +TEST(OpenSSLStreamAdapterTest, + GetDefaultEphemeralKeyExchangeCipherGroupsWithPQC) { + std::vector expected = { +#ifdef SSL_GROUP_X25519_MLKEM768 + SSL_GROUP_X25519_MLKEM768, +#endif +#ifdef SSL_GROUP_X25519 + SSL_GROUP_X25519, +#endif +#ifdef SSL_GROUP_SECP256R1 + SSL_GROUP_SECP256R1, +#endif +#ifdef SSL_GROUP_SECP384R1 + SSL_GROUP_SECP384R1, +#endif + }; + webrtc::test::ScopedKeyValueConfig field_trials( + "WebRTC-EnableDtlsPqc/Enabled/"); + EXPECT_EQ( + webrtc::SSLStreamAdapter::GetDefaultEphemeralKeyExchangeCipherGroups( + &field_trials), + expected); +} diff --git a/rtc_base/ssl_stream_adapter.cc b/rtc_base/ssl_stream_adapter.cc index 4f74d87274..02ee72948b 100644 --- a/rtc_base/ssl_stream_adapter.cc +++ b/rtc_base/ssl_stream_adapter.cc @@ -13,8 +13,11 @@ #include #include #include +#include +#include #include #include +#include #include "absl/functional/any_invocable.h" #include "absl/strings/string_view.h" @@ -100,6 +103,76 @@ bool SSLStreamAdapter::IsAcceptableCipher(absl::string_view cipher, return OpenSSLStreamAdapter::IsAcceptableCipher(cipher, key_type); } +std::optional +SSLStreamAdapter::GetEphemeralKeyExchangeCipherGroupName(uint16_t group_id) { +#if defined(OPENSSL_IS_BORINGSSL) + auto val = SSL_get_group_name(group_id); + if (val != nullptr) { + return std::string(val); + } +#endif + return std::nullopt; +} + +std::set +SSLStreamAdapter::GetSupportedEphemeralKeyExchangeCipherGroups() { + return { + // It would be nice if BoringSSL had a function like this! +#ifdef SSL_GROUP_SECP224R1 + SSL_GROUP_SECP224R1, +#endif +#ifdef SSL_GROUP_SECP256R1 + SSL_GROUP_SECP256R1, +#endif +#ifdef SSL_GROUP_SECP384R1 + SSL_GROUP_SECP384R1, +#endif +#ifdef SSL_GROUP_SECP521R1 + SSL_GROUP_SECP521R1, +#endif +#ifdef SSL_GROUP_X25519 + SSL_GROUP_X25519, +#endif +#ifdef SSL_GROUP_X25519_MLKEM768 + SSL_GROUP_X25519_MLKEM768, +#endif + }; +} + +std::vector +SSLStreamAdapter::GetDefaultEphemeralKeyExchangeCipherGroups( + const FieldTrialsView* field_trials) { + // It would be nice if BoringSSL had a function like this! + // from boringssl/src/ssl/extensions.cc kDefaultGroups. + if (field_trials && field_trials->IsEnabled("WebRTC-EnableDtlsPqc")) { + return { +#ifdef SSL_GROUP_X25519_MLKEM768 + SSL_GROUP_X25519_MLKEM768, +#endif +#ifdef SSL_GROUP_X25519 + SSL_GROUP_X25519, +#endif +#ifdef SSL_GROUP_SECP256R1 + SSL_GROUP_SECP256R1, +#endif +#ifdef SSL_GROUP_SECP384R1 + SSL_GROUP_SECP384R1, +#endif + }; + } + return { +#ifdef SSL_GROUP_X25519 + SSL_GROUP_X25519, +#endif +#ifdef SSL_GROUP_SECP256R1 + SSL_GROUP_SECP256R1, +#endif +#ifdef SSL_GROUP_SECP384R1 + SSL_GROUP_SECP384R1, +#endif + }; +} + // Default shim for backward compat. bool SSLStreamAdapter::SetPeerCertificateDigest( absl::string_view digest_alg, diff --git a/rtc_base/ssl_stream_adapter.h b/rtc_base/ssl_stream_adapter.h index bc62f8eacb..f8f817337a 100644 --- a/rtc_base/ssl_stream_adapter.h +++ b/rtc_base/ssl_stream_adapter.h @@ -16,6 +16,7 @@ #include #include +#include #include #include @@ -244,6 +245,12 @@ class SSLStreamAdapter : public StreamInterface { static bool IsAcceptableCipher(int cipher, KeyType key_type); static bool IsAcceptableCipher(absl::string_view cipher, KeyType key_type); + static std::set GetSupportedEphemeralKeyExchangeCipherGroups(); + static std::optional GetEphemeralKeyExchangeCipherGroupName( + uint16_t); + static std::vector GetDefaultEphemeralKeyExchangeCipherGroups( + const FieldTrialsView* field_trials); + //////////////////////////////////////////////////////////////////////////// // Testing only member functions //////////////////////////////////////////////////////////////////////////// @@ -270,9 +277,13 @@ class SSLStreamAdapter : public StreamInterface { // Used for testing (and maybe put into stats?). virtual int GetRetransmissionCount() const = 0; + // Set cipher group ids to use during DTLS handshake to establish ephemeral + // key, see CryptoOptions::EphemeralKeyExchangeCipherGroups. + virtual bool SetSslGroupIds(const std::vector& group_ids) = 0; + // Return the the ID of the group used by the adapters most recently // completed handshake, or 0 if not applicable (e.g. before the handshake). - virtual uint16_t GetSslGroupIdForTesting() const = 0; + virtual uint16_t GetSslGroupId() const = 0; private: // If true (default), the client is required to provide a certificate during diff --git a/rtc_base/ssl_stream_adapter_unittest.cc b/rtc_base/ssl_stream_adapter_unittest.cc index 8334934b06..fe2430ee35 100644 --- a/rtc_base/ssl_stream_adapter_unittest.cc +++ b/rtc_base/ssl_stream_adapter_unittest.cc @@ -33,6 +33,7 @@ #include "absl/memory/memory.h" #include "absl/strings/string_view.h" #include "api/array_view.h" +#include "api/crypto/crypto_options.h" #include "api/sequence_checker.h" #include "api/task_queue/pending_task_safety_flag.h" #include "api/test/rtc_error_matchers.h" @@ -1589,35 +1590,44 @@ TEST_P(SSLStreamAdapterTestDTLSHandshakeVersion, TestGetSslCipherSuite) { } #ifdef OPENSSL_IS_BORINGSSL -class SSLStreamAdapterTestDTLSPqc : public SSLStreamAdapterTestDTLSBase { - public: - SSLStreamAdapterTestDTLSPqc() : SSLStreamAdapterTestDTLSBase("", "") {} - - protected: - void SetUp() override { - std::string pqc_trial = "WebRTC-EnableDtlsPqc/Enabled/"; - - InitializeClientAndServerStreams(pqc_trial, pqc_trial); - - auto client_identity = - webrtc::SSLIdentity::Create("client", client_key_type_); - auto server_identity = - webrtc::SSLIdentity::Create("server", server_key_type_); - - client_ssl_->SetIdentity(std::move(client_identity)); - server_ssl_->SetIdentity(std::move(server_identity)); +TEST_P(SSLStreamAdapterTestDTLSHandshakeVersion, TestGetSslGroupIdWithPqc) { + auto client_version = ::testing::get<0>(GetParam()); + auto server_version = ::testing::get<1>(GetParam()); + SetupProtocolVersions(client_version, server_version); + + webrtc::CryptoOptions::EphemeralKeyExchangeCipherGroups groups; + std::vector enabled = groups.GetEnabled(); + std::vector groups_with_pqc; + if (std::find(enabled.begin(), enabled.end(), + webrtc::CryptoOptions::EphemeralKeyExchangeCipherGroups:: + kX25519_MLKEM768) == enabled.end()) { + groups_with_pqc.push_back( + webrtc::CryptoOptions::EphemeralKeyExchangeCipherGroups:: + kX25519_MLKEM768); } -}; - -TEST_F(SSLStreamAdapterTestDTLSPqc, TestGetSslGroupId) { - EXPECT_EQ(client_ssl_->GetSslGroupIdForTesting(), 0); - EXPECT_EQ(server_ssl_->GetSslGroupIdForTesting(), 0); + for (auto val : enabled) { + groups_with_pqc.push_back(val); + } + RTC_CHECK(client_ssl_->SetSslGroupIds(groups_with_pqc)); + RTC_CHECK(server_ssl_->SetSslGroupIds(groups_with_pqc)); - SetupProtocolVersions(webrtc::SSL_PROTOCOL_DTLS_13, - webrtc::SSL_PROTOCOL_DTLS_13); + EXPECT_EQ(client_ssl_->GetSslGroupId(), 0); + EXPECT_EQ(server_ssl_->GetSslGroupId(), 0); TestHandshake(); - EXPECT_EQ(client_ssl_->GetSslGroupIdForTesting(), SSL_GROUP_X25519_MLKEM768); - EXPECT_EQ(server_ssl_->GetSslGroupIdForTesting(), SSL_GROUP_X25519_MLKEM768); + if (client_version == webrtc::SSL_PROTOCOL_DTLS_13 && + server_version == webrtc::SSL_PROTOCOL_DTLS_13) { + EXPECT_EQ(client_ssl_->GetSslGroupId(), + webrtc::CryptoOptions::EphemeralKeyExchangeCipherGroups:: + kX25519_MLKEM768); + EXPECT_EQ(server_ssl_->GetSslGroupId(), + webrtc::CryptoOptions::EphemeralKeyExchangeCipherGroups:: + kX25519_MLKEM768); + } else { + EXPECT_EQ(client_ssl_->GetSslGroupId(), + webrtc::CryptoOptions::EphemeralKeyExchangeCipherGroups::kX25519); + EXPECT_EQ(server_ssl_->GetSslGroupId(), + webrtc::CryptoOptions::EphemeralKeyExchangeCipherGroups::kX25519); + } } #endif From 50db72355a2ed5520b73f867bdb072b2fef24188 Mon Sep 17 00:00:00 2001 From: Jonas Oreland Date: Wed, 7 May 2025 03:27:33 -0700 Subject: [PATCH 061/292] Revert "IWYU modules/video_coding/codecs/h264" MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This reverts commit d66b2f9f60329e08046c0c9e021f0bc967e9a3bd. Reason for revert: Breaks import Original change's description: > IWYU modules/video_coding/codecs/h264 > > using > find modules/video_coding/codecs/h264 -name "*.h" -o -name "*.cc" | | xargs tools_webrtc/iwyu/apply-include-cleaner > followed by > tools_webrtc/gn_check_autofix.py -C out/Default/ > and git cl format > > Manual changes to move the includes inside the RTC_USE_H264 define > and wrap libavcodec/libavformat inside extern "C". > Some RTC_USE_H264 includes no longer need to be guarded as the files which are included are fully guarded by that define. > > BUG=webrtc:42226242 > > Change-Id: I3f49028be95e984526d96372acffa3ef21127333 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/385820 > Reviewed-by: Sergey Silkin > Commit-Queue: Philipp Hancke > Reviewed-by: Erik Språng > Cr-Commit-Position: refs/heads/main@{#44526} Bug: webrtc:42226242 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I60f5df9ba6cb3f5754737febd11221af0d33f4d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390280 Reviewed-by: Fredrik Solenberg Commit-Queue: Jonas Oreland Bot-Commit: rubber-stamper@appspot.gserviceaccount.com Reviewed-by: Sergey Silkin Cr-Commit-Position: refs/heads/main@{#44533} --- modules/video_coding/codecs/h264/h264.cc | 13 ++++------ .../codecs/h264/h264_color_space.cc | 7 ----- .../codecs/h264/h264_decoder_impl.cc | 23 ---------------- .../codecs/h264/h264_decoder_impl.h | 4 --- .../codecs/h264/h264_encoder_impl.cc | 26 +++---------------- .../codecs/h264/h264_encoder_impl.h | 12 ++------- .../codecs/h264/h264_encoder_impl_unittest.cc | 3 --- .../codecs/h264/h264_simulcast_unittest.cc | 5 ---- .../codecs/h264/include/h264_globals.h | 3 ++- .../codecs/h264/test/h264_impl_unittest.cc | 4 +-- 10 files changed, 15 insertions(+), 85 deletions(-) diff --git a/modules/video_coding/codecs/h264/h264.cc b/modules/video_coding/codecs/h264/h264.cc index 396c6847f4..ffee2af7f5 100644 --- a/modules/video_coding/codecs/h264/h264.cc +++ b/modules/video_coding/codecs/h264/h264.cc @@ -14,22 +14,19 @@ #include #include #include -#include -#include "absl/base/nullability.h" #include "absl/container/inlined_vector.h" -#include "api/environment/environment.h" -#include "api/video_codecs/h264_profile_level_id.h" -#include "api/video_codecs/scalability_mode.h" #include "api/video_codecs/sdp_video_format.h" -#include "api/video_codecs/video_encoder.h" #include "media/base/media_constants.h" +#include "rtc_base/trace_event.h" + +#if defined(WEBRTC_USE_H264) #include "modules/video_coding/codecs/h264/h264_decoder_impl.h" #include "modules/video_coding/codecs/h264/h264_encoder_impl.h" -#include "modules/video_coding/codecs/h264/include/h264_globals.h" +#endif + #include "rtc_base/checks.h" #include "rtc_base/logging.h" -#include "rtc_base/trace_event.h" namespace webrtc { diff --git a/modules/video_coding/codecs/h264/h264_color_space.cc b/modules/video_coding/codecs/h264/h264_color_space.cc index a93e5e3800..59921263e3 100644 --- a/modules/video_coding/codecs/h264/h264_color_space.cc +++ b/modules/video_coding/codecs/h264/h264_color_space.cc @@ -15,13 +15,6 @@ #include "modules/video_coding/codecs/h264/h264_color_space.h" -extern "C" { -#include -#include -} - -#include "api/video/color_space.h" - namespace webrtc { ColorSpace ExtractH264ColorSpace(AVCodecContext* codec) { diff --git a/modules/video_coding/codecs/h264/h264_decoder_impl.cc b/modules/video_coding/codecs/h264/h264_decoder_impl.cc index b9f81e5a67..f3737a1edd 100644 --- a/modules/video_coding/codecs/h264/h264_decoder_impl.cc +++ b/modules/video_coding/codecs/h264/h264_decoder_impl.cc @@ -18,40 +18,17 @@ extern "C" { #include -#include -#include -#include #include -#include -#include -#include #include -#include } // extern "C" #include -#include -#include -#include #include #include -#include -#include "api/scoped_refptr.h" #include "api/video/color_space.h" -#include "api/video/encoded_image.h" #include "api/video/i010_buffer.h" -#include "api/video/i210_buffer.h" -#include "api/video/i410_buffer.h" #include "api/video/i420_buffer.h" -#include "api/video/i422_buffer.h" -#include "api/video/i444_buffer.h" -#include "api/video/render_resolution.h" -#include "api/video/video_codec_type.h" -#include "api/video/video_frame.h" -#include "api/video/video_frame_buffer.h" -#include "api/video/video_rotation.h" -#include "api/video_codecs/video_decoder.h" #include "common_video/include/video_frame_buffer.h" #include "modules/video_coding/codecs/h264/h264_color_space.h" #include "modules/video_coding/include/video_error_codes.h" diff --git a/modules/video_coding/codecs/h264/h264_decoder_impl.h b/modules/video_coding/codecs/h264/h264_decoder_impl.h index 7446d8fe00..e9da6a7a0d 100644 --- a/modules/video_coding/codecs/h264/h264_decoder_impl.h +++ b/modules/video_coding/codecs/h264/h264_decoder_impl.h @@ -37,14 +37,10 @@ extern "C" { #include -#include } // extern "C" -#include #include -#include "api/video/encoded_image.h" -#include "api/video_codecs/video_decoder.h" #include "common_video/h264/h264_bitstream_parser.h" #include "common_video/include/video_frame_buffer_pool.h" #include "modules/video_coding/codecs/h264/include/h264.h" diff --git a/modules/video_coding/codecs/h264/h264_encoder_impl.cc b/modules/video_coding/codecs/h264/h264_encoder_impl.cc index 19d00a3372..7792d53e32 100644 --- a/modules/video_coding/codecs/h264/h264_encoder_impl.cc +++ b/modules/video_coding/codecs/h264/h264_encoder_impl.cc @@ -17,42 +17,24 @@ #include "modules/video_coding/codecs/h264/h264_encoder_impl.h" #include -#include -#include -#include #include #include #include -#include - -#include "api/environment/environment.h" -#include "api/scoped_refptr.h" -#include "api/units/data_rate.h" -#include "api/video/encoded_image.h" -#include "api/video/i420_buffer.h" -#include "api/video/video_bitrate_allocation.h" -#include "api/video/video_bitrate_allocator.h" + +#include "absl/strings/match.h" #include "api/video/video_codec_constants.h" -#include "api/video/video_codec_type.h" -#include "api/video/video_frame.h" -#include "api/video/video_frame_buffer.h" -#include "api/video/video_frame_type.h" #include "api/video_codecs/scalability_mode.h" -#include "api/video_codecs/video_codec.h" -#include "api/video_codecs/video_encoder.h" #include "common_video/libyuv/include/webrtc_libyuv.h" -#include "modules/video_coding/codecs/h264/include/h264.h" -#include "modules/video_coding/codecs/h264/include/h264_globals.h" -#include "modules/video_coding/codecs/interface/common_constants.h" #include "modules/video_coding/include/video_codec_interface.h" #include "modules/video_coding/include/video_error_codes.h" #include "modules/video_coding/svc/create_scalability_structure.h" -#include "modules/video_coding/svc/scalable_video_controller.h" #include "modules/video_coding/utility/simulcast_rate_allocator.h" #include "modules/video_coding/utility/simulcast_utility.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" +#include "rtc_base/time_utils.h" #include "system_wrappers/include/metrics.h" +#include "third_party/libyuv/include/libyuv/convert.h" #include "third_party/libyuv/include/libyuv/scale.h" #include "third_party/openh264/src/codec/api/wels/codec_api.h" #include "third_party/openh264/src/codec/api/wels/codec_app_def.h" diff --git a/modules/video_coding/codecs/h264/h264_encoder_impl.h b/modules/video_coding/codecs/h264/h264_encoder_impl.h index ea5feb75e1..2d4a95311d 100644 --- a/modules/video_coding/codecs/h264/h264_encoder_impl.h +++ b/modules/video_coding/codecs/h264/h264_encoder_impl.h @@ -21,27 +21,19 @@ #error "See: bugs.webrtc.org/9213#c13." #endif -#include -#include #include -#include #include #include "absl/container/inlined_vector.h" -#include "api/environment/environment.h" -#include "api/scoped_refptr.h" -#include "api/video/encoded_image.h" +#include "api/transport/rtp/dependency_descriptor.h" #include "api/video/i420_buffer.h" #include "api/video/video_codec_constants.h" -#include "api/video/video_frame.h" -#include "api/video/video_frame_type.h" #include "api/video_codecs/scalability_mode.h" -#include "api/video_codecs/video_codec.h" #include "api/video_codecs/video_encoder.h" #include "common_video/h264/h264_bitstream_parser.h" #include "modules/video_coding/codecs/h264/include/h264.h" -#include "modules/video_coding/codecs/h264/include/h264_globals.h" #include "modules/video_coding/svc/scalable_video_controller.h" +#include "modules/video_coding/utility/quality_scaler.h" #include "third_party/openh264/src/codec/api/wels/codec_app_def.h" class ISVCEncoder; diff --git a/modules/video_coding/codecs/h264/h264_encoder_impl_unittest.cc b/modules/video_coding/codecs/h264/h264_encoder_impl_unittest.cc index 2f3dfaf1a6..e714e277e8 100644 --- a/modules/video_coding/codecs/h264/h264_encoder_impl_unittest.cc +++ b/modules/video_coding/codecs/h264/h264_encoder_impl_unittest.cc @@ -12,10 +12,7 @@ #include "modules/video_coding/codecs/h264/h264_encoder_impl.h" #include "api/environment/environment_factory.h" -#include "api/video/video_codec_type.h" -#include "api/video_codecs/video_codec.h" #include "api/video_codecs/video_encoder.h" -#include "modules/video_coding/codecs/h264/include/h264_globals.h" #include "modules/video_coding/include/video_error_codes.h" #include "test/gtest.h" diff --git a/modules/video_coding/codecs/h264/h264_simulcast_unittest.cc b/modules/video_coding/codecs/h264/h264_simulcast_unittest.cc index 39443fcdcc..b9ea324ca2 100644 --- a/modules/video_coding/codecs/h264/h264_simulcast_unittest.cc +++ b/modules/video_coding/codecs/h264/h264_simulcast_unittest.cc @@ -9,16 +9,11 @@ */ #include -#include -#include "api/environment/environment.h" #include "api/test/create_simulcast_test_fixture.h" #include "api/test/simulcast_test_fixture.h" #include "api/test/video/function_video_decoder_factory.h" #include "api/test/video/function_video_encoder_factory.h" -#include "api/video_codecs/sdp_video_format.h" -#include "api/video_codecs/video_decoder_factory.h" -#include "api/video_codecs/video_encoder_factory.h" #include "modules/video_coding/codecs/h264/include/h264.h" #include "test/gtest.h" diff --git a/modules/video_coding/codecs/h264/include/h264_globals.h b/modules/video_coding/codecs/h264/include/h264_globals.h index 9b0b7b6b9e..dac36dd734 100644 --- a/modules/video_coding/codecs/h264/include/h264_globals.h +++ b/modules/video_coding/codecs/h264/include/h264_globals.h @@ -14,10 +14,11 @@ #ifndef MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_GLOBALS_H_ #define MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_GLOBALS_H_ -#include +#include #include #include +#include "modules/video_coding/codecs/interface/common_constants.h" #include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc b/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc index 239aae6548..985224d07e 100644 --- a/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc +++ b/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc @@ -15,13 +15,13 @@ #include "api/video/color_space.h" #include "api/video/encoded_image.h" -#include "api/video/video_codec_type.h" #include "api/video/video_frame.h" -#include "api/video/video_frame_type.h" #include "api/video_codecs/video_codec.h" #include "api/video_codecs/video_decoder.h" #include "api/video_codecs/video_encoder.h" #include "common_video/libyuv/include/webrtc_libyuv.h" +#include "media/base/codec.h" +#include "media/base/media_constants.h" #include "modules/video_coding/codecs/h264/include/h264.h" #include "modules/video_coding/codecs/test/video_codec_unittest.h" #include "modules/video_coding/include/video_codec_interface.h" From 475c74dad32a4418d3608a2bc05796452ddf15ba Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Wed, 7 May 2025 05:02:26 -0700 Subject: [PATCH 062/292] Roll chromium_revision 3bac98a091..413ccd12ab (1456565:1456882) Change log: https://chromium.googlesource.com/chromium/src/+log/3bac98a091..413ccd12ab Full diff: https://chromium.googlesource.com/chromium/src/+/3bac98a091..413ccd12ab Changed dependencies * siso_version: git_revision:70e1167e0e6dad10c8388cace8fd9d9376c43316..git_revision:17b491ae74e86312a96239ff88e852acee83c135 * src/base: https://chromium.googlesource.com/chromium/src/base/+log/6de05cbf0c..db833313b4 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/37f833c4f7..56e78d9e6e * src/buildtools: https://chromium.googlesource.com/chromium/src/buildtools/+log/1df2adafaa..4347ca8fd6 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/19cbcf4234..7a7904b73f * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/ca26dad53b..c4492b5004 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/c01eb5725e..957f2c1a89 * src/third_party/android_build_tools/error_prone/cipd: 2t_u1XEjof609BQE4czlCQzEqRliKS_qY35FOu_VckgC..FLtgIzcLepAc87BaSXIELZIJPT4yxnr8K3MmFaEHrK8C * src/third_party/android_build_tools/lint/cipd: 0xqmMJ1qdP4NPN5s9VmWiNhchvNz_RLBrjXN4jUKN3gC..2fKjMen4v_UU1CDYJqhwJCr6uh-L5w25n9lAKx6SlP0C * src/third_party/android_build_tools/manifest_merger/cipd: 7lstFdOjh1bFmfSZ8le-aZ1a1COQtwirGzuEWa43tnUC..GnPGgBK8kiq32GuJB3iI20yJbFjgt2dqHsmwoPShJ-4C * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/4a839855c5..52ad7cf544 * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/30d9438e7b..18580cf9c4 * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/abf2a0d268..0affb1396d * src/third_party/kotlin_stdlib/cipd: 74gAxjfyJIVJGIA4z0V-jCmMW6A6dDJSoWMh7qhlzp4C..aHCAN6w_ZZCJL7MOPlDbEieRsiUs6ftT33fCI4bbwVwC * src/third_party/libc++/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git/+log/a01c02c9d4..a9cc573e7c * src/third_party/llvm-build/Release+Asserts: Linux_x64/clang-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Linux_x64/clang-tidy-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Linux_x64/clangd-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Mac/clang-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Mac/clang-tidy-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Mac/clangd-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Mac/llvm-code-coverage-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Mac_arm64/clang-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Mac_arm64/clangd-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Win/clang-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Win/clang-tidy-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Win/clang-win-runtime-library-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Win/clangd-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Win/llvm-code-coverage-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz,Win/llvmobjdump-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz..Linux_x64/clang-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Linux_x64/clang-tidy-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Linux_x64/clangd-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Mac/clang-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Mac/clang-tidy-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Mac/clangd-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Mac/llvm-code-coverage-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Mac_arm64/clang-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Mac_arm64/clangd-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Win/clang-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Win/clang-tidy-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Win/clang-win-runtime-library-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Win/clangd-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Win/llvm-code-coverage-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Win/llvmobjdump-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz * src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/c8248a038f..d7bdad4ef8 * src/third_party/rust-toolchain: Linux_x64/rust-toolchain-3350c1eb3fd8fe1bee1ed4c76944d707bd256876-1-llvmorg-21-init-10502-gb2e2ae87.tar.xz,Mac/rust-toolchain-3350c1eb3fd8fe1bee1ed4c76944d707bd256876-1-llvmorg-21-init-10502-gb2e2ae87.tar.xz,Mac_arm64/rust-toolchain-3350c1eb3fd8fe1bee1ed4c76944d707bd256876-1-llvmorg-21-init-10502-gb2e2ae87.tar.xz,Win/rust-toolchain-3350c1eb3fd8fe1bee1ed4c76944d707bd256876-1-llvmorg-21-init-10502-gb2e2ae87.tar.xz..Linux_x64/rust-toolchain-4a0969e06dbeaaa43914d2d00b2e843d49aa3886-1-llvmorg-21-init-10851-gcd6c4b61.tar.xz,Mac/rust-toolchain-4a0969e06dbeaaa43914d2d00b2e843d49aa3886-1-llvmorg-21-init-10851-gcd6c4b61.tar.xz,Mac_arm64/rust-toolchain-4a0969e06dbeaaa43914d2d00b2e843d49aa3886-1-llvmorg-21-init-10851-gcd6c4b61.tar.xz,Win/rust-toolchain-4a0969e06dbeaaa43914d2d00b2e843d49aa3886-1-llvmorg-21-init-10851-gcd6c4b61.tar.xz * src/third_party/siso/cipd: git_revision:70e1167e0e6dad10c8388cace8fd9d9376c43316..git_revision:17b491ae74e86312a96239ff88e852acee83c135 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/678f57506d..ab8639c6e8 DEPS diff: https://chromium.googlesource.com/chromium/src/+/3bac98a091..413ccd12ab/DEPS Clang version changed llvmorg-21-init-10502-gb2e2ae87:llvmorg-21-init-10851-gcd6c4b61 Details: https://chromium.googlesource.com/chromium/src/+/3bac98a091..413ccd12ab/tools/clang/scripts/update.py BUG=None Change-Id: I19a8ad4c1c95d69af75bc357fbb218acb94acd54 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390300 Commit-Queue: Autoroller Bot-Commit: Autoroller Cr-Commit-Position: refs/heads/main@{#44534} --- DEPS | 228 +++++++++++++++++++++++++++++------------------------------ 1 file changed, 114 insertions(+), 114 deletions(-) diff --git a/DEPS b/DEPS index e4660b355d..c2e8822d9c 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '3bac98a09145b27fec236e1b528e883f3fcea876', + 'chromium_revision': '413ccd12abe49ad79521af08e38c1208cc19f44a', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -53,7 +53,7 @@ vars = { # reclient CIPD package version 'reclient_version': 're_client_version:0.177.1.e58c0145-gomaip', # siso CIPD package version. - 'siso_version': 'git_revision:70e1167e0e6dad10c8388cace8fd9d9376c43316', + 'siso_version': 'git_revision:17b491ae74e86312a96239ff88e852acee83c135', # ninja CIPD package. 'ninja_package': 'infra/3pp/tools/ninja/', @@ -73,24 +73,24 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@6de05cbf0c0cf7af590bc79123cca396f28d017d', + 'https://chromium.googlesource.com/chromium/src/base@db833313b460c52046dd0fbcba586e3b9a4a1e9f', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@37f833c4f7b05a39f9cff7029769a4861e7d0fb3', + 'https://chromium.googlesource.com/chromium/src/build@56e78d9e6e7ca1f7320ad2c6467d364061c789d7', 'src/buildtools': - 'https://chromium.googlesource.com/chromium/src/buildtools@1df2adafaa36e2bcff31e28cd814505382e04df6', + 'https://chromium.googlesource.com/chromium/src/buildtools@4347ca8fd6c0c6081dbd9bd05c200c6b170f18bc', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. 'src/examples/androidtests/third_party/gradle': { 'url': 'https://chromium.googlesource.com/external/github.com/gradle/gradle.git@f2d1fb54a951d8b11d25748e4711bec8d128d7e3', 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@19cbcf42345c6d240df1c758328c3484d2bd5e0b', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@7a7904b73f694988f7bad1c6fd5a8e4424467e3a', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@ca26dad53be50e879e65286a00bfed4ef0852ce5', + 'https://chromium.googlesource.com/chromium/src/testing@c4492b50040791ac4dc9cc2125ea550d37c2b3ed', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@c01eb5725e85c8bd4aaac98762e210d0328090ee', + 'https://chromium.googlesource.com/chromium/src/third_party@957f2c1a89321e825753b9efddf3ea3ee6633ef8', 'src/buildtools/linux64': { 'packages': [ @@ -142,143 +142,143 @@ deps = { 'objects': [ { # The Android libclang_rt.builtins libraries are currently only included in the Linux clang package. - 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'sha256sum': '1ed7bf852f8255bf35335c14110c959743e17903e0020c776be3b8d80a564ae9', - 'size_bytes': 2484344, - 'generation': 1746143899795872, + 'object_name': 'Win/clang-win-runtime-library-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz', + 'sha256sum': 'b6a49e302e5c9d562e575e1c1431777d90d4950dc7e6bba9b1c21d0ceba93c29', + 'size_bytes': 2484356, + 'generation': 1746525692368377, 'condition': 'checkout_win and not host_os == "win"', }, { - 'object_name': 'Win/clangd-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz', - 'sha256sum': '90f9035116a665e8a620b259d299c845be42b68c7dce8a6439ba189d550bd6ca', - 'size_bytes': 13855648, - 'generation': 1746143892528921, + 'object_name': 'Win/clangd-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz', + 'sha256sum': '51953564440fd82547a27b1fe0cd76d9a0c200162ee7530fe14d805511613be7', + 'size_bytes': 13886984, + 'generation': 1746525671497511, 'condition': 'host_os == "win" and checkout_clangd', }, { - 'object_name': 'Win/llvm-code-coverage-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz', - 'sha256sum': '100dc0773b81d1a65b40356038d2c85e5aabd613b2e5830846d60c9fa2ca04bf', - 'size_bytes': 2374764, - 'generation': 1746143892621367, + 'object_name': 'Win/llvm-code-coverage-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz', + 'sha256sum': 'e5e5a685fde6a4ddd211913ccf6f8e62ac7e6567fec26d68d5bff3aeac42c330', + 'size_bytes': 2376076, + 'generation': 1746525671928280, 'condition': 'host_os == "win" and checkout_clang_coverage_tools', }, { - 'object_name': 'Win/llvmobjdump-llvmorg-21-init-10502-gb2e2ae87-1.tar.xz', - 'sha256sum': '104ef589d1d1d9a7f1acd38e25c418b12de8b685c4f55f45f626cc64f3f1cc98', - 'size_bytes': 5685804, - 'generation': 1746143892549487, + 'object_name': 'Win/llvmobjdump-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz', + 'sha256sum': 'd02d624d3ecfe5dbaa2e93b2da5a6443e4f25b9fe2fab43b8b540fccc91e4770', + 'size_bytes': 5687508, + 'generation': 1746525671637063, 'condition': '(checkout_linux or checkout_mac or checkout_android) and host_os == "win"', }, ] @@ -290,31 +290,31 @@ deps = { 'bucket': 'chromium-browser-clang', 'objects': [ { - 'object_name': 'Linux_x64/rust-toolchain-3350c1eb3fd8fe1bee1ed4c76944d707bd256876-1-llvmorg-21-init-10502-gb2e2ae87.tar.xz', - 'sha256sum': 'b7b13f2fbd22c5415abec739570acd74066361ce3c82313660be65ab67f1a075', - 'size_bytes': 118912744, - 'generation': 1746143864066683, + 'object_name': 'Linux_x64/rust-toolchain-4a0969e06dbeaaa43914d2d00b2e843d49aa3886-1-llvmorg-21-init-10851-gcd6c4b61.tar.xz', + 'sha256sum': '2d8498b5dcab2954c8212564798252c3edc9c0181ea8b05d3d6ae42b7b6adf7d', + 'size_bytes': 118439104, + 'generation': 1746525613682798, 'condition': 'host_os == "linux" and non_git_source', }, { - 'object_name': 'Mac/rust-toolchain-3350c1eb3fd8fe1bee1ed4c76944d707bd256876-1-llvmorg-21-init-10502-gb2e2ae87.tar.xz', - 'sha256sum': '322dc36f74871d746545a1bb75470968315bce5ed4b385f43033e376f4c12a9b', - 'size_bytes': 112193460, - 'generation': 1746143865768180, + 'object_name': 'Mac/rust-toolchain-4a0969e06dbeaaa43914d2d00b2e843d49aa3886-1-llvmorg-21-init-10851-gcd6c4b61.tar.xz', + 'sha256sum': '93e57858344a978ce2df2d9926c541d8e25cd9def0054328dba4ac13ff633d34', + 'size_bytes': 111691112, + 'generation': 1746525615270255, 'condition': 'host_os == "mac" and host_cpu == "x64"', }, { - 'object_name': 'Mac_arm64/rust-toolchain-3350c1eb3fd8fe1bee1ed4c76944d707bd256876-1-llvmorg-21-init-10502-gb2e2ae87.tar.xz', - 'sha256sum': '2a95ec6fe7623c43e87f06829bae17c45f1c94c19db4c8f085d77cb948c0ee3f', - 'size_bytes': 101896012, - 'generation': 1746143867490688, + 'object_name': 'Mac_arm64/rust-toolchain-4a0969e06dbeaaa43914d2d00b2e843d49aa3886-1-llvmorg-21-init-10851-gcd6c4b61.tar.xz', + 'sha256sum': '4c1f9d7ff8c1febdb755ad971b0ae8d9b3c5bed5b30b96ea00caa784dfb86427', + 'size_bytes': 101778572, + 'generation': 1746525616855514, 'condition': 'host_os == "mac" and host_cpu == "arm64"', }, { - 'object_name': 'Win/rust-toolchain-3350c1eb3fd8fe1bee1ed4c76944d707bd256876-1-llvmorg-21-init-10502-gb2e2ae87.tar.xz', - 'sha256sum': 'da814703ae38af456eeeb63985198cb7ec48740b0a143249830fc880d633f80b', - 'size_bytes': 193736940, - 'generation': 1746143869370201, + 'object_name': 'Win/rust-toolchain-4a0969e06dbeaaa43914d2d00b2e843d49aa3886-1-llvmorg-21-init-10851-gcd6c4b61.tar.xz', + 'sha256sum': '7d2484e229f4015d3711f1294d11048764ec22ebaf8266701abf350cd51c897b', + 'size_bytes': 193318196, + 'generation': 1746525618539172, 'condition': 'host_os == "win"', }, ], @@ -325,11 +325,11 @@ deps = { 'src/third_party/compiler-rt/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@cff6ae8a80960ceca4d83cf6efa271222d2e7d85', 'src/third_party/libc++/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@a01c02c9d4acbdae3b7e8a2f3ee58579a9c29f96', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@a9cc573e7c591795d11b72d8323ba0e573b55bd6', 'src/third_party/libc++abi/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@7ff13dd8c66d5047a3ee1e1ba26ddeb70d3d8740', 'src/third_party/llvm-libc/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@c8248a038fe70b7495d1d52fde500cf2dd6fae82', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@d7bdad4ef86b827a96469b1dfdfcfa1218930e59', 'src/third_party/libunwind/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@81e2cb40a70de2b6978e6d8658891ded9a77f7e3', @@ -427,7 +427,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_build_tools/error_prone', - 'version': '2t_u1XEjof609BQE4czlCQzEqRliKS_qY35FOu_VckgC', + 'version': 'FLtgIzcLepAc87BaSXIELZIJPT4yxnr8K3MmFaEHrK8C', }, ], 'condition': 'checkout_android', @@ -449,7 +449,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_build_tools/lint', - 'version': '0xqmMJ1qdP4NPN5s9VmWiNhchvNz_RLBrjXN4jUKN3gC', + 'version': '2fKjMen4v_UU1CDYJqhwJCr6uh-L5w25n9lAKx6SlP0C', }, ], 'condition': 'checkout_android and non_git_source', @@ -484,7 +484,7 @@ deps = { 'src/third_party/breakpad/breakpad': 'https://chromium.googlesource.com/breakpad/breakpad.git@b319c12317824bfd0e92828db20bbfe0ec6e31f8', 'src/third_party/catapult': - 'https://chromium.googlesource.com/catapult.git@4a839855c5e79d91689784e2c7284f3f6288123b', + 'https://chromium.googlesource.com/catapult.git@52ad7cf544050f01f47c070716b9dae8eb9fab2b', 'src/third_party/ced/src': { 'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5', }, @@ -497,13 +497,13 @@ deps = { 'src/third_party/crc32c/src': 'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558', 'src/third_party/depot_tools': - 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@30d9438e7baf0c045f973de2f517b51554d14d7d', + 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@18580cf9c407a3c23aa072b9d745832516de699e', 'src/third_party/ffmpeg': 'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@dcdd0fa51b65a0b1688ff6b8f0cc81908f09ded2', 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@abf2a0d268dc589f991abd9208f73523e2e14123', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@0affb1396d17e4473ded5b221e66c31d9e4c222f', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -563,7 +563,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/kotlin_stdlib', - 'version': '74gAxjfyJIVJGIA4z0V-jCmMW6A6dDJSoWMh7qhlzp4C', + 'version': 'aHCAN6w_ZZCJL7MOPlDbEieRsiUs6ftT33fCI4bbwVwC', }, ], 'condition': 'checkout_android', @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@678f57506d545d668ad4c34e89dfd616c6e6d4c3', + 'https://chromium.googlesource.com/chromium/src/tools@ab8639c6e8bef8b09642856b1b526e1e4edfcbad', 'src/third_party/espresso': { 'packages': [ @@ -706,7 +706,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_build_tools/manifest_merger', - 'version': '7lstFdOjh1bFmfSZ8le-aZ1a1COQtwirGzuEWa43tnUC', + 'version': 'GnPGgBK8kiq32GuJB3iI20yJbFjgt2dqHsmwoPShJ-4C', }, ], 'condition': 'checkout_android', From 1e750debf5e030bebd9f289b413827bec0ec7ab7 Mon Sep 17 00:00:00 2001 From: Jonas Oreland Date: Wed, 7 May 2025 14:32:34 +0200 Subject: [PATCH 063/292] PQC: Follow up :( 1) Add one missing method 2) Add two missing testcases (one for a previously buggy single-line implementation!) BUG=webrtc:404763475 No-Iwyu: Doesn't seem to understand the #ifdef surrounding open/boring ssl Change-Id: Ic829cac64a15162050c0af272d5c48e01e2eb5ea Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390321 Auto-Submit: Jonas Oreland Reviewed-by: Harald Alvestrand Commit-Queue: Jonas Oreland Cr-Commit-Position: refs/heads/main@{#44535} --- api/crypto/crypto_options.cc | 2 +- api/crypto/crypto_options.h | 2 +- api/crypto/crypto_options_unittest.cc | 41 +++++++++++++++++++++++++++ 3 files changed, 43 insertions(+), 2 deletions(-) diff --git a/api/crypto/crypto_options.cc b/api/crypto/crypto_options.cc index 53c1cf06ae..00c86d57ff 100644 --- a/api/crypto/crypto_options.cc +++ b/api/crypto/crypto_options.cc @@ -112,7 +112,7 @@ CryptoOptions::EphemeralKeyExchangeCipherGroups::GetSupported() { } void CryptoOptions::EphemeralKeyExchangeCipherGroups::AddFirst(uint16_t group) { - enabled_.erase(std::remove(enabled_.begin(), enabled_.end(), group)); + std::erase(enabled_, group); enabled_.insert(enabled_.begin(), group); } diff --git a/api/crypto/crypto_options.h b/api/crypto/crypto_options.h index 0840d8d5df..8b8dc08f59 100644 --- a/api/crypto/crypto_options.h +++ b/api/crypto/crypto_options.h @@ -94,7 +94,7 @@ struct RTC_EXPORT CryptoOptions { // Which cipher groups are enabled in this crypto options. std::vector GetEnabled() const { return enabled_; } - void SetEnabled(const std::vector& groups); + void SetEnabled(const std::vector& groups) { enabled_ = groups; } void AddFirst(uint16_t group); // Update list of enabled groups based on field_trials, diff --git a/api/crypto/crypto_options_unittest.cc b/api/crypto/crypto_options_unittest.cc index 2b79118d63..d7d59ca7b4 100644 --- a/api/crypto/crypto_options_unittest.cc +++ b/api/crypto/crypto_options_unittest.cc @@ -62,6 +62,47 @@ TEST(EphemeralKeyExchangeCipherGroupsTest, GetEnabled) { EXPECT_EQ(groups.GetEnabled(), expected); } +TEST(EphemeralKeyExchangeCipherGroupsTest, SetEnabled) { + std::vector expected = { + webrtc::CryptoOptions::EphemeralKeyExchangeCipherGroups::kX25519, + }; + webrtc::CryptoOptions::EphemeralKeyExchangeCipherGroups groups; + groups.SetEnabled(expected); + EXPECT_EQ(groups.GetEnabled(), expected); +} + +TEST(EphemeralKeyExchangeCipherGroupsTest, AddFirst) { + std::vector initial = { +#ifdef SSL_GROUP_X25519 + SSL_GROUP_X25519, +#endif +#ifdef SSL_GROUP_SECP256R1 + SSL_GROUP_SECP256R1, +#endif +#ifdef SSL_GROUP_SECP384R1 + SSL_GROUP_SECP384R1, +#endif + }; + webrtc::CryptoOptions::EphemeralKeyExchangeCipherGroups groups; + EXPECT_EQ(groups.GetEnabled(), initial); + groups.AddFirst(webrtc::CryptoOptions::EphemeralKeyExchangeCipherGroups:: + kX25519_MLKEM768); + + std::vector expected = { + webrtc::CryptoOptions::EphemeralKeyExchangeCipherGroups::kX25519_MLKEM768, +#ifdef SSL_GROUP_X25519 + SSL_GROUP_X25519, +#endif +#ifdef SSL_GROUP_SECP256R1 + SSL_GROUP_SECP256R1, +#endif +#ifdef SSL_GROUP_SECP384R1 + SSL_GROUP_SECP384R1, +#endif + }; + EXPECT_EQ(groups.GetEnabled(), expected); +} + TEST(EphemeralKeyExchangeCipherGroupsTest, Update) { std::vector expected = { #ifdef SSL_GROUP_X25519_MLKEM768 From 3ba9dc52eeeb6cc171567bf5197994923a6f089b Mon Sep 17 00:00:00 2001 From: Per K Date: Wed, 7 May 2025 11:58:03 +0000 Subject: [PATCH 064/292] Extend FeedBackGenerator to allow multiple network nodes and to log RtcEventLogs MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This cl: - Adds possibility to supply multiple network node builder to the feedback generator. - Adds an RtcEventLog to log all sent and received feedback packets. Bug: None Change-Id: I3e8e0bc114a8ffbb1e8ac21a2cbacbb6db001c86 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390002 Auto-Submit: Per Kjellander Reviewed-by: Björn Terelius Commit-Queue: Björn Terelius Cr-Commit-Position: refs/heads/main@{#44536} --- api/transport/BUILD.gn | 4 + .../test/create_feedback_generator.cc | 9 + .../test/create_feedback_generator.h | 6 + .../test/feedback_generator_interface.h | 29 +++ logging/BUILD.gn | 5 + test/network/BUILD.gn | 41 ++-- test/network/feedback_generator.cc | 181 ++++++++++++++---- test/network/feedback_generator.h | 58 ++++-- test/network/feedback_generator_unittest.cc | 73 +++++++ 9 files changed, 339 insertions(+), 67 deletions(-) diff --git a/api/transport/BUILD.gn b/api/transport/BUILD.gn index 7769526e07..81ff330104 100644 --- a/api/transport/BUILD.gn +++ b/api/transport/BUILD.gn @@ -138,6 +138,8 @@ if (rtc_include_tests) { deps = [ ":network_control", "..:simulated_network_api", + "../../api:network_emulation_manager_api", + "../../api/rtc_event_log:rtc_event_log", "../units:data_rate", "../units:data_size", "../units:time_delta", @@ -154,6 +156,8 @@ if (rtc_include_tests) { deps = [ ":network_control", ":test_feedback_generator_interface", + "../../api:network_emulation_manager_api", + "../../api/rtc_event_log:rtc_event_log", "../../test/network:feedback_generator", ] } diff --git a/api/transport/test/create_feedback_generator.cc b/api/transport/test/create_feedback_generator.cc index 6144bccacf..d7097e80d9 100644 --- a/api/transport/test/create_feedback_generator.cc +++ b/api/transport/test/create_feedback_generator.cc @@ -11,6 +11,7 @@ #include +#include "api/test/network_emulation_manager.h" #include "api/transport/test/feedback_generator_interface.h" #include "test/network/feedback_generator.h" @@ -21,4 +22,12 @@ std::unique_ptr CreateFeedbackGenerator( return std::make_unique(confg); } +std::unique_ptr +CreateFeedbackGeneratorWithoutNetwork( + FeedbackGeneratorWithoutNetwork::Config config, + NetworkEmulationManager& network_emulation_manager) { + return std::make_unique( + config, network_emulation_manager); +} + } // namespace webrtc diff --git a/api/transport/test/create_feedback_generator.h b/api/transport/test/create_feedback_generator.h index a1a2226496..616e8500b8 100644 --- a/api/transport/test/create_feedback_generator.h +++ b/api/transport/test/create_feedback_generator.h @@ -12,10 +12,16 @@ #include +#include "api/test/network_emulation_manager.h" #include "api/transport/test/feedback_generator_interface.h" namespace webrtc { std::unique_ptr CreateFeedbackGenerator( FeedbackGenerator::Config confg); + +std::unique_ptr +CreateFeedbackGeneratorWithoutNetwork( + FeedbackGeneratorWithoutNetwork::Config config, + NetworkEmulationManager& network_emulation_manager); } // namespace webrtc #endif // API_TRANSPORT_TEST_CREATE_FEEDBACK_GENERATOR_H_ diff --git a/api/transport/test/feedback_generator_interface.h b/api/transport/test/feedback_generator_interface.h index 1ffc8b0796..38bee44dc4 100644 --- a/api/transport/test/feedback_generator_interface.h +++ b/api/transport/test/feedback_generator_interface.h @@ -13,6 +13,8 @@ #include #include +#include "api/rtc_event_log/rtc_event_log.h" +#include "api/test/network_emulation_manager.h" #include "api/test/simulated_network.h" #include "api/transport/network_types.h" #include "api/units/data_rate.h" @@ -32,11 +34,38 @@ class FeedbackGenerator { virtual ~FeedbackGenerator() = default; virtual Timestamp Now() = 0; virtual void Sleep(TimeDelta duration) = 0; + // Send a packet of the given size over the simulated network. virtual void SendPacket(size_t size) = 0; virtual std::vector PopFeedback() = 0; virtual void SetSendConfig(BuiltInNetworkBehaviorConfig config) = 0; virtual void SetReturnConfig(BuiltInNetworkBehaviorConfig config) = 0; virtual void SetSendLinkCapacity(DataRate capacity) = 0; + virtual RtcEventLog& event_log() = 0; }; + +// Same as FeedbackGenerator, but NetworkEmulationManager is owned externally. +// Packets can be sent and received via multiple nodes. +class FeedbackGeneratorWithoutNetwork { + public: + struct Config { + std::vector sent_via_nodes; + std::vector received_via_nodes; + TimeDelta feedback_interval = TimeDelta::Millis(50); + DataSize feedback_packet_size = DataSize::Bytes(20); + }; + + virtual ~FeedbackGeneratorWithoutNetwork() = default; + + // Send a packet of the given size over the simulated network. + // The packet size logged in the event log is `total_size` - `overhead`. + // This allows a user to ensure that LoggedPacketInfo.size + + // LoggedPacketInfo.overhead in the event log is `total_size`. + // Note that ParsedRtcEventLog estimate the overhead depending on the + // selected ice candidate. + virtual void SendPacket(size_t total_size, size_t overhead) = 0; + virtual std::vector PopFeedback() = 0; + virtual RtcEventLog& event_log() = 0; +}; + } // namespace webrtc #endif // API_TRANSPORT_TEST_FEEDBACK_GENERATOR_INTERFACE_H_ diff --git a/logging/BUILD.gn b/logging/BUILD.gn index ea483b29a1..a56f9266e4 100644 --- a/logging/BUILD.gn +++ b/logging/BUILD.gn @@ -547,6 +547,11 @@ if (rtc_enable_protobuf) { "//third_party/abseil-cpp/absl/memory", "//third_party/abseil-cpp/absl/strings:string_view", ] + + # Normally public deps should not be used, but the proto must be included + # by all targets depending on rtc_event_log_parser. + public_deps += # no-presubmit-check TODO(webrtc:8603) + [ ":rtc_event_log_proto" ] } if (rtc_include_tests) { diff --git a/test/network/BUILD.gn b/test/network/BUILD.gn index 81ce6e3295..870b56769e 100644 --- a/test/network/BUILD.gn +++ b/test/network/BUILD.gn @@ -194,29 +194,44 @@ if (rtc_include_tests) { deps = [ ":emulated_network", ":simulated_network", + "../../api:create_network_emulation_manager", "../../api:network_emulation_manager_api", + "../../api:rtp_parameters", "../../api:simulated_network_api", + "../../api/environment:environment", + "../../api/environment:environment_factory", + "../../api/rtc_event_log:rtc_event_log", + "../../api/rtc_event_log:rtc_event_log_factory", "../../api/transport:network_control", "../../api/transport:test_feedback_generator_interface", "../../api/units:data_rate", "../../api/units:data_size", "../../api/units:time_delta", "../../api/units:timestamp", + "../../logging:rtc_event_rtp_rtcp", + "../../modules/rtp_rtcp:rtp_rtcp_format", "../../rtc_base:checks", "//third_party/abseil-cpp/absl/memory", ] } - rtc_library("feedback_generator_unittest") { - testonly = true - sources = [ "feedback_generator_unittest.cc" ] - deps = [ - "..:test_support", - "../../api:simulated_network_api", - "../../api/transport:test_feedback_generator", - "../../api/transport:test_feedback_generator_interface", - "../../api/units:time_delta", - ] + if (rtc_enable_protobuf) { + rtc_library("feedback_generator_unittest") { + testonly = true + sources = [ "feedback_generator_unittest.cc" ] + deps = [ + "..:test_support", + "../../api:create_network_emulation_manager", + "../../api:network_emulation_manager_api", + "../../api:simulated_network_api", + "../../api/rtc_event_log:rtc_event_log", + "../../api/transport:test_feedback_generator", + "../../api/transport:test_feedback_generator_interface", + "../../api/units:time_delta", + "../../logging:rtc_event_log_parser", + "../logging:log_writer", + ] + } } if (!build_with_chromium) { @@ -224,13 +239,15 @@ if (rtc_include_tests) { testonly = true deps = [ ":cross_traffic_unittest", - ":feedback_generator_unittest", ":network_emulation_pc_unittest", ":network_emulation_unittest", ":simulated_network_unittest", ] if (rtc_enable_protobuf) { - deps += [ ":schedulable_network_behavior_test" ] + deps += [ + ":feedback_generator_unittest", + ":schedulable_network_behavior_test", + ] } } } diff --git a/test/network/feedback_generator.cc b/test/network/feedback_generator.cc index 7462c99e54..01887d0479 100644 --- a/test/network/feedback_generator.cc +++ b/test/network/feedback_generator.cc @@ -10,90 +10,132 @@ #include "test/network/feedback_generator.h" #include +#include #include #include #include "absl/memory/memory.h" +#include "api/environment/environment.h" +#include "api/environment/environment_factory.h" +#include "api/rtc_event_log/rtc_event_log.h" +#include "api/rtc_event_log/rtc_event_log_factory.h" +#include "api/rtp_parameters.h" +#include "api/test/create_network_emulation_manager.h" #include "api/test/network_emulation_manager.h" #include "api/test/simulated_network.h" #include "api/transport/network_types.h" +#include "api/transport/test/feedback_generator_interface.h" #include "api/units/data_rate.h" #include "api/units/data_size.h" #include "api/units/time_delta.h" #include "api/units/timestamp.h" +#include "logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h" +#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h" +#include "modules/rtp_rtcp/include/rtp_header_extension_map.h" +#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" +#include "modules/rtp_rtcp/source/rtp_header_extensions.h" +#include "modules/rtp_rtcp/source/rtp_packet_to_send.h" #include "rtc_base/checks.h" -#include "test/network/network_emulation_manager.h" +#include "test/network/network_emulation.h" #include "test/network/simulated_network.h" - namespace webrtc { -FeedbackGeneratorImpl::FeedbackGeneratorImpl( - FeedbackGeneratorImpl::Config config) - : conf_(config), - net_({.time_mode = TimeMode::kSimulated}), - send_link_{new SimulatedNetwork(conf_.send_link)}, - ret_link_{new SimulatedNetwork(conf_.return_link)}, - route_(this, - net_.CreateRoute( - {net_.CreateEmulatedNode(absl::WrapUnique(send_link_))}), - net_.CreateRoute( - {net_.CreateEmulatedNode(absl::WrapUnique(ret_link_))})) {} +using ::webrtc::RtcEventRtcpPacketIncoming; +using ::webrtc::RtcEventRtpPacketOutgoing; +using ::webrtc::RtpPacketToSend; +using ::webrtc::TimeDelta; +using ::webrtc::TransportPacketsFeedback; -Timestamp FeedbackGeneratorImpl::Now() { - return net_.Now(); +namespace { +Environment GetEnvironment(NetworkEmulationManager& net) { + EnvironmentFactory factory; + factory.Set(net.time_controller()->GetClock()); + factory.Set(net.time_controller()->GetTaskQueueFactory()); + return factory.Create(); } -void FeedbackGeneratorImpl::Sleep(TimeDelta duration) { - net_.time_controller()->AdvanceTime(duration); +webrtc::EmulatedRoute* CreateRoute( + webrtc::NetworkEmulationManager& net, + std::vector link) { + if (!link.empty()) { + return net.CreateRoute(link); + } else { + return net.CreateRoute({net.CreateUnconstrainedEmulatedNode()}); + } } -void FeedbackGeneratorImpl::SendPacket(size_t size) { +} // namespace + +FeedbackGeneratorWithoutNetworkImpl::FeedbackGeneratorWithoutNetworkImpl( + Config config, + NetworkEmulationManager& net) + : net_(net), + feedback_interval_(config.feedback_interval), + feedback_packet_size_(config.feedback_packet_size), + event_log_(RtcEventLogFactory().Create(GetEnvironment(net))), + route_(this, + CreateRoute(net, std::move(config.sent_via_nodes)), + CreateRoute(net, std::move(config.received_via_nodes))) { + RtpExtension transport_sequence_number_extension; + transport_sequence_number_extension.uri = + RtpExtension::kTransportSequenceNumberUri; + transport_sequence_number_extension.id = 1; + transport_sequence_number_extension.encrypt = false; + rtp_extensions_.RegisterByUri(transport_sequence_number_extension.id, + transport_sequence_number_extension.uri); +} + +void FeedbackGeneratorWithoutNetworkImpl::SendPacket(size_t total_size, + size_t overhead) { SentPacket sent; sent.send_time = Now(); - sent.size = DataSize::Bytes(size); + sent.size = DataSize::Bytes(total_size); sent.sequence_number = sequence_number_++; sent_packets_.push(sent); - route_.SendRequest(size, sent); + + RtpPacketToSend packet_to_send(&rtp_extensions_); + packet_to_send.set_transport_sequence_number(sent.sequence_number); + packet_to_send.SetExtension(sent.sequence_number); + RTC_DCHECK(total_size > packet_to_send.headers_size() + overhead); + if (total_size > packet_to_send.headers_size() + overhead) { + packet_to_send.SetPayloadSize(total_size - packet_to_send.headers_size() - + overhead); + RTC_DCHECK_EQ(packet_to_send.size(), total_size - overhead); + } + event_log_->Log(std::make_unique( + packet_to_send, /*probe_cluster_id*/ 0)); + + route_.SendRequest(total_size, sent); } -std::vector FeedbackGeneratorImpl::PopFeedback() { +std::vector +FeedbackGeneratorWithoutNetworkImpl::PopFeedback() { std::vector ret; ret.swap(feedback_); return ret; } -void FeedbackGeneratorImpl::SetSendConfig(BuiltInNetworkBehaviorConfig config) { - conf_.send_link = config; - send_link_->SetConfig(conf_.send_link); +Timestamp FeedbackGeneratorWithoutNetworkImpl::Now() { + return net_.time_controller()->GetClock()->CurrentTime(); } -void FeedbackGeneratorImpl::SetReturnConfig( - BuiltInNetworkBehaviorConfig config) { - conf_.return_link = config; - ret_link_->SetConfig(conf_.return_link); -} - -void FeedbackGeneratorImpl::SetSendLinkCapacity(DataRate capacity) { - conf_.send_link.link_capacity = capacity; - send_link_->SetConfig(conf_.send_link); -} - -void FeedbackGeneratorImpl::OnRequest(SentPacket packet, - Timestamp arrival_time) { +void FeedbackGeneratorWithoutNetworkImpl::OnRequest(SentPacket packet, + Timestamp arrival_time) { PacketResult result; result.sent_packet = packet; result.receive_time = arrival_time; received_packets_.push_back(result); Timestamp first_recv = received_packets_.front().receive_time; - if (Now() - first_recv > conf_.feedback_interval) { - route_.SendResponse(conf_.feedback_packet_size.bytes(), + if (Now() - first_recv > feedback_interval_) { + route_.SendResponse(feedback_packet_size_.bytes(), std::move(received_packets_)); received_packets_ = {}; } } -void FeedbackGeneratorImpl::OnResponse(std::vector packet_results, - Timestamp arrival_time) { +void FeedbackGeneratorWithoutNetworkImpl::OnResponse( + std::vector packet_results, + Timestamp arrival_time) { TransportPacketsFeedback feedback; feedback.feedback_time = arrival_time; std::vector::const_iterator received_packet_iterator = @@ -117,7 +159,64 @@ void FeedbackGeneratorImpl::OnResponse(std::vector packet_results, } sent_packets_.pop(); } + rtcp::TransportFeedback transport_feedback; + transport_feedback.SetBase( + feedback.ReceivedWithSendInfo()[0].sent_packet.sequence_number, + feedback.ReceivedWithSendInfo()[0].receive_time); + for (const auto& received_packet : feedback.ReceivedWithSendInfo()) { + transport_feedback.AddReceivedPacket( + received_packet.sent_packet.sequence_number, + received_packet.receive_time); + } + event_log_->Log( + std::make_unique(transport_feedback.Build())); feedback_.push_back(feedback); } +FeedbackGeneratorImpl::FeedbackGeneratorImpl( + FeedbackGeneratorImpl::Config config) + : conf_(config), + net_(CreateNetworkEmulationManager({.time_mode = TimeMode::kSimulated})), + send_link_{new SimulatedNetwork(conf_.send_link)}, + ret_link_{new SimulatedNetwork(conf_.return_link)}, + delegate_({.sent_via_nodes = {net_->CreateEmulatedNode( + absl::WrapUnique(send_link_))}, + .received_via_nodes = {net_->CreateEmulatedNode( + absl::WrapUnique(ret_link_))}, + .feedback_interval = config.feedback_interval, + .feedback_packet_size = config.feedback_packet_size}, + *net_) {} + +Timestamp FeedbackGeneratorImpl::Now() { + return delegate_.Now(); +} + +void FeedbackGeneratorImpl::Sleep(TimeDelta duration) { + net_->time_controller()->AdvanceTime(duration); +} + +void FeedbackGeneratorImpl::SendPacket(size_t size) { + delegate_.SendPacket(size, /*overhead=*/0); +} + +std::vector FeedbackGeneratorImpl::PopFeedback() { + return delegate_.PopFeedback(); +} + +void FeedbackGeneratorImpl::SetSendConfig(BuiltInNetworkBehaviorConfig config) { + conf_.send_link = config; + send_link_->SetConfig(conf_.send_link); +} + +void FeedbackGeneratorImpl::SetReturnConfig( + BuiltInNetworkBehaviorConfig config) { + conf_.return_link = config; + ret_link_->SetConfig(conf_.return_link); +} + +void FeedbackGeneratorImpl::SetSendLinkCapacity(DataRate capacity) { + conf_.send_link.link_capacity = capacity; + send_link_->SetConfig(conf_.send_link); +} + } // namespace webrtc diff --git a/test/network/feedback_generator.h b/test/network/feedback_generator.h index c7975f69fa..97dc1441c6 100644 --- a/test/network/feedback_generator.h +++ b/test/network/feedback_generator.h @@ -12,27 +12,63 @@ #include #include +#include #include #include +#include "api/rtc_event_log/rtc_event_log.h" +#include "api/test/network_emulation_manager.h" #include "api/test/simulated_network.h" #include "api/transport/network_types.h" #include "api/transport/test/feedback_generator_interface.h" #include "api/units/data_rate.h" +#include "api/units/data_size.h" #include "api/units/time_delta.h" #include "api/units/timestamp.h" +#include "modules/rtp_rtcp/include/rtp_header_extension_map.h" #include "test/network/network_emulation.h" -#include "test/network/network_emulation_manager.h" #include "test/network/simulated_network.h" namespace webrtc { -class FeedbackGeneratorImpl - : public FeedbackGenerator, - public TwoWayFakeTrafficRoute>:: +class FeedbackGeneratorWithoutNetworkImpl + : public FeedbackGeneratorWithoutNetwork, + TwoWayFakeTrafficRoute>:: TrafficHandlerInterface { + public: + FeedbackGeneratorWithoutNetworkImpl(Config config, + NetworkEmulationManager& net); + ~FeedbackGeneratorWithoutNetworkImpl() override = default; + + void SendPacket(size_t total_size, size_t overhead) override; + std::vector PopFeedback() override; + + RtcEventLog& event_log() override { return *event_log_; } + void OnRequest(SentPacket packet, Timestamp arrival_time) override; + void OnResponse(std::vector packet_results, + Timestamp arrival_time) override; + + Timestamp Now(); + + private: + NetworkEmulationManager& net_; + const TimeDelta feedback_interval_; + const DataSize feedback_packet_size_; + + std::unique_ptr event_log_; + TwoWayFakeTrafficRoute> route_; + + std::queue sent_packets_; + std::vector received_packets_; + std::vector feedback_; + int64_t sequence_number_ = 1; + RtpHeaderExtensionMap rtp_extensions_; +}; + +class FeedbackGeneratorImpl : public FeedbackGenerator { public: explicit FeedbackGeneratorImpl(Config config); + Timestamp Now() override; void Sleep(TimeDelta duration) override; void SendPacket(size_t size) override; @@ -43,21 +79,15 @@ class FeedbackGeneratorImpl void SetSendLinkCapacity(DataRate capacity) override; - void OnRequest(SentPacket packet, Timestamp arrival_time) override; - void OnResponse(std::vector packet_results, - Timestamp arrival_time) override; + RtcEventLog& event_log() override { return delegate_.event_log(); } private: Config conf_; - ::webrtc::test::NetworkEmulationManagerImpl net_; + std::unique_ptr net_; SimulatedNetwork* const send_link_; SimulatedNetwork* const ret_link_; - TwoWayFakeTrafficRoute> route_; - - std::queue sent_packets_; - std::vector received_packets_; - std::vector feedback_; - int64_t sequence_number_ = 1; + FeedbackGeneratorWithoutNetworkImpl delegate_; }; + } // namespace webrtc #endif // TEST_NETWORK_FEEDBACK_GENERATOR_H_ diff --git a/test/network/feedback_generator_unittest.cc b/test/network/feedback_generator_unittest.cc index 6ad41e5696..b2d572567b 100644 --- a/test/network/feedback_generator_unittest.cc +++ b/test/network/feedback_generator_unittest.cc @@ -9,14 +9,28 @@ */ #include +#include +#include +#include "api/rtc_event_log/rtc_event_log.h" +#include "api/test/create_network_emulation_manager.h" +#include "api/test/network_emulation_manager.h" #include "api/test/simulated_network.h" #include "api/transport/test/create_feedback_generator.h" #include "api/transport/test/feedback_generator_interface.h" #include "api/units/time_delta.h" +#include "logging/rtc_event_log/rtc_event_log_parser.h" +#include "test/gmock.h" #include "test/gtest.h" +#include "test/logging/log_writer.h" +#include "test/logging/memory_log_writer.h" namespace webrtc { + +using ::testing::Eq; +using ::testing::Field; +using ::testing::SizeIs; + TEST(FeedbackGeneratorTest, ReportsFeedbackForSentPackets) { size_t kPacketSize = 1000; auto gen = CreateFeedbackGenerator(FeedbackGenerator::Config()); @@ -49,4 +63,63 @@ TEST(FeedbackGeneratorTest, FeedbackIncludesLostPackets) { EXPECT_NEAR(feedback_list[0].LostWithSendInfo().size(), feedback_list[0].ReceivedWithSendInfo().size(), 2); } + +TEST(FeedbackGeneratorTest, WritesToEventLog) { + size_t kPacketSize = 1000; + auto gen = CreateFeedbackGenerator(FeedbackGenerator::Config()); + + MemoryLogStorage log_storage; + std::unique_ptr log_writer_factory = + log_storage.CreateFactory(); + + const std::string rtc_event_log_file_name = "eventlog"; + gen->event_log().StartLogging( + log_writer_factory->Create(rtc_event_log_file_name), + RtcEventLog::kImmediateOutput); + + for (int i = 0; i < 10; ++i) { + gen->SendPacket(kPacketSize); + gen->Sleep(TimeDelta::Millis(50)); + } + auto feedback_list = gen->PopFeedback(); + ASSERT_GT(feedback_list.size(), 0u); + + gen->event_log().StopLogging(); + ParsedRtcEventLog parsed_log; + auto it = log_storage.logs().find(rtc_event_log_file_name); + ASSERT_TRUE(it != log_storage.logs().end()); + ASSERT_TRUE(parsed_log.ParseString(it->second).ok()); + + EXPECT_THAT(parsed_log.GetOutgoingPacketInfos(), SizeIs(10)); + EXPECT_THAT(parsed_log.GetOutgoingPacketInfos(), + Each(Field(&LoggedPacketInfo::size, Eq(kPacketSize)))); + EXPECT_THAT(parsed_log.transport_feedbacks(PacketDirection::kIncomingPacket), + SizeIs(feedback_list.size())); +} + +TEST(FeedbackGeneratorWithoutNetworkTest, ReportsFeedbackForSentPackets) { + auto network_emulation_manager = CreateNetworkEmulationManager( + {.time_mode = webrtc::TimeMode::kSimulated}); + std::unique_ptr gen = + CreateFeedbackGeneratorWithoutNetwork( + FeedbackGeneratorWithoutNetwork::Config(), + *network_emulation_manager); + size_t kTotalPacketSize = 1000; + size_t kOverhead = 40; + + for (int i = 0; i < 10; ++i) { + gen->SendPacket(kTotalPacketSize, kOverhead); + network_emulation_manager->time_controller()->AdvanceTime( + TimeDelta::Millis(50)); + } + auto feedback_list = gen->PopFeedback(); + EXPECT_GT(feedback_list.size(), 0u); + for (const auto& feedback : feedback_list) { + EXPECT_GT(feedback.packet_feedbacks.size(), 0u); + for (const auto& packet : feedback.packet_feedbacks) { + EXPECT_EQ(packet.sent_packet.size.bytes(), kTotalPacketSize); + } + } +} + } // namespace webrtc From d533a0cd41613e1c4923f676bb6faba130abc558 Mon Sep 17 00:00:00 2001 From: Jakob Ivarsson Date: Wed, 7 May 2025 12:31:29 +0000 Subject: [PATCH 065/292] Allow opus_red payload type in NetEqTest. MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Bug: None Change-Id: I8fad06589a2eaf8ea16508456cc03195433d14f3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390320 Commit-Queue: Jakob Ivarsson‎ Reviewed-by: Henrik Lundin Cr-Commit-Position: refs/heads/main@{#44537} --- modules/audio_coding/neteq/tools/neteq_test_factory.cc | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/modules/audio_coding/neteq/tools/neteq_test_factory.cc b/modules/audio_coding/neteq/tools/neteq_test_factory.cc index f74d211089..827ed92931 100644 --- a/modules/audio_coding/neteq/tools/neteq_test_factory.cc +++ b/modules/audio_coding/neteq/tools/neteq_test_factory.cc @@ -298,9 +298,9 @@ std::unique_ptr NetEqTestFactory::InitializeTest( std::set cn_types = std_set_int32_to_uint8( {config.cn_nb, config.cn_wb, config.cn_swb32, config.cn_swb48}); - std::set forbidden_types = std_set_int32_to_uint8( - {config.g722, config.red, config.opus_red, config.avt, config.avt_16, - config.avt_32, config.avt_48}); + std::set forbidden_types = + std_set_int32_to_uint8({config.g722, config.red, config.avt, + config.avt_16, config.avt_32, config.avt_48}); input.reset(new NetEqReplacementInput(std::move(input), replacement_pt, cn_types, forbidden_types)); From 1ccfd191c9b0a0997f5ce56c7667b5f020225755 Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Wed, 7 May 2025 08:59:24 -0700 Subject: [PATCH 066/292] turn rtc_source_set with cc files into rtc_library and add a gn_check_autofix helper to faciliate this. BUG=webrtc:42235055 Change-Id: Ia9c2d4bb7969bce7ec3b902102703d6cf3c8cb33 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/387121 Commit-Queue: Philipp Hancke Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44538} --- api/BUILD.gn | 14 ++-- api/adaptation/BUILD.gn | 2 +- api/audio/BUILD.gn | 6 +- api/neteq/BUILD.gn | 12 ++-- api/test/network_emulation/BUILD.gn | 2 +- api/test/pclf/BUILD.gn | 2 +- api/transport/BUILD.gn | 4 +- api/transport/rtp/BUILD.gn | 2 +- api/video/BUILD.gn | 2 +- api/video_codecs/BUILD.gn | 4 +- call/adaptation/BUILD.gn | 2 +- g3doc/style-guide.md | 4 +- modules/audio_device/BUILD.gn | 4 +- modules/audio_processing/BUILD.gn | 4 +- .../audio_processing/agc2/rnn_vad/BUILD.gn | 2 +- modules/audio_processing/ns/BUILD.gn | 2 +- modules/rtp_rtcp/BUILD.gn | 6 +- modules/utility/BUILD.gn | 2 +- modules/video_capture/BUILD.gn | 2 +- modules/video_coding/BUILD.gn | 2 +- modules/video_coding/svc/BUILD.gn | 16 ++--- net/dcsctp/public/BUILD.gn | 6 +- pc/BUILD.gn | 68 +++++++++---------- rtc_base/BUILD.gn | 10 +-- rtc_base/system/BUILD.gn | 2 +- tools_webrtc/gn_check_autofix.py | 25 ++++++- webrtc.gni | 6 ++ 27 files changed, 121 insertions(+), 92 deletions(-) diff --git a/api/BUILD.gn b/api/BUILD.gn index 3e4bd7762c..125941ac45 100644 --- a/api/BUILD.gn +++ b/api/BUILD.gn @@ -26,7 +26,7 @@ rtc_source_set("call_api") { sources = [ "call/audio_sink.h" ] } -rtc_source_set("enable_media") { +rtc_library("enable_media") { visibility = [ "*" ] sources = [ "enable_media.cc", @@ -46,7 +46,7 @@ rtc_source_set("enable_media") { ] } -rtc_source_set("enable_media_with_defaults") { +rtc_library("enable_media_with_defaults") { visibility = [ "*" ] allow_poison = [ "audio_codecs", @@ -402,7 +402,7 @@ rtc_library("libjingle_peerconnection_api") { ] } -rtc_source_set("frame_transformer_interface") { +rtc_library("frame_transformer_interface") { visibility = [ "*" ] sources = [ "frame_transformer_interface.cc", @@ -654,7 +654,7 @@ rtc_source_set("peer_connection_quality_test_fixture_api") { ] } -rtc_source_set("frame_generator_api") { +rtc_library("frame_generator_api") { visibility = [ "*" ] sources = [ "test/frame_generator_interface.cc", @@ -898,7 +898,7 @@ rtc_source_set("simulated_network_api") { } # TODO(srte): Move to network_emulation sub directory. -rtc_source_set("network_emulation_manager_api") { +rtc_library("network_emulation_manager_api") { visibility = [ "*" ] sources = [ "test/network_emulation_manager.cc", @@ -924,7 +924,7 @@ rtc_source_set("network_emulation_manager_api") { ] } -rtc_source_set("time_controller") { +rtc_library("time_controller") { visibility = [ "*" ] sources = [ "test/time_controller.cc", @@ -1683,7 +1683,7 @@ if (rtc_include_tests) { } } -rtc_source_set("field_trials_registry") { +rtc_library("field_trials_registry") { visibility = [ "*" ] sources = [ "field_trials_registry.cc", diff --git a/api/adaptation/BUILD.gn b/api/adaptation/BUILD.gn index fe66065a36..8e1579f9f1 100644 --- a/api/adaptation/BUILD.gn +++ b/api/adaptation/BUILD.gn @@ -8,7 +8,7 @@ import("../../webrtc.gni") -rtc_source_set("resource_adaptation_api") { +rtc_library("resource_adaptation_api") { visibility = [ "*" ] sources = [ "resource.cc", diff --git a/api/audio/BUILD.gn b/api/audio/BUILD.gn index 5683e4a372..b762e1505b 100644 --- a/api/audio/BUILD.gn +++ b/api/audio/BUILD.gn @@ -77,7 +77,7 @@ rtc_source_set("audio_mixer_api") { ] } -rtc_source_set("audio_processing") { +rtc_library("audio_processing") { visibility = [ "*" ] sources = [ "audio_processing.cc", @@ -123,7 +123,7 @@ rtc_library("builtin_audio_processing_builder") { ] } -rtc_source_set("audio_processing_statistics") { +rtc_library("audio_processing_statistics") { visibility = [ "*" ] sources = [ "audio_processing_statistics.cc", @@ -173,7 +173,7 @@ rtc_source_set("echo_control") { ] } -rtc_source_set("echo_detector_creator") { +rtc_library("echo_detector_creator") { visibility = [ "*" ] allow_poison = [ "default_echo_detector" ] sources = [ diff --git a/api/neteq/BUILD.gn b/api/neteq/BUILD.gn index 9e79a877fd..3fd020b6fe 100644 --- a/api/neteq/BUILD.gn +++ b/api/neteq/BUILD.gn @@ -8,7 +8,7 @@ import("../../webrtc.gni") -rtc_source_set("neteq_api") { +rtc_library("neteq_api") { visibility = [ "*" ] sources = [ "neteq.cc", @@ -28,7 +28,7 @@ rtc_source_set("neteq_api") { ] } -rtc_source_set("default_neteq_factory") { +rtc_library("default_neteq_factory") { visibility = [ "*" ] sources = [ "default_neteq_factory.cc", @@ -44,7 +44,7 @@ rtc_source_set("default_neteq_factory") { ] } -rtc_source_set("custom_neteq_factory") { +rtc_library("custom_neteq_factory") { visibility = [ "*" ] sources = [ "custom_neteq_factory.cc", @@ -75,7 +75,7 @@ rtc_source_set("neteq_controller_api") { ] } -rtc_source_set("default_neteq_controller_factory") { +rtc_library("default_neteq_controller_factory") { visibility = [ "*" ] sources = [ "default_neteq_controller_factory.cc", @@ -89,7 +89,7 @@ rtc_source_set("default_neteq_controller_factory") { ] } -rtc_source_set("tick_timer") { +rtc_library("tick_timer") { visibility = [ "*" ] sources = [ "tick_timer.cc", @@ -98,7 +98,7 @@ rtc_source_set("tick_timer") { deps = [ "../../rtc_base:checks" ] } -rtc_source_set("tick_timer_unittest") { +rtc_library("tick_timer_unittest") { visibility = [ "*" ] testonly = true sources = [ "tick_timer_unittest.cc" ] diff --git a/api/test/network_emulation/BUILD.gn b/api/test/network_emulation/BUILD.gn index e574cb342a..d22cc09d00 100644 --- a/api/test/network_emulation/BUILD.gn +++ b/api/test/network_emulation/BUILD.gn @@ -16,7 +16,7 @@ if (rtc_enable_protobuf) { proto_out_dir = "api/test/network_emulation/" } - rtc_source_set("schedulable_network_node_builder") { + rtc_library("schedulable_network_node_builder") { visibility = [ "*" ] sources = [ "schedulable_network_node_builder.cc", diff --git a/api/test/pclf/BUILD.gn b/api/test/pclf/BUILD.gn index ff8109be57..3082bdd7d3 100644 --- a/api/test/pclf/BUILD.gn +++ b/api/test/pclf/BUILD.gn @@ -8,7 +8,7 @@ import("../../../webrtc.gni") -rtc_source_set("media_configuration") { +rtc_library("media_configuration") { visibility = [ "*" ] testonly = true sources = [ diff --git a/api/transport/BUILD.gn b/api/transport/BUILD.gn index 81ff330104..1dfce618d6 100644 --- a/api/transport/BUILD.gn +++ b/api/transport/BUILD.gn @@ -106,7 +106,7 @@ rtc_source_set("sctp_transport_factory_interface") { deps = [ "../../api/environment" ] } -rtc_source_set("stun_types") { +rtc_library("stun_types") { visibility = [ "*" ] sources = [ "stun.cc", @@ -164,7 +164,7 @@ if (rtc_include_tests) { } if (rtc_include_tests) { - rtc_source_set("stun_unittest") { + rtc_library("stun_unittest") { visibility = [ "*" ] testonly = true sources = [ "stun_unittest.cc" ] diff --git a/api/transport/rtp/BUILD.gn b/api/transport/rtp/BUILD.gn index d4d5e4460c..9982346e12 100644 --- a/api/transport/rtp/BUILD.gn +++ b/api/transport/rtp/BUILD.gn @@ -19,7 +19,7 @@ rtc_source_set("rtp_source") { ] } -rtc_source_set("dependency_descriptor") { +rtc_library("dependency_descriptor") { visibility = [ "*" ] sources = [ "dependency_descriptor.cc", diff --git a/api/video/BUILD.gn b/api/video/BUILD.gn index 98591e6050..d0af76a6a0 100644 --- a/api/video/BUILD.gn +++ b/api/video/BUILD.gn @@ -350,7 +350,7 @@ rtc_source_set("video_stream_encoder") { ] } -rtc_source_set("video_frame_metadata") { +rtc_library("video_frame_metadata") { visibility = [ "*" ] sources = [ "video_frame_metadata.cc", diff --git a/api/video_codecs/BUILD.gn b/api/video_codecs/BUILD.gn index 8351ae424b..3732e6021f 100644 --- a/api/video_codecs/BUILD.gn +++ b/api/video_codecs/BUILD.gn @@ -12,7 +12,7 @@ if (is_android) { import("//build/config/android/rules.gni") } -rtc_source_set("scalability_mode") { +rtc_library("scalability_mode") { visibility = [ "*" ] sources = [ "scalability_mode.cc", @@ -25,7 +25,7 @@ rtc_source_set("scalability_mode") { ] } -rtc_source_set("scalability_mode_helper") { +rtc_library("scalability_mode_helper") { visibility = [ "*" ] sources = [ "scalability_mode_helper.cc", diff --git a/call/adaptation/BUILD.gn b/call/adaptation/BUILD.gn index d3a8ad6f32..c6e5d8f5bf 100644 --- a/call/adaptation/BUILD.gn +++ b/call/adaptation/BUILD.gn @@ -96,7 +96,7 @@ if (rtc_include_tests) { ] } - rtc_source_set("resource_adaptation_test_utilities") { + rtc_library("resource_adaptation_test_utilities") { testonly = true sources = [ diff --git a/g3doc/style-guide.md b/g3doc/style-guide.md index 6af842dee8..0ffaef13e4 100644 --- a/g3doc/style-guide.md +++ b/g3doc/style-guide.md @@ -278,8 +278,8 @@ configuration, only use the following [GN templates][gn-templ]. |------------------|-----------------------------------------------------------------------------------------| | `executable` | `rtc_executable` | | `shared_library` | `rtc_shared_library` | -| `source_set` | `rtc_source_set` (only for header only libraries, for everything else use `rtc_library` | -| `static_library` | `rtc_static_library` (use `rtc_library` unless you really need `rtc_static_library` | +| `source_set` | `rtc_source_set` (only for header only libraries, for everything else use `rtc_library`)| +| `static_library` | `rtc_static_library` (use `rtc_library` unless you really need `rtc_static_library`) | | `test` | `rtc_test` | diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn index 805a6ab6c3..6ddd196126 100644 --- a/modules/audio_device/BUILD.gn +++ b/modules/audio_device/BUILD.gn @@ -101,7 +101,7 @@ rtc_library("audio_device_name") { deps = [ "//third_party/abseil-cpp/absl/strings:string_view" ] } -rtc_source_set("windows_core_audio_utility") { +rtc_library("windows_core_audio_utility") { if (is_win && !build_with_chromium) { sources = [ "win/core_audio_utility_win.cc", @@ -129,7 +129,7 @@ rtc_source_set("windows_core_audio_utility") { # audio_device_impl target. The goal is to use this new structure and # gradually phase out the old design. # TODO(henrika): currently only supported on Windows. -rtc_source_set("audio_device_module_from_input_and_output") { +rtc_library("audio_device_module_from_input_and_output") { visibility = [ "*" ] if (is_win && !build_with_chromium) { sources = [ diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn index 2454689de1..c8b657fe72 100644 --- a/modules/audio_processing/BUILD.gn +++ b/modules/audio_processing/BUILD.gn @@ -82,7 +82,7 @@ rtc_library("high_pass_filter") { ] } -rtc_source_set("aec_dump_interface") { +rtc_library("aec_dump_interface") { visibility = [ "*" ] sources = [ "include/aec_dump.cc", @@ -229,7 +229,7 @@ rtc_library("residual_echo_detector") { ] } -rtc_source_set("rms_level") { +rtc_library("rms_level") { visibility = [ "*" ] sources = [ "rms_level.cc", diff --git a/modules/audio_processing/agc2/rnn_vad/BUILD.gn b/modules/audio_processing/agc2/rnn_vad/BUILD.gn index 025794d262..ddec89972d 100644 --- a/modules/audio_processing/agc2/rnn_vad/BUILD.gn +++ b/modules/audio_processing/agc2/rnn_vad/BUILD.gn @@ -77,7 +77,7 @@ rtc_library("rnn_vad_lp_residual") { ] } -rtc_source_set("rnn_vad_layers") { +rtc_library("rnn_vad_layers") { sources = [ "rnn_fc.cc", "rnn_fc.h", diff --git a/modules/audio_processing/ns/BUILD.gn b/modules/audio_processing/ns/BUILD.gn index 31ad61156d..90aedfabdb 100644 --- a/modules/audio_processing/ns/BUILD.gn +++ b/modules/audio_processing/ns/BUILD.gn @@ -67,7 +67,7 @@ rtc_static_library("ns") { } if (rtc_include_tests) { - rtc_source_set("ns_unittests") { + rtc_library("ns_unittests") { testonly = true configs += [ "..:apm_debug_dump" ] diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn index 96ccf3713c..92c9954dcb 100644 --- a/modules/rtp_rtcp/BUILD.gn +++ b/modules/rtp_rtcp/BUILD.gn @@ -361,7 +361,7 @@ rtc_library("rtp_rtcp") { ] } -rtc_source_set("rtp_rtcp_legacy") { +rtc_library("rtp_rtcp_legacy") { sources = [ "include/rtp_rtcp.h", "source/deprecated/deprecated_rtp_sender_egress.cc", @@ -473,7 +473,7 @@ rtc_library("rtp_video_header") { ] } -rtc_source_set("rtp_video_header_unittest") { +rtc_library("rtp_video_header_unittest") { testonly = true sources = [ "source/rtp_video_header_unittest.cc" ] deps = [ @@ -770,7 +770,7 @@ if (rtc_include_tests) { ] } - rtc_source_set("frame_transformer_factory_unittest") { + rtc_library("frame_transformer_factory_unittest") { testonly = true sources = [ "source/frame_transformer_factory_unittest.cc" ] deps = [ diff --git a/modules/utility/BUILD.gn b/modules/utility/BUILD.gn index 8cefe5653c..b30cb0c6d4 100644 --- a/modules/utility/BUILD.gn +++ b/modules/utility/BUILD.gn @@ -8,7 +8,7 @@ import("../../webrtc.gni") -rtc_source_set("utility") { +rtc_library("utility") { if (is_android) { visibility = [ "*" ] sources = [ diff --git a/modules/video_capture/BUILD.gn b/modules/video_capture/BUILD.gn index 22f5ff2acc..e5e74c9fc3 100644 --- a/modules/video_capture/BUILD.gn +++ b/modules/video_capture/BUILD.gn @@ -51,7 +51,7 @@ rtc_library("video_capture_module") { } if (!build_with_chromium || is_linux || is_chromeos) { - rtc_source_set("video_capture_internal_impl") { + rtc_library("video_capture_internal_impl") { visibility = [ "*" ] deps = [ ":video_capture_module", diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn index d308040153..63661464a0 100644 --- a/modules/video_coding/BUILD.gn +++ b/modules/video_coding/BUILD.gn @@ -619,7 +619,7 @@ rtc_library("webrtc_vp8") { } } -rtc_source_set("webrtc_vp8_scalability") { +rtc_library("webrtc_vp8_scalability") { sources = [ "codecs/vp8/vp8_scalability.cc", "codecs/vp8/vp8_scalability.h", diff --git a/modules/video_coding/svc/BUILD.gn b/modules/video_coding/svc/BUILD.gn index 9ee0223154..81c747d536 100644 --- a/modules/video_coding/svc/BUILD.gn +++ b/modules/video_coding/svc/BUILD.gn @@ -8,7 +8,7 @@ import("../../../webrtc.gni") -rtc_source_set("scalability_mode_util") { +rtc_library("scalability_mode_util") { sources = [ "scalability_mode_util.cc", "scalability_mode_util.h", @@ -23,7 +23,7 @@ rtc_source_set("scalability_mode_util") { ] } -rtc_source_set("scalable_video_controller") { +rtc_library("scalable_video_controller") { sources = [ "scalable_video_controller.h", "scalable_video_controller_no_layering.cc", @@ -39,7 +39,7 @@ rtc_source_set("scalable_video_controller") { ] } -rtc_source_set("scalability_structures") { +rtc_library("scalability_structures") { sources = [ "create_scalability_structure.cc", "create_scalability_structure.h", @@ -66,7 +66,7 @@ rtc_source_set("scalability_structures") { ] } -rtc_source_set("svc_rate_allocator") { +rtc_library("svc_rate_allocator") { sources = [ "svc_rate_allocator.cc", "svc_rate_allocator.h", @@ -88,7 +88,7 @@ rtc_source_set("svc_rate_allocator") { ] } -rtc_source_set("simulcast_to_svc_converter") { +rtc_library("simulcast_to_svc_converter") { sources = [ "simulcast_to_svc_converter.cc", "simulcast_to_svc_converter.h", @@ -110,7 +110,7 @@ rtc_source_set("simulcast_to_svc_converter") { } if (rtc_include_tests) { - rtc_source_set("scalability_structure_tests") { + rtc_library("scalability_structure_tests") { testonly = true sources = [ "scalability_mode_util_unittest.cc", @@ -141,7 +141,7 @@ if (rtc_include_tests) { ] } - rtc_source_set("svc_rate_allocator_tests") { + rtc_library("svc_rate_allocator_tests") { testonly = true sources = [ "svc_rate_allocator_unittest.cc" ] deps = [ @@ -162,7 +162,7 @@ if (rtc_include_tests) { ] } - rtc_source_set("simulcast_to_svc_converter_tests") { + rtc_library("simulcast_to_svc_converter_tests") { testonly = true sources = [ "simulcast_to_svc_converter_unittest.cc" ] deps = [ diff --git a/net/dcsctp/public/BUILD.gn b/net/dcsctp/public/BUILD.gn index 97f7651f15..a744681942 100644 --- a/net/dcsctp/public/BUILD.gn +++ b/net/dcsctp/public/BUILD.gn @@ -21,7 +21,7 @@ rtc_source_set("types") { ] } -rtc_source_set("socket") { +rtc_library("socket") { deps = [ ":types", "../../../api:array_view", @@ -40,7 +40,7 @@ rtc_source_set("socket") { ] } -rtc_source_set("factory") { +rtc_library("factory") { deps = [ ":socket", ":types", @@ -66,7 +66,7 @@ rtc_source_set("mocks") { ] } -rtc_source_set("utils") { +rtc_library("utils") { deps = [ ":socket", ":types", diff --git a/pc/BUILD.gn b/pc/BUILD.gn index 14e96b7541..6d5d0ca9da 100644 --- a/pc/BUILD.gn +++ b/pc/BUILD.gn @@ -60,7 +60,7 @@ rtc_source_set("proxy") { ] } -rtc_source_set("channel") { +rtc_library("channel") { visibility = [ ":*", "../test/peer_scenario", @@ -126,7 +126,7 @@ rtc_source_set("channel_interface") { ] } -rtc_source_set("dtls_srtp_transport") { +rtc_library("dtls_srtp_transport") { visibility = [ ":*" ] sources = [ "dtls_srtp_transport.cc", @@ -145,7 +145,7 @@ rtc_source_set("dtls_srtp_transport") { ] } -rtc_source_set("dtls_transport") { +rtc_library("dtls_transport") { visibility = [ ":*", "../test/*", @@ -171,7 +171,7 @@ rtc_source_set("dtls_transport") { ] } -rtc_source_set("external_hmac") { +rtc_library("external_hmac") { visibility = [ ":*" ] sources = [ "external_hmac.cc", @@ -186,7 +186,7 @@ rtc_source_set("external_hmac") { } } -rtc_source_set("ice_transport") { +rtc_library("ice_transport") { visibility = [ ":*" ] sources = [ "ice_transport.cc", @@ -201,7 +201,7 @@ rtc_source_set("ice_transport") { ] } -rtc_source_set("jsep_transport") { +rtc_library("jsep_transport") { visibility = [ ":*" ] sources = [ "jsep_transport.cc", @@ -245,7 +245,7 @@ rtc_source_set("jsep_transport") { ] } -rtc_source_set("jsep_transport_collection") { +rtc_library("jsep_transport_collection") { visibility = [ ":*" ] sources = [ "jsep_transport_collection.cc", @@ -265,7 +265,7 @@ rtc_source_set("jsep_transport_collection") { ] } -rtc_source_set("jsep_transport_controller") { +rtc_library("jsep_transport_controller") { visibility = [ ":*", "../test/peer_scenario:*", @@ -342,7 +342,7 @@ rtc_source_set("media_factory") { ] } -rtc_source_set("media_session") { +rtc_library("media_session") { visibility = [ "*" ] # Used by Chrome sources = [ "media_session.cc", @@ -511,7 +511,7 @@ rtc_source_set("peer_connection_proxy") { ] } -rtc_source_set("rtcp_mux_filter") { +rtc_library("rtcp_mux_filter") { visibility = [ ":*" ] sources = [ "rtcp_mux_filter.cc", @@ -523,7 +523,7 @@ rtc_source_set("rtcp_mux_filter") { ] } -rtc_source_set("rtp_media_utils") { +rtc_library("rtp_media_utils") { visibility = [ ":*" ] sources = [ "rtp_media_utils.cc", @@ -560,7 +560,7 @@ rtc_source_set("rtp_sender_proxy") { ] } -rtc_source_set("rtp_transport") { +rtc_library("rtp_transport") { visibility = [ ":*" ] sources = [ "rtp_transport.cc", @@ -614,7 +614,7 @@ rtc_source_set("rtp_transport_internal") { ] } -rtc_source_set("sctp_transport") { +rtc_library("sctp_transport") { visibility = [ ":*" ] sources = [ "sctp_transport.cc", @@ -639,7 +639,7 @@ rtc_source_set("sctp_transport") { ] } -rtc_source_set("sctp_utils") { +rtc_library("sctp_utils") { visibility = [ ":*", "../test/fuzzers:*", @@ -661,7 +661,7 @@ rtc_source_set("sctp_utils") { "../rtc_base:ssl_adapter", ] } -rtc_source_set("srtp_session") { +rtc_library("srtp_session") { visibility = [ ":*" ] sources = [ "srtp_session.cc", @@ -691,7 +691,7 @@ rtc_source_set("srtp_session") { deps += [ "//third_party/libsrtp" ] } } -rtc_source_set("srtp_transport") { +rtc_library("srtp_transport") { visibility = [ ":*" ] sources = [ "srtp_transport.cc", @@ -717,7 +717,7 @@ rtc_source_set("srtp_transport") { ] } -rtc_source_set("transport_stats") { +rtc_library("transport_stats") { visibility = [ ":*" ] sources = [ "transport_stats.cc", @@ -742,7 +742,7 @@ rtc_source_set("used_ids") { ] } -rtc_source_set("video_track_source_proxy") { +rtc_library("video_track_source_proxy") { visibility = [ "*" ] # Used by Chrome sources = [ "video_track_source_proxy.cc", @@ -760,7 +760,7 @@ rtc_source_set("video_track_source_proxy") { ] } -rtc_source_set("session_description") { +rtc_library("session_description") { # TODO(bugs.webrtc.org/13661): Reduce visibility if possible visibility = [ "*" ] # Used by Chrome and others @@ -791,7 +791,7 @@ rtc_source_set("session_description") { ] } -rtc_source_set("simulcast_description") { +rtc_library("simulcast_description") { sources = [ "simulcast_description.cc", "simulcast_description.h", @@ -897,7 +897,7 @@ rtc_library("connection_context") { ] } -rtc_source_set("data_channel_controller") { +rtc_library("data_channel_controller") { visibility = [ ":*" ] sources = [ "data_channel_controller.cc", @@ -962,7 +962,7 @@ rtc_source_set("peer_connection_internal") { ] } -rtc_source_set("rtc_stats_collector") { +rtc_library("rtc_stats_collector") { visibility = [ ":*", "../api:*", @@ -1037,7 +1037,7 @@ rtc_source_set("rtc_stats_collector") { ] } -rtc_source_set("rtc_stats_traversal") { +rtc_library("rtc_stats_traversal") { visibility = [ ":*" ] sources = [ "rtc_stats_traversal.cc", @@ -1050,7 +1050,7 @@ rtc_source_set("rtc_stats_traversal") { ] } -rtc_source_set("sdp_munging_detector") { +rtc_library("sdp_munging_detector") { visibility = [ ":*" ] sources = [ "sdp_munging_detector.cc", @@ -1070,7 +1070,7 @@ rtc_source_set("sdp_munging_detector") { "//third_party/abseil-cpp/absl/algorithm:container", ] } -rtc_source_set("sdp_offer_answer") { +rtc_library("sdp_offer_answer") { visibility = [ ":*" ] sources = [ "sdp_offer_answer.cc", @@ -1160,7 +1160,7 @@ rtc_source_set("jsep_ice_candidate") { rtc_source_set("jsep_session_description") { visibility = [ ":*" ] } -rtc_source_set("local_audio_source") { +rtc_library("local_audio_source") { visibility = [ ":*" ] sources = [ "local_audio_source.cc", @@ -1173,7 +1173,7 @@ rtc_source_set("local_audio_source") { "../api:scoped_refptr", ] } -rtc_source_set("peer_connection") { +rtc_library("peer_connection") { visibility = [ ":*" ] sources = [ "peer_connection.cc", @@ -1279,7 +1279,7 @@ rtc_source_set("peer_connection") { ] } -rtc_source_set("simulcast_sdp_serializer") { +rtc_library("simulcast_sdp_serializer") { visibility = [ ":*" ] sources = [ "simulcast_sdp_serializer.cc", @@ -1300,7 +1300,7 @@ rtc_source_set("simulcast_sdp_serializer") { "//third_party/abseil-cpp/absl/strings:string_view", ] } -rtc_source_set("sdp_utils") { +rtc_library("sdp_utils") { sources = [ "sdp_utils.cc", "sdp_utils.h", @@ -1313,7 +1313,7 @@ rtc_source_set("sdp_utils") { "../rtc_base/system:rtc_export", ] } -rtc_source_set("legacy_stats_collector") { +rtc_library("legacy_stats_collector") { visibility = [ ":*" ] sources = [ "legacy_stats_collector.cc", @@ -1367,7 +1367,7 @@ rtc_source_set("stream_collection") { sources = [ "stream_collection.h" ] deps = [ "../api:libjingle_peerconnection_api" ] } -rtc_source_set("track_media_info_map") { +rtc_library("track_media_info_map") { visibility = [ ":*" ] sources = [ "track_media_info_map.cc", @@ -1387,7 +1387,7 @@ rtc_source_set("track_media_info_map") { "../rtc_base:threading", ] } -rtc_source_set("webrtc_sdp") { +rtc_library("webrtc_sdp") { # TODO(bugs.webrtc.org/13661): Reduce visibility if possible visibility = [ "*" ] # Used by Chrome and more @@ -1437,7 +1437,7 @@ rtc_source_set("webrtc_sdp") { "//third_party/abseil-cpp/absl/strings:string_view", ] } -rtc_source_set("webrtc_session_description_factory") { +rtc_library("webrtc_session_description_factory") { visibility = [ ":*" ] sources = [ "webrtc_session_description_factory.cc", @@ -1508,7 +1508,7 @@ rtc_library("media_stream_observer") { "//third_party/abseil-cpp/absl/algorithm:container", ] } -rtc_source_set("peer_connection_factory") { +rtc_library("peer_connection_factory") { # TODO(bugs.webrtc.org/13661): Reduce visibility if possible visibility = [ "*" ] # Known to be used externally allow_poison = [ "environment_construction" ] diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn index 239adb82c7..430060fc3d 100644 --- a/rtc_base/BUILD.gn +++ b/rtc_base/BUILD.gn @@ -21,7 +21,7 @@ rtc_source_set("protobuf_utils") { } } -rtc_source_set("bitstream_reader") { +rtc_library("bitstream_reader") { sources = [ "bitstream_reader.cc", "bitstream_reader.h", @@ -49,7 +49,7 @@ rtc_source_set("untyped_function") { deps = [ "system:assume" ] } -rtc_source_set("callback_list") { +rtc_library("callback_list") { sources = [ "callback_list.cc", "callback_list.h", @@ -607,7 +607,7 @@ rtc_source_set("type_traits") { sources = [ "type_traits.h" ] } -rtc_source_set("rtc_operations_chain") { +rtc_library("rtc_operations_chain") { visibility = [ "*" ] sources = [ "operations_chain.cc", @@ -1050,7 +1050,7 @@ rtc_library("socket") { } } -rtc_source_set("network_constants") { +rtc_library("network_constants") { visibility = [ "*" ] sources = [ "network_constants.cc", @@ -1461,7 +1461,7 @@ rtc_source_set("ssl_header") { } } -rtc_source_set("digest") { +rtc_library("digest") { visibility = [ "*" ] sources = [ "message_digest.cc", diff --git a/rtc_base/system/BUILD.gn b/rtc_base/system/BUILD.gn index b18114f107..c1181618e9 100644 --- a/rtc_base/system/BUILD.gn +++ b/rtc_base/system/BUILD.gn @@ -98,7 +98,7 @@ if (is_mac || is_ios) { } } -rtc_source_set("warn_current_thread_is_deadlocked") { +rtc_library("warn_current_thread_is_deadlocked") { sources = [ "warn_current_thread_is_deadlocked.h" ] deps = [] if (is_android && !build_with_chromium) { diff --git a/tools_webrtc/gn_check_autofix.py b/tools_webrtc/gn_check_autofix.py index d750e46f54..1d594f151b 100755 --- a/tools_webrtc/gn_check_autofix.py +++ b/tools_webrtc/gn_check_autofix.py @@ -101,6 +101,21 @@ def fix_errors(filename, missing_deps, deleted_sources): Run(['gn', 'format', filename]) +def fix_source_set(filename, line_to_replace): + with open(filename) as file: + lines = file.readlines() + fixed_file = '' + for line in lines: + if line.strip() == line_to_replace.strip(): + fixed_file += line.replace('rtc_source_set', 'rtc_library') + else: + fixed_file += line + + with open(filename, 'w') as file: + file.write(fixed_file) + + Run(['gn', 'format', filename]) + def first_non_empty(iterable): """Return first item which evaluates to True, or fallback to None.""" return next((x for x in iterable if x), None) @@ -194,13 +209,15 @@ def main(): target_msg = 'The target:' if target_msg not in error: target_msg = 'It is not in any dependency of' + if target_msg not in error: + target_msg = 'rtc_source_set shall not contain cc files' if target_msg not in error: print('\n'.join(error)) continue index = error.index(target_msg) + 1 - path, target = error[index].strip().split(':') if error[index + 1] in ('is including a file from the target:', 'The include file is in the target(s):'): + path, target = error[index].strip().split(':') dep = error[index + 2].strip() dep_path, dep = dep.split(':') dep = rebase(path, dep_path, dep) @@ -214,6 +231,12 @@ def main(): deleted_file = '"' + os.path.basename( error[index + 2].strip()) + '",' deleted_sources.add(deleted_file) + elif target_msg == 'rtc_source_set shall not contain cc files': + path = error[index].strip().split(':', + maxsplit=1)[0].split(' ')[1][2:] + print('Turning', error[index + 1].strip().split(' ')[0], 'in', + path, 'into an rtc_library...') + fix_source_set(path, error[index + 1]) else: print('\n'.join(error)) continue diff --git a/webrtc.gni b/webrtc.gni index 2a407632d6..11ab193ec7 100644 --- a/webrtc.gni +++ b/webrtc.gni @@ -597,6 +597,12 @@ template("rtc_source_set") { visibility = webrtc_default_visibility } + if (defined(invoker.sources)) { + nonheader_sources = filter_include(invoker.sources, [ "*.cc" ]) + assert(nonheader_sources == [], + "rtc_source_set shall not contain cc files") + } + # What's your poison? if (defined(testonly) && testonly) { assert(!defined(poisonous)) From 9133c1ae8e3b5b768631853dbc885257663f5b22 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Wed, 7 May 2025 11:02:28 -0700 Subject: [PATCH 067/292] Roll chromium_revision 413ccd12ab..34c4b0e9eb (1456882:1457124) Change log: https://chromium.googlesource.com/chromium/src/+log/413ccd12ab..34c4b0e9eb Full diff: https://chromium.googlesource.com/chromium/src/+/413ccd12ab..34c4b0e9eb Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/db833313b4..1fad05b2cd * src/buildtools: https://chromium.googlesource.com/chromium/src/buildtools/+log/4347ca8fd6..077a66f30f * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/7a7904b73f..8e6ebc5db0 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/c4492b5004..5a4686c3d8 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/957f2c1a89..a3660c81d4 * src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/cff6ae8a80..16c0e42913 * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/18580cf9c4..fab0a4296b * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/0affb1396d..6b4da08cef * src/third_party/libc++/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git/+log/a9cc573e7c..a01c02c9d4 * src/third_party/r8/cipd: sxgLKZyJNZMVi8cot5yvRzqCFQxX_HbbR0psaeto5RcC..Is-oa6b7OQ1cVHjvG42iz-Lha-F3MBxBBIyC_eTMTg4C * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/ab8639c6e8..025e0e69fe DEPS diff: https://chromium.googlesource.com/chromium/src/+/413ccd12ab..34c4b0e9eb/DEPS No update to Clang. BUG=None Change-Id: Ifc4068a14be4b1caf1405367cbe599caa516f9ad Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390302 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44539} --- DEPS | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/DEPS b/DEPS index c2e8822d9c..36e1acb387 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '413ccd12abe49ad79521af08e38c1208cc19f44a', + 'chromium_revision': '34c4b0e9ebd7a0681413aef552217ca88701afc5', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,24 +73,24 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@db833313b460c52046dd0fbcba586e3b9a4a1e9f', + 'https://chromium.googlesource.com/chromium/src/base@1fad05b2cd652ec7d039b3fd3e7abd19fb89982a', 'src/build': 'https://chromium.googlesource.com/chromium/src/build@56e78d9e6e7ca1f7320ad2c6467d364061c789d7', 'src/buildtools': - 'https://chromium.googlesource.com/chromium/src/buildtools@4347ca8fd6c0c6081dbd9bd05c200c6b170f18bc', + 'https://chromium.googlesource.com/chromium/src/buildtools@077a66f30fcf281b066fafb6dfc60818c238efb6', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. 'src/examples/androidtests/third_party/gradle': { 'url': 'https://chromium.googlesource.com/external/github.com/gradle/gradle.git@f2d1fb54a951d8b11d25748e4711bec8d128d7e3', 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@7a7904b73f694988f7bad1c6fd5a8e4424467e3a', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@8e6ebc5db04cc19908e9a5ea5ca64da2103212e8', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@c4492b50040791ac4dc9cc2125ea550d37c2b3ed', + 'https://chromium.googlesource.com/chromium/src/testing@5a4686c3d8285395cb649995f98488a587c136f6', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@957f2c1a89321e825753b9efddf3ea3ee6633ef8', + 'https://chromium.googlesource.com/chromium/src/third_party@a3660c81d479913c74e510255744ca0d12cd1355', 'src/buildtools/linux64': { 'packages': [ @@ -323,9 +323,9 @@ deps = { 'src/third_party/clang-format/script': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@37f6e68a107df43b7d7e044fd36a13cbae3413f2', 'src/third_party/compiler-rt/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@cff6ae8a80960ceca4d83cf6efa271222d2e7d85', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@16c0e42913a45357f0e557dcc0b7055129c78af6', 'src/third_party/libc++/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@a9cc573e7c591795d11b72d8323ba0e573b55bd6', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@a01c02c9d4acbdae3b7e8a2f3ee58579a9c29f96', 'src/third_party/libc++abi/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@7ff13dd8c66d5047a3ee1e1ba26ddeb70d3d8740', 'src/third_party/llvm-libc/src': @@ -497,13 +497,13 @@ deps = { 'src/third_party/crc32c/src': 'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558', 'src/third_party/depot_tools': - 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@18580cf9c407a3c23aa072b9d745832516de699e', + 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@fab0a4296b8830c569a4dc82285bfb58a5c2fca8', 'src/third_party/ffmpeg': 'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@dcdd0fa51b65a0b1688ff6b8f0cc81908f09ded2', 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@0affb1396d17e4473ded5b221e66c31d9e4c222f', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@6b4da08cefe089b9283931c6ca9acf08df12859d', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -632,7 +632,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/r8', - 'version': 'sxgLKZyJNZMVi8cot5yvRzqCFQxX_HbbR0psaeto5RcC', + 'version': 'Is-oa6b7OQ1cVHjvG42iz-Lha-F3MBxBBIyC_eTMTg4C', }, ], 'condition': 'checkout_android', @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@ab8639c6e8bef8b09642856b1b526e1e4edfcbad', + 'https://chromium.googlesource.com/chromium/src/tools@025e0e69feeef14e5eda7ba71f7b4c380eed4b05', 'src/third_party/espresso': { 'packages': [ From 8ae8263ecc04b6fd78886e9906194e71a574f88c Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Tue, 6 May 2025 08:45:32 -0700 Subject: [PATCH 068/292] IWYU: fix generated protobuf include as shown by modules/audio_processing/aec_dump/capture_stream_info.h Follow-up to forward fix in https://webrtc-review.googlesource.com/c/src/+/386860 BUG=webrtc:42226242 Change-Id: Iac11f9f679bb694dd4c25c1b87f498a5adef1a0d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/387700 Reviewed-by: Mirko Bonadei Reviewed-by: Artem Titov Commit-Queue: Philipp Hancke Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44540} --- api/DEPS | 4 ++++ .../chrome_perf_dashboard_metrics_exporter_test.cc | 8 ++++++++ 2 files changed, 12 insertions(+) diff --git a/api/DEPS b/api/DEPS index 0efc076c3e..c909547c93 100644 --- a/api/DEPS +++ b/api/DEPS @@ -257,6 +257,10 @@ specific_include_rules = { "+p2p/base/port_allocator.h", ], + "chrome_perf_dashboard_metrics_exporter_test.cc": [ + "+external/webrtc/webrtc", # Android platform build. + ], + # .cc files in api/ should not be restricted in what they can #include, # so we re-add all the top-level directories here. (That's because .h # files leak their #includes to whoever's #including them, but .cc files diff --git a/api/test/metrics/chrome_perf_dashboard_metrics_exporter_test.cc b/api/test/metrics/chrome_perf_dashboard_metrics_exporter_test.cc index ae1ab84b30..f420b7a3d5 100644 --- a/api/test/metrics/chrome_perf_dashboard_metrics_exporter_test.cc +++ b/api/test/metrics/chrome_perf_dashboard_metrics_exporter_test.cc @@ -22,6 +22,14 @@ #include "test/gtest.h" #include "test/testsupport/file_utils.h" #include "third_party/catapult/tracing/tracing/value/histogram.h" +// Files generated at build-time by the protobuf compiler. +#ifdef WEBRTC_ENABLE_PROTOBUF +#ifdef WEBRTC_ANDROID_PLATFORM_BUILD +#include "external/webrtc/webrtc/tracing/tracing/proto/histogram.pb.h" +#else +#include "tracing/tracing/proto/histogram.pb.h" +#endif +#endif namespace webrtc { namespace test { From b4e828d0f8f516fad720b597c18444b41216f9c7 Mon Sep 17 00:00:00 2001 From: webrtc-version-updater Date: Wed, 7 May 2025 21:09:28 -0700 Subject: [PATCH 069/292] Update WebRTC code version (2025-05-08T04:08:56). Bug: None Change-Id: I3a625b0a4db30669beabc4ea6c40e1b20eb6820e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390267 Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Cr-Commit-Position: refs/heads/main@{#44541} --- call/version.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/call/version.cc b/call/version.cc index 93b1799f2e..57f30561de 100644 --- a/call/version.cc +++ b/call/version.cc @@ -13,7 +13,7 @@ namespace webrtc { // The timestamp is always in UTC. -const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-07T04:10:34"; +const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-08T04:08:56"; void LoadWebRTCVersionInRegister() { // Using volatile to instruct the compiler to not optimize `p` away even From a2bdc9f510830147eaa9d4ae86c027a5d1c9745b Mon Sep 17 00:00:00 2001 From: Junji Watanabe Date: Thu, 8 May 2025 12:34:59 +0900 Subject: [PATCH 070/292] infra: Switch WebRTC standalone bots from Ninja/Reclient to Siso/Reclient This CL switches all WebRTC standalone bots from Ninja to Siso. It still uses Reclient. Test: https://ci.chromium.org/b/8715471866428000449 Bug: chromium:412968361 Change-Id: Iab56b6f91fc9f7d0f6805b6af8b034fe627c8d5d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389280 Commit-Queue: Christoffer Dewerin Auto-Submit: Junji Watanabe Reviewed-by: Christoffer Dewerin Reviewed-by: Mirko Bonadei Cr-Commit-Position: refs/heads/main@{#44542} --- infra/config/config.star | 15 +- infra/config/cr-buildbucket.cfg | 1056 +++++++++++++++++++++++++++++-- infra/config/project.cfg | 2 +- 3 files changed, 1019 insertions(+), 54 deletions(-) diff --git a/infra/config/config.star b/infra/config/config.star index ec925311d5..f17a0d57ed 100755 --- a/infra/config/config.star +++ b/infra/config/config.star @@ -18,13 +18,12 @@ DEFAULT_CPU = "x86-64" # Helpers: -def make_rbe_properties(instance, jobs = None, use_siso = None): +def make_rbe_properties(instance, jobs = None): """Makes a default RBE property with the specified argument. Args: instance: RBE insatnce name. jobs: Number of jobs to be used by the builder. - use_siso: Add $build/siso properties to switch from Ninja to Siso. Returns: A dictonary with the reclient properties. """ @@ -44,9 +43,8 @@ def make_rbe_properties(instance, jobs = None, use_siso = None): siso_props["remote_jobs"] = jobs props = { "$build/reclient": reclient_props, + "$biuld/siso": siso_props, } - if use_siso: - props["$build/siso"] = siso_props return props def os_from_name(name): @@ -571,7 +569,6 @@ def ci_builder( perf_cat = None, prioritized = False, enabled = True, - use_siso = None, **kwargs): """Add a post-submit builder. @@ -582,7 +579,6 @@ def ci_builder( perf_cat: the category + name for the /perf/ console, or None to omit from the console. prioritized: True to make this builder have a higher priority and never batch builds. enabled: False to exclude this builder from consoles and failure notifications. - use_siso: True to switch build system from Ninja to Siso. **kwargs: Pass on to webrtc_builder / luci.builder. Returns: A luci.builder. @@ -604,7 +600,7 @@ def ci_builder( dimensions["builderless"] = "1" properties = properties or {} properties["builder_group"] = "client.webrtc" - properties.update(make_rbe_properties("rbe-webrtc-trusted", use_siso = use_siso)) + properties.update(make_rbe_properties("rbe-webrtc-trusted")) notifies = ["post_submit_failure_notifier", "infra_failure_notifier"] notifies += ["webrtc_tree_closer"] if name not in skipped_lkgr_bots else [] @@ -627,7 +623,6 @@ def try_builder( cq = {}, branch_cq = True, builder = None, - use_siso = None, **kwargs): """Add a pre-submit builder. @@ -638,7 +633,6 @@ def try_builder( cq: None to exclude this from all commit queues, or a dict of kwargs for cq_tryjob_verifier. branch_cq: False to exclude this builder just from the release-branch CQ. builder: builder to set in the dimensions, if None, builderless:1 is used. - use_siso: True to switch build system from Ninja to Siso. **kwargs: Pass on to webrtc_builder / luci.builder. Returns: A luci.builder. @@ -651,7 +645,7 @@ def try_builder( dimensions["builderless"] = "1" properties = properties or {} properties["builder_group"] = "tryserver.webrtc" - properties.update(make_rbe_properties("rbe-webrtc-untrusted", use_siso = use_siso)) + properties.update(make_rbe_properties("rbe-webrtc-untrusted")) if cq != None: luci.cq_tryjob_verifier(name, cq_group = "cq", **cq) if branch_cq: @@ -720,7 +714,6 @@ def chromium_try_builder(name, **kwargs): recipe = "chromium_trybot", branch_cq = False, execution_timeout = 3 * time.hour, - use_siso = True, **kwargs ) diff --git a/infra/config/cr-buildbucket.cfg b/infra/config/cr-buildbucket.cfg index 9e2c14b094..3028209508 100644 --- a/infra/config/cr-buildbucket.cfg +++ b/infra/config/cr-buildbucket.cfg @@ -29,6 +29,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -76,6 +85,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -123,6 +141,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -170,6 +197,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -217,6 +253,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -264,6 +309,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -311,6 +365,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -358,6 +421,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -405,6 +477,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -452,6 +533,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -499,6 +589,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -546,6 +645,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -593,6 +701,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -640,6 +757,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -687,6 +813,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -734,6 +869,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -781,6 +925,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -828,6 +981,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -875,6 +1037,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -922,6 +1093,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -969,6 +1149,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -1016,6 +1205,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -1063,6 +1261,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -1110,6 +1317,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -1157,6 +1373,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -1204,6 +1429,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -1251,6 +1485,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -1298,6 +1541,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -1345,6 +1597,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -1392,6 +1653,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -1439,6 +1709,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -1486,6 +1765,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -1533,6 +1821,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -1580,6 +1877,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -1632,6 +1938,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -1684,6 +1999,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -1736,10 +2060,19 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -2014,6 +2347,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -2061,6 +2403,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -2108,6 +2459,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -2155,6 +2515,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -2202,6 +2571,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -2249,6 +2627,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -2294,6 +2681,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -2339,6 +2735,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -2384,6 +2789,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -2429,6 +2843,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -2474,6 +2897,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -2519,6 +2951,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -2564,6 +3005,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -2611,6 +3061,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-trusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-trusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -2678,6 +3137,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -2725,6 +3193,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -2772,6 +3249,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -2819,6 +3305,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -2866,6 +3361,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -2913,11 +3417,7 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' - ' "$build/siso": {' + ' "$biuld/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -2926,6 +3426,10 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -2969,6 +3473,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -3016,6 +3529,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -3063,6 +3585,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -3110,6 +3641,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -3157,6 +3697,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -3204,6 +3753,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -3251,6 +3809,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -3298,6 +3865,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -3345,6 +3921,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -3392,6 +3977,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -3444,6 +4038,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -3496,6 +4099,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -3548,9 +4160,18 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' @@ -3600,6 +4221,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -3647,6 +4277,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -3694,11 +4333,7 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' - ' "$build/siso": {' + ' "$biuld/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -3707,6 +4342,10 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -3750,11 +4389,7 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' - ' "$build/siso": {' + ' "$biuld/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -3763,6 +4398,10 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -3806,6 +4445,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -3853,6 +4501,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -3900,6 +4557,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -3947,6 +4613,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -3994,6 +4669,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -4041,6 +4725,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -4088,6 +4781,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -4135,6 +4837,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -4182,6 +4893,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -4229,6 +4949,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -4276,6 +5005,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -4323,6 +5061,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -4370,6 +5117,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -4417,6 +5173,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -4464,6 +5229,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -4511,6 +5285,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -4558,6 +5341,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -4605,6 +5397,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -4652,11 +5453,7 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' - ' "$build/siso": {' + ' "$biuld/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -4665,6 +5462,10 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -4708,6 +5509,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -4755,6 +5565,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -4802,6 +5621,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -4849,6 +5677,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -4896,6 +5733,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -4943,6 +5789,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -4990,6 +5845,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -5039,11 +5903,7 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' - ' "$build/siso": {' + ' "$biuld/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -5052,6 +5912,10 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -5095,6 +5959,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -5142,6 +6015,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -5189,6 +6071,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -5236,11 +6127,7 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' - ' "$build/siso": {' + ' "$biuld/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -5249,6 +6136,10 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -5292,11 +6183,7 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' - ' "$build/siso": {' + ' "$biuld/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -5305,6 +6192,10 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -5348,6 +6239,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -5395,6 +6295,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -5442,6 +6351,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -5489,6 +6407,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -5536,6 +6463,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -5583,6 +6519,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -5630,6 +6575,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -5677,6 +6631,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' @@ -5724,6 +6687,15 @@ buckets { } properties: '{' + ' "$biuld/siso": {' + ' "configs": [' + ' "builder"' + ' ],' + ' "enable_cloud_profiler": true,' + ' "enable_cloud_trace": true,' + ' "enable_monitoring": true,' + ' "project": "rbe-webrtc-untrusted"' + ' },' ' "$build/reclient": {' ' "instance": "rbe-webrtc-untrusted",' ' "metrics_project": "chromium-reclient-metrics"' diff --git a/infra/config/project.cfg b/infra/config/project.cfg index fae10a3043..24599b5d9e 100644 --- a/infra/config/project.cfg +++ b/infra/config/project.cfg @@ -7,7 +7,7 @@ name: "webrtc" access: "group:all" lucicfg { - version: "1.44.1" + version: "1.45.0" package_dir: "." config_dir: "." entry_point: "config.star" From 781a86ee11dfc460c8539e486aa90dee3a57bd88 Mon Sep 17 00:00:00 2001 From: Junji Watanabe Date: Thu, 8 May 2025 16:26:03 +0900 Subject: [PATCH 071/292] infra: fix typo for siso properties MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Bug: chromium:412968361 No-Try: true Change-Id: I60325c6168728d182b4e574a37971c88cd6ab692 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390360 Reviewed-by: Christoffer Dewerin Reviewed-by: Björn Terelius Commit-Queue: Björn Terelius Auto-Submit: Junji Watanabe Cr-Commit-Position: refs/heads/main@{#44543} --- infra/config/config.star | 2 +- infra/config/cr-buildbucket.cfg | 1150 +++++++++++++++---------------- 2 files changed, 576 insertions(+), 576 deletions(-) diff --git a/infra/config/config.star b/infra/config/config.star index f17a0d57ed..c31054c7b6 100755 --- a/infra/config/config.star +++ b/infra/config/config.star @@ -43,7 +43,7 @@ def make_rbe_properties(instance, jobs = None): siso_props["remote_jobs"] = jobs props = { "$build/reclient": reclient_props, - "$biuld/siso": siso_props, + "$build/siso": siso_props, } return props diff --git a/infra/config/cr-buildbucket.cfg b/infra/config/cr-buildbucket.cfg index 3028209508..1d79a2971e 100644 --- a/infra/config/cr-buildbucket.cfg +++ b/infra/config/cr-buildbucket.cfg @@ -29,7 +29,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -38,10 +42,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -85,7 +85,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -94,10 +98,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -141,7 +141,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -150,10 +154,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -197,7 +197,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -206,10 +210,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -253,7 +253,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -262,10 +266,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -309,7 +309,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -318,10 +322,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -365,7 +365,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -374,10 +378,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -421,7 +421,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -430,10 +434,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -477,7 +477,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -486,10 +490,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -533,7 +533,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -542,10 +546,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -589,7 +589,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -598,10 +602,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -645,7 +645,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -654,10 +658,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -701,7 +701,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -710,10 +714,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -757,7 +757,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -766,10 +770,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -813,7 +813,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -822,10 +826,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -869,7 +869,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -878,10 +882,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -925,7 +925,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -934,10 +938,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -981,7 +981,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -990,10 +994,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -1037,7 +1037,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -1046,10 +1050,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -1093,7 +1093,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -1102,10 +1106,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -1149,7 +1149,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -1158,10 +1162,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -1205,7 +1205,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -1214,10 +1218,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -1261,7 +1261,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -1270,10 +1274,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -1317,7 +1317,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -1326,10 +1330,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -1373,7 +1373,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -1382,10 +1386,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -1429,7 +1429,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -1438,10 +1442,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -1485,7 +1485,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -1494,10 +1498,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -1541,7 +1541,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -1550,10 +1554,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -1597,7 +1597,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -1606,10 +1610,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -1653,7 +1653,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -1662,10 +1666,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -1709,7 +1709,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -1718,10 +1722,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -1765,7 +1765,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -1774,10 +1778,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -1821,7 +1821,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -1830,10 +1834,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -1877,7 +1877,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -1886,10 +1890,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -1938,7 +1938,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -1947,10 +1951,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -1999,7 +1999,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -2008,10 +2012,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -2060,7 +2060,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -2069,10 +2073,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -2347,7 +2347,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -2356,10 +2360,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -2403,7 +2403,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -2412,10 +2416,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -2459,7 +2459,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -2468,10 +2472,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -2515,7 +2515,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -2524,10 +2528,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -2571,7 +2571,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -2580,10 +2584,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -2627,7 +2627,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -2636,10 +2640,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -2681,7 +2681,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -2690,10 +2694,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -2735,7 +2735,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -2744,10 +2748,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -2789,7 +2789,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -2798,10 +2802,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -2843,7 +2843,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -2852,10 +2856,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -2897,7 +2897,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -2906,10 +2910,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -2951,7 +2951,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -2960,10 +2964,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -3005,7 +3005,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -3014,10 +3018,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -3061,7 +3061,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-trusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -3070,10 +3074,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-trusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -3137,7 +3137,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -3146,10 +3150,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -3193,7 +3193,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -3202,10 +3206,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -3249,7 +3249,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -3258,10 +3262,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -3305,7 +3305,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -3314,10 +3318,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -3361,7 +3361,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -3370,10 +3374,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -3417,7 +3417,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -3426,10 +3430,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -3473,7 +3473,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -3482,10 +3486,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -3529,7 +3529,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -3538,10 +3542,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -3585,7 +3585,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -3594,10 +3598,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -3641,7 +3641,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -3650,10 +3654,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -3697,7 +3697,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -3706,10 +3710,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -3753,7 +3753,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -3762,10 +3766,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -3809,7 +3809,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -3818,10 +3822,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -3865,7 +3865,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -3874,10 +3878,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -3921,7 +3921,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -3930,10 +3934,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -3977,7 +3977,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -3986,10 +3990,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -4038,7 +4038,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -4047,10 +4051,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -4099,7 +4099,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -4108,10 +4112,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -4160,7 +4160,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -4169,10 +4173,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -4221,7 +4221,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -4230,10 +4234,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -4277,7 +4277,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -4286,10 +4290,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -4333,7 +4333,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -4342,10 +4346,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -4389,7 +4389,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -4398,10 +4402,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -4445,7 +4445,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -4454,10 +4458,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -4501,7 +4501,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -4510,10 +4514,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -4557,7 +4557,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -4566,10 +4570,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -4613,7 +4613,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -4622,10 +4626,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -4669,7 +4669,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -4678,10 +4682,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -4725,7 +4725,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -4734,10 +4738,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -4781,7 +4781,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -4790,10 +4794,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -4837,7 +4837,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -4846,10 +4850,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -4893,7 +4893,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -4902,10 +4906,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -4949,7 +4949,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -4958,10 +4962,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -5005,7 +5005,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -5014,10 +5018,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -5061,7 +5061,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -5070,10 +5074,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -5117,7 +5117,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -5126,10 +5130,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -5173,7 +5173,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -5182,10 +5186,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -5229,7 +5229,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -5238,10 +5242,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -5285,7 +5285,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -5294,10 +5298,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -5341,7 +5341,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -5350,10 +5354,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -5397,7 +5397,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -5406,10 +5410,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -5453,7 +5453,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -5462,10 +5466,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -5509,7 +5509,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -5518,10 +5522,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -5565,7 +5565,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -5574,10 +5578,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -5621,7 +5621,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -5630,10 +5634,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -5677,7 +5677,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -5686,10 +5690,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -5733,7 +5733,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -5742,10 +5746,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -5789,7 +5789,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -5798,10 +5802,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -5845,7 +5845,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -5854,10 +5858,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -5903,7 +5903,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -5912,10 +5916,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -5959,7 +5959,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -5968,10 +5972,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -6015,7 +6015,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -6024,10 +6028,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -6071,7 +6071,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -6080,10 +6084,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -6127,7 +6127,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -6136,10 +6140,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -6183,7 +6183,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -6192,10 +6196,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -6239,7 +6239,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -6248,10 +6252,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -6295,7 +6295,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -6304,10 +6308,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -6351,7 +6351,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -6360,10 +6364,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -6407,7 +6407,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -6416,10 +6420,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -6463,7 +6463,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -6472,10 +6476,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -6519,7 +6519,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -6528,10 +6532,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -6575,7 +6575,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -6584,10 +6588,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -6631,7 +6631,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -6640,10 +6644,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' @@ -6687,7 +6687,11 @@ buckets { } properties: '{' - ' "$biuld/siso": {' + ' "$build/reclient": {' + ' "instance": "rbe-webrtc-untrusted",' + ' "metrics_project": "chromium-reclient-metrics"' + ' },' + ' "$build/siso": {' ' "configs": [' ' "builder"' ' ],' @@ -6696,10 +6700,6 @@ buckets { ' "enable_monitoring": true,' ' "project": "rbe-webrtc-untrusted"' ' },' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$recipe_engine/resultdb/test_presentation": {' ' "column_keys": [],' ' "grouping_keys": [' From ae1af451b5e745b40e93bc0a549c0765d76787b7 Mon Sep 17 00:00:00 2001 From: Fredrik Solenberg Date: Thu, 8 May 2025 02:10:37 -0700 Subject: [PATCH 072/292] Revert "IWYU: fix generated protobuf include" This reverts commit 8ae8263ecc04b6fd78886e9906194e71a574f88c. Reason for revert: Breaks downstream project Original change's description: > IWYU: fix generated protobuf include > > as shown by > modules/audio_processing/aec_dump/capture_stream_info.h > > Follow-up to forward fix in > https://webrtc-review.googlesource.com/c/src/+/386860 > > BUG=webrtc:42226242 > > Change-Id: Iac11f9f679bb694dd4c25c1b87f498a5adef1a0d > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/387700 > Reviewed-by: Mirko Bonadei > Reviewed-by: Artem Titov > Commit-Queue: Philipp Hancke > Reviewed-by: Harald Alvestrand > Cr-Commit-Position: refs/heads/main@{#44540} Bug: webrtc:42226242 Change-Id: Ie13034442c16f6539aef71206584a649d31efeb7 No-Iwyu: Breaks downstream project Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390380 Reviewed-by: Mirko Bonadei Bot-Commit: rubber-stamper@appspot.gserviceaccount.com Commit-Queue: Fredrik Solenberg Cr-Commit-Position: refs/heads/main@{#44544} --- api/DEPS | 4 ---- .../chrome_perf_dashboard_metrics_exporter_test.cc | 8 -------- 2 files changed, 12 deletions(-) diff --git a/api/DEPS b/api/DEPS index c909547c93..0efc076c3e 100644 --- a/api/DEPS +++ b/api/DEPS @@ -257,10 +257,6 @@ specific_include_rules = { "+p2p/base/port_allocator.h", ], - "chrome_perf_dashboard_metrics_exporter_test.cc": [ - "+external/webrtc/webrtc", # Android platform build. - ], - # .cc files in api/ should not be restricted in what they can #include, # so we re-add all the top-level directories here. (That's because .h # files leak their #includes to whoever's #including them, but .cc files diff --git a/api/test/metrics/chrome_perf_dashboard_metrics_exporter_test.cc b/api/test/metrics/chrome_perf_dashboard_metrics_exporter_test.cc index f420b7a3d5..ae1ab84b30 100644 --- a/api/test/metrics/chrome_perf_dashboard_metrics_exporter_test.cc +++ b/api/test/metrics/chrome_perf_dashboard_metrics_exporter_test.cc @@ -22,14 +22,6 @@ #include "test/gtest.h" #include "test/testsupport/file_utils.h" #include "third_party/catapult/tracing/tracing/value/histogram.h" -// Files generated at build-time by the protobuf compiler. -#ifdef WEBRTC_ENABLE_PROTOBUF -#ifdef WEBRTC_ANDROID_PLATFORM_BUILD -#include "external/webrtc/webrtc/tracing/tracing/proto/histogram.pb.h" -#else -#include "tracing/tracing/proto/histogram.pb.h" -#endif -#endif namespace webrtc { namespace test { From 94db4c40db77695ba23144b19b4834dcdbb798c4 Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Tue, 6 May 2025 13:55:13 -0700 Subject: [PATCH 073/292] IWYU video/ using find video/ -name "*.h" -o -name "*.cc" | xargs tools_webrtc/iwyu/apply-include-cleaner followed by tools_webrtc/gn_check_autofix.py -C out/Default/ and git cl format Manual changes: - video_stream_decoder_impl_unittest.cc: unused and removed - quality_scaler_resource_unittest.cc: added degradation_preference_provider.h - task_queue_frame_decode_scheduler.h: added task_queue_base.h and pending_task_safety_flag.h BUG=webrtc:42226242 Change-Id: I4e2b9a52d48b4a114ce78760cf1cd120870353ab Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/387240 Commit-Queue: Fredrik Solenberg Reviewed-by: Sergey Silkin Reviewed-by: Fredrik Solenberg Cr-Commit-Position: refs/heads/main@{#44545} --- video/BUILD.gn | 44 ++++ video/adaptation/BUILD.gn | 11 + video/adaptation/balanced_constraint.cc | 9 +- video/adaptation/balanced_constraint.h | 4 + .../bandwidth_quality_scaler_resource.cc | 18 +- .../bandwidth_quality_scaler_resource.h | 10 +- video/adaptation/bitrate_constraint.cc | 8 +- video/adaptation/bitrate_constraint.h | 2 + .../adaptation/bitrate_constraint_unittest.cc | 5 + video/adaptation/encode_usage_resource.cc | 11 + video/adaptation/encode_usage_resource.h | 4 +- video/adaptation/overuse_frame_detector.cc | 9 + video/adaptation/overuse_frame_detector.h | 4 +- .../overuse_frame_detector_unittest.cc | 11 +- video/adaptation/pixel_limit_resource.cc | 8 + video/adaptation/pixel_limit_resource.h | 1 + .../pixel_limit_resource_unittest.cc | 6 +- video/adaptation/quality_scaler_resource.cc | 12 +- video/adaptation/quality_scaler_resource.h | 9 +- .../quality_scaler_resource_unittest.cc | 8 +- .../video_stream_encoder_resource.cc | 8 +- .../video_stream_encoder_resource.h | 5 +- .../video_stream_encoder_resource_manager.cc | 39 ++- .../video_stream_encoder_resource_manager.h | 18 +- video/alignment_adjuster.cc | 5 + video/alignment_adjuster.h | 3 + video/alignment_adjuster_unittest.cc | 7 +- video/buffered_frame_decryptor.h | 3 +- video/buffered_frame_decryptor_unittest.cc | 16 +- video/call_stats2.cc | 10 +- video/call_stats2.h | 6 +- video/call_stats2_unittest.cc | 5 +- video/config/BUILD.gn | 3 + video/config/encoder_stream_factory.cc | 21 +- video/config/encoder_stream_factory.h | 3 +- video/config/simulcast.h | 1 + video/config/video_encoder_config.cc | 4 + video/config/video_encoder_config.h | 4 +- video/corruption_detection/BUILD.gn | 2 + .../frame_pair_corruption_score.cc | 1 + .../frame_pair_corruption_score_unittest.cc | 1 - .../generic_mapping_functions.cc | 1 + video/corruption_detection/utils.cc | 2 + video/decode_synchronizer.cc | 8 +- video/decode_synchronizer.h | 6 +- video/decode_synchronizer_unittest.cc | 4 + video/encoder_bitrate_adjuster.cc | 16 ++ video/encoder_bitrate_adjuster.h | 8 +- video/encoder_bitrate_adjuster_unittest.cc | 17 +- video/encoder_overshoot_detector.cc | 6 + video/encoder_overshoot_detector.h | 4 +- video/encoder_overshoot_detector_unittest.cc | 6 + video/encoder_rtcp_feedback.cc | 11 + video/encoder_rtcp_feedback.h | 4 +- video/encoder_rtcp_feedback_unittest.cc | 5 +- .../end_to_end_tests/call_operation_tests.cc | 9 +- video/end_to_end_tests/codec_tests.cc | 19 +- .../extended_reports_tests.cc | 14 +- video/end_to_end_tests/fec_tests.cc | 28 ++- .../frame_encryption_tests.cc | 17 +- video/end_to_end_tests/histogram_tests.cc | 16 ++ .../multi_codec_receive_tests.cc | 24 +- video/end_to_end_tests/multi_stream_tester.cc | 16 +- video/end_to_end_tests/multi_stream_tester.h | 6 + video/end_to_end_tests/multi_stream_tests.cc | 1 + .../resolution_bitrate_limits_tests.cc | 24 +- video/end_to_end_tests/ssrc_tests.cc | 16 +- video/end_to_end_tests/stats_tests.cc | 29 ++- .../transport_feedback_tests.cc | 21 ++ video/frame_cadence_adapter.cc | 10 +- video/frame_cadence_adapter.h | 7 +- video/frame_cadence_adapter_unittest.cc | 8 +- video/frame_decode_timing.cc | 6 +- video/frame_decode_timing.h | 6 +- video/frame_decode_timing_unittest.cc | 4 + video/frame_dumping_decoder.cc | 6 +- video/frame_dumping_encoder.cc | 13 +- video/frame_encode_metadata_writer.cc | 18 ++ video/frame_encode_metadata_writer.h | 8 + .../frame_encode_metadata_writer_unittest.cc | 14 +- video/full_stack_tests.cc | 3 +- video/pc_full_stack_tests.cc | 6 +- video/picture_id_tests.cc | 30 ++- video/quality_convergence_controller.cc | 5 + ...quality_convergence_controller_unittest.cc | 1 + video/quality_convergence_monitor.cc | 5 +- video/quality_convergence_monitor.h | 1 + video/quality_convergence_monitor_unittest.cc | 4 +- video/quality_limitation_reason_tracker.cc | 4 + video/quality_limitation_reason_tracker.h | 1 + ...lity_limitation_reason_tracker_unittest.cc | 2 + video/quality_scaling_tests.cc | 31 ++- video/rate_utilization_tracker.cc | 8 + video/rate_utilization_tracker.h | 1 + video/receive_statistics_proxy.cc | 22 +- video/receive_statistics_proxy_unittest.cc | 19 +- video/render/incoming_video_stream.cc | 6 + video/render/video_render_frames.cc | 6 +- video/report_block_stats.cc | 2 +- video/report_block_stats_unittest.cc | 2 + video/rtp_streams_synchronizer2.cc | 7 + video/rtp_streams_synchronizer2.h | 2 + video/screenshare_loopback.cc | 2 +- video/send_delay_stats.cc | 8 + video/send_delay_stats_unittest.cc | 4 + video/send_statistics_proxy.cc | 31 +++ video/send_statistics_proxy_unittest.cc | 16 ++ video/stats_counter.cc | 2 + video/stats_counter_unittest.cc | 2 + video/stream_synchronization.cc | 2 + video/stream_synchronization_unittest.cc | 2 + video/sv_loopback.cc | 1 - video/task_queue_frame_decode_scheduler.cc | 6 + video/task_queue_frame_decode_scheduler.h | 7 + ...k_queue_frame_decode_scheduler_unittest.cc | 4 +- video/test/mock_video_stream_encoder.h | 13 + video/unique_timestamp_counter_unittest.cc | 2 + video/video_loopback.cc | 1 - video/video_quality_observer2.cc | 4 + video/video_quality_observer2.h | 1 - video/video_receive_stream_timeout_tracker.cc | 7 + video/video_receive_stream_timeout_tracker.h | 2 + ...receive_stream_timeout_tracker_unittest.cc | 4 +- video/video_send_stream_impl.h | 13 + video/video_send_stream_impl_unittest.cc | 2 - video/video_source_sink_controller.cc | 13 +- video/video_source_sink_controller.h | 3 +- .../video_source_sink_controller_unittest.cc | 2 + video/video_stream_buffer_controller.cc | 11 +- video/video_stream_buffer_controller.h | 16 +- ...video_stream_buffer_controller_unittest.cc | 4 + video/video_stream_decoder2.cc | 1 + video/video_stream_decoder2.h | 7 - video/video_stream_decoder_impl_unittest.cc | 222 ------------------ video/video_stream_encoder.h | 27 ++- video/video_stream_encoder_interface.h | 8 +- video/video_stream_encoder_observer.h | 1 + video/video_stream_encoder_unittest.cc | 2 +- 138 files changed, 1008 insertions(+), 410 deletions(-) delete mode 100644 video/video_stream_decoder_impl_unittest.cc diff --git a/video/BUILD.gn b/video/BUILD.gn index 548442e193..04ae5c23de 100644 --- a/video/BUILD.gn +++ b/video/BUILD.gn @@ -26,6 +26,7 @@ rtc_library("video_stream_encoder_interface") { "../api/video:video_bitrate_allocator", "../api/video:video_codec_constants", "../api/video:video_frame", + "../api/video:video_frame_type", "../api/video:video_layers_allocation", "../api/video_codecs:video_codecs_api", "../video/config:encoder_config", @@ -143,6 +144,7 @@ rtc_library("video") { "../modules/rtp_rtcp:rtp_rtcp_format", "../modules/rtp_rtcp:rtp_video_header", "../modules/video_coding", + "../modules/video_coding:codec_globals_headers", "../modules/video_coding:h264_sprop_parameter_sets", "../modules/video_coding:h26x_packet_buffer", "../modules/video_coding:nack_requester", @@ -207,6 +209,7 @@ rtc_library("frame_dumping_decoder") { deps = [ "../api/video:encoded_frame", "../api/video:encoded_image", + "../api/video:video_frame", "../api/video_codecs:video_codecs_api", "../modules/video_coding", "../modules/video_coding:video_codec_interface", @@ -224,17 +227,21 @@ rtc_library("frame_dumping_encoder") { ] deps = [ + "../api:fec_controller_api", "../api:field_trials_view", "../api:sequence_checker", "../api/video:encoded_frame", "../api/video:encoded_image", "../api/video:video_frame", + "../api/video:video_frame_type", "../api/video_codecs:video_codecs_api", "../modules/video_coding", "../modules/video_coding:video_codec_interface", "../modules/video_coding:video_coding_utility", + "../rtc_base:macromagic", "../rtc_base:stringutils", "../rtc_base:timeutils", + "../rtc_base/synchronization:mutex", "../rtc_base/system:file_wrapper", "//third_party/abseil-cpp/absl/algorithm:container", ] @@ -249,7 +256,9 @@ rtc_library("frame_cadence_adapter") { deps = [ "../api:field_trials_view", + "../api:scoped_refptr", "../api:sequence_checker", + "../api:video_track_source_constraints", "../api/metronome", "../api/task_queue", "../api/task_queue:pending_task_safety_flag", @@ -291,6 +300,7 @@ rtc_library("video_stream_buffer_controller") { "../api:sequence_checker", "../api/metronome", "../api/task_queue", + "../api/task_queue:pending_task_safety_flag", "../api/units:data_size", "../api/units:time_delta", "../api/units:timestamp", @@ -306,9 +316,12 @@ rtc_library("video_stream_buffer_controller") { "../rtc_base:checks", "../rtc_base:logging", "../rtc_base:macromagic", + "../rtc_base/experiments:field_trial_parser", + "../rtc_base/system:no_unique_address", "../system_wrappers", "../system_wrappers:field_trial", "//third_party/abseil-cpp/absl/base:core_headers", + "//third_party/abseil-cpp/absl/container:inlined_vector", "//third_party/abseil-cpp/absl/functional:bind_front", ] } @@ -333,6 +346,7 @@ rtc_library("task_queue_frame_decode_scheduler") { "../api:sequence_checker", "../api/task_queue", "../api/task_queue:pending_task_safety_flag", + "../api/units:time_delta", "../api/units:timestamp", "../rtc_base:checks", "../system_wrappers", @@ -348,7 +362,9 @@ rtc_library("frame_decode_timing") { "../api/task_queue", "../api/task_queue:pending_task_safety_flag", "../api/units:time_delta", + "../api/units:timestamp", "../modules/video_coding/timing:timing_module", + "../rtc_base:checks", "../rtc_base:logging", "../system_wrappers", ] @@ -360,10 +376,13 @@ rtc_library("video_receive_stream_timeout_tracker") { "video_receive_stream_timeout_tracker.h", ] deps = [ + "../api:sequence_checker", "../api/task_queue", "../api/units:time_delta", "../api/units:timestamp", + "../rtc_base:checks", "../rtc_base:logging", + "../rtc_base:macromagic", "../rtc_base/task_utils:repeating_task", "../system_wrappers", ] @@ -380,12 +399,14 @@ rtc_library("decode_synchronizer") { "../api:sequence_checker", "../api/metronome", "../api/task_queue", + "../api/task_queue:pending_task_safety_flag", "../api/units:time_delta", "../api/units:timestamp", "../rtc_base:checks", "../rtc_base:event_tracer", "../rtc_base:logging", "../rtc_base:macromagic", + "../system_wrappers", ] } @@ -422,6 +443,7 @@ rtc_library("video_stream_encoder_impl") { "../api:field_trials_view", "../api:make_ref_counted", "../api:rtc_error", + "../api:rtp_packet_info", "../api:rtp_parameters", "../api:rtp_sender_interface", "../api:scoped_refptr", @@ -460,7 +482,9 @@ rtc_library("video_stream_encoder_impl") { "../modules/video_coding:webrtc_vp9_helpers", "../modules/video_coding/svc:scalability_mode_util", "../modules/video_coding/svc:scalability_structures", + "../modules/video_coding/svc:scalable_video_controller", "../modules/video_coding/svc:svc_rate_allocator", + "../rtc_base:buffer", "../rtc_base:checks", "../rtc_base:criticalsection", "../rtc_base:event_tracer", @@ -508,8 +532,18 @@ if (rtc_include_tests) { sources = [ "test/mock_video_stream_encoder.h" ] deps = [ ":video_stream_encoder_interface", + "../api:fec_controller_api", + "../api:rtp_parameters", + "../api:rtp_sender_interface", + "../api:scoped_refptr", + "../api/adaptation:resource_adaptation_api", + "../api/units:data_rate", + "../api/video:video_frame", + "../api/video:video_frame_type", "../api/video:video_stream_encoder", + "../api/video_codecs:video_codecs_api", "../test:test_support", + "config:encoder_config", ] } if (!build_with_chromium) { @@ -632,6 +666,7 @@ if (rtc_include_tests) { sources = [ "full_stack_tests.cc" ] deps = [ ":video_quality_test", + "../api:rtp_parameters", "../api:simulated_network_api", "../api:test_dependency_factory", "../api:video_quality_test_fixture_api", @@ -662,6 +697,7 @@ if (rtc_include_tests) { "../api:create_peer_connection_quality_test_frame_generator", "../api:create_peerconnection_quality_test_fixture", "../api:frame_generator_api", + "../api:function_view", "../api:media_stream_interface", "../api:network_emulation_manager_api", "../api:peer_connection_quality_test_fixture_api", @@ -671,6 +707,9 @@ if (rtc_include_tests) { "../api/test/pclf:media_configuration", "../api/test/pclf:media_quality_test_params", "../api/test/pclf:peer_configurer", + "../api/transport:bitrate_settings", + "../api/units:data_rate", + "../api/units:time_delta", "../api/video_codecs:video_codecs_api", "../media:media_constants", "../modules/video_coding:webrtc_vp9", @@ -885,9 +924,11 @@ if (rtc_include_tests) { "../api:transport_api", "../api:video_track_source_constraints", "../api/adaptation:resource_adaptation_api", + "../api/crypto:frame_decryptor_interface", "../api/crypto:options", "../api/environment", "../api/environment:environment_factory", + "../api/metronome", "../api/metronome/test:fake_metronome", "../api/rtc_event_log", "../api/task_queue", @@ -939,6 +980,7 @@ if (rtc_include_tests) { "../call/adaptation:resource_adaptation", "../call/adaptation:resource_adaptation_test_utilities", "../common_video", + "../common_video:frame_counts", "../common_video:frame_instrumentation_data", "../common_video/test:utilities", "../media:codec", @@ -977,7 +1019,9 @@ if (rtc_include_tests) { "../rtc_base:gunit_helpers", "../rtc_base:logging", "../rtc_base:macromagic", + "../rtc_base:mod_ops", "../rtc_base:network_route", + "../rtc_base:random", "../rtc_base:rate_limiter", "../rtc_base:rate_statistics", "../rtc_base:refcount", diff --git a/video/adaptation/BUILD.gn b/video/adaptation/BUILD.gn index 829d1f1305..1337140c8f 100644 --- a/video/adaptation/BUILD.gn +++ b/video/adaptation/BUILD.gn @@ -32,6 +32,7 @@ rtc_library("video_adaptation") { deps = [ "../../api:field_trials_view", + "../../api:make_ref_counted", "../../api:rtp_parameters", "../../api:scoped_refptr", "../../api:sequence_checker", @@ -39,7 +40,9 @@ rtc_library("video_adaptation") { "../../api/environment", "../../api/task_queue:task_queue", "../../api/units:data_rate", + "../../api/units:data_size", "../../api/units:time_delta", + "../../api/video:encoded_image", "../../api/video:video_adaptation", "../../api/video:video_frame", "../../api/video:video_stream_encoder", @@ -60,6 +63,7 @@ rtc_library("video_adaptation") { "../../rtc_base/experiments:balanced_degradation_settings", "../../rtc_base/experiments:field_trial_parser", "../../rtc_base/experiments:quality_scaler_settings", + "../../rtc_base/experiments:quality_scaling_experiment", "../../rtc_base/synchronization:mutex", "../../rtc_base/system:no_unique_address", "../../rtc_base/task_utils:repeating_task", @@ -84,8 +88,12 @@ if (rtc_include_tests) { ] deps = [ ":video_adaptation", + "..:video_stream_encoder_interface", + "../../api:array_view", "../../api:field_trials_view", + "../../api:rtp_parameters", "../../api:scoped_refptr", + "../../api/adaptation:resource_adaptation_api", "../../api/environment", "../../api/environment:environment_factory", "../../api/task_queue:task_queue", @@ -94,6 +102,7 @@ if (rtc_include_tests) { "../../api/video:encoded_image", "../../api/video:video_adaptation", "../../api/video:video_frame", + "../../api/video:video_rtp_headers", "../../api/video_codecs:scalability_mode", "../../api/video_codecs:video_codecs_api", "../../call/adaptation:resource_adaptation", @@ -107,9 +116,11 @@ if (rtc_include_tests) { "../../rtc_base:rtc_numerics", "../../rtc_base:task_queue_for_test", "../../rtc_base:threading", + "../../rtc_base:timeutils", "../../test:rtc_expect_death", "../../test:test_support", "../../test/time_controller:time_controller", + "../config:encoder_config", "//third_party/abseil-cpp/absl/functional:any_invocable", ] } diff --git a/video/adaptation/balanced_constraint.cc b/video/adaptation/balanced_constraint.cc index 0e8975ca97..27f3eca1a4 100644 --- a/video/adaptation/balanced_constraint.cc +++ b/video/adaptation/balanced_constraint.cc @@ -10,10 +10,17 @@ #include "video/adaptation/balanced_constraint.h" -#include +#include +#include #include +#include "api/field_trials_view.h" +#include "api/rtp_parameters.h" #include "api/sequence_checker.h" +#include "call/adaptation/degradation_preference_provider.h" +#include "call/adaptation/video_source_restrictions.h" +#include "call/adaptation/video_stream_input_state.h" +#include "rtc_base/checks.h" namespace webrtc { diff --git a/video/adaptation/balanced_constraint.h b/video/adaptation/balanced_constraint.h index 294fe6f70c..12375635f6 100644 --- a/video/adaptation/balanced_constraint.h +++ b/video/adaptation/balanced_constraint.h @@ -11,6 +11,7 @@ #ifndef VIDEO_ADAPTATION_BALANCED_CONSTRAINT_H_ #define VIDEO_ADAPTATION_BALANCED_CONSTRAINT_H_ +#include #include #include @@ -18,8 +19,11 @@ #include "api/sequence_checker.h" #include "call/adaptation/adaptation_constraint.h" #include "call/adaptation/degradation_preference_provider.h" +#include "call/adaptation/video_source_restrictions.h" +#include "call/adaptation/video_stream_input_state.h" #include "rtc_base/experiments/balanced_degradation_settings.h" #include "rtc_base/system/no_unique_address.h" +#include "rtc_base/thread_annotations.h" namespace webrtc { diff --git a/video/adaptation/bandwidth_quality_scaler_resource.cc b/video/adaptation/bandwidth_quality_scaler_resource.cc index 815e07ed76..3b104e4b26 100644 --- a/video/adaptation/bandwidth_quality_scaler_resource.cc +++ b/video/adaptation/bandwidth_quality_scaler_resource.cc @@ -10,12 +10,20 @@ #include "video/adaptation/bandwidth_quality_scaler_resource.h" -#include - +#include +#include +#include + +#include "api/adaptation/resource.h" +#include "api/make_ref_counted.h" +#include "api/scoped_refptr.h" +#include "api/sequence_checker.h" +#include "api/video/encoded_image.h" +#include "api/video/video_codec_type.h" +#include "api/video_codecs/video_encoder.h" +#include "modules/video_coding/utility/bandwidth_quality_scaler.h" #include "rtc_base/checks.h" -#include "rtc_base/experiments/balanced_degradation_settings.h" -#include "rtc_base/logging.h" -#include "rtc_base/time_utils.h" +#include "video/adaptation/video_stream_encoder_resource.h" namespace webrtc { diff --git a/video/adaptation/bandwidth_quality_scaler_resource.h b/video/adaptation/bandwidth_quality_scaler_resource.h index 9661e75e6f..14540d668f 100644 --- a/video/adaptation/bandwidth_quality_scaler_resource.h +++ b/video/adaptation/bandwidth_quality_scaler_resource.h @@ -11,18 +11,16 @@ #ifndef VIDEO_ADAPTATION_BANDWIDTH_QUALITY_SCALER_RESOURCE_H_ #define VIDEO_ADAPTATION_BANDWIDTH_QUALITY_SCALER_RESOURCE_H_ +#include #include -#include -#include -#include #include #include "api/scoped_refptr.h" -#include "api/video/video_adaptation_reason.h" +#include "api/video/encoded_image.h" +#include "api/video/video_codec_type.h" #include "api/video_codecs/video_encoder.h" -#include "call/adaptation/degradation_preference_provider.h" -#include "call/adaptation/resource_adaptation_processor_interface.h" #include "modules/video_coding/utility/bandwidth_quality_scaler.h" +#include "rtc_base/thread_annotations.h" #include "video/adaptation/video_stream_encoder_resource.h" namespace webrtc { diff --git a/video/adaptation/bitrate_constraint.cc b/video/adaptation/bitrate_constraint.cc index 6eb480bf7f..cd3f5348ce 100644 --- a/video/adaptation/bitrate_constraint.cc +++ b/video/adaptation/bitrate_constraint.cc @@ -10,11 +10,17 @@ #include "video/adaptation/bitrate_constraint.h" +#include +#include #include -#include #include "api/sequence_checker.h" +#include "api/video_codecs/video_encoder.h" +#include "call/adaptation/encoder_settings.h" +#include "call/adaptation/video_source_restrictions.h" #include "call/adaptation/video_stream_adapter.h" +#include "call/adaptation/video_stream_input_state.h" +#include "rtc_base/checks.h" #include "video/adaptation/video_stream_encoder_resource_manager.h" namespace webrtc { diff --git a/video/adaptation/bitrate_constraint.h b/video/adaptation/bitrate_constraint.h index 8906a16d13..d57500fbcf 100644 --- a/video/adaptation/bitrate_constraint.h +++ b/video/adaptation/bitrate_constraint.h @@ -11,6 +11,7 @@ #ifndef VIDEO_ADAPTATION_BITRATE_CONSTRAINT_H_ #define VIDEO_ADAPTATION_BITRATE_CONSTRAINT_H_ +#include #include #include @@ -20,6 +21,7 @@ #include "call/adaptation/video_source_restrictions.h" #include "call/adaptation/video_stream_input_state.h" #include "rtc_base/system/no_unique_address.h" +#include "rtc_base/thread_annotations.h" namespace webrtc { diff --git a/video/adaptation/bitrate_constraint_unittest.cc b/video/adaptation/bitrate_constraint_unittest.cc index 29ebb6a912..3d4e9fe227 100644 --- a/video/adaptation/bitrate_constraint_unittest.cc +++ b/video/adaptation/bitrate_constraint_unittest.cc @@ -10,16 +10,21 @@ #include "video/adaptation/bitrate_constraint.h" +#include +#include #include #include +#include "api/video/video_codec_type.h" #include "api/video_codecs/scalability_mode.h" +#include "api/video_codecs/video_codec.h" #include "api/video_codecs/video_encoder.h" #include "call/adaptation/encoder_settings.h" #include "call/adaptation/test/fake_frame_rate_provider.h" #include "call/adaptation/video_source_restrictions.h" #include "call/adaptation/video_stream_input_state_provider.h" #include "test/gtest.h" +#include "video/config/video_encoder_config.h" namespace webrtc { diff --git a/video/adaptation/encode_usage_resource.cc b/video/adaptation/encode_usage_resource.cc index 3ca08cbcf2..0496a6c196 100644 --- a/video/adaptation/encode_usage_resource.cc +++ b/video/adaptation/encode_usage_resource.cc @@ -10,10 +10,21 @@ #include "video/adaptation/encode_usage_resource.h" +#include #include +#include +#include #include +#include "api/adaptation/resource.h" +#include "api/make_ref_counted.h" +#include "api/scoped_refptr.h" +#include "api/sequence_checker.h" +#include "api/task_queue/task_queue_base.h" +#include "api/video/video_frame.h" #include "rtc_base/checks.h" +#include "video/adaptation/overuse_frame_detector.h" +#include "video/adaptation/video_stream_encoder_resource.h" namespace webrtc { diff --git a/video/adaptation/encode_usage_resource.h b/video/adaptation/encode_usage_resource.h index 7820859d9d..5b4dc02f33 100644 --- a/video/adaptation/encode_usage_resource.h +++ b/video/adaptation/encode_usage_resource.h @@ -11,12 +11,12 @@ #ifndef VIDEO_ADAPTATION_ENCODE_USAGE_RESOURCE_H_ #define VIDEO_ADAPTATION_ENCODE_USAGE_RESOURCE_H_ +#include #include #include -#include #include "api/scoped_refptr.h" -#include "api/video/video_adaptation_reason.h" +#include "rtc_base/thread_annotations.h" #include "video/adaptation/overuse_frame_detector.h" #include "video/adaptation/video_stream_encoder_resource.h" diff --git a/video/adaptation/overuse_frame_detector.cc b/video/adaptation/overuse_frame_detector.cc index d095d81d85..000f4bfb98 100644 --- a/video/adaptation/overuse_frame_detector.cc +++ b/video/adaptation/overuse_frame_detector.cc @@ -14,20 +14,29 @@ #include #include +#include #include #include #include +#include #include +#include #include #include "api/environment/environment.h" #include "api/field_trials_view.h" +#include "api/sequence_checker.h" +#include "api/task_queue/task_queue_base.h" +#include "api/units/time_delta.h" #include "api/video/video_frame.h" #include "rtc_base/checks.h" +#include "rtc_base/experiments/field_trial_parser.h" #include "rtc_base/logging.h" #include "rtc_base/numerics/exp_filter.h" +#include "rtc_base/task_utils/repeating_task.h" #include "rtc_base/time_utils.h" #include "rtc_base/trace_event.h" +#include "video/video_stream_encoder_observer.h" #if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS) #include diff --git a/video/adaptation/overuse_frame_detector.h b/video/adaptation/overuse_frame_detector.h index bfdc8ebd1d..719f1fa28b 100644 --- a/video/adaptation/overuse_frame_detector.h +++ b/video/adaptation/overuse_frame_detector.h @@ -11,7 +11,7 @@ #ifndef VIDEO_ADAPTATION_OVERUSE_FRAME_DETECTOR_H_ #define VIDEO_ADAPTATION_OVERUSE_FRAME_DETECTOR_H_ -#include +#include #include #include @@ -19,8 +19,8 @@ #include "api/field_trials_view.h" #include "api/sequence_checker.h" #include "api/task_queue/task_queue_base.h" +#include "api/units/time_delta.h" #include "rtc_base/experiments/field_trial_parser.h" -#include "rtc_base/numerics/exp_filter.h" #include "rtc_base/system/no_unique_address.h" #include "rtc_base/task_utils/repeating_task.h" #include "rtc_base/thread_annotations.h" diff --git a/video/adaptation/overuse_frame_detector_unittest.cc b/video/adaptation/overuse_frame_detector_unittest.cc index fa8530f00d..fbcd92e20f 100644 --- a/video/adaptation/overuse_frame_detector_unittest.cc +++ b/video/adaptation/overuse_frame_detector_unittest.cc @@ -10,20 +10,25 @@ #include "video/adaptation/overuse_frame_detector.h" +#include +#include #include +#include +#include "api/array_view.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" -#include "api/video/encoded_image.h" +#include "api/units/time_delta.h" #include "api/video/i420_buffer.h" -#include "api/video/video_adaptation_reason.h" -#include "modules/video_coding/utility/quality_scaler.h" +#include "api/video/video_rotation.h" #include "rtc_base/event.h" #include "rtc_base/fake_clock.h" #include "rtc_base/random.h" #include "rtc_base/task_queue_for_test.h" +#include "rtc_base/time_utils.h" #include "test/gmock.h" #include "test/gtest.h" +#include "video/video_stream_encoder_observer.h" namespace webrtc { diff --git a/video/adaptation/pixel_limit_resource.cc b/video/adaptation/pixel_limit_resource.cc index 446824fa78..04730b6147 100644 --- a/video/adaptation/pixel_limit_resource.cc +++ b/video/adaptation/pixel_limit_resource.cc @@ -10,10 +10,18 @@ #include "video/adaptation/pixel_limit_resource.h" +#include + +#include "api/adaptation/resource.h" +#include "api/make_ref_counted.h" +#include "api/scoped_refptr.h" #include "api/sequence_checker.h" +#include "api/task_queue/task_queue_base.h" #include "api/units/time_delta.h" #include "call/adaptation/video_stream_adapter.h" +#include "call/adaptation/video_stream_input_state_provider.h" #include "rtc_base/checks.h" +#include "rtc_base/task_utils/repeating_task.h" namespace webrtc { diff --git a/video/adaptation/pixel_limit_resource.h b/video/adaptation/pixel_limit_resource.h index 591de753f0..38d20f7bfe 100644 --- a/video/adaptation/pixel_limit_resource.h +++ b/video/adaptation/pixel_limit_resource.h @@ -16,6 +16,7 @@ #include "api/adaptation/resource.h" #include "api/scoped_refptr.h" +#include "api/task_queue/task_queue_base.h" #include "call/adaptation/video_stream_input_state_provider.h" #include "rtc_base/task_utils/repeating_task.h" #include "rtc_base/thread_annotations.h" diff --git a/video/adaptation/pixel_limit_resource_unittest.cc b/video/adaptation/pixel_limit_resource_unittest.cc index 4ed3daa494..ec6d2a8741 100644 --- a/video/adaptation/pixel_limit_resource_unittest.cc +++ b/video/adaptation/pixel_limit_resource_unittest.cc @@ -14,11 +14,15 @@ #include #include "absl/functional/any_invocable.h" +#include "api/adaptation/resource.h" +#include "api/scoped_refptr.h" +#include "api/task_queue/task_queue_base.h" +#include "api/task_queue/task_queue_factory.h" +#include "api/units/time_delta.h" #include "api/units/timestamp.h" #include "call/adaptation/test/fake_video_stream_input_state_provider.h" #include "call/adaptation/test/mock_resource_listener.h" #include "call/adaptation/video_stream_adapter.h" -#include "rtc_base/task_queue_for_test.h" #include "test/gmock.h" #include "test/gtest.h" #include "test/time_controller/simulated_time_controller.h" diff --git a/video/adaptation/quality_scaler_resource.cc b/video/adaptation/quality_scaler_resource.cc index 19ffbf8937..6f6e93caed 100644 --- a/video/adaptation/quality_scaler_resource.cc +++ b/video/adaptation/quality_scaler_resource.cc @@ -10,12 +10,20 @@ #include "video/adaptation/quality_scaler_resource.h" +#include +#include #include +#include "api/adaptation/resource.h" #include "api/field_trials_view.h" +#include "api/make_ref_counted.h" +#include "api/scoped_refptr.h" +#include "api/sequence_checker.h" +#include "api/video/encoded_image.h" +#include "api/video_codecs/video_encoder.h" +#include "modules/video_coding/utility/quality_scaler.h" #include "rtc_base/checks.h" -#include "rtc_base/experiments/balanced_degradation_settings.h" -#include "rtc_base/time_utils.h" +#include "video/adaptation/video_stream_encoder_resource.h" namespace webrtc { diff --git a/video/adaptation/quality_scaler_resource.h b/video/adaptation/quality_scaler_resource.h index 88aa40c936..2e916bb586 100644 --- a/video/adaptation/quality_scaler_resource.h +++ b/video/adaptation/quality_scaler_resource.h @@ -11,18 +11,15 @@ #ifndef VIDEO_ADAPTATION_QUALITY_SCALER_RESOURCE_H_ #define VIDEO_ADAPTATION_QUALITY_SCALER_RESOURCE_H_ +#include #include -#include -#include -#include #include "api/field_trials_view.h" #include "api/scoped_refptr.h" -#include "api/video/video_adaptation_reason.h" +#include "api/video/encoded_image.h" #include "api/video_codecs/video_encoder.h" -#include "call/adaptation/degradation_preference_provider.h" -#include "call/adaptation/resource_adaptation_processor_interface.h" #include "modules/video_coding/utility/quality_scaler.h" +#include "rtc_base/thread_annotations.h" #include "video/adaptation/video_stream_encoder_resource.h" namespace webrtc { diff --git a/video/adaptation/quality_scaler_resource_unittest.cc b/video/adaptation/quality_scaler_resource_unittest.cc index b83db37545..32dfa3bab3 100644 --- a/video/adaptation/quality_scaler_resource_unittest.cc +++ b/video/adaptation/quality_scaler_resource_unittest.cc @@ -10,11 +10,11 @@ #include "video/adaptation/quality_scaler_resource.h" -#include -#include - +#include "api/adaptation/resource.h" +#include "api/rtp_parameters.h" +#include "api/scoped_refptr.h" #include "api/task_queue/task_queue_base.h" -#include "api/video_codecs/video_encoder.h" +#include "call/adaptation/degradation_preference_provider.h" #include "call/adaptation/test/mock_resource_listener.h" #include "rtc_base/thread.h" #include "test/gmock.h" diff --git a/video/adaptation/video_stream_encoder_resource.cc b/video/adaptation/video_stream_encoder_resource.cc index 7793315f93..84b8f04529 100644 --- a/video/adaptation/video_stream_encoder_resource.cc +++ b/video/adaptation/video_stream_encoder_resource.cc @@ -10,9 +10,15 @@ #include "video/adaptation/video_stream_encoder_resource.h" -#include +#include #include +#include "api/adaptation/resource.h" +#include "api/scoped_refptr.h" +#include "api/task_queue/task_queue_base.h" +#include "rtc_base/checks.h" +#include "rtc_base/synchronization/mutex.h" + namespace webrtc { VideoStreamEncoderResource::VideoStreamEncoderResource(std::string name) diff --git a/video/adaptation/video_stream_encoder_resource.h b/video/adaptation/video_stream_encoder_resource.h index 5e93b31ec3..620043dc8b 100644 --- a/video/adaptation/video_stream_encoder_resource.h +++ b/video/adaptation/video_stream_encoder_resource.h @@ -11,15 +11,12 @@ #ifndef VIDEO_ADAPTATION_VIDEO_STREAM_ENCODER_RESOURCE_H_ #define VIDEO_ADAPTATION_VIDEO_STREAM_ENCODER_RESOURCE_H_ -#include #include -#include #include "api/adaptation/resource.h" -#include "api/sequence_checker.h" #include "api/task_queue/task_queue_base.h" -#include "call/adaptation/adaptation_constraint.h" #include "rtc_base/synchronization/mutex.h" +#include "rtc_base/thread_annotations.h" namespace webrtc { diff --git a/video/adaptation/video_stream_encoder_resource_manager.cc b/video/adaptation/video_stream_encoder_resource_manager.cc index ae575de690..20664b91cb 100644 --- a/video/adaptation/video_stream_encoder_resource_manager.cc +++ b/video/adaptation/video_stream_encoder_resource_manager.cc @@ -13,28 +13,55 @@ #include #include -#include -#include +#include +#include #include +#include +#include +#include #include +#include -#include "absl/algorithm/container.h" -#include "absl/base/macros.h" #include "api/adaptation/resource.h" #include "api/field_trials_view.h" +#include "api/rtp_parameters.h" +#include "api/scoped_refptr.h" #include "api/sequence_checker.h" #include "api/task_queue/task_queue_base.h" +#include "api/units/data_rate.h" +#include "api/units/data_size.h" +#include "api/video/encoded_image.h" +#include "api/video/video_adaptation_counters.h" #include "api/video/video_adaptation_reason.h" -#include "api/video/video_source_interface.h" +#include "api/video/video_codec_type.h" +#include "api/video/video_frame.h" +#include "api/video_codecs/video_codec.h" +#include "api/video_codecs/video_encoder.h" +#include "call/adaptation/adaptation_constraint.h" +#include "call/adaptation/degradation_preference_provider.h" +#include "call/adaptation/encoder_settings.h" +#include "call/adaptation/resource_adaptation_processor_interface.h" #include "call/adaptation/video_source_restrictions.h" +#include "call/adaptation/video_stream_adapter.h" +#include "call/adaptation/video_stream_input_state_provider.h" #include "modules/video_coding/svc/scalability_mode_util.h" #include "rtc_base/checks.h" +#include "rtc_base/experiments/quality_scaler_settings.h" +#include "rtc_base/experiments/quality_scaling_experiment.h" #include "rtc_base/logging.h" -#include "rtc_base/numerics/safe_conversions.h" #include "rtc_base/strings/string_builder.h" #include "rtc_base/time_utils.h" #include "rtc_base/trace_event.h" +#include "system_wrappers/include/clock.h" +#include "video/adaptation/balanced_constraint.h" +#include "video/adaptation/bandwidth_quality_scaler_resource.h" +#include "video/adaptation/bitrate_constraint.h" +#include "video/adaptation/encode_usage_resource.h" +#include "video/adaptation/overuse_frame_detector.h" +#include "video/adaptation/pixel_limit_resource.h" #include "video/adaptation/quality_scaler_resource.h" +#include "video/config/video_encoder_config.h" +#include "video/video_stream_encoder_observer.h" namespace webrtc { diff --git a/video/adaptation/video_stream_encoder_resource_manager.h b/video/adaptation/video_stream_encoder_resource_manager.h index 1520bd5aef..34c31af2ab 100644 --- a/video/adaptation/video_stream_encoder_resource_manager.h +++ b/video/adaptation/video_stream_encoder_resource_manager.h @@ -11,13 +11,11 @@ #ifndef VIDEO_ADAPTATION_VIDEO_STREAM_ENCODER_RESOURCE_MANAGER_H_ #define VIDEO_ADAPTATION_VIDEO_STREAM_ENCODER_RESOURCE_MANAGER_H_ -#include +#include #include #include #include #include -#include -#include #include #include "api/adaptation/resource.h" @@ -25,19 +23,22 @@ #include "api/rtp_parameters.h" #include "api/scoped_refptr.h" #include "api/task_queue/task_queue_base.h" +#include "api/units/data_rate.h" +#include "api/units/data_size.h" #include "api/video/video_adaptation_counters.h" #include "api/video/video_adaptation_reason.h" +#include "api/video/video_codec_type.h" #include "api/video/video_frame.h" -#include "api/video/video_source_interface.h" #include "api/video_codecs/video_codec.h" #include "api/video_codecs/video_encoder.h" +#include "call/adaptation/adaptation_constraint.h" +#include "call/adaptation/degradation_preference_provider.h" +#include "call/adaptation/encoder_settings.h" #include "call/adaptation/resource_adaptation_processor_interface.h" +#include "call/adaptation/video_source_restrictions.h" #include "call/adaptation/video_stream_adapter.h" #include "call/adaptation/video_stream_input_state_provider.h" -#include "rtc_base/experiments/quality_scaler_settings.h" -#include "rtc_base/ref_count.h" -#include "rtc_base/strings/string_builder.h" -#include "rtc_base/synchronization/mutex.h" +#include "rtc_base/experiments/balanced_degradation_settings.h" #include "rtc_base/thread_annotations.h" #include "system_wrappers/include/clock.h" #include "video/adaptation/balanced_constraint.h" @@ -47,7 +48,6 @@ #include "video/adaptation/overuse_frame_detector.h" #include "video/adaptation/pixel_limit_resource.h" #include "video/adaptation/quality_scaler_resource.h" -#include "video/adaptation/video_stream_encoder_resource.h" #include "video/config/video_encoder_config.h" #include "video/video_stream_encoder_observer.h" diff --git a/video/alignment_adjuster.cc b/video/alignment_adjuster.cc index de32992837..c51d530422 100644 --- a/video/alignment_adjuster.cc +++ b/video/alignment_adjuster.cc @@ -11,10 +11,15 @@ #include "video/alignment_adjuster.h" #include +#include +#include #include +#include #include "absl/algorithm/container.h" +#include "api/video_codecs/video_encoder.h" #include "rtc_base/logging.h" +#include "video/config/video_encoder_config.h" namespace webrtc { namespace { diff --git a/video/alignment_adjuster.h b/video/alignment_adjuster.h index 086f633f96..fb2bc729c0 100644 --- a/video/alignment_adjuster.h +++ b/video/alignment_adjuster.h @@ -11,6 +11,9 @@ #ifndef VIDEO_ALIGNMENT_ADJUSTER_H_ #define VIDEO_ALIGNMENT_ADJUSTER_H_ +#include +#include + #include "api/video_codecs/video_encoder.h" #include "video/config/video_encoder_config.h" diff --git a/video/alignment_adjuster_unittest.cc b/video/alignment_adjuster_unittest.cc index 95a5e1346e..60e475c9ba 100644 --- a/video/alignment_adjuster_unittest.cc +++ b/video/alignment_adjuster_unittest.cc @@ -10,13 +10,16 @@ #include "video/alignment_adjuster.h" -#include +#include +#include #include #include -#include "rtc_base/numerics/safe_conversions.h" +#include "api/video/video_codec_type.h" +#include "api/video_codecs/video_encoder.h" #include "test/encoder_settings.h" #include "test/gtest.h" +#include "video/config/video_encoder_config.h" namespace webrtc { namespace test { diff --git a/video/buffered_frame_decryptor.h b/video/buffered_frame_decryptor.h index 2c857f8c98..2e28c37a12 100644 --- a/video/buffered_frame_decryptor.h +++ b/video/buffered_frame_decryptor.h @@ -11,12 +11,13 @@ #ifndef VIDEO_BUFFERED_FRAME_DECRYPTOR_H_ #define VIDEO_BUFFERED_FRAME_DECRYPTOR_H_ +#include #include #include -#include "api/crypto/crypto_options.h" #include "api/crypto/frame_decryptor_interface.h" #include "api/field_trials_view.h" +#include "api/scoped_refptr.h" #include "modules/rtp_rtcp/source/frame_object.h" namespace webrtc { diff --git a/video/buffered_frame_decryptor_unittest.cc b/video/buffered_frame_decryptor_unittest.cc index 28e93fe022..781de7512c 100644 --- a/video/buffered_frame_decryptor_unittest.cc +++ b/video/buffered_frame_decryptor_unittest.cc @@ -10,14 +10,24 @@ #include "video/buffered_frame_decryptor.h" -#include +#include +#include #include #include #include +#include "api/crypto/frame_decryptor_interface.h" +#include "api/make_ref_counted.h" +#include "api/rtp_packet_infos.h" +#include "api/scoped_refptr.h" #include "api/test/mock_frame_decryptor.h" -#include "modules/video_coding/packet_buffer.h" -#include "system_wrappers/include/clock.h" +#include "api/video/encoded_image.h" +#include "api/video/video_codec_type.h" +#include "api/video/video_content_type.h" +#include "api/video/video_rotation.h" +#include "api/video/video_timing.h" +#include "modules/rtp_rtcp/source/frame_object.h" +#include "modules/rtp_rtcp/source/rtp_video_header.h" #include "test/gmock.h" #include "test/gtest.h" #include "test/scoped_key_value_config.h" diff --git a/video/call_stats2.cc b/video/call_stats2.cc index ef575d2667..8dd9ce98dd 100644 --- a/video/call_stats2.cc +++ b/video/call_stats2.cc @@ -11,11 +11,19 @@ #include "video/call_stats2.h" #include -#include +#include +#include #include #include "absl/algorithm/container.h" +#include "api/sequence_checker.h" +#include "api/task_queue/pending_task_safety_flag.h" +#include "api/task_queue/task_queue_base.h" +#include "api/units/time_delta.h" +#include "modules/include/module_common_types.h" #include "rtc_base/checks.h" +#include "rtc_base/task_utils/repeating_task.h" +#include "system_wrappers/include/clock.h" #include "system_wrappers/include/metrics.h" namespace webrtc { diff --git a/video/call_stats2.h b/video/call_stats2.h index 7e941d1e75..110f60067f 100644 --- a/video/call_stats2.h +++ b/video/call_stats2.h @@ -11,15 +11,17 @@ #ifndef VIDEO_CALL_STATS2_H_ #define VIDEO_CALL_STATS2_H_ +#include #include -#include #include "api/task_queue/pending_task_safety_flag.h" #include "api/task_queue/task_queue_base.h" -#include "api/units/timestamp.h" +#include "api/units/time_delta.h" #include "modules/include/module_common_types.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "rtc_base/checks.h" #include "rtc_base/task_utils/repeating_task.h" +#include "rtc_base/thread_annotations.h" #include "system_wrappers/include/clock.h" namespace webrtc { diff --git a/video/call_stats2_unittest.cc b/video/call_stats2_unittest.cc index af19f3b883..dd49adf68d 100644 --- a/video/call_stats2_unittest.cc +++ b/video/call_stats2_unittest.cc @@ -10,12 +10,15 @@ #include "video/call_stats2.h" +#include #include #include "api/task_queue/default_task_queue_factory.h" #include "api/task_queue/task_queue_base.h" +#include "api/task_queue/task_queue_factory.h" +#include "modules/include/module_common_types.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" -#include "rtc_base/thread.h" +#include "system_wrappers/include/clock.h" #include "system_wrappers/include/metrics.h" #include "test/gmock.h" #include "test/gtest.h" diff --git a/video/config/BUILD.gn b/video/config/BUILD.gn index 5dcc6316f4..d48bb4931c 100644 --- a/video/config/BUILD.gn +++ b/video/config/BUILD.gn @@ -32,6 +32,7 @@ rtc_library("streams_config") { "../../modules/video_coding:webrtc_vp9_helpers", "../../rtc_base:checks", "../../rtc_base:logging", + "../../rtc_base:safe_conversions", "../../rtc_base/experiments:field_trial_parser", "../../rtc_base/experiments:min_video_bitrate_experiment", "../../rtc_base/experiments:normalize_simulcast_size_experiment", @@ -49,8 +50,10 @@ rtc_library("encoder_config") { deps = [ "../../api:field_trials_view", + "../../api:ref_count", "../../api:scoped_refptr", "../../api/video:resolution", + "../../api/video:video_frame", "../../api/video_codecs:scalability_mode", "../../api/video_codecs:video_codecs_api", "../../rtc_base:checks", diff --git a/video/config/encoder_stream_factory.cc b/video/config/encoder_stream_factory.cc index 558427478a..601c750d36 100644 --- a/video/config/encoder_stream_factory.cc +++ b/video/config/encoder_stream_factory.cc @@ -10,23 +10,34 @@ #include "video/config/encoder_stream_factory.h" #include +#include +#include #include -#include -#include -#include +#include +#include +#include #include "absl/algorithm/container.h" -#include "absl/strings/match.h" #include "api/field_trials_view.h" -#include "api/video/video_codec_constants.h" +#include "api/units/data_rate.h" +#include "api/video/resolution.h" +#include "api/video/video_codec_type.h" +#include "api/video/video_source_interface.h" +#include "api/video_codecs/spatial_layer.h" +#include "api/video_codecs/video_codec.h" +#include "api/video_codecs/video_encoder.h" +#include "call/adaptation/video_source_restrictions.h" #include "media/base/media_constants.h" #include "media/base/video_adapter.h" #include "modules/video_coding/codecs/vp9/svc_config.h" #include "modules/video_coding/utility/simulcast_utility.h" +#include "rtc_base/checks.h" #include "rtc_base/experiments/min_video_bitrate_experiment.h" #include "rtc_base/experiments/normalize_simulcast_size_experiment.h" #include "rtc_base/logging.h" +#include "rtc_base/numerics/safe_conversions.h" #include "video/config/simulcast.h" +#include "video/config/video_encoder_config.h" namespace webrtc { namespace { diff --git a/video/config/encoder_stream_factory.h b/video/config/encoder_stream_factory.h index 192aac6113..abf196bdae 100644 --- a/video/config/encoder_stream_factory.h +++ b/video/config/encoder_stream_factory.h @@ -10,11 +10,12 @@ #ifndef VIDEO_CONFIG_ENCODER_STREAM_FACTORY_H_ #define VIDEO_CONFIG_ENCODER_STREAM_FACTORY_H_ -#include +#include #include #include "api/field_trials_view.h" #include "api/units/data_rate.h" +#include "api/video/resolution.h" #include "api/video_codecs/video_encoder.h" #include "call/adaptation/video_source_restrictions.h" #include "video/config/video_encoder_config.h" diff --git a/video/config/simulcast.h b/video/config/simulcast.h index 1a7f87c768..a8b9143ae7 100644 --- a/video/config/simulcast.h +++ b/video/config/simulcast.h @@ -19,6 +19,7 @@ #include "api/field_trials_view.h" #include "api/units/data_rate.h" #include "api/video/resolution.h" +#include "api/video/video_codec_type.h" #include "video/config/video_encoder_config.h" namespace webrtc { diff --git a/video/config/video_encoder_config.cc b/video/config/video_encoder_config.cc index ca3b20e436..c354014d17 100644 --- a/video/config/video_encoder_config.cc +++ b/video/config/video_encoder_config.cc @@ -9,8 +9,12 @@ */ #include "video/config/video_encoder_config.h" +#include +#include #include +#include "api/video/video_codec_type.h" +#include "api/video_codecs/video_codec.h" #include "rtc_base/checks.h" #include "rtc_base/strings/string_builder.h" diff --git a/video/config/video_encoder_config.h b/video/config/video_encoder_config.h index adb78c7706..7fda03ffa8 100644 --- a/video/config/video_encoder_config.h +++ b/video/config/video_encoder_config.h @@ -18,12 +18,14 @@ #include #include "api/field_trials_view.h" +#include "api/ref_count.h" #include "api/scoped_refptr.h" #include "api/video/resolution.h" +#include "api/video/video_codec_type.h" #include "api/video_codecs/scalability_mode.h" #include "api/video_codecs/sdp_video_format.h" +#include "api/video_codecs/spatial_layer.h" #include "api/video_codecs/video_codec.h" -#include "rtc_base/ref_count.h" namespace webrtc { diff --git a/video/corruption_detection/BUILD.gn b/video/corruption_detection/BUILD.gn index f2beeadf21..eb24a228c7 100644 --- a/video/corruption_detection/BUILD.gn +++ b/video/corruption_detection/BUILD.gn @@ -74,6 +74,7 @@ rtc_library("frame_pair_corruption_score") { ":halton_frame_sampler", ":utils", "../../api:scoped_refptr", + "../../api/video:corruption_detection_filter_settings", "../../api/video:video_frame", "../../rtc_base:checks", "//third_party/abseil-cpp/absl/strings:string_view", @@ -124,6 +125,7 @@ rtc_library("utils") { deps = [ "../../api:scoped_refptr", "../../api/video:video_frame", + "../../rtc_base:checks", "//third_party/abseil-cpp/absl/strings:string_view", "//third_party/abseil-cpp/absl/strings:strings", ] diff --git a/video/corruption_detection/frame_pair_corruption_score.cc b/video/corruption_detection/frame_pair_corruption_score.cc index 91b789cb5e..a6d21c23e9 100644 --- a/video/corruption_detection/frame_pair_corruption_score.cc +++ b/video/corruption_detection/frame_pair_corruption_score.cc @@ -15,6 +15,7 @@ #include "absl/strings/string_view.h" #include "api/scoped_refptr.h" +#include "api/video/corruption_detection_filter_settings.h" #include "api/video/i420_buffer.h" #include "api/video/video_frame_buffer.h" #include "rtc_base/checks.h" diff --git a/video/corruption_detection/frame_pair_corruption_score_unittest.cc b/video/corruption_detection/frame_pair_corruption_score_unittest.cc index cf6de29bda..55ac374d09 100644 --- a/video/corruption_detection/frame_pair_corruption_score_unittest.cc +++ b/video/corruption_detection/frame_pair_corruption_score_unittest.cc @@ -18,7 +18,6 @@ #include "api/scoped_refptr.h" #include "api/video/i420_buffer.h" #include "api/video/video_frame_buffer.h" -#include "test/gmock.h" #include "test/gtest.h" #include "test/testsupport/file_utils.h" #include "test/testsupport/frame_reader.h" diff --git a/video/corruption_detection/generic_mapping_functions.cc b/video/corruption_detection/generic_mapping_functions.cc index 9c75a50281..7dcbfedfef 100644 --- a/video/corruption_detection/generic_mapping_functions.cc +++ b/video/corruption_detection/generic_mapping_functions.cc @@ -12,6 +12,7 @@ #include +#include "api/video/corruption_detection_filter_settings.h" #include "api/video/video_codec_type.h" #include "api/video_codecs/video_codec.h" #include "rtc_base/checks.h" diff --git a/video/corruption_detection/utils.cc b/video/corruption_detection/utils.cc index b213dfb509..b213349c39 100644 --- a/video/corruption_detection/utils.cc +++ b/video/corruption_detection/utils.cc @@ -12,9 +12,11 @@ #include "absl/strings/match.h" #include "absl/strings/string_view.h" +#include "api/scoped_refptr.h" #include "api/video/i420_buffer.h" #include "api/video/video_codec_type.h" #include "api/video/video_frame_buffer.h" +#include "rtc_base/checks.h" namespace webrtc { namespace { diff --git a/video/decode_synchronizer.cc b/video/decode_synchronizer.cc index 94c4902c78..db012822fc 100644 --- a/video/decode_synchronizer.cc +++ b/video/decode_synchronizer.cc @@ -10,17 +10,21 @@ #include "video/decode_synchronizer.h" -#include +#include #include +#include #include -#include +#include "api/metronome/metronome.h" #include "api/sequence_checker.h" +#include "api/task_queue/pending_task_safety_flag.h" +#include "api/task_queue/task_queue_base.h" #include "api/units/time_delta.h" #include "api/units/timestamp.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/trace_event.h" +#include "system_wrappers/include/clock.h" #include "video/frame_decode_scheduler.h" #include "video/frame_decode_timing.h" diff --git a/video/decode_synchronizer.h b/video/decode_synchronizer.h index d441fa0543..821ebfc2b6 100644 --- a/video/decode_synchronizer.h +++ b/video/decode_synchronizer.h @@ -13,18 +13,16 @@ #include -#include #include #include #include -#include #include "api/metronome/metronome.h" -#include "api/sequence_checker.h" +#include "api/task_queue/pending_task_safety_flag.h" #include "api/task_queue/task_queue_base.h" #include "api/units/timestamp.h" -#include "rtc_base/checks.h" #include "rtc_base/thread_annotations.h" +#include "system_wrappers/include/clock.h" #include "video/frame_decode_scheduler.h" #include "video/frame_decode_timing.h" diff --git a/video/decode_synchronizer_unittest.cc b/video/decode_synchronizer_unittest.cc index 78adc77c7f..a725f51aa9 100644 --- a/video/decode_synchronizer_unittest.cc +++ b/video/decode_synchronizer_unittest.cc @@ -12,12 +12,16 @@ #include +#include #include #include #include "absl/functional/any_invocable.h" +#include "api/metronome/metronome.h" #include "api/metronome/test/fake_metronome.h" #include "api/units/time_delta.h" +#include "api/units/timestamp.h" +#include "system_wrappers/include/clock.h" #include "test/gmock.h" #include "test/gtest.h" #include "test/time_controller/simulated_time_controller.h" diff --git a/video/encoder_bitrate_adjuster.cc b/video/encoder_bitrate_adjuster.cc index 8176e65822..babef538e1 100644 --- a/video/encoder_bitrate_adjuster.cc +++ b/video/encoder_bitrate_adjuster.cc @@ -11,14 +11,30 @@ #include "video/encoder_bitrate_adjuster.h" #include +#include +#include #include +#include #include #include "api/field_trials_view.h" +#include "api/units/data_rate.h" +#include "api/units/data_size.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" +#include "api/video/video_bitrate_allocation.h" +#include "api/video/video_codec_constants.h" +#include "api/video/video_codec_type.h" +#include "api/video_codecs/video_codec.h" +#include "api/video_codecs/video_encoder.h" #include "modules/video_coding/svc/scalability_mode_util.h" +#include "rtc_base/checks.h" #include "rtc_base/experiments/rate_control_settings.h" #include "rtc_base/logging.h" #include "rtc_base/time_utils.h" +#include "system_wrappers/include/clock.h" +#include "video/encoder_overshoot_detector.h" +#include "video/rate_utilization_tracker.h" namespace webrtc { namespace { diff --git a/video/encoder_bitrate_adjuster.h b/video/encoder_bitrate_adjuster.h index 9248b0d5eb..3f43047f9c 100644 --- a/video/encoder_bitrate_adjuster.h +++ b/video/encoder_bitrate_adjuster.h @@ -11,12 +11,18 @@ #ifndef VIDEO_ENCODER_BITRATE_ADJUSTER_H_ #define VIDEO_ENCODER_BITRATE_ADJUSTER_H_ +#include +#include #include +#include "absl/container/inlined_vector.h" #include "api/field_trials_view.h" +#include "api/units/data_size.h" #include "api/units/time_delta.h" -#include "api/video/encoded_image.h" #include "api/video/video_bitrate_allocation.h" +#include "api/video/video_codec_constants.h" +#include "api/video/video_codec_type.h" +#include "api/video_codecs/video_codec.h" #include "api/video_codecs/video_encoder.h" #include "system_wrappers/include/clock.h" #include "video/encoder_overshoot_detector.h" diff --git a/video/encoder_bitrate_adjuster_unittest.cc b/video/encoder_bitrate_adjuster_unittest.cc index 8d3195b892..642664a253 100644 --- a/video/encoder_bitrate_adjuster_unittest.cc +++ b/video/encoder_bitrate_adjuster_unittest.cc @@ -10,14 +10,25 @@ #include "video/encoder_bitrate_adjuster.h" +#include +#include +#include #include #include #include -#include "api/field_trials_view.h" #include "api/units/data_rate.h" -#include "rtc_base/logging.h" -#include "rtc_base/numerics/safe_conversions.h" +#include "api/units/data_size.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" +#include "api/video/video_bitrate_allocation.h" +#include "api/video/video_codec_constants.h" +#include "api/video/video_codec_type.h" +#include "api/video_codecs/scalability_mode.h" +#include "api/video_codecs/video_codec.h" +#include "api/video_codecs/video_encoder.h" +#include "rtc_base/checks.h" +#include "rtc_base/time_utils.h" #include "test/gtest.h" #include "test/scoped_key_value_config.h" #include "test/time_controller/simulated_time_controller.h" diff --git a/video/encoder_overshoot_detector.cc b/video/encoder_overshoot_detector.cc index b9697f57de..b31b6be9ee 100644 --- a/video/encoder_overshoot_detector.cc +++ b/video/encoder_overshoot_detector.cc @@ -11,8 +11,14 @@ #include "video/encoder_overshoot_detector.h" #include +#include +#include +#include +#include #include +#include "api/units/data_rate.h" +#include "api/video/video_codec_type.h" #include "system_wrappers/include/metrics.h" namespace webrtc { diff --git a/video/encoder_overshoot_detector.h b/video/encoder_overshoot_detector.h index e78b601d1d..a36a45c144 100644 --- a/video/encoder_overshoot_detector.h +++ b/video/encoder_overshoot_detector.h @@ -11,11 +11,13 @@ #ifndef VIDEO_ENCODER_OVERSHOOT_DETECTOR_H_ #define VIDEO_ENCODER_OVERSHOOT_DETECTOR_H_ +#include +#include #include #include #include "api/units/data_rate.h" -#include "api/video_codecs/video_codec.h" +#include "api/video/video_codec_type.h" namespace webrtc { diff --git a/video/encoder_overshoot_detector_unittest.cc b/video/encoder_overshoot_detector_unittest.cc index 0712194a25..2dc8e4f24e 100644 --- a/video/encoder_overshoot_detector_unittest.cc +++ b/video/encoder_overshoot_detector_unittest.cc @@ -10,9 +10,15 @@ #include "video/encoder_overshoot_detector.h" +#include +#include #include #include "api/units/data_rate.h" +#include "api/units/data_size.h" +#include "api/units/frequency.h" +#include "api/units/time_delta.h" +#include "api/video/video_codec_type.h" #include "rtc_base/fake_clock.h" #include "rtc_base/time_utils.h" #include "system_wrappers/include/metrics.h" diff --git a/video/encoder_rtcp_feedback.cc b/video/encoder_rtcp_feedback.cc index 42827e2df9..be8ee7a975 100644 --- a/video/encoder_rtcp_feedback.cc +++ b/video/encoder_rtcp_feedback.cc @@ -11,13 +11,24 @@ #include "video/encoder_rtcp_feedback.h" #include +#include +#include +#include +#include #include #include +#include #include "api/environment/environment.h" +#include "api/sequence_checker.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" +#include "api/video/video_frame_type.h" #include "api/video_codecs/video_encoder.h" +#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h" #include "rtc_base/checks.h" #include "rtc_base/experiments/keyframe_interval_settings.h" +#include "rtc_base/logging.h" #include "system_wrappers/include/clock.h" namespace webrtc { diff --git a/video/encoder_rtcp_feedback.h b/video/encoder_rtcp_feedback.h index 29832dbd6d..6bea76f78c 100644 --- a/video/encoder_rtcp_feedback.h +++ b/video/encoder_rtcp_feedback.h @@ -10,6 +10,7 @@ #ifndef VIDEO_ENCODER_RTCP_FEEDBACK_H_ #define VIDEO_ENCODER_RTCP_FEEDBACK_H_ +#include #include #include @@ -17,9 +18,10 @@ #include "api/sequence_checker.h" #include "api/units/time_delta.h" #include "api/units/timestamp.h" -#include "call/rtp_video_sender_interface.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h" #include "rtc_base/system/no_unique_address.h" +#include "rtc_base/thread_annotations.h" #include "video/video_stream_encoder_interface.h" namespace webrtc { diff --git a/video/encoder_rtcp_feedback_unittest.cc b/video/encoder_rtcp_feedback_unittest.cc index 62140e81a8..c2ff40265b 100644 --- a/video/encoder_rtcp_feedback_unittest.cc +++ b/video/encoder_rtcp_feedback_unittest.cc @@ -10,9 +10,12 @@ #include "video/encoder_rtcp_feedback.h" -#include +#include +#include #include "api/environment/environment_factory.h" +#include "api/video/video_frame_type.h" +#include "system_wrappers/include/clock.h" #include "test/gmock.h" #include "test/gtest.h" #include "video/test/mock_video_stream_encoder.h" diff --git a/video/end_to_end_tests/call_operation_tests.cc b/video/end_to_end_tests/call_operation_tests.cc index 5b6662bb82..610b871b0d 100644 --- a/video/end_to_end_tests/call_operation_tests.cc +++ b/video/end_to_end_tests/call_operation_tests.cc @@ -9,18 +9,21 @@ */ #include +#include +#include "api/rtp_parameters.h" #include "api/test/create_frame_generator.h" #include "api/test/frame_generator_interface.h" #include "api/test/simulated_network.h" -#include "call/fake_network_pipe.h" +#include "api/video/video_frame.h" +#include "api/video/video_sink_interface.h" +#include "rtc_base/event.h" #include "rtc_base/task_queue_for_test.h" #include "system_wrappers/include/sleep.h" #include "test/call_test.h" -#include "test/field_trial.h" +#include "test/direct_transport.h" #include "test/frame_forwarder.h" #include "test/gtest.h" -#include "test/network/simulated_network.h" #include "test/null_transport.h" #include "test/video_test_constants.h" diff --git a/video/end_to_end_tests/codec_tests.cc b/video/end_to_end_tests/codec_tests.cc index 446b2c784e..e1d440e06c 100644 --- a/video/end_to_end_tests/codec_tests.cc +++ b/video/end_to_end_tests/codec_tests.cc @@ -8,26 +8,35 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include #include +#include +#include #include "api/environment/environment.h" +#include "api/rtp_parameters.h" +#include "api/test/video/function_video_decoder_factory.h" #include "api/test/video/function_video_encoder_factory.h" #include "api/video/color_space.h" +#include "api/video/video_frame.h" #include "api/video/video_rotation.h" +#include "api/video/video_sink_interface.h" +#include "api/video_codecs/sdp_video_format.h" +#include "api/video_codecs/video_codec.h" +#include "api/video_codecs/video_decoder_factory.h" +#include "api/video_codecs/video_encoder_factory.h" +#include "call/video_receive_stream.h" +#include "call/video_send_stream.h" #include "common_video/test/utilities.h" -#include "media/base/codec.h" #include "media/base/media_constants.h" -#include "media/engine/internal_decoder_factory.h" -#include "media/engine/internal_encoder_factory.h" #include "modules/video_coding/codecs/h264/include/h264.h" #include "modules/video_coding/codecs/vp8/include/vp8.h" #include "modules/video_coding/codecs/vp9/include/vp9.h" #include "test/call_test.h" -#include "test/encoder_settings.h" #include "test/field_trial.h" +#include "test/frame_generator_capturer.h" #include "test/gtest.h" #include "test/video_test_constants.h" +#include "video/config/video_encoder_config.h" namespace webrtc { namespace { diff --git a/video/end_to_end_tests/extended_reports_tests.cc b/video/end_to_end_tests/extended_reports_tests.cc index c595074fc6..2a37082f14 100644 --- a/video/end_to_end_tests/extended_reports_tests.cc +++ b/video/end_to_end_tests/extended_reports_tests.cc @@ -11,36 +11,30 @@ #include #include -#include #include #include -#include #include +#include "api/array_view.h" #include "api/rtp_headers.h" -#include "api/task_queue/task_queue_base.h" +#include "api/rtp_parameters.h" #include "api/test/simulated_network.h" #include "api/units/data_rate.h" #include "api/units/time_delta.h" +#include "api/video/video_codec_type.h" #include "api/video_codecs/sdp_video_format.h" -#include "call/call.h" -#include "call/fake_network_pipe.h" #include "call/rtp_config.h" -#include "call/simulated_packet_receiver.h" #include "call/video_receive_stream.h" #include "call/video_send_stream.h" -#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h" #include "modules/rtp_rtcp/source/rtcp_packet/target_bitrate.h" #include "rtc_base/event.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/thread_annotations.h" -#include "system_wrappers/include/clock.h" #include "test/call_test.h" -#include "test/field_trial.h" #include "test/gtest.h" -#include "test/network/simulated_network.h" #include "test/rtcp_packet_parser.h" #include "test/rtp_rtcp_observer.h" +#include "test/scoped_key_value_config.h" #include "test/video_test_constants.h" #include "video/config/video_encoder_config.h" diff --git a/video/end_to_end_tests/fec_tests.cc b/video/end_to_end_tests/fec_tests.cc index 98a9d2a85a..ebad6fe165 100644 --- a/video/end_to_end_tests/fec_tests.cc +++ b/video/end_to_end_tests/fec_tests.cc @@ -8,25 +8,45 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include +#include #include +#include +#include -#include "api/task_queue/task_queue_base.h" +#include "api/array_view.h" +#include "api/environment/environment.h" +#include "api/rtp_parameters.h" #include "api/test/simulated_network.h" #include "api/test/video/function_video_encoder_factory.h" -#include "call/fake_network_pipe.h" +#include "api/transport/bitrate_settings.h" +#include "api/video/video_codec_type.h" +#include "api/video/video_frame.h" +#include "api/video/video_rotation.h" +#include "api/video/video_sink_interface.h" +#include "api/video_codecs/sdp_video_format.h" +#include "api/video_codecs/video_encoder.h" +#include "call/flexfec_receive_stream.h" +#include "call/video_receive_stream.h" +#include "call/video_send_stream.h" #include "media/engine/internal_decoder_factory.h" #include "modules/include/module_common_types_public.h" #include "modules/rtp_rtcp/source/byte_io.h" +#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h" #include "modules/rtp_rtcp/source/rtp_packet.h" #include "modules/video_coding/codecs/vp8/include/vp8.h" +#include "rtc_base/random.h" #include "rtc_base/synchronization/mutex.h" +#include "rtc_base/thread_annotations.h" #include "test/call_test.h" -#include "test/field_trial.h" +#include "test/encoder_settings.h" +#include "test/frame_generator_capturer.h" #include "test/gmock.h" #include "test/gtest.h" -#include "test/network/simulated_network.h" #include "test/rtcp_packet_parser.h" +#include "test/rtp_rtcp_observer.h" #include "test/video_test_constants.h" +#include "video/config/video_encoder_config.h" using ::testing::Contains; using ::testing::Not; diff --git a/video/end_to_end_tests/frame_encryption_tests.cc b/video/end_to_end_tests/frame_encryption_tests.cc index 698d8c14c0..efdb1d8b3d 100644 --- a/video/end_to_end_tests/frame_encryption_tests.cc +++ b/video/end_to_end_tests/frame_encryption_tests.cc @@ -8,14 +8,29 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include +#include + +#include "api/environment/environment.h" +#include "api/make_ref_counted.h" +#include "api/rtp_parameters.h" #include "api/test/fake_frame_decryptor.h" #include "api/test/fake_frame_encryptor.h" +#include "api/test/video/function_video_encoder_factory.h" +#include "api/video/video_codec_type.h" +#include "api/video/video_frame.h" +#include "api/video/video_sink_interface.h" +#include "api/video_codecs/sdp_video_format.h" +#include "api/video_codecs/video_encoder.h" +#include "call/video_receive_stream.h" +#include "call/video_send_stream.h" #include "media/engine/internal_decoder_factory.h" -#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h" #include "modules/video_coding/codecs/vp8/include/vp8.h" #include "test/call_test.h" +#include "test/encoder_settings.h" #include "test/gtest.h" #include "test/video_test_constants.h" +#include "video/config/video_encoder_config.h" namespace webrtc { namespace { diff --git a/video/end_to_end_tests/histogram_tests.cc b/video/end_to_end_tests/histogram_tests.cc index 5f281e267f..65775f1a12 100644 --- a/video/end_to_end_tests/histogram_tests.cc +++ b/video/end_to_end_tests/histogram_tests.cc @@ -8,15 +8,31 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include #include +#include +#include +#include "api/array_view.h" +#include "api/environment/environment.h" +#include "api/rtp_parameters.h" #include "api/test/video/function_video_encoder_factory.h" +#include "api/video/video_codec_type.h" +#include "api/video/video_frame.h" +#include "api/video/video_sink_interface.h" +#include "api/video_codecs/sdp_video_format.h" +#include "call/video_receive_stream.h" +#include "call/video_send_stream.h" #include "modules/video_coding/codecs/vp8/include/vp8.h" #include "rtc_base/synchronization/mutex.h" +#include "rtc_base/thread_annotations.h" +#include "system_wrappers/include/clock.h" #include "system_wrappers/include/metrics.h" #include "test/call_test.h" #include "test/gtest.h" +#include "test/rtp_rtcp_observer.h" #include "test/video_test_constants.h" +#include "video/config/video_encoder_config.h" namespace webrtc { namespace { diff --git a/video/end_to_end_tests/multi_codec_receive_tests.cc b/video/end_to_end_tests/multi_codec_receive_tests.cc index c4e4479d3a..d958c7749d 100644 --- a/video/end_to_end_tests/multi_codec_receive_tests.cc +++ b/video/end_to_end_tests/multi_codec_receive_tests.cc @@ -8,22 +8,42 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include +#include #include +#include +#include +#include +#include +#include "api/array_view.h" +#include "api/environment/environment.h" #include "api/test/simulated_network.h" +#include "api/test/video/function_video_decoder_factory.h" #include "api/test/video/function_video_encoder_factory.h" -#include "call/fake_network_pipe.h" +#include "api/video/video_frame.h" +#include "api/video/video_sink_interface.h" +#include "api/video_codecs/sdp_video_format.h" +#include "api/video_codecs/video_codec.h" +#include "api/video_codecs/video_decoder.h" +#include "api/video_codecs/video_decoder_factory.h" +#include "api/video_codecs/video_encoder.h" +#include "api/video_codecs/video_encoder_factory.h" +#include "call/video_receive_stream.h" #include "modules/include/module_common_types_public.h" #include "modules/rtp_rtcp/source/rtp_packet.h" #include "modules/video_coding/codecs/h264/include/h264.h" #include "modules/video_coding/codecs/vp8/include/vp8.h" #include "modules/video_coding/codecs/vp9/include/vp9.h" +#include "rtc_base/checks.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/task_queue_for_test.h" +#include "rtc_base/thread_annotations.h" #include "test/call_test.h" +#include "test/encoder_settings.h" #include "test/gmock.h" #include "test/gtest.h" -#include "test/network/simulated_network.h" +#include "test/rtp_rtcp_observer.h" #include "test/video_test_constants.h" using ::testing::Contains; diff --git a/video/end_to_end_tests/multi_stream_tester.cc b/video/end_to_end_tests/multi_stream_tester.cc index 7d82085617..a9c423a3db 100644 --- a/video/end_to_end_tests/multi_stream_tester.cc +++ b/video/end_to_end_tests/multi_stream_tester.cc @@ -10,25 +10,39 @@ #include "video/end_to_end_tests/multi_stream_tester.h" +#include +#include #include +#include #include #include #include "api/environment/environment.h" #include "api/environment/environment_factory.h" +#include "api/rtp_parameters.h" #include "api/task_queue/task_queue_base.h" +#include "api/task_queue/task_queue_factory.h" #include "api/test/create_frame_generator.h" #include "api/test/simulated_network.h" #include "api/test/video/function_video_encoder_factory.h" #include "api/video/builtin_video_bitrate_allocator_factory.h" +#include "api/video/video_bitrate_allocator_factory.h" +#include "api/video/video_codec_type.h" +#include "api/video_codecs/sdp_video_format.h" +#include "call/call.h" +#include "call/call_config.h" #include "call/fake_network_pipe.h" +#include "call/video_receive_stream.h" +#include "call/video_send_stream.h" #include "media/engine/internal_decoder_factory.h" #include "modules/video_coding/codecs/vp8/include/vp8.h" #include "rtc_base/task_queue_for_test.h" -#include "test/call_test.h" +#include "test/direct_transport.h" #include "test/encoder_settings.h" +#include "test/frame_generator_capturer.h" #include "test/network/simulated_network.h" #include "test/video_test_constants.h" +#include "video/config/video_encoder_config.h" namespace webrtc { diff --git a/video/end_to_end_tests/multi_stream_tester.h b/video/end_to_end_tests/multi_stream_tester.h index 87200930f4..1a58247f78 100644 --- a/video/end_to_end_tests/multi_stream_tester.h +++ b/video/end_to_end_tests/multi_stream_tester.h @@ -11,13 +11,19 @@ #ifndef VIDEO_END_TO_END_TESTS_MULTI_STREAM_TESTER_H_ #define VIDEO_END_TO_END_TESTS_MULTI_STREAM_TESTER_H_ +#include +#include #include #include +#include "api/media_types.h" #include "api/task_queue/task_queue_base.h" #include "call/call.h" +#include "call/video_receive_stream.h" +#include "call/video_send_stream.h" #include "test/direct_transport.h" #include "test/frame_generator_capturer.h" +#include "video/config/video_encoder_config.h" namespace webrtc { // Test sets up a Call multiple senders with different resolutions and SSRCs. diff --git a/video/end_to_end_tests/multi_stream_tests.cc b/video/end_to_end_tests/multi_stream_tests.cc index 35f017758b..0a20b34b6f 100644 --- a/video/end_to_end_tests/multi_stream_tests.cc +++ b/video/end_to_end_tests/multi_stream_tests.cc @@ -14,6 +14,7 @@ #include #include +#include "api/units/time_delta.h" #include "api/video/video_frame.h" #include "api/video/video_sink_interface.h" #include "call/rtp_config.h" diff --git a/video/end_to_end_tests/resolution_bitrate_limits_tests.cc b/video/end_to_end_tests/resolution_bitrate_limits_tests.cc index 128406249d..a8503d9499 100644 --- a/video/end_to_end_tests/resolution_bitrate_limits_tests.cc +++ b/video/end_to_end_tests/resolution_bitrate_limits_tests.cc @@ -8,18 +8,34 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include - -#include "media/engine/webrtc_video_engine.h" +#include +#include +#include +#include +#include + +#include "api/environment/environment.h" +#include "api/make_ref_counted.h" +#include "api/units/data_rate.h" +#include "api/video/video_codec_type.h" +#include "api/video/video_frame.h" +#include "api/video/video_sink_interface.h" +#include "api/video/video_source_interface.h" +#include "api/video_codecs/scalability_mode.h" +#include "api/video_codecs/video_codec.h" +#include "api/video_codecs/video_encoder.h" +#include "call/video_receive_stream.h" +#include "call/video_send_stream.h" #include "modules/video_coding/svc/scalability_mode_util.h" #include "rtc_base/experiments/encoder_info_settings.h" #include "test/call_test.h" #include "test/fake_encoder.h" #include "test/field_trial.h" +#include "test/frame_generator_capturer.h" #include "test/gtest.h" #include "test/video_encoder_proxy_factory.h" #include "test/video_test_constants.h" -#include "video/config/encoder_stream_factory.h" +#include "video/config/video_encoder_config.h" namespace webrtc { namespace test { diff --git a/video/end_to_end_tests/ssrc_tests.cc b/video/end_to_end_tests/ssrc_tests.cc index f7542ef002..469ea7e068 100644 --- a/video/end_to_end_tests/ssrc_tests.cc +++ b/video/end_to_end_tests/ssrc_tests.cc @@ -8,20 +8,34 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include +#include +#include #include +#include +#include +#include "api/array_view.h" +#include "api/rtp_parameters.h" +#include "api/task_queue/task_queue_base.h" #include "api/test/simulated_network.h" #include "call/fake_network_pipe.h" #include "call/packet_receiver.h" +#include "call/video_receive_stream.h" +#include "call/video_send_stream.h" #include "modules/rtp_rtcp/source/rtp_packet.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" -#include "modules/rtp_rtcp/source/rtp_util.h" +#include "rtc_base/copy_on_write_buffer.h" +#include "rtc_base/event.h" #include "rtc_base/task_queue_for_test.h" #include "test/call_test.h" +#include "test/direct_transport.h" #include "test/gtest.h" #include "test/network/simulated_network.h" #include "test/rtcp_packet_parser.h" +#include "test/rtp_rtcp_observer.h" #include "test/video_test_constants.h" +#include "video/config/video_encoder_config.h" namespace webrtc { class SsrcEndToEndTest : public test::CallTest { diff --git a/video/end_to_end_tests/stats_tests.cc b/video/end_to_end_tests/stats_tests.cc index d06a98be71..8952e9f366 100644 --- a/video/end_to_end_tests/stats_tests.cc +++ b/video/end_to_end_tests/stats_tests.cc @@ -8,27 +8,54 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include +#include +#include #include #include +#include +#include +#include +#include #include "absl/algorithm/container.h" +#include "api/array_view.h" +#include "api/environment/environment.h" +#include "api/rtp_headers.h" +#include "api/rtp_parameters.h" +#include "api/scoped_refptr.h" +#include "api/task_queue/pending_task_safety_flag.h" #include "api/task_queue/task_queue_base.h" #include "api/test/simulated_network.h" #include "api/test/video/function_video_encoder_factory.h" +#include "api/transport/bitrate_settings.h" +#include "api/units/time_delta.h" +#include "api/video/video_frame.h" +#include "api/video/video_sink_interface.h" +#include "api/video_codecs/sdp_video_format.h" +#include "call/call.h" +#include "call/call_config.h" #include "call/fake_network_pipe.h" +#include "call/video_receive_stream.h" +#include "call/video_send_stream.h" #include "modules/rtp_rtcp/source/rtp_packet.h" #include "modules/video_coding/include/video_coding_defines.h" +#include "rtc_base/checks.h" +#include "rtc_base/event.h" #include "rtc_base/strings/string_builder.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/task_queue_for_test.h" +#include "rtc_base/thread_annotations.h" #include "system_wrappers/include/metrics.h" #include "system_wrappers/include/sleep.h" #include "test/call_test.h" +#include "test/fake_decoder.h" #include "test/fake_encoder.h" #include "test/gtest.h" -#include "test/network/simulated_network.h" #include "test/rtcp_packet_parser.h" +#include "test/rtp_rtcp_observer.h" #include "test/video_test_constants.h" +#include "video/config/video_encoder_config.h" namespace webrtc { namespace { diff --git a/video/end_to_end_tests/transport_feedback_tests.cc b/video/end_to_end_tests/transport_feedback_tests.cc index 4f285f2b29..84f2d56d23 100644 --- a/video/end_to_end_tests/transport_feedback_tests.cc +++ b/video/end_to_end_tests/transport_feedback_tests.cc @@ -8,25 +8,46 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include +#include +#include #include +#include #include +#include "api/array_view.h" +#include "api/call/transport.h" #include "api/rtp_parameters.h" #include "api/task_queue/task_queue_base.h" +#include "api/test/simulated_network.h" +#include "api/transport/bitrate_settings.h" #include "api/units/time_delta.h" +#include "call/audio_receive_stream.h" +#include "call/audio_send_stream.h" #include "call/call.h" #include "call/fake_network_pipe.h" +#include "call/video_receive_stream.h" +#include "call/video_send_stream.h" +#include "modules/rtp_rtcp/include/rtp_header_extension_map.h" #include "modules/rtp_rtcp/source/byte_io.h" #include "modules/rtp_rtcp/source/rtp_header_extensions.h" #include "modules/rtp_rtcp/source/rtp_packet.h" +#include "rtc_base/checks.h" +#include "rtc_base/event.h" #include "rtc_base/numerics/sequence_number_unwrapper.h" #include "rtc_base/synchronization/mutex.h" +#include "rtc_base/thread_annotations.h" #include "test/call_test.h" +#include "test/direct_transport.h" +#include "test/fake_videorenderer.h" #include "test/field_trial.h" +#include "test/frame_generator_capturer.h" #include "test/gtest.h" #include "test/network/simulated_network.h" #include "test/rtcp_packet_parser.h" +#include "test/rtp_rtcp_observer.h" #include "test/video_test_constants.h" +#include "video/config/video_encoder_config.h" #include "video/end_to_end_tests/multi_stream_tester.h" namespace webrtc { diff --git a/video/frame_cadence_adapter.cc b/video/frame_cadence_adapter.cc index 2ee785fd53..7e151f6d22 100644 --- a/video/frame_cadence_adapter.cc +++ b/video/frame_cadence_adapter.cc @@ -12,35 +12,37 @@ #include #include +#include #include #include #include +#include #include #include #include "absl/algorithm/container.h" -#include "absl/base/attributes.h" #include "absl/cleanup/cleanup.h" +#include "api/field_trials_view.h" +#include "api/metronome/metronome.h" +#include "api/scoped_refptr.h" #include "api/sequence_checker.h" #include "api/task_queue/pending_task_safety_flag.h" #include "api/task_queue/task_queue_base.h" #include "api/units/time_delta.h" #include "api/units/timestamp.h" #include "api/video/video_frame.h" +#include "api/video_track_source_constraints.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/race_checker.h" #include "rtc_base/rate_statistics.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/system/no_unique_address.h" -#include "rtc_base/system/unused.h" #include "rtc_base/task_utils/repeating_task.h" #include "rtc_base/thread_annotations.h" -#include "rtc_base/time_utils.h" #include "rtc_base/trace_event.h" #include "system_wrappers/include/clock.h" #include "system_wrappers/include/metrics.h" -#include "system_wrappers/include/ntp_time.h" namespace webrtc { namespace { diff --git a/video/frame_cadence_adapter.h b/video/frame_cadence_adapter.h index 2f97b5b49f..2d457a5f93 100644 --- a/video/frame_cadence_adapter.h +++ b/video/frame_cadence_adapter.h @@ -11,17 +11,18 @@ #ifndef VIDEO_FRAME_CADENCE_ADAPTER_H_ #define VIDEO_FRAME_CADENCE_ADAPTER_H_ +#include +#include #include +#include -#include "absl/base/attributes.h" #include "api/field_trials_view.h" #include "api/metronome/metronome.h" #include "api/task_queue/task_queue_base.h" #include "api/units/time_delta.h" +#include "api/units/timestamp.h" #include "api/video/video_frame.h" #include "api/video/video_sink_interface.h" -#include "rtc_base/synchronization/mutex.h" -#include "rtc_base/thread_annotations.h" #include "system_wrappers/include/clock.h" namespace webrtc { diff --git a/video/frame_cadence_adapter_unittest.cc b/video/frame_cadence_adapter_unittest.cc index fb1eefeb37..b40171eaca 100644 --- a/video/frame_cadence_adapter_unittest.cc +++ b/video/frame_cadence_adapter_unittest.cc @@ -10,12 +10,16 @@ #include "video/frame_cadence_adapter.h" +#include #include +#include #include +#include #include -#include #include "absl/functional/any_invocable.h" +#include "api/field_trials_view.h" +#include "api/make_ref_counted.h" #include "api/metronome/test/fake_metronome.h" #include "api/task_queue/default_task_queue_factory.h" #include "api/task_queue/task_queue_base.h" @@ -24,11 +28,13 @@ #include "api/units/timestamp.h" #include "api/video/nv12_buffer.h" #include "api/video/video_frame.h" +#include "api/video_track_source_constraints.h" #include "rtc_base/event.h" #include "rtc_base/logging.h" #include "rtc_base/rate_statistics.h" #include "rtc_base/task_queue_for_test.h" #include "rtc_base/time_utils.h" +#include "system_wrappers/include/clock.h" #include "system_wrappers/include/metrics.h" #include "system_wrappers/include/ntp_time.h" #include "system_wrappers/include/sleep.h" diff --git a/video/frame_decode_timing.cc b/video/frame_decode_timing.cc index e7401a0ca4..04b8e22c5f 100644 --- a/video/frame_decode_timing.cc +++ b/video/frame_decode_timing.cc @@ -10,11 +10,15 @@ #include "video/frame_decode_timing.h" -#include +#include #include #include "api/units/time_delta.h" +#include "api/units/timestamp.h" +#include "modules/video_coding/timing/timing.h" +#include "rtc_base/checks.h" #include "rtc_base/logging.h" +#include "system_wrappers/include/clock.h" namespace webrtc { diff --git a/video/frame_decode_timing.h b/video/frame_decode_timing.h index c5b024f7f2..1679bf524f 100644 --- a/video/frame_decode_timing.h +++ b/video/frame_decode_timing.h @@ -13,10 +13,10 @@ #include -#include +#include -#include "api/task_queue/pending_task_safety_flag.h" -#include "api/task_queue/task_queue_base.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" #include "modules/video_coding/timing/timing.h" #include "system_wrappers/include/clock.h" diff --git a/video/frame_decode_timing_unittest.cc b/video/frame_decode_timing_unittest.cc index cb45d709f1..e4359b5c2a 100644 --- a/video/frame_decode_timing_unittest.cc +++ b/video/frame_decode_timing_unittest.cc @@ -14,9 +14,13 @@ #include +#include "api/field_trials_view.h" #include "api/units/time_delta.h" +#include "api/units/timestamp.h" #include "modules/video_coding/timing/timing.h" +#include "rtc_base/checks.h" #include "rtc_base/containers/flat_map.h" +#include "system_wrappers/include/clock.h" #include "test/gmock.h" #include "test/gtest.h" #include "test/scoped_key_value_config.h" diff --git a/video/frame_dumping_decoder.cc b/video/frame_dumping_decoder.cc index cfb20988d9..a1a4e88426 100644 --- a/video/frame_dumping_decoder.cc +++ b/video/frame_dumping_decoder.cc @@ -10,11 +10,15 @@ #include "video/frame_dumping_decoder.h" +#include #include #include -#include "modules/video_coding/include/video_codec_interface.h" +#include "api/video/encoded_image.h" +#include "api/video/video_codec_type.h" +#include "api/video_codecs/video_decoder.h" #include "modules/video_coding/utility/ivf_file_writer.h" +#include "rtc_base/system/file_wrapper.h" namespace webrtc { namespace { diff --git a/video/frame_dumping_encoder.cc b/video/frame_dumping_encoder.cc index 3785293ee4..dcd5ed6b69 100644 --- a/video/frame_dumping_encoder.cc +++ b/video/frame_dumping_encoder.cc @@ -10,17 +10,26 @@ #include "video/frame_dumping_encoder.h" +#include #include +#include #include #include #include #include "absl/algorithm/container.h" -#include "api/sequence_checker.h" -#include "api/video/video_codec_type.h" +#include "api/fec_controller_override.h" +#include "api/field_trials_view.h" +#include "api/video/encoded_image.h" +#include "api/video/video_frame.h" +#include "api/video/video_frame_type.h" +#include "api/video_codecs/video_codec.h" +#include "api/video_codecs/video_encoder.h" #include "modules/video_coding/utility/ivf_file_writer.h" #include "rtc_base/strings/string_builder.h" +#include "rtc_base/synchronization/mutex.h" #include "rtc_base/system/file_wrapper.h" +#include "rtc_base/thread_annotations.h" #include "rtc_base/time_utils.h" namespace webrtc { diff --git a/video/frame_encode_metadata_writer.cc b/video/frame_encode_metadata_writer.cc index 482ba201de..dbc303ce0b 100644 --- a/video/frame_encode_metadata_writer.cc +++ b/video/frame_encode_metadata_writer.cc @@ -11,14 +11,32 @@ #include "video/frame_encode_metadata_writer.h" #include +#include +#include #include +#include #include +#include "api/make_ref_counted.h" +#include "api/video/encoded_image.h" +#include "api/video/video_bitrate_allocation.h" +#include "api/video/video_codec_type.h" +#include "api/video/video_content_type.h" +#include "api/video/video_frame.h" +#include "api/video/video_frame_type.h" +#include "api/video/video_timing.h" +#include "api/video_codecs/video_codec.h" +#include "api/video_codecs/video_encoder.h" #include "common_video/h264/sps_vui_rewriter.h" #include "modules/include/module_common_types_public.h" +#include "modules/video_coding/include/video_codec_interface.h" #include "modules/video_coding/include/video_coding_defines.h" #include "modules/video_coding/svc/create_scalability_structure.h" +#include "modules/video_coding/svc/scalable_video_controller.h" +#include "rtc_base/buffer.h" +#include "rtc_base/checks.h" #include "rtc_base/logging.h" +#include "rtc_base/synchronization/mutex.h" #include "rtc_base/time_utils.h" namespace webrtc { diff --git a/video/frame_encode_metadata_writer.h b/video/frame_encode_metadata_writer.h index 637d03a40e..1f91568593 100644 --- a/video/frame_encode_metadata_writer.h +++ b/video/frame_encode_metadata_writer.h @@ -11,15 +11,23 @@ #ifndef VIDEO_FRAME_ENCODE_METADATA_WRITER_H_ #define VIDEO_FRAME_ENCODE_METADATA_WRITER_H_ +#include +#include #include #include #include +#include "api/rtp_packet_infos.h" +#include "api/video/color_space.h" #include "api/video/encoded_image.h" +#include "api/video/video_bitrate_allocation.h" +#include "api/video/video_frame.h" +#include "api/video/video_rotation.h" #include "api/video_codecs/video_codec.h" #include "api/video_codecs/video_encoder.h" #include "modules/video_coding/include/video_codec_interface.h" #include "rtc_base/synchronization/mutex.h" +#include "rtc_base/thread_annotations.h" namespace webrtc { diff --git a/video/frame_encode_metadata_writer_unittest.cc b/video/frame_encode_metadata_writer_unittest.cc index 753c3092e0..bb584b8dbd 100644 --- a/video/frame_encode_metadata_writer_unittest.cc +++ b/video/frame_encode_metadata_writer_unittest.cc @@ -11,15 +11,27 @@ #include "video/frame_encode_metadata_writer.h" #include +#include #include +#include "api/rtp_packet_infos.h" +#include "api/scoped_refptr.h" +#include "api/video/color_space.h" +#include "api/video/encoded_image.h" #include "api/video/i420_buffer.h" +#include "api/video/video_bitrate_allocation.h" +#include "api/video/video_codec_type.h" +#include "api/video/video_content_type.h" #include "api/video/video_frame.h" +#include "api/video/video_frame_type.h" +#include "api/video/video_rotation.h" #include "api/video/video_timing.h" +#include "api/video_codecs/video_codec.h" +#include "api/video_codecs/video_encoder.h" #include "common_video/h264/h264_common.h" #include "common_video/test/utilities.h" +#include "modules/video_coding/include/video_codec_interface.h" #include "modules/video_coding/include/video_coding_defines.h" -#include "rtc_base/time_utils.h" #include "test/gmock.h" #include "test/gtest.h" diff --git a/video/full_stack_tests.cc b/video/full_stack_tests.cc index 123ca10f0e..8245e6ff00 100644 --- a/video/full_stack_tests.cc +++ b/video/full_stack_tests.cc @@ -14,12 +14,13 @@ #include #include "absl/flags/flag.h" -#include "absl/flags/parse.h" +#include "api/rtp_parameters.h" #include "api/test/simulated_network.h" #include "api/test/test_dependency_factory.h" #include "api/test/video_quality_test_fixture.h" #include "api/units/data_rate.h" #include "api/video_codecs/sdp_video_format.h" +#include "api/video_codecs/spatial_layer.h" #include "api/video_codecs/video_codec.h" #include "api/video_codecs/vp9_profile.h" #include "modules/video_coding/codecs/vp9/include/vp9.h" diff --git a/video/pc_full_stack_tests.cc b/video/pc_full_stack_tests.cc index bde1a751f8..841f961ff0 100644 --- a/video/pc_full_stack_tests.cc +++ b/video/pc_full_stack_tests.cc @@ -12,6 +12,7 @@ #include #include +#include "api/function_view.h" #include "api/media_stream_interface.h" #include "api/test/create_network_emulation_manager.h" #include "api/test/create_peer_connection_quality_test_frame_generator.h" @@ -25,13 +26,14 @@ #include "api/test/peerconnection_quality_test_fixture.h" #include "api/test/simulated_network.h" #include "api/test/time_controller.h" +#include "api/transport/bitrate_settings.h" +#include "api/units/data_rate.h" +#include "api/units/time_delta.h" #include "api/video_codecs/vp9_profile.h" #include "media/base/media_constants.h" -#include "modules/video_coding/codecs/vp9/include/vp9.h" #include "system_wrappers/include/field_trial.h" #include "test/field_trial.h" #include "test/gtest.h" -#include "test/network/simulated_network.h" #include "test/pc/e2e/network_quality_metrics_reporter.h" #include "test/testsupport/file_utils.h" diff --git a/video/picture_id_tests.cc b/video/picture_id_tests.cc index 46d6499fcf..4e76e5d676 100644 --- a/video/picture_id_tests.cc +++ b/video/picture_id_tests.cc @@ -8,24 +8,46 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include +#include +#include #include - +#include +#include +#include +#include +#include + +#include "api/array_view.h" +#include "api/environment/environment.h" #include "api/test/simulated_network.h" #include "api/test/video/function_video_encoder_factory.h" -#include "call/fake_network_pipe.h" +#include "api/video/video_codec_type.h" +#include "api/video/video_frame_type.h" +#include "api/video_codecs/sdp_video_format.h" +#include "api/video_codecs/video_codec.h" +#include "api/video_codecs/video_encoder_factory.h" #include "media/engine/internal_encoder_factory.h" #include "media/engine/simulcast_encoder_adapter.h" #include "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h" #include "modules/rtp_rtcp/source/rtp_packet.h" +#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h" +#include "modules/video_coding/codecs/interface/common_constants.h" #include "modules/video_coding/codecs/vp8/include/vp8.h" +#include "modules/video_coding/codecs/vp8/include/vp8_globals.h" #include "modules/video_coding/codecs/vp9/include/vp9.h" -#include "rtc_base/numerics/safe_conversions.h" +#include "modules/video_coding/codecs/vp9/include/vp9_globals.h" +#include "rtc_base/checks.h" +#include "rtc_base/numerics/mod_ops.h" #include "rtc_base/numerics/sequence_number_util.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/task_queue_for_test.h" +#include "rtc_base/thread_annotations.h" #include "test/call_test.h" -#include "test/network/simulated_network.h" +#include "test/gtest.h" +#include "test/rtp_rtcp_observer.h" #include "test/video_test_constants.h" +#include "video/config/video_encoder_config.h" namespace webrtc { namespace { diff --git a/video/quality_convergence_controller.cc b/video/quality_convergence_controller.cc index c063f62b34..a3c9afa185 100644 --- a/video/quality_convergence_controller.cc +++ b/video/quality_convergence_controller.cc @@ -11,10 +11,15 @@ #include "video/quality_convergence_controller.h" #include +#include +#include +#include "api/field_trials_view.h" +#include "api/video/video_codec_type.h" #include "rtc_base/checks.h" #include "rtc_base/experiments/struct_parameters_parser.h" #include "rtc_base/logging.h" +#include "video/quality_convergence_monitor.h" namespace webrtc { namespace { diff --git a/video/quality_convergence_controller_unittest.cc b/video/quality_convergence_controller_unittest.cc index 46965d4bff..4f7cf63b57 100644 --- a/video/quality_convergence_controller_unittest.cc +++ b/video/quality_convergence_controller_unittest.cc @@ -12,6 +12,7 @@ #include +#include "api/video/video_codec_type.h" #include "test/gtest.h" #include "test/scoped_key_value_config.h" diff --git a/video/quality_convergence_monitor.cc b/video/quality_convergence_monitor.cc index 25b7bed7a5..2f5d3f242c 100644 --- a/video/quality_convergence_monitor.cc +++ b/video/quality_convergence_monitor.cc @@ -10,9 +10,12 @@ #include "video/quality_convergence_monitor.h" -#include +#include +#include #include +#include "api/field_trials_view.h" +#include "api/video/video_codec_type.h" #include "rtc_base/checks.h" #include "rtc_base/experiments/struct_parameters_parser.h" diff --git a/video/quality_convergence_monitor.h b/video/quality_convergence_monitor.h index 326372db43..ecc59dd18e 100644 --- a/video/quality_convergence_monitor.h +++ b/video/quality_convergence_monitor.h @@ -11,6 +11,7 @@ #ifndef VIDEO_QUALITY_CONVERGENCE_MONITOR_H_ #define VIDEO_QUALITY_CONVERGENCE_MONITOR_H_ +#include #include #include diff --git a/video/quality_convergence_monitor_unittest.cc b/video/quality_convergence_monitor_unittest.cc index e236f1dab9..c8da0a7d38 100644 --- a/video/quality_convergence_monitor_unittest.cc +++ b/video/quality_convergence_monitor_unittest.cc @@ -10,8 +10,10 @@ */ #include "video/quality_convergence_monitor.h" -#include +#include +#include +#include "api/video/video_codec_type.h" #include "test/gtest.h" #include "test/scoped_key_value_config.h" diff --git a/video/quality_limitation_reason_tracker.cc b/video/quality_limitation_reason_tracker.cc index c2b2cc4043..425070f40b 100644 --- a/video/quality_limitation_reason_tracker.cc +++ b/video/quality_limitation_reason_tracker.cc @@ -10,9 +10,13 @@ #include "video/quality_limitation_reason_tracker.h" +#include +#include #include +#include "common_video/include/quality_limitation_reason.h" #include "rtc_base/checks.h" +#include "system_wrappers/include/clock.h" namespace webrtc { diff --git a/video/quality_limitation_reason_tracker.h b/video/quality_limitation_reason_tracker.h index 15bc90773a..738c3139de 100644 --- a/video/quality_limitation_reason_tracker.h +++ b/video/quality_limitation_reason_tracker.h @@ -11,6 +11,7 @@ #ifndef VIDEO_QUALITY_LIMITATION_REASON_TRACKER_H_ #define VIDEO_QUALITY_LIMITATION_REASON_TRACKER_H_ +#include #include #include "common_video/include/quality_limitation_reason.h" diff --git a/video/quality_limitation_reason_tracker_unittest.cc b/video/quality_limitation_reason_tracker_unittest.cc index f550c0d8e4..f39918fa89 100644 --- a/video/quality_limitation_reason_tracker_unittest.cc +++ b/video/quality_limitation_reason_tracker_unittest.cc @@ -10,6 +10,8 @@ #include "video/quality_limitation_reason_tracker.h" +#include + #include "common_video/include/quality_limitation_reason.h" #include "system_wrappers/include/clock.h" #include "test/gtest.h" diff --git a/video/quality_scaling_tests.cc b/video/quality_scaling_tests.cc index 81aeebb2da..bc9b72b549 100644 --- a/video/quality_scaling_tests.cc +++ b/video/quality_scaling_tests.cc @@ -8,19 +8,44 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include +#include +#include +#include +#include +#include #include +#include +#include "api/array_view.h" +#include "api/environment/environment.h" +#include "api/make_ref_counted.h" +#include "api/rtp_parameters.h" #include "api/test/video/function_video_encoder_factory.h" -#include "media/engine/internal_encoder_factory.h" +#include "api/transport/bitrate_settings.h" +#include "api/units/time_delta.h" +#include "api/video/video_codec_type.h" +#include "api/video/video_frame.h" +#include "api/video/video_sink_interface.h" +#include "api/video/video_source_interface.h" +#include "api/video_codecs/scalability_mode.h" +#include "api/video_codecs/sdp_video_format.h" +#include "api/video_codecs/video_codec.h" +#include "api/video_codecs/video_encoder.h" +#include "call/video_receive_stream.h" +#include "call/video_send_stream.h" #include "modules/video_coding/codecs/h264/include/h264.h" #include "modules/video_coding/codecs/vp8/include/vp8.h" #include "modules/video_coding/codecs/vp9/include/vp9.h" +#include "rtc_base/checks.h" #include "rtc_base/experiments/encoder_info_settings.h" #include "test/call_test.h" -#include "test/field_trial.h" #include "test/frame_generator_capturer.h" +#include "test/gtest.h" +#include "test/rtp_rtcp_observer.h" +#include "test/scoped_key_value_config.h" #include "test/video_test_constants.h" -#include "video/config/encoder_stream_factory.h" +#include "video/config/video_encoder_config.h" namespace webrtc { namespace { diff --git a/video/rate_utilization_tracker.cc b/video/rate_utilization_tracker.cc index ad7125a5a1..75c73b06ba 100644 --- a/video/rate_utilization_tracker.cc +++ b/video/rate_utilization_tracker.cc @@ -11,6 +11,14 @@ #include "video/rate_utilization_tracker.h" #include +#include +#include + +#include "api/units/data_rate.h" +#include "api/units/data_size.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" +#include "rtc_base/checks.h" namespace webrtc { diff --git a/video/rate_utilization_tracker.h b/video/rate_utilization_tracker.h index 619d542c4a..3d93eb7d38 100644 --- a/video/rate_utilization_tracker.h +++ b/video/rate_utilization_tracker.h @@ -11,6 +11,7 @@ #ifndef VIDEO_RATE_UTILIZATION_TRACKER_H_ #define VIDEO_RATE_UTILIZATION_TRACKER_H_ +#include #include #include diff --git a/video/receive_statistics_proxy.cc b/video/receive_statistics_proxy.cc index aae6b60f2d..22d46e4cc9 100644 --- a/video/receive_statistics_proxy.cc +++ b/video/receive_statistics_proxy.cc @@ -12,10 +12,28 @@ #include #include +#include +#include +#include #include #include -#include "modules/video_coding/include/video_codec_interface.h" +#include "absl/strings/string_view.h" +#include "api/rtp_packet_info.h" +#include "api/sequence_checker.h" +#include "api/task_queue/pending_task_safety_flag.h" +#include "api/task_queue/task_queue_base.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" +#include "api/video/video_codec_type.h" +#include "api/video/video_content_type.h" +#include "api/video/video_frame.h" +#include "api/video/video_frame_type.h" +#include "api/video/video_timing.h" +#include "api/video_codecs/video_decoder.h" +#include "call/video_receive_stream.h" +#include "modules/rtp_rtcp/include/rtcp_statistics.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/strings/string_builder.h" @@ -23,6 +41,8 @@ #include "rtc_base/time_utils.h" #include "system_wrappers/include/clock.h" #include "system_wrappers/include/metrics.h" +#include "video/stats_counter.h" +#include "video/video_quality_observer2.h" #include "video/video_receive_stream2.h" namespace webrtc { diff --git a/video/receive_statistics_proxy_unittest.cc b/video/receive_statistics_proxy_unittest.cc index 565d076fdb..a198cd6196 100644 --- a/video/receive_statistics_proxy_unittest.cc +++ b/video/receive_statistics_proxy_unittest.cc @@ -10,26 +10,39 @@ #include "video/receive_statistics_proxy.h" +#include +#include +#include #include #include #include #include #include -#include +#include +#include "api/rtp_packet_info.h" +#include "api/rtp_packet_infos.h" #include "api/scoped_refptr.h" #include "api/units/frequency.h" #include "api/units/time_delta.h" +#include "api/units/timestamp.h" #include "api/video/i420_buffer.h" +#include "api/video/video_codec_type.h" +#include "api/video/video_content_type.h" #include "api/video/video_frame.h" -#include "api/video/video_frame_buffer.h" +#include "api/video/video_frame_type.h" #include "api/video/video_rotation.h" -#include "rtc_base/thread.h" +#include "api/video/video_timing.h" +#include "api/video_codecs/video_decoder.h" +#include "call/video_receive_stream.h" +#include "modules/rtp_rtcp/include/rtcp_statistics.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "system_wrappers/include/metrics.h" #include "test/gmock.h" #include "test/gtest.h" #include "test/scoped_key_value_config.h" #include "test/time_controller/simulated_time_controller.h" +#include "video/video_quality_observer2.h" #include "video/video_receive_stream2.h" namespace webrtc { diff --git a/video/render/incoming_video_stream.cc b/video/render/incoming_video_stream.cc index 8d4e1ea5f2..970f303a33 100644 --- a/video/render/incoming_video_stream.cc +++ b/video/render/incoming_video_stream.cc @@ -10,12 +10,18 @@ #include "video/render/incoming_video_stream.h" +#include #include #include #include +#include "api/sequence_checker.h" +#include "api/task_queue/task_queue_factory.h" #include "api/units/time_delta.h" +#include "api/video/video_frame.h" +#include "api/video/video_sink_interface.h" #include "rtc_base/checks.h" +#include "rtc_base/race_checker.h" #include "rtc_base/trace_event.h" #include "video/render/video_render_frames.h" diff --git a/video/render/video_render_frames.cc b/video/render/video_render_frames.cc index d26a050150..ca5385bded 100644 --- a/video/render/video_render_frames.cc +++ b/video/render/video_render_frames.cc @@ -10,10 +10,12 @@ #include "video/render/video_render_frames.h" -#include +#include +#include +#include #include -#include "rtc_base/checks.h" +#include "api/video/video_frame.h" #include "rtc_base/logging.h" #include "rtc_base/time_utils.h" #include "system_wrappers/include/metrics.h" diff --git a/video/report_block_stats.cc b/video/report_block_stats.cc index bf60364682..966e179d79 100644 --- a/video/report_block_stats.cc +++ b/video/report_block_stats.cc @@ -10,7 +10,7 @@ #include "video/report_block_stats.h" -#include +#include namespace webrtc { diff --git a/video/report_block_stats_unittest.cc b/video/report_block_stats_unittest.cc index bd66e571a0..05095f7f13 100644 --- a/video/report_block_stats_unittest.cc +++ b/video/report_block_stats_unittest.cc @@ -10,6 +10,8 @@ #include "video/report_block_stats.h" +#include + #include "test/gtest.h" namespace webrtc { diff --git a/video/rtp_streams_synchronizer2.cc b/video/rtp_streams_synchronizer2.cc index 5fc75e9f54..cbc1a59124 100644 --- a/video/rtp_streams_synchronizer2.cc +++ b/video/rtp_streams_synchronizer2.cc @@ -10,14 +10,21 @@ #include "video/rtp_streams_synchronizer2.h" +#include #include +#include "api/sequence_checker.h" +#include "api/task_queue/task_queue_base.h" +#include "api/units/time_delta.h" #include "call/syncable.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" +#include "rtc_base/task_utils/repeating_task.h" #include "rtc_base/time_utils.h" #include "rtc_base/trace_event.h" +#include "system_wrappers/include/ntp_time.h" #include "system_wrappers/include/rtp_to_ntp_estimator.h" +#include "video/stream_synchronization.h" namespace webrtc { namespace internal { diff --git a/video/rtp_streams_synchronizer2.h b/video/rtp_streams_synchronizer2.h index 0503abe726..ec04a7c341 100644 --- a/video/rtp_streams_synchronizer2.h +++ b/video/rtp_streams_synchronizer2.h @@ -11,12 +11,14 @@ #ifndef VIDEO_RTP_STREAMS_SYNCHRONIZER2_H_ #define VIDEO_RTP_STREAMS_SYNCHRONIZER2_H_ +#include #include #include "api/sequence_checker.h" #include "api/task_queue/task_queue_base.h" #include "rtc_base/system/no_unique_address.h" #include "rtc_base/task_utils/repeating_task.h" +#include "rtc_base/thread_annotations.h" #include "video/stream_synchronization.h" namespace webrtc { diff --git a/video/screenshare_loopback.cc b/video/screenshare_loopback.cc index da2b2a3276..a0bf6e9cf5 100644 --- a/video/screenshare_loopback.cc +++ b/video/screenshare_loopback.cc @@ -21,12 +21,12 @@ #include "api/test/video_quality_test_fixture.h" #include "api/transport/bitrate_settings.h" #include "api/units/data_rate.h" +#include "api/video_codecs/sdp_video_format.h" #include "api/video_codecs/video_codec.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/string_encode.h" #include "system_wrappers/include/field_trial.h" -#include "test/field_trial.h" #include "test/gtest.h" #include "test/run_test.h" #include "test/test_flags.h" diff --git a/video/send_delay_stats.cc b/video/send_delay_stats.cc index 0deeb89c86..c81d457dc0 100644 --- a/video/send_delay_stats.cc +++ b/video/send_delay_stats.cc @@ -10,10 +10,18 @@ #include "video/send_delay_stats.h" +#include +#include #include +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" +#include "call/video_send_stream.h" #include "rtc_base/logging.h" +#include "rtc_base/synchronization/mutex.h" +#include "system_wrappers/include/clock.h" #include "system_wrappers/include/metrics.h" +#include "video/stats_counter.h" namespace webrtc { namespace { diff --git a/video/send_delay_stats_unittest.cc b/video/send_delay_stats_unittest.cc index e23467e71c..c564562e54 100644 --- a/video/send_delay_stats_unittest.cc +++ b/video/send_delay_stats_unittest.cc @@ -14,7 +14,11 @@ #include #include +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" #include "call/rtp_config.h" +#include "call/video_send_stream.h" +#include "system_wrappers/include/clock.h" #include "system_wrappers/include/metrics.h" #include "test/gtest.h" diff --git a/video/send_statistics_proxy.cc b/video/send_statistics_proxy.cc index 57016db103..38873b4fa4 100644 --- a/video/send_statistics_proxy.cc +++ b/video/send_statistics_proxy.cc @@ -13,19 +13,50 @@ #include #include #include +#include +#include +#include #include +#include +#include +#include #include +#include #include "absl/strings/match.h" +#include "api/field_trials_view.h" +#include "api/units/data_rate.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" +#include "api/video/encoded_image.h" +#include "api/video/video_adaptation_counters.h" +#include "api/video/video_adaptation_reason.h" +#include "api/video/video_bitrate_allocation.h" #include "api/video/video_codec_constants.h" #include "api/video/video_codec_type.h" +#include "api/video/video_content_type.h" +#include "api/video/video_frame_type.h" +#include "api/video/video_timing.h" #include "api/video_codecs/video_codec.h" +#include "call/rtp_config.h" +#include "call/video_send_stream.h" +#include "common_video/frame_counts.h" +#include "common_video/include/quality_limitation_reason.h" +#include "modules/rtp_rtcp/include/report_block_data.h" +#include "modules/rtp_rtcp/include/rtcp_statistics.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "modules/video_coding/codecs/interface/common_constants.h" #include "modules/video_coding/include/video_codec_interface.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/numerics/mod_ops.h" #include "rtc_base/strings/string_builder.h" +#include "rtc_base/synchronization/mutex.h" +#include "system_wrappers/include/clock.h" #include "system_wrappers/include/metrics.h" +#include "video/config/video_encoder_config.h" +#include "video/stats_counter.h" +#include "video/video_stream_encoder_observer.h" namespace webrtc { namespace { diff --git a/video/send_statistics_proxy_unittest.cc b/video/send_statistics_proxy_unittest.cc index 39ff0a6ab8..c41a9cdec4 100644 --- a/video/send_statistics_proxy_unittest.cc +++ b/video/send_statistics_proxy_unittest.cc @@ -11,19 +11,35 @@ #include "video/send_statistics_proxy.h" #include +#include +#include #include #include +#include #include #include #include "absl/algorithm/container.h" +#include "api/units/data_rate.h" +#include "api/units/time_delta.h" #include "api/units/timestamp.h" +#include "api/video/video_adaptation_counters.h" #include "api/video/video_adaptation_reason.h" #include "api/video/video_bitrate_allocation.h" #include "api/video/video_codec_type.h" #include "api/video_codecs/scalability_mode.h" #include "api/video_codecs/video_codec.h" +#include "call/video_send_stream.h" +#include "common_video/frame_counts.h" +#include "common_video/include/quality_limitation_reason.h" +#include "modules/rtp_rtcp/include/report_block_data.h" +#include "modules/rtp_rtcp/include/rtcp_statistics.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h" +#include "modules/video_coding/codecs/interface/common_constants.h" +#include "modules/video_coding/include/video_codec_interface.h" #include "rtc_base/fake_clock.h" +#include "system_wrappers/include/clock.h" #include "system_wrappers/include/metrics.h" #include "test/gmock.h" #include "test/gtest.h" diff --git a/video/stats_counter.cc b/video/stats_counter.cc index 09c2d4dd49..73181a4bb9 100644 --- a/video/stats_counter.cc +++ b/video/stats_counter.cc @@ -11,8 +11,10 @@ #include "video/stats_counter.h" #include +#include #include #include +#include #include "rtc_base/checks.h" #include "rtc_base/strings/string_builder.h" diff --git a/video/stats_counter_unittest.cc b/video/stats_counter_unittest.cc index 32f8c8e5e1..9f67afc116 100644 --- a/video/stats_counter_unittest.cc +++ b/video/stats_counter_unittest.cc @@ -10,6 +10,8 @@ #include "video/stats_counter.h" +#include + #include "system_wrappers/include/clock.h" #include "test/gtest.h" diff --git a/video/stream_synchronization.cc b/video/stream_synchronization.cc index d86cc79203..eae9643884 100644 --- a/video/stream_synchronization.cc +++ b/video/stream_synchronization.cc @@ -13,8 +13,10 @@ #include #include +#include #include "rtc_base/logging.h" +#include "system_wrappers/include/ntp_time.h" namespace webrtc { diff --git a/video/stream_synchronization_unittest.cc b/video/stream_synchronization_unittest.cc index b733a1d2cf..0d1d379e4f 100644 --- a/video/stream_synchronization_unittest.cc +++ b/video/stream_synchronization_unittest.cc @@ -11,9 +11,11 @@ #include "video/stream_synchronization.h" #include +#include #include "system_wrappers/include/clock.h" #include "system_wrappers/include/ntp_time.h" +#include "system_wrappers/include/rtp_to_ntp_estimator.h" #include "test/gtest.h" namespace webrtc { diff --git a/video/sv_loopback.cc b/video/sv_loopback.cc index ee8075a7f1..4e9e3c0a91 100644 --- a/video/sv_loopback.cc +++ b/video/sv_loopback.cc @@ -26,7 +26,6 @@ #include "rtc_base/logging.h" #include "rtc_base/string_encode.h" #include "system_wrappers/include/field_trial.h" -#include "test/field_trial.h" #include "test/gtest.h" #include "test/run_test.h" #include "test/test_flags.h" diff --git a/video/task_queue_frame_decode_scheduler.cc b/video/task_queue_frame_decode_scheduler.cc index 624f8abf4a..569e1c62fc 100644 --- a/video/task_queue_frame_decode_scheduler.cc +++ b/video/task_queue_frame_decode_scheduler.cc @@ -11,11 +11,17 @@ #include "video/task_queue_frame_decode_scheduler.h" #include +#include +#include #include #include "api/sequence_checker.h" +#include "api/task_queue/pending_task_safety_flag.h" #include "api/task_queue/task_queue_base.h" +#include "api/units/time_delta.h" #include "rtc_base/checks.h" +#include "system_wrappers/include/clock.h" +#include "video/frame_decode_timing.h" namespace webrtc { diff --git a/video/task_queue_frame_decode_scheduler.h b/video/task_queue_frame_decode_scheduler.h index 1d95c3e344..690a6b1b9e 100644 --- a/video/task_queue_frame_decode_scheduler.h +++ b/video/task_queue_frame_decode_scheduler.h @@ -11,7 +11,14 @@ #ifndef VIDEO_TASK_QUEUE_FRAME_DECODE_SCHEDULER_H_ #define VIDEO_TASK_QUEUE_FRAME_DECODE_SCHEDULER_H_ +#include +#include + +#include "api/task_queue/pending_task_safety_flag.h" +#include "api/task_queue/task_queue_base.h" +#include "system_wrappers/include/clock.h" #include "video/frame_decode_scheduler.h" +#include "video/frame_decode_timing.h" namespace webrtc { diff --git a/video/task_queue_frame_decode_scheduler_unittest.cc b/video/task_queue_frame_decode_scheduler_unittest.cc index 38b43e6e41..a4a36629f4 100644 --- a/video/task_queue_frame_decode_scheduler_unittest.cc +++ b/video/task_queue_frame_decode_scheduler_unittest.cc @@ -12,15 +12,15 @@ #include -#include +#include #include -#include #include "api/units/time_delta.h" #include "api/units/timestamp.h" #include "test/gmock.h" #include "test/gtest.h" #include "test/time_controller/simulated_time_controller.h" +#include "video/frame_decode_timing.h" namespace webrtc { diff --git a/video/test/mock_video_stream_encoder.h b/video/test/mock_video_stream_encoder.h index f1cfc7f4a8..13a6b8fa80 100644 --- a/video/test/mock_video_stream_encoder.h +++ b/video/test/mock_video_stream_encoder.h @@ -10,9 +10,22 @@ #ifndef VIDEO_TEST_MOCK_VIDEO_STREAM_ENCODER_H_ #define VIDEO_TEST_MOCK_VIDEO_STREAM_ENCODER_H_ +#include +#include #include +#include "api/adaptation/resource.h" +#include "api/fec_controller_override.h" +#include "api/rtp_parameters.h" +#include "api/rtp_sender_interface.h" +#include "api/scoped_refptr.h" +#include "api/units/data_rate.h" +#include "api/video/video_frame.h" +#include "api/video/video_frame_type.h" +#include "api/video/video_source_interface.h" +#include "api/video_codecs/video_encoder.h" #include "test/gmock.h" +#include "video/config/video_encoder_config.h" #include "video/video_stream_encoder_interface.h" namespace webrtc { diff --git a/video/unique_timestamp_counter_unittest.cc b/video/unique_timestamp_counter_unittest.cc index b703e84576..d2bca7fd10 100644 --- a/video/unique_timestamp_counter_unittest.cc +++ b/video/unique_timestamp_counter_unittest.cc @@ -9,6 +9,8 @@ */ #include "video/unique_timestamp_counter.h" +#include + #include "test/gtest.h" namespace webrtc { diff --git a/video/video_loopback.cc b/video/video_loopback.cc index 09254d7e73..b8659d13cf 100644 --- a/video/video_loopback.cc +++ b/video/video_loopback.cc @@ -26,7 +26,6 @@ #include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "system_wrappers/include/field_trial.h" -#include "test/field_trial.h" #include "test/gtest.h" #include "test/run_test.h" #include "test/test_flags.h" diff --git a/video/video_quality_observer2.cc b/video/video_quality_observer2.cc index 3c48fd2747..90354d44c0 100644 --- a/video/video_quality_observer2.cc +++ b/video/video_quality_observer2.cc @@ -13,8 +13,12 @@ #include #include #include +#include +#include #include +#include "api/video/video_codec_type.h" +#include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/strings/string_builder.h" #include "system_wrappers/include/metrics.h" diff --git a/video/video_quality_observer2.h b/video/video_quality_observer2.h index e9b66c5dfd..d46d9f669f 100644 --- a/video/video_quality_observer2.h +++ b/video/video_quality_observer2.h @@ -18,7 +18,6 @@ #include #include "api/video/video_codec_type.h" -#include "api/video/video_content_type.h" #include "rtc_base/numerics/moving_average.h" #include "rtc_base/numerics/sample_counter.h" diff --git a/video/video_receive_stream_timeout_tracker.cc b/video/video_receive_stream_timeout_tracker.cc index 0409f26560..e966f2c8b7 100644 --- a/video/video_receive_stream_timeout_tracker.cc +++ b/video/video_receive_stream_timeout_tracker.cc @@ -13,7 +13,14 @@ #include #include +#include "api/sequence_checker.h" +#include "api/task_queue/task_queue_base.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" +#include "rtc_base/checks.h" #include "rtc_base/logging.h" +#include "rtc_base/task_utils/repeating_task.h" +#include "system_wrappers/include/clock.h" namespace webrtc { diff --git a/video/video_receive_stream_timeout_tracker.h b/video/video_receive_stream_timeout_tracker.h index c15aa70e92..7672d1760d 100644 --- a/video/video_receive_stream_timeout_tracker.h +++ b/video/video_receive_stream_timeout_tracker.h @@ -13,10 +13,12 @@ #include +#include "api/sequence_checker.h" #include "api/task_queue/task_queue_base.h" #include "api/units/time_delta.h" #include "api/units/timestamp.h" #include "rtc_base/task_utils/repeating_task.h" +#include "rtc_base/thread_annotations.h" #include "system_wrappers/include/clock.h" namespace webrtc { diff --git a/video/video_receive_stream_timeout_tracker_unittest.cc b/video/video_receive_stream_timeout_tracker_unittest.cc index ea3bf896b8..72a51df162 100644 --- a/video/video_receive_stream_timeout_tracker_unittest.cc +++ b/video/video_receive_stream_timeout_tracker_unittest.cc @@ -10,10 +10,10 @@ #include "video/video_receive_stream_timeout_tracker.h" -#include #include -#include "api/task_queue/task_queue_base.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" #include "test/gmock.h" #include "test/gtest.h" #include "test/time_controller/simulated_time_controller.h" diff --git a/video/video_send_stream_impl.h b/video/video_send_stream_impl.h index fec8962c01..646d4e4c53 100644 --- a/video/video_send_stream_impl.h +++ b/video/video_send_stream_impl.h @@ -20,13 +20,26 @@ #include #include +#include "api/adaptation/resource.h" +#include "api/call/bitrate_allocation.h" #include "api/environment/environment.h" +#include "api/fec_controller.h" #include "api/field_trials_view.h" #include "api/metronome/metronome.h" +#include "api/rtp_parameters.h" +#include "api/rtp_sender_interface.h" +#include "api/scoped_refptr.h" +#include "api/sequence_checker.h" #include "api/task_queue/pending_task_safety_flag.h" #include "api/task_queue/task_queue_base.h" +#include "api/units/data_rate.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" #include "api/video/encoded_image.h" #include "api/video/video_bitrate_allocation.h" +#include "api/video/video_frame.h" +#include "api/video/video_layers_allocation.h" +#include "api/video/video_source_interface.h" #include "api/video_codecs/video_encoder.h" #include "call/bitrate_allocator.h" #include "call/rtp_config.h" diff --git a/video/video_send_stream_impl_unittest.cc b/video/video_send_stream_impl_unittest.cc index a298a017ba..7a11b78eba 100644 --- a/video/video_send_stream_impl_unittest.cc +++ b/video/video_send_stream_impl_unittest.cc @@ -22,7 +22,6 @@ #include "api/array_view.h" #include "api/call/bitrate_allocation.h" -#include "api/environment/environment.h" #include "api/environment/environment_factory.h" #include "api/rtc_event_log/rtc_event_log.h" #include "api/rtp_parameters.h" @@ -56,7 +55,6 @@ #include "video/send_delay_stats.h" #include "video/send_statistics_proxy.h" #include "video/test/mock_video_stream_encoder.h" -#include "video/video_stream_encoder.h" #include "video/video_stream_encoder_interface.h" namespace webrtc { diff --git a/video/video_source_sink_controller.cc b/video/video_source_sink_controller.cc index 51316f8b0c..92c65d8988 100644 --- a/video/video_source_sink_controller.cc +++ b/video/video_source_sink_controller.cc @@ -11,12 +11,19 @@ #include "video/video_source_sink_controller.h" #include +#include #include +#include #include - -#include "rtc_base/logging.h" +#include + +#include "api/sequence_checker.h" +#include "api/video/video_frame.h" +#include "api/video/video_sink_interface.h" +#include "api/video/video_source_interface.h" +#include "call/adaptation/video_source_restrictions.h" +#include "rtc_base/checks.h" #include "rtc_base/numerics/safe_conversions.h" -#include "rtc_base/strings/string_builder.h" namespace webrtc { diff --git a/video/video_source_sink_controller.h b/video/video_source_sink_controller.h index 451c59479e..53d2638c47 100644 --- a/video/video_source_sink_controller.h +++ b/video/video_source_sink_controller.h @@ -11,8 +11,8 @@ #ifndef VIDEO_VIDEO_SOURCE_SINK_CONTROLLER_H_ #define VIDEO_VIDEO_SOURCE_SINK_CONTROLLER_H_ +#include #include -#include #include #include "api/sequence_checker.h" @@ -21,6 +21,7 @@ #include "api/video/video_source_interface.h" #include "call/adaptation/video_source_restrictions.h" #include "rtc_base/system/no_unique_address.h" +#include "rtc_base/thread_annotations.h" namespace webrtc { diff --git a/video/video_source_sink_controller_unittest.cc b/video/video_source_sink_controller_unittest.cc index a1df43a71e..bc4d263166 100644 --- a/video/video_source_sink_controller_unittest.cc +++ b/video/video_source_sink_controller_unittest.cc @@ -11,8 +11,10 @@ #include "video/video_source_sink_controller.h" #include +#include #include "api/video/video_frame.h" +#include "api/video/video_sink_interface.h" #include "api/video/video_source_interface.h" #include "call/adaptation/video_source_restrictions.h" #include "test/gmock.h" diff --git a/video/video_stream_buffer_controller.cc b/video/video_stream_buffer_controller.cc index 83de75ac3f..93bb147802 100644 --- a/video/video_stream_buffer_controller.cc +++ b/video/video_stream_buffer_controller.cc @@ -11,12 +11,15 @@ #include "video/video_stream_buffer_controller.h" #include +#include +#include #include #include #include -#include "absl/base/attributes.h" +#include "absl/container/inlined_vector.h" #include "absl/functional/bind_front.h" +#include "api/field_trials_view.h" #include "api/sequence_checker.h" #include "api/task_queue/task_queue_base.h" #include "api/units/data_size.h" @@ -25,15 +28,17 @@ #include "api/video/encoded_frame.h" #include "api/video/frame_buffer.h" #include "api/video/video_content_type.h" +#include "api/video/video_timing.h" #include "modules/video_coding/frame_helpers.h" +#include "modules/video_coding/include/video_coding_defines.h" #include "modules/video_coding/timing/inter_frame_delay_variation_calculator.h" #include "modules/video_coding/timing/jitter_estimator.h" +#include "modules/video_coding/timing/timing.h" #include "rtc_base/checks.h" +#include "rtc_base/experiments/field_trial_parser.h" #include "rtc_base/logging.h" -#include "rtc_base/thread_annotations.h" #include "video/frame_decode_scheduler.h" #include "video/frame_decode_timing.h" -#include "video/task_queue_frame_decode_scheduler.h" #include "video/video_receive_stream_timeout_tracker.h" namespace webrtc { diff --git a/video/video_stream_buffer_controller.h b/video/video_stream_buffer_controller.h index 43d4544f71..17227871cc 100644 --- a/video/video_stream_buffer_controller.h +++ b/video/video_stream_buffer_controller.h @@ -11,18 +11,32 @@ #ifndef VIDEO_VIDEO_STREAM_BUFFER_CONTROLLER_H_ #define VIDEO_VIDEO_STREAM_BUFFER_CONTROLLER_H_ +#include +#include #include +#include +#include "absl/container/inlined_vector.h" #include "api/field_trials_view.h" +#include "api/sequence_checker.h" +#include "api/task_queue/pending_task_safety_flag.h" #include "api/task_queue/task_queue_base.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" #include "api/video/encoded_frame.h" #include "api/video/frame_buffer.h" +#include "api/video/video_content_type.h" +#include "api/video/video_timing.h" #include "modules/video_coding/include/video_coding_defines.h" #include "modules/video_coding/timing/inter_frame_delay_variation_calculator.h" #include "modules/video_coding/timing/jitter_estimator.h" #include "modules/video_coding/timing/timing.h" +#include "rtc_base/experiments/field_trial_parser.h" +#include "rtc_base/system/no_unique_address.h" +#include "rtc_base/thread_annotations.h" #include "system_wrappers/include/clock.h" -#include "video/decode_synchronizer.h" +#include "video/frame_decode_scheduler.h" +#include "video/frame_decode_timing.h" #include "video/video_receive_stream_timeout_tracker.h" namespace webrtc { diff --git a/video/video_stream_buffer_controller_unittest.cc b/video/video_stream_buffer_controller_unittest.cc index 6987e4a503..08a15338c1 100644 --- a/video/video_stream_buffer_controller_unittest.cc +++ b/video/video_stream_buffer_controller_unittest.cc @@ -12,6 +12,7 @@ #include +#include #include #include #include @@ -25,9 +26,12 @@ #include "api/units/frequency.h" #include "api/units/time_delta.h" #include "api/units/timestamp.h" +#include "api/video/encoded_frame.h" #include "api/video/video_content_type.h" #include "api/video/video_timing.h" +#include "modules/video_coding/timing/timing.h" #include "rtc_base/checks.h" +#include "system_wrappers/include/clock.h" #include "test/fake_encoded_frame.h" #include "test/gmock.h" #include "test/gtest.h" diff --git a/video/video_stream_decoder2.cc b/video/video_stream_decoder2.cc index 51175b7717..02b3db821e 100644 --- a/video/video_stream_decoder2.cc +++ b/video/video_stream_decoder2.cc @@ -15,6 +15,7 @@ #include "api/units/time_delta.h" #include "api/video/video_frame.h" +#include "api/video/video_sink_interface.h" #include "api/video_codecs/video_decoder.h" #include "modules/video_coding/video_receiver2.h" #include "rtc_base/checks.h" diff --git a/video/video_stream_decoder2.h b/video/video_stream_decoder2.h index 616ec12ed1..dad603605a 100644 --- a/video/video_stream_decoder2.h +++ b/video/video_stream_decoder2.h @@ -12,18 +12,11 @@ #define VIDEO_VIDEO_STREAM_DECODER2_H_ #include -#include -#include -#include -#include -#include "api/scoped_refptr.h" #include "api/video/video_frame.h" #include "api/video/video_sink_interface.h" #include "api/video_codecs/video_decoder.h" -#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" #include "modules/video_coding/include/video_coding_defines.h" -#include "rtc_base/platform_thread.h" namespace webrtc { diff --git a/video/video_stream_decoder_impl_unittest.cc b/video/video_stream_decoder_impl_unittest.cc deleted file mode 100644 index 6942e06679..0000000000 --- a/video/video_stream_decoder_impl_unittest.cc +++ /dev/null @@ -1,222 +0,0 @@ -/* - * Copyright (c) 2020 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "video/video_stream_decoder_impl.h" - -#include - -#include "api/video/i420_buffer.h" -#include "api/video_codecs/video_decoder.h" -#include "test/fake_encoded_frame.h" -#include "test/gmock.h" -#include "test/gtest.h" -#include "test/scoped_key_value_config.h" -#include "test/time_controller/simulated_time_controller.h" - -namespace webrtc { -namespace { -using ::testing::_; -using ::testing::NiceMock; -using ::testing::Return; - -class MockVideoStreamDecoderCallbacks - : public VideoStreamDecoderInterface::Callbacks { - public: - MOCK_METHOD(void, OnNonDecodableState, (), (override)); - MOCK_METHOD(void, OnContinuousUntil, (int64_t frame_id), (override)); - MOCK_METHOD( - void, - OnDecodedFrame, - (VideoFrame frame, - const VideoStreamDecoderInterface::Callbacks::FrameInfo& frame_info), - (override)); -}; - -class StubVideoDecoder : public VideoDecoder { - public: - StubVideoDecoder() { ON_CALL(*this, Configure).WillByDefault(Return(true)); } - - MOCK_METHOD(bool, Configure, (const Settings&), (override)); - - int32_t Decode(const EncodedImage& input_image, - bool missing_frames, - int64_t render_time_ms) override { - int32_t ret_code = DecodeCall(input_image, missing_frames, render_time_ms); - if (ret_code == WEBRTC_VIDEO_CODEC_OK || - ret_code == WEBRTC_VIDEO_CODEC_OK_REQUEST_KEYFRAME) { - VideoFrame frame = VideoFrame::Builder() - .set_video_frame_buffer(I420Buffer::Create(1, 1)) - .build(); - callback_->Decoded(frame); - } - return ret_code; - } - - MOCK_METHOD(int32_t, - DecodeCall, - (const EncodedImage& input_image, - bool missing_frames, - int64_t render_time_ms), - ()); - - int32_t Release() override { return 0; } - - int32_t RegisterDecodeCompleteCallback( - DecodedImageCallback* callback) override { - callback_ = callback; - return 0; - } - - private: - DecodedImageCallback* callback_; -}; - -class WrappedVideoDecoder : public VideoDecoder { - public: - explicit WrappedVideoDecoder(StubVideoDecoder* decoder) : decoder_(decoder) {} - - bool Configure(const Settings& settings) override { - return decoder_->Configure(settings); - } - int32_t Decode(const EncodedImage& input_image, - bool missing_frames, - int64_t render_time_ms) override { - return decoder_->Decode(input_image, missing_frames, render_time_ms); - } - int32_t Release() override { return decoder_->Release(); } - - int32_t RegisterDecodeCompleteCallback( - DecodedImageCallback* callback) override { - return decoder_->RegisterDecodeCompleteCallback(callback); - } - - private: - StubVideoDecoder* decoder_; -}; - -class FakeVideoDecoderFactory : public VideoDecoderFactory { - public: - std::vector GetSupportedFormats() const override { - return {}; - } - std::unique_ptr CreateVideoDecoder( - const SdpVideoFormat& format) override { - if (format.name == "VP8") { - return std::make_unique(&vp8_decoder_); - } - - if (format.name == "AV1") { - return std::make_unique(&av1_decoder_); - } - - return {}; - } - - StubVideoDecoder& Vp8Decoder() { return vp8_decoder_; } - StubVideoDecoder& Av1Decoder() { return av1_decoder_; } - - private: - NiceMock vp8_decoder_; - NiceMock av1_decoder_; -}; - -class VideoStreamDecoderImplTest : public ::testing::Test { - public: - VideoStreamDecoderImplTest() - : time_controller_(Timestamp::Seconds(0)), - video_stream_decoder_( - &callbacks_, - &decoder_factory_, - time_controller_.GetTaskQueueFactory(), - {{1, std::make_pair(SdpVideoFormat::VP8(), 1)}, - {2, std::make_pair(SdpVideoFormat::AV1Profile0(), 1)}}, - &field_trials_) { - // Set the min playout delay to a value greater than zero to not activate - // the low-latency renderer. - video_stream_decoder_.SetMinPlayoutDelay(TimeDelta::Millis(10)); - } - - test::ScopedKeyValueConfig field_trials_; - NiceMock callbacks_; - FakeVideoDecoderFactory decoder_factory_; - GlobalSimulatedTimeController time_controller_; - VideoStreamDecoderImpl video_stream_decoder_; -}; - -TEST_F(VideoStreamDecoderImplTest, InsertAndDecodeFrame) { - video_stream_decoder_.OnFrame( - test::FakeFrameBuilder().PayloadType(1).AsLast().Build()); - EXPECT_CALL(callbacks_, OnDecodedFrame); - time_controller_.AdvanceTime(TimeDelta::Millis(1)); -} - -TEST_F(VideoStreamDecoderImplTest, NonDecodableStateWaitingForKeyframe) { - EXPECT_CALL(callbacks_, OnNonDecodableState); - time_controller_.AdvanceTime(TimeDelta::Millis(200)); -} - -TEST_F(VideoStreamDecoderImplTest, NonDecodableStateWaitingForDeltaFrame) { - video_stream_decoder_.OnFrame( - test::FakeFrameBuilder().PayloadType(1).AsLast().Build()); - EXPECT_CALL(callbacks_, OnDecodedFrame); - time_controller_.AdvanceTime(TimeDelta::Millis(1)); - EXPECT_CALL(callbacks_, OnNonDecodableState); - time_controller_.AdvanceTime(TimeDelta::Millis(3000)); -} - -TEST_F(VideoStreamDecoderImplTest, InsertAndDecodeFrameWithKeyframeRequest) { - video_stream_decoder_.OnFrame( - test::FakeFrameBuilder().PayloadType(1).AsLast().Build()); - EXPECT_CALL(decoder_factory_.Vp8Decoder(), DecodeCall) - .WillOnce(Return(WEBRTC_VIDEO_CODEC_OK_REQUEST_KEYFRAME)); - EXPECT_CALL(callbacks_, OnDecodedFrame); - EXPECT_CALL(callbacks_, OnNonDecodableState); - time_controller_.AdvanceTime(TimeDelta::Millis(1)); -} - -TEST_F(VideoStreamDecoderImplTest, FailToInitDecoder) { - video_stream_decoder_.OnFrame( - test::FakeFrameBuilder() - .ReceivedTime(time_controller_.GetClock()->CurrentTime()) - .PayloadType(1) - .AsLast() - .Build()); - ON_CALL(decoder_factory_.Vp8Decoder(), Configure) - .WillByDefault(Return(false)); - EXPECT_CALL(callbacks_, OnNonDecodableState); - time_controller_.AdvanceTime(TimeDelta::Millis(1)); -} - -TEST_F(VideoStreamDecoderImplTest, FailToDecodeFrame) { - video_stream_decoder_.OnFrame( - test::FakeFrameBuilder().PayloadType(1).AsLast().Build()); - ON_CALL(decoder_factory_.Vp8Decoder(), DecodeCall) - .WillByDefault(Return(WEBRTC_VIDEO_CODEC_ERROR)); - EXPECT_CALL(callbacks_, OnNonDecodableState); - time_controller_.AdvanceTime(TimeDelta::Millis(1)); -} - -TEST_F(VideoStreamDecoderImplTest, ChangeFramePayloadType) { - constexpr TimeDelta kFrameInterval = TimeDelta::Millis(1000 / 60); - video_stream_decoder_.OnFrame( - test::FakeFrameBuilder().PayloadType(1).Id(0).AsLast().Build()); - EXPECT_CALL(decoder_factory_.Vp8Decoder(), DecodeCall); - EXPECT_CALL(callbacks_, OnDecodedFrame); - time_controller_.AdvanceTime(kFrameInterval); - - video_stream_decoder_.OnFrame( - test::FakeFrameBuilder().PayloadType(2).Id(1).AsLast().Build()); - EXPECT_CALL(decoder_factory_.Av1Decoder(), DecodeCall); - EXPECT_CALL(callbacks_, OnDecodedFrame); - time_controller_.AdvanceTime(kFrameInterval); -} - -} // namespace -} // namespace webrtc diff --git a/video/video_stream_encoder.h b/video/video_stream_encoder.h index 917d928149..8c0f2ca04d 100644 --- a/video/video_stream_encoder.h +++ b/video/video_stream_encoder.h @@ -12,38 +12,49 @@ #define VIDEO_VIDEO_STREAM_ENCODER_H_ #include -#include +#include +#include #include -#include +#include #include #include "absl/container/inlined_vector.h" #include "api/adaptation/resource.h" #include "api/environment/environment.h" +#include "api/fec_controller_override.h" +#include "api/rtp_parameters.h" #include "api/rtp_sender_interface.h" +#include "api/scoped_refptr.h" #include "api/sequence_checker.h" #include "api/task_queue/pending_task_safety_flag.h" +#include "api/task_queue/task_queue_base.h" #include "api/units/data_rate.h" +#include "api/units/data_size.h" +#include "api/units/timestamp.h" #include "api/video/encoded_image.h" +#include "api/video/video_adaptation_counters.h" +#include "api/video/video_adaptation_reason.h" +#include "api/video/video_bitrate_allocation.h" #include "api/video/video_bitrate_allocator.h" -#include "api/video/video_rotation.h" -#include "api/video/video_sink_interface.h" +#include "api/video/video_frame.h" +#include "api/video/video_frame_type.h" +#include "api/video/video_source_interface.h" #include "api/video/video_stream_encoder_settings.h" #include "api/video_codecs/video_codec.h" #include "api/video_codecs/video_encoder.h" +#include "api/video_codecs/video_encoder_factory.h" #include "call/adaptation/adaptation_constraint.h" -#include "call/adaptation/resource_adaptation_processor.h" #include "call/adaptation/resource_adaptation_processor_interface.h" #include "call/adaptation/video_source_restrictions.h" +#include "call/adaptation/video_stream_adapter.h" #include "call/adaptation/video_stream_input_state_provider.h" #include "modules/video_coding/utility/frame_dropper.h" #include "modules/video_coding/utility/qp_parser.h" #include "rtc_base/experiments/rate_control_settings.h" -#include "rtc_base/numerics/exp_filter.h" -#include "rtc_base/race_checker.h" -#include "rtc_base/rate_statistics.h" #include "rtc_base/thread_annotations.h" +#include "video/adaptation/overuse_frame_detector.h" #include "video/adaptation/video_stream_encoder_resource_manager.h" +#include "video/config/video_encoder_config.h" #include "video/corruption_detection/frame_instrumentation_generator.h" #include "video/encoder_bitrate_adjuster.h" #include "video/frame_cadence_adapter.h" diff --git a/video/video_stream_encoder_interface.h b/video/video_stream_encoder_interface.h index 05da612f26..234ef33d88 100644 --- a/video/video_stream_encoder_interface.h +++ b/video/video_stream_encoder_interface.h @@ -11,18 +11,20 @@ #ifndef VIDEO_VIDEO_STREAM_ENCODER_INTERFACE_H_ #define VIDEO_VIDEO_STREAM_ENCODER_INTERFACE_H_ +#include +#include #include #include "api/adaptation/resource.h" #include "api/fec_controller_override.h" -#include "api/rtc_error.h" #include "api/rtp_parameters.h" // For DegradationPreference. #include "api/rtp_sender_interface.h" #include "api/scoped_refptr.h" #include "api/units/data_rate.h" -#include "api/video/video_bitrate_allocator.h" +#include "api/video/video_bitrate_allocation.h" +#include "api/video/video_frame.h" +#include "api/video/video_frame_type.h" #include "api/video/video_layers_allocation.h" -#include "api/video/video_sink_interface.h" #include "api/video/video_source_interface.h" #include "api/video_codecs/video_encoder.h" #include "video/config/video_encoder_config.h" diff --git a/video/video_stream_encoder_observer.h b/video/video_stream_encoder_observer.h index 95ca5fa887..ec357043c4 100644 --- a/video/video_stream_encoder_observer.h +++ b/video/video_stream_encoder_observer.h @@ -17,6 +17,7 @@ #include "api/video/video_adaptation_counters.h" #include "api/video/video_adaptation_reason.h" #include "api/video/video_bitrate_allocation.h" +#include "api/video_codecs/video_codec.h" #include "api/video_codecs/video_encoder.h" #include "video/config/video_encoder_config.h" diff --git a/video/video_stream_encoder_unittest.cc b/video/video_stream_encoder_unittest.cc index 17f4ea1a2c..ea47223c53 100644 --- a/video/video_stream_encoder_unittest.cc +++ b/video/video_stream_encoder_unittest.cc @@ -94,13 +94,13 @@ #include "rtc_base/event.h" #include "rtc_base/experiments/encoder_info_settings.h" #include "rtc_base/experiments/rate_control_settings.h" -#include "rtc_base/gunit.h" #include "rtc_base/logging.h" #include "rtc_base/ref_counted_object.h" #include "rtc_base/strings/string_builder.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/thread_annotations.h" #include "rtc_base/time_utils.h" +#include "system_wrappers/include/clock.h" #include "system_wrappers/include/metrics.h" #include "test/encoder_settings.h" #include "test/fake_encoder.h" From c4fe8256f3d44eb95e9cacb7eabded04d6aff5e4 Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Tue, 6 May 2025 15:39:54 -0700 Subject: [PATCH 074/292] IWYU modules/audio_coding MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit using find modules/audio_coding -name "*.h" -o -name "*.cc" | grep -v mock_debug_dump_writer.h | xargs tools_webrtc/iwyu/apply-include-cleaner followed by tools_webrtc/gn_check_autofix.py -C out/Default/ and git cl format Manual changes: * controller_manager.h: add api/array_view.h include * RTPFile.cc / dtmf_buffer_unittest.cc: use rtc_base/ip_address.h instea of netinet/in.h * neteq_delay_analyzer.cc: add TODO for stringstream to ostream include * opus: fixup paths and make third_party/opus includes go via opus_interface.h by using IWYU export pragmas The mock_debug_dump_writer has issues with protobuf includes so was ignored for this round. BUG=webrtc:42226242 Change-Id: I5b8613053da6cc0a2a44d1d59bd2efe11e501681 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/387640 Reviewed-by: Jakob Ivarsson‎ Commit-Queue: Fredrik Solenberg Reviewed-by: Fredrik Solenberg Cr-Commit-Position: refs/heads/main@{#44546} --- modules/audio_coding/BUILD.gn | 72 +++++++++++++++++++ modules/audio_coding/acm2/acm_receive_test.cc | 7 ++ modules/audio_coding/acm2/acm_receive_test.h | 2 +- modules/audio_coding/acm2/acm_remixing.cc | 8 +++ modules/audio_coding/acm2/acm_remixing.h | 2 + .../acm2/acm_remixing_unittest.cc | 4 +- modules/audio_coding/acm2/acm_resampler.cc | 5 ++ modules/audio_coding/acm2/acm_resampler.h | 2 + modules/audio_coding/acm2/acm_send_test.cc | 11 ++- modules/audio_coding/acm2/acm_send_test.h | 3 + modules/audio_coding/acm2/call_statistics.cc | 2 + .../acm2/call_statistics_unittest.cc | 2 + .../audio_network_adaptor_impl.cc | 9 ++- .../audio_network_adaptor_impl_unittest.cc | 6 ++ .../bitrate_controller.cc | 2 +- .../bitrate_controller_unittest.cc | 6 +- .../channel_controller.cc | 2 + .../channel_controller_unittest.cc | 4 ++ .../audio_network_adaptor/controller.h | 3 +- .../controller_manager.cc | 11 ++- .../controller_manager.h | 6 +- .../debug_dump_writer.cc | 7 ++ .../audio_network_adaptor/debug_dump_writer.h | 5 +- .../audio_network_adaptor/dtx_controller.cc | 1 + .../dtx_controller_unittest.cc | 3 + .../audio_network_adaptor/event_log_writer.cc | 1 - .../event_log_writer_unittest.cc | 3 + .../fec_controller_plr_based.cc | 6 +- .../frame_length_controller.cc | 4 ++ .../frame_length_controller_unittest.cc | 7 +- .../frame_length_controller_v2.cc | 4 +- .../frame_length_controller_v2.h | 3 +- .../frame_length_controller_v2_unittest.cc | 2 +- .../include/audio_network_adaptor.h | 3 +- .../mock/mock_audio_network_adaptor.h | 5 ++ .../mock/mock_controller.h | 1 + .../mock/mock_controller_manager.h | 1 + .../util/threshold_curve_unittest.cc | 1 + .../builtin_audio_decoder_factory_unittest.cc | 9 +++ .../builtin_audio_encoder_factory_unittest.cc | 9 +++ .../codecs/cng/audio_encoder_cng.cc | 6 ++ .../codecs/cng/audio_encoder_cng_unittest.cc | 12 +++- .../audio_coding/codecs/cng/cng_unittest.cc | 7 +- modules/audio_coding/codecs/cng/webrtc_cng.cc | 4 ++ .../codecs/g711/audio_decoder_pcm.cc | 6 ++ .../codecs/g711/audio_encoder_pcm.cc | 7 ++ .../codecs/g711/audio_encoder_pcm.h | 4 ++ .../audio_coding/codecs/g711/test/testG711.cc | 2 + .../codecs/g722/audio_decoder_g722.cc | 4 ++ .../codecs/g722/audio_decoder_g722.h | 5 ++ .../codecs/g722/audio_encoder_g722.cc | 9 ++- .../codecs/g722/audio_encoder_g722.h | 3 + .../audio_coding/codecs/g722/test/testG722.cc | 2 + .../isac/main/source/os_specific_inline.h | 4 -- .../codecs/isac/main/source/pitch_filter.h | 1 + .../codecs/isac/main/source/structs.h | 2 + .../codecs/legacy_encoded_audio_frame.cc | 7 ++ .../codecs/opus/audio_coder_opus_common.cc | 7 ++ .../codecs/opus/audio_coder_opus_common.h | 4 ++ .../audio_decoder_multi_channel_opus_impl.cc | 13 ++-- .../audio_decoder_multi_channel_opus_impl.h | 2 + ...dio_decoder_multi_channel_opus_unittest.cc | 7 ++ .../codecs/opus/audio_decoder_opus.cc | 7 +- .../opus/audio_decoder_opus_unittest.cc | 5 ++ .../audio_encoder_multi_channel_opus_impl.cc | 11 +++ ...dio_encoder_multi_channel_opus_unittest.cc | 7 ++ .../codecs/opus/audio_encoder_opus.cc | 14 ++++ .../codecs/opus/audio_encoder_opus.h | 7 ++ .../opus/audio_encoder_opus_unittest.cc | 18 ++++- .../codecs/opus/opus_bandwidth_unittest.cc | 13 ++++ .../codecs/opus/opus_complexity_unittest.cc | 8 +++ .../audio_coding/codecs/opus/opus_fec_test.cc | 7 ++ modules/audio_coding/codecs/opus/opus_inst.h | 6 +- .../codecs/opus/opus_interface.cc | 3 +- .../audio_coding/codecs/opus/opus_interface.h | 2 +- .../codecs/opus/opus_speed_test.cc | 8 +++ .../audio_coding/codecs/opus/opus_unittest.cc | 11 ++- .../audio_coding/codecs/opus/test/BUILD.gn | 1 + .../codecs/opus/test/audio_ring_buffer.cc | 2 + .../codecs/opus/test/audio_ring_buffer.h | 1 - .../opus/test/audio_ring_buffer_unittest.cc | 1 + .../audio_coding/codecs/opus/test/blocker.h | 1 + .../codecs/opus/test/blocker_unittest.cc | 3 + .../codecs/opus/test/lapped_transform.cc | 1 - .../codecs/opus/test/lapped_transform.h | 2 + .../opus/test/lapped_transform_unittest.cc | 2 + .../codecs/pcm16b/audio_decoder_pcm16b.cc | 5 ++ .../codecs/pcm16b/audio_encoder_pcm16b.cc | 6 +- .../codecs/pcm16b/audio_encoder_pcm16b.h | 4 ++ .../codecs/pcm16b/pcm16b_common.cc | 3 + .../codecs/red/audio_encoder_copy_red.cc | 13 +++- .../codecs/red/audio_encoder_copy_red.h | 2 + .../red/audio_encoder_copy_red_unittest.cc | 12 +++- .../codecs/tools/audio_codec_speed_test.cc | 6 ++ .../codecs/tools/audio_codec_speed_test.h | 4 ++ .../include/audio_coding_module.h | 6 +- .../include/audio_coding_module_typedefs.h | 2 +- modules/audio_coding/neteq/accelerate.cc | 5 ++ .../neteq/audio_decoder_unittest.cc | 15 +++- .../audio_coding/neteq/audio_multi_vector.cc | 5 ++ .../neteq/audio_multi_vector_unittest.cc | 4 +- modules/audio_coding/neteq/audio_vector.cc | 3 + .../neteq/audio_vector_unittest.cc | 2 +- .../audio_coding/neteq/background_noise.cc | 5 ++ modules/audio_coding/neteq/background_noise.h | 1 + .../neteq/background_noise_unittest.cc | 2 + .../audio_coding/neteq/buffer_level_filter.cc | 1 + modules/audio_coding/neteq/comfort_noise.cc | 2 +- .../neteq/comfort_noise_unittest.cc | 1 + .../audio_coding/neteq/cross_correlation.cc | 3 +- .../neteq/decision_logic_unittest.cc | 6 ++ .../audio_coding/neteq/decoder_database.cc | 11 ++- modules/audio_coding/neteq/decoder_database.h | 6 ++ .../neteq/decoder_database_unittest.cc | 8 +++ modules/audio_coding/neteq/dsp_helper.cc | 4 ++ .../audio_coding/neteq/dsp_helper_unittest.cc | 2 + modules/audio_coding/neteq/dtmf_buffer.cc | 2 + .../neteq/dtmf_buffer_unittest.cc | 6 +- .../audio_coding/neteq/dtmf_tone_generator.cc | 4 ++ .../neteq/dtmf_tone_generator_unittest.cc | 2 + modules/audio_coding/neteq/expand.h | 3 + .../audio_coding/neteq/expand_uma_logger.cc | 5 ++ modules/audio_coding/neteq/expand_unittest.cc | 6 +- modules/audio_coding/neteq/histogram.cc | 3 +- .../audio_coding/neteq/histogram_unittest.cc | 2 + modules/audio_coding/neteq/merge.cc | 6 ++ modules/audio_coding/neteq/merge.h | 4 ++ modules/audio_coding/neteq/merge_unittest.cc | 5 +- .../neteq/mock/mock_buffer_level_filter.h | 2 + .../neteq/mock/mock_decoder_database.h | 6 +- .../neteq/mock/mock_dtmf_buffer.h | 3 + .../neteq/mock/mock_dtmf_tone_generator.h | 3 + modules/audio_coding/neteq/mock/mock_expand.h | 2 + .../audio_coding/neteq/mock/mock_histogram.h | 2 + .../neteq/mock/mock_neteq_controller.h | 5 ++ .../neteq/mock/mock_packet_arrival_history.h | 2 + .../neteq/mock/mock_packet_buffer.h | 5 ++ .../neteq/mock/mock_red_payload_splitter.h | 1 + .../neteq/mock/mock_statistics_calculator.h | 3 + modules/audio_coding/neteq/nack_tracker.cc | 4 ++ .../neteq/nack_tracker_unittest.cc | 3 +- .../neteq/neteq_decoder_plc_unittest.cc | 14 +++- modules/audio_coding/neteq/neteq_impl.h | 1 + .../audio_coding/neteq/neteq_impl_unittest.cc | 21 ++++-- .../neteq/neteq_network_stats_unittest.cc | 15 +++- .../neteq/neteq_stereo_unittest.cc | 8 +++ modules/audio_coding/neteq/neteq_unittest.cc | 13 ++-- modules/audio_coding/neteq/normal.cc | 7 ++ modules/audio_coding/neteq/normal_unittest.cc | 8 ++- modules/audio_coding/neteq/packet.cc | 2 + modules/audio_coding/neteq/packet.h | 1 + .../neteq/packet_arrival_history_unittest.cc | 1 + modules/audio_coding/neteq/packet_buffer.cc | 6 +- modules/audio_coding/neteq/packet_buffer.h | 2 + .../neteq/packet_buffer_unittest.cc | 11 ++- .../audio_coding/neteq/preemptive_expand.cc | 2 + modules/audio_coding/neteq/random_vector.cc | 3 + .../neteq/red_payload_splitter.cc | 1 + .../audio_coding/neteq/red_payload_splitter.h | 2 + .../neteq/red_payload_splitter_unittest.cc | 9 ++- .../audio_coding/neteq/reorder_optimizer.cc | 1 + .../neteq/statistics_calculator.cc | 4 +- .../neteq/statistics_calculator.h | 3 + .../neteq/statistics_calculator_unittest.cc | 4 ++ modules/audio_coding/neteq/sync_buffer.cc | 5 ++ .../neteq/sync_buffer_unittest.cc | 5 ++ .../neteq/test/neteq_decoding_test.cc | 14 +++- .../neteq/test/neteq_decoding_test.h | 3 +- .../neteq/test/neteq_opus_quality_test.cc | 10 +++ .../neteq/test/neteq_pcm16b_quality_test.cc | 7 ++ .../neteq/test/neteq_pcmu_quality_test.cc | 7 ++ .../neteq/test/neteq_performance_unittest.cc | 2 + .../neteq/test/neteq_speed_test.cc | 2 + modules/audio_coding/neteq/time_stretch.cc | 5 ++ modules/audio_coding/neteq/time_stretch.h | 3 + .../neteq/time_stretch_unittest.cc | 5 +- .../audio_coding/neteq/timestamp_scaler.cc | 3 + modules/audio_coding/neteq/timestamp_scaler.h | 2 + .../neteq/timestamp_scaler_unittest.cc | 6 ++ .../audio_coding/neteq/tools/audio_loop.cc | 4 ++ modules/audio_coding/neteq/tools/audio_loop.h | 3 +- .../audio_coding/neteq/tools/audio_sink.cc | 3 + modules/audio_coding/neteq/tools/audio_sink.h | 3 + .../neteq/tools/constant_pcm_packet_source.cc | 6 +- .../neteq/tools/constant_pcm_packet_source.h | 4 +- .../neteq/tools/encode_neteq_input.cc | 7 ++ .../neteq/tools/encode_neteq_input.h | 5 ++ .../neteq/tools/fake_decode_from_file.cc | 11 +++ .../neteq/tools/fake_decode_from_file.h | 5 ++ .../initial_packet_inserter_neteq_input.cc | 6 +- .../initial_packet_inserter_neteq_input.h | 5 +- .../neteq/tools/input_audio_file.cc | 5 ++ .../neteq/tools/input_audio_file.h | 1 - .../neteq/tools/input_audio_file_unittest.cc | 3 + .../neteq/tools/neteq_delay_analyzer.cc | 10 ++- .../neteq/tools/neteq_delay_analyzer.h | 6 +- .../neteq/tools/neteq_event_log_input.cc | 15 ++-- .../neteq/tools/neteq_event_log_input.h | 3 +- .../audio_coding/neteq/tools/neteq_input.cc | 7 ++ .../audio_coding/neteq/tools/neteq_input.h | 4 +- .../neteq/tools/neteq_performance_test.cc | 8 +++ .../neteq/tools/neteq_quality_test.cc | 21 +++++- .../neteq/tools/neteq_quality_test.h | 9 ++- .../neteq/tools/neteq_replacement_input.cc | 9 +++ .../neteq/tools/neteq_replacement_input.h | 3 + .../neteq/tools/neteq_rtp_dump_input.cc | 10 +++ .../neteq/tools/neteq_rtp_dump_input.h | 1 + .../neteq/tools/neteq_stats_getter.cc | 6 ++ .../neteq/tools/neteq_stats_getter.h | 5 ++ .../neteq/tools/neteq_stats_plotter.cc | 5 ++ .../neteq/tools/neteq_stats_plotter.h | 2 +- .../audio_coding/neteq/tools/neteq_test.cc | 19 +++++ modules/audio_coding/neteq/tools/neteq_test.h | 6 +- .../neteq/tools/output_audio_file.h | 2 + .../neteq/tools/output_wav_file.h | 3 +- modules/audio_coding/neteq/tools/packet.cc | 7 ++ modules/audio_coding/neteq/tools/packet.h | 2 + .../audio_coding/neteq/tools/packet_source.cc | 2 + .../audio_coding/neteq/tools/packet_source.h | 1 + .../neteq/tools/packet_unittest.cc | 7 ++ .../neteq/tools/resample_input_audio_file.cc | 3 + .../neteq/tools/resample_input_audio_file.h | 3 +- .../audio_coding/neteq/tools/rtp_analyze.cc | 6 ++ .../neteq/tools/rtp_file_source.h | 4 +- .../audio_coding/neteq/tools/rtp_generator.cc | 6 ++ .../audio_coding/neteq/tools/rtp_generator.h | 3 + .../audio_coding/neteq/tools/rtp_jitter.cc | 4 ++ modules/audio_coding/neteq/tools/rtpcat.cc | 1 + .../audio_coding/neteq/underrun_optimizer.cc | 3 + .../neteq/underrun_optimizer_unittest.cc | 3 + modules/audio_coding/test/Channel.cc | 11 ++- modules/audio_coding/test/Channel.h | 4 +- modules/audio_coding/test/EncodeDecodeTest.cc | 9 +++ modules/audio_coding/test/EncodeDecodeTest.h | 5 +- modules/audio_coding/test/PCMFile.cc | 5 ++ modules/audio_coding/test/PCMFile.h | 1 + modules/audio_coding/test/PacketLossTest.cc | 9 +++ modules/audio_coding/test/PacketLossTest.h | 5 ++ modules/audio_coding/test/RTPFile.cc | 9 ++- modules/audio_coding/test/RTPFile.h | 1 + modules/audio_coding/test/TestAllCodecs.cc | 12 +++- modules/audio_coding/test/TestAllCodecs.h | 3 + modules/audio_coding/test/TestRedFec.cc | 13 +++- modules/audio_coding/test/TestRedFec.h | 6 +- modules/audio_coding/test/TestStereo.cc | 17 ++++- modules/audio_coding/test/TestStereo.h | 3 + modules/audio_coding/test/TestVADDTX.cc | 16 +++++ modules/audio_coding/test/TestVADDTX.h | 5 ++ modules/audio_coding/test/Tester.cc | 4 -- modules/audio_coding/test/opus_test.cc | 7 ++ modules/audio_coding/test/opus_test.h | 4 +- .../test/target_delay_unittest.cc | 7 ++ 252 files changed, 1265 insertions(+), 130 deletions(-) diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn index 6a13b89c4e..31a21d3a7d 100644 --- a/modules/audio_coding/BUILD.gn +++ b/modules/audio_coding/BUILD.gn @@ -100,9 +100,11 @@ rtc_library("audio_encoder_cng") { deps = [ ":webrtc_cng", + "../../api:array_view", "../../api/audio_codecs:audio_codecs_api", "../../api/units:time_delta", "../../common_audio", + "../../rtc_base:buffer", "../../rtc_base:checks", ] } @@ -116,6 +118,7 @@ rtc_library("red") { deps = [ "../../api:array_view", + "../../api:bitrate_allocation", "../../api:field_trials_view", "../../api/audio_codecs:audio_codecs_api", "../../api/units:time_delta", @@ -259,6 +262,7 @@ rtc_library("audio_coding_opus_common") { deps = [ "../../api:array_view", "../../api/audio_codecs:audio_codecs_api", + "../../rtc_base:buffer", "../../rtc_base:checks", "../../rtc_base:stringutils", "//third_party/abseil-cpp/absl/strings:string_view", @@ -279,10 +283,12 @@ rtc_library("webrtc_opus") { ":audio_coding_opus_common", ":audio_network_adaptor", "../../api:array_view", + "../../api:bitrate_allocation", "../../api:field_trials_view", "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs/opus:audio_encoder_opus_config", "../../api/environment", + "../../api/units:time_delta", "../../common_audio", "../../rtc_base:buffer", "../../rtc_base:checks", @@ -297,6 +303,7 @@ rtc_library("webrtc_opus") { "//third_party/abseil-cpp/absl/memory", "//third_party/abseil-cpp/absl/strings", "//third_party/abseil-cpp/absl/strings:string_view", + "//third_party/opus", ] public_deps += # no-presubmit-check TODO(webrtc:8603) [ ":webrtc_opus_wrapper" ] @@ -416,6 +423,7 @@ rtc_library("audio_network_adaptor") { [ ":audio_network_adaptor_config" ] deps = [ + "../../api:array_view", "../../api/audio_codecs:audio_codecs_api", "../../api/rtc_event_log", "../../common_audio", @@ -577,6 +585,7 @@ rtc_library("neteq_tools_minimal") { "../../api:field_trials", "../../api:neteq_simulator_api", "../../api:rtp_headers", + "../../api:scoped_refptr", "../../api/audio:audio_frame_api", "../../api/audio_codecs:audio_codecs_api", "../../api/environment", @@ -593,6 +602,7 @@ rtc_library("neteq_tools_minimal") { "../../rtc_base:stringutils", "../../system_wrappers", "../rtp_rtcp:rtp_rtcp_format", + "//third_party/abseil-cpp/absl/strings:string_view", ] defines = audio_codec_defines } @@ -657,7 +667,14 @@ rtc_library("neteq_tools") { ":neteq_tools_minimal", "..:module_api_public", "../../api:array_view", + "../../api:rtp_headers", + "../../api:rtp_packet_info", + "../../api/audio:audio_frame_api", "../../api/audio_codecs:audio_codecs_api", + "../../api/neteq:neteq_api", + "../../logging:rtc_event_audio", + "../../logging:rtc_event_rtp_rtcp", + "../../rtc_base:buffer", "../../rtc_base:checks", "../../rtc_base:rtc_numerics", "../../rtc_base:safe_conversions", @@ -732,8 +749,11 @@ rtc_library("audio_coding_modules_tests_shared") { ":neteq_tools_minimal", ":webrtc_opus_wrapper", "..:module_api", + "../../api:array_view", "../../api:rtp_headers", + "../../api:rtp_parameters", "../../api/audio:audio_frame_api", + "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs:builtin_audio_decoder_factory", "../../api/audio_codecs:builtin_audio_encoder_factory", "../../api/environment", @@ -785,6 +805,7 @@ if (rtc_include_tests) { ] deps = [ ":audio_network_adaptor", + "../../api/audio_codecs:audio_codecs_api", "../../test:test_support", ] } @@ -849,7 +870,9 @@ if (rtc_include_tests) { ":red", ":webrtc_opus_wrapper", "..:module_api", + "../../api:array_view", "../../api:rtp_headers", + "../../api:scoped_refptr", "../../api/audio:audio_frame_api", "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs:builtin_audio_decoder_factory", @@ -869,6 +892,7 @@ if (rtc_include_tests) { "../../api/units:timestamp", "../../common_audio", "../../rtc_base:checks", + "../../rtc_base:ip_address", "../../rtc_base:logging", "../../rtc_base:macromagic", "../../rtc_base:stringutils", @@ -894,10 +918,14 @@ if (rtc_include_tests) { deps = [ ":neteq_test_support", ":neteq_test_tools", + "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs/opus:audio_encoder_opus", + "../../api/audio_codecs/opus:audio_encoder_opus_config", + "../../api/environment", "../../api/environment:environment_factory", "../../api/test/metrics:global_metrics_logger_and_exporter", "../../api/test/metrics:metric", + "../../rtc_base:buffer", "../../rtc_base:macromagic", "../../rtc_base:timeutils", "../../system_wrappers", @@ -921,6 +949,7 @@ if (rtc_include_tests) { ":audio_coding", ":neteq_tools", ":neteq_tools_minimal", + "../../api:array_view", "../../api:scoped_refptr", "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs:builtin_audio_decoder_factory", @@ -945,6 +974,7 @@ if (rtc_include_tests) { deps = [ ":audio_coding", + ":audio_coding_module_typedefs", ":neteq_input_audio_tools", ":neteq_tools", ":neteq_tools_minimal", @@ -955,6 +985,7 @@ if (rtc_include_tests) { "../../api/environment", "../../api/environment:environment_factory", "../../rtc_base:checks", + "../../rtc_base:copy_on_write_buffer", "../../rtc_base:stringutils", "../../test:test_support", "//testing/gtest", @@ -986,10 +1017,15 @@ if (rtc_include_tests) { ":neteq", ":neteq_input_audio_tools", ":neteq_tools", + "../../api:array_view", "../../api/audio_codecs:audio_codecs_api", + "../../api/audio_codecs/g722:audio_encoder_g722_config", "../../api/audio_codecs/opus:audio_encoder_opus", + "../../api/audio_codecs/opus:audio_encoder_opus_config", "../../api/environment:environment_factory", "../../common_audio", + "../../rtc_base:buffer", + "../../rtc_base:checks", "../../rtc_base/system:arch", "../../test:explicit_key_value_config", "../../test:fileutils", @@ -1088,6 +1124,7 @@ if (rtc_include_tests) { deps = [ "../../rtc_base:macromagic", "../../test:fileutils", + "//third_party/opus", ] sources = [ "codecs/opus/opus_speed_test.cc", @@ -1129,6 +1166,7 @@ if (rtc_include_tests) { ":neteq", ":neteq_test_tools", ":pcm16b", + "../../api:rtp_headers", "../../api/audio:audio_frame_api", "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs:builtin_audio_decoder_factory", @@ -1136,6 +1174,7 @@ if (rtc_include_tests) { "../../api/environment:environment_factory", "../../api/neteq:default_neteq_factory", "../../api/neteq:neteq_api", + "../../api/units:timestamp", "../../rtc_base:checks", "../../system_wrappers", "../../test:fileutils", @@ -1157,11 +1196,16 @@ if (rtc_include_tests) { ":neteq_input_audio_tools", ":neteq_test_tools", ":neteq_tools_minimal", + "../../api:array_view", + "../../api:rtp_headers", + "../../api:scoped_refptr", + "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs:builtin_audio_decoder_factory", "../../api/environment:environment_factory", "../../api/neteq:default_neteq_factory", "../../api/neteq:neteq_api", "../../api/units:timestamp", + "../../rtc_base:buffer", "../../rtc_base:checks", "../../rtc_base:stringutils", "../../system_wrappers", @@ -1207,6 +1251,7 @@ if (rtc_include_tests) { deps = [ "../../api:array_view", "../../rtc_base:buffer", + "../../rtc_base:checks", "../rtp_rtcp:rtp_rtcp_format", ] @@ -1239,6 +1284,9 @@ if (rtc_include_tests) { ":neteq_test_tools", ":neteq_tools_minimal", ":pcm16b", + "../../api:rtp_headers", + "../../rtc_base:checks", + "../rtp_rtcp:rtp_rtcp_format", "//testing/gtest", "//third_party/abseil-cpp/absl/flags:flag", "//third_party/abseil-cpp/absl/flags:parse", @@ -1255,9 +1303,16 @@ if (rtc_include_tests) { ":neteq_quality_test_support", ":neteq_tools", ":webrtc_opus", + "../../api:array_view", + "../../api:rtp_parameters", + "../../api/audio_codecs:audio_codecs_api", + "../../rtc_base:buffer", + "../../rtc_base:checks", "../../test:test_main", + "../../test:test_support", "//testing/gtest", "//third_party/abseil-cpp/absl/flags:flag", + "//third_party/opus", ] } @@ -1285,10 +1340,14 @@ if (rtc_include_tests) { ":g711", ":neteq", ":neteq_quality_test_support", + "../../api:array_view", + "../../api/audio_codecs:audio_codecs_api", + "../../rtc_base:buffer", "../../rtc_base:checks", "../../rtc_base:safe_conversions", "../../test:fileutils", "../../test:test_main", + "../../test:test_support", "//testing/gtest", "//third_party/abseil-cpp/absl/flags:flag", ] @@ -1303,10 +1362,14 @@ if (rtc_include_tests) { ":neteq", ":neteq_quality_test_support", ":pcm16b", + "../../api:array_view", + "../../api/audio_codecs:audio_codecs_api", + "../../rtc_base:buffer", "../../rtc_base:checks", "../../rtc_base:safe_conversions", "../../test:fileutils", "../../test:test_main", + "../../test:test_support", "//testing/gtest", "//third_party/abseil-cpp/absl/flags:flag", ] @@ -1449,7 +1512,12 @@ if (rtc_include_tests) { "..:module_api", "..:module_api_public", "../../api:array_view", + "../../api:bitrate_allocation", + "../../api:field_trials_view", "../../api:make_ref_counted", + "../../api:rtp_headers", + "../../api:rtp_packet_info", + "../../api:rtp_parameters", "../../api:scoped_refptr", "../../api/audio:audio_frame_api", "../../api/audio_codecs:audio_codecs_api", @@ -1469,6 +1537,7 @@ if (rtc_include_tests) { "../../api/neteq:tick_timer", "../../api/neteq:tick_timer_unittest", "../../api/rtc_event_log", + "../../api/units:data_rate", "../../api/units:time_delta", "../../api/units:timestamp", "../../common_audio", @@ -1479,7 +1548,9 @@ if (rtc_include_tests) { "../../modules/rtp_rtcp:rtp_rtcp_format", "../../rtc_base:buffer", "../../rtc_base:checks", + "../../rtc_base:copy_on_write_buffer", "../../rtc_base:digest", + "../../rtc_base:ip_address", "../../rtc_base:macromagic", "../../rtc_base:platform_thread", "../../rtc_base:random", @@ -1510,6 +1581,7 @@ if (rtc_include_tests) { "//third_party/abseil-cpp/absl/flags:flag", "//third_party/abseil-cpp/absl/memory", "//third_party/abseil-cpp/absl/strings:string_view", + "//third_party/opus", ] defines = audio_coding_defines diff --git a/modules/audio_coding/acm2/acm_receive_test.cc b/modules/audio_coding/acm2/acm_receive_test.cc index bf1fa7f1fe..0915ca3cb0 100644 --- a/modules/audio_coding/acm2/acm_receive_test.cc +++ b/modules/audio_coding/acm2/acm_receive_test.cc @@ -12,11 +12,18 @@ #include +#include #include +#include +#include +#include "api/array_view.h" +#include "api/audio_codecs/audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/environment/environment_factory.h" #include "api/neteq/default_neteq_factory.h" +#include "api/neteq/neteq.h" +#include "api/scoped_refptr.h" #include "modules/audio_coding/include/audio_coding_module.h" #include "modules/audio_coding/neteq/tools/audio_sink.h" #include "modules/audio_coding/neteq/tools/packet.h" diff --git a/modules/audio_coding/acm2/acm_receive_test.h b/modules/audio_coding/acm2/acm_receive_test.h index 9cc9019591..f1958124ed 100644 --- a/modules/audio_coding/acm2/acm_receive_test.h +++ b/modules/audio_coding/acm2/acm_receive_test.h @@ -13,8 +13,8 @@ #include // for size_t +#include #include -#include #include "api/audio_codecs/audio_decoder_factory.h" #include "api/neteq/neteq.h" diff --git a/modules/audio_coding/acm2/acm_remixing.cc b/modules/audio_coding/acm2/acm_remixing.cc index 45eb3c1247..3ad995542b 100644 --- a/modules/audio_coding/acm2/acm_remixing.cc +++ b/modules/audio_coding/acm2/acm_remixing.cc @@ -10,7 +10,15 @@ #include "modules/audio_coding/acm2/acm_remixing.h" +#include +#include +#include +#include + +#include "api/array_view.h" +#include "api/audio/audio_frame.h" #include "rtc_base/checks.h" +#include "rtc_base/numerics/safe_conversions.h" namespace webrtc { diff --git a/modules/audio_coding/acm2/acm_remixing.h b/modules/audio_coding/acm2/acm_remixing.h index c6ae2c3844..0ccb7c0bf4 100644 --- a/modules/audio_coding/acm2/acm_remixing.h +++ b/modules/audio_coding/acm2/acm_remixing.h @@ -11,6 +11,8 @@ #ifndef MODULES_AUDIO_CODING_ACM2_ACM_REMIXING_H_ #define MODULES_AUDIO_CODING_ACM2_ACM_REMIXING_H_ +#include +#include #include #include "api/array_view.h" diff --git a/modules/audio_coding/acm2/acm_remixing_unittest.cc b/modules/audio_coding/acm2/acm_remixing_unittest.cc index 7c0c0ef665..dfc3e50ad8 100644 --- a/modules/audio_coding/acm2/acm_remixing_unittest.cc +++ b/modules/audio_coding/acm2/acm_remixing_unittest.cc @@ -10,10 +10,12 @@ #include "modules/audio_coding/acm2/acm_remixing.h" +#include +#include #include #include "api/audio/audio_frame.h" -#include "system_wrappers/include/clock.h" +#include "api/audio/audio_view.h" #include "test/gmock.h" #include "test/gtest.h" #include "test/testsupport/file_utils.h" diff --git a/modules/audio_coding/acm2/acm_resampler.cc b/modules/audio_coding/acm2/acm_resampler.cc index 85f22ac6ef..1425bc94df 100644 --- a/modules/audio_coding/acm2/acm_resampler.cc +++ b/modules/audio_coding/acm2/acm_resampler.cc @@ -12,8 +12,13 @@ #include +#include + #include "api/audio/audio_frame.h" +#include "api/audio/audio_view.h" +#include "rtc_base/checks.h" #include "rtc_base/logging.h" +#include "rtc_base/numerics/safe_conversions.h" namespace webrtc { namespace acm2 { diff --git a/modules/audio_coding/acm2/acm_resampler.h b/modules/audio_coding/acm2/acm_resampler.h index cf7133c38d..244e215df5 100644 --- a/modules/audio_coding/acm2/acm_resampler.h +++ b/modules/audio_coding/acm2/acm_resampler.h @@ -14,6 +14,8 @@ #include #include +#include + #include "api/audio/audio_frame.h" #include "common_audio/resampler/include/push_resampler.h" diff --git a/modules/audio_coding/acm2/acm_send_test.cc b/modules/audio_coding/acm2/acm_send_test.cc index c79bded007..fa635f9522 100644 --- a/modules/audio_coding/acm2/acm_send_test.cc +++ b/modules/audio_coding/acm2/acm_send_test.cc @@ -13,18 +13,23 @@ #include #include +#include +#include +#include + #include "absl/strings/match.h" +#include "absl/strings/str_cat.h" #include "absl/strings/string_view.h" #include "api/audio_codecs/audio_encoder.h" -#include "api/audio_codecs/builtin_audio_decoder_factory.h" +#include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" #include "api/environment/environment_factory.h" #include "modules/audio_coding/include/audio_coding_module.h" +#include "modules/audio_coding/include/audio_coding_module_typedefs.h" #include "modules/audio_coding/neteq/tools/input_audio_file.h" #include "modules/audio_coding/neteq/tools/packet.h" #include "rtc_base/checks.h" -#include "rtc_base/string_encode.h" -#include "test/gtest.h" +#include "rtc_base/copy_on_write_buffer.h" namespace webrtc { namespace test { diff --git a/modules/audio_coding/acm2/acm_send_test.h b/modules/audio_coding/acm2/acm_send_test.h index 1bba67618b..694c1f92ed 100644 --- a/modules/audio_coding/acm2/acm_send_test.h +++ b/modules/audio_coding/acm2/acm_send_test.h @@ -11,6 +11,8 @@ #ifndef MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_ #define MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_ +#include +#include #include #include @@ -18,6 +20,7 @@ #include "api/audio/audio_frame.h" #include "api/environment/environment.h" #include "modules/audio_coding/include/audio_coding_module.h" +#include "modules/audio_coding/include/audio_coding_module_typedefs.h" #include "modules/audio_coding/neteq/tools/packet_source.h" #include "system_wrappers/include/clock.h" diff --git a/modules/audio_coding/acm2/call_statistics.cc b/modules/audio_coding/acm2/call_statistics.cc index 9f3bdadc88..de3e3cc070 100644 --- a/modules/audio_coding/acm2/call_statistics.cc +++ b/modules/audio_coding/acm2/call_statistics.cc @@ -10,6 +10,8 @@ #include "modules/audio_coding/acm2/call_statistics.h" +#include "api/audio/audio_frame.h" +#include "modules/audio_coding/include/audio_coding_module_typedefs.h" #include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/acm2/call_statistics_unittest.cc b/modules/audio_coding/acm2/call_statistics_unittest.cc index b96977b8e4..4cafa90699 100644 --- a/modules/audio_coding/acm2/call_statistics_unittest.cc +++ b/modules/audio_coding/acm2/call_statistics_unittest.cc @@ -10,6 +10,8 @@ #include "modules/audio_coding/acm2/call_statistics.h" +#include "api/audio/audio_frame.h" +#include "modules/audio_coding/include/audio_coding_module_typedefs.h" #include "test/gtest.h" namespace webrtc { diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc index ddcf6c8060..74979843fa 100644 --- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc +++ b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc @@ -12,15 +12,20 @@ #include +#include +#include +#include +#include #include -#include +#include "api/audio_codecs/audio_encoder.h" +#include "modules/audio_coding/audio_network_adaptor/controller.h" #include "modules/audio_coding/audio_network_adaptor/controller_manager.h" #include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h" #include "modules/audio_coding/audio_network_adaptor/event_log_writer.h" +#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" #include "rtc_base/checks.h" #include "rtc_base/time_utils.h" -#include "system_wrappers/include/field_trial.h" namespace webrtc { diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc index 59c5eb8714..ddabcfeba7 100644 --- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc +++ b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc @@ -10,17 +10,23 @@ #include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h" +#include +#include +#include #include #include #include "api/rtc_event_log/rtc_event.h" +#include "api/units/time_delta.h" #include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h" #include "logging/rtc_event_log/mock/mock_rtc_event_log.h" +#include "modules/audio_coding/audio_network_adaptor/controller.h" #include "modules/audio_coding/audio_network_adaptor/mock/mock_controller.h" #include "modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h" #include "modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h" #include "rtc_base/fake_clock.h" #include "test/field_trial.h" +#include "test/gmock.h" #include "test/gtest.h" namespace webrtc { diff --git a/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc b/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc index 88ca38d074..63aed54d8d 100644 --- a/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc +++ b/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc @@ -12,8 +12,8 @@ #include +#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" #include "rtc_base/checks.h" -#include "system_wrappers/include/field_trial.h" namespace webrtc { namespace audio_network_adaptor { diff --git a/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc index b142915c24..304cb0d517 100644 --- a/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc +++ b/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc @@ -10,8 +10,12 @@ #include "modules/audio_coding/audio_network_adaptor/bitrate_controller.h" +#include +#include + +#include "modules/audio_coding/audio_network_adaptor/controller.h" +#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" #include "rtc_base/numerics/safe_conversions.h" -#include "test/field_trial.h" #include "test/gtest.h" namespace webrtc { diff --git a/modules/audio_coding/audio_network_adaptor/channel_controller.cc b/modules/audio_coding/audio_network_adaptor/channel_controller.cc index 2ef2f4c4d8..73d3e39e2a 100644 --- a/modules/audio_coding/audio_network_adaptor/channel_controller.cc +++ b/modules/audio_coding/audio_network_adaptor/channel_controller.cc @@ -11,7 +11,9 @@ #include "modules/audio_coding/audio_network_adaptor/channel_controller.h" #include +#include +#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" #include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc index 1803d449ed..003ab0dd6d 100644 --- a/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc +++ b/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc @@ -10,8 +10,12 @@ #include "modules/audio_coding/audio_network_adaptor/channel_controller.h" +#include #include +#include +#include "modules/audio_coding/audio_network_adaptor/controller.h" +#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" #include "test/gtest.h" namespace webrtc { diff --git a/modules/audio_coding/audio_network_adaptor/controller.h b/modules/audio_coding/audio_network_adaptor/controller.h index fc2cba78ed..80b0435bfb 100644 --- a/modules/audio_coding/audio_network_adaptor/controller.h +++ b/modules/audio_coding/audio_network_adaptor/controller.h @@ -11,9 +11,10 @@ #ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_ #define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_ +#include #include -#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h" +#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" namespace webrtc { diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager.cc b/modules/audio_coding/audio_network_adaptor/controller_manager.cc index 599fcc3653..c126897bfb 100644 --- a/modules/audio_coding/audio_network_adaptor/controller_manager.cc +++ b/modules/audio_coding/audio_network_adaptor/controller_manager.cc @@ -10,21 +10,30 @@ #include "modules/audio_coding/audio_network_adaptor/controller_manager.h" +#include #include +#include +#include +#include #include +#include +#include #include #include +#include #include "absl/strings/string_view.h" +#include "api/array_view.h" #include "modules/audio_coding/audio_network_adaptor/bitrate_controller.h" #include "modules/audio_coding/audio_network_adaptor/channel_controller.h" +#include "modules/audio_coding/audio_network_adaptor/controller.h" #include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h" #include "modules/audio_coding/audio_network_adaptor/dtx_controller.h" #include "modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h" #include "modules/audio_coding/audio_network_adaptor/frame_length_controller.h" #include "modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h" #include "modules/audio_coding/audio_network_adaptor/util/threshold_curve.h" -#include "rtc_base/logging.h" +#include "rtc_base/checks.h" #include "rtc_base/time_utils.h" #if WEBRTC_ENABLE_PROTOBUF diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager.h b/modules/audio_coding/audio_network_adaptor/controller_manager.h index 632cc616a3..7fd617dc69 100644 --- a/modules/audio_coding/audio_network_adaptor/controller_manager.h +++ b/modules/audio_coding/audio_network_adaptor/controller_manager.h @@ -11,12 +11,16 @@ #ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_MANAGER_H_ #define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_MANAGER_H_ +#include +#include #include #include -#include +#include +#include #include #include "absl/strings/string_view.h" +#include "api/array_view.h" #include "modules/audio_coding/audio_network_adaptor/controller.h" namespace webrtc { diff --git a/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc b/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc index 4504e04088..21d0aa750b 100644 --- a/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc +++ b/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc @@ -10,17 +10,24 @@ #include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h" +#include +#include +#include #include #include +#include "modules/audio_coding/audio_network_adaptor/controller.h" +#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" #include "rtc_base/checks.h" #include "rtc_base/numerics/safe_conversions.h" #include "rtc_base/system/file_wrapper.h" #if WEBRTC_ENABLE_PROTOBUF #ifdef WEBRTC_ANDROID_PLATFORM_BUILD +#include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h" #include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.pb.h" #else +#include "modules/audio_coding/audio_network_adaptor/config.pb.h" #include "modules/audio_coding/audio_network_adaptor/debug_dump.pb.h" #endif #endif diff --git a/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h b/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h index fd3a64dbb1..93effbd905 100644 --- a/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h +++ b/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h @@ -11,11 +11,12 @@ #ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP_WRITER_H_ #define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP_WRITER_H_ +#include +#include #include #include "modules/audio_coding/audio_network_adaptor/controller.h" -#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h" -#include "rtc_base/system/file_wrapper.h" +#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" #if WEBRTC_ENABLE_PROTOBUF #ifdef WEBRTC_ANDROID_PLATFORM_BUILD diff --git a/modules/audio_coding/audio_network_adaptor/dtx_controller.cc b/modules/audio_coding/audio_network_adaptor/dtx_controller.cc index b0a7d5d59d..e589657c98 100644 --- a/modules/audio_coding/audio_network_adaptor/dtx_controller.cc +++ b/modules/audio_coding/audio_network_adaptor/dtx_controller.cc @@ -10,6 +10,7 @@ #include "modules/audio_coding/audio_network_adaptor/dtx_controller.h" +#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" #include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc index 46c8f8b844..d92ee6cd81 100644 --- a/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc +++ b/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc @@ -11,7 +11,10 @@ #include "modules/audio_coding/audio_network_adaptor/dtx_controller.h" #include +#include +#include "modules/audio_coding/audio_network_adaptor/controller.h" +#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" #include "test/gtest.h" namespace webrtc { diff --git a/modules/audio_coding/audio_network_adaptor/event_log_writer.cc b/modules/audio_coding/audio_network_adaptor/event_log_writer.cc index 1bef64eaa6..3e12125bc8 100644 --- a/modules/audio_coding/audio_network_adaptor/event_log_writer.cc +++ b/modules/audio_coding/audio_network_adaptor/event_log_writer.cc @@ -18,7 +18,6 @@ #include #include -#include "api/rtc_event_log/rtc_event.h" #include "api/rtc_event_log/rtc_event_log.h" #include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h" #include "rtc_base/checks.h" diff --git a/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc b/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc index 2c344534ea..1ae5794505 100644 --- a/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc +++ b/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc @@ -10,11 +10,14 @@ #include "modules/audio_coding/audio_network_adaptor/event_log_writer.h" +#include #include +#include "api/rtc_event_log/rtc_event.h" #include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h" #include "logging/rtc_event_log/mock/mock_rtc_event_log.h" #include "rtc_base/checks.h" +#include "test/gmock.h" #include "test/gtest.h" namespace webrtc { diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc index bcfbd75c61..5db593abd3 100644 --- a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc +++ b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc @@ -10,9 +10,13 @@ #include "modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h" -#include +#include +#include #include +#include "common_audio/smoothing_filter.h" +#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" +#include "modules/audio_coding/audio_network_adaptor/util/threshold_curve.h" #include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc b/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc index c3ec2d7072..40f98482ad 100644 --- a/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc +++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc @@ -11,9 +11,13 @@ #include "modules/audio_coding/audio_network_adaptor/frame_length_controller.h" #include +#include #include +#include +#include #include +#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" #include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc index 2bf52b21aa..926e4c6f20 100644 --- a/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc +++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc @@ -10,9 +10,14 @@ #include "modules/audio_coding/audio_network_adaptor/frame_length_controller.h" +#include +#include #include -#include +#include +#include +#include "modules/audio_coding/audio_network_adaptor/controller.h" +#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" #include "test/gtest.h" namespace webrtc { diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.cc b/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.cc index f6f4759ba1..bbb0a2de29 100644 --- a/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.cc +++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.cc @@ -10,9 +10,9 @@ #include "modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h" -#include - #include "absl/algorithm/container.h" +#include "api/array_view.h" +#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" #include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h b/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h index c64c75f051..e8ff5adfde 100644 --- a/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h +++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h @@ -14,8 +14,9 @@ #include #include +#include "api/array_view.h" #include "modules/audio_coding/audio_network_adaptor/controller.h" -#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h" +#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" namespace webrtc { diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2_unittest.cc b/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2_unittest.cc index 1bec1acfc9..e038b48af3 100644 --- a/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2_unittest.cc +++ b/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2_unittest.cc @@ -10,8 +10,8 @@ #include "modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h" -#include #include +#include #include "modules/audio_coding/audio_network_adaptor/controller.h" #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" diff --git a/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h b/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h index ed07298e8e..7e5ab54ec0 100644 --- a/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h +++ b/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h @@ -11,7 +11,8 @@ #ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_ #define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_ -#include +#include +#include #include "api/audio_codecs/audio_encoder.h" #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" diff --git a/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h b/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h index 26a9061745..9d0fbab246 100644 --- a/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h +++ b/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h @@ -11,7 +11,12 @@ #ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_AUDIO_NETWORK_ADAPTOR_H_ #define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_AUDIO_NETWORK_ADAPTOR_H_ +#include +#include + +#include "api/audio_codecs/audio_encoder.h" #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h" +#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" #include "test/gmock.h" namespace webrtc { diff --git a/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h b/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h index de554c0517..d7624f3974 100644 --- a/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h +++ b/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h @@ -12,6 +12,7 @@ #define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_H_ #include "modules/audio_coding/audio_network_adaptor/controller.h" +#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" #include "test/gmock.h" namespace webrtc { diff --git a/modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h b/modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h index 9e2fa466fc..71194b0874 100644 --- a/modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h +++ b/modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h @@ -13,6 +13,7 @@ #include +#include "modules/audio_coding/audio_network_adaptor/controller.h" #include "modules/audio_coding/audio_network_adaptor/controller_manager.h" #include "test/gmock.h" diff --git a/modules/audio_coding/audio_network_adaptor/util/threshold_curve_unittest.cc b/modules/audio_coding/audio_network_adaptor/util/threshold_curve_unittest.cc index dc3aec0b18..09e66c9b79 100644 --- a/modules/audio_coding/audio_network_adaptor/util/threshold_curve_unittest.cc +++ b/modules/audio_coding/audio_network_adaptor/util/threshold_curve_unittest.cc @@ -12,6 +12,7 @@ #include +#include "rtc_base/checks.h" #include "test/gtest.h" // A threshold curve divides 2D space into three domains - below, on and above diff --git a/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc b/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc index 55ca237966..fe3adbd823 100644 --- a/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc +++ b/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc @@ -11,9 +11,18 @@ #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include +#include +#include +#include +#include +#include "api/audio_codecs/audio_decoder.h" +#include "api/audio_codecs/audio_decoder_factory.h" +#include "api/audio_codecs/audio_format.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" +#include "api/rtp_parameters.h" +#include "api/scoped_refptr.h" #include "test/gtest.h" namespace webrtc { diff --git a/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc b/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc index 78ba36fe85..f1250a29f4 100644 --- a/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc +++ b/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc @@ -10,12 +10,21 @@ #include "api/audio_codecs/builtin_audio_encoder_factory.h" +#include +#include #include #include +#include #include +#include "api/array_view.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/audio_codecs/audio_encoder_factory.h" +#include "api/audio_codecs/audio_format.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" +#include "api/scoped_refptr.h" +#include "rtc_base/buffer.h" #include "rtc_base/numerics/safe_conversions.h" #include "test/gmock.h" #include "test/gtest.h" diff --git a/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/modules/audio_coding/codecs/cng/audio_encoder_cng.cc index a4d0b9abd4..e93887606f 100644 --- a/modules/audio_coding/codecs/cng/audio_encoder_cng.cc +++ b/modules/audio_coding/codecs/cng/audio_encoder_cng.cc @@ -10,13 +10,19 @@ #include "modules/audio_coding/codecs/cng/audio_encoder_cng.h" +#include #include #include #include #include +#include +#include "api/array_view.h" +#include "api/audio_codecs/audio_encoder.h" #include "api/units/time_delta.h" +#include "common_audio/vad/include/vad.h" #include "modules/audio_coding/codecs/cng/webrtc_cng.h" +#include "rtc_base/buffer.h" #include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc b/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc index c6c2df7f0c..606a6b807c 100644 --- a/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc +++ b/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc @@ -10,11 +10,21 @@ #include "modules/audio_coding/codecs/cng/audio_encoder_cng.h" +#include +#include +#include #include -#include +#include +#include +#include "api/array_view.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/units/time_delta.h" +#include "common_audio/vad/include/vad.h" #include "common_audio/vad/mock/mock_vad.h" +#include "rtc_base/buffer.h" #include "rtc_base/numerics/safe_conversions.h" +#include "test/gmock.h" #include "test/gtest.h" #include "test/mock_audio_encoder.h" #include "test/testsupport/rtc_expect_death.h" diff --git a/modules/audio_coding/codecs/cng/cng_unittest.cc b/modules/audio_coding/codecs/cng/cng_unittest.cc index 76d11d413c..823506ea64 100644 --- a/modules/audio_coding/codecs/cng/cng_unittest.cc +++ b/modules/audio_coding/codecs/cng/cng_unittest.cc @@ -7,10 +7,15 @@ * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ -#include +#include +#include +#include #include +#include "api/array_view.h" #include "modules/audio_coding/codecs/cng/webrtc_cng.h" +#include "rtc_base/buffer.h" +#include "rtc_base/checks.h" #include "test/gtest.h" #include "test/testsupport/file_utils.h" diff --git a/modules/audio_coding/codecs/cng/webrtc_cng.cc b/modules/audio_coding/codecs/cng/webrtc_cng.cc index 4649f16d9e..5961cb0e9c 100644 --- a/modules/audio_coding/codecs/cng/webrtc_cng.cc +++ b/modules/audio_coding/codecs/cng/webrtc_cng.cc @@ -11,8 +11,12 @@ #include "modules/audio_coding/codecs/cng/webrtc_cng.h" #include +#include +#include +#include "api/array_view.h" #include "common_audio/signal_processing/include/signal_processing_library.h" +#include "rtc_base/buffer.h" #include "rtc_base/checks.h" #include "rtc_base/numerics/safe_conversions.h" diff --git a/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc b/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc index 4b97e11c02..f67ff2dfff 100644 --- a/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc +++ b/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc @@ -10,10 +10,16 @@ #include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h" +#include +#include #include +#include +#include "api/audio_codecs/audio_decoder.h" #include "modules/audio_coding/codecs/g711/g711_interface.h" #include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h" +#include "rtc_base/buffer.h" +#include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc index 38232e1c7b..5aa27d0e8e 100644 --- a/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc +++ b/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc @@ -10,9 +10,16 @@ #include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h" +#include #include +#include +#include +#include "api/array_view.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/units/time_delta.h" #include "modules/audio_coding/codecs/g711/g711_interface.h" +#include "rtc_base/buffer.h" #include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/codecs/g711/audio_encoder_pcm.h b/modules/audio_coding/codecs/g711/audio_encoder_pcm.h index 18a4770ec4..0d56e58d02 100644 --- a/modules/audio_coding/codecs/g711/audio_encoder_pcm.h +++ b/modules/audio_coding/codecs/g711/audio_encoder_pcm.h @@ -11,12 +11,16 @@ #ifndef MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_ #define MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_ +#include +#include #include #include #include +#include "api/array_view.h" #include "api/audio_codecs/audio_encoder.h" #include "api/units/time_delta.h" +#include "rtc_base/buffer.h" namespace webrtc { diff --git a/modules/audio_coding/codecs/g711/test/testG711.cc b/modules/audio_coding/codecs/g711/test/testG711.cc index f3a42f5d79..6117045fb4 100644 --- a/modules/audio_coding/codecs/g711/test/testG711.cc +++ b/modules/audio_coding/codecs/g711/test/testG711.cc @@ -21,6 +21,8 @@ /* Runtime statistics */ #include + +#include #define CLOCKS_PER_SEC_G711 1000 /* function for reading audio data from PCM file */ diff --git a/modules/audio_coding/codecs/g722/audio_decoder_g722.cc b/modules/audio_coding/codecs/g722/audio_decoder_g722.cc index cbffc488ec..0fbf91474f 100644 --- a/modules/audio_coding/codecs/g722/audio_decoder_g722.cc +++ b/modules/audio_coding/codecs/g722/audio_decoder_g722.cc @@ -12,10 +12,14 @@ #include +#include #include +#include +#include "api/audio_codecs/audio_decoder.h" #include "modules/audio_coding/codecs/g722/g722_interface.h" #include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h" +#include "rtc_base/buffer.h" #include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/codecs/g722/audio_decoder_g722.h b/modules/audio_coding/codecs/g722/audio_decoder_g722.h index 431d896b18..812f08c9c3 100644 --- a/modules/audio_coding/codecs/g722/audio_decoder_g722.h +++ b/modules/audio_coding/codecs/g722/audio_decoder_g722.h @@ -11,7 +11,12 @@ #ifndef MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_ #define MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_ +#include +#include +#include + #include "api/audio_codecs/audio_decoder.h" +#include "rtc_base/buffer.h" typedef struct WebRtcG722DecInst G722DecInst; diff --git a/modules/audio_coding/codecs/g722/audio_encoder_g722.cc b/modules/audio_coding/codecs/g722/audio_encoder_g722.cc index 699aeb47d1..078ba0a39b 100644 --- a/modules/audio_coding/codecs/g722/audio_encoder_g722.cc +++ b/modules/audio_coding/codecs/g722/audio_encoder_g722.cc @@ -10,11 +10,18 @@ #include "modules/audio_coding/codecs/g722/audio_encoder_g722.h" +#include #include +#include +#include +#include "api/array_view.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/audio_codecs/g722/audio_encoder_g722_config.h" +#include "api/units/time_delta.h" #include "modules/audio_coding/codecs/g722/g722_interface.h" +#include "rtc_base/buffer.h" #include "rtc_base/checks.h" -#include "rtc_base/numerics/safe_conversions.h" namespace webrtc { diff --git a/modules/audio_coding/codecs/g722/audio_encoder_g722.h b/modules/audio_coding/codecs/g722/audio_encoder_g722.h index 8a584e26e8..42ce125d47 100644 --- a/modules/audio_coding/codecs/g722/audio_encoder_g722.h +++ b/modules/audio_coding/codecs/g722/audio_encoder_g722.h @@ -11,10 +11,13 @@ #ifndef MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_ #define MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_ +#include +#include #include #include #include +#include "api/array_view.h" #include "api/audio_codecs/audio_encoder.h" #include "api/audio_codecs/g722/audio_encoder_g722_config.h" #include "api/units/time_delta.h" diff --git a/modules/audio_coding/codecs/g722/test/testG722.cc b/modules/audio_coding/codecs/g722/test/testG722.cc index 9f2155d0f7..4f65e9938c 100644 --- a/modules/audio_coding/codecs/g722/test/testG722.cc +++ b/modules/audio_coding/codecs/g722/test/testG722.cc @@ -21,6 +21,8 @@ /* Runtime statistics */ #include + +#include #define CLOCKS_PER_SEC_G722 100000 // Forward declaration diff --git a/modules/audio_coding/codecs/isac/main/source/os_specific_inline.h b/modules/audio_coding/codecs/isac/main/source/os_specific_inline.h index fe9afa4ba2..7191d9fbab 100644 --- a/modules/audio_coding/codecs/isac/main/source/os_specific_inline.h +++ b/modules/audio_coding/codecs/isac/main/source/os_specific_inline.h @@ -11,10 +11,6 @@ #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_ #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_ -#include - -#include "rtc_base/system/arch.h" - #if defined(WEBRTC_POSIX) #define WebRtcIsac_lrint lrint #elif (defined(WEBRTC_ARCH_X86) && defined(WIN32)) diff --git a/modules/audio_coding/codecs/isac/main/source/pitch_filter.h b/modules/audio_coding/codecs/isac/main/source/pitch_filter.h index 9a232de87b..e1bbb5dad7 100644 --- a/modules/audio_coding/codecs/isac/main/source/pitch_filter.h +++ b/modules/audio_coding/codecs/isac/main/source/pitch_filter.h @@ -11,6 +11,7 @@ #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_FILTER_H_ #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_FILTER_H_ +#include "modules/audio_coding/codecs/isac/main/source/settings.h" #include "modules/audio_coding/codecs/isac/main/source/structs.h" void WebRtcIsac_PitchfilterPre(double* indat, diff --git a/modules/audio_coding/codecs/isac/main/source/structs.h b/modules/audio_coding/codecs/isac/main/source/structs.h index 6861ca42bd..0ca9b68803 100644 --- a/modules/audio_coding/codecs/isac/main/source/structs.h +++ b/modules/audio_coding/codecs/isac/main/source/structs.h @@ -18,6 +18,8 @@ #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_ #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_ +#include + #include "modules/audio_coding/codecs/isac/bandwidth_info.h" #include "modules/audio_coding/codecs/isac/main/source/settings.h" #include "modules/third_party/fft/fft.h" diff --git a/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc b/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc index adbda10e2e..6b884a479a 100644 --- a/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc +++ b/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc @@ -11,9 +11,16 @@ #include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h" #include +#include +#include #include +#include #include +#include +#include "api/array_view.h" +#include "api/audio_codecs/audio_decoder.h" +#include "rtc_base/buffer.h" #include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/codecs/opus/audio_coder_opus_common.cc b/modules/audio_coding/codecs/opus/audio_coder_opus_common.cc index 486f3a9f56..1a6dab0aaa 100644 --- a/modules/audio_coding/codecs/opus/audio_coder_opus_common.cc +++ b/modules/audio_coding/codecs/opus/audio_coder_opus_common.cc @@ -10,7 +10,14 @@ #include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h" +#include +#include +#include +#include + #include "absl/strings/string_view.h" +#include "api/audio_codecs/audio_format.h" +#include "rtc_base/string_to_number.h" namespace webrtc { diff --git a/modules/audio_coding/codecs/opus/audio_coder_opus_common.h b/modules/audio_coding/codecs/opus/audio_coder_opus_common.h index adbfbc6eea..44f019eb2a 100644 --- a/modules/audio_coding/codecs/opus/audio_coder_opus_common.h +++ b/modules/audio_coding/codecs/opus/audio_coder_opus_common.h @@ -11,14 +11,18 @@ #ifndef MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_CODER_OPUS_COMMON_H_ #define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_CODER_OPUS_COMMON_H_ +#include +#include #include #include #include #include #include "absl/strings/string_view.h" +#include "api/array_view.h" #include "api/audio_codecs/audio_decoder.h" #include "api/audio_codecs/audio_format.h" +#include "rtc_base/buffer.h" #include "rtc_base/string_to_number.h" namespace webrtc { diff --git a/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc b/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc index 8acb6d2262..6d9205d8bc 100644 --- a/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc +++ b/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc @@ -10,15 +10,20 @@ #include "modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h" -#include +#include +#include #include -#include +#include #include #include -#include "absl/memory/memory.h" +#include "api/audio_codecs/audio_decoder.h" +#include "api/audio_codecs/audio_format.h" +#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h" #include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h" -#include "rtc_base/string_to_number.h" +#include "modules/audio_coding/codecs/opus/opus_interface.h" +#include "rtc_base/buffer.h" +#include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h b/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h index 7d9d6aadf1..debca5d11d 100644 --- a/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h +++ b/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h @@ -13,7 +13,9 @@ #include +#include #include +#include #include #include "api/audio_codecs/audio_decoder.h" diff --git a/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_unittest.cc index 7b30a392f9..b1bc921af2 100644 --- a/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_unittest.cc +++ b/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_unittest.cc @@ -10,6 +10,13 @@ #include "api/audio_codecs/opus/audio_decoder_multi_channel_opus.h" +#include +#include +#include +#include + +#include "api/audio_codecs/audio_decoder.h" +#include "api/audio_codecs/audio_format.h" #include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h" #include "test/gmock.h" #include "test/gtest.h" diff --git a/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/modules/audio_coding/codecs/opus/audio_decoder_opus.cc index fae0bd47de..c4fb057718 100644 --- a/modules/audio_coding/codecs/opus/audio_decoder_opus.cc +++ b/modules/audio_coding/codecs/opus/audio_decoder_opus.cc @@ -10,13 +10,18 @@ #include "modules/audio_coding/codecs/opus/audio_decoder_opus.h" +#include +#include #include -#include #include +#include #include "api/array_view.h" +#include "api/audio_codecs/audio_decoder.h" #include "api/field_trials_view.h" #include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h" +#include "modules/audio_coding/codecs/opus/opus_interface.h" +#include "rtc_base/buffer.h" #include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/codecs/opus/audio_decoder_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_decoder_opus_unittest.cc index cba690406b..a9355dcfaa 100644 --- a/modules/audio_coding/codecs/opus/audio_decoder_opus_unittest.cc +++ b/modules/audio_coding/codecs/opus/audio_decoder_opus_unittest.cc @@ -10,7 +10,10 @@ #include "modules/audio_coding/codecs/opus/audio_decoder_opus.h" +#include #include +#include +#include #include #include #include @@ -19,6 +22,7 @@ #include "api/array_view.h" #include "api/audio/audio_frame.h" #include "api/audio_codecs/audio_decoder.h" +#include "api/audio_codecs/audio_encoder.h" #include "api/audio_codecs/opus/audio_encoder_opus_config.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" @@ -26,6 +30,7 @@ #include "modules/audio_coding/test/PCMFile.h" #include "rtc_base/buffer.h" #include "rtc_base/checks.h" +#include "rtc_base/numerics/safe_conversions.h" #include "rtc_base/random.h" #include "test/explicit_key_value_config.h" #include "test/gmock.h" diff --git a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc index 13d4cd7826..e0853d315e 100644 --- a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc +++ b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc @@ -20,14 +20,25 @@ #include "modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h" #include +#include +#include #include #include +#include #include +#include #include #include "absl/strings/match.h" +#include "api/array_view.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/audio_codecs/audio_format.h" +#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h" #include "api/audio_codecs/opus/audio_encoder_opus_config.h" +#include "api/units/time_delta.h" #include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h" +#include "modules/audio_coding/codecs/opus/opus_interface.h" +#include "rtc_base/buffer.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/numerics/safe_conversions.h" diff --git a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_unittest.cc index 7601589557..7c4d53dcc1 100644 --- a/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_unittest.cc +++ b/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_unittest.cc @@ -10,7 +10,14 @@ #include "api/audio_codecs/opus/audio_encoder_multi_channel_opus.h" +#include +#include +#include + +#include "api/audio_codecs/audio_encoder.h" +#include "api/audio_codecs/audio_format.h" #include "test/gmock.h" +#include "test/gtest.h" namespace webrtc { using ::testing::NiceMock; diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc index dbab35d4a8..8f95fed01c 100644 --- a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc +++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc @@ -11,20 +11,34 @@ #include "modules/audio_coding/codecs/opus/audio_encoder_opus.h" #include +#include +#include #include #include +#include #include #include +#include #include "absl/memory/memory.h" #include "absl/strings/match.h" #include "absl/strings/string_view.h" +#include "api/array_view.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/audio_codecs/audio_format.h" +#include "api/audio_codecs/opus/audio_encoder_opus_config.h" +#include "api/call/bitrate_allocation.h" +#include "api/environment/environment.h" #include "api/field_trials_view.h" +#include "api/units/time_delta.h" +#include "common_audio/smoothing_filter.h" #include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h" #include "modules/audio_coding/audio_network_adaptor/controller_manager.h" +#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h" #include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h" #include "modules/audio_coding/codecs/opus/opus_interface.h" #include "rtc_base/arraysize.h" +#include "rtc_base/buffer.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/numerics/exp_filter.h" diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/modules/audio_coding/codecs/opus/audio_encoder_opus.h index 706280f2e3..d46d2b5770 100644 --- a/modules/audio_coding/codecs/opus/audio_encoder_opus.h +++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.h @@ -11,20 +11,27 @@ #ifndef MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_ #define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_ +#include +#include #include #include #include #include +#include #include #include "absl/strings/string_view.h" +#include "api/array_view.h" #include "api/audio_codecs/audio_encoder.h" #include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/opus/audio_encoder_opus_config.h" +#include "api/call/bitrate_allocation.h" #include "api/environment/environment.h" +#include "api/units/time_delta.h" #include "common_audio/smoothing_filter.h" #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h" #include "modules/audio_coding/codecs/opus/opus_interface.h" +#include "rtc_base/buffer.h" namespace webrtc { diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc index d4fe4683e1..e3adc6ec71 100644 --- a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc +++ b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc @@ -10,17 +10,33 @@ #include "api/audio_codecs/opus/audio_encoder_opus.h" -#include +#include +#include #include +#include +#include #include +#include #include "absl/strings/string_view.h" +#include "api/array_view.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/audio_codecs/audio_format.h" +#include "api/audio_codecs/opus/audio_encoder_opus_config.h" +#include "api/call/bitrate_allocation.h" #include "api/environment/environment_factory.h" +#include "api/field_trials_view.h" +#include "api/rtp_parameters.h" +#include "api/units/data_rate.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" #include "common_audio/mocks/mock_smoothing_filter.h" +#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" #include "modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h" #include "modules/audio_coding/codecs/opus/audio_encoder_opus.h" #include "modules/audio_coding/codecs/opus/opus_interface.h" #include "modules/audio_coding/neteq/tools/audio_loop.h" +#include "rtc_base/buffer.h" #include "rtc_base/checks.h" #include "rtc_base/fake_clock.h" #include "test/explicit_key_value_config.h" diff --git a/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc b/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc index 2d5aa445ee..00a210fe78 100644 --- a/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc +++ b/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc @@ -8,14 +8,27 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include +#include +#include +#include +#include +#include +#include +#include + +#include "api/audio_codecs/audio_decoder.h" +#include "api/audio_codecs/audio_encoder.h" #include "api/audio_codecs/opus/audio_decoder_opus.h" #include "api/audio_codecs/opus/audio_encoder_opus.h" +#include "api/audio_codecs/opus/audio_encoder_opus_config.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" #include "common_audio/include/audio_util.h" #include "common_audio/window_generator.h" #include "modules/audio_coding/codecs/opus/test/lapped_transform.h" #include "modules/audio_coding/neteq/tools/audio_loop.h" +#include "rtc_base/buffer.h" #include "test/explicit_key_value_config.h" #include "test/gtest.h" #include "test/testsupport/file_utils.h" diff --git a/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc b/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc index b280ec0b17..465cde0901 100644 --- a/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc +++ b/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc @@ -8,11 +8,19 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include +#include +#include + +#include "api/audio_codecs/audio_encoder.h" #include "api/audio_codecs/opus/audio_encoder_opus.h" +#include "api/audio_codecs/opus/audio_encoder_opus_config.h" +#include "api/environment/environment.h" #include "api/environment/environment_factory.h" #include "api/test/metrics/global_metrics_logger_and_exporter.h" #include "api/test/metrics/metric.h" #include "modules/audio_coding/neteq/tools/audio_loop.h" +#include "rtc_base/buffer.h" #include "rtc_base/time_utils.h" #include "test/gtest.h" #include "test/testsupport/file_utils.h" diff --git a/modules/audio_coding/codecs/opus/opus_fec_test.cc b/modules/audio_coding/codecs/opus/opus_fec_test.cc index 815f26e31c..c48f486181 100644 --- a/modules/audio_coding/codecs/opus/opus_fec_test.cc +++ b/modules/audio_coding/codecs/opus/opus_fec_test.cc @@ -8,7 +8,14 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include +#include +#include +#include +#include #include +#include +#include #include "modules/audio_coding/codecs/opus/opus_interface.h" #include "test/gtest.h" diff --git a/modules/audio_coding/codecs/opus/opus_inst.h b/modules/audio_coding/codecs/opus/opus_inst.h index f7b45ea69b..6aea55262d 100644 --- a/modules/audio_coding/codecs/opus/opus_inst.h +++ b/modules/audio_coding/codecs/opus/opus_inst.h @@ -16,8 +16,10 @@ #include "rtc_base/ignore_wundef.h" RTC_PUSH_IGNORING_WUNDEF() -#include "third_party/opus/src/include/opus.h" -#include "third_party/opus/src/include/opus_multistream.h" +#include "third_party/opus/src/include/opus.h" // IWYU pragma: export +#include "third_party/opus/src/include/opus_defines.h" // IWYU pragma: export +#include "third_party/opus/src/include/opus_multistream.h" // IWYU pragma: export +#include "third_party/opus/src/include/opus_types.h" // IWYU pragma: export RTC_POP_IGNORING_WUNDEF() struct WebRtcOpusEncInst { diff --git a/modules/audio_coding/codecs/opus/opus_interface.cc b/modules/audio_coding/codecs/opus/opus_interface.cc index d18b517fe4..9dc579a4b7 100644 --- a/modules/audio_coding/codecs/opus/opus_interface.cc +++ b/modules/audio_coding/codecs/opus/opus_interface.cc @@ -10,10 +10,9 @@ #include "modules/audio_coding/codecs/opus/opus_interface.h" +#include #include -#include -#include "api/array_view.h" #include "rtc_base/checks.h" enum { diff --git a/modules/audio_coding/codecs/opus/opus_interface.h b/modules/audio_coding/codecs/opus/opus_interface.h index dd64e82476..4fb1dff233 100644 --- a/modules/audio_coding/codecs/opus/opus_interface.h +++ b/modules/audio_coding/codecs/opus/opus_interface.h @@ -14,7 +14,7 @@ #include #include -#include "modules/audio_coding/codecs/opus/opus_inst.h" +#include "modules/audio_coding/codecs/opus/opus_inst.h" // IWYU pragma: export #ifdef __cplusplus extern "C" { diff --git a/modules/audio_coding/codecs/opus/opus_speed_test.cc b/modules/audio_coding/codecs/opus/opus_speed_test.cc index 4477e8a5f8..62bf48ce4a 100644 --- a/modules/audio_coding/codecs/opus/opus_speed_test.cc +++ b/modules/audio_coding/codecs/opus/opus_speed_test.cc @@ -8,8 +8,16 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include +#include +#include +#include +#include +#include + #include "modules/audio_coding/codecs/opus/opus_interface.h" #include "modules/audio_coding/codecs/tools/audio_codec_speed_test.h" +#include "test/gtest.h" using ::std::string; diff --git a/modules/audio_coding/codecs/opus/opus_unittest.cc b/modules/audio_coding/codecs/opus/opus_unittest.cc index 0d04f1dc3b..4c03506afb 100644 --- a/modules/audio_coding/codecs/opus/opus_unittest.cc +++ b/modules/audio_coding/codecs/opus/opus_unittest.cc @@ -8,10 +8,19 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include "opus.h" + +#include +#include +#include +#include +#include #include #include +#include +#include -#include "modules/audio_coding/codecs/opus/opus_inst.h" +#include "api/array_view.h" #include "modules/audio_coding/codecs/opus/opus_interface.h" #include "modules/audio_coding/neteq/tools/audio_loop.h" #include "rtc_base/checks.h" diff --git a/modules/audio_coding/codecs/opus/test/BUILD.gn b/modules/audio_coding/codecs/opus/test/BUILD.gn index 8bc0bf5e0e..0b2fc17a36 100644 --- a/modules/audio_coding/codecs/opus/test/BUILD.gn +++ b/modules/audio_coding/codecs/opus/test/BUILD.gn @@ -47,6 +47,7 @@ if (rtc_include_tests) { ":test", "../../../../../common_audio", "../../../../../common_audio:common_audio_c", + "../../../../../rtc_base:checks", "../../../../../rtc_base:macromagic", "../../../../../test:test_support", "//testing/gtest", diff --git a/modules/audio_coding/codecs/opus/test/audio_ring_buffer.cc b/modules/audio_coding/codecs/opus/test/audio_ring_buffer.cc index 2a71b43d2c..616413a529 100644 --- a/modules/audio_coding/codecs/opus/test/audio_ring_buffer.cc +++ b/modules/audio_coding/codecs/opus/test/audio_ring_buffer.cc @@ -10,6 +10,8 @@ #include "modules/audio_coding/codecs/opus/test/audio_ring_buffer.h" +#include + #include "common_audio/ring_buffer.h" #include "rtc_base/checks.h" diff --git a/modules/audio_coding/codecs/opus/test/audio_ring_buffer.h b/modules/audio_coding/codecs/opus/test/audio_ring_buffer.h index a280ca2410..f73613b7d3 100644 --- a/modules/audio_coding/codecs/opus/test/audio_ring_buffer.h +++ b/modules/audio_coding/codecs/opus/test/audio_ring_buffer.h @@ -12,7 +12,6 @@ #include -#include #include struct RingBuffer; diff --git a/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc b/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc index 6dbc8ee9fe..617a281b0a 100644 --- a/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc +++ b/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc @@ -10,6 +10,7 @@ #include "modules/audio_coding/codecs/opus/test/audio_ring_buffer.h" +#include #include #include "common_audio/channel_buffer.h" diff --git a/modules/audio_coding/codecs/opus/test/blocker.h b/modules/audio_coding/codecs/opus/test/blocker.h index 59b7e29621..4f82f6413f 100644 --- a/modules/audio_coding/codecs/opus/test/blocker.h +++ b/modules/audio_coding/codecs/opus/test/blocker.h @@ -11,6 +11,7 @@ #ifndef MODULES_AUDIO_CODING_CODECS_OPUS_TEST_BLOCKER_H_ #define MODULES_AUDIO_CODING_CODECS_OPUS_TEST_BLOCKER_H_ +#include #include #include "common_audio/channel_buffer.h" diff --git a/modules/audio_coding/codecs/opus/test/blocker_unittest.cc b/modules/audio_coding/codecs/opus/test/blocker_unittest.cc index edca73980e..b5a349451c 100644 --- a/modules/audio_coding/codecs/opus/test/blocker_unittest.cc +++ b/modules/audio_coding/codecs/opus/test/blocker_unittest.cc @@ -10,8 +10,11 @@ #include "modules/audio_coding/codecs/opus/test/blocker.h" +#include +#include #include +#include "common_audio/channel_buffer.h" #include "rtc_base/arraysize.h" #include "test/gtest.h" diff --git a/modules/audio_coding/codecs/opus/test/lapped_transform.cc b/modules/audio_coding/codecs/opus/test/lapped_transform.cc index b1a6526bba..405616636c 100644 --- a/modules/audio_coding/codecs/opus/test/lapped_transform.cc +++ b/modules/audio_coding/codecs/opus/test/lapped_transform.cc @@ -10,7 +10,6 @@ #include "modules/audio_coding/codecs/opus/test/lapped_transform.h" -#include #include #include diff --git a/modules/audio_coding/codecs/opus/test/lapped_transform.h b/modules/audio_coding/codecs/opus/test/lapped_transform.h index 5a67d31801..5902abc24c 100644 --- a/modules/audio_coding/codecs/opus/test/lapped_transform.h +++ b/modules/audio_coding/codecs/opus/test/lapped_transform.h @@ -12,10 +12,12 @@ #define MODULES_AUDIO_CODING_CODECS_OPUS_TEST_LAPPED_TRANSFORM_H_ #include +#include #include #include "common_audio/real_fourier.h" #include "modules/audio_coding/codecs/opus/test/blocker.h" +#include "rtc_base/checks.h" #include "rtc_base/memory/aligned_malloc.h" namespace webrtc { diff --git a/modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc b/modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc index 8273f8d7f8..4ccba02475 100644 --- a/modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc +++ b/modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc @@ -12,8 +12,10 @@ #include #include +#include #include +#include "rtc_base/checks.h" #include "test/gtest.h" using std::complex; diff --git a/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc b/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc index 79f3562ed9..f732d03bd1 100644 --- a/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc +++ b/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc @@ -10,10 +10,15 @@ #include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h" +#include +#include #include +#include +#include "api/audio_codecs/audio_decoder.h" #include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h" #include "modules/audio_coding/codecs/pcm16b/pcm16b.h" +#include "rtc_base/buffer.h" #include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc b/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc index 9445b1ee3e..d9b4435cca 100644 --- a/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc +++ b/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc @@ -10,8 +10,12 @@ #include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h" +#include +#include + +#include "api/audio_codecs/audio_encoder.h" +#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h" #include "modules/audio_coding/codecs/pcm16b/pcm16b.h" -#include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h b/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h index c363b40b3f..2d6463707b 100644 --- a/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h +++ b/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h @@ -11,6 +11,10 @@ #ifndef MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_ #define MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_ +#include +#include + +#include "api/audio_codecs/audio_encoder.h" #include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h" namespace webrtc { diff --git a/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc b/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc index ecf91b45ac..28962853e9 100644 --- a/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc +++ b/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc @@ -13,6 +13,9 @@ #include #include +#include + +#include "api/audio_codecs/audio_format.h" namespace webrtc { diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc index 660ab91fa7..725aec0226 100644 --- a/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc +++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc @@ -12,13 +12,24 @@ #include +#include +#include +#include +#include +#include +#include #include #include #include "absl/strings/string_view.h" +#include "api/array_view.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/call/bitrate_allocation.h" +#include "api/field_trials_view.h" +#include "api/units/time_delta.h" +#include "rtc_base/buffer.h" #include "rtc_base/byte_order.h" #include "rtc_base/checks.h" -#include "rtc_base/logging.h" namespace webrtc { static constexpr const int kRedMaxPacketSize = diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/modules/audio_coding/codecs/red/audio_encoder_copy_red.h index 09d21325e2..e5a3d948dd 100644 --- a/modules/audio_coding/codecs/red/audio_encoder_copy_red.h +++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red.h @@ -17,10 +17,12 @@ #include #include #include +#include #include #include "api/array_view.h" #include "api/audio_codecs/audio_encoder.h" +#include "api/call/bitrate_allocation.h" #include "api/field_trials_view.h" #include "api/units/time_delta.h" #include "rtc_base/buffer.h" diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc index 92dec2c835..ea2d6d3850 100644 --- a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc +++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc @@ -10,12 +10,20 @@ #include "modules/audio_coding/codecs/red/audio_encoder_copy_red.h" +#include +#include +#include #include +#include +#include #include -#include "rtc_base/checks.h" +#include "api/array_view.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/units/time_delta.h" +#include "rtc_base/buffer.h" #include "rtc_base/numerics/safe_conversions.h" -#include "test/field_trial.h" +#include "test/gmock.h" #include "test/gtest.h" #include "test/mock_audio_encoder.h" #include "test/scoped_key_value_config.h" diff --git a/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc b/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc index 537e6fcede..f039ab6587 100644 --- a/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc +++ b/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc @@ -10,6 +10,12 @@ #include "modules/audio_coding/codecs/tools/audio_codec_speed_test.h" +#include +#include +#include +#include +#include + #include "rtc_base/checks.h" #include "test/gtest.h" #include "test/testsupport/file_utils.h" diff --git a/modules/audio_coding/codecs/tools/audio_codec_speed_test.h b/modules/audio_coding/codecs/tools/audio_codec_speed_test.h index c5f1d7c259..71d80ba73b 100644 --- a/modules/audio_coding/codecs/tools/audio_codec_speed_test.h +++ b/modules/audio_coding/codecs/tools/audio_codec_speed_test.h @@ -11,8 +11,12 @@ #ifndef MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_ #define MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_ +#include +#include +#include #include #include +#include #include "test/gtest.h" diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h index d4a6a39b66..1493034d54 100644 --- a/modules/audio_coding/include/audio_coding_module.h +++ b/modules/audio_coding/include/audio_coding_module.h @@ -11,15 +11,15 @@ #ifndef MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_ #define MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_ +#include +#include #include -#include -#include #include -#include #include "api/audio_codecs/audio_encoder.h" #include "api/function_view.h" #include "modules/audio_coding/include/audio_coding_module_typedefs.h" +#include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/include/audio_coding_module_typedefs.h b/modules/audio_coding/include/audio_coding_module_typedefs.h index f375108195..f0939c5c94 100644 --- a/modules/audio_coding/include/audio_coding_module_typedefs.h +++ b/modules/audio_coding/include/audio_coding_module_typedefs.h @@ -11,7 +11,7 @@ #ifndef MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_ #define MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_ -#include +#include namespace webrtc { diff --git a/modules/audio_coding/neteq/accelerate.cc b/modules/audio_coding/neteq/accelerate.cc index c15ab73d67..eda4370541 100644 --- a/modules/audio_coding/neteq/accelerate.cc +++ b/modules/audio_coding/neteq/accelerate.cc @@ -10,8 +10,13 @@ #include "modules/audio_coding/neteq/accelerate.h" +#include +#include + #include "api/array_view.h" #include "modules/audio_coding/neteq/audio_multi_vector.h" +#include "modules/audio_coding/neteq/time_stretch.h" +#include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/audio_decoder_unittest.cc b/modules/audio_coding/neteq/audio_decoder_unittest.cc index 06a088299c..9f29394271 100644 --- a/modules/audio_coding/neteq/audio_decoder_unittest.cc +++ b/modules/audio_coding/neteq/audio_decoder_unittest.cc @@ -8,14 +8,23 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include "api/audio_codecs/audio_decoder.h" + #include #include +#include #include -#include +#include +#include +#include #include +#include "api/array_view.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/audio_codecs/g722/audio_encoder_g722_config.h" #include "api/audio_codecs/opus/audio_encoder_opus.h" +#include "api/audio_codecs/opus/audio_encoder_opus_config.h" #include "api/environment/environment_factory.h" #include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h" #include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h" @@ -24,8 +33,10 @@ #include "modules/audio_coding/codecs/opus/audio_decoder_opus.h" #include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h" #include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h" +#include "modules/audio_coding/neteq/tools/input_audio_file.h" #include "modules/audio_coding/neteq/tools/resample_input_audio_file.h" -#include "rtc_base/system/arch.h" +#include "rtc_base/buffer.h" +#include "rtc_base/checks.h" #include "test/explicit_key_value_config.h" #include "test/gtest.h" #include "test/testsupport/file_utils.h" diff --git a/modules/audio_coding/neteq/audio_multi_vector.cc b/modules/audio_coding/neteq/audio_multi_vector.cc index e86d9356ad..aa452808e1 100644 --- a/modules/audio_coding/neteq/audio_multi_vector.cc +++ b/modules/audio_coding/neteq/audio_multi_vector.cc @@ -11,7 +11,12 @@ #include "modules/audio_coding/neteq/audio_multi_vector.h" #include +#include +#include +#include +#include "api/array_view.h" +#include "modules/audio_coding/neteq/audio_vector.h" #include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/audio_multi_vector_unittest.cc b/modules/audio_coding/neteq/audio_multi_vector_unittest.cc index eaf31476b3..c7b6311867 100644 --- a/modules/audio_coding/neteq/audio_multi_vector_unittest.cc +++ b/modules/audio_coding/neteq/audio_multi_vector_unittest.cc @@ -12,9 +12,11 @@ #include -#include +#include +#include #include +#include "modules/audio_coding/neteq/audio_vector.h" #include "rtc_base/numerics/safe_conversions.h" #include "test/gtest.h" diff --git a/modules/audio_coding/neteq/audio_vector.cc b/modules/audio_coding/neteq/audio_vector.cc index 89105949d5..afbd6d4e41 100644 --- a/modules/audio_coding/neteq/audio_vector.cc +++ b/modules/audio_coding/neteq/audio_vector.cc @@ -11,6 +11,9 @@ #include "modules/audio_coding/neteq/audio_vector.h" #include +#include +#include +#include #include #include "rtc_base/checks.h" diff --git a/modules/audio_coding/neteq/audio_vector_unittest.cc b/modules/audio_coding/neteq/audio_vector_unittest.cc index 8221a5f6c7..97c1b7e10e 100644 --- a/modules/audio_coding/neteq/audio_vector_unittest.cc +++ b/modules/audio_coding/neteq/audio_vector_unittest.cc @@ -12,7 +12,7 @@ #include -#include +#include #include "rtc_base/numerics/safe_conversions.h" #include "test/gtest.h" diff --git a/modules/audio_coding/neteq/background_noise.cc b/modules/audio_coding/neteq/background_noise.cc index 812d656c96..50c622eb21 100644 --- a/modules/audio_coding/neteq/background_noise.cc +++ b/modules/audio_coding/neteq/background_noise.cc @@ -13,10 +13,15 @@ #include // memcpy #include // min, max +#include +#include "api/array_view.h" +#include "common_audio/signal_processing/dot_product_with_scale.h" #include "common_audio/signal_processing/include/signal_processing_library.h" +#include "common_audio/signal_processing/include/spl_inl.h" #include "modules/audio_coding/neteq/audio_multi_vector.h" #include "modules/audio_coding/neteq/cross_correlation.h" +#include "rtc_base/checks.h" namespace webrtc { namespace { diff --git a/modules/audio_coding/neteq/background_noise.h b/modules/audio_coding/neteq/background_noise.h index ef2f9cb966..1a506a195b 100644 --- a/modules/audio_coding/neteq/background_noise.h +++ b/modules/audio_coding/neteq/background_noise.h @@ -13,6 +13,7 @@ #include // size_t +#include #include #include "api/array_view.h" diff --git a/modules/audio_coding/neteq/background_noise_unittest.cc b/modules/audio_coding/neteq/background_noise_unittest.cc index e32492f57e..16b0051277 100644 --- a/modules/audio_coding/neteq/background_noise_unittest.cc +++ b/modules/audio_coding/neteq/background_noise_unittest.cc @@ -12,6 +12,8 @@ #include "modules/audio_coding/neteq/background_noise.h" +#include + #include "test/gtest.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/buffer_level_filter.cc b/modules/audio_coding/neteq/buffer_level_filter.cc index 79789feffc..d2070a4bae 100644 --- a/modules/audio_coding/neteq/buffer_level_filter.cc +++ b/modules/audio_coding/neteq/buffer_level_filter.cc @@ -13,6 +13,7 @@ #include #include +#include #include "rtc_base/numerics/safe_conversions.h" diff --git a/modules/audio_coding/neteq/comfort_noise.cc b/modules/audio_coding/neteq/comfort_noise.cc index 49ba21e6de..da94881585 100644 --- a/modules/audio_coding/neteq/comfort_noise.cc +++ b/modules/audio_coding/neteq/comfort_noise.cc @@ -10,6 +10,7 @@ #include "modules/audio_coding/neteq/comfort_noise.h" +#include #include #include @@ -20,7 +21,6 @@ #include "modules/audio_coding/neteq/decoder_database.h" #include "modules/audio_coding/neteq/dsp_helper.h" #include "modules/audio_coding/neteq/sync_buffer.h" -#include "rtc_base/buffer.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" diff --git a/modules/audio_coding/neteq/comfort_noise_unittest.cc b/modules/audio_coding/neteq/comfort_noise_unittest.cc index b436800061..7b376d1f50 100644 --- a/modules/audio_coding/neteq/comfort_noise_unittest.cc +++ b/modules/audio_coding/neteq/comfort_noise_unittest.cc @@ -14,6 +14,7 @@ #include "modules/audio_coding/neteq/mock/mock_decoder_database.h" #include "modules/audio_coding/neteq/sync_buffer.h" +#include "test/gmock.h" #include "test/gtest.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/cross_correlation.cc b/modules/audio_coding/neteq/cross_correlation.cc index 37ed9374f0..121ba210fc 100644 --- a/modules/audio_coding/neteq/cross_correlation.cc +++ b/modules/audio_coding/neteq/cross_correlation.cc @@ -10,10 +10,11 @@ #include "modules/audio_coding/neteq/cross_correlation.h" +#include #include -#include #include "common_audio/signal_processing/include/signal_processing_library.h" +#include "common_audio/signal_processing/include/spl_inl.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/decision_logic_unittest.cc b/modules/audio_coding/neteq/decision_logic_unittest.cc index 2d3170b9ec..363b45c83b 100644 --- a/modules/audio_coding/neteq/decision_logic_unittest.cc +++ b/modules/audio_coding/neteq/decision_logic_unittest.cc @@ -12,6 +12,11 @@ #include "modules/audio_coding/neteq/decision_logic.h" +#include +#include +#include + +#include "api/neteq/neteq.h" #include "api/neteq/neteq_controller.h" #include "api/neteq/tick_timer.h" #include "modules/audio_coding/neteq/delay_manager.h" @@ -19,6 +24,7 @@ #include "modules/audio_coding/neteq/mock/mock_delay_manager.h" #include "modules/audio_coding/neteq/mock/mock_packet_arrival_history.h" #include "test/explicit_key_value_config.h" +#include "test/gmock.h" #include "test/gtest.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/decoder_database.cc b/modules/audio_coding/neteq/decoder_database.cc index 821efec811..070a07b165 100644 --- a/modules/audio_coding/neteq/decoder_database.cc +++ b/modules/audio_coding/neteq/decoder_database.cc @@ -14,13 +14,22 @@ #include #include -#include +#include +#include #include +#include #include "absl/strings/match.h" +#include "absl/strings/str_cat.h" #include "absl/strings/string_view.h" +#include "api/audio_codecs/audio_codec_pair_id.h" #include "api/audio_codecs/audio_decoder.h" +#include "api/audio_codecs/audio_decoder_factory.h" +#include "api/audio_codecs/audio_format.h" #include "api/environment/environment.h" +#include "api/scoped_refptr.h" +#include "modules/audio_coding/codecs/cng/webrtc_cng.h" +#include "modules/audio_coding/neteq/packet.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" diff --git a/modules/audio_coding/neteq/decoder_database.h b/modules/audio_coding/neteq/decoder_database.h index 9c1a1ec1ef..a8970d393d 100644 --- a/modules/audio_coding/neteq/decoder_database.h +++ b/modules/audio_coding/neteq/decoder_database.h @@ -11,17 +11,23 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_DECODER_DATABASE_H_ #define MODULES_AUDIO_CODING_NETEQ_DECODER_DATABASE_H_ +#include #include #include +#include #include +#include #include "absl/strings/string_view.h" +#include "api/audio_codecs/audio_codec_pair_id.h" +#include "api/audio_codecs/audio_decoder.h" #include "api/audio_codecs/audio_decoder_factory.h" #include "api/audio_codecs/audio_format.h" #include "api/environment/environment.h" #include "api/scoped_refptr.h" #include "modules/audio_coding/codecs/cng/webrtc_cng.h" #include "modules/audio_coding/neteq/packet.h" +#include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/decoder_database_unittest.cc b/modules/audio_coding/neteq/decoder_database_unittest.cc index e11b38070e..4718f559af 100644 --- a/modules/audio_coding/neteq/decoder_database_unittest.cc +++ b/modules/audio_coding/neteq/decoder_database_unittest.cc @@ -12,11 +12,19 @@ #include +#include +#include #include +#include #include "absl/memory/memory.h" +#include "api/audio_codecs/audio_decoder.h" +#include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/environment/environment_factory.h" +#include "api/make_ref_counted.h" +#include "modules/audio_coding/codecs/cng/webrtc_cng.h" +#include "modules/audio_coding/neteq/packet.h" #include "test/gmock.h" #include "test/gtest.h" #include "test/mock_audio_decoder.h" diff --git a/modules/audio_coding/neteq/dsp_helper.cc b/modules/audio_coding/neteq/dsp_helper.cc index a979f94214..af3edcb103 100644 --- a/modules/audio_coding/neteq/dsp_helper.cc +++ b/modules/audio_coding/neteq/dsp_helper.cc @@ -13,8 +13,12 @@ #include // Access to memset. #include // Access to min, max. +#include #include "common_audio/signal_processing/include/signal_processing_library.h" +#include "modules/audio_coding/neteq/audio_multi_vector.h" +#include "modules/audio_coding/neteq/audio_vector.h" +#include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/dsp_helper_unittest.cc b/modules/audio_coding/neteq/dsp_helper_unittest.cc index 09247417d3..5f03d81a1b 100644 --- a/modules/audio_coding/neteq/dsp_helper_unittest.cc +++ b/modules/audio_coding/neteq/dsp_helper_unittest.cc @@ -10,6 +10,8 @@ #include "modules/audio_coding/neteq/dsp_helper.h" +#include + #include "modules/audio_coding/neteq/audio_multi_vector.h" #include "test/gtest.h" diff --git a/modules/audio_coding/neteq/dtmf_buffer.cc b/modules/audio_coding/neteq/dtmf_buffer.cc index 9f78aca6e2..5813f6df9a 100644 --- a/modules/audio_coding/neteq/dtmf_buffer.cc +++ b/modules/audio_coding/neteq/dtmf_buffer.cc @@ -11,6 +11,8 @@ #include "modules/audio_coding/neteq/dtmf_buffer.h" #include // max +#include +#include #include "rtc_base/checks.h" #include "rtc_base/logging.h" diff --git a/modules/audio_coding/neteq/dtmf_buffer_unittest.cc b/modules/audio_coding/neteq/dtmf_buffer_unittest.cc index 83745b6c09..cd66116158 100644 --- a/modules/audio_coding/neteq/dtmf_buffer_unittest.cc +++ b/modules/audio_coding/neteq/dtmf_buffer_unittest.cc @@ -10,14 +10,14 @@ #include "modules/audio_coding/neteq/dtmf_buffer.h" +#include + #ifdef WIN32 #include // ntohl() #else -#include // ntohl() #endif -#include - +#include "rtc_base/ip_address.h" #include "test/gtest.h" // Modify the tests so that they pass with the modifications done to DtmfBuffer diff --git a/modules/audio_coding/neteq/dtmf_tone_generator.cc b/modules/audio_coding/neteq/dtmf_tone_generator.cc index 9061e27c67..01f6ead3d9 100644 --- a/modules/audio_coding/neteq/dtmf_tone_generator.cc +++ b/modules/audio_coding/neteq/dtmf_tone_generator.cc @@ -30,6 +30,10 @@ #include "modules/audio_coding/neteq/dtmf_tone_generator.h" +#include +#include + +#include "modules/audio_coding/neteq/audio_multi_vector.h" #include "modules/audio_coding/neteq/audio_vector.h" #include "rtc_base/arraysize.h" #include "rtc_base/checks.h" diff --git a/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc b/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc index 989fe05a11..625610e501 100644 --- a/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc +++ b/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc @@ -14,6 +14,8 @@ #include +#include + #include "common_audio/include/audio_util.h" #include "modules/audio_coding/neteq/audio_multi_vector.h" #include "rtc_base/strings/string_builder.h" diff --git a/modules/audio_coding/neteq/expand.h b/modules/audio_coding/neteq/expand.h index bd58788ae0..09ff46d5d0 100644 --- a/modules/audio_coding/neteq/expand.h +++ b/modules/audio_coding/neteq/expand.h @@ -11,9 +11,12 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_EXPAND_H_ #define MODULES_AUDIO_CODING_NETEQ_EXPAND_H_ +#include +#include #include #include "modules/audio_coding/neteq/audio_vector.h" +#include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/expand_uma_logger.cc b/modules/audio_coding/neteq/expand_uma_logger.cc index e735a45636..282a6e3895 100644 --- a/modules/audio_coding/neteq/expand_uma_logger.cc +++ b/modules/audio_coding/neteq/expand_uma_logger.cc @@ -9,7 +9,12 @@ #include "modules/audio_coding/neteq/expand_uma_logger.h" +#include +#include +#include + #include "absl/strings/string_view.h" +#include "api/neteq/tick_timer.h" #include "rtc_base/checks.h" #include "system_wrappers/include/metrics.h" diff --git a/modules/audio_coding/neteq/expand_unittest.cc b/modules/audio_coding/neteq/expand_unittest.cc index 4a63a020f2..cb0dcaaa88 100644 --- a/modules/audio_coding/neteq/expand_unittest.cc +++ b/modules/audio_coding/neteq/expand_unittest.cc @@ -12,7 +12,11 @@ #include "modules/audio_coding/neteq/expand.h" -#include "common_audio/signal_processing/include/signal_processing_library.h" +#include +#include +#include + +#include "api/neteq/tick_timer.h" #include "modules/audio_coding/neteq/background_noise.h" #include "modules/audio_coding/neteq/random_vector.h" #include "modules/audio_coding/neteq/statistics_calculator.h" diff --git a/modules/audio_coding/neteq/histogram.cc b/modules/audio_coding/neteq/histogram.cc index 825c8ac349..454f3112a4 100644 --- a/modules/audio_coding/neteq/histogram.cc +++ b/modules/audio_coding/neteq/histogram.cc @@ -11,12 +11,11 @@ #include "modules/audio_coding/neteq/histogram.h" #include +#include #include -#include #include #include "rtc_base/checks.h" -#include "rtc_base/numerics/safe_conversions.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/histogram_unittest.cc b/modules/audio_coding/neteq/histogram_unittest.cc index e30a2956dc..793c40b826 100644 --- a/modules/audio_coding/neteq/histogram_unittest.cc +++ b/modules/audio_coding/neteq/histogram_unittest.cc @@ -11,6 +11,8 @@ #include "modules/audio_coding/neteq/histogram.h" #include +#include +#include #include "test/gtest.h" diff --git a/modules/audio_coding/neteq/merge.cc b/modules/audio_coding/neteq/merge.cc index 17767975d7..e6a05ae587 100644 --- a/modules/audio_coding/neteq/merge.cc +++ b/modules/audio_coding/neteq/merge.cc @@ -13,14 +13,20 @@ #include // memmove, memcpy, memset, size_t #include // min, max +#include +#include #include +#include "api/array_view.h" +#include "common_audio/signal_processing/dot_product_with_scale.h" #include "common_audio/signal_processing/include/signal_processing_library.h" +#include "common_audio/signal_processing/include/spl_inl.h" #include "modules/audio_coding/neteq/audio_multi_vector.h" #include "modules/audio_coding/neteq/cross_correlation.h" #include "modules/audio_coding/neteq/dsp_helper.h" #include "modules/audio_coding/neteq/expand.h" #include "modules/audio_coding/neteq/sync_buffer.h" +#include "rtc_base/checks.h" #include "rtc_base/numerics/safe_conversions.h" #include "rtc_base/numerics/safe_minmax.h" diff --git a/modules/audio_coding/neteq/merge.h b/modules/audio_coding/neteq/merge.h index 2f27106bfe..89e171f2b3 100644 --- a/modules/audio_coding/neteq/merge.h +++ b/modules/audio_coding/neteq/merge.h @@ -11,6 +11,10 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_MERGE_H_ #define MODULES_AUDIO_CODING_NETEQ_MERGE_H_ +#include +#include +#include + #include "modules/audio_coding/neteq/audio_multi_vector.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/merge_unittest.cc b/modules/audio_coding/neteq/merge_unittest.cc index 87740af508..eb5ee93455 100644 --- a/modules/audio_coding/neteq/merge_unittest.cc +++ b/modules/audio_coding/neteq/merge_unittest.cc @@ -12,9 +12,12 @@ #include "modules/audio_coding/neteq/merge.h" -#include +#include +#include +#include #include +#include "api/neteq/tick_timer.h" #include "modules/audio_coding/neteq/background_noise.h" #include "modules/audio_coding/neteq/expand.h" #include "modules/audio_coding/neteq/random_vector.h" diff --git a/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h b/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h index 503f6ac6bd..b843910883 100644 --- a/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h +++ b/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h @@ -11,6 +11,8 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_ #define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_ +#include + #include "modules/audio_coding/neteq/buffer_level_filter.h" #include "test/gmock.h" diff --git a/modules/audio_coding/neteq/mock/mock_decoder_database.h b/modules/audio_coding/neteq/mock/mock_decoder_database.h index 08e12156df..9fe73dfcc1 100644 --- a/modules/audio_coding/neteq/mock/mock_decoder_database.h +++ b/modules/audio_coding/neteq/mock/mock_decoder_database.h @@ -11,9 +11,13 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_ #define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_ -#include +#include +#include +#include "api/audio_codecs/audio_decoder.h" +#include "api/audio_codecs/audio_format.h" #include "api/environment/environment_factory.h" +#include "modules/audio_coding/codecs/cng/webrtc_cng.h" #include "modules/audio_coding/neteq/decoder_database.h" #include "test/gmock.h" diff --git a/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h b/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h index c60c56d36b..2f3c595706 100644 --- a/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h +++ b/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h @@ -11,6 +11,9 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_BUFFER_H_ #define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_BUFFER_H_ +#include +#include + #include "modules/audio_coding/neteq/dtmf_buffer.h" #include "test/gmock.h" diff --git a/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h b/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h index 60de167c29..ce37aa7730 100644 --- a/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h +++ b/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h @@ -11,6 +11,9 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_TONE_GENERATOR_H_ #define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_TONE_GENERATOR_H_ +#include + +#include "modules/audio_coding/neteq/audio_multi_vector.h" #include "modules/audio_coding/neteq/dtmf_tone_generator.h" #include "test/gmock.h" diff --git a/modules/audio_coding/neteq/mock/mock_expand.h b/modules/audio_coding/neteq/mock/mock_expand.h index 9d66779021..59b608b5c1 100644 --- a/modules/audio_coding/neteq/mock/mock_expand.h +++ b/modules/audio_coding/neteq/mock/mock_expand.h @@ -11,6 +11,8 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXPAND_H_ #define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXPAND_H_ +#include + #include "modules/audio_coding/neteq/expand.h" #include "test/gmock.h" diff --git a/modules/audio_coding/neteq/mock/mock_histogram.h b/modules/audio_coding/neteq/mock/mock_histogram.h index 03abbc1d4b..365a2675d0 100644 --- a/modules/audio_coding/neteq/mock/mock_histogram.h +++ b/modules/audio_coding/neteq/mock/mock_histogram.h @@ -11,6 +11,8 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_HISTOGRAM_H_ #define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_HISTOGRAM_H_ +#include + #include "modules/audio_coding/neteq/histogram.h" #include "test/gmock.h" diff --git a/modules/audio_coding/neteq/mock/mock_neteq_controller.h b/modules/audio_coding/neteq/mock/mock_neteq_controller.h index ea7809a1b8..fef7c4865e 100644 --- a/modules/audio_coding/neteq/mock/mock_neteq_controller.h +++ b/modules/audio_coding/neteq/mock/mock_neteq_controller.h @@ -11,6 +11,11 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_NETEQ_CONTROLLER_H_ #define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_NETEQ_CONTROLLER_H_ +#include +#include +#include + +#include "api/neteq/neteq.h" #include "api/neteq/neteq_controller.h" #include "test/gmock.h" diff --git a/modules/audio_coding/neteq/mock/mock_packet_arrival_history.h b/modules/audio_coding/neteq/mock/mock_packet_arrival_history.h index 23731d093b..01fe0e3c7d 100644 --- a/modules/audio_coding/neteq/mock/mock_packet_arrival_history.h +++ b/modules/audio_coding/neteq/mock/mock_packet_arrival_history.h @@ -11,6 +11,8 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_ARRIVAL_HISTORY_H_ #define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_ARRIVAL_HISTORY_H_ +#include + #include "api/neteq/tick_timer.h" #include "modules/audio_coding/neteq/packet_arrival_history.h" #include "test/gmock.h" diff --git a/modules/audio_coding/neteq/mock/mock_packet_buffer.h b/modules/audio_coding/neteq/mock/mock_packet_buffer.h index 85bc82bec3..7876ebf6c4 100644 --- a/modules/audio_coding/neteq/mock/mock_packet_buffer.h +++ b/modules/audio_coding/neteq/mock/mock_packet_buffer.h @@ -11,6 +11,11 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_ #define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_ +#include +#include +#include + +#include "modules/audio_coding/neteq/packet.h" #include "modules/audio_coding/neteq/packet_buffer.h" #include "test/gmock.h" diff --git a/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h b/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h index 9daf571a80..6af02c25ce 100644 --- a/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h +++ b/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h @@ -11,6 +11,7 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_RED_PAYLOAD_SPLITTER_H_ #define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_RED_PAYLOAD_SPLITTER_H_ +#include "modules/audio_coding/neteq/packet.h" #include "modules/audio_coding/neteq/red_payload_splitter.h" #include "test/gmock.h" diff --git a/modules/audio_coding/neteq/mock/mock_statistics_calculator.h b/modules/audio_coding/neteq/mock/mock_statistics_calculator.h index 85c70beb6a..70e7a27e0e 100644 --- a/modules/audio_coding/neteq/mock/mock_statistics_calculator.h +++ b/modules/audio_coding/neteq/mock/mock_statistics_calculator.h @@ -11,6 +11,9 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_STATISTICS_CALCULATOR_H_ #define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_STATISTICS_CALCULATOR_H_ +#include + +#include "api/neteq/tick_timer.h" #include "modules/audio_coding/neteq/statistics_calculator.h" #include "test/gmock.h" diff --git a/modules/audio_coding/neteq/nack_tracker.cc b/modules/audio_coding/neteq/nack_tracker.cc index f6f6b80dbc..a85c8af984 100644 --- a/modules/audio_coding/neteq/nack_tracker.cc +++ b/modules/audio_coding/neteq/nack_tracker.cc @@ -10,10 +10,14 @@ #include "modules/audio_coding/neteq/nack_tracker.h" +#include #include +#include #include +#include #include "api/field_trials_view.h" +#include "modules/include/module_common_types_public.h" #include "rtc_base/checks.h" #include "rtc_base/experiments/struct_parameters_parser.h" #include "rtc_base/logging.h" diff --git a/modules/audio_coding/neteq/nack_tracker_unittest.cc b/modules/audio_coding/neteq/nack_tracker_unittest.cc index 89d3555b29..716544a75e 100644 --- a/modules/audio_coding/neteq/nack_tracker_unittest.cc +++ b/modules/audio_coding/neteq/nack_tracker_unittest.cc @@ -13,9 +13,10 @@ #include #include +#include #include +#include -#include "modules/audio_coding/include/audio_coding_module_typedefs.h" #include "test/explicit_key_value_config.h" #include "test/gtest.h" diff --git a/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc b/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc index df22fbdd71..cb2a9d37b2 100644 --- a/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc +++ b/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc @@ -10,18 +10,28 @@ // Test to verify correct operation when using the decoder-internal PLC. +#include +#include #include #include +#include #include #include +#include "api/array_view.h" +#include "api/audio_codecs/audio_decoder.h" +#include "api/audio_codecs/audio_format.h" +#include "api/make_ref_counted.h" +#include "api/neteq/neteq.h" +#include "api/rtp_headers.h" #include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h" #include "modules/audio_coding/neteq/tools/audio_checksum.h" -#include "modules/audio_coding/neteq/tools/audio_sink.h" #include "modules/audio_coding/neteq/tools/encode_neteq_input.h" -#include "modules/audio_coding/neteq/tools/fake_decode_from_file.h" #include "modules/audio_coding/neteq/tools/input_audio_file.h" +#include "modules/audio_coding/neteq/tools/neteq_input.h" #include "modules/audio_coding/neteq/tools/neteq_test.h" +#include "rtc_base/buffer.h" +#include "rtc_base/checks.h" #include "rtc_base/numerics/safe_conversions.h" #include "test/audio_decoder_proxy_factory.h" #include "test/gtest.h" diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h index 9fbf21abcc..722fa876d3 100644 --- a/modules/audio_coding/neteq/neteq_impl.h +++ b/modules/audio_coding/neteq/neteq_impl.h @@ -31,6 +31,7 @@ #include "api/rtp_headers.h" #include "api/rtp_packet_info.h" #include "api/scoped_refptr.h" +#include "api/units/timestamp.h" #include "modules/audio_coding/neteq/audio_multi_vector.h" #include "modules/audio_coding/neteq/packet.h" #include "modules/audio_coding/neteq/packet_buffer.h" diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc index c625dcd898..22413620fd 100644 --- a/modules/audio_coding/neteq/neteq_impl_unittest.cc +++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc @@ -10,31 +10,44 @@ #include "modules/audio_coding/neteq/neteq_impl.h" +#include +#include +#include #include +#include #include #include +#include "api/audio_codecs/audio_decoder.h" +#include "api/audio_codecs/audio_decoder_factory.h" +#include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" +#include "api/environment/environment.h" #include "api/environment/environment_factory.h" +#include "api/make_ref_counted.h" #include "api/neteq/default_neteq_controller_factory.h" #include "api/neteq/default_neteq_factory.h" #include "api/neteq/neteq.h" #include "api/neteq/neteq_controller.h" +#include "api/neteq/tick_timer.h" +#include "api/rtp_headers.h" +#include "api/rtp_packet_info.h" +#include "api/scoped_refptr.h" #include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h" -#include "modules/audio_coding/neteq/accelerate.h" #include "modules/audio_coding/neteq/decision_logic.h" #include "modules/audio_coding/neteq/expand.h" -#include "modules/audio_coding/neteq/histogram.h" #include "modules/audio_coding/neteq/mock/mock_decoder_database.h" #include "modules/audio_coding/neteq/mock/mock_dtmf_buffer.h" #include "modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h" #include "modules/audio_coding/neteq/mock/mock_neteq_controller.h" #include "modules/audio_coding/neteq/mock/mock_packet_buffer.h" #include "modules/audio_coding/neteq/mock/mock_red_payload_splitter.h" -#include "modules/audio_coding/neteq/preemptive_expand.h" -#include "modules/audio_coding/neteq/statistics_calculator.h" +#include "modules/audio_coding/neteq/packet.h" +#include "modules/audio_coding/neteq/packet_buffer.h" +#include "modules/audio_coding/neteq/red_payload_splitter.h" #include "modules/audio_coding/neteq/sync_buffer.h" #include "modules/audio_coding/neteq/timestamp_scaler.h" +#include "rtc_base/checks.h" #include "rtc_base/numerics/safe_conversions.h" #include "system_wrappers/include/clock.h" #include "test/audio_decoder_proxy_factory.h" diff --git a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc index 3fd6541c39..a87590d899 100644 --- a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc +++ b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc @@ -8,18 +8,31 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include +#include +#include #include +#include #include +#include +#include "api/array_view.h" #include "api/audio/audio_frame.h" #include "api/audio_codecs/audio_decoder.h" -#include "api/audio_codecs/builtin_audio_decoder_factory.h" +#include "api/audio_codecs/audio_decoder_factory.h" +#include "api/audio_codecs/audio_format.h" #include "api/environment/environment_factory.h" +#include "api/make_ref_counted.h" #include "api/neteq/default_neteq_factory.h" #include "api/neteq/neteq.h" +#include "api/rtp_headers.h" +#include "api/scoped_refptr.h" +#include "api/units/timestamp.h" #include "modules/audio_coding/neteq/tools/rtp_generator.h" +#include "rtc_base/buffer.h" #include "test/audio_decoder_proxy_factory.h" #include "test/gmock.h" +#include "test/gtest.h" namespace webrtc { namespace test { diff --git a/modules/audio_coding/neteq/neteq_stereo_unittest.cc b/modules/audio_coding/neteq/neteq_stereo_unittest.cc index 1eb810835a..5e13d73cb4 100644 --- a/modules/audio_coding/neteq/neteq_stereo_unittest.cc +++ b/modules/audio_coding/neteq/neteq_stereo_unittest.cc @@ -11,20 +11,28 @@ // Test to verify correct stereo and multi-channel operation. #include +#include +#include +#include #include #include +#include #include +#include "api/array_view.h" #include "api/audio/audio_frame.h" +#include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" #include "api/neteq/default_neteq_factory.h" #include "api/neteq/neteq.h" +#include "api/rtp_headers.h" #include "api/units/timestamp.h" #include "modules/audio_coding/codecs/pcm16b/pcm16b.h" #include "modules/audio_coding/neteq/tools/input_audio_file.h" #include "modules/audio_coding/neteq/tools/rtp_generator.h" +#include "rtc_base/checks.h" #include "rtc_base/strings/string_builder.h" #include "system_wrappers/include/clock.h" #include "test/gtest.h" diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc index b0e0e77c54..ae9ca81234 100644 --- a/modules/audio_coding/neteq/neteq_unittest.cc +++ b/modules/audio_coding/neteq/neteq_unittest.cc @@ -15,27 +15,32 @@ #include // memset #include +#include +#include #include +#include #include #include -#include +#include #include "absl/flags/flag.h" +#include "api/array_view.h" #include "api/audio/audio_frame.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" +#include "api/rtp_headers.h" +#include "api/units/time_delta.h" #include "modules/audio_coding/codecs/pcm16b/pcm16b.h" #include "modules/audio_coding/neteq/test/neteq_decoding_test.h" #include "modules/audio_coding/neteq/tools/audio_loop.h" +#include "modules/audio_coding/neteq/tools/audio_sink.h" +#include "modules/audio_coding/neteq/tools/neteq_input.h" #include "modules/audio_coding/neteq/tools/neteq_rtp_dump_input.h" #include "modules/audio_coding/neteq/tools/neteq_test.h" #include "modules/include/module_common_types_public.h" -#include "modules/rtp_rtcp/include/rtcp_statistics.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" -#include "rtc_base/message_digest.h" #include "rtc_base/numerics/safe_conversions.h" #include "rtc_base/strings/string_builder.h" #include "rtc_base/system/arch.h" -#include "test/field_trial.h" #include "test/gtest.h" #include "test/testsupport/file_utils.h" diff --git a/modules/audio_coding/neteq/normal.cc b/modules/audio_coding/neteq/normal.cc index 489aeee339..c3748d8931 100644 --- a/modules/audio_coding/neteq/normal.cc +++ b/modules/audio_coding/neteq/normal.cc @@ -13,8 +13,15 @@ #include // memset, memcpy #include // min +#include +#include +#include "api/array_view.h" +#include "api/neteq/neteq.h" +#include "common_audio/signal_processing/dot_product_with_scale.h" #include "common_audio/signal_processing/include/signal_processing_library.h" +#include "common_audio/signal_processing/include/spl_inl.h" +#include "modules/audio_coding/codecs/cng/webrtc_cng.h" #include "modules/audio_coding/neteq/audio_multi_vector.h" #include "modules/audio_coding/neteq/background_noise.h" #include "modules/audio_coding/neteq/decoder_database.h" diff --git a/modules/audio_coding/neteq/normal_unittest.cc b/modules/audio_coding/neteq/normal_unittest.cc index f774733b11..dfbf0078fd 100644 --- a/modules/audio_coding/neteq/normal_unittest.cc +++ b/modules/audio_coding/neteq/normal_unittest.cc @@ -12,10 +12,11 @@ #include "modules/audio_coding/neteq/normal.h" -#include -#include +#include +#include -#include "common_audio/signal_processing/include/signal_processing_library.h" +#include "api/neteq/neteq.h" +#include "api/neteq/tick_timer.h" #include "modules/audio_coding/neteq/audio_multi_vector.h" #include "modules/audio_coding/neteq/background_noise.h" #include "modules/audio_coding/neteq/expand.h" @@ -24,6 +25,7 @@ #include "modules/audio_coding/neteq/random_vector.h" #include "modules/audio_coding/neteq/statistics_calculator.h" #include "modules/audio_coding/neteq/sync_buffer.h" +#include "test/gmock.h" #include "test/gtest.h" using ::testing::_; diff --git a/modules/audio_coding/neteq/packet.cc b/modules/audio_coding/neteq/packet.cc index 333f161229..e70824ef28 100644 --- a/modules/audio_coding/neteq/packet.cc +++ b/modules/audio_coding/neteq/packet.cc @@ -10,6 +10,8 @@ #include "modules/audio_coding/neteq/packet.h" +#include "rtc_base/checks.h" + namespace webrtc { Packet::Packet() = default; diff --git a/modules/audio_coding/neteq/packet.h b/modules/audio_coding/neteq/packet.h index f6ee755b77..671ab26cb4 100644 --- a/modules/audio_coding/neteq/packet.h +++ b/modules/audio_coding/neteq/packet.h @@ -15,6 +15,7 @@ #include #include +#include #include "api/audio_codecs/audio_decoder.h" #include "api/neteq/tick_timer.h" diff --git a/modules/audio_coding/neteq/packet_arrival_history_unittest.cc b/modules/audio_coding/neteq/packet_arrival_history_unittest.cc index dd95fec0f7..58a0a5108b 100644 --- a/modules/audio_coding/neteq/packet_arrival_history_unittest.cc +++ b/modules/audio_coding/neteq/packet_arrival_history_unittest.cc @@ -13,6 +13,7 @@ #include #include +#include "api/neteq/tick_timer.h" #include "test/gtest.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/packet_buffer.cc b/modules/audio_coding/neteq/packet_buffer.cc index 894fc5bef6..a2e3a7ee8a 100644 --- a/modules/audio_coding/neteq/packet_buffer.cc +++ b/modules/audio_coding/neteq/packet_buffer.cc @@ -15,17 +15,19 @@ #include "modules/audio_coding/neteq/packet_buffer.h" #include +#include +#include #include #include -#include +#include #include #include "api/audio_codecs/audio_decoder.h" #include "api/neteq/tick_timer.h" #include "modules/audio_coding/neteq/decoder_database.h" +#include "modules/audio_coding/neteq/packet.h" #include "modules/audio_coding/neteq/statistics_calculator.h" #include "rtc_base/checks.h" -#include "rtc_base/experiments/struct_parameters_parser.h" #include "rtc_base/logging.h" #include "rtc_base/numerics/safe_conversions.h" diff --git a/modules/audio_coding/neteq/packet_buffer.h b/modules/audio_coding/neteq/packet_buffer.h index 547e993a4d..d7c98288ea 100644 --- a/modules/audio_coding/neteq/packet_buffer.h +++ b/modules/audio_coding/neteq/packet_buffer.h @@ -11,6 +11,8 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_ #define MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_ +#include +#include #include #include "modules/audio_coding/neteq/decoder_database.h" diff --git a/modules/audio_coding/neteq/packet_buffer_unittest.cc b/modules/audio_coding/neteq/packet_buffer_unittest.cc index 6e134a17ae..5713a6d01d 100644 --- a/modules/audio_coding/neteq/packet_buffer_unittest.cc +++ b/modules/audio_coding/neteq/packet_buffer_unittest.cc @@ -12,14 +12,21 @@ #include "modules/audio_coding/neteq/packet_buffer.h" +#include +#include +#include #include +#include +#include +#include -#include "api/audio_codecs/builtin_audio_decoder_factory.h" +#include "api/array_view.h" +#include "api/audio_codecs/audio_decoder.h" #include "api/neteq/tick_timer.h" #include "modules/audio_coding/neteq/mock/mock_decoder_database.h" #include "modules/audio_coding/neteq/mock/mock_statistics_calculator.h" #include "modules/audio_coding/neteq/packet.h" -#include "test/field_trial.h" +#include "rtc_base/checks.h" #include "test/gmock.h" #include "test/gtest.h" diff --git a/modules/audio_coding/neteq/preemptive_expand.cc b/modules/audio_coding/neteq/preemptive_expand.cc index f5d540bf06..a183c3dd96 100644 --- a/modules/audio_coding/neteq/preemptive_expand.cc +++ b/modules/audio_coding/neteq/preemptive_expand.cc @@ -11,6 +11,8 @@ #include "modules/audio_coding/neteq/preemptive_expand.h" #include +#include +#include #include "api/array_view.h" #include "modules/audio_coding/neteq/audio_multi_vector.h" diff --git a/modules/audio_coding/neteq/random_vector.cc b/modules/audio_coding/neteq/random_vector.cc index ada175831c..f57ede6a85 100644 --- a/modules/audio_coding/neteq/random_vector.cc +++ b/modules/audio_coding/neteq/random_vector.cc @@ -10,6 +10,9 @@ #include "modules/audio_coding/neteq/random_vector.h" +#include +#include + namespace webrtc { const int16_t RandomVector::kRandomTable[RandomVector::kRandomTableSize] = { diff --git a/modules/audio_coding/neteq/red_payload_splitter.cc b/modules/audio_coding/neteq/red_payload_splitter.cc index 4c753f553e..d8c27db218 100644 --- a/modules/audio_coding/neteq/red_payload_splitter.cc +++ b/modules/audio_coding/neteq/red_payload_splitter.cc @@ -20,6 +20,7 @@ #include "modules/audio_coding/neteq/decoder_database.h" #include "modules/audio_coding/neteq/packet.h" #include "rtc_base/buffer.h" +#include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/numerics/safe_conversions.h" diff --git a/modules/audio_coding/neteq/red_payload_splitter.h b/modules/audio_coding/neteq/red_payload_splitter.h index 2f48e4b7d4..db637c204d 100644 --- a/modules/audio_coding/neteq/red_payload_splitter.h +++ b/modules/audio_coding/neteq/red_payload_splitter.h @@ -11,6 +11,8 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_RED_PAYLOAD_SPLITTER_H_ #define MODULES_AUDIO_CODING_NETEQ_RED_PAYLOAD_SPLITTER_H_ +#include + #include "modules/audio_coding/neteq/packet.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/red_payload_splitter_unittest.cc b/modules/audio_coding/neteq/red_payload_splitter_unittest.cc index 7a5dde0056..c902edb8ac 100644 --- a/modules/audio_coding/neteq/red_payload_splitter_unittest.cc +++ b/modules/audio_coding/neteq/red_payload_splitter_unittest.cc @@ -12,14 +12,19 @@ #include "modules/audio_coding/neteq/red_payload_splitter.h" -#include +#include +#include +#include +#include #include // pair -#include "api/audio_codecs/builtin_audio_decoder_factory.h" +#include "api/audio_codecs/audio_format.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" +#include "api/make_ref_counted.h" #include "modules/audio_coding/neteq/decoder_database.h" #include "modules/audio_coding/neteq/packet.h" +#include "rtc_base/checks.h" #include "rtc_base/numerics/safe_conversions.h" #include "test/gtest.h" #include "test/mock_audio_decoder_factory.h" diff --git a/modules/audio_coding/neteq/reorder_optimizer.cc b/modules/audio_coding/neteq/reorder_optimizer.cc index b3d9154a9a..98dbedfaa4 100644 --- a/modules/audio_coding/neteq/reorder_optimizer.cc +++ b/modules/audio_coding/neteq/reorder_optimizer.cc @@ -14,6 +14,7 @@ #include #include +#include #include namespace webrtc { diff --git a/modules/audio_coding/neteq/statistics_calculator.cc b/modules/audio_coding/neteq/statistics_calculator.cc index 69e14608a0..1ae4221a64 100644 --- a/modules/audio_coding/neteq/statistics_calculator.cc +++ b/modules/audio_coding/neteq/statistics_calculator.cc @@ -13,9 +13,11 @@ #include // memset #include +#include #include "absl/strings/string_view.h" -#include "modules/audio_coding/neteq/delay_manager.h" +#include "api/neteq/neteq.h" +#include "api/neteq/tick_timer.h" #include "rtc_base/checks.h" #include "rtc_base/numerics/safe_conversions.h" #include "system_wrappers/include/metrics.h" diff --git a/modules/audio_coding/neteq/statistics_calculator.h b/modules/audio_coding/neteq/statistics_calculator.h index 28ac613bc7..07c8345462 100644 --- a/modules/audio_coding/neteq/statistics_calculator.h +++ b/modules/audio_coding/neteq/statistics_calculator.h @@ -11,11 +11,14 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_STATISTICS_CALCULATOR_H_ #define MODULES_AUDIO_CODING_NETEQ_STATISTICS_CALCULATOR_H_ +#include +#include #include #include #include "absl/strings/string_view.h" #include "api/neteq/neteq.h" +#include "api/neteq/tick_timer.h" #include "modules/audio_coding/neteq/expand_uma_logger.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/statistics_calculator_unittest.cc b/modules/audio_coding/neteq/statistics_calculator_unittest.cc index a8ad359ab0..110c2db5d6 100644 --- a/modules/audio_coding/neteq/statistics_calculator_unittest.cc +++ b/modules/audio_coding/neteq/statistics_calculator_unittest.cc @@ -10,6 +10,10 @@ #include "modules/audio_coding/neteq/statistics_calculator.h" +#include + +#include "api/neteq/neteq.h" +#include "api/neteq/tick_timer.h" #include "test/gtest.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/sync_buffer.cc b/modules/audio_coding/neteq/sync_buffer.cc index 5144462cae..87d56baf21 100644 --- a/modules/audio_coding/neteq/sync_buffer.cc +++ b/modules/audio_coding/neteq/sync_buffer.cc @@ -11,7 +11,12 @@ #include "modules/audio_coding/neteq/sync_buffer.h" #include // Access to min. +#include +#include +#include "api/audio/audio_frame.h" +#include "modules/audio_coding/neteq/audio_multi_vector.h" +#include "rtc_base/buffer.h" #include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/sync_buffer_unittest.cc b/modules/audio_coding/neteq/sync_buffer_unittest.cc index f4057cf3bc..d2137e3f7b 100644 --- a/modules/audio_coding/neteq/sync_buffer_unittest.cc +++ b/modules/audio_coding/neteq/sync_buffer_unittest.cc @@ -10,6 +10,11 @@ #include "modules/audio_coding/neteq/sync_buffer.h" +#include +#include + +#include "api/audio/audio_frame.h" +#include "modules/audio_coding/neteq/audio_multi_vector.h" #include "rtc_base/numerics/safe_conversions.h" #include "test/gtest.h" diff --git a/modules/audio_coding/neteq/test/neteq_decoding_test.cc b/modules/audio_coding/neteq/test/neteq_decoding_test.cc index aca3efc95f..d67888be14 100644 --- a/modules/audio_coding/neteq/test/neteq_decoding_test.cc +++ b/modules/audio_coding/neteq/test/neteq_decoding_test.cc @@ -10,14 +10,26 @@ #include "modules/audio_coding/neteq/test/neteq_decoding_test.h" +#include +#include +#include +#include +#include + #include "absl/strings/string_view.h" +#include "api/array_view.h" +#include "api/audio/audio_frame.h" +#include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/environment/environment_factory.h" #include "api/neteq/default_neteq_factory.h" +#include "api/neteq/neteq.h" #include "api/rtp_headers.h" #include "api/units/timestamp.h" #include "modules/audio_coding/neteq/test/result_sink.h" +#include "modules/audio_coding/neteq/tools/rtp_file_source.h" #include "rtc_base/strings/string_builder.h" +#include "test/gtest.h" #include "test/testsupport/file_utils.h" #ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT @@ -25,7 +37,7 @@ #ifdef WEBRTC_ANDROID_PLATFORM_BUILD #include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h" #else -#include "modules/audio_coding/neteq/neteq_unittest.pb.h" +#include "modules/audio_coding/neteq/neteq_unittest.pb.h" // IWYU pragma: keep #endif #endif diff --git a/modules/audio_coding/neteq/test/neteq_decoding_test.h b/modules/audio_coding/neteq/test/neteq_decoding_test.h index fa348dbc48..a7557dcf0a 100644 --- a/modules/audio_coding/neteq/test/neteq_decoding_test.h +++ b/modules/audio_coding/neteq/test/neteq_decoding_test.h @@ -11,9 +11,10 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_TEST_NETEQ_DECODING_TEST_H_ #define MODULES_AUDIO_CODING_NETEQ_TEST_NETEQ_DECODING_TEST_H_ +#include +#include #include #include -#include #include "absl/strings/string_view.h" #include "api/audio/audio_frame.h" diff --git a/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc b/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc index 59be7cdbf9..e7496a7c3b 100644 --- a/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc +++ b/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc @@ -8,10 +8,20 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include +#include +#include + #include "absl/flags/flag.h" +#include "api/array_view.h" +#include "api/audio_codecs/audio_format.h" +#include "api/rtp_parameters.h" #include "modules/audio_coding/codecs/opus/opus_inst.h" #include "modules/audio_coding/codecs/opus/opus_interface.h" #include "modules/audio_coding/neteq/tools/neteq_quality_test.h" +#include "rtc_base/buffer.h" +#include "rtc_base/checks.h" +#include "test/gtest.h" ABSL_FLAG(int, bit_rate_kbps, 32, "Target bit rate (kbps)."); diff --git a/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc b/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc index 53af21f8ff..6cf73f6bda 100644 --- a/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc +++ b/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc @@ -8,13 +8,20 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include +#include #include #include "absl/flags/flag.h" +#include "api/array_view.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/audio_codecs/audio_format.h" #include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h" #include "modules/audio_coding/neteq/tools/neteq_quality_test.h" +#include "rtc_base/buffer.h" #include "rtc_base/checks.h" #include "rtc_base/numerics/safe_conversions.h" +#include "test/gtest.h" #include "test/testsupport/file_utils.h" ABSL_FLAG(int, frame_size_ms, 20, "Codec frame size (milliseconds)."); diff --git a/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc b/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc index 62ea361d0a..2de5170c26 100644 --- a/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc +++ b/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc @@ -8,13 +8,20 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include +#include #include #include "absl/flags/flag.h" +#include "api/array_view.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/audio_codecs/audio_format.h" #include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h" #include "modules/audio_coding/neteq/tools/neteq_quality_test.h" +#include "rtc_base/buffer.h" #include "rtc_base/checks.h" #include "rtc_base/numerics/safe_conversions.h" +#include "test/gtest.h" #include "test/testsupport/file_utils.h" ABSL_FLAG(int, frame_size_ms, 20, "Codec frame size (milliseconds)."); diff --git a/modules/audio_coding/neteq/test/neteq_performance_unittest.cc b/modules/audio_coding/neteq/test/neteq_performance_unittest.cc index 1b453cf7bf..ad2d42f18c 100644 --- a/modules/audio_coding/neteq/test/neteq_performance_unittest.cc +++ b/modules/audio_coding/neteq/test/neteq_performance_unittest.cc @@ -8,6 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include + #include "absl/flags/flag.h" #include "api/test/metrics/global_metrics_logger_and_exporter.h" #include "api/test/metrics/metric.h" diff --git a/modules/audio_coding/neteq/test/neteq_speed_test.cc b/modules/audio_coding/neteq/test/neteq_speed_test.cc index a72b2009eb..0ffee03aed 100644 --- a/modules/audio_coding/neteq/test/neteq_speed_test.cc +++ b/modules/audio_coding/neteq/test/neteq_speed_test.cc @@ -10,7 +10,9 @@ #include +#include #include +#include #include #include "absl/flags/flag.h" diff --git a/modules/audio_coding/neteq/time_stretch.cc b/modules/audio_coding/neteq/time_stretch.cc index d6dddd5317..39be0404cd 100644 --- a/modules/audio_coding/neteq/time_stretch.cc +++ b/modules/audio_coding/neteq/time_stretch.cc @@ -11,12 +11,17 @@ #include "modules/audio_coding/neteq/time_stretch.h" #include // min, max +#include +#include #include +#include "common_audio/signal_processing/dot_product_with_scale.h" #include "common_audio/signal_processing/include/signal_processing_library.h" +#include "common_audio/signal_processing/include/spl_inl.h" #include "modules/audio_coding/neteq/background_noise.h" #include "modules/audio_coding/neteq/cross_correlation.h" #include "modules/audio_coding/neteq/dsp_helper.h" +#include "rtc_base/checks.h" #include "rtc_base/numerics/safe_conversions.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/time_stretch.h b/modules/audio_coding/neteq/time_stretch.h index f0ddaebeca..9d81a7f55f 100644 --- a/modules/audio_coding/neteq/time_stretch.h +++ b/modules/audio_coding/neteq/time_stretch.h @@ -13,7 +13,10 @@ #include // memset, size_t +#include + #include "modules/audio_coding/neteq/audio_multi_vector.h" +#include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/time_stretch_unittest.cc b/modules/audio_coding/neteq/time_stretch_unittest.cc index da3a98229a..e53bd49724 100644 --- a/modules/audio_coding/neteq/time_stretch_unittest.cc +++ b/modules/audio_coding/neteq/time_stretch_unittest.cc @@ -10,10 +10,13 @@ // Unit tests for Accelerate and PreemptiveExpand classes. +#include "modules/audio_coding/neteq/time_stretch.h" + +#include +#include #include #include -#include "common_audio/signal_processing/include/signal_processing_library.h" #include "modules/audio_coding/neteq/accelerate.h" #include "modules/audio_coding/neteq/background_noise.h" #include "modules/audio_coding/neteq/preemptive_expand.h" diff --git a/modules/audio_coding/neteq/timestamp_scaler.cc b/modules/audio_coding/neteq/timestamp_scaler.cc index 59177d027f..6bb546c026 100644 --- a/modules/audio_coding/neteq/timestamp_scaler.cc +++ b/modules/audio_coding/neteq/timestamp_scaler.cc @@ -10,8 +10,11 @@ #include "modules/audio_coding/neteq/timestamp_scaler.h" +#include + #include "api/audio_codecs/audio_format.h" #include "modules/audio_coding/neteq/decoder_database.h" +#include "modules/audio_coding/neteq/packet.h" #include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/timestamp_scaler.h b/modules/audio_coding/neteq/timestamp_scaler.h index f42ce7207a..833e09be44 100644 --- a/modules/audio_coding/neteq/timestamp_scaler.h +++ b/modules/audio_coding/neteq/timestamp_scaler.h @@ -11,6 +11,8 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_TIMESTAMP_SCALER_H_ #define MODULES_AUDIO_CODING_NETEQ_TIMESTAMP_SCALER_H_ +#include + #include "modules/audio_coding/neteq/packet.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/timestamp_scaler_unittest.cc b/modules/audio_coding/neteq/timestamp_scaler_unittest.cc index e880004a5b..f5d6411ead 100644 --- a/modules/audio_coding/neteq/timestamp_scaler_unittest.cc +++ b/modules/audio_coding/neteq/timestamp_scaler_unittest.cc @@ -10,6 +10,12 @@ #include "modules/audio_coding/neteq/timestamp_scaler.h" +#include +#include +#include +#include + +#include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" diff --git a/modules/audio_coding/neteq/tools/audio_loop.cc b/modules/audio_coding/neteq/tools/audio_loop.cc index cfac0f3300..cff9fd1d1d 100644 --- a/modules/audio_coding/neteq/tools/audio_loop.cc +++ b/modules/audio_coding/neteq/tools/audio_loop.cc @@ -13,7 +13,11 @@ #include #include +#include +#include + #include "absl/strings/string_view.h" +#include "api/array_view.h" namespace webrtc { namespace test { diff --git a/modules/audio_coding/neteq/tools/audio_loop.h b/modules/audio_coding/neteq/tools/audio_loop.h index d722e01666..d5c561d623 100644 --- a/modules/audio_coding/neteq/tools/audio_loop.h +++ b/modules/audio_coding/neteq/tools/audio_loop.h @@ -11,8 +11,9 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_LOOP_H_ #define MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_LOOP_H_ +#include +#include #include -#include #include "absl/strings/string_view.h" #include "api/array_view.h" diff --git a/modules/audio_coding/neteq/tools/audio_sink.cc b/modules/audio_coding/neteq/tools/audio_sink.cc index 656dda43a1..84ab2d6784 100644 --- a/modules/audio_coding/neteq/tools/audio_sink.cc +++ b/modules/audio_coding/neteq/tools/audio_sink.cc @@ -10,6 +10,9 @@ #include "modules/audio_coding/neteq/tools/audio_sink.h" +#include +#include + namespace webrtc { namespace test { diff --git a/modules/audio_coding/neteq/tools/audio_sink.h b/modules/audio_coding/neteq/tools/audio_sink.h index 53729fa920..30df6c27d6 100644 --- a/modules/audio_coding/neteq/tools/audio_sink.h +++ b/modules/audio_coding/neteq/tools/audio_sink.h @@ -11,6 +11,9 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_ #define MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_ +#include +#include + #include "api/audio/audio_frame.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc b/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc index d63e1b05e1..20cd8356d6 100644 --- a/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc +++ b/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc @@ -10,11 +10,15 @@ #include "modules/audio_coding/neteq/tools/constant_pcm_packet_source.h" -#include +#include +#include +#include +#include #include "modules/audio_coding/codecs/pcm16b/pcm16b.h" #include "modules/audio_coding/neteq/tools/packet.h" #include "rtc_base/checks.h" +#include "rtc_base/copy_on_write_buffer.h" namespace webrtc { namespace test { diff --git a/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h b/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h index ab4f5c2281..4727aff583 100644 --- a/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h +++ b/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h @@ -13,8 +13,10 @@ #include -#include +#include +#include +#include "modules/audio_coding/neteq/tools/packet.h" #include "modules/audio_coding/neteq/tools/packet_source.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/tools/encode_neteq_input.cc b/modules/audio_coding/neteq/tools/encode_neteq_input.cc index b4a940bdaf..c76dbbb7e2 100644 --- a/modules/audio_coding/neteq/tools/encode_neteq_input.cc +++ b/modules/audio_coding/neteq/tools/encode_neteq_input.cc @@ -10,8 +10,15 @@ #include "modules/audio_coding/neteq/tools/encode_neteq_input.h" +#include +#include +#include +#include #include +#include "api/audio_codecs/audio_encoder.h" +#include "api/rtp_headers.h" +#include "modules/audio_coding/neteq/tools/neteq_input.h" #include "rtc_base/checks.h" #include "rtc_base/numerics/safe_conversions.h" diff --git a/modules/audio_coding/neteq/tools/encode_neteq_input.h b/modules/audio_coding/neteq/tools/encode_neteq_input.h index 63666e86d8..eb8fea7991 100644 --- a/modules/audio_coding/neteq/tools/encode_neteq_input.h +++ b/modules/audio_coding/neteq/tools/encode_neteq_input.h @@ -11,9 +11,14 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_ #define MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_ +#include +#include #include +#include +#include "api/array_view.h" #include "api/audio_codecs/audio_encoder.h" +#include "api/rtp_headers.h" #include "modules/audio_coding/neteq/tools/neteq_input.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/tools/fake_decode_from_file.cc b/modules/audio_coding/neteq/tools/fake_decode_from_file.cc index 95ce8e9948..37e79fd63c 100644 --- a/modules/audio_coding/neteq/tools/fake_decode_from_file.cc +++ b/modules/audio_coding/neteq/tools/fake_decode_from_file.cc @@ -10,7 +10,18 @@ #include "modules/audio_coding/neteq/tools/fake_decode_from_file.h" +#include +#include +#include +#include +#include +#include + +#include "api/array_view.h" +#include "api/audio_codecs/audio_decoder.h" +#include "modules/audio_coding/neteq/tools/input_audio_file.h" #include "modules/rtp_rtcp/source/byte_io.h" +#include "rtc_base/buffer.h" #include "rtc_base/checks.h" #include "rtc_base/numerics/safe_conversions.h" diff --git a/modules/audio_coding/neteq/tools/fake_decode_from_file.h b/modules/audio_coding/neteq/tools/fake_decode_from_file.h index ea1610e3ec..a5e734f3b8 100644 --- a/modules/audio_coding/neteq/tools/fake_decode_from_file.h +++ b/modules/audio_coding/neteq/tools/fake_decode_from_file.h @@ -11,12 +11,17 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_ #define MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_ +#include +#include #include #include +#include +#include #include "api/array_view.h" #include "api/audio_codecs/audio_decoder.h" #include "modules/audio_coding/neteq/tools/input_audio_file.h" +#include "rtc_base/buffer.h" namespace webrtc { namespace test { diff --git a/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.cc b/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.cc index c41ef590bd..b2bb4f1a92 100644 --- a/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.cc +++ b/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.cc @@ -10,10 +10,14 @@ #include "modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.h" -#include +#include #include +#include #include +#include "api/rtp_headers.h" +#include "modules/audio_coding/neteq/tools/neteq_input.h" +#include "rtc_base/buffer.h" #include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.h b/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.h index f72c21c70f..47939de66b 100644 --- a/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.h +++ b/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.h @@ -11,10 +11,11 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_INITIAL_PACKET_INSERTER_NETEQ_INPUT_H_ #define MODULES_AUDIO_CODING_NETEQ_TOOLS_INITIAL_PACKET_INSERTER_NETEQ_INPUT_H_ -#include +#include #include -#include +#include +#include "api/rtp_headers.h" #include "modules/audio_coding/neteq/tools/neteq_input.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/tools/input_audio_file.cc b/modules/audio_coding/neteq/tools/input_audio_file.cc index b077dbff21..e0000097aa 100644 --- a/modules/audio_coding/neteq/tools/input_audio_file.cc +++ b/modules/audio_coding/neteq/tools/input_audio_file.cc @@ -10,6 +10,11 @@ #include "modules/audio_coding/neteq/tools/input_audio_file.h" +#include +#include +#include +#include + #include "absl/strings/string_view.h" #include "rtc_base/checks.h" diff --git a/modules/audio_coding/neteq/tools/input_audio_file.h b/modules/audio_coding/neteq/tools/input_audio_file.h index e885bb39eb..4fe5b43a3e 100644 --- a/modules/audio_coding/neteq/tools/input_audio_file.h +++ b/modules/audio_coding/neteq/tools/input_audio_file.h @@ -14,7 +14,6 @@ #include #include -#include #include "absl/strings/string_view.h" diff --git a/modules/audio_coding/neteq/tools/input_audio_file_unittest.cc b/modules/audio_coding/neteq/tools/input_audio_file_unittest.cc index 54d5b3fbfa..91cbb2745e 100644 --- a/modules/audio_coding/neteq/tools/input_audio_file_unittest.cc +++ b/modules/audio_coding/neteq/tools/input_audio_file_unittest.cc @@ -12,6 +12,9 @@ #include "modules/audio_coding/neteq/tools/input_audio_file.h" +#include +#include + #include "rtc_base/numerics/safe_conversions.h" #include "test/gtest.h" diff --git a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc index 71e388066f..894eef922c 100644 --- a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc +++ b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc @@ -11,13 +11,21 @@ #include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h" #include +#include +#include #include #include -#include #include +#include // no-presubmit-check TODO(webrtc:8982) +#include #include +#include #include "absl/strings/string_view.h" +#include "api/audio/audio_frame.h" +#include "api/neteq/neteq.h" +#include "api/rtp_packet_info.h" +#include "modules/audio_coding/neteq/tools/neteq_input.h" #include "rtc_base/checks.h" #include "rtc_base/numerics/sequence_number_unwrapper.h" diff --git a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h index cdf178b463..caa3302bb7 100644 --- a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h +++ b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h @@ -11,13 +11,17 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_DELAY_ANALYZER_H_ #define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_DELAY_ANALYZER_H_ +#include +#include #include #include #include -#include +#include #include #include "absl/strings/string_view.h" +#include "api/audio/audio_frame.h" +#include "api/neteq/neteq.h" #include "modules/audio_coding/neteq/tools/neteq_input.h" #include "modules/audio_coding/neteq/tools/neteq_test.h" diff --git a/modules/audio_coding/neteq/tools/neteq_event_log_input.cc b/modules/audio_coding/neteq/tools/neteq_event_log_input.cc index d42a31b7d9..9066bd6323 100644 --- a/modules/audio_coding/neteq/tools/neteq_event_log_input.cc +++ b/modules/audio_coding/neteq/tools/neteq_event_log_input.cc @@ -10,11 +10,18 @@ #include "modules/audio_coding/neteq/tools/neteq_event_log_input.h" -#include +#include +#include #include - -#include "absl/strings/string_view.h" -#include "rtc_base/checks.h" +#include +#include + +#include "api/rtp_headers.h" +#include "logging/rtc_event_log/events/logged_rtp_rtcp.h" +#include "logging/rtc_event_log/events/rtc_event_audio_playout.h" +#include "logging/rtc_event_log/events/rtc_event_neteq_set_minimum_delay.h" +#include "logging/rtc_event_log/rtc_event_log_parser.h" +#include "modules/audio_coding/neteq/tools/neteq_input.h" namespace webrtc { namespace test { diff --git a/modules/audio_coding/neteq/tools/neteq_event_log_input.h b/modules/audio_coding/neteq/tools/neteq_event_log_input.h index cdfd62d80f..23c6388ab0 100644 --- a/modules/audio_coding/neteq/tools/neteq_event_log_input.h +++ b/modules/audio_coding/neteq/tools/neteq_event_log_input.h @@ -11,9 +11,10 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EVENT_LOG_INPUT_H_ #define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EVENT_LOG_INPUT_H_ +#include +#include #include -#include "absl/strings/string_view.h" #include "logging/rtc_event_log/rtc_event_log_parser.h" #include "modules/audio_coding/neteq/tools/neteq_input.h" diff --git a/modules/audio_coding/neteq/tools/neteq_input.cc b/modules/audio_coding/neteq/tools/neteq_input.cc index 707ce1b4c8..8c27b8e014 100644 --- a/modules/audio_coding/neteq/tools/neteq_input.cc +++ b/modules/audio_coding/neteq/tools/neteq_input.cc @@ -10,6 +10,13 @@ #include "modules/audio_coding/neteq/tools/neteq_input.h" +#include +#include +#include +#include +#include + +#include "api/rtp_headers.h" #include "rtc_base/strings/string_builder.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/tools/neteq_input.h b/modules/audio_coding/neteq/tools/neteq_input.h index 0f1b090e94..269d7b6b1f 100644 --- a/modules/audio_coding/neteq/tools/neteq_input.h +++ b/modules/audio_coding/neteq/tools/neteq_input.h @@ -12,12 +12,12 @@ #define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_ #include +#include #include #include #include -#include "modules/audio_coding/neteq/tools/packet.h" -#include "modules/audio_coding/neteq/tools/packet_source.h" +#include "api/rtp_headers.h" #include "rtc_base/buffer.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/tools/neteq_performance_test.cc b/modules/audio_coding/neteq/tools/neteq_performance_test.cc index c3205ded1c..caae9ecc26 100644 --- a/modules/audio_coding/neteq/tools/neteq_performance_test.cc +++ b/modules/audio_coding/neteq/tools/neteq_performance_test.cc @@ -10,12 +10,20 @@ #include "modules/audio_coding/neteq/tools/neteq_performance_test.h" +#include +#include +#include +#include + #include "api/audio/audio_frame.h" +#include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" #include "api/neteq/default_neteq_factory.h" #include "api/neteq/neteq.h" +#include "api/rtp_headers.h" +#include "api/units/timestamp.h" #include "modules/audio_coding/codecs/pcm16b/pcm16b.h" #include "modules/audio_coding/neteq/tools/audio_loop.h" #include "modules/audio_coding/neteq/tools/rtp_generator.h" diff --git a/modules/audio_coding/neteq/tools/neteq_quality_test.cc b/modules/audio_coding/neteq/tools/neteq_quality_test.cc index 4ddef25bab..55ef65e6b4 100644 --- a/modules/audio_coding/neteq/tools/neteq_quality_test.cc +++ b/modules/audio_coding/neteq/tools/neteq_quality_test.cc @@ -12,20 +12,39 @@ #include +#include +#include #include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include #include "absl/flags/flag.h" #include "absl/strings/string_view.h" +#include "api/array_view.h" +#include "api/audio_codecs/audio_decoder_factory.h" +#include "api/audio_codecs/audio_format.h" #include "api/environment/environment_factory.h" #include "api/neteq/default_neteq_factory.h" +#include "api/neteq/neteq.h" +#include "api/scoped_refptr.h" #include "api/units/timestamp.h" #include "modules/audio_coding/neteq/tools/neteq_quality_test.h" #include "modules/audio_coding/neteq/tools/output_audio_file.h" #include "modules/audio_coding/neteq/tools/output_wav_file.h" #include "modules/audio_coding/neteq/tools/resample_input_audio_file.h" +#include "modules/audio_coding/neteq/tools/rtp_generator.h" #include "rtc_base/checks.h" #include "rtc_base/string_encode.h" -#include "system_wrappers/include/clock.h" +#include "test/gtest.h" #include "test/testsupport/file_utils.h" ABSL_FLAG(std::string, diff --git a/modules/audio_coding/neteq/tools/neteq_quality_test.h b/modules/audio_coding/neteq/tools/neteq_quality_test.h index d995095d99..9b84e239f2 100644 --- a/modules/audio_coding/neteq/tools/neteq_quality_test.h +++ b/modules/audio_coding/neteq/tools/neteq_quality_test.h @@ -11,15 +11,22 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_ #define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_ +#include +#include #include #include +#include +#include "api/audio_codecs/audio_decoder_factory.h" +#include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/neteq/neteq.h" +#include "api/rtp_headers.h" +#include "api/scoped_refptr.h" #include "modules/audio_coding/neteq/tools/audio_sink.h" #include "modules/audio_coding/neteq/tools/input_audio_file.h" #include "modules/audio_coding/neteq/tools/rtp_generator.h" -#include "system_wrappers/include/clock.h" +#include "rtc_base/buffer.h" #include "test/gtest.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/tools/neteq_replacement_input.cc b/modules/audio_coding/neteq/tools/neteq_replacement_input.cc index 2fb53d8b3f..374279fb2e 100644 --- a/modules/audio_coding/neteq/tools/neteq_replacement_input.cc +++ b/modules/audio_coding/neteq/tools/neteq_replacement_input.cc @@ -10,7 +10,16 @@ #include "modules/audio_coding/neteq/tools/neteq_replacement_input.h" +#include +#include +#include +#include +#include +#include + +#include "api/rtp_headers.h" #include "modules/audio_coding/neteq/tools/fake_decode_from_file.h" +#include "modules/audio_coding/neteq/tools/neteq_input.h" #include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/tools/neteq_replacement_input.h b/modules/audio_coding/neteq/tools/neteq_replacement_input.h index 33d8138812..a7efbc1f16 100644 --- a/modules/audio_coding/neteq/tools/neteq_replacement_input.h +++ b/modules/audio_coding/neteq/tools/neteq_replacement_input.h @@ -11,9 +11,12 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_REPLACEMENT_INPUT_H_ #define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_REPLACEMENT_INPUT_H_ +#include #include +#include #include +#include "api/rtp_headers.h" #include "modules/audio_coding/neteq/tools/neteq_input.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/tools/neteq_rtp_dump_input.cc b/modules/audio_coding/neteq/tools/neteq_rtp_dump_input.cc index b3c29c1b8e..29831dca6a 100644 --- a/modules/audio_coding/neteq/tools/neteq_rtp_dump_input.cc +++ b/modules/audio_coding/neteq/tools/neteq_rtp_dump_input.cc @@ -10,8 +10,18 @@ #include "modules/audio_coding/neteq/tools/neteq_rtp_dump_input.h" +#include +#include +#include +#include +#include + #include "absl/strings/string_view.h" +#include "api/rtp_headers.h" +#include "modules/audio_coding/neteq/tools/neteq_input.h" +#include "modules/audio_coding/neteq/tools/packet.h" #include "modules/audio_coding/neteq/tools/rtp_file_source.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" namespace webrtc { namespace test { diff --git a/modules/audio_coding/neteq/tools/neteq_rtp_dump_input.h b/modules/audio_coding/neteq/tools/neteq_rtp_dump_input.h index da03732a2a..771be08036 100644 --- a/modules/audio_coding/neteq/tools/neteq_rtp_dump_input.h +++ b/modules/audio_coding/neteq/tools/neteq_rtp_dump_input.h @@ -11,6 +11,7 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_RTP_DUMP_INPUT_H_ #define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_RTP_DUMP_INPUT_H_ +#include #include #include #include diff --git a/modules/audio_coding/neteq/tools/neteq_stats_getter.cc b/modules/audio_coding/neteq/tools/neteq_stats_getter.cc index 2c4e7bc6a4..4c3ced72ee 100644 --- a/modules/audio_coding/neteq/tools/neteq_stats_getter.cc +++ b/modules/audio_coding/neteq/tools/neteq_stats_getter.cc @@ -11,9 +11,15 @@ #include "modules/audio_coding/neteq/tools/neteq_stats_getter.h" #include +#include +#include #include +#include #include +#include "api/audio/audio_frame.h" +#include "api/neteq/neteq.h" +#include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h" #include "rtc_base/checks.h" #include "rtc_base/strings/string_builder.h" #include "rtc_base/time_utils.h" diff --git a/modules/audio_coding/neteq/tools/neteq_stats_getter.h b/modules/audio_coding/neteq/tools/neteq_stats_getter.h index b1b12bb1f8..83ed4f3afa 100644 --- a/modules/audio_coding/neteq/tools/neteq_stats_getter.h +++ b/modules/audio_coding/neteq/tools/neteq_stats_getter.h @@ -11,10 +11,15 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_STATS_GETTER_H_ #define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_STATS_GETTER_H_ +#include +#include #include #include +#include #include +#include "api/audio/audio_frame.h" +#include "api/neteq/neteq.h" #include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h" #include "modules/audio_coding/neteq/tools/neteq_test.h" diff --git a/modules/audio_coding/neteq/tools/neteq_stats_plotter.cc b/modules/audio_coding/neteq/tools/neteq_stats_plotter.cc index 47bcd314bc..f207199226 100644 --- a/modules/audio_coding/neteq/tools/neteq_stats_plotter.cc +++ b/modules/audio_coding/neteq/tools/neteq_stats_plotter.cc @@ -13,9 +13,14 @@ #include #include +#include +#include +#include #include #include "absl/strings/string_view.h" +#include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h" +#include "modules/audio_coding/neteq/tools/neteq_stats_getter.h" namespace webrtc { namespace test { diff --git a/modules/audio_coding/neteq/tools/neteq_stats_plotter.h b/modules/audio_coding/neteq/tools/neteq_stats_plotter.h index 11c16da9d1..d1f5e67a50 100644 --- a/modules/audio_coding/neteq/tools/neteq_stats_plotter.h +++ b/modules/audio_coding/neteq/tools/neteq_stats_plotter.h @@ -11,11 +11,11 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_STATS_PLOTTER_H_ #define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_STATS_PLOTTER_H_ +#include #include #include #include "absl/strings/string_view.h" -#include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h" #include "modules/audio_coding/neteq/tools/neteq_stats_getter.h" #include "modules/audio_coding/neteq/tools/neteq_test.h" diff --git a/modules/audio_coding/neteq/tools/neteq_test.cc b/modules/audio_coding/neteq/tools/neteq_test.cc index 67cd892f85..2baea86b46 100644 --- a/modules/audio_coding/neteq/tools/neteq_test.cc +++ b/modules/audio_coding/neteq/tools/neteq_test.cc @@ -10,15 +10,34 @@ #include "modules/audio_coding/neteq/tools/neteq_test.h" +#include +#include +#include #include #include +#include +#include +#include +#include +#include "absl/strings/string_view.h" +#include "api/array_view.h" +#include "api/audio/audio_frame.h" +#include "api/audio_codecs/audio_decoder_factory.h" +#include "api/audio_codecs/audio_format.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" #include "api/field_trials.h" #include "api/neteq/default_neteq_factory.h" +#include "api/neteq/neteq.h" +#include "api/neteq/neteq_factory.h" +#include "api/scoped_refptr.h" +#include "api/test/neteq_simulator.h" #include "api/units/timestamp.h" +#include "modules/audio_coding/neteq/tools/audio_sink.h" +#include "modules/audio_coding/neteq/tools/neteq_input.h" #include "modules/rtp_rtcp/source/byte_io.h" +#include "rtc_base/checks.h" #include "system_wrappers/include/clock.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/tools/neteq_test.h b/modules/audio_coding/neteq/tools/neteq_test.h index b576b3fea8..8717cd0424 100644 --- a/modules/audio_coding/neteq/tools/neteq_test.h +++ b/modules/audio_coding/neteq/tools/neteq_test.h @@ -11,17 +11,19 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_H_ #define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_H_ +#include #include #include #include #include -#include -#include +#include "absl/strings/string_view.h" #include "api/audio_codecs/audio_decoder_factory.h" +#include "api/audio_codecs/audio_format.h" #include "api/environment/environment.h" #include "api/neteq/neteq.h" #include "api/neteq/neteq_factory.h" +#include "api/scoped_refptr.h" #include "api/test/neteq_simulator.h" #include "modules/audio_coding/neteq/tools/audio_sink.h" #include "modules/audio_coding/neteq/tools/neteq_input.h" diff --git a/modules/audio_coding/neteq/tools/output_audio_file.h b/modules/audio_coding/neteq/tools/output_audio_file.h index 25577fc882..8afc37b92a 100644 --- a/modules/audio_coding/neteq/tools/output_audio_file.h +++ b/modules/audio_coding/neteq/tools/output_audio_file.h @@ -13,10 +13,12 @@ #include +#include #include #include "absl/strings/string_view.h" #include "modules/audio_coding/neteq/tools/audio_sink.h" +#include "rtc_base/checks.h" namespace webrtc { namespace test { diff --git a/modules/audio_coding/neteq/tools/output_wav_file.h b/modules/audio_coding/neteq/tools/output_wav_file.h index 20eedfb554..5a39aef3b9 100644 --- a/modules/audio_coding/neteq/tools/output_wav_file.h +++ b/modules/audio_coding/neteq/tools/output_wav_file.h @@ -11,7 +11,8 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_WAV_FILE_H_ #define MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_WAV_FILE_H_ -#include +#include +#include #include "absl/strings/string_view.h" #include "common_audio/wav_file.h" diff --git a/modules/audio_coding/neteq/tools/packet.cc b/modules/audio_coding/neteq/tools/packet.cc index 42480e0115..f8deb94f92 100644 --- a/modules/audio_coding/neteq/tools/packet.cc +++ b/modules/audio_coding/neteq/tools/packet.cc @@ -10,7 +10,14 @@ #include "modules/audio_coding/neteq/tools/packet.h" +#include +#include +#include +#include + #include "api/array_view.h" +#include "api/rtp_headers.h" +#include "modules/rtp_rtcp/include/rtp_header_extension_map.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "rtc_base/checks.h" #include "rtc_base/copy_on_write_buffer.h" diff --git a/modules/audio_coding/neteq/tools/packet.h b/modules/audio_coding/neteq/tools/packet.h index a94a0fdc5e..6d13ba6d4e 100644 --- a/modules/audio_coding/neteq/tools/packet.h +++ b/modules/audio_coding/neteq/tools/packet.h @@ -11,6 +11,8 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_ #define MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_ +#include +#include #include #include "api/array_view.h" diff --git a/modules/audio_coding/neteq/tools/packet_source.cc b/modules/audio_coding/neteq/tools/packet_source.cc index 598ae6edd4..dd7674c051 100644 --- a/modules/audio_coding/neteq/tools/packet_source.cc +++ b/modules/audio_coding/neteq/tools/packet_source.cc @@ -10,6 +10,8 @@ #include "modules/audio_coding/neteq/tools/packet_source.h" +#include + namespace webrtc { namespace test { diff --git a/modules/audio_coding/neteq/tools/packet_source.h b/modules/audio_coding/neteq/tools/packet_source.h index be1705cae1..68de0d51ed 100644 --- a/modules/audio_coding/neteq/tools/packet_source.h +++ b/modules/audio_coding/neteq/tools/packet_source.h @@ -12,6 +12,7 @@ #define MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_ #include +#include #include #include "modules/audio_coding/neteq/tools/packet.h" diff --git a/modules/audio_coding/neteq/tools/packet_unittest.cc b/modules/audio_coding/neteq/tools/packet_unittest.cc index a935ff6477..df516d885c 100644 --- a/modules/audio_coding/neteq/tools/packet_unittest.cc +++ b/modules/audio_coding/neteq/tools/packet_unittest.cc @@ -12,6 +12,13 @@ #include "modules/audio_coding/neteq/tools/packet.h" +#include +#include +#include +#include + +#include "api/rtp_headers.h" +#include "rtc_base/copy_on_write_buffer.h" #include "test/gtest.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/tools/resample_input_audio_file.cc b/modules/audio_coding/neteq/tools/resample_input_audio_file.cc index 5050e1fb17..49e9015e2f 100644 --- a/modules/audio_coding/neteq/tools/resample_input_audio_file.cc +++ b/modules/audio_coding/neteq/tools/resample_input_audio_file.cc @@ -10,8 +10,11 @@ #include "modules/audio_coding/neteq/tools/resample_input_audio_file.h" +#include +#include #include +#include "modules/audio_coding/neteq/tools/input_audio_file.h" #include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/tools/resample_input_audio_file.h b/modules/audio_coding/neteq/tools/resample_input_audio_file.h index 279fece616..0eae44bd37 100644 --- a/modules/audio_coding/neteq/tools/resample_input_audio_file.h +++ b/modules/audio_coding/neteq/tools/resample_input_audio_file.h @@ -11,7 +11,8 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_ #define MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_ -#include +#include +#include #include "absl/strings/string_view.h" #include "common_audio/resampler/include/resampler.h" diff --git a/modules/audio_coding/neteq/tools/rtp_analyze.cc b/modules/audio_coding/neteq/tools/rtp_analyze.cc index 8e94cbdc05..a4bf3dfe7f 100644 --- a/modules/audio_coding/neteq/tools/rtp_analyze.cc +++ b/modules/audio_coding/neteq/tools/rtp_analyze.cc @@ -10,13 +10,19 @@ #include +#include +#include #include +#include #include #include "absl/flags/flag.h" #include "absl/flags/parse.h" +#include "api/rtp_headers.h" #include "modules/audio_coding/neteq/tools/packet.h" #include "modules/audio_coding/neteq/tools/rtp_file_source.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "rtc_base/checks.h" ABSL_FLAG(int, red, 117, "RTP payload type for RED"); ABSL_FLAG(int, diff --git a/modules/audio_coding/neteq/tools/rtp_file_source.h b/modules/audio_coding/neteq/tools/rtp_file_source.h index 300981a025..47005ca273 100644 --- a/modules/audio_coding/neteq/tools/rtp_file_source.h +++ b/modules/audio_coding/neteq/tools/rtp_file_source.h @@ -13,12 +13,14 @@ #include +#include #include #include -#include #include "absl/strings/string_view.h" +#include "modules/audio_coding/neteq/tools/packet.h" #include "modules/audio_coding/neteq/tools/packet_source.h" +#include "modules/rtp_rtcp/include/rtp_header_extension_map.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/tools/rtp_generator.cc b/modules/audio_coding/neteq/tools/rtp_generator.cc index 5633f11b86..56c799f125 100644 --- a/modules/audio_coding/neteq/tools/rtp_generator.cc +++ b/modules/audio_coding/neteq/tools/rtp_generator.cc @@ -10,6 +10,12 @@ #include "modules/audio_coding/neteq/tools/rtp_generator.h" +#include +#include + +#include "api/rtp_headers.h" +#include "rtc_base/checks.h" + namespace webrtc { namespace test { diff --git a/modules/audio_coding/neteq/tools/rtp_generator.h b/modules/audio_coding/neteq/tools/rtp_generator.h index 2e615adec5..526a995672 100644 --- a/modules/audio_coding/neteq/tools/rtp_generator.h +++ b/modules/audio_coding/neteq/tools/rtp_generator.h @@ -11,6 +11,9 @@ #ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_ #define MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_ +#include +#include + #include "api/rtp_headers.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/tools/rtp_jitter.cc b/modules/audio_coding/neteq/tools/rtp_jitter.cc index 23fc35f657..419a99a4b7 100644 --- a/modules/audio_coding/neteq/tools/rtp_jitter.cc +++ b/modules/audio_coding/neteq/tools/rtp_jitter.cc @@ -11,13 +11,17 @@ #include #include +#include #include #include +#include +#include #include #include "api/array_view.h" #include "modules/rtp_rtcp/source/byte_io.h" #include "rtc_base/buffer.h" +#include "rtc_base/checks.h" namespace webrtc { namespace test { diff --git a/modules/audio_coding/neteq/tools/rtpcat.cc b/modules/audio_coding/neteq/tools/rtpcat.cc index 431de553ae..1c126ff720 100644 --- a/modules/audio_coding/neteq/tools/rtpcat.cc +++ b/modules/audio_coding/neteq/tools/rtpcat.cc @@ -10,6 +10,7 @@ #include +#include #include #include "rtc_base/checks.h" diff --git a/modules/audio_coding/neteq/underrun_optimizer.cc b/modules/audio_coding/neteq/underrun_optimizer.cc index d1f12e9bfd..dcee8e19fe 100644 --- a/modules/audio_coding/neteq/underrun_optimizer.cc +++ b/modules/audio_coding/neteq/underrun_optimizer.cc @@ -11,6 +11,9 @@ #include "modules/audio_coding/neteq/underrun_optimizer.h" #include +#include + +#include "api/neteq/tick_timer.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/underrun_optimizer_unittest.cc b/modules/audio_coding/neteq/underrun_optimizer_unittest.cc index 0988e03d73..e782cdb180 100644 --- a/modules/audio_coding/neteq/underrun_optimizer_unittest.cc +++ b/modules/audio_coding/neteq/underrun_optimizer_unittest.cc @@ -10,6 +10,9 @@ #include "modules/audio_coding/neteq/underrun_optimizer.h" +#include + +#include "api/neteq/tick_timer.h" #include "test/gtest.h" namespace webrtc { diff --git a/modules/audio_coding/test/Channel.cc b/modules/audio_coding/test/Channel.cc index 4c91f5948a..b3be9101b1 100644 --- a/modules/audio_coding/test/Channel.cc +++ b/modules/audio_coding/test/Channel.cc @@ -10,8 +10,17 @@ #include "modules/audio_coding/test/Channel.h" -#include +#include +#include +#include +#include +#include "api/array_view.h" +#include "api/neteq/neteq.h" +#include "api/rtp_headers.h" +#include "api/units/timestamp.h" +#include "modules/audio_coding/include/audio_coding_module_typedefs.h" +#include "rtc_base/checks.h" #include "rtc_base/strings/string_builder.h" #include "rtc_base/time_utils.h" diff --git a/modules/audio_coding/test/Channel.h b/modules/audio_coding/test/Channel.h index 616e283647..364f4e35ea 100644 --- a/modules/audio_coding/test/Channel.h +++ b/modules/audio_coding/test/Channel.h @@ -13,9 +13,11 @@ #include +#include + #include "api/neteq/neteq.h" #include "modules/audio_coding/include/audio_coding_module.h" -#include "modules/include/module_common_types.h" +#include "modules/audio_coding/include/audio_coding_module_typedefs.h" #include "rtc_base/synchronization/mutex.h" namespace webrtc { diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc index f50fb15e48..f1a650566d 100644 --- a/modules/audio_coding/test/EncodeDecodeTest.cc +++ b/modules/audio_coding/test/EncodeDecodeTest.cc @@ -13,15 +13,24 @@ #include #include +#include +#include #include +#include #include "absl/strings/string_view.h" +#include "api/array_view.h" +#include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" #include "api/neteq/default_neteq_factory.h" +#include "api/neteq/neteq.h" +#include "api/units/timestamp.h" #include "modules/audio_coding/include/audio_coding_module.h" +#include "modules/audio_coding/include/audio_coding_module_typedefs.h" +#include "modules/audio_coding/test/RTPFile.h" #include "rtc_base/strings/string_builder.h" #include "test/gtest.h" #include "test/testsupport/file_utils.h" diff --git a/modules/audio_coding/test/EncodeDecodeTest.h b/modules/audio_coding/test/EncodeDecodeTest.h index 8a20998cd7..f3c4580f10 100644 --- a/modules/audio_coding/test/EncodeDecodeTest.h +++ b/modules/audio_coding/test/EncodeDecodeTest.h @@ -14,14 +14,17 @@ #include #include +#include + #include "absl/strings/string_view.h" +#include "api/audio_codecs/audio_format.h" #include "api/environment/environment.h" #include "api/neteq/neteq.h" #include "modules/audio_coding/acm2/acm_resampler.h" #include "modules/audio_coding/include/audio_coding_module.h" +#include "modules/audio_coding/include/audio_coding_module_typedefs.h" #include "modules/audio_coding/test/PCMFile.h" #include "modules/audio_coding/test/RTPFile.h" -#include "modules/include/module_common_types.h" namespace webrtc { diff --git a/modules/audio_coding/test/PCMFile.cc b/modules/audio_coding/test/PCMFile.cc index e069a42de1..513703e7e5 100644 --- a/modules/audio_coding/test/PCMFile.cc +++ b/modules/audio_coding/test/PCMFile.cc @@ -14,7 +14,12 @@ #include #include +#include +#include +#include + #include "absl/strings/string_view.h" +#include "api/audio/audio_frame.h" #include "rtc_base/checks.h" #include "test/gtest.h" diff --git a/modules/audio_coding/test/PCMFile.h b/modules/audio_coding/test/PCMFile.h index 64af5df007..f0212f3b0a 100644 --- a/modules/audio_coding/test/PCMFile.h +++ b/modules/audio_coding/test/PCMFile.h @@ -14,6 +14,7 @@ #include #include +#include #include #include diff --git a/modules/audio_coding/test/PacketLossTest.cc b/modules/audio_coding/test/PacketLossTest.cc index 41737521b7..dc8207e00d 100644 --- a/modules/audio_coding/test/PacketLossTest.cc +++ b/modules/audio_coding/test/PacketLossTest.cc @@ -10,14 +10,23 @@ #include "modules/audio_coding/test/PacketLossTest.h" +#include #include +#include #include "absl/strings/string_view.h" +#include "api/array_view.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" #include "api/neteq/default_neteq_factory.h" +#include "api/neteq/neteq.h" #include "api/units/timestamp.h" +#include "modules/audio_coding/include/audio_coding_module.h" +#include "modules/audio_coding/test/EncodeDecodeTest.h" +#include "modules/audio_coding/test/RTPFile.h" #include "rtc_base/strings/string_builder.h" #include "test/gtest.h" #include "test/testsupport/file_utils.h" diff --git a/modules/audio_coding/test/PacketLossTest.h b/modules/audio_coding/test/PacketLossTest.h index 41a62d519a..8bb80ca5cf 100644 --- a/modules/audio_coding/test/PacketLossTest.h +++ b/modules/audio_coding/test/PacketLossTest.h @@ -14,7 +14,12 @@ #include #include "absl/strings/string_view.h" +#include "api/audio_codecs/audio_format.h" +#include "api/environment/environment.h" +#include "api/neteq/neteq.h" +#include "modules/audio_coding/include/audio_coding_module.h" #include "modules/audio_coding/test/EncodeDecodeTest.h" +#include "modules/audio_coding/test/RTPFile.h" namespace webrtc { diff --git a/modules/audio_coding/test/RTPFile.cc b/modules/audio_coding/test/RTPFile.cc index 0c2ab3c443..24e7f86e35 100644 --- a/modules/audio_coding/test/RTPFile.cc +++ b/modules/audio_coding/test/RTPFile.cc @@ -11,15 +11,22 @@ #include "RTPFile.h" #include +#include +#include +#include +#include #include +#include #include "absl/strings/string_view.h" +#include "api/rtp_headers.h" +#include "rtc_base/ip_address.h" +#include "rtc_base/synchronization/mutex.h" #ifdef WIN32 #include #else -#include #endif // TODO(tlegrand): Consider removing usage of gtest. diff --git a/modules/audio_coding/test/RTPFile.h b/modules/audio_coding/test/RTPFile.h index b796491da9..0598a60e4f 100644 --- a/modules/audio_coding/test/RTPFile.h +++ b/modules/audio_coding/test/RTPFile.h @@ -13,6 +13,7 @@ #include +#include #include #include "absl/strings/string_view.h" diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc index e0ef536b3a..3820098a84 100644 --- a/modules/audio_coding/test/TestAllCodecs.cc +++ b/modules/audio_coding/test/TestAllCodecs.cc @@ -10,21 +10,27 @@ #include "modules/audio_coding/test/TestAllCodecs.h" +#include #include +#include #include #include #include "absl/strings/match.h" +#include "absl/strings/str_cat.h" +#include "api/array_view.h" +#include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" #include "api/environment/environment_factory.h" #include "api/neteq/default_neteq_factory.h" #include "api/neteq/neteq.h" +#include "api/rtp_headers.h" +#include "api/units/timestamp.h" #include "modules/audio_coding/acm2/acm_resampler.h" +#include "modules/audio_coding/include/audio_coding_module.h" #include "modules/audio_coding/include/audio_coding_module_typedefs.h" -#include "modules/include/module_common_types.h" -#include "rtc_base/logging.h" -#include "rtc_base/string_encode.h" +#include "rtc_base/checks.h" #include "rtc_base/strings/string_builder.h" #include "test/gtest.h" #include "test/testsupport/file_utils.h" diff --git a/modules/audio_coding/test/TestAllCodecs.h b/modules/audio_coding/test/TestAllCodecs.h index 0712332a5e..25302ee81f 100644 --- a/modules/audio_coding/test/TestAllCodecs.h +++ b/modules/audio_coding/test/TestAllCodecs.h @@ -11,10 +11,13 @@ #ifndef MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_ #define MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_ +#include +#include #include #include "api/environment/environment.h" #include "modules/audio_coding/include/audio_coding_module.h" +#include "modules/audio_coding/include/audio_coding_module_typedefs.h" #include "modules/audio_coding/test/PCMFile.h" namespace webrtc { diff --git a/modules/audio_coding/test/TestRedFec.cc b/modules/audio_coding/test/TestRedFec.cc index 1900c905fc..404e52c10d 100644 --- a/modules/audio_coding/test/TestRedFec.cc +++ b/modules/audio_coding/test/TestRedFec.cc @@ -10,14 +10,22 @@ #include "modules/audio_coding/test/TestRedFec.h" +#include +#include +#include #include +#include +#include #include #include "absl/strings/match.h" +#include "api/audio/audio_frame.h" #include "api/audio_codecs/L16/audio_decoder_L16.h" #include "api/audio_codecs/L16/audio_encoder_L16.h" #include "api/audio_codecs/audio_decoder_factory_template.h" +#include "api/audio_codecs/audio_encoder.h" #include "api/audio_codecs/audio_encoder_factory_template.h" +#include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/g711/audio_decoder_g711.h" #include "api/audio_codecs/g711/audio_encoder_g711.h" #include "api/audio_codecs/g722/audio_decoder_g722.h" @@ -26,9 +34,12 @@ #include "api/audio_codecs/opus/audio_encoder_opus.h" #include "api/environment/environment_factory.h" #include "api/neteq/default_neteq_factory.h" +#include "api/neteq/neteq.h" +#include "common_audio/vad/include/vad.h" #include "modules/audio_coding/codecs/cng/audio_encoder_cng.h" #include "modules/audio_coding/codecs/red/audio_encoder_copy_red.h" -#include "modules/audio_coding/include/audio_coding_module_typedefs.h" +#include "modules/audio_coding/include/audio_coding_module.h" +#include "modules/audio_coding/test/Channel.h" #include "rtc_base/strings/string_builder.h" #include "test/gtest.h" #include "test/testsupport/file_utils.h" diff --git a/modules/audio_coding/test/TestRedFec.h b/modules/audio_coding/test/TestRedFec.h index 3016b3889a..112de845aa 100644 --- a/modules/audio_coding/test/TestRedFec.h +++ b/modules/audio_coding/test/TestRedFec.h @@ -11,15 +11,19 @@ #ifndef MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_ #define MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_ +#include #include -#include +#include #include "api/audio_codecs/audio_decoder_factory.h" #include "api/audio_codecs/audio_encoder_factory.h" +#include "api/audio_codecs/audio_format.h" #include "api/environment/environment.h" #include "api/neteq/neteq.h" +#include "api/scoped_refptr.h" #include "common_audio/vad/include/vad.h" #include "modules/audio_coding/acm2/acm_resampler.h" +#include "modules/audio_coding/include/audio_coding_module.h" #include "modules/audio_coding/test/Channel.h" #include "modules/audio_coding/test/PCMFile.h" #include "test/scoped_key_value_config.h" diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc index 8a9a973d21..f7206c3eaf 100644 --- a/modules/audio_coding/test/TestStereo.cc +++ b/modules/audio_coding/test/TestStereo.cc @@ -10,15 +10,30 @@ #include "modules/audio_coding/test/TestStereo.h" +#include +#include +#include +#include #include +#include #include "absl/strings/match.h" +#include "absl/strings/str_cat.h" +#include "api/array_view.h" +#include "api/audio/audio_frame.h" +#include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" #include "api/environment/environment_factory.h" #include "api/neteq/default_neteq_factory.h" +#include "api/neteq/neteq.h" +#include "api/rtp_headers.h" +#include "api/rtp_parameters.h" +#include "api/units/timestamp.h" +#include "modules/audio_coding/include/audio_coding_module.h" #include "modules/audio_coding/include/audio_coding_module_typedefs.h" -#include "modules/include/module_common_types.h" +#include "modules/audio_coding/test/PCMFile.h" +#include "rtc_base/checks.h" #include "rtc_base/strings/string_builder.h" #include "test/gtest.h" #include "test/testsupport/file_utils.h" diff --git a/modules/audio_coding/test/TestStereo.h b/modules/audio_coding/test/TestStereo.h index 7f315695d0..e7a10734d8 100644 --- a/modules/audio_coding/test/TestStereo.h +++ b/modules/audio_coding/test/TestStereo.h @@ -13,12 +13,15 @@ #include +#include +#include #include #include "api/environment/environment.h" #include "api/neteq/neteq.h" #include "modules/audio_coding/acm2/acm_resampler.h" #include "modules/audio_coding/include/audio_coding_module.h" +#include "modules/audio_coding/include/audio_coding_module_typedefs.h" #include "modules/audio_coding/test/PCMFile.h" #define PCMA_AND_PCMU diff --git a/modules/audio_coding/test/TestVADDTX.cc b/modules/audio_coding/test/TestVADDTX.cc index d03f1aed49..38df0d12f3 100644 --- a/modules/audio_coding/test/TestVADDTX.cc +++ b/modules/audio_coding/test/TestVADDTX.cc @@ -10,17 +10,33 @@ #include "modules/audio_coding/test/TestVADDTX.h" +#include +#include +#include +#include +#include +#include +#include #include +#include #include "absl/strings/match.h" #include "absl/strings/string_view.h" +#include "api/audio/audio_frame.h" #include "api/audio_codecs/audio_decoder_factory_template.h" +#include "api/audio_codecs/audio_encoder.h" #include "api/audio_codecs/audio_encoder_factory_template.h" +#include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/opus/audio_decoder_opus.h" #include "api/audio_codecs/opus/audio_encoder_opus.h" #include "api/environment/environment_factory.h" #include "api/neteq/default_neteq_factory.h" +#include "api/neteq/neteq.h" +#include "common_audio/vad/include/vad.h" #include "modules/audio_coding/codecs/cng/audio_encoder_cng.h" +#include "modules/audio_coding/include/audio_coding_module.h" +#include "modules/audio_coding/include/audio_coding_module_typedefs.h" +#include "modules/audio_coding/test/Channel.h" #include "modules/audio_coding/test/PCMFile.h" #include "rtc_base/strings/string_builder.h" #include "test/gtest.h" diff --git a/modules/audio_coding/test/TestVADDTX.h b/modules/audio_coding/test/TestVADDTX.h index e37ad03fae..3d2c99e6a0 100644 --- a/modules/audio_coding/test/TestVADDTX.h +++ b/modules/audio_coding/test/TestVADDTX.h @@ -11,13 +11,18 @@ #ifndef MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_ #define MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_ +#include +#include #include +#include #include "absl/strings/string_view.h" #include "api/audio_codecs/audio_decoder_factory.h" #include "api/audio_codecs/audio_encoder_factory.h" +#include "api/audio_codecs/audio_format.h" #include "api/environment/environment.h" #include "api/neteq/neteq.h" +#include "api/scoped_refptr.h" #include "common_audio/vad/include/vad.h" #include "modules/audio_coding/acm2/acm_resampler.h" #include "modules/audio_coding/include/audio_coding_module.h" diff --git a/modules/audio_coding/test/Tester.cc b/modules/audio_coding/test/Tester.cc index 9dbc64500a..f3f2c16081 100644 --- a/modules/audio_coding/test/Tester.cc +++ b/modules/audio_coding/test/Tester.cc @@ -10,10 +10,6 @@ #include -#include -#include - -#include "modules/audio_coding/include/audio_coding_module.h" #include "modules/audio_coding/test/EncodeDecodeTest.h" #include "modules/audio_coding/test/PacketLossTest.h" #include "modules/audio_coding/test/TestAllCodecs.h" diff --git a/modules/audio_coding/test/opus_test.cc b/modules/audio_coding/test/opus_test.cc index 0438514e7d..d6323fdaf1 100644 --- a/modules/audio_coding/test/opus_test.cc +++ b/modules/audio_coding/test/opus_test.cc @@ -10,11 +10,18 @@ #include "modules/audio_coding/test/opus_test.h" +#include +#include +#include +#include #include +#include "api/audio/audio_frame.h" +#include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/environment/environment_factory.h" #include "api/neteq/default_neteq_factory.h" +#include "api/neteq/neteq.h" #include "modules/audio_coding/codecs/opus/opus_interface.h" #include "modules/audio_coding/include/audio_coding_module_typedefs.h" #include "modules/audio_coding/test/TestStereo.h" diff --git a/modules/audio_coding/test/opus_test.h b/modules/audio_coding/test/opus_test.h index a2ae758e55..d6f0e7c31a 100644 --- a/modules/audio_coding/test/opus_test.h +++ b/modules/audio_coding/test/opus_test.h @@ -13,11 +13,13 @@ #include +#include +#include #include #include "api/neteq/neteq.h" #include "modules/audio_coding/acm2/acm_resampler.h" -#include "modules/audio_coding/codecs/opus/opus_interface.h" +#include "modules/audio_coding/codecs/opus/opus_inst.h" #include "modules/audio_coding/test/PCMFile.h" #include "modules/audio_coding/test/TestStereo.h" diff --git a/modules/audio_coding/test/target_delay_unittest.cc b/modules/audio_coding/test/target_delay_unittest.cc index 8da125700a..0b26e888c9 100644 --- a/modules/audio_coding/test/target_delay_unittest.cc +++ b/modules/audio_coding/test/target_delay_unittest.cc @@ -8,12 +8,19 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include +#include +#include +#include #include +#include "api/array_view.h" #include "api/audio/audio_frame.h" +#include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/environment/environment_factory.h" #include "api/neteq/default_neteq_factory.h" +#include "api/neteq/neteq.h" #include "api/rtp_headers.h" #include "api/units/timestamp.h" #include "modules/audio_coding/codecs/pcm16b/pcm16b.h" From e12bd20849d40501e17c33b74adce2aaf6c2a755 Mon Sep 17 00:00:00 2001 From: Harald Alvestrand Date: Thu, 8 May 2025 11:03:33 +0000 Subject: [PATCH 075/292] Use only PayloadTypeSuggester for PT assigment This merges two relands: Reland "Stop using the UsedPayloadTypes class" This reverts commit ebb67395c1a7c81ec592f45d2abfdaf7f8f3e722. Reland "Use only one kind of codec list merging in CodecVendor." This reverts commit 4aa2908fee6a0d286449172565604a04ebfdea7c. The UsedPayloadTypes class has been removed. Bug: webrtc:360058654 Change-Id: I72b8e4f336e37e9a276c6a449482c3b7f46fdc65 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/385223 Reviewed-by: Evan Shrubsole Commit-Queue: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44547} --- pc/codec_vendor.cc | 211 +++++++++++++-------------------------------- pc/codec_vendor.h | 3 - pc/used_ids.h | 38 -------- 3 files changed, 59 insertions(+), 193 deletions(-) diff --git a/pc/codec_vendor.cc b/pc/codec_vendor.cc index 42d52f6487..e45516c767 100644 --- a/pc/codec_vendor.cc +++ b/pc/codec_vendor.cc @@ -39,7 +39,6 @@ #include "pc/rtp_media_utils.h" #include "pc/session_description.h" #include "pc/typed_codec_vendor.h" -#include "pc/used_ids.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/string_encode.h" @@ -54,11 +53,6 @@ namespace webrtc { namespace { -using webrtc::PayloadTypeSuggester; -using webrtc::RTCError; -using webrtc::RTCErrorOr; -using webrtc::RtpTransceiverDirection; - bool IsRtxCodec(const RtpCodecCapability& capability) { return absl::EqualsIgnoreCase(capability.name, kRtxCodecName); } @@ -75,8 +69,7 @@ bool IsComfortNoiseCodec(const Codec& codec) { std::optional FindMatchingCodec(const CodecList& codecs1, const CodecList& codecs2, const Codec& codec_to_match) { - return webrtc::FindMatchingCodec(codecs1.codecs(), codecs2.codecs(), - codec_to_match); + return FindMatchingCodec(codecs1.codecs(), codecs2.codecs(), codec_to_match); } void StripCNCodecs(CodecList& audio_codecs) { @@ -105,7 +98,7 @@ const Codec* GetAssociatedCodecForRtx(const CodecList& codec_list, } int associated_pt; - if (!webrtc::FromString(associated_pt_str, &associated_pt)) { + if (!FromString(associated_pt_str, &associated_pt)) { RTC_LOG(LS_WARNING) << "Couldn't convert payload type " << associated_pt_str << " of RTX codec " << rtx_codec.id << " to an integer."; @@ -136,14 +129,14 @@ const Codec* GetAssociatedCodecForRed(const CodecList& codec_list, return nullptr; } - std::vector redundant_payloads = webrtc::split(fmtp, '/'); + std::vector redundant_payloads = split(fmtp, '/'); if (redundant_payloads.size() < 2) { return nullptr; } absl::string_view associated_pt_str = redundant_payloads[0]; int associated_pt; - if (!webrtc::FromString(associated_pt_str, &associated_pt)) { + if (!FromString(associated_pt_str, &associated_pt)) { RTC_LOG(LS_WARNING) << "Couldn't convert first payload type " << associated_pt_str << " of RED codec " << red_codec.id << " to an integer."; @@ -182,7 +175,21 @@ RTCError MergeCodecs(const CodecList& reference_codecs, return suggestion.MoveError(); } codec.id = suggestion.value(); - offered_codecs.push_back(codec); + // The rewrite of the parameter may have turned the codec into + // one that is already present. + bool skip = false; + for (const Codec& present_codec : offered_codecs) { + if (present_codec.id == codec.id) { + RTC_DCHECK(MatchesWithReferenceAttributes(present_codec, codec)) + << "Codec mismatch: present " << present_codec << ", new " + << codec; + skip = true; + break; + } + } + if (!skip) { + offered_codecs.push_back(codec); + } } } @@ -232,7 +239,7 @@ RTCError MergeCodecs(const CodecList& reference_codecs, } std::string red_param = absl::StrCat(matching_codec->id); red_codec.params[kCodecParamNotInNameValueFormat] = - webrtc::StrJoin(std::vector{red_param, red_param}, "/"); + StrJoin(std::vector{red_param, red_param}, "/"); } RTCErrorOr suggestion = pt_suggester.SuggestPayloadType(mid, red_codec); @@ -240,82 +247,23 @@ RTCError MergeCodecs(const CodecList& reference_codecs, return suggestion.MoveError(); } red_codec.id = suggestion.value(); - offered_codecs.push_back(red_codec); - } - } - offered_codecs.CheckConsistency(); - return RTCError::OK(); -} - -// Adds all codecs from `reference_codecs` to `offered_codecs` that don't -// already exist in `offered_codecs` and ensure the payload types don't -// collide. -// OLD VERSION - uses UsedPayloadTypes -void MergeCodecs(const CodecList& reference_codecs, - CodecList& offered_codecs, - UsedPayloadTypes* used_pltypes) { - // Add all new codecs that are not RTX/RED codecs. - // The two-pass splitting of the loops means preferring payload types - // of actual codecs with respect to collisions. - for (const Codec& reference_codec : reference_codecs) { - if (reference_codec.GetResiliencyType() != Codec::ResiliencyType::kRtx && - reference_codec.GetResiliencyType() != Codec::ResiliencyType::kRed && - !FindMatchingCodec(reference_codecs, offered_codecs, reference_codec)) { - Codec codec = reference_codec; - used_pltypes->FindAndSetIdUsed(&codec); - offered_codecs.push_back(codec); - } - } - - // Add all new RTX or RED codecs. - for (const Codec& reference_codec : reference_codecs) { - if (reference_codec.GetResiliencyType() == Codec::ResiliencyType::kRtx && - !FindMatchingCodec(reference_codecs, offered_codecs, reference_codec)) { - Codec rtx_codec = reference_codec; - const Codec* associated_codec = - GetAssociatedCodecForRtx(reference_codecs, rtx_codec); - if (!associated_codec) { - continue; - } - // Find a codec in the offered list that matches the reference codec. - // Its payload type may be different than the reference codec. - std::optional matching_codec = FindMatchingCodec( - reference_codecs, offered_codecs, *associated_codec); - if (!matching_codec) { - RTC_LOG(LS_WARNING) - << "Couldn't find matching " << associated_codec->name << " codec."; - continue; - } - - rtx_codec.params[kCodecParamAssociatedPayloadType] = - absl::StrCat(matching_codec->id); - used_pltypes->FindAndSetIdUsed(&rtx_codec); - offered_codecs.push_back(rtx_codec); - } else if (reference_codec.GetResiliencyType() == - Codec::ResiliencyType::kRed && - !FindMatchingCodec(reference_codecs, offered_codecs, - reference_codec)) { - Codec red_codec = reference_codec; - const Codec* associated_codec = - GetAssociatedCodecForRed(reference_codecs, red_codec); - if (associated_codec) { - std::optional matching_codec = FindMatchingCodec( - reference_codecs, offered_codecs, *associated_codec); - if (!matching_codec) { - RTC_LOG(LS_WARNING) << "Couldn't find matching " - << associated_codec->name << " codec."; - continue; + // The rewrite of the parameter may have turned the RED codec into + // one that is already present. + bool skip = false; + for (const Codec& present_codec : offered_codecs) { + if (present_codec.id == red_codec.id) { + RTC_DCHECK(MatchesWithReferenceAttributes(present_codec, red_codec)); + skip = true; + break; } - - red_codec.params[kCodecParamNotInNameValueFormat] = - absl::StrCat(matching_codec->id) + "/" + - absl::StrCat(matching_codec->id); } - used_pltypes->FindAndSetIdUsed(&red_codec); - offered_codecs.push_back(red_codec); + if (!skip) { + offered_codecs.push_back(red_codec); + } } } offered_codecs.CheckConsistency(); + return RTCError::OK(); } // `codecs` is a full list of codecs with correct payload type mappings, which @@ -382,7 +330,7 @@ CodecList MatchCodecPreference( codec.params.find(kCodecParamNotInNameValueFormat); if (fmtp != codec.params.end()) { std::vector redundant_payloads = - webrtc::split(fmtp->second, '/'); + split(fmtp->second, '/'); if (!redundant_payloads.empty() && redundant_payloads[0] == id) { if (!red_was_added) { @@ -443,7 +391,7 @@ void NegotiateVideoCodecLevelsForOffer( for (const Codec& supported_codec : supported_codecs) { if (absl::EqualsIgnoreCase(supported_codec.name, kH265CodecName)) { std::optional supported_ptl = - webrtc::ParseSdpForH265ProfileTierLevel(supported_codec.params); + ParseSdpForH265ProfileTierLevel(supported_codec.params); if (supported_ptl.has_value()) { supported_h265_profiles[supported_ptl->profile] = supported_ptl->level; @@ -458,14 +406,14 @@ void NegotiateVideoCodecLevelsForOffer( for (auto& filtered_codec : filtered_codecs) { if (absl::EqualsIgnoreCase(filtered_codec.name, kH265CodecName)) { std::optional filtered_ptl = - webrtc::ParseSdpForH265ProfileTierLevel(filtered_codec.params); + ParseSdpForH265ProfileTierLevel(filtered_codec.params); if (filtered_ptl.has_value()) { auto it = supported_h265_profiles.find(filtered_ptl->profile); if (it != supported_h265_profiles.end() && filtered_ptl->level != it->second) { filtered_codec.params[kH265FmtpLevelId] = - webrtc::H265LevelToString(it->second); + H265LevelToString(it->second); } } } @@ -506,13 +454,13 @@ RTCError NegotiateCodecs(const CodecList& local_codecs, } } if (absl::EqualsIgnoreCase(ours.name, kH264CodecName)) { - webrtc::H264GenerateProfileLevelIdForAnswer(ours.params, theirs->params, - &negotiated.params); + H264GenerateProfileLevelIdForAnswer(ours.params, theirs->params, + &negotiated.params); } #ifdef RTC_ENABLE_H265 if (absl::EqualsIgnoreCase(ours.name, kH265CodecName)) { - webrtc::H265GenerateProfileTierLevelForAnswer( - ours.params, theirs->params, &negotiated.params); + H265GenerateProfileTierLevelForAnswer(ours.params, theirs->params, + &negotiated.params); NegotiateTxMode(ours, *theirs, &negotiated); } #endif @@ -534,7 +482,7 @@ RTCError NegotiateCodecs(const CodecList& local_codecs, continue; } int apt_value; - if (!webrtc::FromString(apt_str, &apt_value)) { + if (!FromString(apt_str, &apt_value)) { RTC_LOG(LS_WARNING) << "Unconvertable apt value"; continue; } @@ -634,9 +582,11 @@ RTCErrorOr> CodecVendor::GetNegotiatedCodecsForOffer( } // Add our codecs that are not in the current description. if (media_description_options.type == MediaType::AUDIO) { - MergeCodecs(all_audio_codecs(), mid, codecs, pt_suggester); + MergeCodecs(audio_recv_codecs(), mid, codecs, pt_suggester); + MergeCodecs(audio_send_codecs(), mid, codecs, pt_suggester); } else { - MergeCodecs(all_video_codecs(), mid, codecs, pt_suggester); + MergeCodecs(video_recv_codecs(), mid, codecs, pt_suggester); + MergeCodecs(video_send_codecs(), mid, codecs, pt_suggester); } CodecList filtered_codecs; CodecList supported_codecs = @@ -667,20 +617,11 @@ RTCErrorOr> CodecVendor::GetNegotiatedCodecsForOffer( const MediaContentDescription* mcd = current_content->media_description(); for (const Codec& codec : mcd->codecs()) { - if (webrtc::FindMatchingCodec(mcd->codecs(), codecs.codecs(), - codec)) { + if (FindMatchingCodec(mcd->codecs(), codecs.codecs(), codec)) { filtered_codecs.push_back(codec); } } } - // Note what PTs are already in use. - UsedPayloadTypes - used_pltypes; // Used to avoid pt collisions in filtered_codecs - for (auto& codec : filtered_codecs) { - // Note: This may change PTs. Doing so woud indicate an error, but - // UsedPayloadTypes doesn't offer a means to make the distinction. - used_pltypes.FindAndSetIdUsed(&codec); - } // Add other supported codecs. for (const Codec& codec : supported_codecs) { std::optional found_codec = @@ -707,8 +648,11 @@ RTCErrorOr> CodecVendor::GetNegotiatedCodecsForOffer( changed_referenced_codec->id); } } - // Quick fix for b/395077842: Remap the codec if it collides. - used_pltypes.FindAndSetIdUsed(&(*found_codec)); + auto pt_or_error = pt_suggester.SuggestPayloadType(mid, *found_codec); + if (!pt_or_error.ok()) { + return pt_or_error.MoveError(); + } + found_codec->id = pt_or_error.value(); filtered_codecs.push_back(*found_codec); } } @@ -763,9 +707,11 @@ RTCErrorOr CodecVendor::GetNegotiatedCodecsForAnswer( } // Add all our supported codecs if (media_description_options.type == MediaType::AUDIO) { - MergeCodecs(all_audio_codecs(), mid, codecs, pt_suggester); + MergeCodecs(audio_recv_codecs(), mid, codecs, pt_suggester); + MergeCodecs(audio_send_codecs(), mid, codecs, pt_suggester); } else { - MergeCodecs(all_video_codecs(), mid, codecs, pt_suggester); + MergeCodecs(video_recv_codecs(), mid, codecs, pt_suggester); + MergeCodecs(video_send_codecs(), mid, codecs, pt_suggester); } CodecList filtered_codecs; CodecList negotiated_codecs; @@ -794,8 +740,8 @@ RTCErrorOr CodecVendor::GetNegotiatedCodecsForAnswer( const MediaContentDescription* mcd = current_content->media_description(); for (const Codec& codec : mcd->codecs()) { - if (std::optional found_codec = webrtc::FindMatchingCodec( - mcd->codecs(), codecs.codecs(), codec)) { + if (std::optional found_codec = + FindMatchingCodec(mcd->codecs(), codecs.codecs(), codec)) { filtered_codecs.push_back(*found_codec); } } @@ -913,8 +859,7 @@ CodecList CodecVendor::GetVideoCodecsForAnswer( case RtpTransceiverDirection::kSendRecv: case RtpTransceiverDirection::kStopped: case RtpTransceiverDirection::kInactive: - return GetVideoCodecsForOffer( - webrtc::RtpTransceiverDirectionReversed(offer)); + return GetVideoCodecsForOffer(RtpTransceiverDirectionReversed(offer)); case RtpTransceiverDirection::kSendOnly: return video_send_codecs_.codecs(); case RtpTransceiverDirection::kRecvOnly: @@ -948,8 +893,7 @@ CodecList CodecVendor::GetAudioCodecsForAnswer( case RtpTransceiverDirection::kSendRecv: case RtpTransceiverDirection::kStopped: case RtpTransceiverDirection::kInactive: - return GetAudioCodecsForOffer( - webrtc::RtpTransceiverDirectionReversed(offer)); + return GetAudioCodecsForOffer(RtpTransceiverDirectionReversed(offer)); case RtpTransceiverDirection::kSendOnly: return audio_send_codecs_.codecs(); case RtpTransceiverDirection::kRecvOnly: @@ -958,43 +902,6 @@ CodecList CodecVendor::GetAudioCodecsForAnswer( RTC_CHECK_NOTREACHED(); } -CodecList CodecVendor::all_video_codecs() const { - CodecList all_codecs; - UsedPayloadTypes used_payload_types; - for (const Codec& codec : video_recv_codecs_.codecs()) { - Codec codec_mutable = codec; - used_payload_types.FindAndSetIdUsed(&codec_mutable); - all_codecs.push_back(codec_mutable); - } - - // Use MergeCodecs to merge the second half of our list as it already checks - // and fixes problems with duplicate payload types. - MergeCodecs(video_send_codecs_.codecs(), all_codecs, &used_payload_types); - - return all_codecs; -} - -CodecList CodecVendor::all_audio_codecs() const { - // Compute the audio codecs union. - CodecList codecs; - for (const Codec& send : audio_send_codecs_.codecs()) { - codecs.push_back(send); - if (!FindMatchingCodec(audio_send_codecs_.codecs(), - audio_recv_codecs_.codecs(), send)) { - // It doesn't make sense to have an RTX codec we support sending but not - // receiving. - RTC_DCHECK(send.GetResiliencyType() != Codec::ResiliencyType::kRtx); - } - } - for (const Codec& recv : audio_recv_codecs_.codecs()) { - if (!FindMatchingCodec(audio_recv_codecs_.codecs(), - audio_send_codecs_.codecs(), recv)) { - codecs.push_back(recv); - } - } - return codecs; -} - CodecList CodecVendor::audio_sendrecv_codecs() const { // Use NegotiateCodecs to merge our codec lists, since the operation is // essentially the same. Put send_codecs as the offered_codecs, which is the diff --git a/pc/codec_vendor.h b/pc/codec_vendor.h index 1472225a00..4a3a8d92b6 100644 --- a/pc/codec_vendor.h +++ b/pc/codec_vendor.h @@ -102,9 +102,6 @@ class CodecVendor { const RtpTransceiverDirection& offer, const RtpTransceiverDirection& answer) const; - CodecList all_video_codecs() const; - CodecList all_audio_codecs() const; - TypedCodecVendor audio_send_codecs_; TypedCodecVendor audio_recv_codecs_; diff --git a/pc/used_ids.h b/pc/used_ids.h index fe80531006..f9d825c957 100644 --- a/pc/used_ids.h +++ b/pc/used_ids.h @@ -14,9 +14,7 @@ #include #include "api/rtp_parameters.h" -#include "media/base/codec.h" #include "rtc_base/checks.h" -#include "rtc_base/logging.h" namespace webrtc { template @@ -88,41 +86,6 @@ class UsedIds { std::set id_set_; }; -// Helper class used for finding duplicate RTP payload types among audio, video -// and data codecs. When bundle is used the payload types may not collide. -class UsedPayloadTypes : public UsedIds { - public: - UsedPayloadTypes() - : UsedIds(kFirstDynamicPayloadTypeLowerRange, - kLastDynamicPayloadTypeUpperRange) {} - - // Check if a payload type is valid. The range [64-95] is forbidden - // when rtcp-mux is used. - static bool IsIdValid(Codec codec, bool rtcp_mux) { - if (rtcp_mux && (codec.id > kLastDynamicPayloadTypeLowerRange && - codec.id < kFirstDynamicPayloadTypeUpperRange)) { - return false; - } - return codec.id >= 0 && codec.id <= kLastDynamicPayloadTypeUpperRange; - } - - protected: - bool IsIdUsed(int new_id) override { - // Range marked for RTCP avoidance is "used". - if (new_id > kLastDynamicPayloadTypeLowerRange && - new_id < kFirstDynamicPayloadTypeUpperRange) - return true; - return UsedIds::IsIdUsed(new_id); - } - - private: - static const int kFirstDynamicPayloadTypeLowerRange = 35; - static const int kLastDynamicPayloadTypeLowerRange = 63; - - static const int kFirstDynamicPayloadTypeUpperRange = 96; - static const int kLastDynamicPayloadTypeUpperRange = 127; -}; - // Helper class used for finding duplicate RTP Header extension ids among // audio and video extensions. class UsedRtpHeaderExtensionIds : public UsedIds { @@ -190,7 +153,6 @@ class UsedRtpHeaderExtensionIds : public UsedIds { #ifdef WEBRTC_ALLOW_DEPRECATED_NAMESPACES namespace cricket { using ::webrtc::UsedIds; -using ::webrtc::UsedPayloadTypes; using ::webrtc::UsedRtpHeaderExtensionIds; } // namespace cricket #endif // WEBRTC_ALLOW_DEPRECATED_NAMESPACES From 868f537136a828656ab7e62add8fe2482fedf38e Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Thu, 8 May 2025 05:02:27 -0700 Subject: [PATCH 076/292] Roll chromium_revision 34c4b0e9eb..079e22aa7b (1457124:1457513) Change log: https://chromium.googlesource.com/chromium/src/+log/34c4b0e9eb..079e22aa7b Full diff: https://chromium.googlesource.com/chromium/src/+/34c4b0e9eb..079e22aa7b Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/1fad05b2cd..d978a5cb7a * src/build: https://chromium.googlesource.com/chromium/src/build/+log/56e78d9e6e..d9c3040247 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/8e6ebc5db0..90aa42cfb5 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/5a4686c3d8..3d17983c5b * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/a3660c81d4..98759af37e * src/third_party/androidx/cipd: maUv18JmcvP3nwKYXghGhozFkEce3pFDH6JtbSR0SqQC..a1d-sHWq2y1nFvYwrBfZHKtnwQXeYeyJXaK6H_dKEtQC * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/0f1d0df618..864a235afc * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/52ad7cf544..64c31ffa4d * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/6b4da08cef..116820911e * src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/d7bdad4ef8..e3e030ec6e * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/025e0e69fe..db4ac9ca2d DEPS diff: https://chromium.googlesource.com/chromium/src/+/34c4b0e9eb..079e22aa7b/DEPS No update to Clang. BUG=None Change-Id: If0238e2a6a4c993091c739128d76df7b37a69c0b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390306 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44548} --- DEPS | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/DEPS b/DEPS index 36e1acb387..92c9e9452c 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '34c4b0e9ebd7a0681413aef552217ca88701afc5', + 'chromium_revision': '079e22aa7b3daddaf84220019be50468ae1d5d7e', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@1fad05b2cd652ec7d039b3fd3e7abd19fb89982a', + 'https://chromium.googlesource.com/chromium/src/base@d978a5cb7a04d54ae8ccfc2e15f77035cfe09e84', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@56e78d9e6e7ca1f7320ad2c6467d364061c789d7', + 'https://chromium.googlesource.com/chromium/src/build@d9c3040247e3b793c2a053d50fe2376e7596cd95', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@077a66f30fcf281b066fafb6dfc60818c238efb6', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@8e6ebc5db04cc19908e9a5ea5ca64da2103212e8', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@90aa42cfb54d0b710925645ecff1e4e03876fe2b', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@5a4686c3d8285395cb649995f98488a587c136f6', + 'https://chromium.googlesource.com/chromium/src/testing@3d17983c5bfeafd053c8310e709b3ec9c0b3a332', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@a3660c81d479913c74e510255744ca0d12cd1355', + 'https://chromium.googlesource.com/chromium/src/third_party@98759af37e9e2a993dbb134139ca0b059d27fa18', 'src/buildtools/linux64': { 'packages': [ @@ -329,7 +329,7 @@ deps = { 'src/third_party/libc++abi/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@7ff13dd8c66d5047a3ee1e1ba26ddeb70d3d8740', 'src/third_party/llvm-libc/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@d7bdad4ef86b827a96469b1dfdfcfa1218930e59', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@e3e030ec6ee1674bf2195d0cfd0a4bf5fee16537', 'src/third_party/libunwind/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@81e2cb40a70de2b6978e6d8658891ded9a77f7e3', @@ -480,11 +480,11 @@ deps = { }, 'src/third_party/boringssl/src': - 'https://boringssl.googlesource.com/boringssl.git@0f1d0df6183d6ddf0b4d7a10bf80122c7ec260e6', + 'https://boringssl.googlesource.com/boringssl.git@864a235afcf4d2575b1eab8de96fbf0d84f6cda9', 'src/third_party/breakpad/breakpad': 'https://chromium.googlesource.com/breakpad/breakpad.git@b319c12317824bfd0e92828db20bbfe0ec6e31f8', 'src/third_party/catapult': - 'https://chromium.googlesource.com/catapult.git@52ad7cf544050f01f47c070716b9dae8eb9fab2b', + 'https://chromium.googlesource.com/catapult.git@64c31ffa4d735add4a7e7520a52e0e3160216132', 'src/third_party/ced/src': { 'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5', }, @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@6b4da08cefe089b9283931c6ca9acf08df12859d', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@116820911e0a0775f3e1be92a6e0896d0c0d61d4', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@025e0e69feeef14e5eda7ba71f7b4c380eed4b05', + 'https://chromium.googlesource.com/chromium/src/tools@db4ac9ca2d95f8cc419fdb252e583dae6f8b950e', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'maUv18JmcvP3nwKYXghGhozFkEce3pFDH6JtbSR0SqQC', + 'version': 'a1d-sHWq2y1nFvYwrBfZHKtnwQXeYeyJXaK6H_dKEtQC', }, ], 'condition': 'checkout_android and non_git_source', From d7985c8f29df59e5c995a8cc12bfb211343aa6b1 Mon Sep 17 00:00:00 2001 From: Sergey Silkin Date: Thu, 8 May 2025 15:58:43 +0200 Subject: [PATCH 077/292] Fix typos in adaptation doc Bug: none Change-Id: Id7cecee62604aff6d5d102e0ec3a7bc67d9c16fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390440 Auto-Submit: Sergey Silkin Reviewed-by: Evan Shrubsole Commit-Queue: Evan Shrubsole Cr-Commit-Position: refs/heads/main@{#44549} --- video/g3doc/adaptation.md | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/video/g3doc/adaptation.md b/video/g3doc/adaptation.md index d34394fdc8..859fc9a50b 100644 --- a/video/g3doc/adaptation.md +++ b/video/g3doc/adaptation.md @@ -1,5 +1,5 @@ - + # Video Adaptation @@ -60,8 +60,8 @@ A custom resource can be injected into the call using the ## Adaptation -When a a *resource* signals the it is over or underused, this signal reaches the -`ResourceAdaptationProcessor` who requests an `Adaptation` proposal from the +When a *resource* signals that it is over or underused, this signal reaches the +`ResourceAdaptationProcessor` which requests an `Adaptation` proposal from the [VideoStreamAdapter][VideoStreamAdapter]. This proposal is based on the degradation preference of the video stream. `ResourceAdaptationProcessor` will determine if the `Adaptation` should be applied based on the current adaptation From 035b4908ea6bfcae363424c40ea3b0f4e2b42cf5 Mon Sep 17 00:00:00 2001 From: Harald Alvestrand Date: Thu, 8 May 2025 13:12:47 +0000 Subject: [PATCH 078/292] Take out some superfluous webrtc:: prefixes Bug: webrtc:42232595 Change-Id: Ib91efb4d7cc952dc5863ac5ebbb485f14920ee6a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390420 Reviewed-by: Evan Shrubsole Auto-Submit: Harald Alvestrand Commit-Queue: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44550} --- pc/used_ids_unittest.cc | 44 +- pc/video_rtp_receiver.h | 5 +- pc/video_track_unittest.cc | 27 +- pc/webrtc_sdp.cc | 156 ++--- pc/webrtc_sdp.h | 6 +- pc/webrtc_sdp_unittest.cc | 821 ++++++++++------------- pc/webrtc_session_description_factory.cc | 6 +- pc/webrtc_session_description_factory.h | 4 +- 8 files changed, 472 insertions(+), 597 deletions(-) diff --git a/pc/used_ids_unittest.cc b/pc/used_ids_unittest.cc index fd73501ab2..4b2b3c20ab 100644 --- a/pc/used_ids_unittest.cc +++ b/pc/used_ids_unittest.cc @@ -15,15 +15,15 @@ #include "rtc_base/checks.h" #include "test/gtest.h" -using ::webrtc::UsedIds; -using ::webrtc::UsedRtpHeaderExtensionIds; +namespace webrtc { +namespace { struct Foo { int id; }; TEST(UsedIdsTest, UniqueIdsAreUnchanged) { - webrtc::UsedIds used_ids(1, 5); + UsedIds used_ids(1, 5); for (int i = 1; i <= 5; ++i) { Foo id = {i}; used_ids.FindAndSetIdUsed(&id); @@ -32,7 +32,7 @@ TEST(UsedIdsTest, UniqueIdsAreUnchanged) { } TEST(UsedIdsTest, IdsOutsideRangeAreUnchanged) { - webrtc::UsedIds used_ids(1, 5); + UsedIds used_ids(1, 5); Foo id_11 = {11}; Foo id_12 = {12}; @@ -53,7 +53,7 @@ TEST(UsedIdsTest, IdsOutsideRangeAreUnchanged) { } TEST(UsedIdsTest, CollisionsAreReassignedIdsInReverseOrder) { - webrtc::UsedIds used_ids(1, 10); + UsedIds used_ids(1, 10); Foo id_1 = {1}; Foo id_2 = {2}; Foo id_2_collision = {2}; @@ -92,7 +92,7 @@ TEST_P(UsedRtpHeaderExtensionIdsTest, UniqueIdsAreUnchanged) { // Fill all IDs. for (int j = 1; j <= GetParam().max_id; ++j) { - webrtc::RtpExtension extension("", j); + RtpExtension extension("", j); used_ids.FindAndSetIdUsed(&extension); EXPECT_EQ(extension.id, j); } @@ -100,11 +100,11 @@ TEST_P(UsedRtpHeaderExtensionIdsTest, UniqueIdsAreUnchanged) { TEST_P(UsedRtpHeaderExtensionIdsTest, PrioritizeReassignmentToOneByteIds) { UsedRtpHeaderExtensionIds used_ids(GetParam().id_domain); - webrtc::RtpExtension id_1("", 1); - webrtc::RtpExtension id_2("", 2); - webrtc::RtpExtension id_2_collision("", 2); - webrtc::RtpExtension id_3("", 3); - webrtc::RtpExtension id_3_collision("", 3); + RtpExtension id_1("", 1); + RtpExtension id_2("", 2); + RtpExtension id_2_collision("", 2); + RtpExtension id_3("", 3); + RtpExtension id_3_collision("", 3); // Expect that colliding IDs are reassigned to one-byte IDs. used_ids.FindAndSetIdUsed(&id_1); @@ -121,16 +121,15 @@ TEST_F(UsedRtpHeaderExtensionIdsTest, TwoByteIdsAllowed) { UsedRtpHeaderExtensionIds::IdDomain::kTwoByteAllowed); // Fill all one byte IDs. - for (int i = 1; i <= webrtc::RtpExtension::kOneByteHeaderExtensionMaxId; - ++i) { - webrtc::RtpExtension id("", i); + for (int i = 1; i <= RtpExtension::kOneByteHeaderExtensionMaxId; ++i) { + RtpExtension id("", i); used_ids.FindAndSetIdUsed(&id); } // Add new extensions with colliding IDs. - webrtc::RtpExtension id1_collision("", 1); - webrtc::RtpExtension id2_collision("", 2); - webrtc::RtpExtension id3_collision("", 3); + RtpExtension id1_collision("", 1); + RtpExtension id2_collision("", 2); + RtpExtension id3_collision("", 3); // Expect to reassign to two-byte header extension IDs. used_ids.FindAndSetIdUsed(&id1_collision); @@ -166,16 +165,19 @@ TEST_P(UsedRtpHeaderExtensionIdsDeathTest, DieWhenAllIdsAreOccupied) { // Fill all IDs. for (int j = 1; j <= GetParam().max_id; ++j) { - webrtc::RtpExtension id("", j); + RtpExtension id("", j); used_ids.FindAndSetIdUsed(&id); } - webrtc::RtpExtension id1_collision("", 1); - webrtc::RtpExtension id2_collision("", 2); - webrtc::RtpExtension id3_collision("", GetParam().max_id); + RtpExtension id1_collision("", 1); + RtpExtension id2_collision("", 2); + RtpExtension id3_collision("", GetParam().max_id); EXPECT_DEATH(used_ids.FindAndSetIdUsed(&id1_collision), ""); EXPECT_DEATH(used_ids.FindAndSetIdUsed(&id2_collision), ""); EXPECT_DEATH(used_ids.FindAndSetIdUsed(&id3_collision), ""); } #endif // RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) + +} // namespace +} // namespace webrtc diff --git a/pc/video_rtp_receiver.h b/pc/video_rtp_receiver.h index 5bd2e6a766..04a1aac70c 100644 --- a/pc/video_rtp_receiver.h +++ b/pc/video_rtp_receiver.h @@ -29,7 +29,6 @@ #include "api/transport/rtp/rtp_source.h" #include "api/video/video_frame.h" #include "api/video/video_sink_interface.h" -#include "api/video/video_source_interface.h" #include "media/base/media_channel.h" #include "pc/jitter_buffer_delay.h" #include "pc/media_stream_track_proxy.h" @@ -67,9 +66,7 @@ class VideoRtpReceiver : public RtpReceiverInternal { scoped_refptr dtls_transport() const override; std::vector stream_ids() const override; std::vector> streams() const override; - webrtc::MediaType media_type() const override { - return webrtc::MediaType::VIDEO; - } + MediaType media_type() const override { return MediaType::VIDEO; } std::string id() const override { return id_; } diff --git a/pc/video_track_unittest.cc b/pc/video_track_unittest.cc index c003a51bb5..bdf8f72881 100644 --- a/pc/video_track_unittest.cc +++ b/pc/video_track_unittest.cc @@ -18,34 +18,28 @@ #include "media/base/fake_frame_source.h" #include "pc/test/fake_video_track_renderer.h" #include "pc/test/fake_video_track_source.h" -#include "pc/video_track_source.h" #include "rtc_base/thread.h" #include "rtc_base/time_utils.h" #include "test/gtest.h" -using webrtc::FakeVideoTrackRenderer; -using webrtc::FakeVideoTrackSource; -using webrtc::MediaSourceInterface; -using webrtc::MediaStreamTrackInterface; -using webrtc::VideoTrack; -using webrtc::VideoTrackInterface; -using webrtc::VideoTrackSource; +namespace webrtc { +namespace { class VideoTrackTest : public ::testing::Test { public: - VideoTrackTest() : frame_source_(640, 480, webrtc::kNumMicrosecsPerSec / 30) { + VideoTrackTest() : frame_source_(640, 480, kNumMicrosecsPerSec / 30) { static const char kVideoTrackId[] = "track_id"; - video_track_source_ = webrtc::make_ref_counted( + video_track_source_ = make_ref_counted( /*is_screencast=*/false); video_track_ = VideoTrack::Create(kVideoTrackId, video_track_source_, - webrtc::Thread::Current()); + Thread::Current()); } protected: - webrtc::AutoThread main_thread_; - webrtc::scoped_refptr video_track_source_; - webrtc::scoped_refptr video_track_; - webrtc::FakeFrameSource frame_source_; + AutoThread main_thread_; + scoped_refptr video_track_source_; + scoped_refptr video_track_; + FakeFrameSource frame_source_; }; // VideoTrack::Create will create an API proxy around the source object. @@ -105,3 +99,6 @@ TEST_F(VideoTrackTest, DisableTrackBlackout) { EXPECT_EQ(3, renderer->num_rendered_frames()); EXPECT_FALSE(renderer->black_frame()); } + +} // namespace +} // namespace webrtc diff --git a/pc/webrtc_sdp.cc b/pc/webrtc_sdp.cc index a87a82c6a9..eb6a07d0a8 100644 --- a/pc/webrtc_sdp.cc +++ b/pc/webrtc_sdp.cc @@ -68,34 +68,6 @@ #include "rtc_base/string_encode.h" #include "rtc_base/strings/string_builder.h" -using ::webrtc::AudioContentDescription; -using webrtc::Candidate; -using ::webrtc::Candidates; -using ::webrtc::ContentInfo; -using ::webrtc::ICE_CANDIDATE_COMPONENT_RTCP; -using ::webrtc::ICE_CANDIDATE_COMPONENT_RTP; -using ::webrtc::kApplicationSpecificBandwidth; -using ::webrtc::kCodecParamMaxPTime; -using ::webrtc::kCodecParamMinPTime; -using ::webrtc::kCodecParamPTime; -using ::webrtc::kTransportSpecificBandwidth; -using ::webrtc::MediaContentDescription; -using ::webrtc::MediaProtocolType; -using ::webrtc::RidDescription; -using ::webrtc::RtpHeaderExtensions; -using ::webrtc::SctpDataContentDescription; -using ::webrtc::SimulcastDescription; -using ::webrtc::SimulcastLayer; -using ::webrtc::SimulcastLayerList; -using ::webrtc::SocketAddress; -using ::webrtc::SsrcGroup; -using ::webrtc::StreamParams; -using ::webrtc::StreamParamsVec; -using ::webrtc::TransportDescription; -using ::webrtc::TransportInfo; -using ::webrtc::UnsupportedContentDescription; -using ::webrtc::VideoContentDescription; - // TODO(deadbeef): Switch to using anonymous namespace rather than declaring // everything "static". namespace webrtc { @@ -203,8 +175,8 @@ static const char kCandidatePrflx[] = "prflx"; static const char kCandidateRelay[] = "relay"; static const char kTcpCandidateType[] = "tcptype"; -// webrtc::StringBuilder doesn't have a << overload for chars, while -// webrtc::split and webrtc::tokenize_first both take a char delimiter. To +// StringBuilder doesn't have a << overload for chars, while +// split and tokenize_first both take a char delimiter. To // handle both cases these constants come in pairs of a chars and length-one // strings. static const char kSdpDelimiterEqual[] = "="; @@ -268,22 +240,22 @@ using SsrcGroupVec = std::vector; static void BuildMediaDescription(const ContentInfo* content_info, const TransportInfo* transport_info, - const webrtc::MediaType media_type, + const MediaType media_type, const std::vector& candidates, int msid_signaling, std::string* message); -static void BuildMediaLine(const webrtc::MediaType media_type, +static void BuildMediaLine(const MediaType media_type, const ContentInfo* content_info, const MediaContentDescription* media_desc, std::string* message); static void BuildRtpContentAttributes(const MediaContentDescription* media_desc, - const webrtc::MediaType media_type, + const MediaType media_type, int msid_signaling, std::string* message); static void BuildRtpHeaderExtensions(const RtpHeaderExtensions& extensions, std::string* message); static void BuildRtpmap(const MediaContentDescription* media_desc, - const webrtc::MediaType media_type, + const MediaType media_type, std::string* message); static void BuildCandidate(const std::vector& candidates, bool include_ufrag, @@ -314,7 +286,7 @@ static bool ParseMediaDescription( SdpParseError* error); static bool ParseContent( absl::string_view message, - const webrtc::MediaType media_type, + const MediaType media_type, int mline_index, absl::string_view protocol, const std::vector& payload_types, @@ -337,12 +309,12 @@ static bool ParseSsrcGroupAttribute(absl::string_view line, SsrcGroupVec* ssrc_groups, SdpParseError* error); static bool ParseRtpmapAttribute(absl::string_view line, - const webrtc::MediaType media_type, + const MediaType media_type, const std::vector& payload_types, MediaContentDescription* media_desc, SdpParseError* error); static bool ParseFmtpAttributes(absl::string_view line, - const webrtc::MediaType media_type, + const MediaType media_type, MediaContentDescription* media_desc, SdpParseError* error); static bool ParseFmtpParam(absl::string_view line, @@ -350,11 +322,11 @@ static bool ParseFmtpParam(absl::string_view line, std::string* value, SdpParseError* error); static bool ParsePacketizationAttribute(absl::string_view line, - const webrtc::MediaType media_type, + const MediaType media_type, MediaContentDescription* media_desc, SdpParseError* error); static bool ParseRtcpFbAttribute(absl::string_view line, - const webrtc::MediaType media_type, + const MediaType media_type, MediaContentDescription* media_desc, SdpParseError* error); static bool ParseIceOptions(absl::string_view line, @@ -891,7 +863,7 @@ std::string SdpSerialize(const JsepSessionDescription& jdesc) { } // MediaStream semantics. - // TODO(bugs.webrtc.org/10421): Change to & webrtc::kMsidSignalingSemantic + // TODO(bugs.webrtc.org/10421): Change to & kMsidSignalingSemantic // when we think it's safe to do so, so that we gradually fade out this old // line that was removed from the specification. if (desc->msid_signaling() != kMsidSignalingNotUsed) { @@ -1398,7 +1370,7 @@ void BuildDtlsFingerprintSetup(const TransportInfo* transport_info, } } -void BuildMediaLine(const webrtc::MediaType media_type, +void BuildMediaLine(const MediaType media_type, const ContentInfo* content_info, const MediaContentDescription* media_desc, std::string* message) { @@ -1409,15 +1381,14 @@ void BuildMediaLine(const webrtc::MediaType media_type, // fmt is a list of payload type numbers that MAY be used in the session. std::string type; std::string fmt; - if (media_type == webrtc::MediaType::AUDIO || - media_type == webrtc::MediaType::VIDEO) { - type = media_type == webrtc::MediaType::AUDIO ? kSdpMediaTypeAudio - : kSdpMediaTypeVideo; + if (media_type == MediaType::AUDIO || media_type == MediaType::VIDEO) { + type = media_type == MediaType::AUDIO ? kSdpMediaTypeAudio + : kSdpMediaTypeVideo; for (const Codec& codec : media_desc->codecs()) { fmt.append(" "); fmt.append(absl::StrCat(codec.id)); } - } else if (media_type == webrtc::MediaType::DATA) { + } else if (media_type == MediaType::DATA) { type = kSdpMediaTypeData; const SctpDataContentDescription* sctp_data_desc = media_desc->as_sctp(); if (sctp_data_desc) { @@ -1431,7 +1402,7 @@ void BuildMediaLine(const webrtc::MediaType media_type, } else { RTC_DCHECK_NOTREACHED() << "Data description without SCTP"; } - } else if (media_type == webrtc::MediaType::UNSUPPORTED) { + } else if (media_type == MediaType::UNSUPPORTED) { const UnsupportedContentDescription* unsupported_desc = media_desc->as_unsupported(); type = unsupported_desc->media_type(); @@ -1469,7 +1440,7 @@ void BuildMediaLine(const webrtc::MediaType media_type, void BuildMediaDescription(const ContentInfo* content_info, const TransportInfo* transport_info, - const webrtc::MediaType media_type, + const MediaType media_type, const std::vector& candidates, int msid_signaling, std::string* message) { @@ -1561,7 +1532,7 @@ void BuildMediaDescription(const ContentInfo* content_info, } void BuildRtpContentAttributes(const MediaContentDescription* media_desc, - const webrtc::MediaType media_type, + const MediaType media_type, int msid_signaling, std::string* message) { SimulcastSdpSerializer serializer; @@ -1798,7 +1769,7 @@ bool IsFmtpParam(absl::string_view name) { return name != kCodecParamPTime && name != kCodecParamMaxPTime; } -bool WriteFmtpParameters(const webrtc::CodecParameterMap& parameters, +bool WriteFmtpParameters(const CodecParameterMap& parameters, StringBuilder* os) { bool empty = true; const char* delimiter = ""; // No delimiter before first parameter. @@ -1861,7 +1832,7 @@ bool GetMinValue(const std::vector& values, int* value) { } bool GetParameter(const std::string& name, - const webrtc::CodecParameterMap& params, + const CodecParameterMap& params, int* value) { std::map::const_iterator found = params.find(name); if (found == params.end()) { @@ -1874,12 +1845,12 @@ bool GetParameter(const std::string& name, } void BuildRtpmap(const MediaContentDescription* media_desc, - const webrtc::MediaType media_type, + const MediaType media_type, std::string* message) { RTC_DCHECK(message != NULL); RTC_DCHECK(media_desc != NULL); StringBuilder os; - if (media_type == webrtc::MediaType::VIDEO) { + if (media_type == MediaType::VIDEO) { for (const Codec& codec : media_desc->codecs()) { // RFC 4566 // a=rtpmap: / @@ -1894,7 +1865,7 @@ void BuildRtpmap(const MediaContentDescription* media_desc, AddRtcpFbLines(codec, message); AddFmtpLine(codec, message); } - } else if (media_type == webrtc::MediaType::AUDIO) { + } else if (media_type == MediaType::AUDIO) { std::vector ptimes; std::vector maxptimes; int max_minptime = 0; @@ -2580,7 +2551,7 @@ static void BackfillCodecParameters(std::vector& codecs) { static std::unique_ptr ParseContentDescription( absl::string_view message, - const webrtc::MediaType media_type, + const MediaType media_type, int mline_index, absl::string_view protocol, const std::vector& payload_types, @@ -2592,9 +2563,9 @@ static std::unique_ptr ParseContentDescription( std::vector>* candidates, SdpParseError* error) { std::unique_ptr media_desc; - if (media_type == webrtc::MediaType::AUDIO) { + if (media_type == MediaType::AUDIO) { media_desc = std::make_unique(); - } else if (media_type == webrtc::MediaType::VIDEO) { + } else if (media_type == MediaType::VIDEO) { media_desc = std::make_unique(); } else { RTC_DCHECK_NOTREACHED(); @@ -2714,14 +2685,14 @@ bool ParseMediaDescription( } if (media_type == kSdpMediaTypeVideo) { content = ParseContentDescription( - message, webrtc::MediaType::VIDEO, mline_index, protocol, - payload_types, pos, &content_name, &bundle_only, - §ion_msid_signaling, &transport, candidates, error); + message, MediaType::VIDEO, mline_index, protocol, payload_types, pos, + &content_name, &bundle_only, §ion_msid_signaling, &transport, + candidates, error); } else if (media_type == kSdpMediaTypeAudio) { content = ParseContentDescription( - message, webrtc::MediaType::AUDIO, mline_index, protocol, - payload_types, pos, &content_name, &bundle_only, - §ion_msid_signaling, &transport, candidates, error); + message, MediaType::AUDIO, mline_index, protocol, payload_types, pos, + &content_name, &bundle_only, §ion_msid_signaling, &transport, + candidates, error); } else if (media_type == kSdpMediaTypeData && IsDtlsSctp(protocol)) { // The draft-03 format is: // m=application DTLS/SCTP ... @@ -2739,7 +2710,7 @@ bool ParseMediaDescription( } else if (fields[3] == kDefaultSctpmapProtocol) { data_desc->set_use_sctpmap(false); } - if (!ParseContent(message, webrtc::MediaType::DATA, mline_index, protocol, + if (!ParseContent(message, MediaType::DATA, mline_index, protocol, payload_types, pos, &content_name, &bundle_only, §ion_msid_signaling, data_desc.get(), &transport, candidates, error)) { @@ -2751,11 +2722,10 @@ bool ParseMediaDescription( RTC_LOG(LS_WARNING) << "Unsupported media type: " << *mline; auto unsupported_desc = std::make_unique(media_type); - if (!ParseContent(message, webrtc::MediaType::UNSUPPORTED, mline_index, - protocol, payload_types, pos, &content_name, - &bundle_only, §ion_msid_signaling, - unsupported_desc.get(), &transport, candidates, - error)) { + if (!ParseContent(message, MediaType::UNSUPPORTED, mline_index, protocol, + payload_types, pos, &content_name, &bundle_only, + §ion_msid_signaling, unsupported_desc.get(), + &transport, candidates, error)) { return false; } unsupported_desc->set_protocol(protocol); @@ -2841,7 +2811,7 @@ bool ParseMediaDescription( return true; } -void AddParameters(const webrtc::CodecParameterMap& parameters, Codec* codec) { +void AddParameters(const CodecParameterMap& parameters, Codec* codec) { for (const auto& entry : parameters) { const std::string& key = entry.first; const std::string& value = entry.second; @@ -2863,14 +2833,14 @@ void AddFeedbackParameters(const FeedbackParams& feedback_params, // Gets the current codec setting associated with `payload_type`. If there // is no Codec associated with that payload type it returns an empty codec // with that payload type. -Codec GetCodecWithPayloadType(webrtc::MediaType type, +Codec GetCodecWithPayloadType(MediaType type, const std::vector& codecs, int payload_type) { const Codec* codec = FindCodecById(codecs, payload_type); if (codec) return *codec; // Return empty codec with `payload_type`. - if (type == webrtc::MediaType::AUDIO) { + if (type == MediaType::AUDIO) { return CreateAudioCodec(payload_type, "", 0, 0); } else { return CreateVideoCodec(payload_type, ""); @@ -2901,7 +2871,7 @@ void AddOrReplaceCodec(MediaContentDescription* content_desc, // to `parameters`. void UpdateCodec(MediaContentDescription* content_desc, int payload_type, - const webrtc::CodecParameterMap& parameters) { + const CodecParameterMap& parameters) { // Codec might already have been populated (from rtpmap). Codec new_codec = GetCodecWithPayloadType( content_desc->type(), content_desc->codecs(), payload_type); @@ -2982,7 +2952,7 @@ void AddAudioAttribute(const std::string& name, } bool ParseContent(absl::string_view message, - const webrtc::MediaType media_type, + const MediaType media_type, int mline_index, absl::string_view protocol, const std::vector& payload_types, @@ -2998,7 +2968,7 @@ bool ParseContent(absl::string_view message, RTC_DCHECK(content_name != NULL); RTC_DCHECK(transport != NULL); - if (media_type == webrtc::MediaType::AUDIO) { + if (media_type == MediaType::AUDIO) { MaybeCreateStaticPayloadAudioCodecs(payload_types, media_desc); } @@ -3142,7 +3112,7 @@ bool ParseContent(absl::string_view message, if (!ParseDtlsSetup(*line, &(transport->connection_role), error)) { return false; } - } else if (IsDtlsSctp(protocol) && media_type == webrtc::MediaType::DATA) { + } else if (IsDtlsSctp(protocol) && media_type == MediaType::DATA) { // // SCTP specific attributes // @@ -3344,7 +3314,7 @@ bool ParseContent(absl::string_view message, if (!ssrc_infos.empty()) { CreateTracksFromSsrcInfos(ssrc_infos, stream_ids, track_id, &tracks, *msid_signaling); - } else if (media_type != webrtc::MediaType::DATA && + } else if (media_type != MediaType::DATA && (*msid_signaling & kMsidSignalingMediaSection)) { // If the stream_ids/track_id was signaled but SSRCs were unsignaled we // still create a track. This isn't done for data media types because @@ -3383,7 +3353,7 @@ bool ParseContent(absl::string_view message, [](const Codec codec) { return !codec.name.empty(); })) { return ParseFailed("Failed to parse codecs correctly.", error); } - if (media_type == webrtc::MediaType::AUDIO) { + if (media_type == MediaType::AUDIO) { AddAudioAttribute(kCodecParamMaxPTime, maxptime_as_string, media_desc); AddAudioAttribute(kCodecParamPTime, ptime_as_string, media_desc); } @@ -3542,7 +3512,7 @@ void UpdateCodec(int payload_type, } bool ParseRtpmapAttribute(absl::string_view line, - const webrtc::MediaType media_type, + const MediaType media_type, const std::vector& payload_types, MediaContentDescription* media_desc, SdpParseError* error) { @@ -3586,7 +3556,7 @@ bool ParseRtpmapAttribute(absl::string_view line, return false; } - if (media_type == webrtc::MediaType::VIDEO) { + if (media_type == MediaType::VIDEO) { for (const Codec& existing_codec : media_desc->codecs()) { if (!existing_codec.name.empty() && payload_type == existing_codec.id && (!absl::EqualsIgnoreCase(encoding_name, existing_codec.name) || @@ -3602,7 +3572,7 @@ bool ParseRtpmapAttribute(absl::string_view line, } } UpdateCodec(payload_type, encoding_name, media_desc); - } else if (media_type == webrtc::MediaType::AUDIO) { + } else if (media_type == MediaType::AUDIO) { // RFC 4566 // For audio streams, indicates the number // of audio channels. This parameter is OPTIONAL and may be @@ -3654,7 +3624,7 @@ bool ParseFmtpParam(absl::string_view line, } bool ParseFmtpParameterSet(absl::string_view line_params, - webrtc::CodecParameterMap& codec_params, + CodecParameterMap& codec_params, SdpParseError* error) { // Parse out format specific parameters. for (absl::string_view param : @@ -3675,11 +3645,10 @@ bool ParseFmtpParameterSet(absl::string_view line_params, } bool ParseFmtpAttributes(absl::string_view line, - const webrtc::MediaType media_type, + const MediaType media_type, MediaContentDescription* media_desc, SdpParseError* error) { - if (media_type != webrtc::MediaType::AUDIO && - media_type != webrtc::MediaType::VIDEO) { + if (media_type != MediaType::AUDIO && media_type != MediaType::VIDEO) { return true; } @@ -3709,23 +3678,22 @@ bool ParseFmtpAttributes(absl::string_view line, } // Parse out format specific parameters. - webrtc::CodecParameterMap codec_params; + CodecParameterMap codec_params; if (!ParseFmtpParameterSet(line_params, codec_params, error)) { return false; } - if (media_type == webrtc::MediaType::AUDIO || - media_type == webrtc::MediaType::VIDEO) { + if (media_type == MediaType::AUDIO || media_type == MediaType::VIDEO) { UpdateCodec(media_desc, payload_type, codec_params); } return true; } bool ParsePacketizationAttribute(absl::string_view line, - const webrtc::MediaType media_type, + const MediaType media_type, MediaContentDescription* media_desc, SdpParseError* error) { - if (media_type != webrtc::MediaType::VIDEO) { + if (media_type != MediaType::VIDEO) { return true; } std::vector packetization_fields = @@ -3749,11 +3717,10 @@ bool ParsePacketizationAttribute(absl::string_view line, } bool ParseRtcpFbAttribute(absl::string_view line, - const webrtc::MediaType media_type, + const MediaType media_type, MediaContentDescription* media_desc, SdpParseError* error) { - if (media_type != webrtc::MediaType::AUDIO && - media_type != webrtc::MediaType::VIDEO) { + if (media_type != MediaType::AUDIO && media_type != MediaType::VIDEO) { return true; } std::vector rtcp_fb_fields = @@ -3781,8 +3748,7 @@ bool ParseRtcpFbAttribute(absl::string_view line, } const FeedbackParam feedback_param(id, param); - if (media_type == webrtc::MediaType::AUDIO || - media_type == webrtc::MediaType::VIDEO) { + if (media_type == MediaType::AUDIO || media_type == MediaType::VIDEO) { UpdateCodec(media_desc, payload_type, feedback_param); } return true; diff --git a/pc/webrtc_sdp.h b/pc/webrtc_sdp.h index 884b45414a..f44df2aa46 100644 --- a/pc/webrtc_sdp.h +++ b/pc/webrtc_sdp.h @@ -27,7 +27,7 @@ #include "api/jsep.h" #include "api/jsep_ice_candidate.h" #include "api/jsep_session_description.h" -#include "media/base/codec.h" +#include "api/rtp_parameters.h" #include "rtc_base/strings/string_builder.h" #include "rtc_base/system/rtc_export.h" @@ -100,12 +100,12 @@ RTC_EXPORT bool ParseCandidate(absl::string_view message, // parameters are not considered to be part of the FMTP line, see the function // IsFmtpParam(). Returns true if the set of FMTP parameters is nonempty, false // otherwise. -bool WriteFmtpParameters(const webrtc::CodecParameterMap& parameters, +bool WriteFmtpParameters(const CodecParameterMap& parameters, StringBuilder* os); // Parses a string into an FMTP parameter set, in key-value format. bool ParseFmtpParameterSet(absl::string_view line_params, - webrtc::CodecParameterMap& codec_params, + CodecParameterMap& codec_params, SdpParseError* error); } // namespace webrtc diff --git a/pc/webrtc_sdp_unittest.cc b/pc/webrtc_sdp_unittest.cc index ba696d4b41..2ce4765b66 100644 --- a/pc/webrtc_sdp_unittest.cc +++ b/pc/webrtc_sdp_unittest.cc @@ -56,37 +56,13 @@ #endif #include "pc/webrtc_sdp.h" +namespace webrtc { + +namespace { + using ::testing::ElementsAre; using ::testing::Field; using ::testing::Property; -using ::webrtc::AudioContentDescription; -using webrtc::Candidate; -using ::webrtc::ContentGroup; -using ::webrtc::ContentInfo; -using ::webrtc::ICE_CANDIDATE_COMPONENT_RTCP; -using ::webrtc::ICE_CANDIDATE_COMPONENT_RTP; -using webrtc::IceCandidateCollection; -using webrtc::IceCandidateInterface; -using webrtc::IceCandidateType; -using webrtc::JsepIceCandidate; -using webrtc::JsepSessionDescription; -using ::webrtc::kFecSsrcGroupSemantics; -using ::webrtc::MediaProtocolType; -using ::webrtc::RidDescription; -using ::webrtc::RidDirection; -using webrtc::RtpExtension; -using webrtc::RtpTransceiverDirection; -using ::webrtc::SctpDataContentDescription; -using webrtc::SdpParseError; -using webrtc::SdpType; -using ::webrtc::SessionDescription; -using webrtc::SessionDescriptionInterface; -using ::webrtc::SimulcastDescription; -using ::webrtc::SimulcastLayer; -using ::webrtc::StreamParams; -using ::webrtc::TransportDescription; -using ::webrtc::TransportInfo; -using ::webrtc::VideoContentDescription; static const uint32_t kDefaultSctpPort = 5000; static const uint16_t kUnusualSctpPort = 9556; @@ -860,12 +836,12 @@ static SdpType kDummyType = SdpType::kOffer; static bool SdpDeserialize(const std::string& message, JsepSessionDescription* jdesc) { - return webrtc::SdpDeserialize(message, jdesc, NULL); + return SdpDeserialize(message, jdesc, NULL); } static bool SdpDeserializeCandidate(const std::string& message, JsepIceCandidate* candidate) { - return webrtc::SdpDeserializeCandidate(message, candidate, NULL); + return SdpDeserializeCandidate(message, candidate, NULL); } // Add some extra `newlines` to the `message` after `line`. @@ -887,7 +863,7 @@ static void ExpectParseFailure(const std::string& bad_sdp, const std::string& bad_part) { JsepSessionDescription desc(kDummyType); SdpParseError error; - bool ret = webrtc::SdpDeserialize(bad_sdp, &desc, &error); + bool ret = SdpDeserialize(bad_sdp, &desc, &error); ASSERT_FALSE(ret); EXPECT_NE(std::string::npos, error.line.find(bad_part.c_str())) << "Did not find " << bad_part << " in " << error.line; @@ -951,10 +927,9 @@ static void ReplaceRejected(bool audio_rejected, static TransportDescription MakeTransportDescription(std::string ufrag, std::string pwd) { - webrtc::SSLFingerprint fingerprint(webrtc::DIGEST_SHA_1, kIdentityDigest); + SSLFingerprint fingerprint(DIGEST_SHA_1, kIdentityDigest); return TransportDescription(std::vector(), ufrag, pwd, - webrtc::ICEMODE_FULL, webrtc::CONNECTIONROLE_NONE, - &fingerprint); + ICEMODE_FULL, CONNECTIONROLE_NONE, &fingerprint); } // WebRtcSdpTest @@ -963,7 +938,7 @@ class WebRtcSdpTest : public ::testing::Test { public: WebRtcSdpTest() : jdesc_(kDummyType) { #ifdef WEBRTC_ANDROID - webrtc::InitializeAndroidObjects(); + InitializeAndroidObjects(); #endif // AudioContentDescription audio_desc_ = CreateAudioContentDescription(); @@ -973,7 +948,7 @@ class WebRtcSdpTest : public ::testing::Test { audio_stream.set_stream_ids({kStreamId1}); audio_stream.ssrcs.push_back(kAudioTrack1Ssrc); audio_desc_->AddStream(audio_stream); - webrtc::SocketAddress audio_addr("74.125.127.126", 2345); + SocketAddress audio_addr("74.125.127.126", 2345); audio_desc_->set_connection_address(audio_addr); desc_.AddContent(kAudioContentName, MediaProtocolType::kRtp, absl::WrapUnique(audio_desc_)); @@ -986,17 +961,16 @@ class WebRtcSdpTest : public ::testing::Test { video_stream.set_stream_ids({kStreamId1}); video_stream.ssrcs.push_back(kVideoTrack1Ssrc1); video_stream.ssrcs.push_back(kVideoTrack1Ssrc2); - webrtc::SsrcGroup ssrc_group(webrtc::kFecSsrcGroupSemantics, - video_stream.ssrcs); + SsrcGroup ssrc_group(kFecSsrcGroupSemantics, video_stream.ssrcs); video_stream.ssrc_groups.push_back(ssrc_group); video_desc_->AddStream(video_stream); - webrtc::SocketAddress video_addr("74.125.224.39", 3457); + SocketAddress video_addr("74.125.224.39", 3457); video_desc_->set_connection_address(video_addr); desc_.AddContent(kVideoContentName, MediaProtocolType::kRtp, absl::WrapUnique(video_desc_)); // TransportInfo, with fingerprint - webrtc::SSLFingerprint fingerprint(webrtc::DIGEST_SHA_1, kIdentityDigest); + SSLFingerprint fingerprint(DIGEST_SHA_1, kIdentityDigest); desc_.AddTransportInfo(TransportInfo( kAudioContentName, MakeTransportDescription(kUfragVoice, kPwdVoice))); desc_.AddTransportInfo(TransportInfo( @@ -1004,75 +978,67 @@ class WebRtcSdpTest : public ::testing::Test { // v4 host int port = 1234; - webrtc::SocketAddress address("192.168.1.5", port++); - Candidate candidate1(webrtc::ICE_CANDIDATE_COMPONENT_RTP, "udp", address, + SocketAddress address("192.168.1.5", port++); + Candidate candidate1(ICE_CANDIDATE_COMPONENT_RTP, "udp", address, kCandidatePriority, "", "", IceCandidateType::kHost, kCandidateGeneration, kCandidateFoundation1); address.SetPort(port++); - Candidate candidate2(webrtc::ICE_CANDIDATE_COMPONENT_RTCP, "udp", address, + Candidate candidate2(ICE_CANDIDATE_COMPONENT_RTCP, "udp", address, kCandidatePriority, "", "", IceCandidateType::kHost, kCandidateGeneration, kCandidateFoundation1); address.SetPort(port++); - Candidate candidate3(webrtc::ICE_CANDIDATE_COMPONENT_RTCP, "udp", address, + Candidate candidate3(ICE_CANDIDATE_COMPONENT_RTCP, "udp", address, kCandidatePriority, "", "", IceCandidateType::kHost, kCandidateGeneration, kCandidateFoundation1); address.SetPort(port++); - Candidate candidate4(webrtc::ICE_CANDIDATE_COMPONENT_RTP, "udp", address, + Candidate candidate4(ICE_CANDIDATE_COMPONENT_RTP, "udp", address, kCandidatePriority, "", "", IceCandidateType::kHost, kCandidateGeneration, kCandidateFoundation1); // v6 host - webrtc::SocketAddress v6_address("::1", port++); - webrtc::Candidate candidate5(webrtc::ICE_CANDIDATE_COMPONENT_RTP, "udp", - v6_address, kCandidatePriority, "", "", - IceCandidateType::kHost, kCandidateGeneration, - kCandidateFoundation2); + SocketAddress v6_address("::1", port++); + Candidate candidate5(ICE_CANDIDATE_COMPONENT_RTP, "udp", v6_address, + kCandidatePriority, "", "", IceCandidateType::kHost, + kCandidateGeneration, kCandidateFoundation2); v6_address.SetPort(port++); - webrtc::Candidate candidate6(webrtc::ICE_CANDIDATE_COMPONENT_RTCP, "udp", - v6_address, kCandidatePriority, "", "", - IceCandidateType::kHost, kCandidateGeneration, - kCandidateFoundation2); + Candidate candidate6(ICE_CANDIDATE_COMPONENT_RTCP, "udp", v6_address, + kCandidatePriority, "", "", IceCandidateType::kHost, + kCandidateGeneration, kCandidateFoundation2); v6_address.SetPort(port++); - webrtc::Candidate candidate7(webrtc::ICE_CANDIDATE_COMPONENT_RTCP, "udp", - v6_address, kCandidatePriority, "", "", - IceCandidateType::kHost, kCandidateGeneration, - kCandidateFoundation2); + Candidate candidate7(ICE_CANDIDATE_COMPONENT_RTCP, "udp", v6_address, + kCandidatePriority, "", "", IceCandidateType::kHost, + kCandidateGeneration, kCandidateFoundation2); v6_address.SetPort(port++); - webrtc::Candidate candidate8(webrtc::ICE_CANDIDATE_COMPONENT_RTP, "udp", - v6_address, kCandidatePriority, "", "", - IceCandidateType::kHost, kCandidateGeneration, - kCandidateFoundation2); + Candidate candidate8(ICE_CANDIDATE_COMPONENT_RTP, "udp", v6_address, + kCandidatePriority, "", "", IceCandidateType::kHost, + kCandidateGeneration, kCandidateFoundation2); // stun int port_stun = 2345; - webrtc::SocketAddress address_stun("74.125.127.126", port_stun++); - webrtc::SocketAddress rel_address_stun("192.168.1.5", port_stun++); - webrtc::Candidate candidate9(webrtc::ICE_CANDIDATE_COMPONENT_RTP, "udp", - address_stun, kCandidatePriority, "", "", - IceCandidateType::kSrflx, kCandidateGeneration, - kCandidateFoundation3); + SocketAddress address_stun("74.125.127.126", port_stun++); + SocketAddress rel_address_stun("192.168.1.5", port_stun++); + Candidate candidate9(ICE_CANDIDATE_COMPONENT_RTP, "udp", address_stun, + kCandidatePriority, "", "", IceCandidateType::kSrflx, + kCandidateGeneration, kCandidateFoundation3); candidate9.set_related_address(rel_address_stun); address_stun.SetPort(port_stun++); rel_address_stun.SetPort(port_stun++); - webrtc::Candidate candidate10(webrtc::ICE_CANDIDATE_COMPONENT_RTCP, "udp", - address_stun, kCandidatePriority, "", "", - IceCandidateType::kSrflx, - kCandidateGeneration, kCandidateFoundation3); + Candidate candidate10(ICE_CANDIDATE_COMPONENT_RTCP, "udp", address_stun, + kCandidatePriority, "", "", IceCandidateType::kSrflx, + kCandidateGeneration, kCandidateFoundation3); candidate10.set_related_address(rel_address_stun); // relay int port_relay = 3456; - webrtc::SocketAddress address_relay("74.125.224.39", port_relay++); - webrtc::Candidate candidate11(webrtc::ICE_CANDIDATE_COMPONENT_RTCP, "udp", - address_relay, kCandidatePriority, "", "", - IceCandidateType::kRelay, - kCandidateGeneration, kCandidateFoundation4); + SocketAddress address_relay("74.125.224.39", port_relay++); + Candidate candidate11(ICE_CANDIDATE_COMPONENT_RTCP, "udp", address_relay, + kCandidatePriority, "", "", IceCandidateType::kRelay, + kCandidateGeneration, kCandidateFoundation4); address_relay.SetPort(port_relay++); - webrtc::Candidate candidate12(webrtc::ICE_CANDIDATE_COMPONENT_RTP, "udp", - address_relay, kCandidatePriority, "", "", - IceCandidateType::kRelay, - kCandidateGeneration, kCandidateFoundation4); + Candidate candidate12(ICE_CANDIDATE_COMPONENT_RTP, "udp", address_relay, + kCandidatePriority, "", "", IceCandidateType::kRelay, + kCandidateGeneration, kCandidateFoundation4); // voice candidates_.push_back(candidate1); @@ -1112,9 +1078,9 @@ class WebRtcSdpTest : public ::testing::Test { const IceCandidateCollection* video_candidates_collection = jdesc_.candidates(1); ASSERT_NE(nullptr, video_candidates_collection); - std::vector video_candidates; + std::vector video_candidates; for (size_t i = 0; i < video_candidates_collection->count(); ++i) { - webrtc::Candidate c = video_candidates_collection->at(i)->candidate(); + Candidate c = video_candidates_collection->at(i)->candidate(); c.set_transport_name("video_content_name"); video_candidates.push_back(c); } @@ -1131,13 +1097,13 @@ class WebRtcSdpTest : public ::testing::Test { desc_.transport_infos()[1].description.ice_ufrag.clear(); desc_.transport_infos()[1].description.ice_pwd.clear(); desc_.transport_infos()[1].description.connection_role = - webrtc::CONNECTIONROLE_NONE; + CONNECTIONROLE_NONE; // Set bundle-only flag. desc_.contents()[1].bundle_only = true; // Add BUNDLE group. - ContentGroup group(webrtc::GROUP_TYPE_BUNDLE); + ContentGroup group(GROUP_TYPE_BUNDLE); group.AddContentName(kAudioContentName); group.AddContentName(kVideoContentName); desc_.AddGroup(group); @@ -1179,8 +1145,8 @@ class WebRtcSdpTest : public ::testing::Test { absl::WrapUnique(audio_desc_)); desc_.AddContent(kVideoContentName, MediaProtocolType::kRtp, absl::WrapUnique(video_desc_)); - desc_.set_msid_signaling(webrtc::kMsidSignalingSsrcAttribute | - webrtc::kMsidSignalingSemantic); + desc_.set_msid_signaling(kMsidSignalingSsrcAttribute | + kMsidSignalingSemantic); ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(), jdesc_.session_version())); } @@ -1234,8 +1200,8 @@ class WebRtcSdpTest : public ::testing::Test { desc_.AddTransportInfo( TransportInfo(kVideoContentName3, MakeTransportDescription(kUfragVideo3, kPwdVideo3))); - desc_.set_msid_signaling(webrtc::kMsidSignalingMediaSection | - webrtc::kMsidSignalingSemantic); + desc_.set_msid_signaling(kMsidSignalingMediaSection | + kMsidSignalingSemantic); ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(), jdesc_.session_version())); @@ -1247,10 +1213,10 @@ class WebRtcSdpTest : public ::testing::Test { AudioContentDescription* audio = new AudioContentDescription(); audio->set_rtcp_mux(true); audio->set_rtcp_reduced_size(true); - audio->set_protocol(webrtc::kMediaProtocolSavpf); - audio->AddCodec(webrtc::CreateAudioCodec(111, "opus", 48000, 2)); - audio->AddCodec(webrtc::CreateAudioCodec(103, "ISAC", 16000, 1)); - audio->AddCodec(webrtc::CreateAudioCodec(104, "ISAC", 32000, 1)); + audio->set_protocol(kMediaProtocolSavpf); + audio->AddCodec(CreateAudioCodec(111, "opus", 48000, 2)); + audio->AddCodec(CreateAudioCodec(103, "ISAC", 16000, 1)); + audio->AddCodec(CreateAudioCodec(104, "ISAC", 32000, 1)); return audio; } @@ -1313,8 +1279,8 @@ class WebRtcSdpTest : public ::testing::Test { absl::WrapUnique(audio_desc)); // Enable signaling a=msid lines. - desc_.set_msid_signaling(webrtc::kMsidSignalingMediaSection | - webrtc::kMsidSignalingSemantic); + desc_.set_msid_signaling(kMsidSignalingMediaSection | + kMsidSignalingSemantic); ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(), jdesc_.session_version())); } @@ -1323,14 +1289,13 @@ class WebRtcSdpTest : public ::testing::Test { // configuration. VideoContentDescription* CreateVideoContentDescription() { VideoContentDescription* video = new VideoContentDescription(); - video->set_protocol(webrtc::kMediaProtocolSavpf); - video->AddCodec(webrtc::CreateVideoCodec(120, "VP8")); + video->set_protocol(kMediaProtocolSavpf); + video->AddCodec(CreateVideoCodec(120, "VP8")); return video; } - void CompareMediaContentDescription( - const webrtc::MediaContentDescription* cd1, - const webrtc::MediaContentDescription* cd2) { + void CompareMediaContentDescription(const MediaContentDescription* cd1, + const MediaContentDescription* cd2) { // type EXPECT_EQ(cd1->type(), cd2->type()); @@ -1346,13 +1311,13 @@ class WebRtcSdpTest : public ::testing::Test { // protocol // Use an equivalence class here, for old and new versions of the // protocol description. - if (cd1->protocol() == webrtc::kMediaProtocolDtlsSctp || - cd1->protocol() == webrtc::kMediaProtocolUdpDtlsSctp || - cd1->protocol() == webrtc::kMediaProtocolTcpDtlsSctp) { + if (cd1->protocol() == kMediaProtocolDtlsSctp || + cd1->protocol() == kMediaProtocolUdpDtlsSctp || + cd1->protocol() == kMediaProtocolTcpDtlsSctp) { const bool cd2_is_also_dtls_sctp = - cd2->protocol() == webrtc::kMediaProtocolDtlsSctp || - cd2->protocol() == webrtc::kMediaProtocolUdpDtlsSctp || - cd2->protocol() == webrtc::kMediaProtocolTcpDtlsSctp; + cd2->protocol() == kMediaProtocolDtlsSctp || + cd2->protocol() == kMediaProtocolUdpDtlsSctp || + cd2->protocol() == kMediaProtocolTcpDtlsSctp; EXPECT_TRUE(cd2_is_also_dtls_sctp); } else { EXPECT_EQ(cd1->protocol(), cd2->protocol()); @@ -1416,27 +1381,27 @@ class WebRtcSdpTest : public ::testing::Test { return; } for (size_t i = 0; i < desc1.contents().size(); ++i) { - const webrtc::ContentInfo& c1 = desc1.contents().at(i); - const webrtc::ContentInfo& c2 = desc2.contents().at(i); + const ContentInfo& c1 = desc1.contents().at(i); + const ContentInfo& c2 = desc2.contents().at(i); // ContentInfo properties. EXPECT_EQ(c1.mid(), c2.mid()); EXPECT_EQ(c1.type, c2.type); EXPECT_EQ(c1.rejected, c2.rejected); EXPECT_EQ(c1.bundle_only, c2.bundle_only); - ASSERT_EQ(webrtc::IsAudioContent(&c1), webrtc::IsAudioContent(&c2)); - if (webrtc::IsAudioContent(&c1)) { + ASSERT_EQ(IsAudioContent(&c1), IsAudioContent(&c2)); + if (IsAudioContent(&c1)) { CompareMediaContentDescription(c1.media_description(), c2.media_description()); } - ASSERT_EQ(webrtc::IsVideoContent(&c1), webrtc::IsVideoContent(&c2)); - if (webrtc::IsVideoContent(&c1)) { + ASSERT_EQ(IsVideoContent(&c1), IsVideoContent(&c2)); + if (IsVideoContent(&c1)) { CompareMediaContentDescription(c1.media_description(), c2.media_description()); } - ASSERT_EQ(webrtc::IsDataContent(&c1), webrtc::IsDataContent(&c2)); + ASSERT_EQ(IsDataContent(&c1), IsDataContent(&c2)); if (c1.media_description()->as_sctp()) { ASSERT_TRUE(c2.media_description()->as_sctp()); const SctpDataContentDescription* scd1 = @@ -1452,42 +1417,42 @@ class WebRtcSdpTest : public ::testing::Test { } // group - const webrtc::ContentGroups groups1 = desc1.groups(); - const webrtc::ContentGroups groups2 = desc2.groups(); + const ContentGroups groups1 = desc1.groups(); + const ContentGroups groups2 = desc2.groups(); EXPECT_EQ(groups1.size(), groups1.size()); if (groups1.size() != groups2.size()) { ADD_FAILURE(); return; } for (size_t i = 0; i < groups1.size(); ++i) { - const webrtc::ContentGroup group1 = groups1.at(i); - const webrtc::ContentGroup group2 = groups2.at(i); + const ContentGroup group1 = groups1.at(i); + const ContentGroup group2 = groups2.at(i); EXPECT_EQ(group1.semantics(), group2.semantics()); - const webrtc::ContentNames names1 = group1.content_names(); - const webrtc::ContentNames names2 = group2.content_names(); + const ContentNames names1 = group1.content_names(); + const ContentNames names2 = group2.content_names(); EXPECT_EQ(names1.size(), names2.size()); if (names1.size() != names2.size()) { ADD_FAILURE(); return; } - webrtc::ContentNames::const_iterator iter1 = names1.begin(); - webrtc::ContentNames::const_iterator iter2 = names2.begin(); + ContentNames::const_iterator iter1 = names1.begin(); + ContentNames::const_iterator iter2 = names2.begin(); while (iter1 != names1.end()) { EXPECT_EQ(*iter1++, *iter2++); } } // transport info - const webrtc::TransportInfos transports1 = desc1.transport_infos(); - const webrtc::TransportInfos transports2 = desc2.transport_infos(); + const TransportInfos transports1 = desc1.transport_infos(); + const TransportInfos transports2 = desc2.transport_infos(); EXPECT_EQ(transports1.size(), transports2.size()); if (transports1.size() != transports2.size()) { ADD_FAILURE(); return; } for (size_t i = 0; i < transports1.size(); ++i) { - const webrtc::TransportInfo transport1 = transports1.at(i); - const webrtc::TransportInfo transport2 = transports2.at(i); + const TransportInfo transport1 = transports1.at(i); + const TransportInfo transport2 = transports2.at(i); EXPECT_EQ(transport1.content_name, transport2.content_name); EXPECT_EQ(transport1.description.ice_ufrag, transport2.description.ice_ufrag); @@ -1583,7 +1548,7 @@ class WebRtcSdpTest : public ::testing::Test { void AddIceOptions(const std::string& content_name, const std::vector& transport_options) { ASSERT_TRUE(desc_.GetTransportInfoByName(content_name) != NULL); - webrtc::TransportInfo transport_info = + TransportInfo transport_info = *(desc_.GetTransportInfoByName(content_name)); desc_.RemoveTransportInfoByName(content_name); transport_info.description.transport_options = transport_options; @@ -1594,7 +1559,7 @@ class WebRtcSdpTest : public ::testing::Test { const std::string& ice_ufrag, const std::string& ice_pwd) { ASSERT_TRUE(desc_.GetTransportInfoByName(content_name) != NULL); - webrtc::TransportInfo transport_info = + TransportInfo transport_info = *(desc_.GetTransportInfoByName(content_name)); desc_.RemoveTransportInfoByName(content_name); transport_info.description.ice_ufrag = ice_ufrag; @@ -1620,7 +1585,7 @@ class WebRtcSdpTest : public ::testing::Test { // Removes everything in StreamParams from the session description that is // used for a=ssrc lines. void RemoveSsrcSignalingFromStreamParams() { - for (webrtc::ContentInfo& content_info : jdesc_.description()->contents()) { + for (ContentInfo& content_info : jdesc_.description()->contents()) { // With Unified Plan there should be one StreamParams per m= section. StreamParams& stream = content_info.media_description()->mutable_streams()[0]; @@ -1672,7 +1637,7 @@ class WebRtcSdpTest : public ::testing::Test { jdesc_.session_version())) { return false; } - std::string message = webrtc::SdpSerialize(jdesc_); + std::string message = SdpSerialize(jdesc_); EXPECT_EQ(new_sdp, message); return true; } @@ -1697,7 +1662,7 @@ class WebRtcSdpTest : public ::testing::Test { JsepSessionDescription jdesc_no_candidates(kDummyType); MakeDescriptionWithoutCandidates(&jdesc_no_candidates); - std::string message = webrtc::SdpSerialize(jdesc_no_candidates); + std::string message = SdpSerialize(jdesc_no_candidates); EXPECT_EQ(new_sdp, message); return true; } @@ -1707,7 +1672,7 @@ class WebRtcSdpTest : public ::testing::Test { new SctpDataContentDescription()); sctp_desc_ = data.get(); sctp_desc_->set_use_sctpmap(use_sctpmap); - sctp_desc_->set_protocol(webrtc::kMediaProtocolUdpDtlsSctp); + sctp_desc_->set_protocol(kMediaProtocolUdpDtlsSctp); sctp_desc_->set_port(kDefaultSctpPort); desc_.AddContent(kDataContentName, MediaProtocolType::kSctp, std::move(data)); @@ -1788,8 +1753,7 @@ class WebRtcSdpTest : public ::testing::Test { // media level. if (session_level && media_level) { SdpParseError error; - EXPECT_FALSE( - webrtc::SdpDeserialize(sdp_with_extmap, &jdesc_with_extmap, &error)); + EXPECT_FALSE(SdpDeserialize(sdp_with_extmap, &jdesc_with_extmap, &error)); EXPECT_NE(std::string::npos, error.description.find("a=extmap")); } else { EXPECT_TRUE(SdpDeserialize(sdp_with_extmap, &jdesc_with_extmap)); @@ -1797,10 +1761,10 @@ class WebRtcSdpTest : public ::testing::Test { } } - void VerifyCodecParameter(const webrtc::CodecParameterMap& params, + void VerifyCodecParameter(const CodecParameterMap& params, const std::string& name, int expected_value) { - webrtc::CodecParameterMap::const_iterator found = params.find(name); + CodecParameterMap::const_iterator found = params.find(name); ASSERT_TRUE(found != params.end()); EXPECT_EQ(found->second, absl::StrCat(expected_value)); } @@ -1851,13 +1815,13 @@ class WebRtcSdpTest : public ::testing::Test { // Deserialize SdpParseError error; - EXPECT_TRUE(webrtc::SdpDeserialize(sdp, jdesc_output, &error)); + EXPECT_TRUE(SdpDeserialize(sdp, jdesc_output, &error)); const AudioContentDescription* acd = - webrtc::GetFirstAudioContentDescription(jdesc_output->description()); + GetFirstAudioContentDescription(jdesc_output->description()); ASSERT_TRUE(acd); ASSERT_FALSE(acd->codecs().empty()); - webrtc::Codec opus = acd->codecs()[0]; + Codec opus = acd->codecs()[0]; EXPECT_EQ("opus", opus.name); EXPECT_EQ(111, opus.id); VerifyCodecParameter(opus.params, "minptime", params.min_ptime); @@ -1871,7 +1835,7 @@ class WebRtcSdpTest : public ::testing::Test { VerifyCodecParameter(codec.params, "maxptime", params.max_ptime); } - webrtc::Codec dtmf = acd->codecs()[3]; + Codec dtmf = acd->codecs()[3]; EXPECT_EQ("telephone-event", dtmf.name); EXPECT_EQ(105, dtmf.id); EXPECT_EQ(3u, @@ -1880,18 +1844,18 @@ class WebRtcSdpTest : public ::testing::Test { EXPECT_EQ(dtmf.params.begin()->second, "0-15,66,70"); const VideoContentDescription* vcd = - webrtc::GetFirstVideoContentDescription(jdesc_output->description()); + GetFirstVideoContentDescription(jdesc_output->description()); ASSERT_TRUE(vcd); ASSERT_FALSE(vcd->codecs().empty()); - webrtc::Codec vp8 = vcd->codecs()[0]; + Codec vp8 = vcd->codecs()[0]; EXPECT_EQ("VP8", vp8.name); EXPECT_EQ(99, vp8.id); - webrtc::Codec rtx = vcd->codecs()[1]; + Codec rtx = vcd->codecs()[1]; EXPECT_EQ("RTX", rtx.name); EXPECT_EQ(95, rtx.id); VerifyCodecParameter(rtx.params, "apt", vp8.id); // VP9 is listed last in the m= line so should come after VP8 and RTX. - webrtc::Codec vp9 = vcd->codecs()[2]; + Codec vp9 = vcd->codecs()[2]; EXPECT_EQ("VP9", vp9.name); EXPECT_EQ(96, vp9.id); } @@ -1924,33 +1888,33 @@ class WebRtcSdpTest : public ::testing::Test { std::string sdp = os.str(); // Deserialize SdpParseError error; - EXPECT_TRUE(webrtc::SdpDeserialize(sdp, jdesc_output, &error)); + EXPECT_TRUE(SdpDeserialize(sdp, jdesc_output, &error)); const AudioContentDescription* acd = - webrtc::GetFirstAudioContentDescription(jdesc_output->description()); + GetFirstAudioContentDescription(jdesc_output->description()); ASSERT_TRUE(acd); ASSERT_FALSE(acd->codecs().empty()); - webrtc::Codec opus = acd->codecs()[0]; + Codec opus = acd->codecs()[0]; EXPECT_EQ(111, opus.id); - EXPECT_TRUE(opus.HasFeedbackParam(webrtc::FeedbackParam( - webrtc::kRtcpFbParamNack, webrtc::kParamValueEmpty))); + EXPECT_TRUE(opus.HasFeedbackParam( + FeedbackParam(kRtcpFbParamNack, kParamValueEmpty))); const VideoContentDescription* vcd = - webrtc::GetFirstVideoContentDescription(jdesc_output->description()); + GetFirstVideoContentDescription(jdesc_output->description()); ASSERT_TRUE(vcd); ASSERT_FALSE(vcd->codecs().empty()); - webrtc::Codec vp8 = vcd->codecs()[0]; + Codec vp8 = vcd->codecs()[0]; EXPECT_EQ(vp8.name, "VP8"); EXPECT_EQ(101, vp8.id); - EXPECT_TRUE(vp8.HasFeedbackParam(webrtc::FeedbackParam( - webrtc::kRtcpFbParamLntf, webrtc::kParamValueEmpty))); - EXPECT_TRUE(vp8.HasFeedbackParam(webrtc::FeedbackParam( - webrtc::kRtcpFbParamNack, webrtc::kParamValueEmpty))); - EXPECT_TRUE(vp8.HasFeedbackParam(webrtc::FeedbackParam( - webrtc::kRtcpFbParamNack, webrtc::kRtcpFbNackParamPli))); - EXPECT_TRUE(vp8.HasFeedbackParam(webrtc::FeedbackParam( - webrtc::kRtcpFbParamRemb, webrtc::kParamValueEmpty))); - EXPECT_TRUE(vp8.HasFeedbackParam(webrtc::FeedbackParam( - webrtc::kRtcpFbParamCcm, webrtc::kRtcpFbCcmParamFir))); + EXPECT_TRUE(vp8.HasFeedbackParam( + FeedbackParam(kRtcpFbParamLntf, kParamValueEmpty))); + EXPECT_TRUE(vp8.HasFeedbackParam( + FeedbackParam(kRtcpFbParamNack, kParamValueEmpty))); + EXPECT_TRUE(vp8.HasFeedbackParam( + FeedbackParam(kRtcpFbParamNack, kRtcpFbNackParamPli))); + EXPECT_TRUE(vp8.HasFeedbackParam( + FeedbackParam(kRtcpFbParamRemb, kParamValueEmpty))); + EXPECT_TRUE(vp8.HasFeedbackParam( + FeedbackParam(kRtcpFbParamCcm, kRtcpFbCcmParamFir))); } // Two SDP messages can mean the same thing but be different strings, e.g. @@ -1960,10 +1924,10 @@ class WebRtcSdpTest : public ::testing::Test { // deserializing and comparing JsepSessionDescription will test // the serializer sufficiently. void TestSerialize(const JsepSessionDescription& jdesc) { - std::string message = webrtc::SdpSerialize(jdesc); + std::string message = SdpSerialize(jdesc); JsepSessionDescription jdesc_output_des(kDummyType); SdpParseError error; - EXPECT_TRUE(webrtc::SdpDeserialize(message, &jdesc_output_des, &error)); + EXPECT_TRUE(SdpDeserialize(message, &jdesc_output_des, &error)); EXPECT_TRUE(CompareSessionDescription(jdesc, jdesc_output_des)); } @@ -1972,8 +1936,8 @@ class WebRtcSdpTest : public ::testing::Test { // 'connection address' field, previously set from the candidates, must also // be reset. void MakeDescriptionWithoutCandidates(JsepSessionDescription* jdesc) { - webrtc::SocketAddress audio_addr("0.0.0.0", 9); - webrtc::SocketAddress video_addr("0.0.0.0", 9); + SocketAddress audio_addr("0.0.0.0", 9); + SocketAddress video_addr("0.0.0.0", 9); audio_desc_->set_connection_address(audio_addr); video_desc_->set_connection_address(video_addr); ASSERT_TRUE(jdesc->Initialize(desc_.Clone(), kSessionId, kSessionVersion)); @@ -2008,34 +1972,34 @@ void TestMismatch(const std::string& string1, const std::string& string2) { TEST_F(WebRtcSdpTest, SerializeSessionDescription) { // SessionDescription with desc and candidates. - std::string message = webrtc::SdpSerialize(jdesc_); + std::string message = SdpSerialize(jdesc_); TestMismatch(std::string(kSdpFullString), message); } TEST_F(WebRtcSdpTest, SerializeSessionDescriptionEmpty) { JsepSessionDescription jdesc_empty(kDummyType); - EXPECT_EQ("", webrtc::SdpSerialize(jdesc_empty)); + EXPECT_EQ("", SdpSerialize(jdesc_empty)); } TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithoutCandidates) { // JsepSessionDescription with desc but without candidates. JsepSessionDescription jdesc_no_candidates(kDummyType); MakeDescriptionWithoutCandidates(&jdesc_no_candidates); - std::string message = webrtc::SdpSerialize(jdesc_no_candidates); + std::string message = SdpSerialize(jdesc_no_candidates); EXPECT_EQ(std::string(kSdpString), message); } TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithBundles) { - ContentGroup group1(webrtc::GROUP_TYPE_BUNDLE); + ContentGroup group1(GROUP_TYPE_BUNDLE); group1.AddContentName(kAudioContentName); group1.AddContentName(kVideoContentName); desc_.AddGroup(group1); - ContentGroup group2(webrtc::GROUP_TYPE_BUNDLE); + ContentGroup group2(GROUP_TYPE_BUNDLE); group2.AddContentName(kAudioContentName2); desc_.AddGroup(group2); ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(), jdesc_.session_version())); - std::string message = webrtc::SdpSerialize(jdesc_); + std::string message = SdpSerialize(jdesc_); std::string sdp_with_bundle = kSdpFullString; InjectAfter(kSessionTime, "a=group:BUNDLE audio_content_name video_content_name\r\n" @@ -2045,17 +2009,15 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithBundles) { } TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithBandwidth) { - VideoContentDescription* vcd = - webrtc::GetFirstVideoContentDescription(&desc_); + VideoContentDescription* vcd = GetFirstVideoContentDescription(&desc_); vcd->set_bandwidth(100 * 1000 + 755); // Integer division will drop the 755. vcd->set_bandwidth_type("AS"); - AudioContentDescription* acd = - webrtc::GetFirstAudioContentDescription(&desc_); + AudioContentDescription* acd = GetFirstAudioContentDescription(&desc_); acd->set_bandwidth(555); acd->set_bandwidth_type("TIAS"); ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(), jdesc_.session_version())); - std::string message = webrtc::SdpSerialize(jdesc_); + std::string message = SdpSerialize(jdesc_); std::string sdp_with_bandwidth = kSdpFullString; InjectAfter("c=IN IP4 74.125.224.39\r\n", "b=AS:100\r\n", &sdp_with_bandwidth); @@ -2066,12 +2028,11 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithBandwidth) { // Should default to b=AS if bandwidth_type isn't set. TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithMissingBandwidthType) { - VideoContentDescription* vcd = - webrtc::GetFirstVideoContentDescription(&desc_); + VideoContentDescription* vcd = GetFirstVideoContentDescription(&desc_); vcd->set_bandwidth(100 * 1000); ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(), jdesc_.session_version())); - std::string message = webrtc::SdpSerialize(jdesc_); + std::string message = SdpSerialize(jdesc_); std::string sdp_with_bandwidth = kSdpFullString; InjectAfter("c=IN IP4 74.125.224.39\r\n", "b=AS:100\r\n", &sdp_with_bandwidth); @@ -2089,7 +2050,7 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithIceOptions) { AddIceOptions(kVideoContentName, transport_options); ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(), jdesc_.session_version())); - std::string message = webrtc::SdpSerialize(jdesc_); + std::string message = SdpSerialize(jdesc_); std::string sdp_with_ice_options = kSdpFullString; InjectAfter(kAttributeIcePwdVoice, "a=ice-options:iceoption1 iceoption3\r\n", &sdp_with_ice_options); @@ -2128,7 +2089,7 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithSctpDataChannel) { JsepSessionDescription jsep_desc(kDummyType); MakeDescriptionWithoutCandidates(&jsep_desc); - std::string message = webrtc::SdpSerialize(jsep_desc); + std::string message = SdpSerialize(jsep_desc); std::string expected_sdp = kSdpString; expected_sdp.append(kSdpSctpDataChannelString); @@ -2139,7 +2100,7 @@ void MutateJsepSctpPort(JsepSessionDescription* jdesc, const SessionDescription& desc, int port) { // Take our pre-built session description and change the SCTP port. - std::unique_ptr mutant = desc.Clone(); + std::unique_ptr mutant = desc.Clone(); SctpDataContentDescription* dcdesc = mutant->GetContentDescriptionByName(kDataContentName)->as_sctp(); dcdesc->set_port(port); @@ -2156,7 +2117,7 @@ TEST_F(WebRtcSdpTest, SerializeWithSctpDataChannelAndNewPort) { const int kNewPort = 1234; MutateJsepSctpPort(&jsep_desc, desc_, kNewPort); - std::string message = webrtc::SdpSerialize(jsep_desc); + std::string message = SdpSerialize(jsep_desc); std::string expected_sdp = kSdpString; expected_sdp.append(kSdpSctpDataChannelString); @@ -2174,16 +2135,14 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithExtmapAllowMixed) { } TEST_F(WebRtcSdpTest, SerializeMediaContentDescriptionWithExtmapAllowMixed) { - webrtc::MediaContentDescription* video_desc = + MediaContentDescription* video_desc = jdesc_.description()->GetContentDescriptionByName(kVideoContentName); ASSERT_TRUE(video_desc); - webrtc::MediaContentDescription* audio_desc = + MediaContentDescription* audio_desc = jdesc_.description()->GetContentDescriptionByName(kAudioContentName); ASSERT_TRUE(audio_desc); - video_desc->set_extmap_allow_mixed_enum( - webrtc::MediaContentDescription::kMedia); - audio_desc->set_extmap_allow_mixed_enum( - webrtc::MediaContentDescription::kMedia); + video_desc->set_extmap_allow_mixed_enum(MediaContentDescription::kMedia); + audio_desc->set_extmap_allow_mixed_enum(MediaContentDescription::kMedia); TestSerialize(jdesc_); } @@ -2192,7 +2151,7 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithExtmap) { AddExtmap(encrypted); JsepSessionDescription desc_with_extmap(kDummyType); MakeDescriptionWithoutCandidates(&desc_with_extmap); - std::string message = webrtc::SdpSerialize(desc_with_extmap); + std::string message = SdpSerialize(desc_with_extmap); std::string sdp_with_extmap = kSdpString; InjectAfter("a=mid:audio_content_name\r\n", kExtmap, &sdp_with_extmap); @@ -2211,51 +2170,50 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithExtmapEncrypted) { } TEST_F(WebRtcSdpTest, SerializeCandidates) { - std::string message = webrtc::SdpSerializeCandidate(*jcandidate_); + std::string message = SdpSerializeCandidate(*jcandidate_); EXPECT_EQ(std::string(kRawCandidate), message); Candidate candidate_with_ufrag(candidates_.front()); candidate_with_ufrag.set_username("ABC"); jcandidate_.reset(new JsepIceCandidate(std::string("audio_content_name"), 0, candidate_with_ufrag)); - message = webrtc::SdpSerializeCandidate(*jcandidate_); + message = SdpSerializeCandidate(*jcandidate_); EXPECT_EQ(std::string(kRawCandidate) + " ufrag ABC", message); Candidate candidate_with_network_info(candidates_.front()); candidate_with_network_info.set_network_id(1); jcandidate_.reset(new JsepIceCandidate(std::string("audio"), 0, candidate_with_network_info)); - message = webrtc::SdpSerializeCandidate(*jcandidate_); + message = SdpSerializeCandidate(*jcandidate_); EXPECT_EQ(std::string(kRawCandidate) + " network-id 1", message); candidate_with_network_info.set_network_cost(999); jcandidate_.reset(new JsepIceCandidate(std::string("audio"), 0, candidate_with_network_info)); - message = webrtc::SdpSerializeCandidate(*jcandidate_); + message = SdpSerializeCandidate(*jcandidate_); EXPECT_EQ(std::string(kRawCandidate) + " network-id 1 network-cost 999", message); } TEST_F(WebRtcSdpTest, SerializeHostnameCandidate) { - webrtc::SocketAddress address("a.test", 1234); - webrtc::Candidate candidate(webrtc::ICE_CANDIDATE_COMPONENT_RTP, "udp", - address, kCandidatePriority, "", "", - IceCandidateType::kHost, kCandidateGeneration, - kCandidateFoundation1); + SocketAddress address("a.test", 1234); + Candidate candidate(ICE_CANDIDATE_COMPONENT_RTP, "udp", address, + kCandidatePriority, "", "", IceCandidateType::kHost, + kCandidateGeneration, kCandidateFoundation1); JsepIceCandidate jcandidate(std::string("audio_content_name"), 0, candidate); - std::string message = webrtc::SdpSerializeCandidate(jcandidate); + std::string message = SdpSerializeCandidate(jcandidate); EXPECT_EQ(std::string(kRawHostnameCandidate), message); } TEST_F(WebRtcSdpTest, SerializeTcpCandidates) { - Candidate candidate(webrtc::ICE_CANDIDATE_COMPONENT_RTP, "tcp", - webrtc::SocketAddress("192.168.1.5", 9), - kCandidatePriority, "", "", IceCandidateType::kHost, - kCandidateGeneration, kCandidateFoundation1); - candidate.set_tcptype(webrtc::TCPTYPE_ACTIVE_STR); + Candidate candidate(ICE_CANDIDATE_COMPONENT_RTP, "tcp", + SocketAddress("192.168.1.5", 9), kCandidatePriority, "", + "", IceCandidateType::kHost, kCandidateGeneration, + kCandidateFoundation1); + candidate.set_tcptype(TCPTYPE_ACTIVE_STR); std::unique_ptr jcandidate( new JsepIceCandidate(std::string("audio_content_name"), 0, candidate)); - std::string message = webrtc::SdpSerializeCandidate(*jcandidate); + std::string message = SdpSerializeCandidate(*jcandidate); EXPECT_EQ(std::string(kSdpTcpActiveCandidate), message); } @@ -2271,8 +2229,7 @@ TEST_F(WebRtcSdpTest, ParseTcpCandidateWithoutTcptype) { JsepIceCandidate jcandidate(kDummyMid, kDummyIndex); EXPECT_TRUE(SdpDeserializeCandidate(missing_tcptype, &jcandidate)); - EXPECT_EQ(std::string(webrtc::TCPTYPE_PASSIVE_STR), - jcandidate.candidate().tcptype()); + EXPECT_EQ(std::string(TCPTYPE_PASSIVE_STR), jcandidate.candidate().tcptype()); } TEST_F(WebRtcSdpTest, ParseSslTcpCandidate) { @@ -2286,7 +2243,7 @@ TEST_F(WebRtcSdpTest, ParseSslTcpCandidate) { } TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithH264) { - webrtc::Codec h264_codec = webrtc::CreateVideoCodec("H264"); + Codec h264_codec = CreateVideoCodec("H264"); // Id must be valid, but value doesn't matter. h264_codec.id = 123; h264_codec.SetParam("profile-level-id", "42e01f"); @@ -2296,7 +2253,7 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithH264) { jdesc_.Initialize(desc_.Clone(), kSessionId, kSessionVersion); - std::string message = webrtc::SdpSerialize(jdesc_); + std::string message = SdpSerialize(jdesc_); size_t after_pt = message.find(" H264/90000"); ASSERT_NE(after_pt, std::string::npos); size_t before_pt = message.rfind("a=rtpmap:", after_pt); @@ -2374,14 +2331,14 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithoutRtpmap) { JsepSessionDescription jdesc(kDummyType); EXPECT_TRUE(SdpDeserialize(kSdpNoRtpmapString, &jdesc)); - webrtc::AudioContentDescription* audio = - webrtc::GetFirstAudioContentDescription(jdesc.description()); - webrtc::Codecs ref_codecs; + AudioContentDescription* audio = + GetFirstAudioContentDescription(jdesc.description()); + Codecs ref_codecs; // The codecs in the AudioContentDescription should be in the same order as // the payload types (s) on the m= line. - ref_codecs.push_back(webrtc::CreateAudioCodec(0, "PCMU", 8000, 1)); - ref_codecs.push_back(webrtc::CreateAudioCodec(18, "G729", 8000, 1)); - ref_codecs.push_back(webrtc::CreateAudioCodec(103, "ISAC", 16000, 1)); + ref_codecs.push_back(CreateAudioCodec(0, "PCMU", 8000, 1)); + ref_codecs.push_back(CreateAudioCodec(18, "G729", 8000, 1)); + ref_codecs.push_back(CreateAudioCodec(103, "ISAC", 16000, 1)); EXPECT_EQ(ref_codecs, audio->codecs()); } @@ -2397,18 +2354,18 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithoutRtpmapButWithFmtp) { JsepSessionDescription jdesc(kDummyType); EXPECT_TRUE(SdpDeserialize(kSdpNoRtpmapString, &jdesc)); - webrtc::AudioContentDescription* audio = - webrtc::GetFirstAudioContentDescription(jdesc.description()); + AudioContentDescription* audio = + GetFirstAudioContentDescription(jdesc.description()); - webrtc::Codec g729 = audio->codecs()[0]; + Codec g729 = audio->codecs()[0]; EXPECT_EQ("G729", g729.name); EXPECT_EQ(8000, g729.clockrate); EXPECT_EQ(18, g729.id); - webrtc::CodecParameterMap::iterator found = g729.params.find("annexb"); + CodecParameterMap::iterator found = g729.params.find("annexb"); ASSERT_TRUE(found != g729.params.end()); EXPECT_EQ(found->second, "yes"); - webrtc::Codec isac = audio->codecs()[1]; + Codec isac = audio->codecs()[1]; EXPECT_EQ("ISAC", isac.name); EXPECT_EQ(103, isac.id); EXPECT_EQ(16000, isac.clockrate); @@ -2435,7 +2392,7 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithBundle) { "a=group:BUNDLE audio_content_name video_content_name\r\n", &sdp_with_bundle); EXPECT_TRUE(SdpDeserialize(sdp_with_bundle, &jdesc_with_bundle)); - ContentGroup group(webrtc::GROUP_TYPE_BUNDLE); + ContentGroup group(GROUP_TYPE_BUNDLE); group.AddContentName(kAudioContentName); group.AddContentName(kVideoContentName); desc_.AddGroup(group); @@ -2452,11 +2409,9 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithBandwidth) { InjectAfter("a=mid:audio_content_name\r\na=sendrecv\r\n", "b=AS:50\r\n", &sdp_with_bandwidth); EXPECT_TRUE(SdpDeserialize(sdp_with_bandwidth, &jdesc_with_bandwidth)); - VideoContentDescription* vcd = - webrtc::GetFirstVideoContentDescription(&desc_); + VideoContentDescription* vcd = GetFirstVideoContentDescription(&desc_); vcd->set_bandwidth(100 * 1000); - AudioContentDescription* acd = - webrtc::GetFirstAudioContentDescription(&desc_); + AudioContentDescription* acd = GetFirstAudioContentDescription(&desc_); acd->set_bandwidth(50 * 1000); ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(), jdesc_.session_version())); @@ -2471,11 +2426,9 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithTiasBandwidth) { InjectAfter("a=mid:audio_content_name\r\na=sendrecv\r\n", "b=TIAS:50000\r\n", &sdp_with_bandwidth); EXPECT_TRUE(SdpDeserialize(sdp_with_bandwidth, &jdesc_with_bandwidth)); - VideoContentDescription* vcd = - webrtc::GetFirstVideoContentDescription(&desc_); + VideoContentDescription* vcd = GetFirstVideoContentDescription(&desc_); vcd->set_bandwidth(100 * 1000); - AudioContentDescription* acd = - webrtc::GetFirstAudioContentDescription(&desc_); + AudioContentDescription* acd = GetFirstAudioContentDescription(&desc_); acd->set_bandwidth(50 * 1000); ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(), jdesc_.session_version())); @@ -2491,11 +2444,9 @@ TEST_F(WebRtcSdpTest, InjectAfter("a=mid:audio_content_name\r\na=sendrecv\r\n", "b=unknown:50000\r\n", &sdp_with_bandwidth); EXPECT_TRUE(SdpDeserialize(sdp_with_bandwidth, &jdesc_with_bandwidth)); - VideoContentDescription* vcd = - webrtc::GetFirstVideoContentDescription(&desc_); + VideoContentDescription* vcd = GetFirstVideoContentDescription(&desc_); vcd->set_bandwidth(-1); - AudioContentDescription* acd = - webrtc::GetFirstAudioContentDescription(&desc_); + AudioContentDescription* acd = GetFirstAudioContentDescription(&desc_); acd->set_bandwidth(-1); ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(), jdesc_.session_version())); @@ -2596,16 +2547,14 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithoutExtmapAllowMixed) { } TEST_F(WebRtcSdpTest, DeserializeMediaContentDescriptionWithExtmapAllowMixed) { - webrtc::MediaContentDescription* video_desc = + MediaContentDescription* video_desc = jdesc_.description()->GetContentDescriptionByName(kVideoContentName); ASSERT_TRUE(video_desc); - webrtc::MediaContentDescription* audio_desc = + MediaContentDescription* audio_desc = jdesc_.description()->GetContentDescriptionByName(kAudioContentName); ASSERT_TRUE(audio_desc); - video_desc->set_extmap_allow_mixed_enum( - webrtc::MediaContentDescription::kMedia); - audio_desc->set_extmap_allow_mixed_enum( - webrtc::MediaContentDescription::kMedia); + video_desc->set_extmap_allow_mixed_enum(MediaContentDescription::kMedia); + audio_desc->set_extmap_allow_mixed_enum(MediaContentDescription::kMedia); std::string sdp_with_extmap_allow_mixed = kSdpFullString; InjectAfter("a=mid:audio_content_name\r\n", kExtmapAllowMixed, @@ -2658,11 +2607,11 @@ TEST_F(WebRtcSdpTest, DeserializeCandidate) { sdp = kSdpTcpActiveCandidate; EXPECT_TRUE(SdpDeserializeCandidate(sdp, &jcandidate)); - // Make a webrtc::Candidate equivalent to kSdpTcpCandidate string. - Candidate candidate(webrtc::ICE_CANDIDATE_COMPONENT_RTP, "tcp", - webrtc::SocketAddress("192.168.1.5", 9), - kCandidatePriority, "", "", IceCandidateType::kHost, - kCandidateGeneration, kCandidateFoundation1); + // Make a Candidate equivalent to kSdpTcpCandidate string. + Candidate candidate(ICE_CANDIDATE_COMPONENT_RTP, "tcp", + SocketAddress("192.168.1.5", 9), kCandidatePriority, "", + "", IceCandidateType::kHost, kCandidateGeneration, + kCandidateFoundation1); std::unique_ptr jcandidate_template( new JsepIceCandidate(std::string("audio_content_name"), 0, candidate)); EXPECT_TRUE( @@ -2809,7 +2758,7 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithSctpDataChannelsButWrongMediaType) { void MutateJsepSctpMaxMessageSize(const SessionDescription& desc, int new_value, JsepSessionDescription* jdesc) { - std::unique_ptr mutant = desc.Clone(); + std::unique_ptr mutant = desc.Clone(); SctpDataContentDescription* dcdesc = mutant->GetContentDescriptionByName(kDataContentName)->as_sctp(); dcdesc->set_max_message_size(new_value); @@ -2836,7 +2785,7 @@ TEST_F(WebRtcSdpTest, SerializeSdpWithSctpDataChannelWithMaxMessageSize) { AddSctpDataChannel(use_sctpmap); JsepSessionDescription jdesc(kDummyType); MutateJsepSctpMaxMessageSize(desc_, 12345, &jdesc); - std::string message = webrtc::SdpSerialize(jdesc); + std::string message = SdpSerialize(jdesc); EXPECT_NE(std::string::npos, message.find("\r\na=max-message-size:12345\r\n")); JsepSessionDescription jdesc_output(kDummyType); @@ -2852,7 +2801,7 @@ TEST_F(WebRtcSdpTest, AddSctpDataChannel(use_sctpmap); JsepSessionDescription jdesc(kDummyType); MutateJsepSctpMaxMessageSize(desc_, 65536, &jdesc); - std::string message = webrtc::SdpSerialize(jdesc); + std::string message = SdpSerialize(jdesc); EXPECT_EQ(std::string::npos, message.find("\r\na=max-message-size:")); JsepSessionDescription jdesc_output(kDummyType); EXPECT_TRUE(SdpDeserialize(message, &jdesc_output)); @@ -2948,8 +2897,7 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithSctpDataChannelsAndBandwidth) { bool use_sctpmap = true; AddSctpDataChannel(use_sctpmap); JsepSessionDescription jdesc(kDummyType); - SctpDataContentDescription* dcd = - webrtc::GetFirstSctpDataContentDescription(&desc_); + SctpDataContentDescription* dcd = GetFirstSctpDataContentDescription(&desc_); dcd->set_bandwidth(100 * 1000); ASSERT_TRUE(jdesc.Initialize(desc_.Clone(), kSessionId, kSessionVersion)); @@ -2994,7 +2942,7 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithoutEndLineBreak) { sdp = sdp.substr(0, sdp.size() - 2); // Remove \r\n at the end. // Deserialize SdpParseError error; - EXPECT_FALSE(webrtc::SdpDeserialize(sdp, &jdesc, &error)); + EXPECT_FALSE(SdpDeserialize(sdp, &jdesc, &error)); const std::string lastline = "a=ssrc:3 cname:stream_1_cname"; EXPECT_EQ(lastline, error.line); EXPECT_EQ("Invalid SDP line.", error.description); @@ -3032,12 +2980,12 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithConferenceFlag) { EXPECT_TRUE(SdpDeserialize(kSdpConferenceString, &jdesc)); // Verify - webrtc::AudioContentDescription* audio = - webrtc::GetFirstAudioContentDescription(jdesc.description()); + AudioContentDescription* audio = + GetFirstAudioContentDescription(jdesc.description()); EXPECT_TRUE(audio->conference_mode()); - webrtc::VideoContentDescription* video = - webrtc::GetFirstVideoContentDescription(jdesc.description()); + VideoContentDescription* video = + GetFirstVideoContentDescription(jdesc.description()); EXPECT_TRUE(video->conference_mode()); } @@ -3047,16 +2995,16 @@ TEST_F(WebRtcSdpTest, SerializeSdpWithConferenceFlag) { // We tested deserialization already above, so just test that if we serialize // and deserialize the flag doesn't disappear. EXPECT_TRUE(SdpDeserialize(kSdpConferenceString, &jdesc)); - std::string reserialized = webrtc::SdpSerialize(jdesc); + std::string reserialized = SdpSerialize(jdesc); EXPECT_TRUE(SdpDeserialize(reserialized, &jdesc)); // Verify. - webrtc::AudioContentDescription* audio = - webrtc::GetFirstAudioContentDescription(jdesc.description()); + AudioContentDescription* audio = + GetFirstAudioContentDescription(jdesc.description()); EXPECT_TRUE(audio->conference_mode()); - webrtc::VideoContentDescription* video = - webrtc::GetFirstVideoContentDescription(jdesc.description()); + VideoContentDescription* video = + GetFirstVideoContentDescription(jdesc.description()); EXPECT_TRUE(video->conference_mode()); } @@ -3064,18 +3012,18 @@ TEST_F(WebRtcSdpTest, SerializeAndDeserializeRemoteNetEstimate) { { // By default remote estimates are disabled. JsepSessionDescription dst(kDummyType); - SdpDeserialize(webrtc::SdpSerialize(jdesc_), &dst); - EXPECT_FALSE(webrtc::GetFirstVideoContentDescription(dst.description()) - ->remote_estimate()); + SdpDeserialize(SdpSerialize(jdesc_), &dst); + EXPECT_FALSE( + GetFirstVideoContentDescription(dst.description())->remote_estimate()); } { // When remote estimate is enabled, the setting is propagated via SDP. - webrtc::GetFirstVideoContentDescription(jdesc_.description()) + GetFirstVideoContentDescription(jdesc_.description()) ->set_remote_estimate(true); JsepSessionDescription dst(kDummyType); - SdpDeserialize(webrtc::SdpSerialize(jdesc_), &dst); - EXPECT_TRUE(webrtc::GetFirstVideoContentDescription(dst.description()) - ->remote_estimate()); + SdpDeserialize(SdpSerialize(jdesc_), &dst); + EXPECT_TRUE( + GetFirstVideoContentDescription(dst.description())->remote_estimate()); } } @@ -3190,7 +3138,7 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithReorderedPltypes) { EXPECT_TRUE(SdpDeserialize(kSdpWithReorderedPlTypesString, &jdesc_output)); const AudioContentDescription* acd = - webrtc::GetFirstAudioContentDescription(jdesc_output.description()); + GetFirstAudioContentDescription(jdesc_output.description()); ASSERT_TRUE(acd); ASSERT_FALSE(acd->codecs().empty()); EXPECT_EQ("ISAC", acd->codecs()[0].name); @@ -3240,18 +3188,16 @@ TEST_F(WebRtcSdpTest, DeserializeVideoFmtp) { // Deserialize SdpParseError error; - EXPECT_TRUE( - webrtc::SdpDeserialize(kSdpWithFmtpString, &jdesc_output, &error)); + EXPECT_TRUE(SdpDeserialize(kSdpWithFmtpString, &jdesc_output, &error)); const VideoContentDescription* vcd = - webrtc::GetFirstVideoContentDescription(jdesc_output.description()); + GetFirstVideoContentDescription(jdesc_output.description()); ASSERT_TRUE(vcd); ASSERT_FALSE(vcd->codecs().empty()); - webrtc::Codec vp8 = vcd->codecs()[0]; + Codec vp8 = vcd->codecs()[0]; EXPECT_EQ("VP8", vp8.name); EXPECT_EQ(120, vp8.id); - webrtc::CodecParameterMap::iterator found = - vp8.params.find("x-google-min-bitrate"); + CodecParameterMap::iterator found = vp8.params.find("x-google-min-bitrate"); ASSERT_TRUE(found != vp8.params.end()); EXPECT_EQ(found->second, "10"); found = vp8.params.find("x-google-max-quantization"); @@ -3274,17 +3220,16 @@ TEST_F(WebRtcSdpTest, DeserializeVideoFmtpWithSprops) { // Deserialize. SdpParseError error; - EXPECT_TRUE( - webrtc::SdpDeserialize(kSdpWithFmtpString, &jdesc_output, &error)); + EXPECT_TRUE(SdpDeserialize(kSdpWithFmtpString, &jdesc_output, &error)); const VideoContentDescription* vcd = - webrtc::GetFirstVideoContentDescription(jdesc_output.description()); + GetFirstVideoContentDescription(jdesc_output.description()); ASSERT_TRUE(vcd); ASSERT_FALSE(vcd->codecs().empty()); - webrtc::Codec h264 = vcd->codecs()[0]; + Codec h264 = vcd->codecs()[0]; EXPECT_EQ("H264", h264.name); EXPECT_EQ(98, h264.id); - webrtc::CodecParameterMap::const_iterator found = + CodecParameterMap::const_iterator found = h264.params.find("profile-level-id"); ASSERT_TRUE(found != h264.params.end()); EXPECT_EQ(found->second, "42A01E"); @@ -3307,18 +3252,16 @@ TEST_F(WebRtcSdpTest, DeserializeVideoFmtpWithSpace) { // Deserialize SdpParseError error; - EXPECT_TRUE( - webrtc::SdpDeserialize(kSdpWithFmtpString, &jdesc_output, &error)); + EXPECT_TRUE(SdpDeserialize(kSdpWithFmtpString, &jdesc_output, &error)); const VideoContentDescription* vcd = - webrtc::GetFirstVideoContentDescription(jdesc_output.description()); + GetFirstVideoContentDescription(jdesc_output.description()); ASSERT_TRUE(vcd); ASSERT_FALSE(vcd->codecs().empty()); - webrtc::Codec vp8 = vcd->codecs()[0]; + Codec vp8 = vcd->codecs()[0]; EXPECT_EQ("VP8", vp8.name); EXPECT_EQ(120, vp8.id); - webrtc::CodecParameterMap::iterator found = - vp8.params.find("x-google-min-bitrate"); + CodecParameterMap::iterator found = vp8.params.find("x-google-min-bitrate"); ASSERT_TRUE(found != vp8.params.end()); EXPECT_EQ(found->second, "10"); found = vp8.params.find("x-google-max-quantization"); @@ -3345,46 +3288,45 @@ TEST_F(WebRtcSdpTest, DeserializePacketizationAttributeWithIllegalValue) { "a=packetization:122 unknownpacketizationattributevalue\r\n"; SdpParseError error; - EXPECT_TRUE(webrtc::SdpDeserialize(kSdpWithPacketizationString, &jdesc_output, - &error)); + EXPECT_TRUE( + SdpDeserialize(kSdpWithPacketizationString, &jdesc_output, &error)); AudioContentDescription* acd = - webrtc::GetFirstAudioContentDescription(jdesc_output.description()); + GetFirstAudioContentDescription(jdesc_output.description()); ASSERT_TRUE(acd); ASSERT_THAT(acd->codecs(), testing::SizeIs(1)); - webrtc::Codec opus = acd->codecs()[0]; + Codec opus = acd->codecs()[0]; EXPECT_EQ(opus.name, "opus"); EXPECT_EQ(opus.id, 111); const VideoContentDescription* vcd = - webrtc::GetFirstVideoContentDescription(jdesc_output.description()); + GetFirstVideoContentDescription(jdesc_output.description()); ASSERT_TRUE(vcd); ASSERT_THAT(vcd->codecs(), testing::SizeIs(3)); - webrtc::Codec vp8 = vcd->codecs()[0]; + Codec vp8 = vcd->codecs()[0]; EXPECT_EQ(vp8.name, "VP8"); EXPECT_EQ(vp8.id, 120); EXPECT_EQ(vp8.packetization, "raw"); - webrtc::Codec vp9 = vcd->codecs()[1]; + Codec vp9 = vcd->codecs()[1]; EXPECT_EQ(vp9.name, "VP9"); EXPECT_EQ(vp9.id, 121); EXPECT_EQ(vp9.packetization, std::nullopt); - webrtc::Codec h264 = vcd->codecs()[2]; + Codec h264 = vcd->codecs()[2]; EXPECT_EQ(h264.name, "H264"); EXPECT_EQ(h264.id, 122); EXPECT_EQ(h264.packetization, std::nullopt); } TEST_F(WebRtcSdpTest, SerializeAudioFmtpWithUnknownParameter) { - AudioContentDescription* acd = - webrtc::GetFirstAudioContentDescription(&desc_); + AudioContentDescription* acd = GetFirstAudioContentDescription(&desc_); - webrtc::Codecs codecs = acd->codecs(); + Codecs codecs = acd->codecs(); codecs[0].params["unknown-future-parameter"] = "SomeFutureValue"; acd->set_codecs(codecs); ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(), jdesc_.session_version())); - std::string message = webrtc::SdpSerialize(jdesc_); + std::string message = SdpSerialize(jdesc_); std::string sdp_with_fmtp = kSdpFullString; InjectAfter("a=rtpmap:111 opus/48000/2\r\n", "a=fmtp:111 unknown-future-parameter=SomeFutureValue\r\n", @@ -3393,16 +3335,15 @@ TEST_F(WebRtcSdpTest, SerializeAudioFmtpWithUnknownParameter) { } TEST_F(WebRtcSdpTest, SerializeAudioFmtpWithKnownFmtpParameter) { - AudioContentDescription* acd = - webrtc::GetFirstAudioContentDescription(&desc_); + AudioContentDescription* acd = GetFirstAudioContentDescription(&desc_); - webrtc::Codecs codecs = acd->codecs(); + Codecs codecs = acd->codecs(); codecs[0].params["stereo"] = "1"; acd->set_codecs(codecs); ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(), jdesc_.session_version())); - std::string message = webrtc::SdpSerialize(jdesc_); + std::string message = SdpSerialize(jdesc_); std::string sdp_with_fmtp = kSdpFullString; InjectAfter("a=rtpmap:111 opus/48000/2\r\n", "a=fmtp:111 stereo=1\r\n", &sdp_with_fmtp); @@ -3410,17 +3351,16 @@ TEST_F(WebRtcSdpTest, SerializeAudioFmtpWithKnownFmtpParameter) { } TEST_F(WebRtcSdpTest, SerializeAudioFmtpWithPTimeAndMaxPTime) { - AudioContentDescription* acd = - webrtc::GetFirstAudioContentDescription(&desc_); + AudioContentDescription* acd = GetFirstAudioContentDescription(&desc_); - webrtc::Codecs codecs = acd->codecs(); + Codecs codecs = acd->codecs(); codecs[0].params["ptime"] = "20"; codecs[0].params["maxptime"] = "120"; acd->set_codecs(codecs); ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(), jdesc_.session_version())); - std::string message = webrtc::SdpSerialize(jdesc_); + std::string message = SdpSerialize(jdesc_); std::string sdp_with_fmtp = kSdpFullString; InjectAfter("a=rtpmap:104 ISAC/32000\r\n", "a=maxptime:120\r\n" // No comma here. String merging! @@ -3430,19 +3370,17 @@ TEST_F(WebRtcSdpTest, SerializeAudioFmtpWithPTimeAndMaxPTime) { } TEST_F(WebRtcSdpTest, SerializeAudioFmtpWithTelephoneEvent) { - AudioContentDescription* acd = - webrtc::GetFirstAudioContentDescription(&desc_); + AudioContentDescription* acd = GetFirstAudioContentDescription(&desc_); - webrtc::Codecs codecs = acd->codecs(); - webrtc::Codec dtmf = - webrtc::CreateAudioCodec(105, "telephone-event", 8000, 1); + Codecs codecs = acd->codecs(); + Codec dtmf = CreateAudioCodec(105, "telephone-event", 8000, 1); dtmf.params[""] = "0-15"; codecs.push_back(dtmf); acd->set_codecs(codecs); ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(), jdesc_.session_version())); - std::string message = webrtc::SdpSerialize(jdesc_); + std::string message = SdpSerialize(jdesc_); std::string sdp_with_fmtp = kSdpFullString; InjectAfter("m=audio 2345 RTP/SAVPF 111 103 104", " 105", &sdp_with_fmtp); InjectAfter( @@ -3454,16 +3392,15 @@ TEST_F(WebRtcSdpTest, SerializeAudioFmtpWithTelephoneEvent) { } TEST_F(WebRtcSdpTest, SerializeVideoFmtp) { - VideoContentDescription* vcd = - webrtc::GetFirstVideoContentDescription(&desc_); + VideoContentDescription* vcd = GetFirstVideoContentDescription(&desc_); - webrtc::Codecs codecs = vcd->codecs(); + Codecs codecs = vcd->codecs(); codecs[0].params["x-google-min-bitrate"] = "10"; vcd->set_codecs(codecs); ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(), jdesc_.session_version())); - std::string message = webrtc::SdpSerialize(jdesc_); + std::string message = SdpSerialize(jdesc_); std::string sdp_with_fmtp = kSdpFullString; InjectAfter("a=rtpmap:120 VP8/90000\r\n", "a=fmtp:120 x-google-min-bitrate=10\r\n", &sdp_with_fmtp); @@ -3471,16 +3408,15 @@ TEST_F(WebRtcSdpTest, SerializeVideoFmtp) { } TEST_F(WebRtcSdpTest, SerializeVideoPacketizationAttribute) { - VideoContentDescription* vcd = - webrtc::GetFirstVideoContentDescription(&desc_); + VideoContentDescription* vcd = GetFirstVideoContentDescription(&desc_); - webrtc::Codecs codecs = vcd->codecs(); + Codecs codecs = vcd->codecs(); codecs[0].packetization = "raw"; vcd->set_codecs(codecs); ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(), jdesc_.session_version())); - std::string message = webrtc::SdpSerialize(jdesc_); + std::string message = SdpSerialize(jdesc_); std::string sdp_with_packetization = kSdpFullString; InjectAfter("a=rtpmap:120 VP8/90000\r\n", "a=packetization:120 raw\r\n", &sdp_with_packetization); @@ -3492,24 +3428,24 @@ TEST_F(WebRtcSdpTest, DeserializeAndSerializeSdpWithIceLite) { JsepSessionDescription jdesc_with_icelite(kDummyType); std::string sdp_with_icelite = kSdpFullString; EXPECT_TRUE(SdpDeserialize(sdp_with_icelite, &jdesc_with_icelite)); - webrtc::SessionDescription* desc = jdesc_with_icelite.description(); - const webrtc::TransportInfo* tinfo1 = + SessionDescription* desc = jdesc_with_icelite.description(); + const TransportInfo* tinfo1 = desc->GetTransportInfoByName("audio_content_name"); - EXPECT_EQ(webrtc::ICEMODE_FULL, tinfo1->description.ice_mode); - const webrtc::TransportInfo* tinfo2 = + EXPECT_EQ(ICEMODE_FULL, tinfo1->description.ice_mode); + const TransportInfo* tinfo2 = desc->GetTransportInfoByName("video_content_name"); - EXPECT_EQ(webrtc::ICEMODE_FULL, tinfo2->description.ice_mode); + EXPECT_EQ(ICEMODE_FULL, tinfo2->description.ice_mode); // Add "a=ice-lite" and deserialize, making sure it's ICE lite. InjectAfter(kSessionTime, "a=ice-lite\r\n", &sdp_with_icelite); EXPECT_TRUE(SdpDeserialize(sdp_with_icelite, &jdesc_with_icelite)); desc = jdesc_with_icelite.description(); - const webrtc::TransportInfo* atinfo = + const TransportInfo* atinfo = desc->GetTransportInfoByName("audio_content_name"); - EXPECT_EQ(webrtc::ICEMODE_LITE, atinfo->description.ice_mode); - const webrtc::TransportInfo* vtinfo = + EXPECT_EQ(ICEMODE_LITE, atinfo->description.ice_mode); + const TransportInfo* vtinfo = desc->GetTransportInfoByName("video_content_name"); - EXPECT_EQ(webrtc::ICEMODE_LITE, vtinfo->description.ice_mode); + EXPECT_EQ(ICEMODE_LITE, vtinfo->description.ice_mode); // Now that we know deserialization works, we can use TestSerialize to test // serialization. @@ -3524,23 +3460,21 @@ TEST_F(WebRtcSdpTest, RoundTripSdpWithSctpDataChannelsWithCandidates) { JsepSessionDescription jdesc_output(kDummyType); EXPECT_TRUE(SdpDeserialize(sdp_with_data, &jdesc_output)); - EXPECT_EQ(sdp_with_data, webrtc::SdpSerialize(jdesc_output)); + EXPECT_EQ(sdp_with_data, SdpSerialize(jdesc_output)); } TEST_F(WebRtcSdpTest, SerializeDtlsSetupAttribute) { TransportInfo audio_transport_info = *(desc_.GetTransportInfoByName(kAudioContentName)); - EXPECT_EQ(webrtc::CONNECTIONROLE_NONE, + EXPECT_EQ(CONNECTIONROLE_NONE, audio_transport_info.description.connection_role); - audio_transport_info.description.connection_role = - webrtc::CONNECTIONROLE_ACTIVE; + audio_transport_info.description.connection_role = CONNECTIONROLE_ACTIVE; TransportInfo video_transport_info = *(desc_.GetTransportInfoByName(kVideoContentName)); - EXPECT_EQ(webrtc::CONNECTIONROLE_NONE, + EXPECT_EQ(CONNECTIONROLE_NONE, video_transport_info.description.connection_role); - video_transport_info.description.connection_role = - webrtc::CONNECTIONROLE_ACTIVE; + video_transport_info.description.connection_role = CONNECTIONROLE_ACTIVE; desc_.RemoveTransportInfoByName(kAudioContentName); desc_.RemoveTransportInfoByName(kVideoContentName); @@ -3550,7 +3484,7 @@ TEST_F(WebRtcSdpTest, SerializeDtlsSetupAttribute) { ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(), jdesc_.session_version())); - std::string message = webrtc::SdpSerialize(jdesc_); + std::string message = SdpSerialize(jdesc_); std::string sdp_with_dtlssetup = kSdpFullString; // Now adding `setup` attribute. @@ -3563,15 +3497,13 @@ TEST_F(WebRtcSdpTest, DeserializeDtlsSetupAttributeActpass) { std::string sdp_with_dtlssetup = kSdpFullString; InjectAfter(kSessionTime, "a=setup:actpass\r\n", &sdp_with_dtlssetup); EXPECT_TRUE(SdpDeserialize(sdp_with_dtlssetup, &jdesc_with_dtlssetup)); - webrtc::SessionDescription* desc = jdesc_with_dtlssetup.description(); - const webrtc::TransportInfo* atinfo = + SessionDescription* desc = jdesc_with_dtlssetup.description(); + const TransportInfo* atinfo = desc->GetTransportInfoByName("audio_content_name"); - EXPECT_EQ(webrtc::CONNECTIONROLE_ACTPASS, - atinfo->description.connection_role); - const webrtc::TransportInfo* vtinfo = + EXPECT_EQ(CONNECTIONROLE_ACTPASS, atinfo->description.connection_role); + const TransportInfo* vtinfo = desc->GetTransportInfoByName("video_content_name"); - EXPECT_EQ(webrtc::CONNECTIONROLE_ACTPASS, - vtinfo->description.connection_role); + EXPECT_EQ(CONNECTIONROLE_ACTPASS, vtinfo->description.connection_role); } TEST_F(WebRtcSdpTest, DeserializeDtlsSetupAttributeActive) { @@ -3579,28 +3511,26 @@ TEST_F(WebRtcSdpTest, DeserializeDtlsSetupAttributeActive) { std::string sdp_with_dtlssetup = kSdpFullString; InjectAfter(kSessionTime, "a=setup:active\r\n", &sdp_with_dtlssetup); EXPECT_TRUE(SdpDeserialize(sdp_with_dtlssetup, &jdesc_with_dtlssetup)); - webrtc::SessionDescription* desc = jdesc_with_dtlssetup.description(); - const webrtc::TransportInfo* atinfo = + SessionDescription* desc = jdesc_with_dtlssetup.description(); + const TransportInfo* atinfo = desc->GetTransportInfoByName("audio_content_name"); - EXPECT_EQ(webrtc::CONNECTIONROLE_ACTIVE, atinfo->description.connection_role); - const webrtc::TransportInfo* vtinfo = + EXPECT_EQ(CONNECTIONROLE_ACTIVE, atinfo->description.connection_role); + const TransportInfo* vtinfo = desc->GetTransportInfoByName("video_content_name"); - EXPECT_EQ(webrtc::CONNECTIONROLE_ACTIVE, vtinfo->description.connection_role); + EXPECT_EQ(CONNECTIONROLE_ACTIVE, vtinfo->description.connection_role); } TEST_F(WebRtcSdpTest, DeserializeDtlsSetupAttributePassive) { JsepSessionDescription jdesc_with_dtlssetup(kDummyType); std::string sdp_with_dtlssetup = kSdpFullString; InjectAfter(kSessionTime, "a=setup:passive\r\n", &sdp_with_dtlssetup); EXPECT_TRUE(SdpDeserialize(sdp_with_dtlssetup, &jdesc_with_dtlssetup)); - webrtc::SessionDescription* desc = jdesc_with_dtlssetup.description(); - const webrtc::TransportInfo* atinfo = + SessionDescription* desc = jdesc_with_dtlssetup.description(); + const TransportInfo* atinfo = desc->GetTransportInfoByName("audio_content_name"); - EXPECT_EQ(webrtc::CONNECTIONROLE_PASSIVE, - atinfo->description.connection_role); - const webrtc::TransportInfo* vtinfo = + EXPECT_EQ(CONNECTIONROLE_PASSIVE, atinfo->description.connection_role); + const TransportInfo* vtinfo = desc->GetTransportInfoByName("video_content_name"); - EXPECT_EQ(webrtc::CONNECTIONROLE_PASSIVE, - vtinfo->description.connection_role); + EXPECT_EQ(CONNECTIONROLE_PASSIVE, vtinfo->description.connection_role); } // Verifies that the order of the serialized m-lines follows the order of the @@ -3609,9 +3539,8 @@ TEST_F(WebRtcSdpTest, MediaContentOrderMaintainedRoundTrip) { JsepSessionDescription jdesc(kDummyType); const std::string media_content_sdps[3] = {kSdpAudioString, kSdpVideoString, kSdpSctpDataChannelString}; - const webrtc::MediaType media_types[3] = {webrtc::MediaType::AUDIO, - webrtc::MediaType::VIDEO, - webrtc::MediaType::DATA}; + const MediaType media_types[3] = {MediaType::AUDIO, MediaType::VIDEO, + MediaType::DATA}; // Verifies all 6 permutations. for (size_t i = 0; i < 6; ++i) { @@ -3628,16 +3557,16 @@ TEST_F(WebRtcSdpTest, MediaContentOrderMaintainedRoundTrip) { sdp_string += media_content_sdps[media_content_in_sdp[j]]; EXPECT_TRUE(SdpDeserialize(sdp_string, &jdesc)); - webrtc::SessionDescription* desc = jdesc.description(); + SessionDescription* desc = jdesc.description(); EXPECT_EQ(3u, desc->contents().size()); for (size_t j = 0; j < 3; ++j) { - const webrtc::MediaContentDescription* mdesc = + const MediaContentDescription* mdesc = desc->contents()[j].media_description(); EXPECT_EQ(media_types[media_content_in_sdp[j]], mdesc->type()); } - std::string serialized_sdp = webrtc::SdpSerialize(jdesc); + std::string serialized_sdp = SdpSerialize(jdesc); EXPECT_EQ(sdp_string, serialized_sdp); } } @@ -3707,18 +3636,17 @@ TEST_F(WebRtcSdpTest, SerializeUnifiedPlanSessionDescription) { // lines do not support multiple stream ids and no stream ids. TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionSpecialMsid) { // Create both msid lines for Plan B and Unified Plan support. - MakeUnifiedPlanDescriptionMultipleStreamIds( - webrtc::kMsidSignalingMediaSection | webrtc::kMsidSignalingSsrcAttribute | - webrtc::kMsidSignalingSemantic); + MakeUnifiedPlanDescriptionMultipleStreamIds(kMsidSignalingMediaSection | + kMsidSignalingSsrcAttribute | + kMsidSignalingSemantic); JsepSessionDescription deserialized_description(kDummyType); EXPECT_TRUE(SdpDeserialize(kUnifiedPlanSdpFullStringWithSpecialMsid, &deserialized_description)); EXPECT_TRUE(CompareSessionDescription(jdesc_, deserialized_description)); - EXPECT_EQ(webrtc::kMsidSignalingMediaSection | - webrtc::kMsidSignalingSsrcAttribute | - webrtc::kMsidSignalingSemantic, + EXPECT_EQ(kMsidSignalingMediaSection | kMsidSignalingSsrcAttribute | + kMsidSignalingSemantic, deserialized_description.description()->msid_signaling()); } @@ -3728,10 +3656,10 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionSpecialMsid) { // multiple stream ids. TEST_F(WebRtcSdpTest, SerializeSessionDescriptionSpecialMsid) { // Create both msid lines for Plan B and Unified Plan support. - MakeUnifiedPlanDescriptionMultipleStreamIds( - webrtc::kMsidSignalingMediaSection | webrtc::kMsidSignalingSsrcAttribute | - webrtc::kMsidSignalingSemantic); - std::string serialized_sdp = webrtc::SdpSerialize(jdesc_); + MakeUnifiedPlanDescriptionMultipleStreamIds(kMsidSignalingMediaSection | + kMsidSignalingSsrcAttribute | + kMsidSignalingSemantic); + std::string serialized_sdp = SdpSerialize(jdesc_); // We explicitly test that the serialized SDP string is equal to the hard // coded SDP string. This is necessary, because in the parser "a=msid" lines // take priority over "a=ssrc msid" lines. This means if we just used @@ -3745,8 +3673,8 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionSpecialMsid) { // no stream ids and multiple stream ids. TEST_F(WebRtcSdpTest, UnifiedPlanDeserializeSessionDescriptionSpecialMsid) { // Only create a=msid lines for strictly Unified Plan stream ID support. - MakeUnifiedPlanDescriptionMultipleStreamIds( - webrtc::kMsidSignalingMediaSection | webrtc::kMsidSignalingSemantic); + MakeUnifiedPlanDescriptionMultipleStreamIds(kMsidSignalingMediaSection | + kMsidSignalingSemantic); JsepSessionDescription deserialized_description(kDummyType); std::string unified_plan_sdp_string = @@ -3763,8 +3691,8 @@ TEST_F(WebRtcSdpTest, UnifiedPlanDeserializeSessionDescriptionSpecialMsid) { // stream ids and multiple stream ids. TEST_F(WebRtcSdpTest, UnifiedPlanSerializeSessionDescriptionSpecialMsid) { // Only create a=msid lines for strictly Unified Plan stream ID support. - MakeUnifiedPlanDescriptionMultipleStreamIds( - webrtc::kMsidSignalingMediaSection | webrtc::kMsidSignalingSemantic); + MakeUnifiedPlanDescriptionMultipleStreamIds(kMsidSignalingMediaSection | + kMsidSignalingSemantic); TestSerialize(jdesc_); } @@ -3796,8 +3724,7 @@ TEST_F(WebRtcSdpTest, SerializeUnifiedPlanSessionDescriptionNoSsrcSignaling) { TEST_F(WebRtcSdpTest, EmptyDescriptionHasNoMsidSignaling) { JsepSessionDescription jsep_desc(kDummyType); ASSERT_TRUE(SdpDeserialize(kSdpSessionString, &jsep_desc)); - EXPECT_EQ(webrtc::kMsidSignalingSemantic, - jsep_desc.description()->msid_signaling()); + EXPECT_EQ(kMsidSignalingSemantic, jsep_desc.description()->msid_signaling()); } TEST_F(WebRtcSdpTest, DataChannelOnlyHasNoMsidSignaling) { @@ -3805,22 +3732,20 @@ TEST_F(WebRtcSdpTest, DataChannelOnlyHasNoMsidSignaling) { std::string sdp = kSdpSessionString; sdp += kSdpSctpDataChannelString; ASSERT_TRUE(SdpDeserialize(sdp, &jsep_desc)); - EXPECT_EQ(webrtc::kMsidSignalingSemantic, - jsep_desc.description()->msid_signaling()); + EXPECT_EQ(kMsidSignalingSemantic, jsep_desc.description()->msid_signaling()); } TEST_F(WebRtcSdpTest, PlanBHasSsrcAttributeMsidSignaling) { JsepSessionDescription jsep_desc(kDummyType); ASSERT_TRUE(SdpDeserialize(kPlanBSdpFullString, &jsep_desc)); - EXPECT_EQ( - webrtc::kMsidSignalingSsrcAttribute | webrtc::kMsidSignalingSemantic, - jsep_desc.description()->msid_signaling()); + EXPECT_EQ(kMsidSignalingSsrcAttribute | kMsidSignalingSemantic, + jsep_desc.description()->msid_signaling()); } TEST_F(WebRtcSdpTest, UnifiedPlanHasMediaSectionMsidSignaling) { JsepSessionDescription jsep_desc(kDummyType); ASSERT_TRUE(SdpDeserialize(kUnifiedPlanSdpFullString, &jsep_desc)); - EXPECT_EQ(webrtc::kMsidSignalingMediaSection | webrtc::kMsidSignalingSemantic, + EXPECT_EQ(kMsidSignalingMediaSection | kMsidSignalingSemantic, jsep_desc.description()->msid_signaling()); } @@ -3829,33 +3754,33 @@ const char kSsrcAttributeMsidLine[] = "a=ssrc:1 msid:local_stream_1 audio_track_id_1"; TEST_F(WebRtcSdpTest, SerializeOnlyMediaSectionMsid) { - jdesc_.description()->set_msid_signaling(webrtc::kMsidSignalingMediaSection); - std::string sdp = webrtc::SdpSerialize(jdesc_); + jdesc_.description()->set_msid_signaling(kMsidSignalingMediaSection); + std::string sdp = SdpSerialize(jdesc_); EXPECT_NE(std::string::npos, sdp.find(kMediaSectionMsidLine)); EXPECT_EQ(std::string::npos, sdp.find(kSsrcAttributeMsidLine)); } TEST_F(WebRtcSdpTest, SerializeOnlySsrcAttributeMsid) { - jdesc_.description()->set_msid_signaling(webrtc::kMsidSignalingSsrcAttribute); - std::string sdp = webrtc::SdpSerialize(jdesc_); + jdesc_.description()->set_msid_signaling(kMsidSignalingSsrcAttribute); + std::string sdp = SdpSerialize(jdesc_); EXPECT_EQ(std::string::npos, sdp.find(kMediaSectionMsidLine)); EXPECT_NE(std::string::npos, sdp.find(kSsrcAttributeMsidLine)); } TEST_F(WebRtcSdpTest, SerializeBothMediaSectionAndSsrcAttributeMsid) { - jdesc_.description()->set_msid_signaling(webrtc::kMsidSignalingMediaSection | - webrtc::kMsidSignalingSsrcAttribute); - std::string sdp = webrtc::SdpSerialize(jdesc_); + jdesc_.description()->set_msid_signaling(kMsidSignalingMediaSection | + kMsidSignalingSsrcAttribute); + std::string sdp = SdpSerialize(jdesc_); EXPECT_NE(std::string::npos, sdp.find(kMediaSectionMsidLine)); EXPECT_NE(std::string::npos, sdp.find(kSsrcAttributeMsidLine)); } TEST_F(WebRtcSdpTest, SerializeWithoutMsidSemantics) { - jdesc_.description()->set_msid_signaling(webrtc::kMsidSignalingNotUsed); - std::string sdp = webrtc::SdpSerialize(jdesc_); + jdesc_.description()->set_msid_signaling(kMsidSignalingNotUsed); + std::string sdp = SdpSerialize(jdesc_); EXPECT_EQ(std::string::npos, sdp.find("a=msid-semantic:")); } @@ -3909,9 +3834,9 @@ TEST_F(WebRtcSdpTest, BandwidthLimitOfNegativeOneIgnored) { JsepSessionDescription jdesc_output(kDummyType); EXPECT_TRUE(SdpDeserialize(kSdpWithBandwidthOfNegativeOne, &jdesc_output)); const VideoContentDescription* vcd = - webrtc::GetFirstVideoContentDescription(jdesc_output.description()); + GetFirstVideoContentDescription(jdesc_output.description()); ASSERT_TRUE(vcd); - EXPECT_EQ(webrtc::kAutoBandwidth, vcd->bandwidth()); + EXPECT_EQ(kAutoBandwidth, vcd->bandwidth()); } // Test that "ufrag"/"pwd" in the candidate line itself are ignored, and only @@ -3939,7 +3864,7 @@ TEST_F(WebRtcSdpTest, IceCredentialsInCandidateStringIgnored) { const IceCandidateCollection* candidates = jdesc_output.candidates(0); ASSERT_NE(nullptr, candidates); ASSERT_EQ(1U, candidates->count()); - webrtc::Candidate c = candidates->at(0)->candidate(); + Candidate c = candidates->at(0)->candidate(); EXPECT_EQ("ufrag_voice", c.username()); EXPECT_EQ("pwd_voice", c.password()); } @@ -3968,7 +3893,7 @@ TEST_F(WebRtcSdpTest, AttributeWithPartialMatchingNameIsIgnored) { const IceCandidateCollection* candidates = jdesc_output.candidates(0); ASSERT_NE(nullptr, candidates); ASSERT_EQ(1U, candidates->count()); - webrtc::Candidate c = candidates->at(0)->candidate(); + Candidate c = candidates->at(0)->candidate(); EXPECT_EQ("ufrag_voice", c.username()); EXPECT_EQ("pwd_voice", c.password()); } @@ -4420,7 +4345,7 @@ TEST_F(WebRtcSdpTest, SerializeAndDeserializeWithConnectionAddress) { JsepSessionDescription expected_jsep(kDummyType); MakeDescriptionWithoutCandidates(&expected_jsep); // Serialization. - std::string message = webrtc::SdpSerialize(expected_jsep); + std::string message = SdpSerialize(expected_jsep); // Deserialization. JsepSessionDescription jdesc(kDummyType); EXPECT_TRUE(SdpDeserialize(message, &jdesc)); @@ -4470,10 +4395,9 @@ TEST_F(WebRtcSdpTest, TestDeserializeSimulcastAttribute) { sdp += "a=simulcast:send 1,2;3 recv 4;5;6\r\n"; JsepSessionDescription output(kDummyType); SdpParseError error; - EXPECT_TRUE(webrtc::SdpDeserialize(sdp, &output, &error)); - const webrtc::ContentInfos& contents = output.description()->contents(); - const webrtc::MediaContentDescription* media = - contents.back().media_description(); + EXPECT_TRUE(SdpDeserialize(sdp, &output, &error)); + const ContentInfos& contents = output.description()->contents(); + const MediaContentDescription* media = contents.back().media_description(); EXPECT_TRUE(media->HasSimulcast()); EXPECT_EQ(2ul, media->simulcast_description().send_layers().size()); EXPECT_EQ(3ul, media->simulcast_description().receive_layers().size()); @@ -4491,10 +4415,9 @@ TEST_F(WebRtcSdpTest, TestDeserializeSimulcastAttributeRemovesUnknownRids) { sdp += "a=simulcast:send 1,2;3 recv 4;5,6\r\n"; JsepSessionDescription output(kDummyType); SdpParseError error; - EXPECT_TRUE(webrtc::SdpDeserialize(sdp, &output, &error)); - const webrtc::ContentInfos& contents = output.description()->contents(); - const webrtc::MediaContentDescription* media = - contents.back().media_description(); + EXPECT_TRUE(SdpDeserialize(sdp, &output, &error)); + const ContentInfos& contents = output.description()->contents(); + const MediaContentDescription* media = contents.back().media_description(); EXPECT_TRUE(media->HasSimulcast()); const SimulcastDescription& simulcast = media->simulcast_description(); EXPECT_EQ(2ul, simulcast.send_layers().size()); @@ -4529,10 +4452,9 @@ TEST_F(WebRtcSdpTest, sdp += "a=simulcast:send 1;2;3 recv 2;4\r\n"; JsepSessionDescription output(kDummyType); SdpParseError error; - EXPECT_TRUE(webrtc::SdpDeserialize(sdp, &output, &error)); - const webrtc::ContentInfos& contents = output.description()->contents(); - const webrtc::MediaContentDescription* media = - contents.back().media_description(); + EXPECT_TRUE(SdpDeserialize(sdp, &output, &error)); + const ContentInfos& contents = output.description()->contents(); + const MediaContentDescription* media = contents.back().media_description(); EXPECT_TRUE(media->HasSimulcast()); const SimulcastDescription& simulcast = media->simulcast_description(); EXPECT_EQ(2ul, simulcast.send_layers().size()); @@ -4555,10 +4477,9 @@ TEST_F(WebRtcSdpTest, TestDeserializeIgnoresEmptyRidLines) { sdp += "a=simulcast:send 1;2\r\n"; JsepSessionDescription output(kDummyType); SdpParseError error; - EXPECT_TRUE(webrtc::SdpDeserialize(sdp, &output, &error)); - const webrtc::ContentInfos& contents = output.description()->contents(); - const webrtc::MediaContentDescription* media = - contents.back().media_description(); + EXPECT_TRUE(SdpDeserialize(sdp, &output, &error)); + const ContentInfos& contents = output.description()->contents(); + const MediaContentDescription* media = contents.back().media_description(); EXPECT_TRUE(media->HasSimulcast()); const SimulcastDescription& simulcast = media->simulcast_description(); EXPECT_TRUE(simulcast.receive_layers().empty()); @@ -4581,10 +4502,9 @@ TEST_F(WebRtcSdpTest, TestDeserializeIgnoresMalformedRidLines) { sdp += "a=simulcast:send 1,2,3;4,5\r\n"; JsepSessionDescription output(kDummyType); SdpParseError error; - EXPECT_TRUE(webrtc::SdpDeserialize(sdp, &output, &error)); - const webrtc::ContentInfos& contents = output.description()->contents(); - const webrtc::MediaContentDescription* media = - contents.back().media_description(); + EXPECT_TRUE(SdpDeserialize(sdp, &output, &error)); + const ContentInfos& contents = output.description()->contents(); + const MediaContentDescription* media = contents.back().media_description(); EXPECT_TRUE(media->HasSimulcast()); const SimulcastDescription& simulcast = media->simulcast_description(); EXPECT_TRUE(simulcast.receive_layers().empty()); @@ -4604,10 +4524,9 @@ TEST_F(WebRtcSdpTest, TestDeserializeIgnoresInvalidPayloadTypesInRid) { sdp += "a=simulcast:send 1;2\r\n"; JsepSessionDescription output(kDummyType); SdpParseError error; - EXPECT_TRUE(webrtc::SdpDeserialize(sdp, &output, &error)); - const webrtc::ContentInfos& contents = output.description()->contents(); - const webrtc::MediaContentDescription* media = - contents.back().media_description(); + EXPECT_TRUE(SdpDeserialize(sdp, &output, &error)); + const ContentInfos& contents = output.description()->contents(); + const MediaContentDescription* media = contents.back().media_description(); EXPECT_TRUE(media->HasSimulcast()); const SimulcastDescription& simulcast = media->simulcast_description(); EXPECT_TRUE(simulcast.receive_layers().empty()); @@ -4635,10 +4554,9 @@ TEST_F(WebRtcSdpTest, TestDeserializeIgnoresDuplicateRidLines) { sdp += "a=simulcast:send 1,2;3 recv 4\r\n"; JsepSessionDescription output(kDummyType); SdpParseError error; - EXPECT_TRUE(webrtc::SdpDeserialize(sdp, &output, &error)); - const webrtc::ContentInfos& contents = output.description()->contents(); - const webrtc::MediaContentDescription* media = - contents.back().media_description(); + EXPECT_TRUE(SdpDeserialize(sdp, &output, &error)); + const ContentInfos& contents = output.description()->contents(); + const MediaContentDescription* media = contents.back().media_description(); EXPECT_TRUE(media->HasSimulcast()); const SimulcastDescription& simulcast = media->simulcast_description(); EXPECT_EQ(2ul, simulcast.send_layers().size()); @@ -4658,10 +4576,9 @@ TEST_F(WebRtcSdpTest, TestDeserializeRidSendDirection) { sdp += "a=simulcast:send 1;2\r\n"; JsepSessionDescription output(kDummyType); SdpParseError error; - EXPECT_TRUE(webrtc::SdpDeserialize(sdp, &output, &error)); - const webrtc::ContentInfos& contents = output.description()->contents(); - const webrtc::MediaContentDescription* media = - contents.back().media_description(); + EXPECT_TRUE(SdpDeserialize(sdp, &output, &error)); + const ContentInfos& contents = output.description()->contents(); + const MediaContentDescription* media = contents.back().media_description(); EXPECT_FALSE(media->HasSimulcast()); } @@ -4672,10 +4589,9 @@ TEST_F(WebRtcSdpTest, TestDeserializeRidRecvDirection) { sdp += "a=simulcast:recv 1;2\r\n"; JsepSessionDescription output(kDummyType); SdpParseError error; - EXPECT_TRUE(webrtc::SdpDeserialize(sdp, &output, &error)); - const webrtc::ContentInfos& contents = output.description()->contents(); - const webrtc::MediaContentDescription* media = - contents.back().media_description(); + EXPECT_TRUE(SdpDeserialize(sdp, &output, &error)); + const ContentInfos& contents = output.description()->contents(); + const MediaContentDescription* media = contents.back().media_description(); EXPECT_FALSE(media->HasSimulcast()); } @@ -4690,10 +4606,9 @@ TEST_F(WebRtcSdpTest, TestDeserializeIgnoresWrongRidDirectionLines) { sdp += "a=simulcast:send 1;5;3 recv 4;2;6\r\n"; JsepSessionDescription output(kDummyType); SdpParseError error; - EXPECT_TRUE(webrtc::SdpDeserialize(sdp, &output, &error)); - const webrtc::ContentInfos& contents = output.description()->contents(); - const webrtc::MediaContentDescription* media = - contents.back().media_description(); + EXPECT_TRUE(SdpDeserialize(sdp, &output, &error)); + const ContentInfos& contents = output.description()->contents(); + const MediaContentDescription* media = contents.back().media_description(); EXPECT_TRUE(media->HasSimulcast()); const SimulcastDescription& simulcast = media->simulcast_description(); EXPECT_EQ(2ul, simulcast.send_layers().size()); @@ -4749,35 +4664,34 @@ TEST_F(WebRtcSdpTest, ParseNoMid) { JsepSessionDescription output(kDummyType); SdpParseError error; - ASSERT_TRUE(webrtc::SdpDeserialize(sdp, &output, &error)); + ASSERT_TRUE(SdpDeserialize(sdp, &output, &error)); EXPECT_THAT(output.description()->contents(), - ElementsAre(Property("name", &webrtc::ContentInfo::mid, ""), - Property("name", &webrtc::ContentInfo::mid, ""))); + ElementsAre(Property("name", &ContentInfo::mid, ""), + Property("name", &ContentInfo::mid, ""))); } TEST_F(WebRtcSdpTest, SerializeWithDefaultSctpProtocol) { AddSctpDataChannel(false); // Don't use sctpmap JsepSessionDescription jsep_desc(kDummyType); MakeDescriptionWithoutCandidates(&jsep_desc); - std::string message = webrtc::SdpSerialize(jsep_desc); - EXPECT_NE(std::string::npos, message.find(webrtc::kMediaProtocolUdpDtlsSctp)); + std::string message = SdpSerialize(jsep_desc); + EXPECT_NE(std::string::npos, message.find(kMediaProtocolUdpDtlsSctp)); } TEST_F(WebRtcSdpTest, DeserializeWithAllSctpProtocols) { AddSctpDataChannel(false); - std::string protocols[] = {webrtc::kMediaProtocolDtlsSctp, - webrtc::kMediaProtocolUdpDtlsSctp, - webrtc::kMediaProtocolTcpDtlsSctp}; + std::string protocols[] = {kMediaProtocolDtlsSctp, kMediaProtocolUdpDtlsSctp, + kMediaProtocolTcpDtlsSctp}; for (const auto& protocol : protocols) { sctp_desc_->set_protocol(protocol); JsepSessionDescription jsep_desc(kDummyType); MakeDescriptionWithoutCandidates(&jsep_desc); - std::string message = webrtc::SdpSerialize(jsep_desc); + std::string message = SdpSerialize(jsep_desc); EXPECT_NE(std::string::npos, message.find(protocol)); JsepSessionDescription jsep_output(kDummyType); SdpParseError error; - EXPECT_TRUE(webrtc::SdpDeserialize(message, &jsep_output, &error)); + EXPECT_TRUE(SdpDeserialize(message, &jsep_output, &error)); } } @@ -4971,7 +4885,7 @@ TEST_F(WebRtcSdpTest, ParseIgnoreUnknownSsrcSpecificAttribute) { JsepSessionDescription output(kDummyType); SdpParseError error; - ASSERT_TRUE(webrtc::SdpDeserialize(sdp, &output, &error)); + ASSERT_TRUE(SdpDeserialize(sdp, &output, &error)); } TEST_F(WebRtcSdpTest, ParseSessionLevelExtmapAttributes) { @@ -5114,7 +5028,7 @@ TEST_F(WebRtcSdpTest, BackfillsDefaultFmtpValues) { const auto content = jdesc.description()->contents()[0]; const auto* description = content.media_description(); ASSERT_NE(description, nullptr); - const std::vector codecs = description->codecs(); + const std::vector codecs = description->codecs(); ASSERT_EQ(codecs.size(), 4u); std::string value; @@ -5143,10 +5057,10 @@ TEST_F(WebRtcSdpTest, BackfillsDefaultFmtpValues) { TEST_F(WebRtcSdpTest, ParsesKeyValueFmtpParameterSet) { std::string params = "key1=value1;key2=value2"; - webrtc::CodecParameterMap codec_params; + CodecParameterMap codec_params; SdpParseError error; - ASSERT_TRUE(webrtc::ParseFmtpParameterSet(params, codec_params, &error)); + ASSERT_TRUE(ParseFmtpParameterSet(params, codec_params, &error)); EXPECT_EQ(2U, codec_params.size()); EXPECT_EQ(codec_params["key1"], "value1"); EXPECT_EQ(codec_params["key2"], "value2"); @@ -5154,10 +5068,13 @@ TEST_F(WebRtcSdpTest, ParsesKeyValueFmtpParameterSet) { TEST_F(WebRtcSdpTest, ParsesNonKeyValueFmtpParameterSet) { std::string params = "not-in-key-value-format"; - webrtc::CodecParameterMap codec_params; + CodecParameterMap codec_params; SdpParseError error; - ASSERT_TRUE(webrtc::ParseFmtpParameterSet(params, codec_params, &error)); + ASSERT_TRUE(ParseFmtpParameterSet(params, codec_params, &error)); EXPECT_EQ(1U, codec_params.size()); EXPECT_EQ(codec_params[""], "not-in-key-value-format"); } + +} // namespace +} // namespace webrtc diff --git a/pc/webrtc_session_description_factory.cc b/pc/webrtc_session_description_factory.cc index 0ca42e4802..5c90efd063 100644 --- a/pc/webrtc_session_description_factory.cc +++ b/pc/webrtc_session_description_factory.cc @@ -43,10 +43,6 @@ #include "rtc_base/rtc_certificate_generator.h" #include "rtc_base/ssl_identity.h" #include "rtc_base/ssl_stream_adapter.h" -#include "rtc_base/unique_id_generator.h" - -using ::webrtc::MediaSessionOptions; -using webrtc::UniqueRandomIdGenerator; namespace webrtc { namespace { @@ -116,7 +112,7 @@ WebRtcSessionDescriptionFactory::WebRtcSessionDescriptionFactory( bool dtls_enabled, std::unique_ptr cert_generator, scoped_refptr certificate, - std::function&)> + std::function&)> on_certificate_ready, CodecLookupHelper* codec_lookup_helper, const FieldTrialsView& field_trials) diff --git a/pc/webrtc_session_description_factory.h b/pc/webrtc_session_description_factory.h index 3f3e6b62aa..c897a81ac2 100644 --- a/pc/webrtc_session_description_factory.h +++ b/pc/webrtc_session_description_factory.h @@ -53,7 +53,7 @@ class WebRtcSessionDescriptionFactory { bool dtls_enabled, std::unique_ptr cert_generator, scoped_refptr certificate, - std::function&)> + std::function&)> on_certificate_ready, CodecLookupHelper* codec_lookup_helper, const FieldTrialsView& field_trials); @@ -145,7 +145,7 @@ class WebRtcSessionDescriptionFactory { CertificateRequestState certificate_request_state_; std::queue> callbacks_; - std::function&)> + std::function&)> on_certificate_ready_; WeakPtrFactory weak_factory_{this}; From 4f1aa9f4a548f9bf5b6286a1dada5f07aa6d181e Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Thu, 8 May 2025 09:04:06 -0700 Subject: [PATCH 079/292] Roll chromium_revision 079e22aa7b..a543dd4ebe (1457513:1457639) Change log: https://chromium.googlesource.com/chromium/src/+log/079e22aa7b..a543dd4ebe Full diff: https://chromium.googlesource.com/chromium/src/+/079e22aa7b..a543dd4ebe Changed dependencies * fuchsia_version: version:27.20250501.3.1..version:28.20250508.3.1 * src/base: https://chromium.googlesource.com/chromium/src/base/+log/d978a5cb7a..d20957e106 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/90aa42cfb5..0ec92f3b3d * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/98759af37e..cb05a5b3b1 * src/third_party/androidx/cipd: a1d-sHWq2y1nFvYwrBfZHKtnwQXeYeyJXaK6H_dKEtQC..0kpSWsOkTy_kz_7BrmXaMj0RWlcTCIC9WOKWl3eSM4cC * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/116820911e..e21fb6701e * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/f4cde1b052..e2d7750acd * src/third_party/r8/cipd: Is-oa6b7OQ1cVHjvG42iz-Lha-F3MBxBBIyC_eTMTg4C..UDiEQW04Q7U0lSlAQAik0xbIV-AdjAnCHl7GcecroU4C * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/db4ac9ca2d..779ed26f14 DEPS diff: https://chromium.googlesource.com/chromium/src/+/079e22aa7b..a543dd4ebe/DEPS No update to Clang. BUG=None Change-Id: I6fc8c5c9b7f745ef96a0b3985377c0321b08cdd5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390480 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44551} --- DEPS | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/DEPS b/DEPS index 92c9e9452c..21a3cb614c 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '079e22aa7b3daddaf84220019be50468ae1d5d7e', + 'chromium_revision': 'a543dd4ebed1c7abc66739418691b8392fc9cca7', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -36,7 +36,7 @@ vars = { # By default, download the fuchsia sdk from the public sdk directory. 'fuchsia_sdk_cipd_prefix': 'fuchsia/sdk/core/', - 'fuchsia_version': 'version:27.20250501.3.1', + 'fuchsia_version': 'version:28.20250508.3.1', # By default, download the fuchsia images from the fuchsia GCS bucket. 'fuchsia_images_bucket': 'fuchsia', 'checkout_fuchsia': False, @@ -73,7 +73,7 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@d978a5cb7a04d54ae8ccfc2e15f77035cfe09e84', + 'https://chromium.googlesource.com/chromium/src/base@d20957e106d7e5274e7d635c42844ed07651822f', 'src/build': 'https://chromium.googlesource.com/chromium/src/build@d9c3040247e3b793c2a053d50fe2376e7596cd95', 'src/buildtools': @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@90aa42cfb54d0b710925645ecff1e4e03876fe2b', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@0ec92f3b3d6944e8c440720453be2d0ec4c79a3d', 'condition': 'checkout_ios', }, 'src/testing': 'https://chromium.googlesource.com/chromium/src/testing@3d17983c5bfeafd053c8310e709b3ec9c0b3a332', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@98759af37e9e2a993dbb134139ca0b059d27fa18', + 'https://chromium.googlesource.com/chromium/src/third_party@cb05a5b3b1f60b4d4e2d60a9f4f8c1503122df94', 'src/buildtools/linux64': { 'packages': [ @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@116820911e0a0775f3e1be92a6e0896d0c0d61d4', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@e21fb6701eba9e55f33b427d82010b8f8ce857d7', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'f4cde1b052fd591a5c50fe11dbcda61c51c4cccf', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'e2d7750acd15d57c6c88e3c8fb8c5a45f424ba96', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -632,7 +632,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/r8', - 'version': 'Is-oa6b7OQ1cVHjvG42iz-Lha-F3MBxBBIyC_eTMTg4C', + 'version': 'UDiEQW04Q7U0lSlAQAik0xbIV-AdjAnCHl7GcecroU4C', }, ], 'condition': 'checkout_android', @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@db4ac9ca2d95f8cc419fdb252e583dae6f8b950e', + 'https://chromium.googlesource.com/chromium/src/tools@779ed26f147aae6da64703617484084a7db117db', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'a1d-sHWq2y1nFvYwrBfZHKtnwQXeYeyJXaK6H_dKEtQC', + 'version': '0kpSWsOkTy_kz_7BrmXaMj0RWlcTCIC9WOKWl3eSM4cC', }, ], 'condition': 'checkout_android and non_git_source', From 361d125b97dd8e456f9f62e162323eb502ebaa46 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Thu, 8 May 2025 13:02:01 -0700 Subject: [PATCH 080/292] Roll chromium_revision a543dd4ebe..2cf9839a86 (1457639:1457781) Change log: https://chromium.googlesource.com/chromium/src/+log/a543dd4ebe..2cf9839a86 Full diff: https://chromium.googlesource.com/chromium/src/+/a543dd4ebe..2cf9839a86 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/d20957e106..cb47eca645 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/d9c3040247..0db6c04986 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/0ec92f3b3d..8659cb6bc9 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/3d17983c5b..a6361fe2c1 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/cb05a5b3b1..150313ee04 * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/864a235afc..154bb5860a * src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/16c0e42913..2fdf452fd1 * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/e21fb6701e..ede59d0345 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/779ed26f14..1fc313a7be DEPS diff: https://chromium.googlesource.com/chromium/src/+/a543dd4ebe..2cf9839a86/DEPS No update to Clang. BUG=None Change-Id: I324e0b461b26819d50c922a5b91b358ce13f5454 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390540 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44552} --- DEPS | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/DEPS b/DEPS index 21a3cb614c..9ea348872e 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': 'a543dd4ebed1c7abc66739418691b8392fc9cca7', + 'chromium_revision': '2cf9839a863db931568ffb9365ecc90d8d38238a', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@d20957e106d7e5274e7d635c42844ed07651822f', + 'https://chromium.googlesource.com/chromium/src/base@cb47eca645e45e8a686f7d4d5d9fa10d6b5342b6', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@d9c3040247e3b793c2a053d50fe2376e7596cd95', + 'https://chromium.googlesource.com/chromium/src/build@0db6c0498659d81821aabcf35dbb1504897f1264', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@077a66f30fcf281b066fafb6dfc60818c238efb6', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@0ec92f3b3d6944e8c440720453be2d0ec4c79a3d', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@8659cb6bc9e27f920c25a2a5e3942863196126c3', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@3d17983c5bfeafd053c8310e709b3ec9c0b3a332', + 'https://chromium.googlesource.com/chromium/src/testing@a6361fe2c1126b449e594025018228c1d5ae1f87', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@cb05a5b3b1f60b4d4e2d60a9f4f8c1503122df94', + 'https://chromium.googlesource.com/chromium/src/third_party@150313ee04814ad1551916bfb103a6f09e31cb3a', 'src/buildtools/linux64': { 'packages': [ @@ -323,7 +323,7 @@ deps = { 'src/third_party/clang-format/script': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@37f6e68a107df43b7d7e044fd36a13cbae3413f2', 'src/third_party/compiler-rt/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@16c0e42913a45357f0e557dcc0b7055129c78af6', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@2fdf452fd1b09f2e0bb9fdfe8a162c86cc70dbc2', 'src/third_party/libc++/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@a01c02c9d4acbdae3b7e8a2f3ee58579a9c29f96', 'src/third_party/libc++abi/src': @@ -480,7 +480,7 @@ deps = { }, 'src/third_party/boringssl/src': - 'https://boringssl.googlesource.com/boringssl.git@864a235afcf4d2575b1eab8de96fbf0d84f6cda9', + 'https://boringssl.googlesource.com/boringssl.git@154bb5860a47c160c3be09003908c64c1de11bff', 'src/third_party/breakpad/breakpad': 'https://chromium.googlesource.com/breakpad/breakpad.git@b319c12317824bfd0e92828db20bbfe0ec6e31f8', 'src/third_party/catapult': @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@e21fb6701eba9e55f33b427d82010b8f8ce857d7', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@ede59d0345c0bbeb2832c7f78e2137fb9b3c195e', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@779ed26f147aae6da64703617484084a7db117db', + 'https://chromium.googlesource.com/chromium/src/tools@1fc313a7be3e633f6f3ecebf45f04ec9514b2871', 'src/third_party/espresso': { 'packages': [ From 40d95f45aba66376be9ca33f7931cc0c9564fe79 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Thu, 8 May 2025 19:02:12 -0700 Subject: [PATCH 081/292] Roll chromium_revision 2cf9839a86..efebdb6499 (1457781:1457935) Change log: https://chromium.googlesource.com/chromium/src/+log/2cf9839a86..efebdb6499 Full diff: https://chromium.googlesource.com/chromium/src/+/2cf9839a86..efebdb6499 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/cb47eca645..07cde36d11 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/0db6c04986..009c5fbcdd * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/8659cb6bc9..8d03237122 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/a6361fe2c1..62c918c015 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/150313ee04..4694d84c9b * src/third_party/android_build_tools/lint/cipd: 2fKjMen4v_UU1CDYJqhwJCr6uh-L5w25n9lAKx6SlP0C..QltgzNfGb4l4ekv1_GjIWtuew1hrBof1WVy2B8zrkYwC * src/third_party/android_build_tools/manifest_merger/cipd: GnPGgBK8kiq32GuJB3iI20yJbFjgt2dqHsmwoPShJ-4C..hVPp69VTDru05sisN9HcrOf67Tk_Mhu6P6bsR24gOrEC * src/third_party/androidx/cipd: 0kpSWsOkTy_kz_7BrmXaMj0RWlcTCIC9WOKWl3eSM4cC..3f6bb_ds1Ab_INPp6lc8i778IRbWZSwVm2QBzfQFTMoC * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/154bb5860a..f371c85a8d * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/64c31ffa4d..74f545d009 * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/fab0a4296b..a7571b1596 * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/ede59d0345..b58c61966d * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/e2d7750acd..993d41d0c3 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/1fc313a7be..40cdeeca2e DEPS diff: https://chromium.googlesource.com/chromium/src/+/2cf9839a86..efebdb6499/DEPS No update to Clang. BUG=None Change-Id: I6301d0205fcecac94a837a7ed672a954f2a27023 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390542 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44553} --- DEPS | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/DEPS b/DEPS index 9ea348872e..6588ed5c3b 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '2cf9839a863db931568ffb9365ecc90d8d38238a', + 'chromium_revision': 'efebdb649977730f38af9c17a2a4e53e22f02ee9', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@cb47eca645e45e8a686f7d4d5d9fa10d6b5342b6', + 'https://chromium.googlesource.com/chromium/src/base@07cde36d11f84e1ca905028b7b96ab273fe76520', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@0db6c0498659d81821aabcf35dbb1504897f1264', + 'https://chromium.googlesource.com/chromium/src/build@009c5fbcdd84d02ac6099abfa9c59efd4aa6391b', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@077a66f30fcf281b066fafb6dfc60818c238efb6', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@8659cb6bc9e27f920c25a2a5e3942863196126c3', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@8d03237122a4d1490aee8079b839feb54ec8cda8', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@a6361fe2c1126b449e594025018228c1d5ae1f87', + 'https://chromium.googlesource.com/chromium/src/testing@62c918c0155b79bb0edc172b6e41396e143e22a8', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@150313ee04814ad1551916bfb103a6f09e31cb3a', + 'https://chromium.googlesource.com/chromium/src/third_party@4694d84c9bd8f526078f36d441053bfe6373214f', 'src/buildtools/linux64': { 'packages': [ @@ -449,7 +449,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_build_tools/lint', - 'version': '2fKjMen4v_UU1CDYJqhwJCr6uh-L5w25n9lAKx6SlP0C', + 'version': 'QltgzNfGb4l4ekv1_GjIWtuew1hrBof1WVy2B8zrkYwC', }, ], 'condition': 'checkout_android and non_git_source', @@ -480,11 +480,11 @@ deps = { }, 'src/third_party/boringssl/src': - 'https://boringssl.googlesource.com/boringssl.git@154bb5860a47c160c3be09003908c64c1de11bff', + 'https://boringssl.googlesource.com/boringssl.git@f371c85a8d58bf911cba6073bcca6d497af81dc9', 'src/third_party/breakpad/breakpad': 'https://chromium.googlesource.com/breakpad/breakpad.git@b319c12317824bfd0e92828db20bbfe0ec6e31f8', 'src/third_party/catapult': - 'https://chromium.googlesource.com/catapult.git@64c31ffa4d735add4a7e7520a52e0e3160216132', + 'https://chromium.googlesource.com/catapult.git@74f545d009ceef464f840938ae079ffbc112a102', 'src/third_party/ced/src': { 'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5', }, @@ -497,13 +497,13 @@ deps = { 'src/third_party/crc32c/src': 'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558', 'src/third_party/depot_tools': - 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@fab0a4296b8830c569a4dc82285bfb58a5c2fca8', + 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@a7571b1596305d6a6fe26c788f24e40c579fcda8', 'src/third_party/ffmpeg': 'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@dcdd0fa51b65a0b1688ff6b8f0cc81908f09ded2', 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@ede59d0345c0bbeb2832c7f78e2137fb9b3c195e', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@b58c61966d84dfefbf75f8fb0d2fadae948ac4db', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'e2d7750acd15d57c6c88e3c8fb8c5a45f424ba96', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '993d41d0c39335e35fe577808e8f704e58d9dbfb', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@1fc313a7be3e633f6f3ecebf45f04ec9514b2871', + 'https://chromium.googlesource.com/chromium/src/tools@40cdeeca2ea585f4a1057b559081baa457e10c99', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': '0kpSWsOkTy_kz_7BrmXaMj0RWlcTCIC9WOKWl3eSM4cC', + 'version': '3f6bb_ds1Ab_INPp6lc8i778IRbWZSwVm2QBzfQFTMoC', }, ], 'condition': 'checkout_android and non_git_source', @@ -706,7 +706,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_build_tools/manifest_merger', - 'version': 'GnPGgBK8kiq32GuJB3iI20yJbFjgt2dqHsmwoPShJ-4C', + 'version': 'hVPp69VTDru05sisN9HcrOf67Tk_Mhu6P6bsR24gOrEC', }, ], 'condition': 'checkout_android', From ff4222ed4c7a5082ded120a2be49922653aa7c3c Mon Sep 17 00:00:00 2001 From: webrtc-version-updater Date: Thu, 8 May 2025 21:02:35 -0700 Subject: [PATCH 082/292] Update WebRTC code version (2025-05-09T04:02:12). Bug: None Change-Id: I8042c7f5060a2ef2845b7f8609f53eeee951ce7d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390502 Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Cr-Commit-Position: refs/heads/main@{#44554} --- call/version.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/call/version.cc b/call/version.cc index 57f30561de..210d105d53 100644 --- a/call/version.cc +++ b/call/version.cc @@ -13,7 +13,7 @@ namespace webrtc { // The timestamp is always in UTC. -const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-08T04:08:56"; +const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-09T04:02:12"; void LoadWebRTCVersionInRegister() { // Using volatile to instruct the compiler to not optimize `p` away even From e7162754c0229d10c903525be3f900edb184dfb7 Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Wed, 7 May 2025 08:42:46 -0700 Subject: [PATCH 083/292] document footgun behavior of codec munging and setCodecPreferences which drops the munged codec BUG=webrtc:396344370 Change-Id: I4980c6980ad7df29d4d1451777a61f90dac73f74 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389440 Reviewed-by: Evan Shrubsole Reviewed-by: Harald Alvestrand Commit-Queue: Philipp Hancke Cr-Commit-Position: refs/heads/main@{#44555} --- pc/peer_connection_integrationtest.cc | 98 ++++++++++++++++++++++++++- 1 file changed, 96 insertions(+), 2 deletions(-) diff --git a/pc/peer_connection_integrationtest.cc b/pc/peer_connection_integrationtest.cc index 95bd835301..a3e975086b 100644 --- a/pc/peer_connection_integrationtest.cc +++ b/pc/peer_connection_integrationtest.cc @@ -18,6 +18,7 @@ #include #include +#include #include #include #include @@ -4793,6 +4794,99 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan, EXPECT_TRUE(caller()->SetLocalDescriptionAndSendSdpMessage(std::move(offer))); } +TEST_F(PeerConnectionIntegrationTestUnifiedPlan, + MungeOfferCodecAndReOfferWorksWithSetCodecPreferencesIsAFootgun) { + ASSERT_TRUE(CreatePeerConnectionWrappers()); + ConnectFakeSignaling(); + caller()->AddVideoTrack(); + auto munger = [](std::unique_ptr& sdp) { + auto video = GetFirstVideoContentDescription(sdp->description()); + auto codecs = video->codecs(); + std::optional replacement_codec; + for (auto&& codec : codecs) { + if (codec.name == "AV1") { + replacement_codec = codec; + break; + } + } + if (replacement_codec) { + for (auto&& codec : codecs) { + if (codec.name == "VP8") { + RTC_LOG(LS_INFO) << "Remapping VP8 codec " << codec << " to AV1"; + codec.name = replacement_codec->name; + codec.params = replacement_codec->params; + break; + } + } + video->set_codecs(codecs); + } else { + RTC_LOG(LS_INFO) << "Skipping munge, no AV1 codec found"; + } + }; + caller()->SetGeneratedSdpMunger(munger); + caller()->CreateAndSetAndSignalOffer(); + ASSERT_THAT( + WaitUntil([&] { return SignalingStateStable(); }, ::testing::IsTrue()), + IsRtcOk()); + caller()->SetGeneratedSdpMunger(nullptr); + + // Note currently negotiated codecs and count VP9 and AV1. + auto codecs_after_munge = caller() + ->pc() + ->local_description() + ->description() + ->contents()[0] + .media_description() + ->codecs(); + size_t av1_munge = 0; + size_t vp8_munge = 0; + for (const auto& codec : codecs_after_munge) { + if (codec.name == "AV1") + av1_munge++; + else if (codec.name == "VP8") + vp8_munge++; + } + // We should have replaced VP8 with AV1. + EXPECT_EQ(av1_munge, 2u); + EXPECT_EQ(vp8_munge, 0u); + + // Call setCodecPreferences. + std::vector codecs = + caller() + ->pc_factory() + ->GetRtpReceiverCapabilities(webrtc::MediaType::VIDEO) + .codecs; + auto transceivers = caller()->pc()->GetTransceivers(); + ASSERT_EQ(transceivers.size(), 1u); + transceivers[0]->SetCodecPreferences(codecs); + + auto offer = caller()->CreateOfferAndWait(); + ASSERT_NE(offer, nullptr); + // The offer should be acceptable. + EXPECT_TRUE(caller()->SetLocalDescriptionAndSendSdpMessage(std::move(offer))); + + auto codecs_after_scp = caller() + ->pc() + ->local_description() + ->description() + ->contents()[0] + .media_description() + ->codecs(); + size_t av1_scp = 0; + size_t vp8_scp = 0; + for (const auto& codec : codecs_after_scp) { + if (codec.name == "AV1") + av1_scp++; + else if (codec.name == "VP8") + vp8_scp++; + } + // The SDP munging modification was reverted by sCP. + // This is a footgun but please do not mix such munging with + // setCodecPreferences, munging is on the way out. + EXPECT_EQ(av1_scp, 1u); + EXPECT_EQ(vp8_scp, 1u); +} + TEST_F(PeerConnectionIntegrationTestUnifiedPlan, SensibleRtxWithDuplicateCodecs) { ASSERT_TRUE(CreatePeerConnectionWrappers()); @@ -4837,7 +4931,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan, // associated RTX codec. std::unique_ptr answer = caller()->CreateAnswerForTest(); - ASSERT_THAT(answer, NotNull()); + ASSERT_NE(answer, nullptr); RTC_LOG(LS_ERROR) << "Answer is " << *answer; ASSERT_THAT(answer->description()->contents().size(), Eq(1)); auto codecs = @@ -4896,7 +4990,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan, caller()->AddVideoTrack(); auto offer2 = caller()->CreateOfferAndWait(); // Observe that packetization is raw on BOTH media sections. - ASSERT_THAT(offer2, NotNull()); + ASSERT_NE(offer2, nullptr); EXPECT_EQ(offer2->description()->contents().size(), 2U); for (const auto& content : offer2->description()->contents()) { for (const auto& codec : content.media_description()->codecs()) { From 9f3b8b648208df0d61256cdd6c328c32174fd673 Mon Sep 17 00:00:00 2001 From: Junji Watanabe Date: Fri, 9 May 2025 11:29:10 +0900 Subject: [PATCH 084/292] infra: Switch Reclient to Siso's builtin RBE client This CL stops using Reclient and starts using Siso's builtin RBE client to be aligned with Chromium's remote execution system. Bug: chromium:412968361 Change-Id: I8917e2e3147351f5e1cb9919544040b1a8b13a3a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/388480 Reviewed-by: Mirko Bonadei Auto-Submit: Junji Watanabe Reviewed-by: Christoffer Dewerin Commit-Queue: Mirko Bonadei Cr-Commit-Position: refs/heads/main@{#44556} --- tools_webrtc/mb/mb_config.pyl | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/tools_webrtc/mb/mb_config.pyl b/tools_webrtc/mb/mb_config.pyl index 3d1fd7ccb2..1fdaaf5c84 100644 --- a/tools_webrtc/mb/mb_config.pyl +++ b/tools_webrtc/mb/mb_config.pyl @@ -524,7 +524,7 @@ 'debug_bot': { 'mixins': [ 'debug', - 'reclient', + 'remoteexec', 'strict_field_trials', 'disallow_deprecated_namespaces' ], @@ -533,7 +533,7 @@ 'mixins': [ 'debug', 'minimal_symbols', - 'reclient', + 'remoteexec', 'strict_field_trials', 'disallow_deprecated_namespaces', ], @@ -620,10 +620,7 @@ # 100% release build without DCHECKs while 'release_bot' is a partial # release configs since `dcheck_always_on` is set to true. 'pure_release_bot': { - 'mixins': ['release', 'reclient', 'dcheck_off'], - }, - 'reclient': { - 'gn_args': 'use_remoteexec=true', + 'mixins': ['release', 'remoteexec', 'dcheck_off'], }, 'release': { 'gn_args': 'is_debug=false', @@ -636,6 +633,9 @@ 'disallow_deprecated_namespaces', ], }, + 'remoteexec': { + 'gn_args': 'use_remoteexec=true use_reclient=false use_siso=true', + }, 'rtc_objc_test_prefix': { 'gn_args': 'rtc_objc_prefix="RTC_TESTING"', }, From cc24329a5ad386e2736a476f3fc72995cdb2bae3 Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Wed, 30 Apr 2025 10:15:39 -0700 Subject: [PATCH 085/292] Remove platform-specific variable shadowing MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Windows: * main_wnd.cc * device_info_ds.cc * rtc_base/network.cc * rtc_base/physical_socket_server.cc iOS: * lots of AppRTCDemo fixes * SRWebSocket.m (somewhat in line with upstream) * audio_device_ios.mm * RTCNetworkMonitor.mm * RTCPeerConnectionFactory_xctest.m Android: * android_network_monitor.c Linux: * fuzzers/rtp_packet_fuzzer.cc BUG=webrtc:42223409 No-IWYU: refactoring of platform specific code Change-Id: I5fd0138f2c727fb268a989b7a42f4b388c883835 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/387924 Reviewed-by: Harald Alvestrand Commit-Queue: Philipp Hancke Reviewed-by: Kári Helgason Cr-Commit-Position: refs/heads/main@{#44557} --- examples/objc/AppRTCMobile/ARDAppClient.m | 22 ++++---- .../objc/AppRTCMobile/ARDAppEngineClient.m | 54 +++++++++---------- .../objc/AppRTCMobile/ARDWebSocketChannel.m | 12 ++--- .../objc/AppRTCMobile/common/ARDUtilities.m | 12 ++--- .../AppRTCMobile/mac/APPRTCViewController.m | 4 +- .../third_party/SocketRocket/SRWebSocket.m | 54 +++++++++---------- examples/peerconnection/client/main_wnd.cc | 12 ++--- .../video_capture/windows/device_info_ds.cc | 11 ++-- rtc_base/network.cc | 24 ++++----- rtc_base/physical_socket_server.cc | 2 +- .../src/jni/android_network_monitor.cc | 6 +-- .../components/network/RTCNetworkMonitor.mm | 7 +-- sdk/objc/native/src/audio/audio_device_ios.mm | 4 +- .../RTCPeerConnectionFactory_xctest.m | 43 ++++++++------- test/fuzzers/rtp_packet_fuzzer.cc | 12 +++-- 15 files changed, 142 insertions(+), 137 deletions(-) diff --git a/examples/objc/AppRTCMobile/ARDAppClient.m b/examples/objc/AppRTCMobile/ARDAppClient.m index 72961abad9..614037e6f7 100644 --- a/examples/objc/AppRTCMobile/ARDAppClient.m +++ b/examples/objc/AppRTCMobile/ARDAppClient.m @@ -200,9 +200,9 @@ - (void)setShouldGetStats:(BOOL)shouldGetStats { statisticsWithCompletionHandler:^( RTC_OBJC_TYPE(RTCStatisticsReport) * stats) { dispatch_async(dispatch_get_main_queue(), ^{ - ARDAppClient *strongSelf = weakSelf; - [strongSelf.delegate appClient:strongSelf - didGetStats:stats]; + ARDAppClient *strongerSelf = weakSelf; + [strongerSelf.delegate appClient:strongerSelf + didGetStats:stats]; }); }]; }]; @@ -497,10 +497,10 @@ - (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection __weak ARDAppClient *weakSelf = self; [self.peerConnection setLocalDescription:sdp - completionHandler:^(NSError *error) { + completionHandler:^(NSError *sldError) { ARDAppClient *strongSelf = weakSelf; [strongSelf peerConnection:strongSelf.peerConnection - didSetSessionDescriptionWithError:error]; + didSetSessionDescriptionWithError:sldError]; }]; ARDSessionDescriptionMessage *message = [[ARDSessionDescriptionMessage alloc] initWithDescription:sdp]; @@ -534,11 +534,11 @@ - (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection [self.peerConnection answerForConstraints:constraints completionHandler:^(RTC_OBJC_TYPE(RTCSessionDescription) * sdp, - NSError * error) { + NSError * answerError) { ARDAppClient *strongSelf = weakSelf; [strongSelf peerConnection:strongSelf.peerConnection didCreateSessionDescription:sdp - error:error]; + error:answerError]; }]; } }); @@ -768,9 +768,9 @@ - (void)createMediaSenders { [self defaultMediaAudioConstraints]; RTC_OBJC_TYPE(RTCAudioSource) *source = [_factory audioSourceWithConstraints:constraints]; - RTC_OBJC_TYPE(RTCAudioTrack) *track = + RTC_OBJC_TYPE(RTCAudioTrack) *audioTrack = [_factory audioTrackWithSource:source trackId:kARDAudioTrackId]; - [_peerConnection addTrack:track streamIds:@[ kARDMediaStreamId ]]; + [_peerConnection addTrack:audioTrack streamIds:@[ kARDMediaStreamId ]]; _localVideoTrack = [self createLocalVideoTrack]; if (_localVideoTrack) { [_peerConnection addTrack:_localVideoTrack @@ -780,9 +780,9 @@ - (void)createMediaSenders { // transceiver already has an RTC_OBJC_TYPE(RTCRtpReceiver) with a track. // The track will automatically get unmuted and produce frames once RTP is // received. - RTC_OBJC_TYPE(RTCVideoTrack) *track = (RTC_OBJC_TYPE(RTCVideoTrack) *)( + RTC_OBJC_TYPE(RTCVideoTrack) *videoTrack = (RTC_OBJC_TYPE(RTCVideoTrack) *)( [self videoTransceiver].receiver.track); - [_delegate appClient:self didReceiveRemoteVideoTrack:track]; + [_delegate appClient:self didReceiveRemoteVideoTrack:videoTrack]; } } diff --git a/examples/objc/AppRTCMobile/ARDAppEngineClient.m b/examples/objc/AppRTCMobile/ARDAppEngineClient.m index b7f8dcf784..e65b39a718 100644 --- a/examples/objc/AppRTCMobile/ARDAppEngineClient.m +++ b/examples/objc/AppRTCMobile/ARDAppEngineClient.m @@ -66,8 +66,8 @@ - (void)joinRoomWithRoomId:(NSString *)roomId [ARDJoinResponse responseFromJSONData:data]; if (!joinResponse) { if (completionHandler) { - NSError *error = [[self class] badResponseError]; - completionHandler(nil, error); + NSError *err = [[self class] badResponseError]; + completionHandler(nil, err); } return; } @@ -86,37 +86,37 @@ - (void)sendMessage:(ARDSignalingMessage *)message NSParameterAssert(roomId.length); NSParameterAssert(clientId.length); - NSData *data = [message JSONData]; + NSData *messageData = [message JSONData]; NSString *urlString = [NSString stringWithFormat:kARDRoomServerMessageFormat, roomId, clientId]; NSURL *url = [NSURL URLWithString:urlString]; RTCLog(@"C->RS POST: %@", message); NSMutableURLRequest *request = [NSMutableURLRequest requestWithURL:url]; request.HTTPMethod = @"POST"; - request.HTTPBody = data; - [NSURLConnection - sendAsyncRequest:request - completionHandler:^( - NSURLResponse *response __unused, NSData *data, NSError *error) { - if (error) { - if (completionHandler) { - completionHandler(nil, error); - } - return; - } - ARDMessageResponse *messageResponse = - [ARDMessageResponse responseFromJSONData:data]; - if (!messageResponse) { - if (completionHandler) { - NSError *error = [[self class] badResponseError]; - completionHandler(nil, error); - } - return; - } - if (completionHandler) { - completionHandler(messageResponse, nil); - } - }]; + request.HTTPBody = messageData; + [NSURLConnection sendAsyncRequest:request + completionHandler:^(NSURLResponse *response __unused, + NSData *data, + NSError *responseError) { + if (responseError) { + if (completionHandler) { + completionHandler(nil, responseError); + } + return; + } + ARDMessageResponse *messageResponse = + [ARDMessageResponse responseFromJSONData:data]; + if (!messageResponse) { + if (completionHandler) { + NSError *err = [[self class] badResponseError]; + completionHandler(nil, err); + } + return; + } + if (completionHandler) { + completionHandler(messageResponse, nil); + } + }]; } - (void)leaveRoomWithRoomId:(NSString *)roomId diff --git a/examples/objc/AppRTCMobile/ARDWebSocketChannel.m b/examples/objc/AppRTCMobile/ARDWebSocketChannel.m index 37260d63ba..e21b30deed 100644 --- a/examples/objc/AppRTCMobile/ARDWebSocketChannel.m +++ b/examples/objc/AppRTCMobile/ARDWebSocketChannel.m @@ -80,14 +80,10 @@ - (void)sendMessage:(ARDSignalingMessage *)message { if (_state == kARDSignalingChannelStateRegistered) { NSString *payload = [[NSString alloc] initWithData:data encoding:NSUTF8StringEncoding]; - NSDictionary *message = @{ - @"cmd" : @"send", - @"msg" : payload, - }; - NSData *messageJSONObject = - [NSJSONSerialization dataWithJSONObject:message - options:NSJSONWritingPrettyPrinted - error:nil]; + NSData *messageJSONObject = [NSJSONSerialization + dataWithJSONObject:@{@"cmd" : @"send", @"msg" : payload} + options:NSJSONWritingPrettyPrinted + error:nil]; NSString *messageString = [[NSString alloc] initWithData:messageJSONObject encoding:NSUTF8StringEncoding]; diff --git a/examples/objc/AppRTCMobile/common/ARDUtilities.m b/examples/objc/AppRTCMobile/common/ARDUtilities.m index 9ac5508a51..539445ead7 100644 --- a/examples/objc/AppRTCMobile/common/ARDUtilities.m +++ b/examples/objc/AppRTCMobile/common/ARDUtilities.m @@ -70,28 +70,28 @@ + (void)sendAsyncPostToURL:(NSURL *)url [[self class] sendAsyncRequest:request completionHandler:^( - NSURLResponse *response, NSData *data, NSError *error) { + NSURLResponse *response, NSData *responseData, NSError *error) { if (error) { RTCLogError(@"Error posting data: %@", error.localizedDescription); if (completionHandler) { - completionHandler(NO, data); + completionHandler(NO, responseData); } return; } NSHTTPURLResponse *httpResponse = (NSHTTPURLResponse *)response; if (httpResponse.statusCode != 200) { - NSString *serverResponse = data.length > 0 ? - [[NSString alloc] initWithData:data + NSString *serverResponse = responseData.length > 0 ? + [[NSString alloc] initWithData:responseData encoding:NSUTF8StringEncoding] : nil; RTCLogError(@"Received bad response: %@", serverResponse); if (completionHandler) { - completionHandler(NO, data); + completionHandler(NO, responseData); } return; } if (completionHandler) { - completionHandler(YES, data); + completionHandler(YES, responseData); } }]; } diff --git a/examples/objc/AppRTCMobile/mac/APPRTCViewController.m b/examples/objc/AppRTCMobile/mac/APPRTCViewController.m index cf87b49ac7..f8bb55e6a2 100644 --- a/examples/objc/AppRTCMobile/mac/APPRTCViewController.m +++ b/examples/objc/AppRTCMobile/mac/APPRTCViewController.m @@ -149,11 +149,11 @@ - (void)updateConstraints { #pragma mark - Constraints helper -+ (void)addConstraints:(NSArray*)constraints ++ (void)addConstraints:(NSArray*)constraintFormats toView:(NSView*)view viewsDictionary:(NSDictionary*)viewsDictionary metrics:(NSDictionary*)metrics { - for (NSString* constraintFormat in constraints) { + for (NSString* constraintFormat in constraintFormats) { NSArray* constraints = [NSLayoutConstraint constraintsWithVisualFormat:constraintFormat options:0 diff --git a/examples/objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.m b/examples/objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.m index 68f3e93f7f..3888ecfd27 100644 --- a/examples/objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.m +++ b/examples/objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.m @@ -483,15 +483,15 @@ - (void)_readHTTPHeader; _receivedHTTPHeaders = CFHTTPMessageCreateEmpty(NULL, NO); } - [self _readUntilHeaderCompleteWithCallback:^(SRWebSocket *self, NSData *data) { - CFHTTPMessageAppendBytes(self->_receivedHTTPHeaders, (const UInt8 *)data.bytes, data.length); + [self _readUntilHeaderCompleteWithCallback:^(SRWebSocket *socket, NSData *data) { + CFHTTPMessageAppendBytes(socket->_receivedHTTPHeaders, (const UInt8 *)data.bytes, data.length); - if (CFHTTPMessageIsHeaderComplete(self->_receivedHTTPHeaders)) { + if (CFHTTPMessageIsHeaderComplete(socket->_receivedHTTPHeaders)) { SRFastLog(@"Finished reading headers %@", - CFBridgingRelease(CFHTTPMessageCopyAllHeaderFields(self->_receivedHTTPHeaders))); - [self _HTTPHeadersDidFinish]; + CFBridgingRelease(CFHTTPMessageCopyAllHeaderFields(socket->_receivedHTTPHeaders))); + [socket _HTTPHeadersDidFinish]; } else { - [self _readHTTPHeader]; + [socket _readHTTPHeader]; } }]; } @@ -929,15 +929,15 @@ - (void)_handleFrameHeader:(frame_header)frame_header curData:(NSData *)curData; } } else { assert(frame_header.payload_length <= SIZE_T_MAX); - [self _addConsumerWithDataLength:(size_t)frame_header.payload_length callback:^(SRWebSocket *self, NSData *newData) { + [self _addConsumerWithDataLength:(size_t)frame_header.payload_length callback:^(SRWebSocket *socket, NSData *newData) { if (isControlFrame) { - [self _handleFrameWithData:newData opCode:frame_header.opcode]; + [socket _handleFrameWithData:newData opCode:frame_header.opcode]; } else { if (frame_header.fin) { - [self _handleFrameWithData:self->_currentFrameData opCode:frame_header.opcode]; + [socket _handleFrameWithData:socket->_currentFrameData opCode:frame_header.opcode]; } else { // TODO add assert that opcode is not a control; - [self _readFrameContinue]; + [socket _readFrameContinue]; } } @@ -978,14 +978,14 @@ - (void)_readFrameContinue; { assert((_currentFrameCount == 0 && _currentFrameOpcode == 0) || (_currentFrameCount > 0 && _currentFrameOpcode > 0)); - [self _addConsumerWithDataLength:2 callback:^(SRWebSocket *self, NSData *data) { + [self _addConsumerWithDataLength:2 callback:^(SRWebSocket *sself, NSData *data) { __block frame_header header = {0}; const uint8_t *headerBuffer = data.bytes; assert(data.length >= 2); if (headerBuffer[0] & SRRsvMask) { - [self _closeWithProtocolError:@"Server used RSV bits"]; + [sself _closeWithProtocolError:@"Server used RSV bits"]; return; } @@ -993,17 +993,17 @@ - (void)_readFrameContinue; BOOL isControlFrame = (receivedOpcode == SROpCodePing || receivedOpcode == SROpCodePong || receivedOpcode == SROpCodeConnectionClose); - if (!isControlFrame && receivedOpcode != 0 && self->_currentFrameCount > 0) { - [self _closeWithProtocolError:@"all data frames after the initial data frame must have opcode 0"]; + if (!isControlFrame && receivedOpcode != 0 && sself->_currentFrameCount > 0) { + [sself _closeWithProtocolError:@"all data frames after the initial data frame must have opcode 0"]; return; } - if (receivedOpcode == 0 && self->_currentFrameCount == 0) { - [self _closeWithProtocolError:@"cannot continue a message"]; + if (receivedOpcode == 0 && sself->_currentFrameCount == 0) { + [sself _closeWithProtocolError:@"cannot continue a message"]; return; } - header.opcode = receivedOpcode == 0 ? self->_currentFrameOpcode : receivedOpcode; + header.opcode = receivedOpcode == 0 ? sself->_currentFrameOpcode : receivedOpcode; header.fin = !!(SRFinMask & headerBuffer[0]); @@ -1014,10 +1014,10 @@ - (void)_readFrameContinue; headerBuffer = NULL; if (header.masked) { - [self _closeWithProtocolError:@"Client must receive unmasked data"]; + [sself _closeWithProtocolError:@"Client must receive unmasked data"]; } - size_t extra_bytes_needed = header.masked ? sizeof(self->_currentReadMaskKey) : 0; + size_t extra_bytes_needed = header.masked ? sizeof(sself->_currentReadMaskKey) : 0; if (header.payload_length == 126) { extra_bytes_needed += sizeof(uint16_t); @@ -1026,11 +1026,11 @@ - (void)_readFrameContinue; } if (extra_bytes_needed == 0) { - [self _handleFrameHeader:header curData:self->_currentFrameData]; + [sself _handleFrameHeader:header curData:sself->_currentFrameData]; } else { - [self _addConsumerWithDataLength:extra_bytes_needed callback:^(SRWebSocket *self, NSData *data) { - size_t mapped_size = data.length; - const void *mapped_buffer = data.bytes; + [sself _addConsumerWithDataLength:extra_bytes_needed callback:^(SRWebSocket *eself, NSData *edata) { + size_t mapped_size = edata.length; + const void *mapped_buffer = edata.bytes; size_t offset = 0; if (header.payload_length == 126) { @@ -1048,13 +1048,13 @@ - (void)_readFrameContinue; if (header.masked) { - assert(mapped_size >= sizeof(self->_currentReadMaskOffset) + offset); - memcpy(self->_currentReadMaskKey, + assert(mapped_size >= sizeof(eself->_currentReadMaskOffset) + offset); + memcpy(eself->_currentReadMaskKey, ((uint8_t *)mapped_buffer) + offset, - sizeof(self->_currentReadMaskKey)); + sizeof(eself->_currentReadMaskKey)); } - [self _handleFrameHeader:header curData:self->_currentFrameData]; + [eself _handleFrameHeader:header curData:eself->_currentFrameData]; } readToCurrentFrame:NO unmaskBytes:NO]; } } readToCurrentFrame:NO unmaskBytes:NO]; diff --git a/examples/peerconnection/client/main_wnd.cc b/examples/peerconnection/client/main_wnd.cc index 59854875fb..6b834bc466 100644 --- a/examples/peerconnection/client/main_wnd.cc +++ b/examples/peerconnection/client/main_wnd.cc @@ -279,15 +279,15 @@ void MainWnd::OnPaint() { &bmi, DIB_RGB_COLORS, SRCCOPY); if ((rc.right - rc.left) > 200 && (rc.bottom - rc.top) > 200) { - const BITMAPINFO& bmi = local_renderer->bmi(); + const BITMAPINFO& local_bmi = local_renderer->bmi(); image = local_renderer->image(); - int thumb_width = bmi.bmiHeader.biWidth / 4; - int thumb_height = abs(bmi.bmiHeader.biHeight) / 4; + int thumb_width = local_bmi.bmiHeader.biWidth / 4; + int thumb_height = abs(local_bmi.bmiHeader.biHeight) / 4; StretchDIBits(dc_mem, logical_area.x - thumb_width - 10, logical_area.y - thumb_height - 10, thumb_width, - thumb_height, 0, 0, bmi.bmiHeader.biWidth, - -bmi.bmiHeader.biHeight, image, &bmi, DIB_RGB_COLORS, - SRCCOPY); + thumb_height, 0, 0, local_bmi.bmiHeader.biWidth, + -local_bmi.bmiHeader.biHeight, image, &local_bmi, + DIB_RGB_COLORS, SRCCOPY); } BitBlt(ps.hdc, 0, 0, logical_area.x, logical_area.y, dc_mem, 0, 0, diff --git a/modules/video_capture/windows/device_info_ds.cc b/modules/video_capture/windows/device_info_ds.cc index c4bef2d6ca..9a2c322137 100644 --- a/modules/video_capture/windows/device_info_ds.cc +++ b/modules/video_capture/windows/device_info_ds.cc @@ -442,17 +442,18 @@ int32_t DeviceInfoDS::CreateCapabilityMap(const char* deviceUniqueIdUTF8) LONGLONG* frameDurationList = NULL; LONGLONG maxFPS = 0; long listSize = 0; - SIZE size; - size.cx = capability.width; - size.cy = capability.height; + SIZE requested_size; + requested_size.cx = capability.width; + requested_size.cy = capability.height; // GetMaxAvailableFrameRate doesn't return max frame rate always // eg: Logitech Notebook. This may be due to a bug in that API // because GetFrameRateList array is reversed in the above camera. So // a util method written. Can't assume the first value will return // the max fps. - hrVC = videoControlConfig->GetFrameRateList( - outputCapturePin, tmp, size, &listSize, &frameDurationList); + hrVC = videoControlConfig->GetFrameRateList(outputCapturePin, tmp, + requested_size, &listSize, + &frameDurationList); if (hrVC == S_OK) { maxFPS = GetMaxOfFrameArray(frameDurationList, listSize); diff --git a/rtc_base/network.cc b/rtc_base/network.cc index 19741366f8..cc796b7558 100644 --- a/rtc_base/network.cc +++ b/rtc_base/network.cc @@ -907,18 +907,18 @@ bool BasicNetworkManager::CreateNetworks( adapter_type = ADAPTER_TYPE_VPN; } - auto network = CreateNetwork(name, description, prefix, prefix_length, - adapter_type); - network->set_underlying_type_for_vpn(underlying_type_for_vpn); - network->set_default_local_address_provider(this); - network->set_mdns_responder_provider(this); - network->set_scope_id(scope_id); - network->AddIP(ip); - bool ignored = IsIgnoredNetwork(*network); - network->set_ignored(ignored); - if (include_ignored || !network->ignored()) { - current_networks[key] = network.get(); - networks->push_back(std::move(network)); + auto new_network = CreateNetwork(name, description, prefix, + prefix_length, adapter_type); + new_network->set_underlying_type_for_vpn(underlying_type_for_vpn); + new_network->set_default_local_address_provider(this); + new_network->set_mdns_responder_provider(this); + new_network->set_scope_id(scope_id); + new_network->AddIP(ip); + bool ignored = IsIgnoredNetwork(*new_network); + new_network->set_ignored(ignored); + if (include_ignored || !new_network->ignored()) { + current_networks[key] = new_network.get(); + networks->push_back(std::move(new_network)); } } else { (*existing_network).second->AddIP(ip); diff --git a/rtc_base/physical_socket_server.cc b/rtc_base/physical_socket_server.cc index e69311b790..4769f41bc0 100644 --- a/rtc_base/physical_socket_server.cc +++ b/rtc_base/physical_socket_server.cc @@ -1901,7 +1901,7 @@ bool PhysicalSocketServer::Wait(webrtc::TimeDelta max_wait_duration, bool process_io) { // We don't support reentrant waiting. RTC_DCHECK(!waiting_); - ScopedSetTrue s(&waiting_); + ScopedSetTrue set(&waiting_); int cmsWait = ToCmsWait(max_wait_duration); int64_t cmsTotal = cmsWait; diff --git a/sdk/android/src/jni/android_network_monitor.cc b/sdk/android/src/jni/android_network_monitor.cc index f2f14b3b18..f8dc19eaec 100644 --- a/sdk/android/src/jni/android_network_monitor.cc +++ b/sdk/android/src/jni/android_network_monitor.cc @@ -478,11 +478,11 @@ std::optional AndroidNetworkMonitor::FindNetworkHandleFromIfname( } if (bind_using_ifname_) { - for (auto const& iter : network_handle_by_if_name_) { + for (auto const& c_iter : network_handle_by_if_name_) { // Use substring match so that e.g if_name="v4-wlan0" is matched // agains iter="wlan0" - if (if_name.find(iter.first) != absl::string_view::npos) { - return std::make_optional(iter.second); + if (if_name.find(c_iter.first) != absl::string_view::npos) { + return std::make_optional(c_iter.second); } } } diff --git a/sdk/objc/components/network/RTCNetworkMonitor.mm b/sdk/objc/components/network/RTCNetworkMonitor.mm index e3f312362f..df93636eca 100644 --- a/sdk/objc/components/network/RTCNetworkMonitor.mm +++ b/sdk/objc/components/network/RTCNetworkMonitor.mm @@ -95,9 +95,10 @@ - (instancetype)initWithObserver:(webrtc::NetworkMonitorObserver *)observer { return true; }); @synchronized(strongSelf) { - webrtc::NetworkMonitorObserver *observer = strongSelf->_observer; - if (observer) { - observer->OnPathUpdate(std::move(owned_map)); + webrtc::NetworkMonitorObserver *strongObserver = + strongSelf->_observer; + if (strongObserver) { + strongObserver->OnPathUpdate(std::move(owned_map)); } } }); diff --git a/sdk/objc/native/src/audio/audio_device_ios.mm b/sdk/objc/native/src/audio/audio_device_ios.mm index 25c1e02022..c7e8b62b83 100644 --- a/sdk/objc/native/src/audio/audio_device_ios.mm +++ b/sdk/objc/native/src/audio/audio_device_ios.mm @@ -688,9 +688,9 @@ static void LogDeviceInfo() { if (restart_audio_unit) { OSStatus result = audio_unit_->Start(); if (result != noErr) { - RTC_OBJC_TYPE(RTCAudioSession)* session = + RTC_OBJC_TYPE(RTCAudioSession)* new_session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance]; - [session notifyAudioUnitStartFailedWithError:result]; + [new_session notifyAudioUnitStartFailedWithError:result]; RTCLogError(@"Failed to start audio unit with sample rate: %d, reason %d", playout_parameters_.sample_rate(), result); diff --git a/sdk/objc/unittests/RTCPeerConnectionFactory_xctest.m b/sdk/objc/unittests/RTCPeerConnectionFactory_xctest.m index c5e39a6a30..ec88d6d257 100644 --- a/sdk/objc/unittests/RTCPeerConnectionFactory_xctest.m +++ b/sdk/objc/unittests/RTCPeerConnectionFactory_xctest.m @@ -370,11 +370,11 @@ - (void)testRollback { __weak RTC_OBJC_TYPE(RTCPeerConnection) *weakPC1 = pc1; [pc1 setLocalDescription:offer - completionHandler:^(NSError *error) { - XCTAssertNil(error); + completionHandler:^(NSError *sldError) { + XCTAssertNil(sldError); [weakPC1 setLocalDescription:rollback - completionHandler:^(NSError *error) { - XCTAssertNil(error); + completionHandler:^(NSError *rollbackError) { + XCTAssertNil(rollbackError); }]; }]; NSTimeInterval negotiationTimeout = 15; @@ -522,8 +522,8 @@ - (void)testSetCodecPreferences { offerForConstraints:constraints completionHandler:^( RTC_OBJC_TYPE(RTCSessionDescription) *_Nullable sdp, - NSError *_Nullable error) { - XCTAssertNil(error); + NSError *_Nullable createOfferError) { + XCTAssertNil(createOfferError); XCTAssertNotNil(sdp); NSArray *rtpMaps = [self rtpMapsFromSDP:sdp.sdp]; @@ -611,8 +611,8 @@ __block RTC_OBJC_TYPE(RTCRtpHeaderExtensionCapability) *targetExtension = offerForConstraints:constraints completionHandler:^( RTC_OBJC_TYPE(RTCSessionDescription) *_Nullable sdp, - NSError *_Nullable error) { - XCTAssertNil(error); + NSError *_Nullable createOfferError) { + XCTAssertNil(createOfferError); XCTAssertNotNil(sdp); NSArray *extMaps = [self extMapsFromSDP:sdp.sdp]; @@ -712,34 +712,37 @@ - (bool)negotiatePeerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)pc1 [weakPC1 offerForConstraints:sdpConstraints completionHandler:^(RTC_OBJC_TYPE(RTCSessionDescription) * offer, - NSError * error) { - XCTAssertNil(error); + NSError * createOfferError) { + XCTAssertNil(createOfferError); XCTAssertNotNil(offer); [weakPC1 setLocalDescription:offer - completionHandler:^(NSError *error) { - XCTAssertNil(error); + completionHandler:^(NSError *sldError) { + XCTAssertNil(sldError); [weakPC2 setRemoteDescription:offer - completionHandler:^(NSError *error) { - XCTAssertNil(error); + completionHandler:^(NSError *srdError) { + XCTAssertNil(srdError); [weakPC2 answerForConstraints:sdpConstraints completionHandler:^( RTC_OBJC_TYPE(RTCSessionDescription) * answer, - NSError * error) { - XCTAssertNil(error); + NSError * createAnswerError) { + XCTAssertNil(createAnswerError); XCTAssertNotNil(answer); [weakPC2 setLocalDescription:answer - completionHandler:^(NSError *error) { - XCTAssertNil(error); + completionHandler:^( + NSError *sldAnswerError) { + XCTAssertNil(sldAnswerError); [weakPC1 setRemoteDescription:answer completionHandler:^( - NSError *error) { - XCTAssertNil(error); + NSError + *srdAnswerError) { + XCTAssertNil( + srdAnswerError); dispatch_semaphore_signal( negotiatedSem); }]; diff --git a/test/fuzzers/rtp_packet_fuzzer.cc b/test/fuzzers/rtp_packet_fuzzer.cc index d2d4376bed..2b9c95a5b7 100644 --- a/test/fuzzers/rtp_packet_fuzzer.cc +++ b/test/fuzzers/rtp_packet_fuzzer.cc @@ -79,10 +79,11 @@ void FuzzOneInput(const uint8_t* data, size_t size) { case kRtpExtensionNone: case kRtpExtensionNumberOfExtensions: break; - case kRtpExtensionTransmissionTimeOffset: + case kRtpExtensionTransmissionTimeOffset: { int32_t offset; packet.GetExtension(&offset); break; + } case kRtpExtensionAudioLevel: { AudioLevel audio_level; packet.GetExtension(&audio_level); @@ -102,14 +103,16 @@ void FuzzOneInput(const uint8_t* data, size_t size) { packet.GetExtension(&extension); break; } - case kRtpExtensionVideoRotation: + case kRtpExtensionVideoRotation: { uint8_t rotation; packet.GetExtension(&rotation); break; - case kRtpExtensionTransportSequenceNumber: + } + case kRtpExtensionTransportSequenceNumber: { uint16_t seqnum; packet.GetExtension(&seqnum); break; + } case kRtpExtensionTransportSequenceNumber02: { uint16_t seqnum; std::optional feedback_request; @@ -122,10 +125,11 @@ void FuzzOneInput(const uint8_t* data, size_t size) { packet.GetExtension(&playout); break; } - case kRtpExtensionVideoContentType: + case kRtpExtensionVideoContentType: { VideoContentType content_type; packet.GetExtension(&content_type); break; + } case kRtpExtensionVideoTiming: { VideoSendTiming timing; packet.GetExtension(&timing); From e6f4fab81c89a23a91e673443e0540e3b82f0063 Mon Sep 17 00:00:00 2001 From: Tom Van Goethem Date: Thu, 8 May 2025 16:02:13 +0000 Subject: [PATCH 086/292] Add metrics for SDP munging outcome The outcome of SDP munging is logged to metrics. This checks whether any munging was done, and if so, whether it was accepted or rejected. The metrics are defined in https://crrev.com/c/6520602. Bug: chromium:40567530 Change-Id: Icf55f7bd089214160b0231b232a191ee3940bf34 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390401 Commit-Queue: Harald Alvestrand Reviewed-by: Johannes Kron Auto-Submit: Tom Van Goethem Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44558} --- api/uma_metrics.h | 10 +++++++ pc/sdp_munging_detector_unittest.cc | 42 +++++++++++++++++++++++++++++ pc/sdp_offer_answer.cc | 13 +++++++++ 3 files changed, 65 insertions(+) diff --git a/api/uma_metrics.h b/api/uma_metrics.h index fdca7dcceb..e6f722ca43 100644 --- a/api/uma_metrics.h +++ b/api/uma_metrics.h @@ -227,6 +227,16 @@ enum SdpMungingType { kMaxValue, }; +// The outcome of setting the local description, whether SDP munging is detected +// and if the should be accepted or rejected. Keep in sync with +// SdpMungingOutcome from tools/metrics/histograms/metadata/web_rtc/enums.xml +enum class SdpMungingOutcome { + kNoMunge = 0, + kAccepted = 1, + kRejected = 2, + kMaxValue = kRejected, +}; + // When adding new metrics please consider using the style described in // https://chromium.googlesource.com/chromium/src.git/+/HEAD/tools/metrics/histograms/README.md#usage // instead of the legacy enums used above. diff --git a/pc/sdp_munging_detector_unittest.cc b/pc/sdp_munging_detector_unittest.cc index a06735b6f8..ea2ac1ecbd 100644 --- a/pc/sdp_munging_detector_unittest.cc +++ b/pc/sdp_munging_detector_unittest.cc @@ -273,6 +273,12 @@ TEST_F(SdpMungingTest, IceUfrag) { EXPECT_THAT( metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), ElementsAre(Pair(SdpMungingType::kIceUfrag, 1))); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.SdpOutcome.Rejected"), + ElementsAre(Pair(SdpMungingType::kIceUfrag, 1))); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Outcome"), + ElementsAre(Pair(static_cast(SdpMungingOutcome::kRejected), 1))); } TEST_F(SdpMungingTest, IceUfragCheckDisabledByFieldTrial) { @@ -291,6 +297,12 @@ TEST_F(SdpMungingTest, IceUfragCheckDisabledByFieldTrial) { EXPECT_THAT( metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), ElementsAre(Pair(SdpMungingType::kIceUfrag, 1))); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.SdpOutcome.Accepted"), + ElementsAre(Pair(SdpMungingType::kIceUfrag, 1))); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Outcome"), + ElementsAre(Pair(static_cast(SdpMungingOutcome::kAccepted), 1))); } TEST_F(SdpMungingTest, IceUfragWithCheckDisabledForTesting) { @@ -324,6 +336,12 @@ TEST_F(SdpMungingTest, IcePwdCheckDisabledByFieldTrial) { EXPECT_THAT( metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), ElementsAre(Pair(SdpMungingType::kIcePwd, 1))); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.SdpOutcome.Accepted"), + ElementsAre(Pair(SdpMungingType::kIcePwd, 1))); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Outcome"), + ElementsAre(Pair(static_cast(SdpMungingOutcome::kAccepted), 1))); } TEST_F(SdpMungingTest, IcePwd) { @@ -340,6 +358,12 @@ TEST_F(SdpMungingTest, IcePwd) { EXPECT_THAT( metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), ElementsAre(Pair(SdpMungingType::kIcePwd, 1))); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.SdpOutcome.Rejected"), + ElementsAre(Pair(SdpMungingType::kIcePwd, 1))); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Outcome"), + ElementsAre(Pair(static_cast(SdpMungingOutcome::kRejected), 1))); } TEST_F(SdpMungingTest, IceUfragRestrictedAddresses) { @@ -416,6 +440,12 @@ TEST_F(SdpMungingTest, IceUfragSdpRejectedAndRestrictedAddresses) { EXPECT_THAT( metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), ElementsAre(Pair(SdpMungingType::kIceUfrag, 1))); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.SdpOutcome.Rejected"), + ElementsAre(Pair(SdpMungingType::kIceUfrag, 1))); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Outcome"), + ElementsAre(Pair(static_cast(SdpMungingOutcome::kRejected), 1))); } TEST_F(SdpMungingTest, IceMode) { @@ -500,6 +530,12 @@ TEST_F(SdpMungingTest, RemoveContentDefault) { EXPECT_THAT( metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), ElementsAre(Pair(SdpMungingType::kNumberOfContents, 1))); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.SdpOutcome.Rejected"), + ElementsAre(Pair(SdpMungingType::kNumberOfContents, 1))); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Outcome"), + ElementsAre(Pair(static_cast(SdpMungingOutcome::kRejected), 1))); } TEST_F(SdpMungingTest, RemoveContentKillswitch) { @@ -523,6 +559,12 @@ TEST_F(SdpMungingTest, RemoveContentKillswitch) { EXPECT_THAT( metrics::Samples("WebRTC.PeerConnection.SdpMunging.Offer.Initial"), ElementsAre(Pair(SdpMungingType::kNumberOfContents, 1))); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.SdpOutcome.Accepted"), + ElementsAre(Pair(SdpMungingType::kNumberOfContents, 1))); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.SdpMunging.Outcome"), + ElementsAre(Pair(static_cast(SdpMungingOutcome::kAccepted), 1))); } TEST_F(SdpMungingTest, TransceiverDirection) { diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc index 6f19d1bee6..ecb4e8b836 100644 --- a/pc/sdp_offer_answer.cc +++ b/pc/sdp_offer_answer.cc @@ -2486,14 +2486,27 @@ void SdpOfferAnswerHandler::DoSetLocalDescription( "WebRTC-NoSdpMangleNumberOfContents")) { reject_error = true; } + SdpMungingOutcome outcome = reject_error ? SdpMungingOutcome::kRejected + : SdpMungingOutcome::kAccepted; + RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SdpMunging.Outcome", + static_cast(outcome), + static_cast(SdpMungingOutcome::kMaxValue)); if (reject_error) { observer->OnSetLocalDescriptionComplete( RTCError(RTCErrorType::INVALID_MODIFICATION, "SDP is modified in a non-acceptable way")); last_sdp_munging_type_ = sdp_munging_type; ReportInitialSdpMunging(had_local_description, desc->GetType()); + RTC_HISTOGRAM_ENUMERATION_SPARSE( + "WebRTC.PeerConnection.SdpMunging.SdpOutcome.Rejected", + sdp_munging_type, SdpMungingType::kMaxValue); return; } + if (sdp_munging_type != kNoModification) { + RTC_HISTOGRAM_ENUMERATION_SPARSE( + "WebRTC.PeerConnection.SdpMunging.SdpOutcome.Accepted", + sdp_munging_type, SdpMungingType::kMaxValue); + } } // Workaround for isses.webrtc.org/412904801 - detect if packetization:raw From d5b3b1ed4594821451d2ebb1d61cc1e6dab83b56 Mon Sep 17 00:00:00 2001 From: Tom Van Goethem Date: Thu, 8 May 2025 14:19:56 +0000 Subject: [PATCH 087/292] Add metrics for restricted candidates This adds two UMA metrics for the type of SDP munging that occurred and the port of the candidate that was restricted. The metrics descriptions are being added here: crrev.com/c/6521706. Bug: b/409713509 Change-Id: I3232cb0cee6848074ab103f4d4edb2e080e09568 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390340 Reviewed-by: Harald Alvestrand Reviewed-by: Johannes Kron Commit-Queue: Tom Van Goethem Cr-Commit-Position: refs/heads/main@{#44559} --- pc/sdp_munging_detector_unittest.cc | 16 ++++++++++++++++ pc/sdp_offer_answer.cc | 12 ++++++++++++ 2 files changed, 28 insertions(+) diff --git a/pc/sdp_munging_detector_unittest.cc b/pc/sdp_munging_detector_unittest.cc index ea2ac1ecbd..340021ebfb 100644 --- a/pc/sdp_munging_detector_unittest.cc +++ b/pc/sdp_munging_detector_unittest.cc @@ -17,8 +17,11 @@ #include #include +#include "absl/strings/numbers.h" #include "absl/strings/str_cat.h" #include "absl/strings/str_replace.h" +#include "absl/strings/str_split.h" +#include "absl/strings/string_view.h" #include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" @@ -399,6 +402,7 @@ TEST_F(SdpMungingTest, IceUfragRestrictedAddresses) { {"127.0.1.1:23456", true}, {"8.8.8.8:3456", true}, }; + int num_blocked = 0; for (const auto& address_test : address_tests) { std::optional result; const std::string candidate = StringFormat( @@ -414,8 +418,20 @@ TEST_F(SdpMungingTest, IceUfragRestrictedAddresses) { if (address_test.second == true) { EXPECT_TRUE(result.value().ok()); } else { + std::pair host = + absl::StrSplit(address_test.first, ":"); + int port; + ASSERT_TRUE(absl::SimpleAtoi(host.second, &port)); EXPECT_FALSE(result.value().ok()); EXPECT_EQ(result.value().type(), RTCErrorType::UNSUPPORTED_OPERATION); + num_blocked++; + EXPECT_THAT( + metrics::Samples( + "WebRTC.PeerConnection.RestrictedCandidates.SdpMungingType"), + ElementsAre(Pair(SdpMungingType::kIceUfrag, num_blocked))); + EXPECT_THAT( + metrics::Samples("WebRTC.PeerConnection.RestrictedCandidates.Port"), + Contains(Pair(port, 1))); } } } diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc index ecb4e8b836..5d4ce93856 100644 --- a/pc/sdp_offer_answer.cc +++ b/pc/sdp_offer_answer.cc @@ -5357,6 +5357,18 @@ bool SdpOfferAnswerHandler::ReadyToUseRemoteCandidate( RTC_LOG(LS_ERROR) << "ReadyToUseRemoteCandidate: Candidate not valid " "because of SDP munging."; *valid = false; + // There might be other types of SDP munging, but here we're only + // interested in IceUfrag and IcePwd. + SdpMungingType sdp_munging_type = + last_sdp_munging_type_ == SdpMungingType::kIcePwd + ? SdpMungingType::kIcePwd + : SdpMungingType::kIceUfrag; + RTC_HISTOGRAM_ENUMERATION_SPARSE( + "WebRTC.PeerConnection.RestrictedCandidates.SdpMungingType", + sdp_munging_type, SdpMungingType::kMaxValue); + RTC_HISTOGRAM_ENUMERATION_SPARSE( + "WebRTC.PeerConnection.RestrictedCandidates.Port", + candidate->candidate().address().port(), 65536); return false; } } From e4e347dc78ee62bab8a8a231fd119f4ffe03672a Mon Sep 17 00:00:00 2001 From: Fredrik Solenberg Date: Fri, 9 May 2025 04:11:54 -0700 Subject: [PATCH 088/292] Revert "document footgun behavior of codec munging and setCodecPreferences" This reverts commit e7162754c0229d10c903525be3f900edb184dfb7. Reason for revert: Breaks downstream project Original change's description: > document footgun behavior of codec munging and setCodecPreferences > > which drops the munged codec > > BUG=webrtc:396344370 > > Change-Id: I4980c6980ad7df29d4d1451777a61f90dac73f74 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389440 > Reviewed-by: Evan Shrubsole > Reviewed-by: Harald Alvestrand > Commit-Queue: Philipp Hancke > Cr-Commit-Position: refs/heads/main@{#44555} Bug: webrtc:396344370 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I86e7929217420bf4f1741910e628430b660d4418 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390660 Commit-Queue: Harald Alvestrand Auto-Submit: Fredrik Solenberg Reviewed-by: Evan Shrubsole Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44560} --- pc/peer_connection_integrationtest.cc | 98 +-------------------------- 1 file changed, 2 insertions(+), 96 deletions(-) diff --git a/pc/peer_connection_integrationtest.cc b/pc/peer_connection_integrationtest.cc index a3e975086b..95bd835301 100644 --- a/pc/peer_connection_integrationtest.cc +++ b/pc/peer_connection_integrationtest.cc @@ -18,7 +18,6 @@ #include #include -#include #include #include #include @@ -4794,99 +4793,6 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan, EXPECT_TRUE(caller()->SetLocalDescriptionAndSendSdpMessage(std::move(offer))); } -TEST_F(PeerConnectionIntegrationTestUnifiedPlan, - MungeOfferCodecAndReOfferWorksWithSetCodecPreferencesIsAFootgun) { - ASSERT_TRUE(CreatePeerConnectionWrappers()); - ConnectFakeSignaling(); - caller()->AddVideoTrack(); - auto munger = [](std::unique_ptr& sdp) { - auto video = GetFirstVideoContentDescription(sdp->description()); - auto codecs = video->codecs(); - std::optional replacement_codec; - for (auto&& codec : codecs) { - if (codec.name == "AV1") { - replacement_codec = codec; - break; - } - } - if (replacement_codec) { - for (auto&& codec : codecs) { - if (codec.name == "VP8") { - RTC_LOG(LS_INFO) << "Remapping VP8 codec " << codec << " to AV1"; - codec.name = replacement_codec->name; - codec.params = replacement_codec->params; - break; - } - } - video->set_codecs(codecs); - } else { - RTC_LOG(LS_INFO) << "Skipping munge, no AV1 codec found"; - } - }; - caller()->SetGeneratedSdpMunger(munger); - caller()->CreateAndSetAndSignalOffer(); - ASSERT_THAT( - WaitUntil([&] { return SignalingStateStable(); }, ::testing::IsTrue()), - IsRtcOk()); - caller()->SetGeneratedSdpMunger(nullptr); - - // Note currently negotiated codecs and count VP9 and AV1. - auto codecs_after_munge = caller() - ->pc() - ->local_description() - ->description() - ->contents()[0] - .media_description() - ->codecs(); - size_t av1_munge = 0; - size_t vp8_munge = 0; - for (const auto& codec : codecs_after_munge) { - if (codec.name == "AV1") - av1_munge++; - else if (codec.name == "VP8") - vp8_munge++; - } - // We should have replaced VP8 with AV1. - EXPECT_EQ(av1_munge, 2u); - EXPECT_EQ(vp8_munge, 0u); - - // Call setCodecPreferences. - std::vector codecs = - caller() - ->pc_factory() - ->GetRtpReceiverCapabilities(webrtc::MediaType::VIDEO) - .codecs; - auto transceivers = caller()->pc()->GetTransceivers(); - ASSERT_EQ(transceivers.size(), 1u); - transceivers[0]->SetCodecPreferences(codecs); - - auto offer = caller()->CreateOfferAndWait(); - ASSERT_NE(offer, nullptr); - // The offer should be acceptable. - EXPECT_TRUE(caller()->SetLocalDescriptionAndSendSdpMessage(std::move(offer))); - - auto codecs_after_scp = caller() - ->pc() - ->local_description() - ->description() - ->contents()[0] - .media_description() - ->codecs(); - size_t av1_scp = 0; - size_t vp8_scp = 0; - for (const auto& codec : codecs_after_scp) { - if (codec.name == "AV1") - av1_scp++; - else if (codec.name == "VP8") - vp8_scp++; - } - // The SDP munging modification was reverted by sCP. - // This is a footgun but please do not mix such munging with - // setCodecPreferences, munging is on the way out. - EXPECT_EQ(av1_scp, 1u); - EXPECT_EQ(vp8_scp, 1u); -} - TEST_F(PeerConnectionIntegrationTestUnifiedPlan, SensibleRtxWithDuplicateCodecs) { ASSERT_TRUE(CreatePeerConnectionWrappers()); @@ -4931,7 +4837,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan, // associated RTX codec. std::unique_ptr answer = caller()->CreateAnswerForTest(); - ASSERT_NE(answer, nullptr); + ASSERT_THAT(answer, NotNull()); RTC_LOG(LS_ERROR) << "Answer is " << *answer; ASSERT_THAT(answer->description()->contents().size(), Eq(1)); auto codecs = @@ -4990,7 +4896,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan, caller()->AddVideoTrack(); auto offer2 = caller()->CreateOfferAndWait(); // Observe that packetization is raw on BOTH media sections. - ASSERT_NE(offer2, nullptr); + ASSERT_THAT(offer2, NotNull()); EXPECT_EQ(offer2->description()->contents().size(), 2U); for (const auto& content : offer2->description()->contents()) { for (const auto& codec : content.media_description()->codecs()) { From 7b785551ec7e55fbeaa2eafaa873d6453d8bef70 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Fri, 9 May 2025 10:34:11 +0000 Subject: [PATCH 089/292] [cleanup] Fix redundant webrtc name specifier This CL was uploaded by git cl split. R=hta@webrtc.org Bug: webrtc:42232595 Change-Id: If9392739ec311ae2c2d32bb6244e9586c3b8216c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390561 Commit-Queue: Harald Alvestrand Auto-Submit: Evan Shrubsole Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44561} --- api/video/encoded_image.cc | 2 +- api/video/i010_buffer.cc | 8 +++----- api/video/i210_buffer.cc | 5 ++--- api/video/i410_buffer.cc | 5 ++--- api/video/i420_buffer.cc | 5 ++--- api/video/i422_buffer.cc | 5 ++--- api/video/i444_buffer.cc | 5 ++--- api/video/rtp_video_frame_assembler.cc | 2 +- api/video/video_frame.cc | 2 +- api/video_codecs/libaom_av1_encoder_factory.cc | 13 ++++++------- api/video_codecs/sdp_video_format.cc | 2 +- api/video_codecs/simple_encoder_wrapper.cc | 7 +++---- ...ecoder_software_fallback_wrapper_unittest.cc | 4 ++-- ...ncoder_software_fallback_wrapper_unittest.cc | 11 +++++------ .../video_encoder_software_fallback_wrapper.cc | 17 ++++++++--------- 15 files changed, 41 insertions(+), 52 deletions(-) diff --git a/api/video/encoded_image.cc b/api/video/encoded_image.cc index c5f1787f1a..163250a87e 100644 --- a/api/video/encoded_image.cc +++ b/api/video/encoded_image.cc @@ -77,7 +77,7 @@ void EncodedImage::SetEncodeTime(int64_t encode_start_ms, timing_.encode_finish_ms = encode_finish_ms; } -webrtc::Timestamp EncodedImage::CaptureTime() const { +Timestamp EncodedImage::CaptureTime() const { return capture_time_ms_ > 0 ? Timestamp::Millis(capture_time_ms_) : Timestamp::MinusInfinity(); } diff --git a/api/video/i010_buffer.cc b/api/video/i010_buffer.cc index e9cbe09298..59da7fb114 100644 --- a/api/video/i010_buffer.cc +++ b/api/video/i010_buffer.cc @@ -103,7 +103,7 @@ scoped_refptr I010Buffer::Copy(const I420BufferInterface& source) { // static scoped_refptr I010Buffer::Rotate(const I010BufferInterface& src, VideoRotation rotation) { - if (rotation == webrtc::kVideoRotation_0) + if (rotation == kVideoRotation_0) return Copy(src); RTC_CHECK(src.DataY()); @@ -111,13 +111,11 @@ scoped_refptr I010Buffer::Rotate(const I010BufferInterface& src, RTC_CHECK(src.DataV()); int rotated_width = src.width(); int rotated_height = src.height(); - if (rotation == webrtc::kVideoRotation_90 || - rotation == webrtc::kVideoRotation_270) { + if (rotation == kVideoRotation_90 || rotation == kVideoRotation_270) { std::swap(rotated_width, rotated_height); } - scoped_refptr buffer = - Create(rotated_width, rotated_height); + scoped_refptr buffer = Create(rotated_width, rotated_height); int res = libyuv::I010Rotate( src.DataY(), src.StrideY(), src.DataU(), src.StrideU(), src.DataV(), diff --git a/api/video/i210_buffer.cc b/api/video/i210_buffer.cc index c01214eff8..1f7919e3b4 100644 --- a/api/video/i210_buffer.cc +++ b/api/video/i210_buffer.cc @@ -110,12 +110,11 @@ scoped_refptr I210Buffer::Rotate(const I210BufferInterface& src, int rotated_width = src.width(); int rotated_height = src.height(); - if (rotation == webrtc::kVideoRotation_90 || - rotation == webrtc::kVideoRotation_270) { + if (rotation == kVideoRotation_90 || rotation == kVideoRotation_270) { std::swap(rotated_width, rotated_height); } - scoped_refptr buffer = + scoped_refptr buffer = I210Buffer::Create(rotated_width, rotated_height); RTC_CHECK_EQ(0, diff --git a/api/video/i410_buffer.cc b/api/video/i410_buffer.cc index f6fc13f25e..37564b9f3a 100644 --- a/api/video/i410_buffer.cc +++ b/api/video/i410_buffer.cc @@ -122,12 +122,11 @@ scoped_refptr I410Buffer::Rotate(const I410BufferInterface& src, int rotated_width = src.width(); int rotated_height = src.height(); - if (rotation == webrtc::kVideoRotation_90 || - rotation == webrtc::kVideoRotation_270) { + if (rotation == kVideoRotation_90 || rotation == kVideoRotation_270) { std::swap(rotated_width, rotated_height); } - scoped_refptr buffer = + scoped_refptr buffer = I410Buffer::Create(rotated_width, rotated_height); int res = libyuv::I410Rotate( diff --git a/api/video/i420_buffer.cc b/api/video/i420_buffer.cc index 39260b125a..33b2ea666e 100644 --- a/api/video/i420_buffer.cc +++ b/api/video/i420_buffer.cc @@ -120,12 +120,11 @@ scoped_refptr I420Buffer::Rotate(const I420BufferInterface& src, int rotated_width = src.width(); int rotated_height = src.height(); - if (rotation == webrtc::kVideoRotation_90 || - rotation == webrtc::kVideoRotation_270) { + if (rotation == kVideoRotation_90 || rotation == kVideoRotation_270) { std::swap(rotated_width, rotated_height); } - scoped_refptr buffer = + scoped_refptr buffer = I420Buffer::Create(rotated_width, rotated_height); RTC_CHECK_EQ(0, diff --git a/api/video/i422_buffer.cc b/api/video/i422_buffer.cc index 4322ace00c..25814a88d9 100644 --- a/api/video/i422_buffer.cc +++ b/api/video/i422_buffer.cc @@ -136,12 +136,11 @@ scoped_refptr I422Buffer::Rotate(const I422BufferInterface& src, int rotated_width = src.width(); int rotated_height = src.height(); - if (rotation == webrtc::kVideoRotation_90 || - rotation == webrtc::kVideoRotation_270) { + if (rotation == kVideoRotation_90 || rotation == kVideoRotation_270) { std::swap(rotated_width, rotated_height); } - scoped_refptr buffer = + scoped_refptr buffer = I422Buffer::Create(rotated_width, rotated_height); int res = libyuv::I422Rotate( diff --git a/api/video/i444_buffer.cc b/api/video/i444_buffer.cc index 73486622bc..ace6af1ea1 100644 --- a/api/video/i444_buffer.cc +++ b/api/video/i444_buffer.cc @@ -119,12 +119,11 @@ scoped_refptr I444Buffer::Rotate(const I444BufferInterface& src, int rotated_width = src.width(); int rotated_height = src.height(); - if (rotation == webrtc::kVideoRotation_90 || - rotation == webrtc::kVideoRotation_270) { + if (rotation == kVideoRotation_90 || rotation == kVideoRotation_270) { std::swap(rotated_width, rotated_height); } - scoped_refptr buffer = + scoped_refptr buffer = I444Buffer::Create(rotated_width, rotated_height); RTC_CHECK_EQ(0, diff --git a/api/video/rtp_video_frame_assembler.cc b/api/video/rtp_video_frame_assembler.cc index 7d79b7a5aa..e09688a438 100644 --- a/api/video/rtp_video_frame_assembler.cc +++ b/api/video/rtp_video_frame_assembler.cc @@ -239,7 +239,7 @@ RtpVideoFrameAssembler::Impl::UpdateWithPadding(uint16_t seq_num) { bool RtpVideoFrameAssembler::Impl::ParseDependenciesDescriptorExtension( const RtpPacketReceived& rtp_packet, RTPVideoHeader& video_header) { - webrtc::DependencyDescriptor dependency_descriptor; + DependencyDescriptor dependency_descriptor; if (!rtp_packet.GetExtension( video_structure_.get(), &dependency_descriptor)) { diff --git a/api/video/video_frame.cc b/api/video/video_frame.cc index 8483ed011a..8ae4e3348a 100644 --- a/api/video/video_frame.cc +++ b/api/video/video_frame.cc @@ -265,7 +265,7 @@ VideoFrame::Builder& VideoFrame::Builder::set_packet_infos( } VideoFrame::VideoFrame(const scoped_refptr& buffer, - webrtc::VideoRotation rotation, + VideoRotation rotation, int64_t timestamp_us) : video_frame_buffer_(buffer), timestamp_rtp_(0), diff --git a/api/video_codecs/libaom_av1_encoder_factory.cc b/api/video_codecs/libaom_av1_encoder_factory.cc index 7652aae250..f82538f5c8 100644 --- a/api/video_codecs/libaom_av1_encoder_factory.cc +++ b/api/video_codecs/libaom_av1_encoder_factory.cc @@ -105,7 +105,7 @@ class LibaomAv1Encoder : public VideoEncoderInterface { const VideoEncoderFactoryInterface::StaticEncoderSettings& settings, const std::map& encoder_specific_settings); - void Encode(scoped_refptr frame_buffer, + void Encode(scoped_refptr frame_buffer, const TemporalUnitSettings& tu_settings, std::vector frame_settings) override; @@ -269,7 +269,7 @@ ThreadTilesAndSuperblockSizeInfo GetThreadingTilesAndSuperblockSize( } bool ValidateEncodeParams( - const webrtc::VideoFrameBuffer& /* frame_buffer */, + const VideoFrameBuffer& /* frame_buffer */, const VideoEncoderInterface::TemporalUnitSettings& /* tu_settings */, const std::vector& frame_settings, @@ -537,7 +537,7 @@ aom_svc_ref_frame_config_t GetSvcRefFrameConfig( } aom_svc_params_t GetSvcParams( - const webrtc::VideoFrameBuffer& frame_buffer, + const VideoFrameBuffer& frame_buffer, const std::vector& frame_settings) { aom_svc_params_t svc_params = {}; @@ -634,10 +634,9 @@ aom_svc_params_t GetSvcParams( return svc_params; } -void LibaomAv1Encoder::Encode( - scoped_refptr frame_buffer, - const TemporalUnitSettings& tu_settings, - std::vector frame_settings) { +void LibaomAv1Encoder::Encode(scoped_refptr frame_buffer, + const TemporalUnitSettings& tu_settings, + std::vector frame_settings) { absl::Cleanup on_return = [&] { // On return call `EncodeComplete` with EncodingError result unless they // were already called with an EncodedData result. diff --git a/api/video_codecs/sdp_video_format.cc b/api/video_codecs/sdp_video_format.cc index 5c2c5dac3c..6d70ae3850 100644 --- a/api/video_codecs/sdp_video_format.cc +++ b/api/video_codecs/sdp_video_format.cc @@ -191,7 +191,7 @@ bool SdpVideoFormat::IsSameCodec(const SdpVideoFormat& other) const { } bool SdpVideoFormat::IsCodecInList( - ArrayView formats) const { + ArrayView formats) const { for (const auto& format : formats) { if (IsSameCodec(format)) { return true; diff --git a/api/video_codecs/simple_encoder_wrapper.cc b/api/video_codecs/simple_encoder_wrapper.cc index cf0fd80f8e..7ecec6d134 100644 --- a/api/video_codecs/simple_encoder_wrapper.cc +++ b/api/video_codecs/simple_encoder_wrapper.cc @@ -152,10 +152,9 @@ void SimpleEncoderWrapper::SetEncodeFps(int fps) { fps_ = fps; } -void SimpleEncoderWrapper::Encode( - scoped_refptr frame_buffer, - bool force_keyframe, - EncodeResultCallback callback) { +void SimpleEncoderWrapper::Encode(scoped_refptr frame_buffer, + bool force_keyframe, + EncodeResultCallback callback) { std::vector configs = svc_controller_->NextFrameConfig(force_keyframe); std::vector encode_settings; diff --git a/api/video_codecs/test/video_decoder_software_fallback_wrapper_unittest.cc b/api/video_codecs/test/video_decoder_software_fallback_wrapper_unittest.cc index d452884c11..5ff941f273 100644 --- a/api/video_codecs/test/video_decoder_software_fallback_wrapper_unittest.cc +++ b/api/video_codecs/test/video_decoder_software_fallback_wrapper_unittest.cc @@ -184,12 +184,12 @@ TEST_F(VideoDecoderSoftwareFallbackWrapperTest, ForwardsRegisterDecodeCompleteCallback) { class FakeDecodedImageCallback : public DecodedImageCallback { int32_t Decoded(VideoFrame& /* decodedImage */) override { return 0; } - int32_t Decoded(webrtc::VideoFrame& /* decodedImage */, + int32_t Decoded(VideoFrame& /* decodedImage */, int64_t /* decode_time_ms */) override { RTC_DCHECK_NOTREACHED(); return -1; } - void Decoded(webrtc::VideoFrame& /* decodedImage */, + void Decoded(VideoFrame& /* decodedImage */, std::optional /* decode_time_ms */, std::optional /* qp */) override { RTC_DCHECK_NOTREACHED(); diff --git a/api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc b/api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc index 94e31ecf8b..903e7fe1d4 100644 --- a/api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc +++ b/api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc @@ -219,12 +219,11 @@ void VideoEncoderSoftwareFallbackWrapperTestBase::EncodeFrame( I420Buffer::SetBlack(buffer.get()); std::vector types(1, VideoFrameType::kVideoFrameKey); - frame_ = - std::make_unique(VideoFrame::Builder() - .set_video_frame_buffer(buffer) - .set_rotation(webrtc::kVideoRotation_0) - .set_timestamp_us(0) - .build()); + frame_ = std::make_unique(VideoFrame::Builder() + .set_video_frame_buffer(buffer) + .set_rotation(kVideoRotation_0) + .set_timestamp_us(0) + .build()); EXPECT_EQ(expected_ret, fallback_wrapper_->Encode(*frame_, &types)); } diff --git a/api/video_codecs/video_encoder_software_fallback_wrapper.cc b/api/video_codecs/video_encoder_software_fallback_wrapper.cc index 1ef2c609df..e1b7910b16 100644 --- a/api/video_codecs/video_encoder_software_fallback_wrapper.cc +++ b/api/video_codecs/video_encoder_software_fallback_wrapper.cc @@ -144,11 +144,10 @@ std::optional GetForcedFallbackParams( class VideoEncoderSoftwareFallbackWrapper final : public VideoEncoder { public: - VideoEncoderSoftwareFallbackWrapper( - const FieldTrialsView& field_trials, - std::unique_ptr sw_encoder, - std::unique_ptr hw_encoder, - bool prefer_temporal_support); + VideoEncoderSoftwareFallbackWrapper(const FieldTrialsView& field_trials, + std::unique_ptr sw_encoder, + std::unique_ptr hw_encoder, + bool prefer_temporal_support); ~VideoEncoderSoftwareFallbackWrapper() override; void SetFecControllerOverride( @@ -221,8 +220,8 @@ class VideoEncoderSoftwareFallbackWrapper final : public VideoEncoder { }; EncoderState encoder_state_; - const std::unique_ptr encoder_; - const std::unique_ptr fallback_encoder_; + const std::unique_ptr encoder_; + const std::unique_ptr fallback_encoder_; EncodedImageCallback* callback_; @@ -233,8 +232,8 @@ class VideoEncoderSoftwareFallbackWrapper final : public VideoEncoder { VideoEncoderSoftwareFallbackWrapper::VideoEncoderSoftwareFallbackWrapper( const FieldTrialsView& field_trials, - std::unique_ptr sw_encoder, - std::unique_ptr hw_encoder, + std::unique_ptr sw_encoder, + std::unique_ptr hw_encoder, bool prefer_temporal_support) : encoder_state_(EncoderState::kUninitialized), encoder_(std::move(hw_encoder)), From ef14b26160ce263dc2623991be994eb0203dfb37 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Fri, 9 May 2025 10:36:40 +0000 Subject: [PATCH 090/292] [cleanup] Fix redundant webrtc name specifier This CL was uploaded by git cl split. R=hta@webrtc.org Bug: webrtc:42232595 Change-Id: I9abba6e77795e7ecefec13279ef81ef2eae3dd25 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390603 Auto-Submit: Evan Shrubsole Reviewed-by: Harald Alvestrand Commit-Queue: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44562} --- modules/audio_device/audio_device_unittest.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/modules/audio_device/audio_device_unittest.cc b/modules/audio_device/audio_device_unittest.cc index e7c1386f81..c7b1cb3c62 100644 --- a/modules/audio_device/audio_device_unittest.cc +++ b/modules/audio_device/audio_device_unittest.cc @@ -522,7 +522,7 @@ class MockAudioTransport : public test::MockAudioTransport { #endif class MAYBE_AudioDeviceTest - : public ::testing::TestWithParam { + : public ::testing::TestWithParam { protected: MAYBE_AudioDeviceTest() : audio_layer_(GetParam()), env_(CreateEnvironment()) { From 4dd4ac53dcc51112bddb450296c2fec66ad2d87b Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Thu, 17 Apr 2025 08:40:18 -0700 Subject: [PATCH 091/292] build: enable -Wshadow which prevents shadowing variables. BUG=webrtc:42223409 Change-Id: Icfaa101dc25717a89d1ac50d446dd9b69e18764c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/387100 Reviewed-by: Christoffer Dewerin Reviewed-by: Mirko Bonadei Commit-Queue: Mirko Bonadei Cr-Commit-Position: refs/heads/main@{#44563} --- BUILD.gn | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/BUILD.gn b/BUILD.gn index 2d84f3cae7..4ad7f3e00c 100644 --- a/BUILD.gn +++ b/BUILD.gn @@ -377,8 +377,7 @@ config("common_config") { if (is_clang) { cflags += [ - # TODO(webrtc:13219): Fix -Wshadow instances and enable. - "-Wno-shadow", + "-Wshadow", # See https://reviews.llvm.org/D56731 for details about this # warning. From bbb7c2e39a0a7ade2b83e6d4b9c554dd14104867 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Fri, 9 May 2025 10:36:25 +0000 Subject: [PATCH 092/292] [cleanup] Fix redundant webrtc name specifier This CL was uploaded by git cl split. R=hta@webrtc.org Bug: webrtc:42232595 Change-Id: I0058f780e712d9bf42ea9f9c9eec68ea23d3ea0a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390443 Commit-Queue: Harald Alvestrand Auto-Submit: Evan Shrubsole Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44564} --- modules/audio_coding/acm2/acm_receive_test.cc | 2 +- modules/audio_coding/acm2/acm_send_test.cc | 2 +- .../audio_coding/acm2/audio_coding_module.cc | 3 +-- .../acm2/audio_coding_module_unittest.cc | 24 +++++++++---------- .../audio_coding/codecs/cng/cng_unittest.cc | 2 +- .../opus/audio_decoder_opus_unittest.cc | 4 ++-- .../opus/audio_encoder_opus_unittest.cc | 9 ++++--- .../codecs/opus/opus_bandwidth_unittest.cc | 2 +- .../codecs/opus/opus_complexity_unittest.cc | 8 +++---- .../audio_coding/codecs/opus/opus_unittest.cc | 2 +- .../red/audio_encoder_copy_red_unittest.cc | 10 ++++---- .../neteq/audio_decoder_unittest.cc | 5 ++-- .../neteq/neteq_decoder_plc_unittest.cc | 2 +- .../neteq/neteq_stereo_unittest.cc | 2 +- modules/audio_coding/neteq/neteq_unittest.cc | 16 ++++++------- .../neteq/test/neteq_decoding_test.cc | 6 ++--- .../neteq/test/neteq_performance_unittest.cc | 6 ++--- .../audio_coding/neteq/test/result_sink.cc | 4 ++-- .../neteq/tools/neteq_performance_test.cc | 2 +- .../neteq/tools/neteq_quality_test.cc | 9 ++++--- .../neteq/tools/neteq_test_factory.cc | 5 ++-- modules/audio_coding/neteq/tools/packet.cc | 2 +- modules/audio_coding/test/EncodeDecodeTest.cc | 9 ++++--- modules/audio_coding/test/PacketLossTest.cc | 4 ++-- modules/audio_coding/test/TestAllCodecs.cc | 4 ++-- modules/audio_coding/test/TestRedFec.cc | 4 ++-- modules/audio_coding/test/TestStereo.cc | 6 ++--- modules/audio_coding/test/TestVADDTX.cc | 10 ++++---- modules/audio_coding/test/opus_test.cc | 11 ++++----- 29 files changed, 83 insertions(+), 92 deletions(-) diff --git a/modules/audio_coding/acm2/acm_receive_test.cc b/modules/audio_coding/acm2/acm_receive_test.cc index 0915ca3cb0..afa6f77d45 100644 --- a/modules/audio_coding/acm2/acm_receive_test.cc +++ b/modules/audio_coding/acm2/acm_receive_test.cc @@ -105,7 +105,7 @@ void AcmReceiveTestOldApi::Run() { static_cast(output_freq_hz_ * 10 / 1000); EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_); if (exptected_output_channels_ != kArbitraryChannels) { - if (output_frame.speech_type_ == webrtc::AudioFrame::kPLC) { + if (output_frame.speech_type_ == AudioFrame::kPLC) { // Don't check number of channels for PLC output, since each test run // usually starts with a short period of mono PLC before decoding the // first packet. diff --git a/modules/audio_coding/acm2/acm_send_test.cc b/modules/audio_coding/acm2/acm_send_test.cc index fa635f9522..64cd530934 100644 --- a/modules/audio_coding/acm2/acm_send_test.cc +++ b/modules/audio_coding/acm2/acm_send_test.cc @@ -39,7 +39,7 @@ AcmSendTestOldApi::AcmSendTestOldApi(InputAudioFile* audio_source, int test_duration_ms) : clock_(0), env_(CreateEnvironment(&clock_)), - acm_(webrtc::AudioCodingModule::Create()), + acm_(AudioCodingModule::Create()), audio_source_(audio_source), source_rate_hz_(source_rate_hz), input_block_size_samples_( diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc index f2b4237301..21ec232eed 100644 --- a/modules/audio_coding/acm2/audio_coding_module.cc +++ b/modules/audio_coding/acm2/audio_coding_module.cc @@ -180,8 +180,7 @@ class AudioCodingModuleImpl final : public AudioCodingModule { void UpdateCodecTypeHistogram(size_t codec_type) { RTC_HISTOGRAM_ENUMERATION( "WebRTC.Audio.Encoder.CodecType", static_cast(codec_type), - static_cast( - webrtc::AudioEncoder::CodecType::kMaxLoggedAudioCodecTypes)); + static_cast(AudioEncoder::CodecType::kMaxLoggedAudioCodecTypes)); } void AudioCodingModuleImpl::ChangeLogger::MaybeLog(int value) { diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc index 51f49b1112..301da4aef4 100644 --- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc +++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc @@ -707,7 +707,7 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test, // output. test::AudioChecksum audio_checksum; const std::string output_file_name = - webrtc::test::OutputPath() + + test::OutputPath() + ::testing::UnitTest::GetInstance() ->current_test_info() ->test_case_name() + @@ -817,13 +817,12 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test, uint32_t last_timestamp_; std::unique_ptr payload_checksum_; const std::string kTestFileMono32kHz = - webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); + test::ResourcePath("audio_coding/testfile32kHz", "pcm"); const std::string kTestFileFakeStereo32kHz = - webrtc::test::ResourcePath("audio_coding/testfile_fake_stereo_32kHz", - "pcm"); - const std::string kTestFileQuad48kHz = webrtc::test::ResourcePath( - "audio_coding/speech_4_channels_48k_one_second", - "wav"); + test::ResourcePath("audio_coding/testfile_fake_stereo_32kHz", "pcm"); + const std::string kTestFileQuad48kHz = + test::ResourcePath("audio_coding/speech_4_channels_48k_one_second", + "wav"); }; class AcmSenderBitExactnessNewApi : public AcmSenderBitExactnessOldApi {}; @@ -1047,7 +1046,7 @@ class AcmSetBitRateTest : public ::testing::Test { // false. bool SetUpSender() { const std::string input_file_name = - webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); + test::ResourcePath("audio_coding/testfile32kHz", "pcm"); // Note that `audio_source_` will loop forever. The test duration is set // explicitly by `kTestDurationMs`. audio_source_.reset(new test::InputAudioFile(input_file_name)); @@ -1270,10 +1269,9 @@ TEST_F(AcmSenderBitExactnessOldApi, External_Pcmu_20ms) { EXPECT_CALL(*mock_encoder, EncodeImpl(_, _, _)) .Times(AtLeast(1)) .WillRepeatedly(Invoke( - &encoder, - static_cast, webrtc::Buffer*)>( - &AudioEncoderPcmU::Encode))); + &encoder, static_cast, Buffer*)>( + &AudioEncoderPcmU::Encode))); ASSERT_TRUE(SetUpSender(kTestFileMono32kHz, 32000)); ASSERT_NO_FATAL_FAILURE( SetUpTestExternalEncoder(std::move(mock_encoder), config.payload_type)); @@ -1310,7 +1308,7 @@ class AcmSwitchingOutputFrequencyOldApi : public ::testing::Test, // Set up the receiver used to decode the packets and verify the decoded // output. const std::string output_file_name = - webrtc::test::OutputPath() + + test::OutputPath() + ::testing::UnitTest::GetInstance() ->current_test_info() ->test_case_name() + diff --git a/modules/audio_coding/codecs/cng/cng_unittest.cc b/modules/audio_coding/codecs/cng/cng_unittest.cc index 823506ea64..a486cc01b1 100644 --- a/modules/audio_coding/codecs/cng/cng_unittest.cc +++ b/modules/audio_coding/codecs/cng/cng_unittest.cc @@ -50,7 +50,7 @@ class CngDeathTest : public CngTest {}; void CngTest::SetUp() { FILE* input_file; const std::string file_name = - webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); + test::ResourcePath("audio_coding/testfile32kHz", "pcm"); input_file = fopen(file_name.c_str(), "rb"); ASSERT_TRUE(input_file != NULL); ASSERT_EQ(640, static_cast( diff --git a/modules/audio_coding/codecs/opus/audio_decoder_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_decoder_opus_unittest.cc index a9355dcfaa..56b846f8cf 100644 --- a/modules/audio_coding/codecs/opus/audio_decoder_opus_unittest.cc +++ b/modules/audio_coding/codecs/opus/audio_decoder_opus_unittest.cc @@ -43,8 +43,8 @@ namespace { using test::ExplicitKeyValueConfig; using testing::SizeIs; -using DecodeResult = ::webrtc::AudioDecoder::EncodedAudioFrame::DecodeResult; -using ParseResult = ::webrtc::AudioDecoder::ParseResult; +using DecodeResult = AudioDecoder::EncodedAudioFrame::DecodeResult; +using ParseResult = AudioDecoder::ParseResult; constexpr int kSampleRateHz = 48000; diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc index e3adc6ec71..1d777112c4 100644 --- a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc +++ b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc @@ -630,7 +630,7 @@ TEST(AudioEncoderOpusTest, TestConfigFromParams) { } TEST(AudioEncoderOpusTest, TestConfigFromInvalidParams) { - const webrtc::SdpAudioFormat format("opus", 48000, 2); + const SdpAudioFormat format("opus", 48000, 2); const auto default_config = *AudioEncoderOpus::SdpToConfig(format); #if WEBRTC_OPUS_SUPPORT_120MS_PTIME const std::vector default_supported_frame_lengths_ms({20, 40, 60, 120}); @@ -690,9 +690,8 @@ TEST(AudioEncoderOpusTest, GetFrameLenghtRange) { CreateConfigWithParameters({{"maxptime", "10"}, {"ptime", "10"}}); std::unique_ptr encoder = AudioEncoderOpus::MakeAudioEncoder( CreateEnvironment(), config, {.payload_type = kDefaultOpusPayloadType}); - auto ptime = webrtc::TimeDelta::Millis(10); - std::optional> range = { - {ptime, ptime}}; + auto ptime = TimeDelta::Millis(10); + std::optional> range = {{ptime, ptime}}; EXPECT_EQ(encoder->GetFrameLengthRange(), range); } @@ -785,7 +784,7 @@ TEST_P(AudioEncoderOpusTest, OpusFlagDtxAsNonSpeech) { // Open file containing speech and silence. const std::string kInputFileName = - webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); + test::ResourcePath("audio_coding/testfile32kHz", "pcm"); test::AudioLoop audio_loop; // Use the file as if it were sampled at our desired input rate. const size_t max_loop_length_samples = diff --git a/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc b/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc index 00a210fe78..12783a656d 100644 --- a/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc +++ b/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc @@ -139,7 +139,7 @@ TEST(BandwidthAdaptationTest, BandwidthAdaptationTest) { // Open speech file. const std::string kInputFileName = - webrtc::test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm"); + test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm"); test::AudioLoop audio_loop; EXPECT_EQ(kSampleRateHz, encoder->SampleRateHz()); ASSERT_TRUE(audio_loop.Init(kInputFileName, kMaxLoopLengthSamples, diff --git a/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc b/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc index 465cde0901..bf116813f6 100644 --- a/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc +++ b/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc @@ -28,9 +28,9 @@ namespace webrtc { namespace { -using ::webrtc::test::GetGlobalMetricsLogger; -using ::webrtc::test::ImprovementDirection; -using ::webrtc::test::Unit; +using test::GetGlobalMetricsLogger; +using test::ImprovementDirection; +using test::Unit; int64_t RunComplexityTest(const Environment& env, const AudioEncoderOpusConfig& config) { @@ -39,7 +39,7 @@ int64_t RunComplexityTest(const Environment& env, AudioEncoderOpus::MakeAudioEncoder(env, config, {.payload_type = 17}); // Open speech file. const std::string kInputFileName = - webrtc::test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm"); + test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm"); test::AudioLoop audio_loop; constexpr int kSampleRateHz = 48000; EXPECT_EQ(kSampleRateHz, encoder->SampleRateHz()); diff --git a/modules/audio_coding/codecs/opus/opus_unittest.cc b/modules/audio_coding/codecs/opus/opus_unittest.cc index 4c03506afb..674e50e0f4 100644 --- a/modules/audio_coding/codecs/opus/opus_unittest.cc +++ b/modules/audio_coding/codecs/opus/opus_unittest.cc @@ -184,7 +184,7 @@ void OpusTest::PrepareSpeechData(int block_length_ms, int loop_length_ms) { {4, "audio_coding/speech_4_channels_48k_one_second"}}; std::map channel_to_suffix = { {1, "pcm"}, {2, "pcm"}, {4, "wav"}}; - const std::string file_name = webrtc::test::ResourcePath( + const std::string file_name = test::ResourcePath( channel_to_basename[channels_], channel_to_suffix[channels_]); if (loop_length_ms < block_length_ms) { loop_length_ms = block_length_ms; diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc index ea2d6d3850..1ba411beb0 100644 --- a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc +++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc @@ -207,7 +207,7 @@ TEST_F(AudioEncoderCopyRedTest, CheckPayloadSizes1) { // Checks that the correct payload sizes are populated into the redundancy // information for a redundancy level of 0. TEST_F(AudioEncoderCopyRedTest, CheckPayloadSizes0) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( field_trials_, "WebRTC-Audio-Red-For-Opus/Enabled-0/"); // Recreate the RED encoder to take the new field trial setting into account. AudioEncoderCopyRed::Config config; @@ -233,7 +233,7 @@ TEST_F(AudioEncoderCopyRedTest, CheckPayloadSizes0) { // Checks that the correct payload sizes are populated into the redundancy // information for a redundancy level of 2. TEST_F(AudioEncoderCopyRedTest, CheckPayloadSizes2) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( field_trials_, "WebRTC-Audio-Red-For-Opus/Enabled-2/"); // Recreate the RED encoder to take the new field trial setting into account. AudioEncoderCopyRed::Config config; @@ -275,7 +275,7 @@ TEST_F(AudioEncoderCopyRedTest, CheckPayloadSizes2) { // Checks that the correct payload sizes are populated into the redundancy // information for a redundancy level of 3. TEST_F(AudioEncoderCopyRedTest, CheckPayloadSizes3) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( field_trials_, "WebRTC-Audio-Red-For-Opus/Enabled-3/"); // Recreate the RED encoder to take the new field trial setting into account. AudioEncoderCopyRed::Config config; @@ -489,7 +489,7 @@ TEST_F(AudioEncoderCopyRedTest, CheckRFC2198Header) { // Variant with a redundancy of 0. TEST_F(AudioEncoderCopyRedTest, CheckRFC2198Header0) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( field_trials_, "WebRTC-Audio-Red-For-Opus/Enabled-0/"); // Recreate the RED encoder to take the new field trial setting into account. AudioEncoderCopyRed::Config config; @@ -517,7 +517,7 @@ TEST_F(AudioEncoderCopyRedTest, CheckRFC2198Header0) { } // Variant with a redundancy of 2. TEST_F(AudioEncoderCopyRedTest, CheckRFC2198Header2) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( field_trials_, "WebRTC-Audio-Red-For-Opus/Enabled-2/"); // Recreate the RED encoder to take the new field trial setting into account. AudioEncoderCopyRed::Config config; diff --git a/modules/audio_coding/neteq/audio_decoder_unittest.cc b/modules/audio_coding/neteq/audio_decoder_unittest.cc index 9f29394271..400035353b 100644 --- a/modules/audio_coding/neteq/audio_decoder_unittest.cc +++ b/modules/audio_coding/neteq/audio_decoder_unittest.cc @@ -103,9 +103,8 @@ double MseInputOutput(const std::vector& input, class AudioDecoderTest : public ::testing::Test { protected: AudioDecoderTest() - : input_audio_( - webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"), - 32000), + : input_audio_(test::ResourcePath("audio_coding/testfile32kHz", "pcm"), + 32000), codec_input_rate_hz_(32000), // Legacy default value. frame_size_(0), data_length_(0), diff --git a/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc b/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc index cb2a9d37b2..6417cf45e7 100644 --- a/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc +++ b/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc @@ -212,7 +212,7 @@ TestStatistics RunTest(int loss_cadence, NetEqTest::DecoderMap decoders; // Using a fake decoder which simply reads the output audio from a file. auto input_file = std::make_unique( - webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm")); + test::ResourcePath("audio_coding/testfile32kHz", "pcm")); AudioDecoderPlc dec(std::move(input_file), kSampleRateHz); // Masquerading as a PCM16b decoder. decoders.emplace(kPayloadType, SdpAudioFormat("l16", 32000, 1)); diff --git a/modules/audio_coding/neteq/neteq_stereo_unittest.cc b/modules/audio_coding/neteq/neteq_stereo_unittest.cc index 5e13d73cb4..6cab630474 100644 --- a/modules/audio_coding/neteq/neteq_stereo_unittest.cc +++ b/modules/audio_coding/neteq/neteq_stereo_unittest.cc @@ -100,7 +100,7 @@ class NetEqStereoTest : public ::testing::TestWithParam { virtual void SetUp() { const std::string file_name = - webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); + test::ResourcePath("audio_coding/testfile32kHz", "pcm"); input_file_.reset(new test::InputAudioFile(file_name)); RTC_CHECK_GE(num_channels_, 2); ASSERT_TRUE(neteq_mono_->RegisterPayloadType( diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc index ae9ca81234..2b1d28302f 100644 --- a/modules/audio_coding/neteq/neteq_unittest.cc +++ b/modules/audio_coding/neteq/neteq_unittest.cc @@ -51,7 +51,7 @@ namespace webrtc { // TODO(bugs.webrtc.org/345525069): Either fix/enable or remove. TEST_F(NetEqDecodingTest, DISABLED_TestBitExactness) { const std::string input_rtp_file = - webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp"); + test::ResourcePath("audio_coding/neteq_universal_new", "rtp"); const std::string output_checksum = "dee7a10ab92526876a70a85bc48a4906901af3df"; @@ -72,7 +72,7 @@ TEST_F(NetEqDecodingTest, DISABLED_TestBitExactness) { #endif TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) { const std::string input_rtp_file = - webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp"); + test::ResourcePath("audio_coding/neteq_opus", "rtp"); const std::string output_checksum = "434bdc4ec08546510ee903d001c8be1a01c44e24|" @@ -96,7 +96,7 @@ TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) { #endif TEST_F(NetEqDecodingTest, MAYBE_TestOpusDtxBitExactness) { const std::string input_rtp_file = - webrtc::test::ResourcePath("audio_coding/neteq_opus_dtx", "rtp"); + test::ResourcePath("audio_coding/neteq_opus_dtx", "rtp"); const std::string output_checksum = "7eddce841cbfa500964c91cdae78b01b9f448948|" @@ -322,10 +322,10 @@ class NetEqBgnTest : public NetEqDecodingTest { // We are using the same 32 kHz input file for all tests, regardless of // `sampling_rate_hz`. The output may sound weird, but the test is still // valid. - ASSERT_TRUE(input.Init( - webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"), - 10 * sampling_rate_hz, // Max 10 seconds loop length. - expected_samples_per_channel)); + ASSERT_TRUE( + input.Init(test::ResourcePath("audio_coding/testfile32kHz", "pcm"), + 10 * sampling_rate_hz, // Max 10 seconds loop length. + expected_samples_per_channel)); // Payload of 10 ms of PCM16 32 kHz. uint8_t payload[kBlockSize32kHz * sizeof(int16_t)]; @@ -1001,7 +1001,7 @@ TEST(NetEqNoTimeStretchingMode, RunTest) { {7, kRtpExtensionVideoContentType}, {8, kRtpExtensionVideoTiming}}; std::unique_ptr input = CreateNetEqRtpDumpInput( - webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp"), + test::ResourcePath("audio_coding/neteq_universal_new", "rtp"), rtp_ext_map, std::nullopt /*No SSRC filter*/); std::unique_ptr input_time_limit( new TimeLimitedNetEqInput(std::move(input), 20000)); diff --git a/modules/audio_coding/neteq/test/neteq_decoding_test.cc b/modules/audio_coding/neteq/test/neteq_decoding_test.cc index d67888be14..eb0b196499 100644 --- a/modules/audio_coding/neteq/test/neteq_decoding_test.cc +++ b/modules/audio_coding/neteq/test/neteq_decoding_test.cc @@ -46,7 +46,7 @@ namespace webrtc { namespace { -void LoadDecoders(webrtc::NetEq* neteq) { +void LoadDecoders(NetEq* neteq) { ASSERT_EQ(true, neteq->RegisterPayloadType(0, SdpAudioFormat("pcmu", 8000, 1))); ASSERT_EQ(true, @@ -150,11 +150,11 @@ void NetEqDecodingTest::DecodeAndCompare( OpenInputFile(rtp_file); std::string ref_out_file = - gen_ref ? webrtc::test::OutputPath() + "neteq_universal_ref.pcm" : ""; + gen_ref ? test::OutputPath() + "neteq_universal_ref.pcm" : ""; ResultSink output(ref_out_file); std::string stat_out_file = - gen_ref ? webrtc::test::OutputPath() + "neteq_network_stats.dat" : ""; + gen_ref ? test::OutputPath() + "neteq_network_stats.dat" : ""; ResultSink network_stats(stat_out_file); packet_ = rtp_source_->NextPacket(); diff --git a/modules/audio_coding/neteq/test/neteq_performance_unittest.cc b/modules/audio_coding/neteq/test/neteq_performance_unittest.cc index ad2d42f18c..6664137a8c 100644 --- a/modules/audio_coding/neteq/test/neteq_performance_unittest.cc +++ b/modules/audio_coding/neteq/test/neteq_performance_unittest.cc @@ -20,9 +20,9 @@ namespace webrtc { namespace { -using ::webrtc::test::GetGlobalMetricsLogger; -using ::webrtc::test::ImprovementDirection; -using ::webrtc::test::Unit; +using test::GetGlobalMetricsLogger; +using test::ImprovementDirection; +using test::Unit; // Runs a test with 10% packet losses and 10% clock drift, to exercise // both loss concealment and time-stretching code. diff --git a/modules/audio_coding/neteq/test/result_sink.cc b/modules/audio_coding/neteq/test/result_sink.cc index 98683b1d8d..72dfc318a2 100644 --- a/modules/audio_coding/neteq/test/result_sink.cc +++ b/modules/audio_coding/neteq/test/result_sink.cc @@ -34,8 +34,8 @@ namespace webrtc { #ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT -void Convert(const webrtc::NetEqNetworkStatistics& stats_raw, - webrtc::neteq_unittest::NetEqNetworkStatistics* stats) { +void Convert(const NetEqNetworkStatistics& stats_raw, + neteq_unittest::NetEqNetworkStatistics* stats) { stats->set_current_buffer_size_ms(stats_raw.current_buffer_size_ms); stats->set_preferred_buffer_size_ms(stats_raw.preferred_buffer_size_ms); stats->set_jitter_peaks_found(stats_raw.jitter_peaks_found); diff --git a/modules/audio_coding/neteq/tools/neteq_performance_test.cc b/modules/audio_coding/neteq/tools/neteq_performance_test.cc index caae9ecc26..1b4b259014 100644 --- a/modules/audio_coding/neteq/tools/neteq_performance_test.cc +++ b/modules/audio_coding/neteq/tools/neteq_performance_test.cc @@ -42,7 +42,7 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms, int lossrate, double drift_factor) { const std::string kInputFileName = - webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); + test::ResourcePath("audio_coding/testfile32kHz", "pcm"); const int kSampRateHz = 32000; const std::string kDecoderName = "pcm16-swb32"; const int kPayloadType = 95; diff --git a/modules/audio_coding/neteq/tools/neteq_quality_test.cc b/modules/audio_coding/neteq/tools/neteq_quality_test.cc index 55ef65e6b4..161914995a 100644 --- a/modules/audio_coding/neteq/tools/neteq_quality_test.cc +++ b/modules/audio_coding/neteq/tools/neteq_quality_test.cc @@ -113,13 +113,12 @@ const std::string& GetInFilenamePath(absl::string_view file_name) { std::vector name_parts = split(file_name, '.'); RTC_CHECK_EQ(name_parts.size(), 2); static const std::string path = - ::webrtc::test::ResourcePath(name_parts[0], name_parts[1]); + test::ResourcePath(name_parts[0], name_parts[1]); return path; } const std::string& GetOutFilenamePath(absl::string_view file_name) { - static const std::string path = - ::webrtc::test::OutputPath() + std::string(file_name); + static const std::string path = test::OutputPath() + std::string(file_name); return path; } @@ -261,10 +260,10 @@ NetEqQualityTest::NetEqQualityTest( out_filename.substr(out_filename.size() - 4) == ".wav") { // Open a wav file. output_.reset( - new webrtc::test::OutputWavFile(out_filename, 1000 * out_sampling_khz)); + new test::OutputWavFile(out_filename, 1000 * out_sampling_khz)); } else { // Open a pcm file. - output_.reset(new webrtc::test::OutputAudioFile(out_filename)); + output_.reset(new test::OutputAudioFile(out_filename)); } NetEq::Config config; diff --git a/modules/audio_coding/neteq/tools/neteq_test_factory.cc b/modules/audio_coding/neteq/tools/neteq_test_factory.cc index 827ed92931..857c633859 100644 --- a/modules/audio_coding/neteq/tools/neteq_test_factory.cc +++ b/modules/audio_coding/neteq/tools/neteq_test_factory.cc @@ -60,9 +60,8 @@ namespace webrtc { namespace test { namespace { -std::optional CodecSampleRate( - uint8_t payload_type, - webrtc::test::NetEqTestFactory::Config config) { +std::optional CodecSampleRate(uint8_t payload_type, + test::NetEqTestFactory::Config config) { if (payload_type == config.pcmu || payload_type == config.pcma || payload_type == config.pcm16b || payload_type == config.cn_nb || payload_type == config.avt) diff --git a/modules/audio_coding/neteq/tools/packet.cc b/modules/audio_coding/neteq/tools/packet.cc index f8deb94f92..3e9ef26f66 100644 --- a/modules/audio_coding/neteq/tools/packet.cc +++ b/modules/audio_coding/neteq/tools/packet.cc @@ -95,7 +95,7 @@ void Packet::DeleteRedHeaders(std::list* headers) { bool Packet::ParseHeader(const RtpHeaderExtensionMap* extension_map) { // Use RtpPacketReceived instead of RtpPacket because former already has a // converter into legacy RTPHeader. - webrtc::RtpPacketReceived rtp_packet(extension_map); + RtpPacketReceived rtp_packet(extension_map); // Because of the special case of dummy packets that have padding marked in // the RTP header, but do not have rtp payload with the padding size, handle diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc index f1a650566d..1de3da55bb 100644 --- a/modules/audio_coding/test/EncodeDecodeTest.cc +++ b/modules/audio_coding/test/EncodeDecodeTest.cc @@ -71,7 +71,7 @@ void Sender::Setup(const Environment& env, int payload_type, SdpAudioFormat format) { // Open input file - const std::string file_name = webrtc::test::ResourcePath(in_file_name, "pcm"); + const std::string file_name = test::ResourcePath(in_file_name, "pcm"); _pcmFile.Open(file_name, in_sample_rate, "rb"); if (format.num_channels == 2) { _pcmFile.ReadStereo(true); @@ -146,8 +146,7 @@ void Receiver::Setup(NetEq* neteq, int playSampFreq; std::string file_name; StringBuilder file_stream; - file_stream << webrtc::test::OutputPath() << out_file_name << file_num - << ".pcm"; + file_stream << test::OutputPath() << out_file_name << file_num << ".pcm"; file_name = file_stream.str(); _rtpStream = rtpStream; @@ -259,8 +258,8 @@ void EncodeDecodeTest::Perform() { RTPFile rtpFile; std::unique_ptr acm(AudioCodingModule::Create()); - std::string fileName = webrtc::test::TempFilename( - webrtc::test::OutputPath(), "encode_decode_rtp"); + std::string fileName = + test::TempFilename(test::OutputPath(), "encode_decode_rtp"); rtpFile.Open(fileName.c_str(), "wb+"); rtpFile.WriteHeader(); Sender sender; diff --git a/modules/audio_coding/test/PacketLossTest.cc b/modules/audio_coding/test/PacketLossTest.cc index dc8207e00d..59b5309c80 100644 --- a/modules/audio_coding/test/PacketLossTest.cc +++ b/modules/audio_coding/test/PacketLossTest.cc @@ -159,8 +159,8 @@ void PacketLossTest::Perform() { send_format.parameters = {{"stereo", "1"}}; } - std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(), - "packet_loss_test"); + std::string fileName = + test::TempFilename(test::OutputPath(), "packet_loss_test"); rtpFile.Open(fileName.c_str(), "wb+"); rtpFile.WriteHeader(); SenderWithFEC sender; diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc index 3820098a84..786ae2b552 100644 --- a/modules/audio_coding/test/TestAllCodecs.cc +++ b/modules/audio_coding/test/TestAllCodecs.cc @@ -135,7 +135,7 @@ TestAllCodecs::~TestAllCodecs() { void TestAllCodecs::Perform() { const std::string file_name = - webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); + test::ResourcePath("audio_coding/testfile32kHz", "pcm"); infile_a_.Open(file_name, 32000, "rb"); neteq_->SetCodecs({{107, {"L16", 8000, 1}}, @@ -378,7 +378,7 @@ void TestAllCodecs::Run(TestPack* channel) { } void TestAllCodecs::OpenOutFile(int test_number) { - std::string filename = webrtc::test::OutputPath(); + std::string filename = test::OutputPath(); StringBuilder test_number_str; test_number_str << test_number; filename += "testallcodecs_out_"; diff --git a/modules/audio_coding/test/TestRedFec.cc b/modules/audio_coding/test/TestRedFec.cc index 404e52c10d..7d93fee2d4 100644 --- a/modules/audio_coding/test/TestRedFec.cc +++ b/modules/audio_coding/test/TestRedFec.cc @@ -72,7 +72,7 @@ TestRedFec::~TestRedFec() { void TestRedFec::Perform() { const std::string file_name = - webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); + test::ResourcePath("audio_coding/testfile32kHz", "pcm"); _inFileA.Open(file_name, 32000, "rb"); // Create and connect the channel @@ -204,7 +204,7 @@ void TestRedFec::Run() { void TestRedFec::OpenOutFile(int16_t test_number) { std::string file_name; StringBuilder file_stream; - file_stream << webrtc::test::OutputPath(); + file_stream << test::OutputPath(); file_stream << "TestRedFec_outFile_"; file_stream << test_number << ".pcm"; file_name = file_stream.str(); diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc index f7206c3eaf..8625276cc2 100644 --- a/modules/audio_coding/test/TestStereo.cc +++ b/modules/audio_coding/test/TestStereo.cc @@ -140,9 +140,9 @@ void TestStereo::Perform() { // Open both mono and stereo test files in 32 kHz. const std::string file_name_stereo = - webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"); + test::ResourcePath("audio_coding/teststereo32kHz", "pcm"); const std::string file_name_mono = - webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); + test::ResourcePath("audio_coding/testfile32kHz", "pcm"); frequency_hz = 32000; in_file_stereo_ = new PCMFile(); in_file_mono_ = new PCMFile(); @@ -618,7 +618,7 @@ void TestStereo::Run(TestPackStereo* channel, void TestStereo::OpenOutFile(int16_t test_number) { std::string file_name; StringBuilder file_stream; - file_stream << webrtc::test::OutputPath() << "teststereo_out_" << test_number + file_stream << test::OutputPath() << "teststereo_out_" << test_number << ".pcm"; file_name = file_stream.str(); out_file_.Open(file_name, 32000, "wb"); diff --git a/modules/audio_coding/test/TestVADDTX.cc b/modules/audio_coding/test/TestVADDTX.cc index 38df0d12f3..ee0fb79f5c 100644 --- a/modules/audio_coding/test/TestVADDTX.cc +++ b/modules/audio_coding/test/TestVADDTX.cc @@ -222,9 +222,9 @@ void TestWebRtcVadDtx::Test(bool new_outfile, bool expect_dtx_enabled) { output_file_num_++; } StringBuilder out_filename; - out_filename << webrtc::test::OutputPath() << "testWebRtcVadDtx_outFile_" + out_filename << test::OutputPath() << "testWebRtcVadDtx_outFile_" << output_file_num_ << ".pcm"; - Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"), 32000, 1, + Run(test::ResourcePath("audio_coding/testfile32kHz", "pcm"), 32000, 1, out_filename.str(), !new_outfile, expects); } @@ -234,13 +234,13 @@ void TestOpusDtx::Perform() { // Register Opus as send codec std::string out_filename = - webrtc::test::OutputPath() + "testOpusDtx_outFile_mono.pcm"; + test::OutputPath() + "testOpusDtx_outFile_mono.pcm"; RegisterCodec({"opus", 48000, 2}, std::nullopt); acm_send_->ModifyEncoder([](std::unique_ptr* encoder_ptr) { (*encoder_ptr)->SetDtx(false); }); - Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"), 32000, 1, + Run(test::ResourcePath("audio_coding/testfile32kHz", "pcm"), 32000, 1, out_filename, false, expects); acm_send_->ModifyEncoder([](std::unique_ptr* encoder_ptr) { @@ -248,7 +248,7 @@ void TestOpusDtx::Perform() { }); expects[static_cast(AudioFrameType::kEmptyFrame)] = 1; expects[static_cast(AudioFrameType::kAudioFrameCN)] = 1; - Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"), 32000, 1, + Run(test::ResourcePath("audio_coding/testfile32kHz", "pcm"), 32000, 1, out_filename, true, expects); } diff --git a/modules/audio_coding/test/opus_test.cc b/modules/audio_coding/test/opus_test.cc index d6323fdaf1..8bde843e8e 100644 --- a/modules/audio_coding/test/opus_test.cc +++ b/modules/audio_coding/test/opus_test.cc @@ -73,9 +73,9 @@ void OpusTest::Perform() { // Open both mono and stereo test files in 32 kHz. const std::string file_name_stereo = - webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"); + test::ResourcePath("audio_coding/teststereo32kHz", "pcm"); const std::string file_name_mono = - webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); + test::ResourcePath("audio_coding/testfile32kHz", "pcm"); frequency_hz = 32000; in_file_stereo_.Open(file_name_stereo, frequency_hz, "rb"); in_file_stereo_.ReadStereo(true); @@ -396,14 +396,13 @@ void OpusTest::Run(TestPackStereo* channel, void OpusTest::OpenOutFile(int test_number) { std::string file_name; std::stringstream file_stream; - file_stream << webrtc::test::OutputPath() << "opustest_out_" << test_number - << ".pcm"; + file_stream << test::OutputPath() << "opustest_out_" << test_number << ".pcm"; file_name = file_stream.str(); out_file_.Open(file_name, 48000, "wb"); file_stream.str(""); file_name = file_stream.str(); - file_stream << webrtc::test::OutputPath() << "opusstandalone_out_" - << test_number << ".pcm"; + file_stream << test::OutputPath() << "opusstandalone_out_" << test_number + << ".pcm"; file_name = file_stream.str(); out_file_standalone_.Open(file_name, 48000, "wb"); } From 9cdfd8558e7b87a5dcf3e75dd3db9ef41749322a Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Fri, 9 May 2025 10:37:27 +0000 Subject: [PATCH 093/292] [cleanup] Fix redundant webrtc name specifier This CL was uploaded by git cl split. R=hta@webrtc.org Bug: webrtc:42232595 Change-Id: If6bbf2538f0a95088b06d7d216e59c0dc7b3eb7b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390605 Commit-Queue: Harald Alvestrand Auto-Submit: Evan Shrubsole Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44565} --- modules/desktop_capture/screen_capturer_unittest.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/modules/desktop_capture/screen_capturer_unittest.cc b/modules/desktop_capture/screen_capturer_unittest.cc index 4654000a82..2cd8184e6b 100644 --- a/modules/desktop_capture/screen_capturer_unittest.cc +++ b/modules/desktop_capture/screen_capturer_unittest.cc @@ -101,7 +101,7 @@ ACTION_P(SaveUniquePtrArg, dest) { #define MAYBE_GetScreenListAndSelectScreen GetScreenListAndSelectScreen #endif TEST_F(ScreenCapturerTest, MAYBE_GetScreenListAndSelectScreen) { - webrtc::DesktopCapturer::SourceList screens; + DesktopCapturer::SourceList screens; EXPECT_TRUE(capturer_->GetSourceList(&screens)); for (const auto& screen : screens) { EXPECT_TRUE(capturer_->SelectSource(screen.id)); From 989ba456168e9a1414eaf31052f5268b09a3554e Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Fri, 9 May 2025 10:37:43 +0000 Subject: [PATCH 094/292] [cleanup] Fix redundant webrtc name specifier This CL was uploaded by git cl split. R=hta@webrtc.org Bug: webrtc:42232595 Change-Id: I0852d75dbab3768d3f508a7c37715eb0bd4cdc04 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390641 Reviewed-by: Harald Alvestrand Commit-Queue: Harald Alvestrand Auto-Submit: Evan Shrubsole Cr-Commit-Position: refs/heads/main@{#44566} --- modules/pacing/pacing_controller_unittest.cc | 6 +-- modules/rtp_rtcp/source/fec_test_helper.cc | 6 +-- .../frame_transformer_factory_unittest.cc | 2 +- modules/rtp_rtcp/source/nack_rtx_unittest.cc | 2 +- .../source/receive_statistics_unittest.cc | 2 +- .../source/rtcp_packet/app_unittest.cc | 2 +- .../rtcp_packet/loss_notification_unittest.cc | 2 +- .../source/rtcp_packet/nack_unittest.cc | 2 +- .../rtp_rtcp/source/rtcp_receiver_unittest.cc | 2 +- .../source/rtcp_transceiver_impl_unittest.cc | 38 +++++++++---------- .../source/rtp_sender_audio_unittest.cc | 2 +- modules/rtp_rtcp/source/rtp_sender_video.cc | 6 +-- .../source/rtp_sender_video_unittest.cc | 5 +-- modules/rtp_rtcp/test/testFec/test_fec.cc | 2 +- 14 files changed, 39 insertions(+), 40 deletions(-) diff --git a/modules/pacing/pacing_controller_unittest.cc b/modules/pacing/pacing_controller_unittest.cc index 2b755ff507..a51e7047c1 100644 --- a/modules/pacing/pacing_controller_unittest.cc +++ b/modules/pacing/pacing_controller_unittest.cc @@ -140,7 +140,7 @@ class MockPacingControllerCallback : public PacingController::PacketSender { MOCK_METHOD(size_t, SendPadding, (size_t target_size)); MOCK_METHOD(void, OnAbortedRetransmissions, - (uint32_t, webrtc::ArrayView), + (uint32_t, ArrayView), (override)); MOCK_METHOD(std::optional, GetRtxSsrcForMedia, @@ -168,7 +168,7 @@ class MockPacketSender : public PacingController::PacketSender { (override)); MOCK_METHOD(void, OnAbortedRetransmissions, - (uint32_t, webrtc::ArrayView), + (uint32_t, ArrayView), (override)); MOCK_METHOD(std::optional, GetRtxSsrcForMedia, @@ -304,7 +304,7 @@ class PacingControllerTest : public ::testing::Test { type == RtpPacketMediaType::kRetransmission, false)); } - void AdvanceTimeUntil(webrtc::Timestamp time) { + void AdvanceTimeUntil(Timestamp time) { Timestamp now = clock_.CurrentTime(); clock_.AdvanceTime(std::max(TimeDelta::Zero(), time - now)); } diff --git a/modules/rtp_rtcp/source/fec_test_helper.cc b/modules/rtp_rtcp/source/fec_test_helper.cc index 3186a476b6..aa07e6b293 100644 --- a/modules/rtp_rtcp/source/fec_test_helper.cc +++ b/modules/rtp_rtcp/source/fec_test_helper.cc @@ -84,9 +84,9 @@ ForwardErrorCorrection::PacketList MediaPacketGenerator::ConstructMediaPackets( // Only push one (fake) frame to the FEC. data[1] &= 0x7f; - webrtc::ByteWriter::WriteBigEndian(&data[2], seq_num); - webrtc::ByteWriter::WriteBigEndian(&data[4], time_stamp); - webrtc::ByteWriter::WriteBigEndian(&data[8], ssrc_); + ByteWriter::WriteBigEndian(&data[2], seq_num); + ByteWriter::WriteBigEndian(&data[4], time_stamp); + ByteWriter::WriteBigEndian(&data[8], ssrc_); // Generate random values for payload. for (size_t j = 12; j < media_packet->data.size(); ++j) diff --git a/modules/rtp_rtcp/source/frame_transformer_factory_unittest.cc b/modules/rtp_rtcp/source/frame_transformer_factory_unittest.cc index 312bdcbb86..03a3608037 100644 --- a/modules/rtp_rtcp/source/frame_transformer_factory_unittest.cc +++ b/modules/rtp_rtcp/source/frame_transformer_factory_unittest.cc @@ -48,7 +48,7 @@ TEST(FrameTransformerFactory, CloneVideoFrame) { std::fill_n(data, 10, 5); ArrayView data_view(data); EXPECT_CALL(original_frame, GetData()).WillRepeatedly(Return(data_view)); - webrtc::VideoFrameMetadata metadata; + VideoFrameMetadata metadata; std::vector csrcs{123, 321}; // Copy csrcs rather than moving so we can compare in an EXPECT_EQ later. metadata.SetCsrcs(csrcs); diff --git a/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/modules/rtp_rtcp/source/nack_rtx_unittest.cc index ea0a21a6b9..90e75a3f45 100644 --- a/modules/rtp_rtcp/source/nack_rtx_unittest.cc +++ b/modules/rtp_rtcp/source/nack_rtx_unittest.cc @@ -67,7 +67,7 @@ class VerifyingMediaStream : public RtpPacketSinkInterface { std::list sequence_numbers_; }; -class RtxLoopBackTransport : public webrtc::Transport { +class RtxLoopBackTransport : public Transport { public: explicit RtxLoopBackTransport(uint32_t rtx_ssrc) : count_(0), diff --git a/modules/rtp_rtcp/source/receive_statistics_unittest.cc b/modules/rtp_rtcp/source/receive_statistics_unittest.cc index 918c74de7f..492eaae29d 100644 --- a/modules/rtp_rtcp/source/receive_statistics_unittest.cc +++ b/modules/rtp_rtcp/source/receive_statistics_unittest.cc @@ -631,7 +631,7 @@ TEST_P(ReceiveStatisticsTest, SimpleJitterComputation) { // See jitter caluculation in https://www.rfc-editor.org/rfc/rfc3550 6.4.1. const uint32_t expected_jitter = (kLateArrivalDeltaSamples) / 16; EXPECT_EQ(expected_jitter, statistician->GetStats().jitter); - EXPECT_EQ(webrtc::TimeDelta::Seconds(expected_jitter) / kCodecSampleRate, + EXPECT_EQ(TimeDelta::Seconds(expected_jitter) / kCodecSampleRate, statistician->GetStats().interarrival_jitter); } diff --git a/modules/rtp_rtcp/source/rtcp_packet/app_unittest.cc b/modules/rtp_rtcp/source/rtcp_packet/app_unittest.cc index 4448f346fa..384f1c03f5 100644 --- a/modules/rtp_rtcp/source/rtcp_packet/app_unittest.cc +++ b/modules/rtp_rtcp/source/rtcp_packet/app_unittest.cc @@ -20,9 +20,9 @@ namespace webrtc { namespace { +using rtcp::App; using ::testing::ElementsAreArray; using ::testing::make_tuple; -using ::webrtc::rtcp::App; constexpr uint32_t kName = ((uint32_t)'n' << 24) | ((uint32_t)'a' << 16) | ((uint32_t)'m' << 8) | (uint32_t)'e'; diff --git a/modules/rtp_rtcp/source/rtcp_packet/loss_notification_unittest.cc b/modules/rtp_rtcp/source/rtcp_packet/loss_notification_unittest.cc index b7b9c54acf..312f051554 100644 --- a/modules/rtp_rtcp/source/rtcp_packet/loss_notification_unittest.cc +++ b/modules/rtp_rtcp/source/rtcp_packet/loss_notification_unittest.cc @@ -21,9 +21,9 @@ namespace webrtc { +using rtcp::LossNotification; using ::testing::ElementsAreArray; using ::testing::make_tuple; -using ::webrtc::rtcp::LossNotification; TEST(RtcpPacketLossNotificationTest, SetWithIllegalValuesFails) { constexpr uint16_t kLastDecoded = 0x3c7b; diff --git a/modules/rtp_rtcp/source/rtcp_packet/nack_unittest.cc b/modules/rtp_rtcp/source/rtcp_packet/nack_unittest.cc index de7b942ddf..a41700b577 100644 --- a/modules/rtp_rtcp/source/rtcp_packet/nack_unittest.cc +++ b/modules/rtp_rtcp/source/rtcp_packet/nack_unittest.cc @@ -23,6 +23,7 @@ namespace webrtc { namespace { +using rtcp::Nack; using ::testing::_; using ::testing::ElementsAre; using ::testing::ElementsAreArray; @@ -30,7 +31,6 @@ using ::testing::Invoke; using ::testing::make_tuple; using ::testing::MockFunction; using ::testing::UnorderedElementsAreArray; -using ::webrtc::rtcp::Nack; constexpr uint32_t kSenderSsrc = 0x12345678; constexpr uint32_t kRemoteSsrc = 0x23456789; diff --git a/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc index 4da2dceeb8..31d218eea2 100644 --- a/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc +++ b/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc @@ -68,6 +68,7 @@ namespace webrtc { namespace { using rtcp::ReceiveTimeInfo; +using test::ExplicitKeyValueConfig; using ::testing::_; using ::testing::AllOf; using ::testing::ElementsAre; @@ -83,7 +84,6 @@ using ::testing::SizeIs; using ::testing::StrEq; using ::testing::StrictMock; using ::testing::UnorderedElementsAre; -using ::webrtc::test::ExplicitKeyValueConfig; class MockRtcpPacketTypeCounterObserver : public RtcpPacketTypeCounterObserver { public: diff --git a/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc b/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc index 3a10304b4b..be485b87da 100644 --- a/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc +++ b/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc @@ -59,6 +59,11 @@ namespace webrtc { namespace { +using rtcp::Bye; +using rtcp::CompoundPacket; +using rtcp::ReportBlock; +using rtcp::SenderReport; +using test::RtcpPacketParser; using ::testing::_; using ::testing::ElementsAre; using ::testing::ElementsAreArray; @@ -71,11 +76,6 @@ using ::testing::SizeIs; using ::testing::StrictMock; using ::testing::UnorderedElementsAre; using ::testing::WithArg; -using ::webrtc::rtcp::Bye; -using ::webrtc::rtcp::CompoundPacket; -using ::webrtc::rtcp::ReportBlock; -using ::webrtc::rtcp::SenderReport; -using ::webrtc::test::RtcpPacketParser; class MockReceiveStatisticsProvider : public ReceiveStatisticsProvider { public: @@ -432,7 +432,7 @@ TEST_F(RtcpTransceiverImplTest, AvoidsEmptyPacketsInReducedMode) { RtcpTransceiverConfig config = DefaultTestConfig(); config.rtcp_transport = transport.AsStdFunction(); - config.rtcp_mode = webrtc::RtcpMode::kReducedSize; + config.rtcp_mode = RtcpMode::kReducedSize; config.receive_statistics = &receive_statistics; RtcpTransceiverImpl rtcp_transceiver(config); @@ -445,7 +445,7 @@ TEST_F(RtcpTransceiverImplTest, AvoidsEmptyReceiverReportsInReducedMode) { RtcpTransceiverConfig config = DefaultTestConfig(); config.rtcp_transport = RtcpParserTransport(rtcp_parser); - config.rtcp_mode = webrtc::RtcpMode::kReducedSize; + config.rtcp_mode = RtcpMode::kReducedSize; config.receive_statistics = &receive_statistics; // Set it to produce something (RRTR) in the "periodic" rtcp packets. config.non_sender_rtt_measurement = true; @@ -545,7 +545,7 @@ TEST_F(RtcpTransceiverImplTest, const uint32_t kSenderSsrc = 12345; RtcpTransceiverConfig config = DefaultTestConfig(); config.send_remb_on_change = true; - config.rtcp_mode = webrtc::RtcpMode::kReducedSize; + config.rtcp_mode = RtcpMode::kReducedSize; config.feedback_ssrc = kSenderSsrc; RtcpPacketParser rtcp_parser; config.rtcp_transport = RtcpParserTransport(rtcp_parser); @@ -717,12 +717,12 @@ TEST_F(RtcpTransceiverImplTest, CallsObserverOnTargetBitrateBySenderSsrc) { rtcp_transceiver.AddMediaReceiverRtcpObserver(kRemoteSsrc1, &observer1); rtcp_transceiver.AddMediaReceiverRtcpObserver(kRemoteSsrc2, &observer2); - webrtc::rtcp::TargetBitrate target_bitrate; + rtcp::TargetBitrate target_bitrate; target_bitrate.AddTargetBitrate(0, 0, /*target_bitrate_kbps=*/10); target_bitrate.AddTargetBitrate(0, 1, /*target_bitrate_kbps=*/20); target_bitrate.AddTargetBitrate(1, 0, /*target_bitrate_kbps=*/40); target_bitrate.AddTargetBitrate(1, 1, /*target_bitrate_kbps=*/80); - webrtc::rtcp::ExtendedReports xr; + rtcp::ExtendedReports xr; xr.SetSenderSsrc(kRemoteSsrc1); xr.SetTargetBitrate(target_bitrate); auto raw_packet = xr.Build(); @@ -744,12 +744,12 @@ TEST_F(RtcpTransceiverImplTest, SkipsIncorrectTargetBitrateEntries) { RtcpTransceiverImpl rtcp_transceiver(config); rtcp_transceiver.AddMediaReceiverRtcpObserver(kRemoteSsrc, &observer); - webrtc::rtcp::TargetBitrate target_bitrate; + rtcp::TargetBitrate target_bitrate; target_bitrate.AddTargetBitrate(0, 0, /*target_bitrate_kbps=*/10); - target_bitrate.AddTargetBitrate(0, webrtc::kMaxTemporalStreams, 20); - target_bitrate.AddTargetBitrate(webrtc::kMaxSpatialLayers, 0, 40); + target_bitrate.AddTargetBitrate(0, kMaxTemporalStreams, 20); + target_bitrate.AddTargetBitrate(kMaxSpatialLayers, 0, 40); - webrtc::rtcp::ExtendedReports xr; + rtcp::ExtendedReports xr; xr.SetTargetBitrate(target_bitrate); xr.SetSenderSsrc(kRemoteSsrc); auto raw_packet = xr.Build(); @@ -790,7 +790,7 @@ TEST_F(RtcpTransceiverImplTest, CallsObserverOnByeBehindUnknownRtcpPacket) { CompoundPacket compound; // Use Application-Defined rtcp packet as unknown. - auto app = std::make_unique(); + auto app = std::make_unique(); compound.Append(std::move(app)); auto bye = std::make_unique(); bye->SetSenderSsrc(kRemoteSsrc); @@ -1180,7 +1180,7 @@ TEST_F(RtcpTransceiverImplTest, KeyFrameRequestCreatesCompoundPacket) { RtcpPacketParser rtcp_parser; config.rtcp_transport = RtcpParserTransport(rtcp_parser); - config.rtcp_mode = webrtc::RtcpMode::kCompound; + config.rtcp_mode = RtcpMode::kCompound; RtcpTransceiverImpl rtcp_transceiver(config); rtcp_transceiver.SendFullIntraRequest(kRemoteSsrcs, true); @@ -1198,7 +1198,7 @@ TEST_F(RtcpTransceiverImplTest, KeyFrameRequestCreatesReducedSizePacket) { RtcpPacketParser rtcp_parser; config.rtcp_transport = RtcpParserTransport(rtcp_parser); - config.rtcp_mode = webrtc::RtcpMode::kReducedSize; + config.rtcp_mode = RtcpMode::kReducedSize; RtcpTransceiverImpl rtcp_transceiver(config); rtcp_transceiver.SendFullIntraRequest(kRemoteSsrcs, true); @@ -1393,10 +1393,10 @@ TEST_F(RtcpTransceiverImplTest, IgnoresUnknownSsrcInDlrr) { RtcpTransceiverImpl rtcp_transceiver(config); Timestamp time = Timestamp::Micros(12345678); - webrtc::rtcp::ReceiveTimeInfo rti; + rtcp::ReceiveTimeInfo rti; rti.ssrc = kUnknownSsrc; rti.last_rr = CompactNtp(config.clock->ConvertTimestampToNtpTime(time)); - webrtc::rtcp::ExtendedReports xr; + rtcp::ExtendedReports xr; xr.AddDlrrItem(rti); auto raw_packet = xr.Build(); diff --git a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc index 83e5a1ffcd..2cb508497f 100644 --- a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc @@ -45,7 +45,7 @@ const uint64_t kStartTime = 123456789; using ::testing::ElementsAreArray; -class LoopbackTransportTest : public webrtc::Transport { +class LoopbackTransportTest : public Transport { public: LoopbackTransportTest() { receivers_extensions_.Register(kAudioLevelExtensionId); diff --git a/modules/rtp_rtcp/source/rtp_sender_video.cc b/modules/rtp_rtcp/source/rtp_sender_video.cc index acb4c7321c..96662d1743 100644 --- a/modules/rtp_rtcp/source/rtp_sender_video.cc +++ b/modules/rtp_rtcp/source/rtp_sender_video.cc @@ -669,7 +669,7 @@ bool RTPSenderVideo::SendVideo(int payload_type, Buffer encrypted_video_payload; if (frame_encryptor_ != nullptr) { const size_t max_ciphertext_size = - frame_encryptor_->GetMaxCiphertextByteSize(webrtc::MediaType::VIDEO, + frame_encryptor_->GetMaxCiphertextByteSize(MediaType::VIDEO, payload.size()); encrypted_video_payload.SetSize(max_ciphertext_size); @@ -682,8 +682,8 @@ bool RTPSenderVideo::SendVideo(int payload_type, } if (frame_encryptor_->Encrypt( - webrtc::MediaType::VIDEO, first_packet->Ssrc(), additional_data, - payload, encrypted_video_payload, &bytes_written) != 0) { + MediaType::VIDEO, first_packet->Ssrc(), additional_data, payload, + encrypted_video_payload, &bytes_written) != 0) { return false; } diff --git a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc index a67ebfd35c..79432a97fa 100644 --- a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc @@ -115,7 +115,7 @@ constexpr int kMaxPacketLength = 1500; constexpr Timestamp kStartTime = Timestamp::Millis(123456789); constexpr TimeDelta kDefaultExpectedRetransmissionTime = TimeDelta::Millis(125); -class LoopbackTransportTest : public webrtc::Transport { +class LoopbackTransportTest : public Transport { public: LoopbackTransportTest() { receivers_extensions_.Register( @@ -1577,8 +1577,7 @@ class RtpSenderVideoWithFrameTransformerTest : public ::testing::Test { std::unique_ptr CreateDefaultEncodedImage() { const uint8_t data[] = {1, 2, 3, 4}; auto encoded_image = std::make_unique(); - encoded_image->SetEncodedData( - webrtc::EncodedImageBuffer::Create(data, sizeof(data))); + encoded_image->SetEncodedData(EncodedImageBuffer::Create(data, sizeof(data))); return encoded_image; } diff --git a/modules/rtp_rtcp/test/testFec/test_fec.cc b/modules/rtp_rtcp/test/testFec/test_fec.cc index bd6998d71f..2898a47940 100644 --- a/modules/rtp_rtcp/test/testFec/test_fec.cc +++ b/modules/rtp_rtcp/test/testFec/test_fec.cc @@ -133,7 +133,7 @@ void RunTest(bool use_flexfec) { // reproduce past results. const unsigned int random_seed = static_cast(time(nullptr)); Random random(random_seed); - std::string filename = webrtc::test::OutputPath() + "randomSeedLog.txt"; + std::string filename = test::OutputPath() + "randomSeedLog.txt"; FILE* random_seed_file = fopen(filename.c_str(), "a"); fprintf(random_seed_file, "%u\n", random_seed); fclose(random_seed_file); From 2eb3fbfbc49a45fedb3ea4f1c53f1da006ca06e4 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Fri, 9 May 2025 10:39:04 +0000 Subject: [PATCH 095/292] [cleanup] Fix redundant webrtc name specifier This CL was uploaded by git cl split. R=hta@webrtc.org Bug: webrtc:42232595 Change-Id: I1383f90786b7b4c077f3c0f0a1c58fdb6dc31534 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390607 Commit-Queue: Harald Alvestrand Auto-Submit: Evan Shrubsole Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44567} --- sdk/android/src/jni/audio_device/audio_track_jni.cc | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sdk/android/src/jni/audio_device/audio_track_jni.cc b/sdk/android/src/jni/audio_device/audio_track_jni.cc index 464805f912..097c07c5d5 100644 --- a/sdk/android/src/jni/audio_device/audio_track_jni.cc +++ b/sdk/android/src/jni/audio_device/audio_track_jni.cc @@ -84,11 +84,10 @@ int32_t AudioTrackJni::InitPlayout() { return 0; } RTC_DCHECK(!playing_); - double buffer_size_factor = - strtod(webrtc::field_trial::FindFullName( - "WebRTC-AudioDevicePlayoutBufferSizeFactor") - .c_str(), - nullptr); + double buffer_size_factor = strtod( + field_trial::FindFullName("WebRTC-AudioDevicePlayoutBufferSizeFactor") + .c_str(), + nullptr); if (buffer_size_factor == 0) buffer_size_factor = 1.0; int requested_buffer_size_bytes = Java_WebRtcAudioTrack_initPlayout( From 2af2759a467a61d5749e48bf45ab8343406229af Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Fri, 9 May 2025 10:38:48 +0000 Subject: [PATCH 096/292] [cleanup] Fix redundant webrtc name specifier This CL was uploaded by git cl split. R=hta@webrtc.org Bug: webrtc:42232595 Change-Id: I065a85350583d051a092041f8bdebf500336c687 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390643 Reviewed-by: Harald Alvestrand Auto-Submit: Evan Shrubsole Commit-Queue: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44568} --- .../peerconnection/peer_connection_factory.cc | 4 ++-- sdk/android/native_api/video/video_source.cc | 4 ++-- sdk/android/src/jni/dav1d_codec.cc | 2 +- sdk/android/src/jni/jni_onload.cc | 2 +- .../src/jni/pc/add_ice_candidate_observer.cc | 2 +- sdk/android/src/jni/pc/ice_candidate.cc | 10 +++++----- sdk/android/src/jni/pc/media_stream_track.cc | 12 +++++------- sdk/android/src/jni/pc/peer_connection_factory.cc | 6 +++--- sdk/android/src/jni/pc/rtp_parameters.cc | 14 +++++++------- sdk/android/src/jni/pc/rtp_receiver.cc | 2 +- sdk/android/src/jni/pc/rtp_sender.cc | 7 +++---- sdk/android/src/jni/pc/rtp_transceiver.cc | 2 +- sdk/android/src/jni/pc/sdp_observer.cc | 2 +- sdk/android/src/jni/pc/turn_customizer.cc | 2 +- .../src/jni/software_video_decoder_factory.cc | 9 ++++----- .../src/jni/software_video_encoder_factory.cc | 9 ++++----- 16 files changed, 42 insertions(+), 47 deletions(-) diff --git a/sdk/android/native_api/peerconnection/peer_connection_factory.cc b/sdk/android/native_api/peerconnection/peer_connection_factory.cc index ca87f809d2..c2208a06be 100644 --- a/sdk/android/native_api/peerconnection/peer_connection_factory.cc +++ b/sdk/android/native_api/peerconnection/peer_connection_factory.cc @@ -20,12 +20,12 @@ namespace webrtc { jobject NativeToJavaPeerConnectionFactory( JNIEnv* jni, - scoped_refptr pcf, + scoped_refptr pcf, std::unique_ptr socket_factory, std::unique_ptr network_thread, std::unique_ptr worker_thread, std::unique_ptr signaling_thread) { - return webrtc::jni::NativeToJavaPeerConnectionFactory( + return jni::NativeToJavaPeerConnectionFactory( jni, pcf, std::move(socket_factory), std::move(network_thread), std::move(worker_thread), std::move(signaling_thread)); } diff --git a/sdk/android/native_api/video/video_source.cc b/sdk/android/native_api/video/video_source.cc index ba2d486aef..2ca8ddcded 100644 --- a/sdk/android/native_api/video/video_source.cc +++ b/sdk/android/native_api/video/video_source.cc @@ -92,9 +92,9 @@ class JavaVideoTrackSourceImpl : public JavaVideoTrackSourceInterface { bool SupportsEncodedOutput() const override { return false; } void GenerateKeyFrame() override {} void AddEncodedSink( - VideoSinkInterface* sink) override {} + VideoSinkInterface* sink) override {} void RemoveEncodedSink( - VideoSinkInterface* sink) override {} + VideoSinkInterface* sink) override {} scoped_refptr android_video_track_source_; ScopedJavaGlobalRef native_capturer_observer_; diff --git a/sdk/android/src/jni/dav1d_codec.cc b/sdk/android/src/jni/dav1d_codec.cc index 1246d88c0b..99ad2f50ae 100644 --- a/sdk/android/src/jni/dav1d_codec.cc +++ b/sdk/android/src/jni/dav1d_codec.cc @@ -18,7 +18,7 @@ namespace webrtc { namespace jni { static jlong JNI_Dav1dDecoder_CreateDecoder(JNIEnv* jni) { - return jlongFromPointer(webrtc::CreateDav1dDecoder().release()); + return jlongFromPointer(CreateDav1dDecoder().release()); } } // namespace jni diff --git a/sdk/android/src/jni/jni_onload.cc b/sdk/android/src/jni/jni_onload.cc index 5c44784077..6945e7601f 100644 --- a/sdk/android/src/jni/jni_onload.cc +++ b/sdk/android/src/jni/jni_onload.cc @@ -28,7 +28,7 @@ extern "C" jint JNIEXPORT JNICALL JNI_OnLoad(JavaVM* jvm, void* reserved) { return -1; RTC_CHECK(InitializeSSL()) << "Failed to InitializeSSL()"; - webrtc::InitClassLoader(GetEnv()); + InitClassLoader(GetEnv()); return ret; } diff --git a/sdk/android/src/jni/pc/add_ice_candidate_observer.cc b/sdk/android/src/jni/pc/add_ice_candidate_observer.cc index 7f3dddbb28..57a053aa9a 100644 --- a/sdk/android/src/jni/pc/add_ice_candidate_observer.cc +++ b/sdk/android/src/jni/pc/add_ice_candidate_observer.cc @@ -25,7 +25,7 @@ AddIceCandidateObserverJni::AddIceCandidateObserverJni( const JavaRef& j_observer) : j_observer_global_(env, j_observer) {} -void AddIceCandidateObserverJni::OnComplete(webrtc::RTCError error) { +void AddIceCandidateObserverJni::OnComplete(RTCError error) { JNIEnv* env = AttachCurrentThreadIfNeeded(); if (error.ok()) { Java_AddIceObserver_onAddSuccess(env, j_observer_global_); diff --git a/sdk/android/src/jni/pc/ice_candidate.cc b/sdk/android/src/jni/pc/ice_candidate.cc index 10c1a11243..4e6e5706c7 100644 --- a/sdk/android/src/jni/pc/ice_candidate.cc +++ b/sdk/android/src/jni/pc/ice_candidate.cc @@ -192,23 +192,23 @@ JavaToNativeContinualGatheringPolicy( return PeerConnectionInterface::GATHER_ONCE; } -webrtc::PortPrunePolicy JavaToNativePortPrunePolicy( +PortPrunePolicy JavaToNativePortPrunePolicy( JNIEnv* jni, const JavaRef& j_port_prune_policy) { std::string enum_name = GetJavaEnumName(jni, j_port_prune_policy); if (enum_name == "NO_PRUNE") { - return webrtc::NO_PRUNE; + return NO_PRUNE; } if (enum_name == "PRUNE_BASED_ON_PRIORITY") { - return webrtc::PRUNE_BASED_ON_PRIORITY; + return PRUNE_BASED_ON_PRIORITY; } if (enum_name == "KEEP_FIRST_READY") { - return webrtc::KEEP_FIRST_READY; + return KEEP_FIRST_READY; } RTC_CHECK(false) << " Unexpected PortPrunePolicy enum name " << enum_name; - return webrtc::NO_PRUNE; + return NO_PRUNE; } PeerConnectionInterface::TlsCertPolicy JavaToNativeTlsCertPolicy( diff --git a/sdk/android/src/jni/pc/media_stream_track.cc b/sdk/android/src/jni/pc/media_stream_track.cc index 20c02536c5..d3eb4e8d3b 100644 --- a/sdk/android/src/jni/pc/media_stream_track.cc +++ b/sdk/android/src/jni/pc/media_stream_track.cc @@ -18,16 +18,14 @@ namespace webrtc { namespace jni { -ScopedJavaLocalRef NativeToJavaMediaType( - JNIEnv* jni, - webrtc::MediaType media_type) { +ScopedJavaLocalRef NativeToJavaMediaType(JNIEnv* jni, + MediaType media_type) { return Java_MediaType_fromNativeIndex(jni, static_cast(media_type)); } -webrtc::MediaType JavaToNativeMediaType(JNIEnv* jni, - const JavaRef& j_media_type) { - return static_cast( - Java_MediaType_getNative(jni, j_media_type)); +MediaType JavaToNativeMediaType(JNIEnv* jni, + const JavaRef& j_media_type) { + return static_cast(Java_MediaType_getNative(jni, j_media_type)); } static ScopedJavaLocalRef JNI_MediaStreamTrack_GetId(JNIEnv* jni, diff --git a/sdk/android/src/jni/pc/peer_connection_factory.cc b/sdk/android/src/jni/pc/peer_connection_factory.cc index f05259b478..a09859afbe 100644 --- a/sdk/android/src/jni/pc/peer_connection_factory.cc +++ b/sdk/android/src/jni/pc/peer_connection_factory.cc @@ -139,7 +139,7 @@ StaticObjectContainer& GetStaticObjects() { ScopedJavaLocalRef NativeToScopedJavaPeerConnectionFactory( JNIEnv* env, - scoped_refptr pcf, + scoped_refptr pcf, std::unique_ptr socket_factory, std::unique_ptr network_thread, std::unique_ptr worker_thread, @@ -179,7 +179,7 @@ static bool factory_static_initialized = false; jobject NativeToJavaPeerConnectionFactory( JNIEnv* jni, - scoped_refptr pcf, + scoped_refptr pcf, std::unique_ptr socket_factory, std::unique_ptr network_thread, std::unique_ptr worker_thread, @@ -318,7 +318,7 @@ ScopedJavaLocalRef CreatePeerConnectionFactoryForJava( #ifndef WEBRTC_EXCLUDE_AUDIO_PROCESSING_MODULE } else { dependencies.audio_processing_builder = - std::make_unique(); + std::make_unique(); #endif } dependencies.video_encoder_factory = diff --git a/sdk/android/src/jni/pc/rtp_parameters.cc b/sdk/android/src/jni/pc/rtp_parameters.cc index 4bd9ee0e1d..c83ea8599c 100644 --- a/sdk/android/src/jni/pc/rtp_parameters.cc +++ b/sdk/android/src/jni/pc/rtp_parameters.cc @@ -20,26 +20,26 @@ namespace jni { namespace { -webrtc::DegradationPreference JavaToNativeDegradationPreference( +DegradationPreference JavaToNativeDegradationPreference( JNIEnv* jni, const JavaRef& j_degradation_preference) { std::string enum_name = GetJavaEnumName(jni, j_degradation_preference); if (enum_name == "DISABLED") - return webrtc::DegradationPreference::DISABLED; + return DegradationPreference::DISABLED; if (enum_name == "MAINTAIN_FRAMERATE") - return webrtc::DegradationPreference::MAINTAIN_FRAMERATE; + return DegradationPreference::MAINTAIN_FRAMERATE; if (enum_name == "MAINTAIN_RESOLUTION") - return webrtc::DegradationPreference::MAINTAIN_RESOLUTION; + return DegradationPreference::MAINTAIN_RESOLUTION; if (enum_name == "BALANCED") - return webrtc::DegradationPreference::BALANCED; + return DegradationPreference::BALANCED; RTC_CHECK(false) << "Unexpected DegradationPreference enum_name " << enum_name; - return webrtc::DegradationPreference::DISABLED; + return DegradationPreference::DISABLED; } ScopedJavaLocalRef NativeToJavaRtpEncodingParameter( @@ -99,7 +99,7 @@ RtpEncodingParameters JavaToNativeRtpEncodingParameters( Java_Encoding_getMaxBitrateBps(jni, j_encoding_parameters); encoding.bitrate_priority = Java_Encoding_getBitratePriority(jni, j_encoding_parameters); - encoding.network_priority = static_cast( + encoding.network_priority = static_cast( Java_Encoding_getNetworkPriority(jni, j_encoding_parameters)); encoding.max_bitrate_bps = JavaToNativeOptionalInt(jni, j_max_bitrate); ScopedJavaLocalRef j_min_bitrate = diff --git a/sdk/android/src/jni/pc/rtp_receiver.cc b/sdk/android/src/jni/pc/rtp_receiver.cc index 56dc78de13..81dcd4abc2 100644 --- a/sdk/android/src/jni/pc/rtp_receiver.cc +++ b/sdk/android/src/jni/pc/rtp_receiver.cc @@ -33,7 +33,7 @@ class RtpReceiverObserverJni : public RtpReceiverObserverInterface { ~RtpReceiverObserverJni() override = default; - void OnFirstPacketReceived(webrtc::MediaType media_type) override { + void OnFirstPacketReceived(MediaType media_type) override { JNIEnv* const env = AttachCurrentThreadIfNeeded(); Java_Observer_onFirstPacketReceived(env, j_observer_global_, NativeToJavaMediaType(env, media_type)); diff --git a/sdk/android/src/jni/pc/rtp_sender.cc b/sdk/android/src/jni/pc/rtp_sender.cc index e63d0cc688..c3dd3a2a63 100644 --- a/sdk/android/src/jni/pc/rtp_sender.cc +++ b/sdk/android/src/jni/pc/rtp_sender.cc @@ -112,11 +112,10 @@ static void JNI_RtpSender_SetFrameEncryptor(JNIEnv* jni, static jni_zero::ScopedJavaLocalRef JNI_RtpSender_GetMediaType( JNIEnv* jni, jlong j_rtp_sender_pointer) { - webrtc::MediaType media_type = + MediaType media_type = reinterpret_cast(j_rtp_sender_pointer)->media_type(); - return media_type == webrtc::MediaType::AUDIO - ? NativeToJavaString(jni, "audio") - : NativeToJavaString(jni, "video"); + return media_type == MediaType::AUDIO ? NativeToJavaString(jni, "audio") + : NativeToJavaString(jni, "video"); } } // namespace jni diff --git a/sdk/android/src/jni/pc/rtp_transceiver.cc b/sdk/android/src/jni/pc/rtp_transceiver.cc index d7425b97b7..8818438b85 100644 --- a/sdk/android/src/jni/pc/rtp_transceiver.cc +++ b/sdk/android/src/jni/pc/rtp_transceiver.cc @@ -182,7 +182,7 @@ jboolean JNI_RtpTransceiver_SetDirection( RtpTransceiverDirection direction = static_cast( Java_RtpTransceiverDirection_getNativeIndex(jni, j_rtp_transceiver_direction)); - webrtc::RTCError error = + RTCError error = reinterpret_cast(j_rtp_transceiver_pointer) ->SetDirectionWithError(direction); if (!error.ok()) { diff --git a/sdk/android/src/jni/pc/sdp_observer.cc b/sdk/android/src/jni/pc/sdp_observer.cc index c8b4345af4..2b5d8e23b4 100644 --- a/sdk/android/src/jni/pc/sdp_observer.cc +++ b/sdk/android/src/jni/pc/sdp_observer.cc @@ -41,7 +41,7 @@ void CreateSdpObserverJni::OnSuccess(SessionDescriptionInterface* desc) { delete desc; } -void CreateSdpObserverJni::OnFailure(webrtc::RTCError error) { +void CreateSdpObserverJni::OnFailure(RTCError error) { JNIEnv* env = AttachCurrentThreadIfNeeded(); Java_SdpObserver_onCreateFailure(env, j_observer_global_, NativeToJavaString(env, error.message())); diff --git a/sdk/android/src/jni/pc/turn_customizer.cc b/sdk/android/src/jni/pc/turn_customizer.cc index 94ebf0140d..b79fe515fd 100644 --- a/sdk/android/src/jni/pc/turn_customizer.cc +++ b/sdk/android/src/jni/pc/turn_customizer.cc @@ -22,7 +22,7 @@ TurnCustomizer* GetNativeTurnCustomizer( const JavaRef& j_turn_customizer) { if (IsNull(env, j_turn_customizer)) return nullptr; - return reinterpret_cast( + return reinterpret_cast( Java_TurnCustomizer_getNativeTurnCustomizer(env, j_turn_customizer)); } diff --git a/sdk/android/src/jni/software_video_decoder_factory.cc b/sdk/android/src/jni/software_video_decoder_factory.cc index 08e94a964f..cc82e51481 100644 --- a/sdk/android/src/jni/software_video_decoder_factory.cc +++ b/sdk/android/src/jni/software_video_decoder_factory.cc @@ -24,8 +24,7 @@ namespace webrtc { namespace jni { static jlong JNI_SoftwareVideoDecoderFactory_CreateFactory(JNIEnv* env) { - return webrtc::NativeToJavaPointer( - CreateBuiltinVideoDecoderFactory().release()); + return NativeToJavaPointer(CreateBuiltinVideoDecoderFactory().release()); } jboolean JNI_SoftwareVideoDecoderFactory_IsSupported( @@ -53,10 +52,10 @@ static jni_zero::ScopedJavaLocalRef JNI_SoftwareVideoDecoderFactory_GetSupportedCodecs(JNIEnv* env, jlong j_factory) { auto* const native_factory = - reinterpret_cast(j_factory); + reinterpret_cast(j_factory); - return webrtc::NativeToJavaList(env, native_factory->GetSupportedFormats(), - &webrtc::jni::SdpVideoFormatToVideoCodecInfo); + return NativeToJavaList(env, native_factory->GetSupportedFormats(), + &jni::SdpVideoFormatToVideoCodecInfo); } } // namespace jni diff --git a/sdk/android/src/jni/software_video_encoder_factory.cc b/sdk/android/src/jni/software_video_encoder_factory.cc index a2d44d932d..344a459259 100644 --- a/sdk/android/src/jni/software_video_encoder_factory.cc +++ b/sdk/android/src/jni/software_video_encoder_factory.cc @@ -21,8 +21,7 @@ namespace webrtc { namespace jni { static jlong JNI_SoftwareVideoEncoderFactory_CreateFactory(JNIEnv* env) { - return webrtc::NativeToJavaPointer( - CreateBuiltinVideoEncoderFactory().release()); + return NativeToJavaPointer(CreateBuiltinVideoEncoderFactory().release()); } jboolean JNI_SoftwareVideoEncoderFactory_IsSupported( @@ -50,10 +49,10 @@ static jni_zero::ScopedJavaLocalRef JNI_SoftwareVideoEncoderFactory_GetSupportedCodecs(JNIEnv* env, jlong j_factory) { auto* const native_factory = - reinterpret_cast(j_factory); + reinterpret_cast(j_factory); - return webrtc::NativeToJavaList(env, native_factory->GetSupportedFormats(), - &webrtc::jni::SdpVideoFormatToVideoCodecInfo); + return NativeToJavaList(env, native_factory->GetSupportedFormats(), + &jni::SdpVideoFormatToVideoCodecInfo); } } // namespace jni From 55263c8e5b141b276865c0f423346a9e06ed6211 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Fri, 9 May 2025 10:37:11 +0000 Subject: [PATCH 097/292] [cleanup] Fix redundant webrtc name specifier This CL was uploaded by git cl split. R=hta@webrtc.org Bug: webrtc:42232595 Change-Id: Ib0bf8c795d80fd31cd153a9ccb30389f6b57f823 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390640 Auto-Submit: Evan Shrubsole Commit-Queue: Harald Alvestrand Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44569} --- .../congestion_controller/goog_cc/alr_detector_unittest.cc | 6 +++--- .../goog_cc/goog_cc_network_control.cc | 2 +- .../goog_cc/loss_based_bandwidth_estimation.cc | 2 +- .../congestion_controller/goog_cc/loss_based_bwe_v2_test.cc | 2 +- modules/congestion_controller/goog_cc/probe_controller.cc | 3 +-- .../rtp/transport_feedback_demuxer_unittest.cc | 2 +- 6 files changed, 8 insertions(+), 9 deletions(-) diff --git a/modules/congestion_controller/goog_cc/alr_detector_unittest.cc b/modules/congestion_controller/goog_cc/alr_detector_unittest.cc index 57c583dd67..725bd342d6 100644 --- a/modules/congestion_controller/goog_cc/alr_detector_unittest.cc +++ b/modules/congestion_controller/goog_cc/alr_detector_unittest.cc @@ -156,7 +156,7 @@ TEST(AlrDetectorTest, BandwidthEstimateChanges) { } TEST(AlrDetectorTest, ParseControlFieldTrial) { - webrtc::test::ScopedFieldTrials scoped_field_trial( + test::ScopedFieldTrials scoped_field_trial( "WebRTC-ProbingScreenshareBwe/Control/"); std::optional parsed_params = AlrExperimentSettings::CreateFromFieldTrial( @@ -165,7 +165,7 @@ TEST(AlrDetectorTest, ParseControlFieldTrial) { } TEST(AlrDetectorTest, ParseActiveFieldTrial) { - webrtc::test::ScopedFieldTrials scoped_field_trial( + test::ScopedFieldTrials scoped_field_trial( "WebRTC-ProbingScreenshareBwe/1.1,2875,85,20,-20,1/"); std::optional parsed_params = AlrExperimentSettings::CreateFromFieldTrial( @@ -180,7 +180,7 @@ TEST(AlrDetectorTest, ParseActiveFieldTrial) { } TEST(AlrDetectorTest, ParseAlrSpecificFieldTrial) { - webrtc::test::ScopedFieldTrials scoped_field_trial( + test::ScopedFieldTrials scoped_field_trial( "WebRTC-AlrDetectorParameters/" "bw_usage:90%,start:0%,stop:-10%/"); FieldTrialBasedConfig field_trials; diff --git a/modules/congestion_controller/goog_cc/goog_cc_network_control.cc b/modules/congestion_controller/goog_cc/goog_cc_network_control.cc index a50a9b6cc6..f318e85612 100644 --- a/modules/congestion_controller/goog_cc/goog_cc_network_control.cc +++ b/modules/congestion_controller/goog_cc/goog_cc_network_control.cc @@ -77,7 +77,7 @@ BandwidthLimitedCause GetBandwidthLimitedCause(LossBasedState loss_based_state, case LossBasedState::kDecreasing: // Probes may not be sent in this state. return BandwidthLimitedCause::kLossLimitedBwe; - case webrtc::LossBasedState::kIncreaseUsingPadding: + case LossBasedState::kIncreaseUsingPadding: // Probes may not be sent in this state. return BandwidthLimitedCause::kLossLimitedBwe; case LossBasedState::kIncreasing: diff --git a/modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.cc b/modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.cc index 81cdad512e..08bd4b3fc4 100644 --- a/modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.cc +++ b/modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.cc @@ -81,7 +81,7 @@ double ExponentialUpdate(TimeDelta window, TimeDelta interval) { return 1.0f - exp(interval / window * -1.0); } -bool IsEnabled(const webrtc::FieldTrialsView& key_value_config, +bool IsEnabled(const FieldTrialsView& key_value_config, absl::string_view name) { return absl::StartsWith(key_value_config.Lookup(name), "Enabled"); } diff --git a/modules/congestion_controller/goog_cc/loss_based_bwe_v2_test.cc b/modules/congestion_controller/goog_cc/loss_based_bwe_v2_test.cc index f900473db8..a061eb6a3a 100644 --- a/modules/congestion_controller/goog_cc/loss_based_bwe_v2_test.cc +++ b/modules/congestion_controller/goog_cc/loss_based_bwe_v2_test.cc @@ -27,7 +27,7 @@ namespace webrtc { namespace { -using ::webrtc::test::ExplicitKeyValueConfig; +using test::ExplicitKeyValueConfig; constexpr TimeDelta kObservationDurationLowerBound = TimeDelta::Millis(250); constexpr TimeDelta kDelayedIncreaseWindow = TimeDelta::Millis(300); diff --git a/modules/congestion_controller/goog_cc/probe_controller.cc b/modules/congestion_controller/goog_cc/probe_controller.cc index bf8253ded5..0847c19734 100644 --- a/modules/congestion_controller/goog_cc/probe_controller.cc +++ b/modules/congestion_controller/goog_cc/probe_controller.cc @@ -426,8 +426,7 @@ std::vector ProbeController::RequestProbe( return std::vector(); } -void ProbeController::SetNetworkStateEstimate( - webrtc::NetworkStateEstimate estimate) { +void ProbeController::SetNetworkStateEstimate(NetworkStateEstimate estimate) { network_estimate_ = estimate; } diff --git a/modules/congestion_controller/rtp/transport_feedback_demuxer_unittest.cc b/modules/congestion_controller/rtp/transport_feedback_demuxer_unittest.cc index 79f25a9879..204ce328d1 100644 --- a/modules/congestion_controller/rtp/transport_feedback_demuxer_unittest.cc +++ b/modules/congestion_controller/rtp/transport_feedback_demuxer_unittest.cc @@ -28,7 +28,7 @@ using PacketInfo = StreamFeedbackObserver::StreamPacketInfo; static constexpr uint32_t kSsrc = 8492; -class MockStreamFeedbackObserver : public webrtc::StreamFeedbackObserver { +class MockStreamFeedbackObserver : public StreamFeedbackObserver { public: MOCK_METHOD(void, OnPacketFeedbackVector, From 48ca1258900942210cdfa9aab6d050020c0c4d68 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Fri, 9 May 2025 10:35:54 +0000 Subject: [PATCH 098/292] [cleanup] Fix redundant webrtc name specifier This CL was uploaded by git cl split. R=hta@webrtc.org No-IWYU: LSC Bug: webrtc:42232595 Change-Id: Ief513a87b95f13d4b30177b4bbfc02c17f7e49fb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390442 Auto-Submit: Evan Shrubsole Reviewed-by: Harald Alvestrand Commit-Queue: Evan Shrubsole Cr-Commit-Position: refs/heads/main@{#44570} --- common_video/h264/sps_vui_rewriter.cc | 10 +++++----- common_video/h264/sps_vui_rewriter_unittest.cc | 12 ++++++------ .../codecs/h264/test/h264_impl_unittest.cc | 2 +- 3 files changed, 12 insertions(+), 12 deletions(-) diff --git a/common_video/h264/sps_vui_rewriter.cc b/common_video/h264/sps_vui_rewriter.cc index b102cf4c78..0684b0d86e 100644 --- a/common_video/h264/sps_vui_rewriter.cc +++ b/common_video/h264/sps_vui_rewriter.cc @@ -85,7 +85,7 @@ uint32_t CopyBits(int bits, bool CopyAndRewriteVui(const SpsParser::SpsState& sps, BitstreamReader& source, BitBufferWriter& destination, - const webrtc::ColorSpace* color_space, + const ColorSpace* color_space, SpsVuiRewriter::ParseResult& out_vui_rewritten); void CopyHrdParameters(BitstreamReader& source, BitBufferWriter& destination); @@ -134,7 +134,7 @@ void SpsVuiRewriter::UpdateStats(ParseResult result, Direction direction) { SpsVuiRewriter::ParseResult SpsVuiRewriter::ParseAndRewriteSps( ArrayView buffer, std::optional* sps, - const webrtc::ColorSpace* color_space, + const ColorSpace* color_space, Buffer* destination) { // Create temporary RBSP decoded buffer of the payload (exlcuding the // leading nalu type header byte (the SpsParser uses only the payload). @@ -210,7 +210,7 @@ SpsVuiRewriter::ParseResult SpsVuiRewriter::ParseAndRewriteSps( SpsVuiRewriter::ParseResult SpsVuiRewriter::ParseAndRewriteSps( ArrayView buffer, std::optional* sps, - const webrtc::ColorSpace* color_space, + const ColorSpace* color_space, Buffer* destination, Direction direction) { ParseResult result = @@ -221,7 +221,7 @@ SpsVuiRewriter::ParseResult SpsVuiRewriter::ParseAndRewriteSps( Buffer SpsVuiRewriter::ParseOutgoingBitstreamAndRewrite( ArrayView buffer, - const webrtc::ColorSpace* color_space) { + const ColorSpace* color_space) { std::vector nalus = H264::FindNaluIndices(buffer); // Allocate some extra space for potentially adding a missing VUI. @@ -281,7 +281,7 @@ namespace { bool CopyAndRewriteVui(const SpsParser::SpsState& sps, BitstreamReader& source, BitBufferWriter& destination, - const webrtc::ColorSpace* color_space, + const ColorSpace* color_space, SpsVuiRewriter::ParseResult& out_vui_rewritten) { out_vui_rewritten = SpsVuiRewriter::ParseResult::kVuiOk; diff --git a/common_video/h264/sps_vui_rewriter_unittest.cc b/common_video/h264/sps_vui_rewriter_unittest.cc index 581b9ef276..2b0f06d5f7 100644 --- a/common_video/h264/sps_vui_rewriter_unittest.cc +++ b/common_video/h264/sps_vui_rewriter_unittest.cc @@ -150,37 +150,37 @@ static const VuiHeader kVuiLimitedRangeBt709Color = { /* transfer_characteristics= */ 1, /* matrix_coefficients= */ 1}; -static const webrtc::ColorSpace kColorSpaceH264Default( +static const ColorSpace kColorSpaceH264Default( ColorSpace::PrimaryID::kUnspecified, ColorSpace::TransferID::kUnspecified, ColorSpace::MatrixID::kUnspecified, ColorSpace::RangeID::kLimited); -static const webrtc::ColorSpace kColorSpacePrimariesBt709( +static const ColorSpace kColorSpacePrimariesBt709( ColorSpace::PrimaryID::kBT709, ColorSpace::TransferID::kUnspecified, ColorSpace::MatrixID::kUnspecified, ColorSpace::RangeID::kLimited); -static const webrtc::ColorSpace kColorSpaceTransferBt709( +static const ColorSpace kColorSpaceTransferBt709( ColorSpace::PrimaryID::kUnspecified, ColorSpace::TransferID::kBT709, ColorSpace::MatrixID::kUnspecified, ColorSpace::RangeID::kLimited); -static const webrtc::ColorSpace kColorSpaceMatrixBt709( +static const ColorSpace kColorSpaceMatrixBt709( ColorSpace::PrimaryID::kUnspecified, ColorSpace::TransferID::kUnspecified, ColorSpace::MatrixID::kBT709, ColorSpace::RangeID::kLimited); -static const webrtc::ColorSpace kColorSpaceFullRange( +static const ColorSpace kColorSpaceFullRange( ColorSpace::PrimaryID::kBT709, ColorSpace::TransferID::kUnspecified, ColorSpace::MatrixID::kUnspecified, ColorSpace::RangeID::kFull); -static const webrtc::ColorSpace kColorSpaceBt709LimitedRange( +static const ColorSpace kColorSpaceBt709LimitedRange( ColorSpace::PrimaryID::kBT709, ColorSpace::TransferID::kBT709, ColorSpace::MatrixID::kBT709, diff --git a/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc b/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc index 985224d07e..6e9e58e717 100644 --- a/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc +++ b/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc @@ -42,7 +42,7 @@ class TestH264Impl : public VideoCodecUnitTest { } void ModifyCodecSettings(VideoCodec* codec_settings) override { - webrtc::test::CodecSettings(kVideoCodecH264, codec_settings); + test::CodecSettings(kVideoCodecH264, codec_settings); } }; From 3504db9472f6ee2804f53c13a455304652f90882 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Fri, 9 May 2025 10:34:28 +0000 Subject: [PATCH 099/292] [cleanup] Fix redundant webrtc name specifier This CL was uploaded by git cl split. R=hta@webrtc.org Bug: webrtc:42232595 No-IWYU: LSC Change-Id: I5df65e5bcf25eac9c8a34b7d785c53ab9b115e47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390441 Auto-Submit: Evan Shrubsole Reviewed-by: Harald Alvestrand Commit-Queue: Evan Shrubsole Cr-Commit-Position: refs/heads/main@{#44571} --- audio/audio_receive_stream.cc | 27 +++++++------- audio/audio_state_unittest.cc | 2 +- audio/channel_receive.cc | 37 +++++++++---------- ...ive_frame_transformer_delegate_unittest.cc | 2 +- audio/channel_send.cc | 23 ++++++------ ...end_frame_transformer_delegate_unittest.cc | 6 +-- audio/channel_send_unittest.cc | 2 +- audio/test/audio_end_to_end_test.cc | 3 +- audio/test/non_sender_rtt_test.cc | 2 +- 9 files changed, 50 insertions(+), 54 deletions(-) diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc index 344c4b4428..1c4d634729 100644 --- a/audio/audio_receive_stream.cc +++ b/audio/audio_receive_stream.cc @@ -77,9 +77,9 @@ std::string AudioReceiveStreamInterface::Config::ToString() const { namespace { std::unique_ptr CreateChannelReceive( const Environment& env, - webrtc::AudioState* audio_state, + AudioState* audio_state, NetEqFactory* neteq_factory, - const webrtc::AudioReceiveStreamInterface::Config& config) { + const AudioReceiveStreamInterface::Config& config) { RTC_DCHECK(audio_state); internal::AudioState* internal_audio_state = static_cast(audio_state); @@ -98,8 +98,8 @@ AudioReceiveStreamImpl::AudioReceiveStreamImpl( const Environment& env, PacketRouter* packet_router, NetEqFactory* neteq_factory, - const webrtc::AudioReceiveStreamInterface::Config& config, - const scoped_refptr& audio_state) + const AudioReceiveStreamInterface::Config& config, + const scoped_refptr& audio_state) : AudioReceiveStreamImpl( env, packet_router, @@ -111,8 +111,8 @@ AudioReceiveStreamImpl::AudioReceiveStreamImpl( AudioReceiveStreamImpl::AudioReceiveStreamImpl( const Environment& /* env */, PacketRouter* packet_router, - const webrtc::AudioReceiveStreamInterface::Config& config, - const scoped_refptr& audio_state, + const AudioReceiveStreamInterface::Config& config, + const scoped_refptr& audio_state, std::unique_ptr channel_receive) : config_(config), audio_state_(audio_state), @@ -159,7 +159,7 @@ void AudioReceiveStreamImpl::UnregisterFromTransport() { } void AudioReceiveStreamImpl::ReconfigureForTesting( - const webrtc::AudioReceiveStreamInterface::Config& config) { + const AudioReceiveStreamInterface::Config& config) { RTC_DCHECK_RUN_ON(&worker_thread_checker_); // SSRC can't be changed mid-stream. @@ -212,7 +212,7 @@ bool AudioReceiveStreamImpl::IsRunning() const { } void AudioReceiveStreamImpl::SetDepacketizerToDecoderFrameTransformer( - scoped_refptr frame_transformer) { + scoped_refptr frame_transformer) { RTC_DCHECK_RUN_ON(&worker_thread_checker_); channel_receive_->SetDepacketizerToDecoderFrameTransformer( std::move(frame_transformer)); @@ -238,7 +238,7 @@ void AudioReceiveStreamImpl::SetNackHistory(int history_ms) { channel_receive_->SetNACKStatus(history_ms != 0, history_ms / 20); } -void AudioReceiveStreamImpl::SetRtcpMode(webrtc::RtcpMode mode) { +void AudioReceiveStreamImpl::SetRtcpMode(RtcpMode mode) { RTC_DCHECK_RUN_ON(&worker_thread_checker_); if (config_.rtp.rtcp_mode == mode) @@ -255,17 +255,17 @@ void AudioReceiveStreamImpl::SetNonSenderRttMeasurement(bool enabled) { } void AudioReceiveStreamImpl::SetFrameDecryptor( - scoped_refptr frame_decryptor) { + scoped_refptr frame_decryptor) { // TODO(bugs.webrtc.org/11993): This is called via WebRtcAudioReceiveStream, // expect to be called on the network thread. RTC_DCHECK_RUN_ON(&worker_thread_checker_); channel_receive_->SetFrameDecryptor(std::move(frame_decryptor)); } -webrtc::AudioReceiveStreamInterface::Stats AudioReceiveStreamImpl::GetStats( +AudioReceiveStreamInterface::Stats AudioReceiveStreamImpl::GetStats( bool get_and_clear_legacy_stats) const { RTC_DCHECK_RUN_ON(&worker_thread_checker_); - webrtc::AudioReceiveStreamInterface::Stats stats; + AudioReceiveStreamInterface::Stats stats; stats.remote_ssrc = remote_ssrc(); auto receive_codec = channel_receive_->GetReceiveCodec(); @@ -274,8 +274,7 @@ webrtc::AudioReceiveStreamInterface::Stats AudioReceiveStreamImpl::GetStats( stats.codec_payload_type = receive_codec->first; } - webrtc::CallReceiveStatistics call_stats = - channel_receive_->GetRTCPStatistics(); + CallReceiveStatistics call_stats = channel_receive_->GetRTCPStatistics(); stats.payload_bytes_received = call_stats.payload_bytes_received; stats.header_and_padding_bytes_received = call_stats.header_and_padding_bytes_received; diff --git a/audio/audio_state_unittest.cc b/audio/audio_state_unittest.cc index 0c278d770c..6a98bdd357 100644 --- a/audio/audio_state_unittest.cc +++ b/audio/audio_state_unittest.cc @@ -57,7 +57,7 @@ struct FakeAsyncAudioProcessingHelper { std::unique_ptr CreateTaskQueue( absl::string_view /* name */, Priority /* priority */) const override { - return std::unique_ptr( + return std::unique_ptr( new FakeTaskQueue()); } }; diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc index cfbf843032..832f86aaab 100644 --- a/audio/channel_receive.cc +++ b/audio/channel_receive.cc @@ -132,7 +132,7 @@ class ChannelReceive : public ChannelReceiveInterface, scoped_refptr decoder_factory, std::optional codec_pair_id, scoped_refptr frame_decryptor, - const webrtc::CryptoOptions& crypto_options, + const CryptoOptions& crypto_options, scoped_refptr frame_transformer); ~ChannelReceive() override; @@ -190,7 +190,7 @@ class ChannelReceive : public ChannelReceiveInterface, CallReceiveStatistics GetRTCPStatistics() const override; void SetNACKStatus(bool enable, int max_packets) override; - void SetRtcpMode(webrtc::RtcpMode mode) override; + void SetRtcpMode(RtcpMode mode) override; void SetNonSenderRttMeasurement(bool enabled) override; AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo( @@ -204,11 +204,10 @@ class ChannelReceive : public ChannelReceiveInterface, // Sets a frame transformer between the depacketizer and the decoder, to // transform the received frames before decoding them. void SetDepacketizerToDecoderFrameTransformer( - scoped_refptr frame_transformer) - override; + scoped_refptr frame_transformer) override; void SetFrameDecryptor( - scoped_refptr frame_decryptor) override; + scoped_refptr frame_decryptor) override; void OnLocalSsrcChange(uint32_t local_ssrc) override; @@ -233,7 +232,7 @@ class ChannelReceive : public ChannelReceiveInterface, RTC_RUN_ON(worker_thread_checker_); void InitFrameTransformerDelegate( - scoped_refptr frame_transformer) + scoped_refptr frame_transformer) RTC_RUN_ON(worker_thread_checker_); // Thread checkers document and lock usage of some methods to specific threads @@ -293,7 +292,7 @@ class ChannelReceive : public ChannelReceiveInterface, mutable Mutex ts_stats_lock_; - webrtc::RtpTimestampUnwrapper rtp_ts_wraparound_handler_; + RtpTimestampUnwrapper rtp_ts_wraparound_handler_; // The rtp timestamp of the first played out audio frame. int64_t capture_start_rtp_time_stamp_; // The capture ntp time (in local timebase) of the first played out audio @@ -310,12 +309,12 @@ class ChannelReceive : public ChannelReceiveInterface, // E2EE Audio Frame Decryption scoped_refptr frame_decryptor_ RTC_GUARDED_BY(worker_thread_checker_); - webrtc::CryptoOptions crypto_options_; + CryptoOptions crypto_options_; - webrtc::AbsoluteCaptureTimeInterpolator absolute_capture_time_interpolator_ + AbsoluteCaptureTimeInterpolator absolute_capture_time_interpolator_ RTC_GUARDED_BY(worker_thread_checker_); - webrtc::CaptureClockOffsetUpdater capture_clock_offset_updater_ + CaptureClockOffsetUpdater capture_clock_offset_updater_ RTC_GUARDED_BY(ts_stats_lock_); scoped_refptr @@ -381,7 +380,7 @@ void ChannelReceive::OnReceivedPayloadData(ArrayView payload, } void ChannelReceive::InitFrameTransformerDelegate( - scoped_refptr frame_transformer) { + scoped_refptr frame_transformer) { RTC_DCHECK(frame_transformer); RTC_DCHECK(!frame_transformer_delegate_); RTC_DCHECK(worker_thread_->IsCurrent()); @@ -565,7 +564,7 @@ ChannelReceive::ChannelReceive( scoped_refptr decoder_factory, std::optional codec_pair_id, scoped_refptr frame_decryptor, - const webrtc::CryptoOptions& crypto_options, + const CryptoOptions& crypto_options, scoped_refptr frame_transformer) : env_(env), worker_thread_(TaskQueueBase::Current()), @@ -713,14 +712,14 @@ void ChannelReceive::ReceivePacket(const uint8_t* packet, Buffer decrypted_audio_payload; if (frame_decryptor_ != nullptr) { const size_t max_plaintext_size = frame_decryptor_->GetMaxPlaintextByteSize( - webrtc::MediaType::AUDIO, payload_length); + MediaType::AUDIO, payload_length); decrypted_audio_payload.SetSize(max_plaintext_size); const std::vector csrcs(header.arrOfCSRCs, header.arrOfCSRCs + header.numCSRCs); const FrameDecryptorInterface::Result decrypt_result = frame_decryptor_->Decrypt( - webrtc::MediaType::AUDIO, csrcs, + MediaType::AUDIO, csrcs, /*additional_data=*/ nullptr, ArrayView(payload, payload_data_length), decrypted_audio_payload); @@ -747,7 +746,7 @@ void ChannelReceive::ReceivePacket(const uint8_t* packet, char buf[1024]; SimpleStringBuilder mime_type(buf); auto it = payload_type_map_.find(header.payloadType); - mime_type << webrtc::MediaTypeToString(webrtc::MediaType::AUDIO) << "/" + mime_type << MediaTypeToString(MediaType::AUDIO) << "/" << (it != payload_type_map_.end() ? it->second.name : "x-unknown"); frame_transformer_delegate_->Transform(payload_data, header, remote_ssrc_, @@ -906,7 +905,7 @@ void ChannelReceive::SetNACKStatus(bool enable, int max_packets) { } } -void ChannelReceive::SetRtcpMode(webrtc::RtcpMode mode) { +void ChannelReceive::SetRtcpMode(RtcpMode mode) { RTC_DCHECK_RUN_ON(&worker_thread_checker_); rtp_rtcp_->SetRTCPStatus(mode); } @@ -933,7 +932,7 @@ void ChannelReceive::RtcpPacketTypesCounterUpdated( } void ChannelReceive::SetDepacketizerToDecoderFrameTransformer( - scoped_refptr frame_transformer) { + scoped_refptr frame_transformer) { RTC_DCHECK_RUN_ON(&worker_thread_checker_); if (!frame_transformer) { RTC_DCHECK_NOTREACHED() << "Not setting the transformer?"; @@ -953,7 +952,7 @@ void ChannelReceive::SetDepacketizerToDecoderFrameTransformer( } void ChannelReceive::SetFrameDecryptor( - scoped_refptr frame_decryptor) { + scoped_refptr frame_decryptor) { RTC_DCHECK_RUN_ON(&worker_thread_checker_); frame_decryptor_ = std::move(frame_decryptor); } @@ -1189,7 +1188,7 @@ std::unique_ptr CreateChannelReceive( scoped_refptr decoder_factory, std::optional codec_pair_id, scoped_refptr frame_decryptor, - const webrtc::CryptoOptions& crypto_options, + const CryptoOptions& crypto_options, scoped_refptr frame_transformer) { return std::make_unique( env, neteq_factory, audio_device_module, rtcp_send_transport, local_ssrc, diff --git a/audio/channel_receive_frame_transformer_delegate_unittest.cc b/audio/channel_receive_frame_transformer_delegate_unittest.cc index 95bfd50b13..4b60ea210a 100644 --- a/audio/channel_receive_frame_transformer_delegate_unittest.cc +++ b/audio/channel_receive_frame_transformer_delegate_unittest.cc @@ -42,7 +42,7 @@ class MockChannelReceive { public: MOCK_METHOD(void, ReceiveFrame, - (webrtc::ArrayView packet, + (ArrayView packet, const RTPHeader& header, Timestamp receive_time)); diff --git a/audio/channel_send.cc b/audio/channel_send.cc index 52817f31ff..e953549ecf 100644 --- a/audio/channel_send.cc +++ b/audio/channel_send.cc @@ -132,7 +132,7 @@ class ChannelSend : public ChannelSendInterface, Transport* rtp_transport, RtcpRttStats* rtcp_rtt_stats, FrameEncryptorInterface* frame_encryptor, - const webrtc::CryptoOptions& crypto_options, + const CryptoOptions& crypto_options, bool extmap_allow_mixed, int rtcp_report_interval_ms, uint32_t ssrc, @@ -204,8 +204,7 @@ class ChannelSend : public ChannelSendInterface, // Sets a frame transformer between encoder and packetizer, to transform // encoded frames before sending them out the network. void SetEncoderToPacketizerFrameTransformer( - scoped_refptr frame_transformer) - override; + scoped_refptr frame_transformer) override; // RtcpPacketTypeCounterObserver. void RtcpPacketTypesCounterUpdated( @@ -249,7 +248,7 @@ class ChannelSend : public ChannelSendInterface, void OnReceivedRtt(int64_t rtt_ms); void InitFrameTransformerDelegate( - scoped_refptr frame_transformer); + scoped_refptr frame_transformer); // Calls the encoder on the encoder queue (instead of blocking). void CallEncoderAsync(absl::AnyInvocable modifier); @@ -301,7 +300,7 @@ class ChannelSend : public ChannelSendInterface, scoped_refptr frame_encryptor_ RTC_GUARDED_BY(encoder_queue_checker_); // E2EE Frame Encryption Options - const webrtc::CryptoOptions crypto_options_; + const CryptoOptions crypto_options_; // Delegates calls to a frame transformer to transform audio, and // receives callbacks with the transformed frames; delegates calls to @@ -389,7 +388,7 @@ int32_t ChannelSend::SendData(AudioFrameType frameType, // is transformed, the delegate will call SendRtpAudio to send it. char buf[1024]; SimpleStringBuilder mime_type(buf); - mime_type << webrtc::MediaTypeToString(webrtc::MediaType::AUDIO) << "/" + mime_type << MediaTypeToString(MediaType::AUDIO) << "/" << encoder_format_.name; frame_transformer_delegate_->Transform( frameType, payloadType, rtp_timestamp + rtp_rtcp_->StartTimestamp(), @@ -421,13 +420,13 @@ int32_t ChannelSend::SendRtpAudio(AudioFrameType frameType, // TODO(benwright@webrtc.org) - Allocate enough to always encrypt inline. // Allocate a buffer to hold the maximum possible encrypted payload. size_t max_ciphertext_size = frame_encryptor_->GetMaxCiphertextByteSize( - webrtc::MediaType::AUDIO, payload.size()); + MediaType::AUDIO, payload.size()); encrypted_audio_payload.SetSize(max_ciphertext_size); // Encrypt the audio payload into the buffer. size_t bytes_written = 0; int encrypt_status = - frame_encryptor_->Encrypt(webrtc::MediaType::AUDIO, rtp_rtcp_->SSRC(), + frame_encryptor_->Encrypt(MediaType::AUDIO, rtp_rtcp_->SSRC(), /*additional_data=*/nullptr, payload, encrypted_audio_payload, &bytes_written); if (encrypt_status != 0) { @@ -489,7 +488,7 @@ ChannelSend::ChannelSend( Transport* rtp_transport, RtcpRttStats* rtcp_rtt_stats, FrameEncryptorInterface* frame_encryptor, - const webrtc::CryptoOptions& crypto_options, + const CryptoOptions& crypto_options, bool extmap_allow_mixed, int rtcp_report_interval_ms, uint32_t ssrc, @@ -919,7 +918,7 @@ void ChannelSend::SetFrameEncryptor( } void ChannelSend::SetEncoderToPacketizerFrameTransformer( - scoped_refptr frame_transformer) { + scoped_refptr frame_transformer) { RTC_DCHECK_RUN_ON(&worker_thread_checker_); if (!frame_transformer) return; @@ -937,7 +936,7 @@ void ChannelSend::OnReceivedRtt(int64_t rtt_ms) { } void ChannelSend::InitFrameTransformerDelegate( - scoped_refptr frame_transformer) { + scoped_refptr frame_transformer) { RTC_DCHECK_RUN_ON(&encoder_queue_checker_); RTC_DCHECK(frame_transformer); RTC_DCHECK(!frame_transformer_delegate_); @@ -971,7 +970,7 @@ std::unique_ptr CreateChannelSend( Transport* rtp_transport, RtcpRttStats* rtcp_rtt_stats, FrameEncryptorInterface* frame_encryptor, - const webrtc::CryptoOptions& crypto_options, + const CryptoOptions& crypto_options, bool extmap_allow_mixed, int rtcp_report_interval_ms, uint32_t ssrc, diff --git a/audio/channel_send_frame_transformer_delegate_unittest.cc b/audio/channel_send_frame_transformer_delegate_unittest.cc index 2a526f0570..e9575c1857 100644 --- a/audio/channel_send_frame_transformer_delegate_unittest.cc +++ b/audio/channel_send_frame_transformer_delegate_unittest.cc @@ -51,9 +51,9 @@ class MockChannelSend { (AudioFrameType frameType, uint8_t payloadType, uint32_t rtp_timestamp, - webrtc::ArrayView payload, + ArrayView payload, int64_t absolute_capture_timestamp_ms, - webrtc::ArrayView csrcs, + ArrayView csrcs, std::optional audio_level_dbov)); ChannelSendFrameTransformerDelegate::SendFrameCallback callback() { @@ -104,7 +104,7 @@ std::unique_ptr CreateFrame() { AudioFrameType::kEmptyFrame, 0, 0, mock_data, sizeof(mock_data), 0, /*ssrc=*/0, /*mimeType=*/"audio/opus", /*audio_level_dbov=*/123); return absl::WrapUnique( - static_cast(frame.release())); + static_cast(frame.release())); } // Test that the delegate registers itself with the frame transformer on Init(). diff --git a/audio/channel_send_unittest.cc b/audio/channel_send_unittest.cc index 03ce5bceeb..7636ee32c6 100644 --- a/audio/channel_send_unittest.cc +++ b/audio/channel_send_unittest.cc @@ -123,7 +123,7 @@ class ChannelSendTest : public ::testing::Test { void ProcessNextFrame() { ProcessNextFrame(CreateAudioFrame()); } GlobalSimulatedTimeController time_controller_; - webrtc::test::ScopedKeyValueConfig field_trials_; + test::ScopedKeyValueConfig field_trials_; Environment env_; NiceMock transport_; CryptoOptions crypto_options_; diff --git a/audio/test/audio_end_to_end_test.cc b/audio/test/audio_end_to_end_test.cc index 746ae3fea7..d9c4594935 100644 --- a/audio/test/audio_end_to_end_test.cc +++ b/audio/test/audio_end_to_end_test.cc @@ -63,8 +63,7 @@ void AudioEndToEndTest::ModifyAudioConfigs( AudioSendStream::Config* send_config, std::vector* /* receive_configs */) { // Large bitrate by default. - const webrtc::SdpAudioFormat kDefaultFormat("opus", 48000, 2, - {{"stereo", "1"}}); + const SdpAudioFormat kDefaultFormat("opus", 48000, 2, {{"stereo", "1"}}); send_config->send_codec_spec = AudioSendStream::Config::SendCodecSpec( test::VideoTestConstants::kAudioSendPayloadType, kDefaultFormat); send_config->min_bitrate_bps = 32000; diff --git a/audio/test/non_sender_rtt_test.cc b/audio/test/non_sender_rtt_test.cc index 17dd97ef09..49b6cae92e 100644 --- a/audio/test/non_sender_rtt_test.cc +++ b/audio/test/non_sender_rtt_test.cc @@ -60,7 +60,7 @@ TEST_F(NonSenderRttTest, NonSenderRttStats) { // cases it can take more than 10 seconds. EXPECT_THAT( WaitUntil([&] { return HasRoundTripTimeMeasurement(); }, IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kLongTimeoutMs)}), + {.timeout = TimeDelta::Millis(kLongTimeoutMs)}), IsRtcOk()); } From 469d89cf1b1de27e58c6528d971d26d2e9c0a83e Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Fri, 9 May 2025 10:34:45 +0000 Subject: [PATCH 100/292] [cleanup] Fix redundant webrtc name specifier This CL was uploaded by git cl split. R=hta@webrtc.org No-IWYU: LSC Bug: webrtc:42232595 Change-Id: Idcf2617a0e73bd4adcfd40fc804e37a249f5d427 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390601 Reviewed-by: Harald Alvestrand Auto-Submit: Evan Shrubsole Commit-Queue: Evan Shrubsole Cr-Commit-Position: refs/heads/main@{#44572} --- .../resource_adaptation_processor_unittest.cc | 2 +- .../video_stream_adapter_unittest.cc | 6 ++--- .../overuse_frame_detector_unittest.cc | 22 +++++++++---------- .../video_stream_encoder_resource_manager.cc | 2 +- 4 files changed, 16 insertions(+), 16 deletions(-) diff --git a/call/adaptation/resource_adaptation_processor_unittest.cc b/call/adaptation/resource_adaptation_processor_unittest.cc index 79433c7bb3..fe60171c2a 100644 --- a/call/adaptation/resource_adaptation_processor_unittest.cc +++ b/call/adaptation/resource_adaptation_processor_unittest.cc @@ -147,7 +147,7 @@ class ResourceAdaptationProcessorTest : public ::testing::Test { protected: AutoThread main_thread_; - webrtc::test::ScopedKeyValueConfig field_trials_; + test::ScopedKeyValueConfig field_trials_; FakeFrameRateProvider frame_rate_provider_; VideoStreamInputStateProvider input_state_provider_; scoped_refptr resource_; diff --git a/call/adaptation/video_stream_adapter_unittest.cc b/call/adaptation/video_stream_adapter_unittest.cc index 83720b4318..4e57883d89 100644 --- a/call/adaptation/video_stream_adapter_unittest.cc +++ b/call/adaptation/video_stream_adapter_unittest.cc @@ -155,7 +155,7 @@ class VideoStreamAdapterTest : public ::testing::Test { field_trials_) {} protected: - webrtc::test::ScopedKeyValueConfig field_trials_; + test::ScopedKeyValueConfig field_trials_; FakeVideoStreamInputStateProvider input_state_provider_; scoped_refptr resource_; testing::StrictMock encoder_stats_observer_; @@ -918,7 +918,7 @@ TEST_F(VideoStreamAdapterTest, AdaptationConstraintDisallowsAdaptationsUp) { TEST(VideoStreamAdapterDeathTest, SetDegradationPreferenceInvalidatesAdaptations) { - webrtc::test::ScopedKeyValueConfig field_trials; + test::ScopedKeyValueConfig field_trials; FakeVideoStreamInputStateProvider input_state_provider; testing::StrictMock encoder_stats_observer_; VideoStreamAdapter adapter(&input_state_provider, &encoder_stats_observer_, @@ -931,7 +931,7 @@ TEST(VideoStreamAdapterDeathTest, } TEST(VideoStreamAdapterDeathTest, AdaptDownInvalidatesAdaptations) { - webrtc::test::ScopedKeyValueConfig field_trials; + test::ScopedKeyValueConfig field_trials; FakeVideoStreamInputStateProvider input_state_provider; testing::StrictMock encoder_stats_observer_; VideoStreamAdapter adapter(&input_state_provider, &encoder_stats_observer_, diff --git a/video/adaptation/overuse_frame_detector_unittest.cc b/video/adaptation/overuse_frame_detector_unittest.cc index fbcd92e20f..f1b522f105 100644 --- a/video/adaptation/overuse_frame_detector_unittest.cc +++ b/video/adaptation/overuse_frame_detector_unittest.cc @@ -108,7 +108,7 @@ class OveruseFrameDetectorTest : public ::testing::Test, VideoFrame frame = VideoFrame::Builder() .set_video_frame_buffer(I420Buffer::Create(width, height)) - .set_rotation(webrtc::kVideoRotation_0) + .set_rotation(kVideoRotation_0) .set_timestamp_us(0) .build(); uint32_t timestamp = 0; @@ -134,7 +134,7 @@ class OveruseFrameDetectorTest : public ::testing::Test, VideoFrame frame = VideoFrame::Builder() .set_video_frame_buffer(I420Buffer::Create(width, height)) - .set_rotation(webrtc::kVideoRotation_0) + .set_rotation(kVideoRotation_0) .set_timestamp_us(0) .build(); uint32_t timestamp = 0; @@ -164,12 +164,12 @@ class OveruseFrameDetectorTest : public ::testing::Test, int width, int height, int delay_us) { - webrtc::Random random(17); + Random random(17); VideoFrame frame = VideoFrame::Builder() .set_video_frame_buffer(I420Buffer::Create(width, height)) - .set_rotation(webrtc::kVideoRotation_0) + .set_rotation(kVideoRotation_0) .set_timestamp_us(0) .build(); uint32_t timestamp = 0; @@ -382,7 +382,7 @@ TEST_F(OveruseFrameDetectorTest, MeasuresMultipleConcurrentSamples) { VideoFrame frame = VideoFrame::Builder() .set_video_frame_buffer(I420Buffer::Create(kWidth, kHeight)) - .set_rotation(webrtc::kVideoRotation_0) + .set_rotation(kVideoRotation_0) .set_timestamp_us(0) .build(); for (size_t i = 0; i < 1000; ++i) { @@ -409,7 +409,7 @@ TEST_F(OveruseFrameDetectorTest, UpdatesExistingSamples) { VideoFrame frame = VideoFrame::Builder() .set_video_frame_buffer(I420Buffer::Create(kWidth, kHeight)) - .set_rotation(webrtc::kVideoRotation_0) + .set_rotation(kVideoRotation_0) .set_timestamp_us(0) .build(); uint32_t timestamp = 0; @@ -675,7 +675,7 @@ class OveruseFrameDetectorTest2 : public OveruseFrameDetectorTest { VideoFrame frame = VideoFrame::Builder() .set_video_frame_buffer(I420Buffer::Create(width, height)) - .set_rotation(webrtc::kVideoRotation_0) + .set_rotation(kVideoRotation_0) .set_timestamp_us(0) .build(); while (num_frames-- > 0) { @@ -694,12 +694,12 @@ class OveruseFrameDetectorTest2 : public OveruseFrameDetectorTest { int width, int height, int delay_us) override { - webrtc::Random random(17); + Random random(17); VideoFrame frame = VideoFrame::Builder() .set_video_frame_buffer(I420Buffer::Create(width, height)) - .set_rotation(webrtc::kVideoRotation_0) + .set_rotation(kVideoRotation_0) .set_timestamp_us(0) .build(); for (int i = 0; i < num_frames; i++) { @@ -863,7 +863,7 @@ TEST_F(OveruseFrameDetectorTest2, MeasuresMultipleConcurrentSamples) { VideoFrame frame = VideoFrame::Builder() .set_video_frame_buffer(I420Buffer::Create(kWidth, kHeight)) - .set_rotation(webrtc::kVideoRotation_0) + .set_rotation(kVideoRotation_0) .set_timestamp_us(0) .build(); for (size_t i = 0; i < 1000; ++i) { @@ -890,7 +890,7 @@ TEST_F(OveruseFrameDetectorTest2, UpdatesExistingSamples) { VideoFrame frame = VideoFrame::Builder() .set_video_frame_buffer(I420Buffer::Create(kWidth, kHeight)) - .set_rotation(webrtc::kVideoRotation_0) + .set_rotation(kVideoRotation_0) .set_timestamp_us(0) .build(); uint32_t timestamp = 0; diff --git a/video/adaptation/video_stream_encoder_resource_manager.cc b/video/adaptation/video_stream_encoder_resource_manager.cc index 20664b91cb..437f50f69e 100644 --- a/video/adaptation/video_stream_encoder_resource_manager.cc +++ b/video/adaptation/video_stream_encoder_resource_manager.cc @@ -579,7 +579,7 @@ void VideoStreamEncoderResourceManager::UpdateBandwidthQualityScalerSettings( // Before executing "StartCheckForOveruse",we must execute "AddResource" // firstly,because it can make the listener valid. AddResource(bandwidth_quality_scaler_resource_, - webrtc::VideoAdaptationReason::kQuality); + VideoAdaptationReason::kQuality); bandwidth_quality_scaler_resource_->StartCheckForOveruse( resolution_bitrate_limits, codec_type); } From 8aa99b141c269a5ef02da4ca84789fecbadb3a4c Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Fri, 9 May 2025 10:36:55 +0000 Subject: [PATCH 101/292] [cleanup] Fix redundant webrtc name specifier This CL was uploaded by git cl split. R=hta@webrtc.org No-IWYU: LSC Bug: webrtc:42232595 Change-Id: I79a0259f1ead29262eb143ad4073689856d4925e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390604 Commit-Queue: Evan Shrubsole Auto-Submit: Evan Shrubsole Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44573} --- .../aec3/block_framer_unittest.cc | 15 +++-- .../aec3/block_processor_unittest.cc | 24 ++++---- .../aec3/echo_canceller3_unittest.cc | 15 ++--- .../aec3/frame_blocker_unittest.cc | 18 +++--- .../aec_dump/aec_dump_impl.cc | 4 +- .../aec_dump/aec_dump_unittest.cc | 20 +++---- .../agc2/rnn_vad/rnn_fc_unittest.cc | 2 +- .../agc2/rnn_vad/rnn_gru_unittest.cc | 2 +- .../agc2/rnn_vad/rnn_vad_unittest.cc | 2 +- .../agc2/rnn_vad/test_utils.cc | 2 +- .../agc2/vad_wrapper_unittest.cc | 5 +- .../audio_processing_impl_unittest.cc | 20 +++---- .../audio_processing_performance_unittest.cc | 12 ++-- .../audio_processing_unittest.cc | 2 +- .../test/aec_dump_based_simulator.cc | 59 +++++++++---------- .../test/audio_processing_simulator.cc | 2 +- .../test/performance_timer.cc | 2 +- .../test/runtime_setting_util.cc | 2 +- 18 files changed, 96 insertions(+), 112 deletions(-) diff --git a/modules/audio_processing/aec3/block_framer_unittest.cc b/modules/audio_processing/aec3/block_framer_unittest.cc index c9b60588d1..d45b96e980 100644 --- a/modules/audio_processing/aec3/block_framer_unittest.cc +++ b/modules/audio_processing/aec3/block_framer_unittest.cc @@ -124,9 +124,9 @@ void RunWronglySizedInsertAndExtractParametersTest( num_sub_frame_bands, std::vector>( num_sub_frame_channels, std::vector(sub_frame_length, 0.f))); - std::vector>> output_sub_frame_view( + std::vector>> output_sub_frame_view( output_sub_frame.size(), - std::vector>(num_sub_frame_channels)); + std::vector>(num_sub_frame_channels)); SetupSubFrameView(&output_sub_frame, &output_sub_frame_view); BlockFramer framer(correct_num_bands, correct_num_channels); EXPECT_DEATH( @@ -147,9 +147,9 @@ void RunWronglySizedInsertParameterTest(int sample_rate_hz, correct_num_bands, std::vector>( correct_num_channels, std::vector(kSubFrameLength, 0.f))); - std::vector>> output_sub_frame_view( + std::vector>> output_sub_frame_view( output_sub_frame.size(), - std::vector>(correct_num_channels)); + std::vector>(correct_num_channels)); SetupSubFrameView(&output_sub_frame, &output_sub_frame_view); BlockFramer framer(correct_num_bands, correct_num_channels); framer.InsertBlockAndExtractSubFrame(correct_block, &output_sub_frame_view); @@ -174,9 +174,8 @@ void RunWronglyInsertOrderTest(int sample_rate_hz, correct_num_bands, std::vector>( num_channels, std::vector(kSubFrameLength, 0.f))); - std::vector>> output_sub_frame_view( - output_sub_frame.size(), - std::vector>(num_channels)); + std::vector>> output_sub_frame_view( + output_sub_frame.size(), std::vector>(num_channels)); SetupSubFrameView(&output_sub_frame, &output_sub_frame_view); BlockFramer framer(correct_num_bands, num_channels); for (size_t k = 0; k < num_preceeding_api_calls; ++k) { @@ -293,7 +292,7 @@ TEST(BlockFramerDeathTest, WrongNumberOfPreceedingApiCallsForInsertBlock) { for (size_t num_channels : {1, 2, 8}) { for (auto rate : {16000, 32000, 48000}) { for (size_t num_calls = 0; num_calls < 4; ++num_calls) { - webrtc::StringBuilder ss; + StringBuilder ss; ss << "Sample rate: " << rate; ss << ", Num channels: " << num_channels; ss << ", Num preceeding InsertBlockAndExtractSubFrame calls: " diff --git a/modules/audio_processing/aec3/block_processor_unittest.cc b/modules/audio_processing/aec3/block_processor_unittest.cc index a56eaac9fa..eece30c9a7 100644 --- a/modules/audio_processing/aec3/block_processor_unittest.cc +++ b/modules/audio_processing/aec3/block_processor_unittest.cc @@ -132,9 +132,9 @@ TEST(BlockProcessor, DISABLED_DelayControllerIntegration) { Random random_generator(42U); for (auto rate : {16000, 32000, 48000}) { SCOPED_TRACE(ProduceDebugText(rate)); - std::unique_ptr> + std::unique_ptr> render_delay_buffer_mock( - new StrictMock(rate, 1)); + new StrictMock(rate, 1)); EXPECT_CALL(*render_delay_buffer_mock, Insert(_)) .Times(kNumBlocks) .WillRepeatedly(Return(RenderDelayBuffer::BufferingEvent::kNone)); @@ -171,15 +171,14 @@ TEST(BlockProcessor, DISABLED_SubmoduleIntegration) { Random random_generator(42U); for (auto rate : {16000, 32000, 48000}) { SCOPED_TRACE(ProduceDebugText(rate)); - std::unique_ptr> + std::unique_ptr> render_delay_buffer_mock( - new StrictMock(rate, 1)); - std::unique_ptr< - ::testing::StrictMock> + new StrictMock(rate, 1)); + std::unique_ptr<::testing::StrictMock> render_delay_controller_mock( - new StrictMock()); - std::unique_ptr> - echo_remover_mock(new StrictMock()); + new StrictMock()); + std::unique_ptr> + echo_remover_mock(new StrictMock()); EXPECT_CALL(*render_delay_buffer_mock, Insert(_)) .Times(kNumBlocks - 1) @@ -291,10 +290,9 @@ TEST(BlockProcessor, ExternalDelayAppliedCorrectlyWithInitialCaptureCalls) { std::unique_ptr delay_buffer( RenderDelayBuffer::Create(config, kSampleRateHz, kNumRenderChannels)); - std::unique_ptr> - echo_remover_mock(new NiceMock()); - webrtc::test::MockEchoRemover* echo_remover_mock_pointer = - echo_remover_mock.get(); + std::unique_ptr> echo_remover_mock( + new NiceMock()); + test::MockEchoRemover* echo_remover_mock_pointer = echo_remover_mock.get(); std::unique_ptr block_processor(BlockProcessor::Create( config, kSampleRateHz, kNumRenderChannels, kNumCaptureChannels, diff --git a/modules/audio_processing/aec3/echo_canceller3_unittest.cc b/modules/audio_processing/aec3/echo_canceller3_unittest.cc index 486c64cd3c..f0fa73de42 100644 --- a/modules/audio_processing/aec3/echo_canceller3_unittest.cc +++ b/modules/audio_processing/aec3/echo_canceller3_unittest.cc @@ -316,9 +316,8 @@ class EchoCanceller3Tester { constexpr size_t kNumFullBlocksPerFrame = 160 / kBlockSize; constexpr size_t kExpectedNumBlocksToProcess = (kNumFramesToProcess * 160) / kBlockSize; - std::unique_ptr> - block_processor_mock( - new StrictMock()); + std::unique_ptr> + block_processor_mock(new StrictMock()); EXPECT_CALL(*block_processor_mock, BufferRender(_)) .Times(kExpectedNumBlocksToProcess); EXPECT_CALL(*block_processor_mock, UpdateEchoLeakageStatus(_)).Times(0); @@ -393,9 +392,8 @@ class EchoCanceller3Tester { EchoLeakageTestVariant leakage_report_variant) { constexpr size_t kExpectedNumBlocksToProcess = (kNumFramesToProcess * 160) / kBlockSize; - std::unique_ptr> - block_processor_mock( - new StrictMock()); + std::unique_ptr> + block_processor_mock(new StrictMock()); EXPECT_CALL(*block_processor_mock, BufferRender(_)) .Times(kExpectedNumBlocksToProcess); EXPECT_CALL(*block_processor_mock, ProcessCapture(_, _, _, _)) @@ -482,9 +480,8 @@ class EchoCanceller3Tester { const size_t kNumFullBlocksPerFrame = 160 / kBlockSize; const size_t kExpectedNumBlocksToProcess = (kNumFramesToProcess * 160) / kBlockSize; - std::unique_ptr> - block_processor_mock( - new StrictMock()); + std::unique_ptr> + block_processor_mock(new StrictMock()); EXPECT_CALL(*block_processor_mock, BufferRender(_)) .Times(kExpectedNumBlocksToProcess); EXPECT_CALL(*block_processor_mock, UpdateEchoLeakageStatus(_)).Times(0); diff --git a/modules/audio_processing/aec3/frame_blocker_unittest.cc b/modules/audio_processing/aec3/frame_blocker_unittest.cc index 135018412a..ee6296978f 100644 --- a/modules/audio_processing/aec3/frame_blocker_unittest.cc +++ b/modules/audio_processing/aec3/frame_blocker_unittest.cc @@ -199,9 +199,9 @@ void RunWronglySizedInsertAndExtractParametersTest( num_sub_frame_bands, std::vector>( num_sub_frame_channels, std::vector(sub_frame_length, 0.f))); - std::vector>> input_sub_frame_view( + std::vector>> input_sub_frame_view( input_sub_frame.size(), - std::vector>(num_sub_frame_channels)); + std::vector>(num_sub_frame_channels)); FillSubFrameView(0, 0, &input_sub_frame, &input_sub_frame_view); FrameBlocker blocker(correct_num_bands, correct_num_channels); EXPECT_DEATH( @@ -222,9 +222,9 @@ void RunWronglySizedExtractParameterTest(int sample_rate_hz, correct_num_bands, std::vector>( correct_num_channels, std::vector(kSubFrameLength, 0.f))); - std::vector>> input_sub_frame_view( + std::vector>> input_sub_frame_view( input_sub_frame.size(), - std::vector>(correct_num_channels)); + std::vector>(correct_num_channels)); FillSubFrameView(0, 0, &input_sub_frame, &input_sub_frame_view); FrameBlocker blocker(correct_num_bands, correct_num_channels); blocker.InsertSubFrameAndExtractBlock(input_sub_frame_view, &correct_block); @@ -247,9 +247,8 @@ void RunWrongExtractOrderTest(int sample_rate_hz, std::vector>> input_sub_frame( num_bands, std::vector>( num_channels, std::vector(kSubFrameLength, 0.f))); - std::vector>> input_sub_frame_view( - input_sub_frame.size(), - std::vector>(num_channels)); + std::vector>> input_sub_frame_view( + input_sub_frame.size(), std::vector>(num_channels)); FillSubFrameView(0, 0, &input_sub_frame, &input_sub_frame_view); FrameBlocker blocker(num_bands, num_channels); for (size_t k = 0; k < num_preceeding_api_calls; ++k) { @@ -367,7 +366,7 @@ TEST(FrameBlockerDeathTest, WrongNumberOfPreceedingApiCallsForExtractBlock) { for (auto rate : {16000, 32000, 48000}) { for (size_t num_channels : {1, 2, 4, 8}) { for (size_t num_calls = 0; num_calls < 4; ++num_calls) { - webrtc::StringBuilder ss; + StringBuilder ss; ss << "Sample rate: " << rate; ss << "Num channels: " << num_channels; ss << ", Num preceeding InsertSubFrameAndExtractBlock calls: " @@ -395,8 +394,7 @@ TEST(FrameBlockerDeathTest, NullBlockParameter) { std::vector>> sub_frame( 1, std::vector>( 1, std::vector(kSubFrameLength, 0.f))); - std::vector>> sub_frame_view( - sub_frame.size()); + std::vector>> sub_frame_view(sub_frame.size()); FillSubFrameView(0, 0, &sub_frame, &sub_frame_view); EXPECT_DEATH( FrameBlocker(1, 1).InsertSubFrameAndExtractBlock(sub_frame_view, nullptr), diff --git a/modules/audio_processing/aec_dump/aec_dump_impl.cc b/modules/audio_processing/aec_dump/aec_dump_impl.cc index 0c196506fd..31826ff824 100644 --- a/modules/audio_processing/aec_dump/aec_dump_impl.cc +++ b/modules/audio_processing/aec_dump/aec_dump_impl.cc @@ -23,8 +23,8 @@ namespace webrtc { namespace { -void CopyFromConfigToEvent(const webrtc::InternalAPMConfig& config, - webrtc::audioproc::Config* pb_cfg) { +void CopyFromConfigToEvent(const InternalAPMConfig& config, + audioproc::Config* pb_cfg) { pb_cfg->set_aec_enabled(config.aec_enabled); pb_cfg->set_aec_delay_agnostic_enabled(config.aec_delay_agnostic_enabled); pb_cfg->set_aec_drift_compensation_enabled( diff --git a/modules/audio_processing/aec_dump/aec_dump_unittest.cc b/modules/audio_processing/aec_dump/aec_dump_unittest.cc index 2a8110c4fc..f928ae8faa 100644 --- a/modules/audio_processing/aec_dump/aec_dump_unittest.cc +++ b/modules/audio_processing/aec_dump/aec_dump_unittest.cc @@ -21,14 +21,14 @@ namespace webrtc { TEST(AecDumper, APICallsDoNotCrash) { // Note order of initialization: Task queue has to be initialized // before AecDump. - webrtc::TaskQueueForTest file_writer_queue("file_writer_queue"); + TaskQueueForTest file_writer_queue("file_writer_queue"); const std::string filename = - webrtc::test::TempFilename(webrtc::test::OutputPath(), "aec_dump"); + test::TempFilename(test::OutputPath(), "aec_dump"); { - std::unique_ptr aec_dump = - webrtc::AecDumpFactory::Create(filename, -1, file_writer_queue.Get()); + std::unique_ptr aec_dump = + AecDumpFactory::Create(filename, -1, file_writer_queue.Get()); constexpr int kNumChannels = 1; constexpr int kNumSamplesPerChannel = 160; @@ -44,10 +44,10 @@ TEST(AecDumper, APICallsDoNotCrash) { aec_dump->WriteCaptureStreamMessage(); - webrtc::InternalAPMConfig apm_config; + InternalAPMConfig apm_config; aec_dump->WriteConfig(apm_config); - webrtc::ProcessingConfig api_format; + ProcessingConfig api_format; constexpr int64_t kTimeNowMs = 123456789ll; aec_dump->WriteInitMessage(api_format, kTimeNowMs); } @@ -56,14 +56,14 @@ TEST(AecDumper, APICallsDoNotCrash) { } TEST(AecDumper, WriteToFile) { - webrtc::TaskQueueForTest file_writer_queue("file_writer_queue"); + TaskQueueForTest file_writer_queue("file_writer_queue"); const std::string filename = - webrtc::test::TempFilename(webrtc::test::OutputPath(), "aec_dump"); + test::TempFilename(test::OutputPath(), "aec_dump"); { - std::unique_ptr aec_dump = - webrtc::AecDumpFactory::Create(filename, -1, file_writer_queue.Get()); + std::unique_ptr aec_dump = + AecDumpFactory::Create(filename, -1, file_writer_queue.Get()); constexpr int kNumChannels = 1; constexpr int kNumSamplesPerChannel = 160; diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_fc_unittest.cc b/modules/audio_processing/agc2/rnn_vad/rnn_fc_unittest.cc index ff9bb18bc2..672090bad3 100644 --- a/modules/audio_processing/agc2/rnn_vad/rnn_fc_unittest.cc +++ b/modules/audio_processing/agc2/rnn_vad/rnn_fc_unittest.cc @@ -69,7 +69,7 @@ TEST_P(RnnFcParametrization, DISABLED_BenchmarkFullyConnectedLayer) { /*layer_name=*/"FC"); constexpr int kNumTests = 10000; - ::webrtc::test::PerformanceTimer perf_timer(kNumTests); + test::PerformanceTimer perf_timer(kNumTests); for (int k = 0; k < kNumTests; ++k) { perf_timer.StartTimer(); fc.ComputeOutput(kFullyConnectedInputVector); diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc b/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc index 43b3e6885a..5afd4383d2 100644 --- a/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc +++ b/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc @@ -141,7 +141,7 @@ TEST_P(RnnGruParametrization, DISABLED_BenchmarkGatedRecurrentLayer) { input_sequence.size() / kInputLayerOutputSize; constexpr int kNumTests = 100; - ::webrtc::test::PerformanceTimer perf_timer(kNumTests); + test::PerformanceTimer perf_timer(kNumTests); for (int k = 0; k < kNumTests; ++k) { perf_timer.StartTimer(); for (int i = 0; i < input_sequence_length; ++i) { diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc b/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc index f33cd14a8a..b9991eab91 100644 --- a/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc +++ b/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc @@ -135,7 +135,7 @@ TEST_P(RnnVadProbabilityParametrization, DISABLED_RnnVadPerformance) { std::array feature_vector; RnnVad rnn_vad(cpu_features); constexpr int number_of_tests = 100; - ::webrtc::test::PerformanceTimer perf_timer(number_of_tests); + test::PerformanceTimer perf_timer(number_of_tests); for (int k = 0; k < number_of_tests; ++k) { features_extractor.Reset(); rnn_vad.Reset(); diff --git a/modules/audio_processing/agc2/rnn_vad/test_utils.cc b/modules/audio_processing/agc2/rnn_vad/test_utils.cc index a3951bf260..aa78b70ecb 100644 --- a/modules/audio_processing/agc2/rnn_vad/test_utils.cc +++ b/modules/audio_processing/agc2/rnn_vad/test_utils.cc @@ -68,7 +68,7 @@ class FloatFileReader : public FileReader { } // namespace -using webrtc::test::ResourcePath; +using test::ResourcePath; void ExpectEqualFloatArray(ArrayView expected, ArrayView computed) { diff --git a/modules/audio_processing/agc2/vad_wrapper_unittest.cc b/modules/audio_processing/agc2/vad_wrapper_unittest.cc index a4c1dcba21..4cac82d2db 100644 --- a/modules/audio_processing/agc2/vad_wrapper_unittest.cc +++ b/modules/audio_processing/agc2/vad_wrapper_unittest.cc @@ -42,10 +42,7 @@ class MockVad : public VoiceActivityDetectorWrapper::MonoVad { public: MOCK_METHOD(int, SampleRateHz, (), (const, override)); MOCK_METHOD(void, Reset, (), (override)); - MOCK_METHOD(float, - Analyze, - (webrtc::ArrayView frame), - (override)); + MOCK_METHOD(float, Analyze, (ArrayView frame), (override)); }; // Checks that the ctor and `Initialize()` read the sample rate of the wrapped diff --git a/modules/audio_processing/audio_processing_impl_unittest.cc b/modules/audio_processing/audio_processing_impl_unittest.cc index 5d4b04e692..e761f7609b 100644 --- a/modules/audio_processing/audio_processing_impl_unittest.cc +++ b/modules/audio_processing/audio_processing_impl_unittest.cc @@ -204,7 +204,7 @@ TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) { TEST(AudioProcessingImplTest, UpdateCapturePreGainRuntimeSetting) { scoped_refptr apm = BuiltinAudioProcessingBuilder().Build(CreateEnvironment()); - webrtc::AudioProcessing::Config apm_config; + AudioProcessing::Config apm_config; apm_config.pre_amplifier.enabled = true; apm_config.pre_amplifier.fixed_gain_factor = 1.f; apm->ApplyConfig(apm_config); @@ -237,7 +237,7 @@ TEST(AudioProcessingImplTest, LevelAdjustmentUpdateCapturePreGainRuntimeSetting) { scoped_refptr apm = BuiltinAudioProcessingBuilder().Build(CreateEnvironment()); - webrtc::AudioProcessing::Config apm_config; + AudioProcessing::Config apm_config; apm_config.capture_level_adjustment.enabled = true; apm_config.capture_level_adjustment.pre_gain_factor = 1.f; apm->ApplyConfig(apm_config); @@ -270,7 +270,7 @@ TEST(AudioProcessingImplTest, LevelAdjustmentUpdateCapturePostGainRuntimeSetting) { scoped_refptr apm = BuiltinAudioProcessingBuilder().Build(CreateEnvironment()); - webrtc::AudioProcessing::Config apm_config; + AudioProcessing::Config apm_config; apm_config.capture_level_adjustment.enabled = true; apm_config.capture_level_adjustment.post_gain_factor = 1.f; apm->ApplyConfig(apm_config); @@ -393,7 +393,7 @@ TEST(AudioProcessingImplTest, .SetEchoControlFactory(std::move(echo_control_factory)) .Build(CreateEnvironment()); // Disable AGC. - webrtc::AudioProcessing::Config apm_config; + AudioProcessing::Config apm_config; apm_config.gain_controller1.enabled = false; apm_config.gain_controller2.enabled = false; apm_config.pre_amplifier.enabled = true; @@ -436,7 +436,7 @@ TEST(AudioProcessingImplTest, .SetEchoControlFactory(std::move(echo_control_factory)) .Build(CreateEnvironment()); // Disable AGC. - webrtc::AudioProcessing::Config apm_config; + AudioProcessing::Config apm_config; apm_config.gain_controller1.enabled = false; apm_config.gain_controller2.enabled = false; apm_config.capture_level_adjustment.enabled = true; @@ -478,7 +478,7 @@ TEST(AudioProcessingImplTest, BuiltinAudioProcessingBuilder() .SetEchoControlFactory(std::move(echo_control_factory)) .Build(CreateEnvironment()); - webrtc::AudioProcessing::Config apm_config; + AudioProcessing::Config apm_config; // Enable AGC1. apm_config.gain_controller1.enabled = true; apm_config.gain_controller1.analog_gain_controller.enabled = true; @@ -534,7 +534,7 @@ TEST(AudioProcessingImplTest, EchoControllerObservesPlayoutVolumeChange) { .SetEchoControlFactory(std::move(echo_control_factory)) .Build(CreateEnvironment()); // Disable AGC. - webrtc::AudioProcessing::Config apm_config; + AudioProcessing::Config apm_config; apm_config.gain_controller1.enabled = false; apm_config.gain_controller2.enabled = false; apm->ApplyConfig(apm_config); @@ -591,7 +591,7 @@ TEST(AudioProcessingImplTest, RenderPreProcessorBeforeEchoDetector) { .SetEchoDetector(test_echo_detector) .SetRenderPreProcessing(std::move(test_render_pre_processor)) .Build(CreateEnvironment()); - webrtc::AudioProcessing::Config apm_config; + AudioProcessing::Config apm_config; apm_config.pre_amplifier.enabled = true; apm->ApplyConfig(apm_config); @@ -643,7 +643,7 @@ class StartupInputVolumeParameterizedTest // is never modified. TEST_P(StartupInputVolumeParameterizedTest, WithNoInputVolumeControllerStartupVolumeNotModified) { - webrtc::AudioProcessing::Config config; + AudioProcessing::Config config; config.gain_controller1.enabled = false; config.gain_controller2.enabled = false; auto apm = BuiltinAudioProcessingBuilder(config).Build(CreateEnvironment()); @@ -662,7 +662,7 @@ INSTANTIATE_TEST_SUITE_P(AudioProcessingImplTest, // volume always matches the applied one. TEST(AudioProcessingImplTest, WithNoInputVolumeControllerAppliedAndRecommendedVolumesMatch) { - webrtc::AudioProcessing::Config config; + AudioProcessing::Config config; config.gain_controller1.enabled = false; config.gain_controller2.enabled = false; auto apm = BuiltinAudioProcessingBuilder(config).Build(CreateEnvironment()); diff --git a/modules/audio_processing/audio_processing_performance_unittest.cc b/modules/audio_processing/audio_processing_performance_unittest.cc index d12302cb58..820d403eee 100644 --- a/modules/audio_processing/audio_processing_performance_unittest.cc +++ b/modules/audio_processing/audio_processing_performance_unittest.cc @@ -33,10 +33,10 @@ namespace webrtc { namespace { -using ::webrtc::test::GetGlobalMetricsLogger; -using ::webrtc::test::ImprovementDirection; -using ::webrtc::test::Metric; -using ::webrtc::test::Unit; +using test::GetGlobalMetricsLogger; +using test::ImprovementDirection; +using test::Metric; +using test::Unit; class CallSimulator; @@ -206,7 +206,7 @@ class TimedThreadApiProcessor { simulation_config_(simulation_config), apm_(apm), frame_data_(kMaxFrameSize), - clock_(webrtc::Clock::GetRealTimeClock()), + clock_(Clock::GetRealTimeClock()), num_durations_to_store_(num_durations_to_store), api_call_durations_(num_durations_to_store_ - kNumInitializationFrames), samples_count_(0), @@ -355,7 +355,7 @@ class TimedThreadApiProcessor { const SimulationConfig* const simulation_config_ = nullptr; AudioProcessing* apm_ = nullptr; AudioFrameData frame_data_; - webrtc::Clock* clock_; + Clock* clock_; const size_t num_durations_to_store_; SamplesStatsCounter api_call_durations_; size_t samples_count_ = 0; diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc index 8e32d34add..373b574d7a 100644 --- a/modules/audio_processing/audio_processing_unittest.cc +++ b/modules/audio_processing/audio_processing_unittest.cc @@ -2369,7 +2369,7 @@ std::string ProduceDebugText(int render_input_sample_rate_hz, void RunApmRateAndChannelTest(ArrayView sample_rates_hz, ArrayView render_channel_counts, ArrayView capture_channel_counts) { - webrtc::AudioProcessing::Config apm_config; + AudioProcessing::Config apm_config; apm_config.pipeline.multi_channel_render = true; apm_config.pipeline.multi_channel_capture = true; apm_config.echo_canceller.enabled = true; diff --git a/modules/audio_processing/test/aec_dump_based_simulator.cc b/modules/audio_processing/test/aec_dump_based_simulator.cc index 896c4b8be3..6a249eeecc 100644 --- a/modules/audio_processing/test/aec_dump_based_simulator.cc +++ b/modules/audio_processing/test/aec_dump_based_simulator.cc @@ -45,7 +45,7 @@ namespace { // Verify output bitexactness for the fixed interface. // TODO(peah): Check whether it would make sense to add a threshold // to use for checking the bitexactness in a soft manner. -bool VerifyFixedBitExactness(const webrtc::audioproc::Stream& msg, +bool VerifyFixedBitExactness(const audioproc::Stream& msg, const Int16Frame& frame) { if (sizeof(frame.data[0]) * frame.data.size() != msg.output_data().size()) { return false; @@ -61,7 +61,7 @@ bool VerifyFixedBitExactness(const webrtc::audioproc::Stream& msg, } // Verify output bitexactness for the float interface. -bool VerifyFloatBitExactness(const webrtc::audioproc::Stream& msg, +bool VerifyFloatBitExactness(const audioproc::Stream& msg, const StreamConfig& out_config, const ChannelBuffer& out_buf) { if (static_cast(msg.output_channel_size()) != @@ -86,7 +86,7 @@ bool VerifyFloatBitExactness(const webrtc::audioproc::Stream& msg, bool ReadNextMessage(bool use_dump_file, FILE* dump_input_file, std::stringstream& input, - webrtc::audioproc::Event& event_msg) { + audioproc::Event& event_msg) { if (use_dump_file) { return ReadMessageFromFile(dump_input_file, &event_msg); } @@ -105,7 +105,7 @@ AecDumpBasedSimulator::AecDumpBasedSimulator( AecDumpBasedSimulator::~AecDumpBasedSimulator() = default; void AecDumpBasedSimulator::PrepareProcessStreamCall( - const webrtc::audioproc::Stream& msg) { + const audioproc::Stream& msg) { if (msg.has_input_data()) { // Fixed interface processing. // Verify interface invariance. @@ -196,7 +196,7 @@ void AecDumpBasedSimulator::PrepareProcessStreamCall( } void AecDumpBasedSimulator::VerifyProcessStreamBitExactness( - const webrtc::audioproc::Stream& msg) { + const audioproc::Stream& msg) { if (bitexact_output_) { if (interface_used_ == InterfaceType::kFixedInterface) { bitexact_output_ = VerifyFixedBitExactness(msg, fwd_frame_); @@ -207,7 +207,7 @@ void AecDumpBasedSimulator::VerifyProcessStreamBitExactness( } void AecDumpBasedSimulator::PrepareReverseProcessStreamCall( - const webrtc::audioproc::ReverseStream& msg) { + const audioproc::ReverseStream& msg) { if (msg.has_data()) { // Fixed interface processing. // Verify interface invariance. @@ -267,7 +267,7 @@ void AecDumpBasedSimulator::Process() { input << settings_.aec_dump_input_string.value(); } - webrtc::audioproc::Event event_msg; + audioproc::Event event_msg; int capture_frames_since_init = 0; int init_index = 0; while (ReadNextMessage(use_dump_file, dump_input_file_, input, event_msg)) { @@ -300,12 +300,12 @@ void AecDumpBasedSimulator::Analyze() { input << settings_.aec_dump_input_string.value(); } - webrtc::audioproc::Event event_msg; + audioproc::Event event_msg; int num_capture_frames = 0; int num_render_frames = 0; int init_index = 0; while (ReadNextMessage(use_dump_file, dump_input_file_, input, event_msg)) { - if (event_msg.type() == webrtc::audioproc::Event::INIT) { + if (event_msg.type() == audioproc::Event::INIT) { ++init_index; constexpr float kNumFramesPerSecond = 100.f; float capture_time_seconds = num_capture_frames / kNumFramesPerSecond; @@ -318,9 +318,9 @@ void AecDumpBasedSimulator::Analyze() { << num_capture_frames << " frames) " << std::endl; std::cout << " Render: " << render_time_seconds << " s (" << num_render_frames << " frames) " << std::endl; - } else if (event_msg.type() == webrtc::audioproc::Event::STREAM) { + } else if (event_msg.type() == audioproc::Event::STREAM) { ++num_capture_frames; - } else if (event_msg.type() == webrtc::audioproc::Event::REVERSE_STREAM) { + } else if (event_msg.type() == audioproc::Event::REVERSE_STREAM) { ++num_render_frames; } } @@ -330,40 +330,38 @@ void AecDumpBasedSimulator::Analyze() { } } -void AecDumpBasedSimulator::HandleEvent( - const webrtc::audioproc::Event& event_msg, - int& capture_frames_since_init, - int& init_index) { +void AecDumpBasedSimulator::HandleEvent(const audioproc::Event& event_msg, + int& capture_frames_since_init, + int& init_index) { switch (event_msg.type()) { - case webrtc::audioproc::Event::INIT: + case audioproc::Event::INIT: RTC_CHECK(event_msg.has_init()); ++init_index; capture_frames_since_init = 0; HandleMessage(event_msg.init(), init_index); break; - case webrtc::audioproc::Event::STREAM: + case audioproc::Event::STREAM: RTC_CHECK(event_msg.has_stream()); ++capture_frames_since_init; HandleMessage(event_msg.stream()); break; - case webrtc::audioproc::Event::REVERSE_STREAM: + case audioproc::Event::REVERSE_STREAM: RTC_CHECK(event_msg.has_reverse_stream()); HandleMessage(event_msg.reverse_stream()); break; - case webrtc::audioproc::Event::CONFIG: + case audioproc::Event::CONFIG: RTC_CHECK(event_msg.has_config()); HandleMessage(event_msg.config()); break; - case webrtc::audioproc::Event::RUNTIME_SETTING: + case audioproc::Event::RUNTIME_SETTING: HandleMessage(event_msg.runtime_setting()); break; - case webrtc::audioproc::Event::UNKNOWN_EVENT: + case audioproc::Event::UNKNOWN_EVENT: RTC_CHECK_NOTREACHED(); } } -void AecDumpBasedSimulator::HandleMessage( - const webrtc::audioproc::Config& msg) { +void AecDumpBasedSimulator::HandleMessage(const audioproc::Config& msg) { if (settings_.use_verbose_logging) { std::cout << "Config at frame:" << std::endl; std::cout << " Forward: " << get_num_process_stream_calls() << std::endl; @@ -403,7 +401,7 @@ void AecDumpBasedSimulator::HandleMessage( } if (msg.has_aecm_routing_mode() && - static_cast( + static_cast( msg.aecm_routing_mode()) != EchoControlMobileImpl::kSpeakerphone) { RTC_LOG(LS_ERROR) << "Ignoring deprecated setting: AECM routing mode: " << msg.aecm_routing_mode(); @@ -421,8 +419,7 @@ void AecDumpBasedSimulator::HandleMessage( if (msg.has_agc_mode() || settings_.agc_mode) { int mode = settings_.agc_mode ? *settings_.agc_mode : msg.agc_mode(); apm_config.gain_controller1.mode = - static_cast( - mode); + static_cast(mode); if (settings_.use_verbose_logging) { std::cout << " agc_mode: " << mode << std::endl; } @@ -524,7 +521,7 @@ void AecDumpBasedSimulator::HandleMessage( } } -void AecDumpBasedSimulator::HandleMessage(const webrtc::audioproc::Init& msg, +void AecDumpBasedSimulator::HandleMessage(const audioproc::Init& msg, int init_index) { RTC_CHECK(msg.has_sample_rate()); RTC_CHECK(msg.has_num_input_channels()); @@ -586,8 +583,7 @@ void AecDumpBasedSimulator::HandleMessage(const webrtc::audioproc::Init& msg, msg.num_reverse_channels(), num_reverse_output_channels); } -void AecDumpBasedSimulator::HandleMessage( - const webrtc::audioproc::Stream& msg) { +void AecDumpBasedSimulator::HandleMessage(const audioproc::Stream& msg) { if (call_order_output_file_) { *call_order_output_file_ << "c"; } @@ -596,8 +592,7 @@ void AecDumpBasedSimulator::HandleMessage( VerifyProcessStreamBitExactness(msg); } -void AecDumpBasedSimulator::HandleMessage( - const webrtc::audioproc::ReverseStream& msg) { +void AecDumpBasedSimulator::HandleMessage(const audioproc::ReverseStream& msg) { if (call_order_output_file_) { *call_order_output_file_ << "r"; } @@ -606,7 +601,7 @@ void AecDumpBasedSimulator::HandleMessage( } void AecDumpBasedSimulator::HandleMessage( - const webrtc::audioproc::RuntimeSetting& msg) { + const audioproc::RuntimeSetting& msg) { RTC_CHECK(ap_.get()); if (msg.has_capture_pre_gain()) { // Handle capture pre-gain runtime setting only if not overridden. diff --git a/modules/audio_processing/test/audio_processing_simulator.cc b/modules/audio_processing/test/audio_processing_simulator.cc index a273083720..914edb775f 100644 --- a/modules/audio_processing/test/audio_processing_simulator.cc +++ b/modules/audio_processing/test/audio_processing_simulator.cc @@ -503,7 +503,7 @@ void AudioProcessingSimulator::ConfigureAudioProcessor() { } if (settings_.agc_mode) { apm_config.gain_controller1.mode = - static_cast( + static_cast( *settings_.agc_mode); } if (settings_.use_agc_limiter) { diff --git a/modules/audio_processing/test/performance_timer.cc b/modules/audio_processing/test/performance_timer.cc index 1a82258903..cc12e00d6c 100644 --- a/modules/audio_processing/test/performance_timer.cc +++ b/modules/audio_processing/test/performance_timer.cc @@ -20,7 +20,7 @@ namespace webrtc { namespace test { PerformanceTimer::PerformanceTimer(int num_frames_to_process) - : clock_(webrtc::Clock::GetRealTimeClock()) { + : clock_(Clock::GetRealTimeClock()) { timestamps_us_.reserve(num_frames_to_process); } diff --git a/modules/audio_processing/test/runtime_setting_util.cc b/modules/audio_processing/test/runtime_setting_util.cc index 4899d2d459..2240768d74 100644 --- a/modules/audio_processing/test/runtime_setting_util.cc +++ b/modules/audio_processing/test/runtime_setting_util.cc @@ -15,7 +15,7 @@ namespace webrtc { void ReplayRuntimeSetting(AudioProcessing* apm, - const webrtc::audioproc::RuntimeSetting& setting) { + const audioproc::RuntimeSetting& setting) { RTC_CHECK(apm); // TODO(bugs.webrtc.org/9138): Add ability to handle different types // of settings. Currently CapturePreGain, CaptureFixedPostGain and From a9f96a4221e36b57f5b5aa62ffa5260485d542b1 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Fri, 9 May 2025 10:37:58 +0000 Subject: [PATCH 102/292] [cleanup] Fix redundant webrtc name specifier This CL was uploaded by git cl split. R=hta@webrtc.org No-IWYU: LSC Bug: webrtc:42232595 Change-Id: I517c8776a2d5dab65a34cbf0b6e51e5581a013e9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390642 Auto-Submit: Evan Shrubsole Commit-Queue: Evan Shrubsole Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44574} --- .../remote_bitrate_estimator/aimd_rate_control_unittest.cc | 2 +- .../congestion_control_feedback_generator.cc | 2 +- .../transport_sequence_number_feedback_generator_unittest.cc | 3 +-- test/fuzzers/utils/rtp_replayer.cc | 4 ++-- 4 files changed, 5 insertions(+), 6 deletions(-) diff --git a/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc b/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc index 26c0a869da..6a830f1418 100644 --- a/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc +++ b/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc @@ -22,7 +22,7 @@ namespace webrtc { namespace { -using ::webrtc::test::ExplicitKeyValueConfig; +using test::ExplicitKeyValueConfig; constexpr Timestamp kInitialTime = Timestamp::Millis(123'456); diff --git a/modules/remote_bitrate_estimator/congestion_control_feedback_generator.cc b/modules/remote_bitrate_estimator/congestion_control_feedback_generator.cc index bf13fa352f..fbfbf6f890 100644 --- a/modules/remote_bitrate_estimator/congestion_control_feedback_generator.cc +++ b/modules/remote_bitrate_estimator/congestion_control_feedback_generator.cc @@ -32,7 +32,7 @@ namespace webrtc { -constexpr DataRate kMaxFeedbackRate = webrtc::DataRate::KilobitsPerSec(500); +constexpr DataRate kMaxFeedbackRate = DataRate::KilobitsPerSec(500); CongestionControlFeedbackGenerator::CongestionControlFeedbackGenerator( const Environment& env, diff --git a/modules/remote_bitrate_estimator/transport_sequence_number_feedback_generator_unittest.cc b/modules/remote_bitrate_estimator/transport_sequence_number_feedback_generator_unittest.cc index 3c908af6bc..2f683be1ca 100644 --- a/modules/remote_bitrate_estimator/transport_sequence_number_feedback_generator_unittest.cc +++ b/modules/remote_bitrate_estimator/transport_sequence_number_feedback_generator_unittest.cc @@ -107,8 +107,7 @@ class TransportSequenceNumberFeedbackGeneneratorTest : public ::testing::Test { map.Register(1); RtpPacketReceived packet(&map, arrival_time); packet.SetSsrc(kMediaSsrc); - packet.SetExtension(seq, - feedback_request); + packet.SetExtension(seq, feedback_request); feedback_generator_.OnReceivedPacket(packet); } diff --git a/test/fuzzers/utils/rtp_replayer.cc b/test/fuzzers/utils/rtp_replayer.cc index 6af6d330de..081d31ba80 100644 --- a/test/fuzzers/utils/rtp_replayer.cc +++ b/test/fuzzers/utils/rtp_replayer.cc @@ -52,7 +52,7 @@ void RtpReplayer::Replay( // Work around: webrtc calls webrtc::Random(clock.TimeInMicroseconds()) // everywhere and Random expects non-zero seed. Let's set the clock non-zero // to make them happy. - fake_clock.SetTime(webrtc::Timestamp::Millis(1)); + fake_clock.SetTime(Timestamp::Millis(1)); // Attempt to create an RtpReader from the input file. auto rtp_reader = CreateRtpReader(rtp_dump_data, rtp_dump_size); @@ -170,7 +170,7 @@ void RtpReplayer::ReplayPackets( if (deliver_in_ms > 0) { // StatsCounter::ReportMetricToAggregatedCounter is O(elapsed time). // Set an upper limit to prevent waste time. - clock->AdvanceTime(webrtc::TimeDelta::Millis( + clock->AdvanceTime(TimeDelta::Millis( std::min(deliver_in_ms, static_cast(100)))); } From 318cc33c71536717c536dd997bca01fa0596848b Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Fri, 9 May 2025 10:38:33 +0000 Subject: [PATCH 103/292] [cleanup] Fix redundant webrtc name specifier This CL was uploaded by git cl split. R=hta@webrtc.org No-IWYU: LSC Bug: webrtc:42232595 Change-Id: I34e14bba2b09d3b344d434b9779effffba7b745c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390606 Commit-Queue: Evan Shrubsole Reviewed-by: Harald Alvestrand Auto-Submit: Evan Shrubsole Cr-Commit-Position: refs/heads/main@{#44575} --- rtc_base/async_dns_resolver.cc | 8 +- rtc_base/async_dns_resolver_unittest.cc | 2 +- rtc_base/async_packet_socket.cc | 5 +- rtc_base/async_packet_socket_unittest.cc | 7 +- rtc_base/async_tcp_socket.cc | 4 +- rtc_base/async_udp_socket.cc | 4 +- rtc_base/bit_buffer_unittest.cc | 1 - rtc_base/boringssl_certificate.cc | 2 +- rtc_base/boringssl_identity.cc | 5 +- rtc_base/byte_buffer.cc | 8 +- rtc_base/byte_buffer_unittest.cc | 38 +- rtc_base/containers/flat_set_unittest.cc | 2 +- rtc_base/cpu_time.cc | 4 +- rtc_base/cpu_time_unittest.cc | 8 +- rtc_base/event.cc | 4 +- rtc_base/file_rotating_stream_unittest.cc | 25 +- rtc_base/ifaddrs_converter.cc | 4 +- rtc_base/ip_address.cc | 18 +- rtc_base/ip_address_unittest.cc | 2 +- rtc_base/memory/fifo_buffer.cc | 32 +- rtc_base/memory_stream.cc | 18 +- rtc_base/network.cc | 198 +++++---- rtc_base/network_unittest.cc | 257 ++++++------ rtc_base/null_socket_server_unittest.cc | 8 +- rtc_base/openssl_adapter.cc | 20 +- rtc_base/openssl_digest.cc | 24 +- rtc_base/openssl_stream_adapter.cc | 191 +++++---- rtc_base/openssl_utility.cc | 9 +- rtc_base/openssl_utility_unittest.cc | 2 +- rtc_base/operations_chain_unittest.cc | 20 +- rtc_base/physical_socket_server.cc | 23 +- rtc_base/physical_socket_server_unittest.cc | 30 +- rtc_base/rate_tracker.cc | 6 +- .../rtc_certificate_generator_unittest.cc | 10 +- rtc_base/rtc_certificate_unittest.cc | 2 +- rtc_base/socket_address.cc | 42 +- rtc_base/socket_unittest.cc | 239 +++++------ rtc_base/ssl_fingerprint.cc | 2 +- rtc_base/ssl_identity.cc | 2 +- rtc_base/ssl_stream_adapter.cc | 2 +- rtc_base/strings/string_builder_unittest.cc | 32 +- rtc_base/strong_alias_unittest.cc | 2 +- rtc_base/synchronization/mutex_unittest.cc | 2 - .../sequence_checker_internal.cc | 6 +- rtc_base/task_queue_stdlib_unittest.cc | 2 +- rtc_base/test_client.cc | 2 +- rtc_base/test_client_unittest.cc | 2 +- rtc_base/thread.cc | 18 +- rtc_base/thread_unittest.cc | 30 +- rtc_base/unique_id_generator.cc | 2 +- rtc_base/unique_id_generator_unittest.cc | 4 +- rtc_base/virtual_socket_server.cc | 26 +- rtc_base/virtual_socket_unittest.cc | 379 +++++++++--------- 53 files changed, 843 insertions(+), 952 deletions(-) diff --git a/rtc_base/async_dns_resolver.cc b/rtc_base/async_dns_resolver.cc index 625721a2d9..9c9610d47d 100644 --- a/rtc_base/async_dns_resolver.cc +++ b/rtc_base/async_dns_resolver.cc @@ -112,7 +112,7 @@ class AsyncDnsResolver::State : public RefCountedBase { // Execute the passed function if the state is Active. void Finish(absl::AnyInvocable function) { - webrtc::MutexLock lock(&mutex_); + MutexLock lock(&mutex_); if (status_ != Status::kActive) { return; } @@ -120,12 +120,12 @@ class AsyncDnsResolver::State : public RefCountedBase { function(); } void Kill() { - webrtc::MutexLock lock(&mutex_); + MutexLock lock(&mutex_); status_ = Status::kDead; } private: - webrtc::Mutex mutex_; + Mutex mutex_; Status status_ RTC_GUARDED_BY(mutex_) = Status::kActive; }; @@ -148,7 +148,7 @@ void AsyncDnsResolver::Start(const SocketAddress& addr, result_.addr_ = addr; callback_ = std::move(callback); auto thread_function = [this, addr, family, flag = safety_.flag(), - caller_task_queue = webrtc::TaskQueueBase::Current(), + caller_task_queue = TaskQueueBase::Current(), state = state_] { std::vector addresses; int error = ResolveHostname(addr.hostname(), family, addresses); diff --git a/rtc_base/async_dns_resolver_unittest.cc b/rtc_base/async_dns_resolver_unittest.cc index e9f1d2b463..ae03b204b2 100644 --- a/rtc_base/async_dns_resolver_unittest.cc +++ b/rtc_base/async_dns_resolver_unittest.cc @@ -28,7 +28,7 @@ namespace { using ::testing::IsTrue; -const webrtc::TimeDelta kDefaultTimeout = webrtc::TimeDelta::Millis(1000); +const TimeDelta kDefaultTimeout = TimeDelta::Millis(1000); const int kPortNumber = 3027; TEST(AsyncDnsResolver, ConstructorWorks) { diff --git a/rtc_base/async_packet_socket.cc b/rtc_base/async_packet_socket.cc index 2ead920705..397e197ec6 100644 --- a/rtc_base/async_packet_socket.cc +++ b/rtc_base/async_packet_socket.cc @@ -41,7 +41,7 @@ AsyncPacketSocket::~AsyncPacketSocket() = default; void AsyncPacketSocket::SubscribeCloseEvent( const void* removal_tag, - std::function callback) { + std::function callback) { RTC_DCHECK_RUN_ON(&network_checker_); on_close_.AddReceiver(removal_tag, std::move(callback)); } @@ -52,8 +52,7 @@ void AsyncPacketSocket::UnsubscribeCloseEvent(const void* removal_tag) { } void AsyncPacketSocket::RegisterReceivedPacketCallback( - absl::AnyInvocable + absl::AnyInvocable received_packet_callback) { RTC_DCHECK_RUN_ON(&network_checker_); RTC_CHECK(!received_packet_callback_); diff --git a/rtc_base/async_packet_socket_unittest.cc b/rtc_base/async_packet_socket_unittest.cc index 9ed491efb5..b37334aa4e 100644 --- a/rtc_base/async_packet_socket_unittest.cc +++ b/rtc_base/async_packet_socket_unittest.cc @@ -30,7 +30,7 @@ class MockAsyncPacketSocket : public AsyncPacketSocket { Send, (const void* pv, size_t cb, - const webrtc::AsyncSocketPacketOptions& options), + const AsyncSocketPacketOptions& options), (override)); MOCK_METHOD(int, @@ -38,7 +38,7 @@ class MockAsyncPacketSocket : public AsyncPacketSocket { (const void* pv, size_t cb, const SocketAddress& addr, - const webrtc::AsyncSocketPacketOptions& options), + const AsyncSocketPacketOptions& options), (override)); MOCK_METHOD(int, Close, (), (override)); MOCK_METHOD(State, GetState, (), (const, override)); @@ -52,8 +52,7 @@ class MockAsyncPacketSocket : public AsyncPacketSocket { TEST(AsyncPacketSocket, RegisteredCallbackReceivePacketsFromNotify) { MockAsyncPacketSocket mock_socket; - MockFunction + MockFunction received_packet; EXPECT_CALL(received_packet, Call); diff --git a/rtc_base/async_tcp_socket.cc b/rtc_base/async_tcp_socket.cc index 420d3595ca..e07bfc1f20 100644 --- a/rtc_base/async_tcp_socket.cc +++ b/rtc_base/async_tcp_socket.cc @@ -266,7 +266,7 @@ int AsyncTCPSocket::Send(const void* pv, if (!IsOutBufferEmpty()) return static_cast(cb); - PacketLength pkt_len = webrtc::HostToNetwork16(static_cast(cb)); + PacketLength pkt_len = HostToNetwork16(static_cast(cb)); AppendToOutBuffer(&pkt_len, kPacketLenSize); AppendToOutBuffer(pv, cb); @@ -295,7 +295,7 @@ size_t AsyncTCPSocket::ProcessInput(ArrayView data) { if (bytes_left < kPacketLenSize) return processed_bytes; - PacketLength pkt_len = webrtc::GetBE16(data.data() + processed_bytes); + PacketLength pkt_len = GetBE16(data.data() + processed_bytes); if (bytes_left < kPacketLenSize + pkt_len) return processed_bytes; diff --git a/rtc_base/async_udp_socket.cc b/rtc_base/async_udp_socket.cc index f0846875f9..652c104933 100644 --- a/rtc_base/async_udp_socket.cc +++ b/rtc_base/async_udp_socket.cc @@ -67,7 +67,7 @@ int AsyncUDPSocket::Send(const void* pv, const AsyncSocketPacketOptions& options) { SentPacketInfo sent_packet(options.packet_id, TimeMillis(), options.info_signaled_after_sent); - webrtc::CopySocketInformationToPacketInfo(cb, *this, &sent_packet.info); + CopySocketInformationToPacketInfo(cb, *this, &sent_packet.info); int ret = socket_->Send(pv, cb); SignalSentPacket(this, sent_packet); return ret; @@ -79,7 +79,7 @@ int AsyncUDPSocket::SendTo(const void* pv, const AsyncSocketPacketOptions& options) { SentPacketInfo sent_packet(options.packet_id, TimeMillis(), options.info_signaled_after_sent); - webrtc::CopySocketInformationToPacketInfo(cb, *this, &sent_packet.info); + CopySocketInformationToPacketInfo(cb, *this, &sent_packet.info); if (has_set_ect1_options_ != options.ecn_1) { // It is unclear what is most efficient, setting options on every sent // packet or when changed. Potentially, can separate send sockets be used? diff --git a/rtc_base/bit_buffer_unittest.cc b/rtc_base/bit_buffer_unittest.cc index 923a3df30a..670275c879 100644 --- a/rtc_base/bit_buffer_unittest.cc +++ b/rtc_base/bit_buffer_unittest.cc @@ -22,7 +22,6 @@ namespace webrtc { using ::testing::ElementsAre; -using ::webrtc::BitstreamReader; TEST(BitBufferWriterTest, ConsumeBits) { uint8_t bytes[64] = {0}; diff --git a/rtc_base/boringssl_certificate.cc b/rtc_base/boringssl_certificate.cc index 016b4dabba..4a34d05448 100644 --- a/rtc_base/boringssl_certificate.cc +++ b/rtc_base/boringssl_certificate.cc @@ -265,7 +265,7 @@ std::unique_ptr BoringSSLCertificate::Generate( SSLIdentityParams actual_params(params); if (actual_params.common_name.empty()) { // Use a random string, arbitrarily 8 chars long. - actual_params.common_name = webrtc::CreateRandomString(8); + actual_params.common_name = CreateRandomString(8); } bssl::UniquePtr cert_buffer = MakeCertificate(key_pair->pkey(), actual_params); diff --git a/rtc_base/boringssl_identity.cc b/rtc_base/boringssl_identity.cc index 5c8f60e362..4406962fac 100644 --- a/rtc_base/boringssl_identity.cc +++ b/rtc_base/boringssl_identity.cc @@ -116,9 +116,8 @@ std::unique_ptr BoringSSLIdentity::CreateFromPEMStrings( std::unique_ptr BoringSSLIdentity::CreateFromPEMChainStrings( absl::string_view private_key, absl::string_view certificate_chain) { - bssl::UniquePtr bio( - BIO_new_mem_buf(certificate_chain.data(), - webrtc::dchecked_cast(certificate_chain.size()))); + bssl::UniquePtr bio(BIO_new_mem_buf( + certificate_chain.data(), dchecked_cast(certificate_chain.size()))); if (!bio) { return nullptr; } diff --git a/rtc_base/byte_buffer.cc b/rtc_base/byte_buffer.cc index 0f5ea525f6..0cf153fe7e 100644 --- a/rtc_base/byte_buffer.cc +++ b/rtc_base/byte_buffer.cc @@ -56,7 +56,7 @@ bool ByteBufferReader::ReadUInt16(uint16_t* val) { if (!ReadBytes(reinterpret_cast(&v), 2)) { return false; } else { - *val = webrtc::NetworkToHost16(v); + *val = NetworkToHost16(v); return true; } } @@ -72,7 +72,7 @@ bool ByteBufferReader::ReadUInt24(uint32_t* val) { if (!ReadBytes(read_into, 3)) { return false; } else { - *val = webrtc::NetworkToHost32(v); + *val = NetworkToHost32(v); return true; } } @@ -85,7 +85,7 @@ bool ByteBufferReader::ReadUInt32(uint32_t* val) { if (!ReadBytes(reinterpret_cast(&v), 4)) { return false; } else { - *val = webrtc::NetworkToHost32(v); + *val = NetworkToHost32(v); return true; } } @@ -98,7 +98,7 @@ bool ByteBufferReader::ReadUInt64(uint64_t* val) { if (!ReadBytes(reinterpret_cast(&v), 8)) { return false; } else { - *val = webrtc::NetworkToHost64(v); + *val = NetworkToHost64(v); return true; } } diff --git a/rtc_base/byte_buffer_unittest.cc b/rtc_base/byte_buffer_unittest.cc index 6bf532deb9..d0b93e2e22 100644 --- a/rtc_base/byte_buffer_unittest.cc +++ b/rtc_base/byte_buffer_unittest.cc @@ -46,35 +46,35 @@ TEST(ByteBufferTest, TestByteOrder) { uint32_t n32 = 1; uint64_t n64 = 1; - EXPECT_EQ(n16, webrtc::NetworkToHost16(webrtc::HostToNetwork16(n16))); - EXPECT_EQ(n32, webrtc::NetworkToHost32(webrtc::HostToNetwork32(n32))); - EXPECT_EQ(n64, webrtc::NetworkToHost64(webrtc::HostToNetwork64(n64))); + EXPECT_EQ(n16, NetworkToHost16(HostToNetwork16(n16))); + EXPECT_EQ(n32, NetworkToHost32(HostToNetwork32(n32))); + EXPECT_EQ(n64, NetworkToHost64(HostToNetwork64(n64))); - if (webrtc::IsHostBigEndian()) { + if (IsHostBigEndian()) { // The host is the network (big) endian. - EXPECT_EQ(n16, webrtc::HostToNetwork16(n16)); - EXPECT_EQ(n32, webrtc::HostToNetwork32(n32)); - EXPECT_EQ(n64, webrtc::HostToNetwork64(n64)); + EXPECT_EQ(n16, HostToNetwork16(n16)); + EXPECT_EQ(n32, HostToNetwork32(n32)); + EXPECT_EQ(n64, HostToNetwork64(n64)); // GetBE converts big endian to little endian here. - EXPECT_EQ(n16 >> 8, webrtc::GetBE16(&n16)); - EXPECT_EQ(n32 >> 24, webrtc::GetBE32(&n32)); - EXPECT_EQ(n64 >> 56, webrtc::GetBE64(&n64)); + EXPECT_EQ(n16 >> 8, GetBE16(&n16)); + EXPECT_EQ(n32 >> 24, GetBE32(&n32)); + EXPECT_EQ(n64 >> 56, GetBE64(&n64)); } else { // The host is little endian. - EXPECT_NE(n16, webrtc::HostToNetwork16(n16)); - EXPECT_NE(n32, webrtc::HostToNetwork32(n32)); - EXPECT_NE(n64, webrtc::HostToNetwork64(n64)); + EXPECT_NE(n16, HostToNetwork16(n16)); + EXPECT_NE(n32, HostToNetwork32(n32)); + EXPECT_NE(n64, HostToNetwork64(n64)); // GetBE converts little endian to big endian here. - EXPECT_EQ(webrtc::GetBE16(&n16), webrtc::HostToNetwork16(n16)); - EXPECT_EQ(webrtc::GetBE32(&n32), webrtc::HostToNetwork32(n32)); - EXPECT_EQ(webrtc::GetBE64(&n64), webrtc::HostToNetwork64(n64)); + EXPECT_EQ(GetBE16(&n16), HostToNetwork16(n16)); + EXPECT_EQ(GetBE32(&n32), HostToNetwork32(n32)); + EXPECT_EQ(GetBE64(&n64), HostToNetwork64(n64)); // GetBE converts little endian to big endian here. - EXPECT_EQ(n16 << 8, webrtc::GetBE16(&n16)); - EXPECT_EQ(n32 << 24, webrtc::GetBE32(&n32)); - EXPECT_EQ(n64 << 56, webrtc::GetBE64(&n64)); + EXPECT_EQ(n16 << 8, GetBE16(&n16)); + EXPECT_EQ(n32 << 24, GetBE32(&n32)); + EXPECT_EQ(n64 << 56, GetBE64(&n64)); } } diff --git a/rtc_base/containers/flat_set_unittest.cc b/rtc_base/containers/flat_set_unittest.cc index 22c051a3c8..6b0a2f7203 100644 --- a/rtc_base/containers/flat_set_unittest.cc +++ b/rtc_base/containers/flat_set_unittest.cc @@ -99,7 +99,7 @@ TEST(FlatSet, CopySwap) { } TEST(FlatSet, UsingTransparentCompare) { - using ExplicitInt = webrtc::MoveOnlyInt; + using ExplicitInt = MoveOnlyInt; flat_set s; const auto& s1 = s; int x = 0; diff --git a/rtc_base/cpu_time.cc b/rtc_base/cpu_time.cc index b0d187c411..0ae890632e 100644 --- a/rtc_base/cpu_time.cc +++ b/rtc_base/cpu_time.cc @@ -58,7 +58,7 @@ int64_t GetProcessCpuTimeNanos() { #elif defined(WEBRTC_LINUX) struct timespec ts; if (clock_gettime(CLOCK_PROCESS_CPUTIME_ID, &ts) == 0) { - return ts.tv_sec * webrtc::kNumNanosecsPerSec + ts.tv_nsec; + return ts.tv_sec * kNumNanosecsPerSec + ts.tv_nsec; } else { RTC_LOG_ERR(LS_ERROR) << "clock_gettime() failed."; } @@ -106,7 +106,7 @@ int64_t GetThreadCpuTimeNanos() { #elif defined(WEBRTC_LINUX) struct timespec ts; if (clock_gettime(CLOCK_THREAD_CPUTIME_ID, &ts) == 0) { - return ts.tv_sec * webrtc::kNumNanosecsPerSec + ts.tv_nsec; + return ts.tv_sec * kNumNanosecsPerSec + ts.tv_nsec; } else { RTC_LOG_ERR(LS_ERROR) << "clock_gettime() failed."; } diff --git a/rtc_base/cpu_time_unittest.cc b/rtc_base/cpu_time_unittest.cc index fbdd5faa39..9d2f4adac4 100644 --- a/rtc_base/cpu_time_unittest.cc +++ b/rtc_base/cpu_time_unittest.cc @@ -80,25 +80,25 @@ TEST(CpuTimeTest, MAYBE_TEST(TwoThreads)) { // Therefore GetThreadCpuTime is not a wall clock. EXPECT_LE(thread_duration_nanos, (kProcessingTimeMillisecs - kAllowedErrorMillisecs) * - webrtc::kNumNanosecsPerMillisec); + kNumNanosecsPerMillisec); // Total process time is at least twice working threads' CPU time. // Therefore process and thread times are correctly related. EXPECT_GE(process_duration_nanos, kWorkingThreads * (kProcessingTimeMillisecs - kAllowedErrorMillisecs) * - webrtc::kNumNanosecsPerMillisec); + kNumNanosecsPerMillisec); } TEST(CpuTimeTest, MAYBE_TEST(Sleeping)) { int64_t process_start_time_nanos = GetProcessCpuTimeNanos(); - webrtc::SleepMs(kProcessingTimeMillisecs); + SleepMs(kProcessingTimeMillisecs); int64_t process_duration_nanos = GetProcessCpuTimeNanos() - process_start_time_nanos; // Sleeping should not introduce any additional CPU time. // Therefore GetProcessCpuTime is not a wall clock. EXPECT_LE(process_duration_nanos, (kProcessingTimeMillisecs - kAllowedErrorMillisecs) * - webrtc::kNumNanosecsPerMillisec); + kNumNanosecsPerMillisec); } } // namespace webrtc diff --git a/rtc_base/event.cc b/rtc_base/event.cc index b78e249cc0..7338670d66 100644 --- a/rtc_base/event.cc +++ b/rtc_base/event.cc @@ -30,8 +30,6 @@ namespace webrtc { -using ::webrtc::TimeDelta; - Event::Event() : Event(false, false) {} #if defined(WEBRTC_WIN) @@ -189,7 +187,7 @@ bool Event::Wait(TimeDelta give_up_after, TimeDelta warn_after) { } else { error = wait(warn_ts); if (error == ETIMEDOUT) { - webrtc::WarnThatTheCurrentThreadIsProbablyDeadlocked(); + WarnThatTheCurrentThreadIsProbablyDeadlocked(); error = wait(give_up_ts); } } diff --git a/rtc_base/file_rotating_stream_unittest.cc b/rtc_base/file_rotating_stream_unittest.cc index 1422fd1db1..4ee54dc0a4 100644 --- a/rtc_base/file_rotating_stream_unittest.cc +++ b/rtc_base/file_rotating_stream_unittest.cc @@ -30,7 +30,7 @@ namespace { void CleanupLogDirectory(const FileRotatingStream& stream) { for (size_t i = 0; i < stream.GetNumFiles(); ++i) { // Ignore return value, not all files are expected to exist. - webrtc::test::RemoveFile(stream.GetFilePath(i)); + test::RemoveFile(stream.GetFilePath(i)); } } @@ -53,14 +53,14 @@ class MAYBE_FileRotatingStreamTest : public ::testing::Test { size_t max_file_size, size_t num_log_files, bool ensure_trailing_delimiter = true) { - dir_path_ = webrtc::test::OutputPath(); + dir_path_ = test::OutputPath(); // Append per-test output path in order to run within gtest parallel. dir_path_.append(dir_name.begin(), dir_name.end()); if (ensure_trailing_delimiter) { - dir_path_.append(std::string(webrtc::test::kPathDelimiter)); + dir_path_.append(std::string(test::kPathDelimiter)); } - ASSERT_TRUE(webrtc::test::CreateDir(dir_path_)); + ASSERT_TRUE(test::CreateDir(dir_path_)); stream_.reset(new FileRotatingStream(dir_path_, file_prefix, max_file_size, num_log_files)); } @@ -69,7 +69,7 @@ class MAYBE_FileRotatingStreamTest : public ::testing::Test { // On windows, open files can't be removed. stream_->Close(); CleanupLogDirectory(*stream_); - EXPECT_TRUE(webrtc::test::RemoveDir(dir_path_)); + EXPECT_TRUE(test::RemoveDir(dir_path_)); stream_.reset(); } @@ -158,12 +158,12 @@ TEST_F(MAYBE_FileRotatingStreamTest, WriteAndRead) { } // Check that exactly three files exist. for (size_t i = 0; i < arraysize(messages); ++i) { - EXPECT_TRUE(webrtc::test::FileExists(stream_->GetFilePath(i))); + EXPECT_TRUE(test::FileExists(stream_->GetFilePath(i))); } std::string message("d"); WriteAndFlush(message.c_str(), message.size()); for (size_t i = 0; i < arraysize(messages); ++i) { - EXPECT_TRUE(webrtc::test::FileExists(stream_->GetFilePath(i))); + EXPECT_TRUE(test::FileExists(stream_->GetFilePath(i))); } // TODO(tkchin): Maybe check all the files in the dir. @@ -193,8 +193,7 @@ TEST_F(MAYBE_FileRotatingStreamTest, WriteWithoutDelimiterAndRead) { // Reopen for read. std::string expected_contents("bbccd"); VerifyStreamRead(expected_contents, - dir_path_ + std::string(webrtc::test::kPathDelimiter), - kFilePrefix); + dir_path_ + std::string(test::kPathDelimiter), kFilePrefix); } // Tests that a write operation followed by a read (without trailing delimiter) @@ -258,12 +257,12 @@ TEST_F(MAYBE_FileRotatingStreamTest, GetFilePath) { class MAYBE_CallSessionFileRotatingStreamTest : public ::testing::Test { protected: void Init(absl::string_view dir_name, size_t max_total_log_size) { - dir_path_ = webrtc::test::OutputPath(); + dir_path_ = test::OutputPath(); // Append per-test output path in order to run within gtest parallel. dir_path_.append(dir_name.begin(), dir_name.end()); - dir_path_.append(std::string(webrtc::test::kPathDelimiter)); - ASSERT_TRUE(webrtc::test::CreateDir(dir_path_)); + dir_path_.append(std::string(test::kPathDelimiter)); + ASSERT_TRUE(test::CreateDir(dir_path_)); stream_.reset( new CallSessionFileRotatingStream(dir_path_, max_total_log_size)); } @@ -272,7 +271,7 @@ class MAYBE_CallSessionFileRotatingStreamTest : public ::testing::Test { // On windows, open files can't be removed. stream_->Close(); CleanupLogDirectory(*stream_); - EXPECT_TRUE(webrtc::test::RemoveDir(dir_path_)); + EXPECT_TRUE(test::RemoveDir(dir_path_)); stream_.reset(); } diff --git a/rtc_base/ifaddrs_converter.cc b/rtc_base/ifaddrs_converter.cc index de949aa575..8b50e3d508 100644 --- a/rtc_base/ifaddrs_converter.cc +++ b/rtc_base/ifaddrs_converter.cc @@ -31,7 +31,7 @@ bool IfAddrsConverter::ConvertIfAddrsToIPAddress( return true; } case AF_INET6: { - int ip_attributes = webrtc::IPV6_ADDRESS_FLAG_NONE; + int ip_attributes = IPV6_ADDRESS_FLAG_NONE; if (!ConvertNativeAttributesToIPAttributes(interface, &ip_attributes)) { return false; } @@ -51,7 +51,7 @@ bool IfAddrsConverter::ConvertIfAddrsToIPAddress( bool IfAddrsConverter::ConvertNativeAttributesToIPAttributes( const struct ifaddrs* /* interface */, int* ip_attributes) { - *ip_attributes = webrtc::IPV6_ADDRESS_FLAG_NONE; + *ip_attributes = IPV6_ADDRESS_FLAG_NONE; return true; } diff --git a/rtc_base/ip_address.cc b/rtc_base/ip_address.cc index 92be158e70..5c349dc1f8 100644 --- a/rtc_base/ip_address.cc +++ b/rtc_base/ip_address.cc @@ -46,7 +46,7 @@ static in_addr ExtractMappedAddress(const in6_addr& addr); uint32_t IPAddress::v4AddressAsHostOrderInteger() const { if (family_ == AF_INET) { - return webrtc::NetworkToHost32(u_.ip4.s_addr); + return NetworkToHost32(u_.ip4.s_addr); } else { return 0; } @@ -112,8 +112,8 @@ bool IPAddress::operator<(const IPAddress& other) const { // Comparing addresses of the same family. switch (family_) { case AF_INET: { - return webrtc::NetworkToHost32(u_.ip4.s_addr) < - webrtc::NetworkToHost32(other.u_.ip4.s_addr); + return NetworkToHost32(u_.ip4.s_addr) < + NetworkToHost32(other.u_.ip4.s_addr); } case AF_INET6: { return memcmp(&u_.ip6.s6_addr, &other.u_.ip6.s6_addr, 16) < 0; @@ -371,9 +371,9 @@ IPAddress TruncateIP(const IPAddress& ip, int length) { return IPAddress(INADDR_ANY); } int mask = (0xFFFFFFFF << (32 - length)); - uint32_t host_order_ip = webrtc::NetworkToHost32(ip.ipv4_address().s_addr); + uint32_t host_order_ip = NetworkToHost32(ip.ipv4_address().s_addr); in_addr masked; - masked.s_addr = webrtc::HostToNetwork32(host_order_ip & mask); + masked.s_addr = HostToNetwork32(host_order_ip & mask); return IPAddress(masked); } else if (ip.family() == AF_INET6) { if (length > 127) { @@ -390,8 +390,8 @@ IPAddress TruncateIP(const IPAddress& ip, int length) { uint32_t* v6_as_ints = reinterpret_cast(&v6addr.s6_addr); for (int i = 0; i < 4; ++i) { if (i == position) { - uint32_t host_order_inner = webrtc::NetworkToHost32(v6_as_ints[i]); - v6_as_ints[i] = webrtc::HostToNetwork32(host_order_inner & inner_mask); + uint32_t host_order_inner = NetworkToHost32(v6_as_ints[i]); + v6_as_ints[i] = HostToNetwork32(host_order_inner & inner_mask); } else if (i > position) { v6_as_ints[i] = 0; } @@ -406,7 +406,7 @@ int CountIPMaskBits(const IPAddress& mask) { int bits = 0; switch (mask.family()) { case AF_INET: { - word_to_count = webrtc::NetworkToHost32(mask.ipv4_address().s_addr); + word_to_count = NetworkToHost32(mask.ipv4_address().s_addr); break; } case AF_INET6: { @@ -420,7 +420,7 @@ int CountIPMaskBits(const IPAddress& mask) { } } if (i < 4) { - word_to_count = webrtc::NetworkToHost32(v6_as_ints[i]); + word_to_count = NetworkToHost32(v6_as_ints[i]); } bits = (i * 32); break; diff --git a/rtc_base/ip_address_unittest.cc b/rtc_base/ip_address_unittest.cc index 5f5988d260..ff6a1829a9 100644 --- a/rtc_base/ip_address_unittest.cc +++ b/rtc_base/ip_address_unittest.cc @@ -522,7 +522,7 @@ TEST(IPAddressTest, TestIPFromAddrInfo) { test_info.ai_next = &next_info; // Check that we can get an IPv4 address out. test_info.ai_addr = reinterpret_cast(&expected4); - expected4.sin_addr.s_addr = webrtc::HostToNetwork32(kIPv4PublicAddr); + expected4.sin_addr.s_addr = HostToNetwork32(kIPv4PublicAddr); expected4.sin_family = AF_INET; IPAddress expected(kIPv4PublicAddr); IPAddress addr; diff --git a/rtc_base/memory/fifo_buffer.cc b/rtc_base/memory/fifo_buffer.cc index 8c102001e3..4eb0dfa054 100644 --- a/rtc_base/memory/fifo_buffer.cc +++ b/rtc_base/memory/fifo_buffer.cc @@ -24,7 +24,7 @@ namespace webrtc { FifoBuffer::FifoBuffer(size_t size) - : state_(webrtc::SS_OPEN), + : state_(SS_OPEN), buffer_(new char[size]), buffer_length_(size), data_length_(0), @@ -34,7 +34,7 @@ FifoBuffer::FifoBuffer(size_t size) } FifoBuffer::FifoBuffer(size_t size, Thread* owner) - : state_(webrtc::SS_OPEN), + : state_(SS_OPEN), buffer_(new char[size]), buffer_length_(size), data_length_(0), @@ -64,7 +64,7 @@ StreamResult FifoBuffer::Read(ArrayView buffer, size_t copy = 0; StreamResult result = ReadLocked(buffer.data(), buffer.size(), ©); - if (result == webrtc::SR_SUCCESS) { + if (result == SR_SUCCESS) { // If read was successful then adjust the read position and number of // bytes buffered. read_position_ = (read_position_ + copy) % buffer_length_; @@ -73,7 +73,7 @@ StreamResult FifoBuffer::Read(ArrayView buffer, // if we were full before, and now we're not, post an event if (!was_writable && copy > 0) { - PostEvent(webrtc::SE_WRITE, 0); + PostEvent(SE_WRITE, 0); } } return result; @@ -88,13 +88,13 @@ StreamResult FifoBuffer::Write(ArrayView buffer, size_t copy = 0; StreamResult result = WriteLocked(buffer.data(), buffer.size(), ©); - if (result == webrtc::SR_SUCCESS) { + if (result == SR_SUCCESS) { // If write was successful then adjust the number of readable bytes. data_length_ += copy; bytes_written = copy; // if we didn't have any data to read before, and now we do, post an event if (!was_readable && copy > 0) { - PostEvent(webrtc::SE_READ, 0); + PostEvent(SE_READ, 0); } } return result; @@ -102,7 +102,7 @@ StreamResult FifoBuffer::Write(ArrayView buffer, void FifoBuffer::Close() { RTC_DCHECK_RUN_ON(&callback_sequence_); - state_ = webrtc::SS_CLOSED; + state_ = SS_CLOSED; } const void* FifoBuffer::GetReadData(size_t* size) { @@ -120,14 +120,14 @@ void FifoBuffer::ConsumeReadData(size_t size) { read_position_ = (read_position_ + size) % buffer_length_; data_length_ -= size; if (!was_writable && size > 0) { - PostEvent(webrtc::SE_WRITE, 0); + PostEvent(SE_WRITE, 0); } } void* FifoBuffer::GetWriteBuffer(size_t* size) { RTC_DCHECK_RUN_ON(&callback_sequence_); - if (state_ == webrtc::SS_CLOSED) { + if (state_ == SS_CLOSED) { return nullptr; } @@ -151,7 +151,7 @@ void FifoBuffer::ConsumeWriteBuffer(size_t size) { const bool was_readable = (data_length_ > 0); data_length_ += size; if (!was_readable && size > 0) { - PostEvent(webrtc::SE_READ, 0); + PostEvent(SE_READ, 0); } } @@ -159,7 +159,7 @@ StreamResult FifoBuffer::ReadLocked(void* buffer, size_t bytes, size_t* bytes_read) { if (data_length_ == 0) { - return (state_ != webrtc::SS_CLOSED) ? webrtc::SR_BLOCK : webrtc::SR_EOS; + return (state_ != SS_CLOSED) ? SR_BLOCK : SR_EOS; } const size_t available = data_length_; @@ -173,18 +173,18 @@ StreamResult FifoBuffer::ReadLocked(void* buffer, if (bytes_read) { *bytes_read = copy; } - return webrtc::SR_SUCCESS; + return SR_SUCCESS; } StreamResult FifoBuffer::WriteLocked(const void* buffer, size_t bytes, size_t* bytes_written) { - if (state_ == webrtc::SS_CLOSED) { - return webrtc::SR_EOS; + if (state_ == SS_CLOSED) { + return SR_EOS; } if (data_length_ >= buffer_length_) { - return webrtc::SR_BLOCK; + return SR_BLOCK; } const size_t available = buffer_length_ - data_length_; @@ -199,7 +199,7 @@ StreamResult FifoBuffer::WriteLocked(const void* buffer, if (bytes_written) { *bytes_written = copy; } - return webrtc::SR_SUCCESS; + return SR_SUCCESS; } } // namespace webrtc diff --git a/rtc_base/memory_stream.cc b/rtc_base/memory_stream.cc index ad6793e07c..d0b6acbff6 100644 --- a/rtc_base/memory_stream.cc +++ b/rtc_base/memory_stream.cc @@ -23,14 +23,14 @@ namespace webrtc { StreamState MemoryStream::GetState() const { - return webrtc::SS_OPEN; + return SS_OPEN; } StreamResult MemoryStream::Read(ArrayView buffer, size_t& bytes_read, int& error) { if (seek_position_ >= data_length_) { - return webrtc::SR_EOS; + return SR_EOS; } size_t available = data_length_ - seek_position_; size_t bytes; @@ -43,7 +43,7 @@ StreamResult MemoryStream::Read(ArrayView buffer, memcpy(buffer.data(), &buffer_[seek_position_], bytes); seek_position_ += bytes; bytes_read = bytes; - return webrtc::SR_SUCCESS; + return SR_SUCCESS; } StreamResult MemoryStream::Write(ArrayView buffer, @@ -57,7 +57,7 @@ StreamResult MemoryStream::Write(ArrayView buffer, size_t new_buffer_length = std::max( ((seek_position_ + buffer.size()) | 0xFF) + 1, buffer_length_ * 2); StreamResult result = DoReserve(new_buffer_length, &error); - if (webrtc::SR_SUCCESS != result) { + if (SR_SUCCESS != result) { return result; } RTC_DCHECK(buffer_length_ >= new_buffer_length); @@ -74,7 +74,7 @@ StreamResult MemoryStream::Write(ArrayView buffer, data_length_ = seek_position_; } bytes_written = bytes; - return webrtc::SR_SUCCESS; + return SR_SUCCESS; } void MemoryStream::Close() { @@ -105,7 +105,7 @@ bool MemoryStream::GetSize(size_t* size) const { } bool MemoryStream::ReserveSize(size_t size) { - return (webrtc::SR_SUCCESS == DoReserve(size, nullptr)); + return (SR_SUCCESS == DoReserve(size, nullptr)); } /////////////////////////////////////////////////////////////////////////////// @@ -126,7 +126,7 @@ void MemoryStream::SetData(const void* data, size_t length) { StreamResult MemoryStream::DoReserve(size_t size, int* error) { if (buffer_length_ >= size) - return webrtc::SR_SUCCESS; + return SR_SUCCESS; if (char* new_buffer = new char[size]) { if (buffer_ != nullptr && data_length_ > 0) { @@ -135,13 +135,13 @@ StreamResult MemoryStream::DoReserve(size_t size, int* error) { delete[] buffer_; buffer_ = new_buffer; buffer_length_ = size; - return webrtc::SR_SUCCESS; + return SR_SUCCESS; } if (error) { *error = ENOMEM; } - return webrtc::SR_ERROR; + return SR_ERROR; } } // namespace webrtc diff --git a/rtc_base/network.cc b/rtc_base/network.cc index cc796b7558..edb78eb4a4 100644 --- a/rtc_base/network.cc +++ b/rtc_base/network.cc @@ -62,10 +62,6 @@ namespace webrtc { namespace { -using ::webrtc::Environment; -using ::webrtc::SafeTask; -using ::webrtc::SocketFactory; -using ::webrtc::TimeDelta; // List of MAC addresses of known VPN (for windows). constexpr uint8_t kVpns[3][6] = { @@ -101,9 +97,8 @@ bool SortNetworks(const Network* a, const Network* b) { // After type, networks are sorted by IP address precedence values // from RFC 3484-bis - if (webrtc::IPAddressPrecedence(ip_a) != webrtc::IPAddressPrecedence(ip_b)) { - return webrtc::IPAddressPrecedence(ip_a) > - webrtc::IPAddressPrecedence(ip_b); + if (IPAddressPrecedence(ip_a) != IPAddressPrecedence(ip_b)) { + return IPAddressPrecedence(ip_a) > IPAddressPrecedence(ip_b); } // TODO(mallinath) - Add VPN and Link speed conditions while sorting. @@ -118,37 +113,32 @@ uint16_t ComputeNetworkCostByType(int type, bool add_network_cost_to_vpn) { // TODO(jonaso) : Rollout support for cellular network cost using A/B // experiment to make sure it does not introduce regressions. - int vpnCost = - (is_vpn && add_network_cost_to_vpn) ? webrtc::kNetworkCostVpn : 0; + int vpnCost = (is_vpn && add_network_cost_to_vpn) ? kNetworkCostVpn : 0; switch (type) { - case webrtc::ADAPTER_TYPE_ETHERNET: - case webrtc::ADAPTER_TYPE_LOOPBACK: - return webrtc::kNetworkCostMin + vpnCost; - case webrtc::ADAPTER_TYPE_WIFI: - return webrtc::kNetworkCostLow + vpnCost; - case webrtc::ADAPTER_TYPE_CELLULAR: - return webrtc::kNetworkCostCellular + vpnCost; - case webrtc::ADAPTER_TYPE_CELLULAR_2G: - return (use_differentiated_cellular_costs - ? webrtc::kNetworkCostCellular2G - : webrtc::kNetworkCostCellular) + + case ADAPTER_TYPE_ETHERNET: + case ADAPTER_TYPE_LOOPBACK: + return kNetworkCostMin + vpnCost; + case ADAPTER_TYPE_WIFI: + return kNetworkCostLow + vpnCost; + case ADAPTER_TYPE_CELLULAR: + return kNetworkCostCellular + vpnCost; + case ADAPTER_TYPE_CELLULAR_2G: + return (use_differentiated_cellular_costs ? kNetworkCostCellular2G + : kNetworkCostCellular) + vpnCost; - case webrtc::ADAPTER_TYPE_CELLULAR_3G: - return (use_differentiated_cellular_costs - ? webrtc::kNetworkCostCellular3G - : webrtc::kNetworkCostCellular) + + case ADAPTER_TYPE_CELLULAR_3G: + return (use_differentiated_cellular_costs ? kNetworkCostCellular3G + : kNetworkCostCellular) + vpnCost; - case webrtc::ADAPTER_TYPE_CELLULAR_4G: - return (use_differentiated_cellular_costs - ? webrtc::kNetworkCostCellular4G - : webrtc::kNetworkCostCellular) + + case ADAPTER_TYPE_CELLULAR_4G: + return (use_differentiated_cellular_costs ? kNetworkCostCellular4G + : kNetworkCostCellular) + vpnCost; - case webrtc::ADAPTER_TYPE_CELLULAR_5G: - return (use_differentiated_cellular_costs - ? webrtc::kNetworkCostCellular5G - : webrtc::kNetworkCostCellular) + + case ADAPTER_TYPE_CELLULAR_5G: + return (use_differentiated_cellular_costs ? kNetworkCostCellular5G + : kNetworkCostCellular) + vpnCost; - case webrtc::ADAPTER_TYPE_ANY: + case ADAPTER_TYPE_ANY: // Candidates gathered from the any-address/wildcard ports, as backups, // are given the maximum cost so that if there are other candidates with // known interface types, we would not select candidate pairs using these @@ -158,13 +148,13 @@ uint16_t ComputeNetworkCostByType(int type, // ADAPTER_TYPE_CELLULAR would then have a higher cost. See // P2PTransportChannel::SortConnectionsAndUpdateState for how we rank and // select candidate pairs, where the network cost is among the criteria. - return webrtc::kNetworkCostMax + vpnCost; - case webrtc::ADAPTER_TYPE_VPN: + return kNetworkCostMax + vpnCost; + case ADAPTER_TYPE_VPN: // The cost of a VPN should be computed using its underlying network type. RTC_DCHECK_NOTREACHED(); - return webrtc::kNetworkCostUnknown; + return kNetworkCostUnknown; default: - return webrtc::kNetworkCostUnknown + vpnCost; + return kNetworkCostUnknown + vpnCost; } } @@ -177,19 +167,19 @@ bool IsIgnoredIPv6(bool allow_mac_based_ipv6, const InterfaceAddress& ip) { // Link-local addresses require scope id to be bound successfully. // However, our IPAddress structure doesn't carry that so the // information is lost and causes binding failure. - if (webrtc::IPIsLinkLocal(ip)) { + if (IPIsLinkLocal(ip)) { RTC_LOG(LS_VERBOSE) << "Ignore link local IP:" << ip.ToSensitiveString(); return true; } // Any MAC based IPv6 should be avoided to prevent the MAC tracking. - if (webrtc::IPIsMacBased(ip) && !allow_mac_based_ipv6) { + if (IPIsMacBased(ip) && !allow_mac_based_ipv6) { RTC_LOG(LS_INFO) << "Ignore Mac based IP:" << ip.ToSensitiveString(); return true; } // Ignore deprecated IPv6. - if (ip.ipv6_flags() & webrtc::IPV6_ADDRESS_FLAG_DEPRECATED) { + if (ip.ipv6_flags() & IPV6_ADDRESS_FLAG_DEPRECATED) { RTC_LOG(LS_INFO) << "Ignore deprecated IP:" << ip.ToSensitiveString(); return true; } @@ -275,23 +265,23 @@ AdapterType GetAdapterTypeFromName(absl::string_view network_name) { // Note that we have a more robust way to determine if a network interface // is a loopback interface by checking the flag IFF_LOOPBACK in ifa_flags of // an ifaddr struct. See ConvertIfAddrs in this file. - return webrtc::ADAPTER_TYPE_LOOPBACK; + return ADAPTER_TYPE_LOOPBACK; } if (MatchTypeNameWithIndexPattern(network_name, "eth")) { - return webrtc::ADAPTER_TYPE_ETHERNET; + return ADAPTER_TYPE_ETHERNET; } if (MatchTypeNameWithIndexPattern(network_name, "wlan") || MatchTypeNameWithIndexPattern(network_name, "v4-wlan")) { - return webrtc::ADAPTER_TYPE_WIFI; + return ADAPTER_TYPE_WIFI; } if (MatchTypeNameWithIndexPattern(network_name, "ipsec") || MatchTypeNameWithIndexPattern(network_name, "tun") || MatchTypeNameWithIndexPattern(network_name, "utun") || MatchTypeNameWithIndexPattern(network_name, "tap")) { - return webrtc::ADAPTER_TYPE_VPN; + return ADAPTER_TYPE_VPN; } #if defined(WEBRTC_IOS) // Cell networks are pdp_ipN on iOS. @@ -316,7 +306,7 @@ AdapterType GetAdapterTypeFromName(absl::string_view network_name) { } #endif - return webrtc::ADAPTER_TYPE_UNKNOWN; + return ADAPTER_TYPE_UNKNOWN; } NetworkManager::EnumerationPermission NetworkManager::enumeration_permission() @@ -355,8 +345,8 @@ std::vector NetworkManagerBase::GetAnyAddressNetworks() { std::vector networks; if (!ipv4_any_address_network_) { const IPAddress ipv4_any_address(INADDR_ANY); - ipv4_any_address_network_ = CreateNetwork("any", "any", ipv4_any_address, 0, - webrtc::ADAPTER_TYPE_ANY); + ipv4_any_address_network_ = + CreateNetwork("any", "any", ipv4_any_address, 0, ADAPTER_TYPE_ANY); ipv4_any_address_network_->set_default_local_address_provider(this); ipv4_any_address_network_->set_mdns_responder_provider(this); ipv4_any_address_network_->AddIP(ipv4_any_address); @@ -365,8 +355,8 @@ std::vector NetworkManagerBase::GetAnyAddressNetworks() { if (!ipv6_any_address_network_) { const IPAddress ipv6_any_address(in6addr_any); - ipv6_any_address_network_ = CreateNetwork("any", "any", ipv6_any_address, 0, - webrtc::ADAPTER_TYPE_ANY); + ipv6_any_address_network_ = + CreateNetwork("any", "any", ipv6_any_address, 0, ADAPTER_TYPE_ANY); ipv6_any_address_network_->set_default_local_address_provider(this); ipv6_any_address_network_->set_mdns_responder_provider(this); ipv6_any_address_network_->AddIP(ipv6_any_address); @@ -446,7 +436,7 @@ void NetworkManagerBase::MergeNetworkList( Network* existing_net = existing->second.get(); *changed = existing_net->SetIPs(kv.second.ips, *changed); merged_list.push_back(existing_net); - if (net->type() != webrtc::ADAPTER_TYPE_UNKNOWN && + if (net->type() != ADAPTER_TYPE_UNKNOWN && net->type() != existing_net->type()) { if (ShouldAdapterChangeTriggerNetworkChange(existing_net->type(), net->type())) { @@ -599,8 +589,8 @@ NetworkMonitorInterface::InterfaceInfo BasicNetworkManager::GetInterfaceInfo( struct ifaddrs* cursor) const { if (cursor->ifa_flags & IFF_LOOPBACK) { return { - .adapter_type = webrtc::ADAPTER_TYPE_LOOPBACK, - .underlying_type_for_vpn = webrtc::ADAPTER_TYPE_UNKNOWN, + .adapter_type = ADAPTER_TYPE_LOOPBACK, + .underlying_type_for_vpn = ADAPTER_TYPE_UNKNOWN, .network_preference = NetworkPreference::NEUTRAL, .available = true, }; @@ -608,7 +598,7 @@ NetworkMonitorInterface::InterfaceInfo BasicNetworkManager::GetInterfaceInfo( return network_monitor_->GetInterfaceInfo(cursor->ifa_name); } else { return {.adapter_type = GetAdapterTypeFromName(cursor->ifa_name), - .underlying_type_for_vpn = webrtc::ADAPTER_TYPE_UNKNOWN, + .underlying_type_for_vpn = ADAPTER_TYPE_UNKNOWN, .network_preference = NetworkPreference::NEUTRAL, .available = true}; } @@ -658,8 +648,8 @@ void BasicNetworkManager::ConvertIfAddrs( reinterpret_cast(cursor->ifa_addr)->sin6_scope_id; } - int prefix_length = webrtc::CountIPMaskBits(mask); - prefix = webrtc::TruncateIP(ip, prefix_length); + int prefix_length = CountIPMaskBits(mask); + prefix = TruncateIP(ip, prefix_length); std::string key = MakeNetworkKey(std::string(cursor->ifa_name), prefix, prefix_length); @@ -694,10 +684,10 @@ void BasicNetworkManager::ConvertIfAddrs( NetworkMonitorInterface::InterfaceInfo if_info = GetInterfaceInfo(cursor); // Check manually configured VPN override. - if (if_info.adapter_type != webrtc::ADAPTER_TYPE_VPN && + if (if_info.adapter_type != ADAPTER_TYPE_VPN && IsConfiguredVpn(prefix, prefix_length)) { if_info.underlying_type_for_vpn = if_info.adapter_type; - if_info.adapter_type = webrtc::ADAPTER_TYPE_VPN; + if_info.adapter_type = ADAPTER_TYPE_VPN; } auto network = CreateNetwork(cursor->ifa_name, cursor->ifa_name, prefix, @@ -1177,32 +1167,32 @@ IPAddress Network::GetBestIP() const { for (const InterfaceAddress& ip : ips_) { // Ignore any address which has been deprecated already. - if (ip.ipv6_flags() & webrtc::IPV6_ADDRESS_FLAG_DEPRECATED) + if (ip.ipv6_flags() & IPV6_ADDRESS_FLAG_DEPRECATED) continue; - if (webrtc::IPIsLinkLocal(ip)) { + if (IPIsLinkLocal(ip)) { link_local_ip = ip; continue; } // ULA address should only be returned when we have no other // global IP. - if (webrtc::IPIsULA(static_cast(ip))) { + if (IPIsULA(static_cast(ip))) { ula_ip = ip; continue; } selected_ip = ip; // Search could stop once a temporary non-deprecated one is found. - if (ip.ipv6_flags() & webrtc::IPV6_ADDRESS_FLAG_TEMPORARY) + if (ip.ipv6_flags() & IPV6_ADDRESS_FLAG_TEMPORARY) break; } - if (webrtc::IPIsUnspec(selected_ip)) { - if (!webrtc::IPIsUnspec(link_local_ip)) { + if (IPIsUnspec(selected_ip)) { + if (!IPIsUnspec(link_local_ip)) { // No proper global IPv6 address found, use link local address instead. selected_ip = link_local_ip; - } else if (!webrtc::IPIsUnspec(ula_ip)) { + } else if (!IPIsUnspec(ula_ip)) { // No proper global and link local address found, use ULA instead. selected_ip = ula_ip; } @@ -1233,45 +1223,45 @@ uint16_t Network::GetCost(const FieldTrialsView& field_trials) const { std::pair Network::GuessAdapterFromNetworkCost( int network_cost) { switch (network_cost) { - case webrtc::kNetworkCostMin: - return {webrtc::ADAPTER_TYPE_ETHERNET, false}; - case webrtc::kNetworkCostMin + webrtc::kNetworkCostVpn: - return {webrtc::ADAPTER_TYPE_ETHERNET, true}; - case webrtc::kNetworkCostLow: - return {webrtc::ADAPTER_TYPE_WIFI, false}; - case webrtc::kNetworkCostLow + webrtc::kNetworkCostVpn: - return {webrtc::ADAPTER_TYPE_WIFI, true}; - case webrtc::kNetworkCostCellular: - return {webrtc::ADAPTER_TYPE_CELLULAR, false}; - case webrtc::kNetworkCostCellular + webrtc::kNetworkCostVpn: - return {webrtc::ADAPTER_TYPE_CELLULAR, true}; - case webrtc::kNetworkCostCellular2G: - return {webrtc::ADAPTER_TYPE_CELLULAR_2G, false}; - case webrtc::kNetworkCostCellular2G + webrtc::kNetworkCostVpn: - return {webrtc::ADAPTER_TYPE_CELLULAR_2G, true}; - case webrtc::kNetworkCostCellular3G: - return {webrtc::ADAPTER_TYPE_CELLULAR_3G, false}; - case webrtc::kNetworkCostCellular3G + webrtc::kNetworkCostVpn: - return {webrtc::ADAPTER_TYPE_CELLULAR_3G, true}; - case webrtc::kNetworkCostCellular4G: - return {webrtc::ADAPTER_TYPE_CELLULAR_4G, false}; - case webrtc::kNetworkCostCellular4G + webrtc::kNetworkCostVpn: - return {webrtc::ADAPTER_TYPE_CELLULAR_4G, true}; - case webrtc::kNetworkCostCellular5G: - return {webrtc::ADAPTER_TYPE_CELLULAR_5G, false}; - case webrtc::kNetworkCostCellular5G + webrtc::kNetworkCostVpn: - return {webrtc::ADAPTER_TYPE_CELLULAR_5G, true}; - case webrtc::kNetworkCostUnknown: - return {webrtc::ADAPTER_TYPE_UNKNOWN, false}; - case webrtc::kNetworkCostUnknown + webrtc::kNetworkCostVpn: - return {webrtc::ADAPTER_TYPE_UNKNOWN, true}; - case webrtc::kNetworkCostMax: - return {webrtc::ADAPTER_TYPE_ANY, false}; - case webrtc::kNetworkCostMax + webrtc::kNetworkCostVpn: - return {webrtc::ADAPTER_TYPE_ANY, true}; + case kNetworkCostMin: + return {ADAPTER_TYPE_ETHERNET, false}; + case kNetworkCostMin + kNetworkCostVpn: + return {ADAPTER_TYPE_ETHERNET, true}; + case kNetworkCostLow: + return {ADAPTER_TYPE_WIFI, false}; + case kNetworkCostLow + kNetworkCostVpn: + return {ADAPTER_TYPE_WIFI, true}; + case kNetworkCostCellular: + return {ADAPTER_TYPE_CELLULAR, false}; + case kNetworkCostCellular + kNetworkCostVpn: + return {ADAPTER_TYPE_CELLULAR, true}; + case kNetworkCostCellular2G: + return {ADAPTER_TYPE_CELLULAR_2G, false}; + case kNetworkCostCellular2G + kNetworkCostVpn: + return {ADAPTER_TYPE_CELLULAR_2G, true}; + case kNetworkCostCellular3G: + return {ADAPTER_TYPE_CELLULAR_3G, false}; + case kNetworkCostCellular3G + kNetworkCostVpn: + return {ADAPTER_TYPE_CELLULAR_3G, true}; + case kNetworkCostCellular4G: + return {ADAPTER_TYPE_CELLULAR_4G, false}; + case kNetworkCostCellular4G + kNetworkCostVpn: + return {ADAPTER_TYPE_CELLULAR_4G, true}; + case kNetworkCostCellular5G: + return {ADAPTER_TYPE_CELLULAR_5G, false}; + case kNetworkCostCellular5G + kNetworkCostVpn: + return {ADAPTER_TYPE_CELLULAR_5G, true}; + case kNetworkCostUnknown: + return {ADAPTER_TYPE_UNKNOWN, false}; + case kNetworkCostUnknown + kNetworkCostVpn: + return {ADAPTER_TYPE_UNKNOWN, true}; + case kNetworkCostMax: + return {ADAPTER_TYPE_ANY, false}; + case kNetworkCostMax + kNetworkCostVpn: + return {ADAPTER_TYPE_ANY, true}; } RTC_LOG(LS_VERBOSE) << "Unknown network cost: " << network_cost; - return {webrtc::ADAPTER_TYPE_UNKNOWN, false}; + return {ADAPTER_TYPE_UNKNOWN, false}; } std::string Network::ToString() const { @@ -1280,9 +1270,9 @@ std::string Network::ToString() const { // the IP address. ss << "Net[" << description_.substr(0, description_.find(' ')) << ":" << prefix_.ToSensitiveString() << "/" << prefix_length_ << ":" - << webrtc::AdapterTypeToString(type_); + << AdapterTypeToString(type_); if (IsVpn()) { - ss << "/" << webrtc::AdapterTypeToString(underlying_type_for_vpn_); + ss << "/" << AdapterTypeToString(underlying_type_for_vpn_); } ss << ":id=" << id_ << "]"; return ss.Release(); @@ -1301,7 +1291,7 @@ bool BasicNetworkManager::IsConfiguredVpn(IPAddress prefix, RTC_DCHECK_RUN_ON(thread_); for (const auto& vpn : vpn_) { if (prefix_length >= vpn.prefix_length()) { - auto copy = webrtc::TruncateIP(prefix, vpn.prefix_length()); + auto copy = TruncateIP(prefix, vpn.prefix_length()); if (copy == vpn.address()) { return true; } diff --git a/rtc_base/network_unittest.cc b/rtc_base/network_unittest.cc index 02c6ee0013..35795edd20 100644 --- a/rtc_base/network_unittest.cc +++ b/rtc_base/network_unittest.cc @@ -79,7 +79,7 @@ namespace { IPAddress IPFromString(absl::string_view str) { IPAddress ip; - RTC_CHECK(webrtc::IPFromString(str, &ip)); + RTC_CHECK(IPFromString(str, &ip)); return ip; } @@ -90,13 +90,13 @@ class FakeNetworkMonitor : public NetworkMonitorInterface { bool started() { return started_; } InterfaceInfo GetInterfaceInfo(absl::string_view if_name) override { InterfaceInfo if_info = { - .adapter_type = webrtc::ADAPTER_TYPE_UNKNOWN, + .adapter_type = ADAPTER_TYPE_UNKNOWN, .available = absl::c_count(unavailable_adapters_, if_name) == 0, }; if (absl::StartsWith(if_name, "wifi")) { - if_info.adapter_type = webrtc::ADAPTER_TYPE_WIFI; + if_info.adapter_type = ADAPTER_TYPE_WIFI; } else if (absl::StartsWith(if_name, "cellular")) { - if_info.adapter_type = webrtc::ADAPTER_TYPE_CELLULAR; + if_info.adapter_type = ADAPTER_TYPE_CELLULAR; } return if_info; } @@ -247,7 +247,7 @@ class NetworkTest : public ::testing::Test, public sigslot::has_slots<> { ipv6_addr->sin6_family = AF_INET6; ipv6_addr->sin6_scope_id = scope_id; IPAddress ip; - webrtc::IPFromString(ip_string, &ip); + IPFromString(ip_string, &ip); ipv6_addr->sin6_addr = ip.ipv6_address(); return ipv6_addr; } @@ -290,7 +290,7 @@ class NetworkTest : public ::testing::Test, public sigslot::has_slots<> { memset(ipv4_addr, 0, sizeof(struct sockaddr_in)); ipv4_addr->sin_family = AF_INET; IPAddress ip; - webrtc::IPFromString(ip_string, &ip); + IPFromString(ip_string, &ip); ipv4_addr->sin_addr = ip.ipv4_address(); return ipv4_addr; } @@ -365,11 +365,9 @@ TEST_F(NetworkTest, TestNetworkConstruct) { TEST_F(NetworkTest, TestIsIgnoredNetworkIgnoresIPsStartingWith0) { Network ipv4_network1("test_eth0", "Test Network Adapter 1", - IPAddress(0x12345600U), 24, - webrtc::ADAPTER_TYPE_ETHERNET); + IPAddress(0x12345600U), 24, ADAPTER_TYPE_ETHERNET); Network ipv4_network2("test_eth1", "Test Network Adapter 2", - IPAddress(0x010000U), 24, - webrtc::ADAPTER_TYPE_ETHERNET); + IPAddress(0x010000U), 24, ADAPTER_TYPE_ETHERNET); PhysicalSocketServer socket_server; BasicNetworkManager network_manager(CreateEnvironment(), &socket_server); network_manager.StartUpdating(); @@ -553,25 +551,24 @@ TEST_F(NetworkTest, TestBasicMergeNetworkList) { void SetupNetworks(std::vector>* list) { IPAddress ip; IPAddress prefix; - EXPECT_TRUE(webrtc::IPFromString("abcd::1234:5678:abcd:ef12", &ip)); - EXPECT_TRUE(webrtc::IPFromString("abcd::", &prefix)); + EXPECT_TRUE(IPFromString("abcd::1234:5678:abcd:ef12", &ip)); + EXPECT_TRUE(IPFromString("abcd::", &prefix)); // First, fake link-locals. Network ipv6_eth0_linklocalnetwork("test_eth0", "Test NetworkAdapter 1", prefix, 64); ipv6_eth0_linklocalnetwork.AddIP(ip); - EXPECT_TRUE(webrtc::IPFromString("abcd::5678:abcd:ef12:3456", &ip)); + EXPECT_TRUE(IPFromString("abcd::5678:abcd:ef12:3456", &ip)); Network ipv6_eth1_linklocalnetwork("test_eth1", "Test NetworkAdapter 2", prefix, 64); ipv6_eth1_linklocalnetwork.AddIP(ip); // Public networks: - EXPECT_TRUE(webrtc::IPFromString("2401:fa00:4:1000:be30:5bff:fee5:c3", &ip)); - prefix = webrtc::TruncateIP(ip, 64); + EXPECT_TRUE(IPFromString("2401:fa00:4:1000:be30:5bff:fee5:c3", &ip)); + prefix = TruncateIP(ip, 64); Network ipv6_eth0_publicnetwork1_ip1("test_eth0", "Test NetworkAdapter 1", prefix, 64); ipv6_eth0_publicnetwork1_ip1.AddIP(ip); - EXPECT_TRUE( - webrtc::IPFromString("2400:4030:1:2c00:be30:abcd:efab:cdef", &ip)); - prefix = webrtc::TruncateIP(ip, 64); + EXPECT_TRUE(IPFromString("2400:4030:1:2c00:be30:abcd:efab:cdef", &ip)); + prefix = TruncateIP(ip, 64); Network ipv6_eth1_publicnetwork1_ip1("test_eth1", "Test NetworkAdapter 1", prefix, 64); ipv6_eth1_publicnetwork1_ip1.AddIP(ip); @@ -644,16 +641,15 @@ TEST_F(NetworkTest, MergeWithChangedIP) { SetupNetworks(&original_list); // Make a network that we're going to change. IPAddress ip; - EXPECT_TRUE(webrtc::IPFromString("2401:fa01:4:1000:be30:faa:fee:faa", &ip)); - IPAddress prefix = webrtc::TruncateIP(ip, 64); + EXPECT_TRUE(IPFromString("2401:fa01:4:1000:be30:faa:fee:faa", &ip)); + IPAddress prefix = TruncateIP(ip, 64); std::unique_ptr network_to_change = std::make_unique( "test_eth0", "Test Network Adapter 1", prefix, 64); std::unique_ptr changed_network = std::make_unique(*network_to_change); network_to_change->AddIP(ip); IPAddress changed_ip; - EXPECT_TRUE( - webrtc::IPFromString("2401:fa01:4:1000:be30:f00:f00:f00", &changed_ip)); + EXPECT_TRUE(IPFromString("2401:fa01:4:1000:be30:f00:f00:f00", &changed_ip)); changed_network->AddIP(changed_ip); const Network* const network_to_change_ptr = network_to_change.get(); original_list.push_back(std::move(network_to_change)); @@ -688,13 +684,12 @@ TEST_F(NetworkTest, TestMultipleIPMergeNetworkList) { IPAddress check_ip; IPAddress prefix; // Add a second IP to the public network on eth0 (2401:fa00:4:1000/64). - EXPECT_TRUE(webrtc::IPFromString("2401:fa00:4:1000:be30:5bff:fee5:c6", &ip)); - prefix = webrtc::TruncateIP(ip, 64); + EXPECT_TRUE(IPFromString("2401:fa00:4:1000:be30:5bff:fee5:c6", &ip)); + prefix = TruncateIP(ip, 64); Network ipv6_eth0_publicnetwork1_ip2("test_eth0", "Test NetworkAdapter 1", prefix, 64); // This is the IP that already existed in the public network on eth0. - EXPECT_TRUE( - webrtc::IPFromString("2401:fa00:4:1000:be30:5bff:fee5:c3", &check_ip)); + EXPECT_TRUE(IPFromString("2401:fa00:4:1000:be30:5bff:fee5:c3", &check_ip)); ipv6_eth0_publicnetwork1_ip2.AddIP(ip); std::vector> second_list; @@ -741,8 +736,8 @@ TEST_F(NetworkTest, TestMultiplePublicNetworksOnOneInterfaceMerge) { IPAddress ip; IPAddress prefix; // A second network for eth0. - EXPECT_TRUE(webrtc::IPFromString("2400:4030:1:2c00:be30:5bff:fee5:c3", &ip)); - prefix = webrtc::TruncateIP(ip, 64); + EXPECT_TRUE(IPFromString("2400:4030:1:2c00:be30:5bff:fee5:c3", &ip)); + prefix = TruncateIP(ip, 64); Network ipv6_eth0_publicnetwork2_ip1("test_eth0", "Test NetworkAdapter 1", prefix, 64); ipv6_eth0_publicnetwork2_ip1.AddIP(ip); @@ -806,9 +801,8 @@ TEST_F(NetworkTest, IPv6NetworksPreferredOverIPv4) { IPAddress ip; IPAddress prefix; - EXPECT_TRUE( - webrtc::IPFromString("2400:4030:1:2c00:be30:abcd:efab:cdef", &ip)); - prefix = webrtc::TruncateIP(ip, 64); + EXPECT_TRUE(IPFromString("2400:4030:1:2c00:be30:abcd:efab:cdef", &ip)); + prefix = TruncateIP(ip, 64); Network ipv6_eth1_publicnetwork1_ip1("test_eth1", "Test NetworkAdapter 2", prefix, 64); ipv6_eth1_publicnetwork1_ip1.AddIP(ip); @@ -854,20 +848,20 @@ TEST_F(NetworkTest, NetworksSortedByInterfaceName) { TEST_F(NetworkTest, TestNetworkAdapterTypes) { Network wifi("wlan0", "Wireless Adapter", IPAddress(0x12345600U), 24, - webrtc::ADAPTER_TYPE_WIFI); - EXPECT_EQ(webrtc::ADAPTER_TYPE_WIFI, wifi.type()); + ADAPTER_TYPE_WIFI); + EXPECT_EQ(ADAPTER_TYPE_WIFI, wifi.type()); Network ethernet("eth0", "Ethernet", IPAddress(0x12345600U), 24, - webrtc::ADAPTER_TYPE_ETHERNET); - EXPECT_EQ(webrtc::ADAPTER_TYPE_ETHERNET, ethernet.type()); + ADAPTER_TYPE_ETHERNET); + EXPECT_EQ(ADAPTER_TYPE_ETHERNET, ethernet.type()); Network cellular("test_cell", "Cellular Adapter", IPAddress(0x12345600U), 24, - webrtc::ADAPTER_TYPE_CELLULAR); - EXPECT_EQ(webrtc::ADAPTER_TYPE_CELLULAR, cellular.type()); + ADAPTER_TYPE_CELLULAR); + EXPECT_EQ(ADAPTER_TYPE_CELLULAR, cellular.type()); Network vpn("bridge_test", "VPN Adapter", IPAddress(0x12345600U), 24, - webrtc::ADAPTER_TYPE_VPN); - EXPECT_EQ(webrtc::ADAPTER_TYPE_VPN, vpn.type()); + ADAPTER_TYPE_VPN); + EXPECT_EQ(ADAPTER_TYPE_VPN, vpn.type()); Network unknown("test", "Test Adapter", IPAddress(0x12345600U), 24, - webrtc::ADAPTER_TYPE_UNKNOWN); - EXPECT_EQ(webrtc::ADAPTER_TYPE_UNKNOWN, unknown.type()); + ADAPTER_TYPE_UNKNOWN); + EXPECT_EQ(ADAPTER_TYPE_UNKNOWN, unknown.type()); } #if defined(WEBRTC_POSIX) @@ -953,8 +947,7 @@ TEST_F(NetworkTest, TestGetAdapterTypeFromNetworkMonitor) { // A network created without a network monitor will get UNKNOWN type. ifaddrs* addr_list = InstallIpv6Network(if_name, ipv6_address, ipv6_mask, manager_without_monitor); - EXPECT_EQ(webrtc::ADAPTER_TYPE_UNKNOWN, - GetAdapterType(manager_without_monitor)); + EXPECT_EQ(ADAPTER_TYPE_UNKNOWN, GetAdapterType(manager_without_monitor)); ReleaseIfAddrs(addr_list); // With the fake network monitor the type should be correctly determined. @@ -965,7 +958,7 @@ TEST_F(NetworkTest, TestGetAdapterTypeFromNetworkMonitor) { // detected by the network monitor now. addr_list = InstallIpv6Network(if_name, ipv6_address, ipv6_mask, manager_with_monitor); - EXPECT_EQ(webrtc::ADAPTER_TYPE_WIFI, GetAdapterType(manager_with_monitor)); + EXPECT_EQ(ADAPTER_TYPE_WIFI, GetAdapterType(manager_with_monitor)); ReleaseIfAddrs(addr_list); } @@ -986,25 +979,25 @@ TEST_F(NetworkTest, TestGetAdapterTypeFromNameMatching) { char if_name[20] = "ipsec11"; ifaddrs* addr_list = InstallIpv6Network(if_name, ipv6_address1, ipv6_mask, manager); - EXPECT_EQ(webrtc::ADAPTER_TYPE_VPN, GetAdapterType(manager)); + EXPECT_EQ(ADAPTER_TYPE_VPN, GetAdapterType(manager)); ClearNetworks(manager); ReleaseIfAddrs(addr_list); strcpy(if_name, "lo0"); addr_list = InstallIpv6Network(if_name, ipv6_address1, ipv6_mask, manager); - EXPECT_EQ(webrtc::ADAPTER_TYPE_LOOPBACK, GetAdapterType(manager)); + EXPECT_EQ(ADAPTER_TYPE_LOOPBACK, GetAdapterType(manager)); ClearNetworks(manager); ReleaseIfAddrs(addr_list); strcpy(if_name, "eth0"); addr_list = InstallIpv4Network(if_name, ipv4_address1, ipv4_mask, manager); - EXPECT_EQ(webrtc::ADAPTER_TYPE_ETHERNET, GetAdapterType(manager)); + EXPECT_EQ(ADAPTER_TYPE_ETHERNET, GetAdapterType(manager)); ClearNetworks(manager); ReleaseIfAddrs(addr_list); strcpy(if_name, "wlan0"); addr_list = InstallIpv6Network(if_name, ipv6_address1, ipv6_mask, manager); - EXPECT_EQ(webrtc::ADAPTER_TYPE_WIFI, GetAdapterType(manager)); + EXPECT_EQ(ADAPTER_TYPE_WIFI, GetAdapterType(manager)); ClearNetworks(manager); ReleaseIfAddrs(addr_list); @@ -1088,14 +1081,12 @@ TEST_F(NetworkTest, TestMergeNetworkList) { // Create 2 IPAddress classes with only last digit different. IPAddress ip1, ip2; - EXPECT_TRUE(webrtc::IPFromString("2400:4030:1:2c00:be30:0:0:1", &ip1)); - EXPECT_TRUE(webrtc::IPFromString("2400:4030:1:2c00:be30:0:0:2", &ip2)); + EXPECT_TRUE(IPFromString("2400:4030:1:2c00:be30:0:0:1", &ip1)); + EXPECT_TRUE(IPFromString("2400:4030:1:2c00:be30:0:0:2", &ip2)); // Create 2 networks with the same prefix and length. - auto net1 = - std::make_unique("em1", "em1", webrtc::TruncateIP(ip1, 64), 64); - auto net2 = - std::make_unique("em1", "em1", webrtc::TruncateIP(ip1, 64), 64); + auto net1 = std::make_unique("em1", "em1", TruncateIP(ip1, 64), 64); + auto net2 = std::make_unique("em1", "em1", TruncateIP(ip1, 64), 64); // Add different IP into each. net1->AddIP(ip1); @@ -1170,18 +1161,17 @@ TEST_F(NetworkTest, TestIPv6Selection) { std::string ipstr; ipstr = "2401:fa00:4:1000:be30:5bff:fee5:c3"; - ASSERT_TRUE( - webrtc::IPFromString(ipstr, webrtc::IPV6_ADDRESS_FLAG_DEPRECATED, &ip)); + ASSERT_TRUE(IPFromString(ipstr, IPV6_ADDRESS_FLAG_DEPRECATED, &ip)); // Create a network with this prefix. - Network ipv6_network("test_eth0", "Test NetworkAdapter", - webrtc::TruncateIP(ip, 64), 64); + Network ipv6_network("test_eth0", "Test NetworkAdapter", TruncateIP(ip, 64), + 64); EXPECT_EQ(AF_INET6, ipv6_network.family()); // When there is no address added, it should return an unspecified // address. EXPECT_EQ(ipv6_network.GetBestIP(), IPAddress()); - EXPECT_TRUE(webrtc::IPIsUnspec(ipv6_network.GetBestIP())); + EXPECT_TRUE(IPIsUnspec(ipv6_network.GetBestIP())); // Deprecated one should not be returned. ipv6_network.AddIP(ip); @@ -1190,20 +1180,19 @@ TEST_F(NetworkTest, TestIPv6Selection) { // Add ULA one. ULA is unique local address which is starting either // with 0xfc or 0xfd. ipstr = "fd00:fa00:4:1000:be30:5bff:fee5:c4"; - ASSERT_TRUE(webrtc::IPFromString(ipstr, webrtc::IPV6_ADDRESS_FLAG_NONE, &ip)); + ASSERT_TRUE(IPFromString(ipstr, IPV6_ADDRESS_FLAG_NONE, &ip)); ipv6_network.AddIP(ip); EXPECT_EQ(ipv6_network.GetBestIP(), static_cast(ip)); // Add global one. ipstr = "2401:fa00:4:1000:be30:5bff:fee5:c5"; - ASSERT_TRUE(webrtc::IPFromString(ipstr, webrtc::IPV6_ADDRESS_FLAG_NONE, &ip)); + ASSERT_TRUE(IPFromString(ipstr, IPV6_ADDRESS_FLAG_NONE, &ip)); ipv6_network.AddIP(ip); EXPECT_EQ(ipv6_network.GetBestIP(), static_cast(ip)); // Add global dynamic temporary one. ipstr = "2401:fa00:4:1000:be30:5bff:fee5:c6"; - ASSERT_TRUE( - webrtc::IPFromString(ipstr, webrtc::IPV6_ADDRESS_FLAG_TEMPORARY, &ip)); + ASSERT_TRUE(IPFromString(ipstr, IPV6_ADDRESS_FLAG_TEMPORARY, &ip)); ipv6_network.AddIP(ip); EXPECT_EQ(ipv6_network.GetBestIP(), static_cast(ip)); } @@ -1214,18 +1203,16 @@ TEST_F(NetworkTest, TestGetBestIPWithPreferGlobalIPv6ToLinkLocalEnabled) { std::string ipstr; ipstr = "2401:fa00:4:1000:be30:5bff:fee5:c3"; - ASSERT_TRUE( - webrtc::IPFromString(ipstr, webrtc::IPV6_ADDRESS_FLAG_DEPRECATED, &ip)); + ASSERT_TRUE(IPFromString(ipstr, IPV6_ADDRESS_FLAG_DEPRECATED, &ip)); // Create a network with this prefix. - Network ipv6_network("test_eth0", "Test NetworkAdapter", - webrtc::TruncateIP(ip, 64), 64, - webrtc::ADAPTER_TYPE_UNKNOWN); + Network ipv6_network("test_eth0", "Test NetworkAdapter", TruncateIP(ip, 64), + 64, ADAPTER_TYPE_UNKNOWN); // When there is no address added, it should return an unspecified // address. EXPECT_EQ(ipv6_network.GetBestIP(), IPAddress()); - EXPECT_TRUE(webrtc::IPIsUnspec(ipv6_network.GetBestIP())); + EXPECT_TRUE(IPIsUnspec(ipv6_network.GetBestIP())); // Deprecated one should not be returned. ipv6_network.AddIP(ip); @@ -1234,43 +1221,39 @@ TEST_F(NetworkTest, TestGetBestIPWithPreferGlobalIPv6ToLinkLocalEnabled) { // Add ULA one. ULA is unique local address which is starting either // with 0xfc or 0xfd. ipstr = "fd00:fa00:4:1000:be30:5bff:fee5:c4"; - ASSERT_TRUE(webrtc::IPFromString(ipstr, webrtc::IPV6_ADDRESS_FLAG_NONE, &ip)); + ASSERT_TRUE(IPFromString(ipstr, IPV6_ADDRESS_FLAG_NONE, &ip)); ipv6_network.AddIP(ip); EXPECT_EQ(ipv6_network.GetBestIP(), static_cast(ip)); // Add link local one. ipstr = "fe80::aabb:ccff:fedd:eeff"; - ASSERT_TRUE( - webrtc::IPFromString(ipstr, webrtc::IPV6_ADDRESS_FLAG_NONE, &link_local)); + ASSERT_TRUE(IPFromString(ipstr, IPV6_ADDRESS_FLAG_NONE, &link_local)); ipv6_network.AddIP(link_local); EXPECT_EQ(ipv6_network.GetBestIP(), static_cast(link_local)); // Add global one. ipstr = "2401:fa00:4:1000:be30:5bff:fee5:c5"; - ASSERT_TRUE(webrtc::IPFromString(ipstr, webrtc::IPV6_ADDRESS_FLAG_NONE, &ip)); + ASSERT_TRUE(IPFromString(ipstr, IPV6_ADDRESS_FLAG_NONE, &ip)); ipv6_network.AddIP(ip); EXPECT_EQ(ipv6_network.GetBestIP(), static_cast(ip)); // Add another link local address, then the compatible address is still global // one. ipstr = "fe80::aabb:ccff:fedd:eedd"; - ASSERT_TRUE( - webrtc::IPFromString(ipstr, webrtc::IPV6_ADDRESS_FLAG_NONE, &link_local)); + ASSERT_TRUE(IPFromString(ipstr, IPV6_ADDRESS_FLAG_NONE, &link_local)); ipv6_network.AddIP(link_local); EXPECT_EQ(ipv6_network.GetBestIP(), static_cast(ip)); // Add global dynamic temporary one. ipstr = "2401:fa00:4:1000:be30:5bff:fee5:c6"; - ASSERT_TRUE( - webrtc::IPFromString(ipstr, webrtc::IPV6_ADDRESS_FLAG_TEMPORARY, &ip)); + ASSERT_TRUE(IPFromString(ipstr, IPV6_ADDRESS_FLAG_TEMPORARY, &ip)); ipv6_network.AddIP(ip); EXPECT_EQ(ipv6_network.GetBestIP(), static_cast(ip)); // Add another link local address, then the compatible address is still global // dynamic one. ipstr = "fe80::aabb:ccff:fedd:eedd"; - ASSERT_TRUE( - webrtc::IPFromString(ipstr, webrtc::IPV6_ADDRESS_FLAG_NONE, &link_local)); + ASSERT_TRUE(IPFromString(ipstr, IPV6_ADDRESS_FLAG_NONE, &link_local)); ipv6_network.AddIP(link_local); EXPECT_EQ(ipv6_network.GetBestIP(), static_cast(ip)); } @@ -1284,8 +1267,7 @@ TEST_F(NetworkTest, TestNetworkMonitoring) { manager.StartUpdating(); FakeNetworkMonitor* network_monitor = GetNetworkMonitor(manager); EXPECT_TRUE(network_monitor && network_monitor->started()); - EXPECT_THAT(webrtc::WaitUntil([&] { return callback_called_; }, IsTrue()), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return callback_called_; }, IsTrue()), IsRtcOk()); callback_called_ = false; // Clear the networks so that there will be network changes below. @@ -1293,8 +1275,7 @@ TEST_F(NetworkTest, TestNetworkMonitoring) { // Network manager is started, so the callback is called when the network // monitor fires the network-change event. network_monitor->InovkeNetworksChangedCallbackForTesting(); - EXPECT_THAT(webrtc::WaitUntil([&] { return callback_called_; }, IsTrue()), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return callback_called_; }, IsTrue()), IsRtcOk()); // Network manager is stopped. manager.StopUpdating(); @@ -1317,8 +1298,7 @@ TEST_F(NetworkTest, MAYBE_DefaultLocalAddress) { manager.SignalNetworksChanged.connect(static_cast(this), &NetworkTest::OnNetworksChanged); manager.StartUpdating(); - EXPECT_THAT(webrtc::WaitUntil([&] { return callback_called_; }, IsTrue()), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return callback_called_; }, IsTrue()), IsRtcOk()); // Make sure we can query default local address when an address for such // address family exists. @@ -1328,7 +1308,7 @@ TEST_F(NetworkTest, MAYBE_DefaultLocalAddress) { if (network->GetBestIP().family() == AF_INET) { EXPECT_TRUE(QueryDefaultLocalAddress(manager, AF_INET) != IPAddress()); } else if (network->GetBestIP().family() == AF_INET6 && - !webrtc::IPIsLoopback(network->GetBestIP())) { + !IPIsLoopback(network->GetBestIP())) { // Existence of an IPv6 loopback address doesn't mean it has IPv6 network // enabled. EXPECT_TRUE(QueryDefaultLocalAddress(manager, AF_INET6) != IPAddress()); @@ -1336,23 +1316,23 @@ TEST_F(NetworkTest, MAYBE_DefaultLocalAddress) { } // GetDefaultLocalAddress should return the valid default address after set. - manager.set_default_local_addresses(webrtc::GetLoopbackIP(AF_INET), - webrtc::GetLoopbackIP(AF_INET6)); + manager.set_default_local_addresses(GetLoopbackIP(AF_INET), + GetLoopbackIP(AF_INET6)); EXPECT_TRUE(manager.GetDefaultLocalAddress(AF_INET, &ip)); - EXPECT_EQ(ip, webrtc::GetLoopbackIP(AF_INET)); + EXPECT_EQ(ip, GetLoopbackIP(AF_INET)); EXPECT_TRUE(manager.GetDefaultLocalAddress(AF_INET6, &ip)); - EXPECT_EQ(ip, webrtc::GetLoopbackIP(AF_INET6)); + EXPECT_EQ(ip, GetLoopbackIP(AF_INET6)); // More tests on GetDefaultLocalAddress with ipv6 addresses where the set // default address may be different from the best IP address of any network. InterfaceAddress ip1; - EXPECT_TRUE(webrtc::IPFromString("abcd::1234:5678:abcd:1111", - webrtc::IPV6_ADDRESS_FLAG_TEMPORARY, &ip1)); + EXPECT_TRUE(IPFromString("abcd::1234:5678:abcd:1111", + IPV6_ADDRESS_FLAG_TEMPORARY, &ip1)); // Create a network with a prefix of ip1. - Network ipv6_network("test_eth0", "Test NetworkAdapter", - webrtc::TruncateIP(ip1, 64), 64); + Network ipv6_network("test_eth0", "Test NetworkAdapter", TruncateIP(ip1, 64), + 64); IPAddress ip2; - EXPECT_TRUE(webrtc::IPFromString("abcd::1234:5678:abcd:2222", &ip2)); + EXPECT_TRUE(IPFromString("abcd::1234:5678:abcd:2222", &ip2)); ipv6_network.AddIP(ip1); ipv6_network.AddIP(ip2); std::vector> list; @@ -1362,13 +1342,13 @@ TEST_F(NetworkTest, MAYBE_DefaultLocalAddress) { // If the set default address is not in any network, GetDefaultLocalAddress // should return it. IPAddress ip3; - EXPECT_TRUE(webrtc::IPFromString("abcd::1234:5678:abcd:3333", &ip3)); - manager.set_default_local_addresses(webrtc::GetLoopbackIP(AF_INET), ip3); + EXPECT_TRUE(IPFromString("abcd::1234:5678:abcd:3333", &ip3)); + manager.set_default_local_addresses(GetLoopbackIP(AF_INET), ip3); EXPECT_TRUE(manager.GetDefaultLocalAddress(AF_INET6, &ip)); EXPECT_EQ(ip3, ip); // If the set default address is in a network, GetDefaultLocalAddress will // return the best IP in that network. - manager.set_default_local_addresses(webrtc::GetLoopbackIP(AF_INET), ip2); + manager.set_default_local_addresses(GetLoopbackIP(AF_INET), ip2); EXPECT_TRUE(manager.GetDefaultLocalAddress(AF_INET6, &ip)); EXPECT_EQ(static_cast(ip1), ip); @@ -1382,10 +1362,9 @@ TEST_F(NetworkTest, TestWhenNetworkListChangeReturnsChangedFlag) { BasicNetworkManager manager(CreateEnvironment(), &socket_server); IPAddress ip1; - EXPECT_TRUE(webrtc::IPFromString("2400:4030:1:2c00:be30:0:0:1", &ip1)); - auto net1 = - std::make_unique("em1", "em1", webrtc::TruncateIP(ip1, 64), 64); - net1->set_type(webrtc::ADAPTER_TYPE_CELLULAR_3G); + EXPECT_TRUE(IPFromString("2400:4030:1:2c00:be30:0:0:1", &ip1)); + auto net1 = std::make_unique("em1", "em1", TruncateIP(ip1, 64), 64); + net1->set_type(ADAPTER_TYPE_CELLULAR_3G); net1->AddIP(ip1); std::vector> list; list.push_back(std::move(net1)); @@ -1396,14 +1375,14 @@ TEST_F(NetworkTest, TestWhenNetworkListChangeReturnsChangedFlag) { EXPECT_TRUE(changed); std::vector list2 = manager.GetNetworks(); EXPECT_EQ(list2.size(), 1uL); - EXPECT_EQ(webrtc::ADAPTER_TYPE_CELLULAR_3G, list2[0]->type()); + EXPECT_EQ(ADAPTER_TYPE_CELLULAR_3G, list2[0]->type()); } // Modify net1 from 3G to 4G { - auto net2 = std::make_unique("em1", "em1", - webrtc::TruncateIP(ip1, 64), 64); - net2->set_type(webrtc::ADAPTER_TYPE_CELLULAR_4G); + auto net2 = + std::make_unique("em1", "em1", TruncateIP(ip1, 64), 64); + net2->set_type(ADAPTER_TYPE_CELLULAR_4G); net2->AddIP(ip1); list.clear(); list.push_back(std::move(net2)); @@ -1415,14 +1394,14 @@ TEST_F(NetworkTest, TestWhenNetworkListChangeReturnsChangedFlag) { EXPECT_FALSE(changed); std::vector list2 = manager.GetNetworks(); ASSERT_EQ(list2.size(), 1uL); - EXPECT_EQ(webrtc::ADAPTER_TYPE_CELLULAR_4G, list2[0]->type()); + EXPECT_EQ(ADAPTER_TYPE_CELLULAR_4G, list2[0]->type()); } // Don't modify. { - auto net2 = std::make_unique("em1", "em1", - webrtc::TruncateIP(ip1, 64), 64); - net2->set_type(webrtc::ADAPTER_TYPE_CELLULAR_4G); + auto net2 = + std::make_unique("em1", "em1", TruncateIP(ip1, 64), 64); + net2->set_type(ADAPTER_TYPE_CELLULAR_4G); net2->AddIP(ip1); list.clear(); list.push_back(std::move(net2)); @@ -1433,7 +1412,7 @@ TEST_F(NetworkTest, TestWhenNetworkListChangeReturnsChangedFlag) { EXPECT_FALSE(changed); std::vector list2 = manager.GetNetworks(); ASSERT_EQ(list2.size(), 1uL); - EXPECT_EQ(webrtc::ADAPTER_TYPE_CELLULAR_4G, list2[0]->type()); + EXPECT_EQ(ADAPTER_TYPE_CELLULAR_4G, list2[0]->type()); } } @@ -1500,9 +1479,9 @@ TEST_F(NetworkTest, WebRTC_BindUsingInterfaceName) { FakeNetworkMonitor* network_monitor = GetNetworkMonitor(manager); IPAddress ipv6; - EXPECT_TRUE(webrtc::IPFromString("1000:2000:3000:4000:0:0:0:1", &ipv6)); + EXPECT_TRUE(IPFromString("1000:2000:3000:4000:0:0:0:1", &ipv6)); IPAddress ipv4; - EXPECT_TRUE(webrtc::IPFromString("192.168.0.2", &ipv4)); + EXPECT_TRUE(IPFromString("192.168.0.2", &ipv4)); // The network monitor only knwos about the ipv6 address, interface. network_monitor->set_adapters({"wlan0"}); @@ -1518,15 +1497,15 @@ TEST_F(NetworkTest, WebRTC_BindUsingInterfaceName) { TEST_F(NetworkTest, NetworkCostVpn_Default) { IPAddress ip1; - EXPECT_TRUE(webrtc::IPFromString("2400:4030:1:2c00:be30:0:0:1", &ip1)); + EXPECT_TRUE(IPFromString("2400:4030:1:2c00:be30:0:0:1", &ip1)); ScopedKeyValueConfig field_trials; - Network* net1 = new Network("em1", "em1", webrtc::TruncateIP(ip1, 64), 64); - net1->set_type(webrtc::ADAPTER_TYPE_VPN); - net1->set_underlying_type_for_vpn(webrtc::ADAPTER_TYPE_ETHERNET); + Network* net1 = new Network("em1", "em1", TruncateIP(ip1, 64), 64); + net1->set_type(ADAPTER_TYPE_VPN); + net1->set_underlying_type_for_vpn(ADAPTER_TYPE_ETHERNET); - Network* net2 = new Network("em1", "em1", webrtc::TruncateIP(ip1, 64), 64); - net2->set_type(webrtc::ADAPTER_TYPE_ETHERNET); + Network* net2 = new Network("em1", "em1", TruncateIP(ip1, 64), 64); + net2->set_type(ADAPTER_TYPE_ETHERNET); EXPECT_EQ(net1->GetCost(field_trials), net2->GetCost(field_trials)); delete net1; @@ -1537,14 +1516,14 @@ TEST_F(NetworkTest, NetworkCostVpn_VpnMoreExpensive) { ScopedKeyValueConfig field_trials("WebRTC-AddNetworkCostToVpn/Enabled/"); IPAddress ip1; - EXPECT_TRUE(webrtc::IPFromString("2400:4030:1:2c00:be30:0:0:1", &ip1)); + EXPECT_TRUE(IPFromString("2400:4030:1:2c00:be30:0:0:1", &ip1)); - Network* net1 = new Network("em1", "em1", webrtc::TruncateIP(ip1, 64), 64); - net1->set_type(webrtc::ADAPTER_TYPE_VPN); - net1->set_underlying_type_for_vpn(webrtc::ADAPTER_TYPE_ETHERNET); + Network* net1 = new Network("em1", "em1", TruncateIP(ip1, 64), 64); + net1->set_type(ADAPTER_TYPE_VPN); + net1->set_underlying_type_for_vpn(ADAPTER_TYPE_ETHERNET); - Network* net2 = new Network("em1", "em1", webrtc::TruncateIP(ip1, 64), 64); - net2->set_type(webrtc::ADAPTER_TYPE_ETHERNET); + Network* net2 = new Network("em1", "em1", TruncateIP(ip1, 64), 64); + net2->set_type(ADAPTER_TYPE_ETHERNET); EXPECT_GT(net1->GetCost(field_trials), net2->GetCost(field_trials)); delete net1; @@ -1557,35 +1536,35 @@ TEST_F(NetworkTest, GuessAdapterFromNetworkCost) { "WebRTC-UseDifferentiatedCellularCosts/Enabled/"); IPAddress ip1; - EXPECT_TRUE(webrtc::IPFromString("2400:4030:1:2c00:be30:0:0:1", &ip1)); + EXPECT_TRUE(IPFromString("2400:4030:1:2c00:be30:0:0:1", &ip1)); - for (auto type : webrtc::kAllAdapterTypes) { - if (type == webrtc::ADAPTER_TYPE_VPN) + for (auto type : kAllAdapterTypes) { + if (type == ADAPTER_TYPE_VPN) continue; - Network net1("em1", "em1", webrtc::TruncateIP(ip1, 64), 64); + Network net1("em1", "em1", TruncateIP(ip1, 64), 64); net1.set_type(type); auto [guess, vpn] = Network::GuessAdapterFromNetworkCost(net1.GetCost(field_trials)); EXPECT_FALSE(vpn); - if (type == webrtc::ADAPTER_TYPE_LOOPBACK) { - EXPECT_EQ(guess, webrtc::ADAPTER_TYPE_ETHERNET); + if (type == ADAPTER_TYPE_LOOPBACK) { + EXPECT_EQ(guess, ADAPTER_TYPE_ETHERNET); } else { EXPECT_EQ(type, guess); } } // VPN - for (auto type : webrtc::kAllAdapterTypes) { - if (type == webrtc::ADAPTER_TYPE_VPN) + for (auto type : kAllAdapterTypes) { + if (type == ADAPTER_TYPE_VPN) continue; - Network net1("em1", "em1", webrtc::TruncateIP(ip1, 64), 64); - net1.set_type(webrtc::ADAPTER_TYPE_VPN); + Network net1("em1", "em1", TruncateIP(ip1, 64), 64); + net1.set_type(ADAPTER_TYPE_VPN); net1.set_underlying_type_for_vpn(type); auto [guess, vpn] = Network::GuessAdapterFromNetworkCost(net1.GetCost(field_trials)); EXPECT_TRUE(vpn); - if (type == webrtc::ADAPTER_TYPE_LOOPBACK) { - EXPECT_EQ(guess, webrtc::ADAPTER_TYPE_ETHERNET); + if (type == ADAPTER_TYPE_LOOPBACK) { + EXPECT_EQ(guess, ADAPTER_TYPE_ETHERNET); } else { EXPECT_EQ(type, guess); } @@ -1630,8 +1609,8 @@ TEST_F(NetworkTest, VpnListOverrideAdapterType) { std::vector list = manager.GetNetworks(); ASSERT_EQ(1u, list.size()); - EXPECT_EQ(webrtc::ADAPTER_TYPE_VPN, list[0]->type()); - EXPECT_EQ(webrtc::ADAPTER_TYPE_ETHERNET, list[0]->underlying_type_for_vpn()); + EXPECT_EQ(ADAPTER_TYPE_VPN, list[0]->type()); + EXPECT_EQ(ADAPTER_TYPE_ETHERNET, list[0]->underlying_type_for_vpn()); ClearNetworks(manager); ReleaseIfAddrs(addr_list); } diff --git a/rtc_base/null_socket_server_unittest.cc b/rtc_base/null_socket_server_unittest.cc index b89a2415e6..81da9c3128 100644 --- a/rtc_base/null_socket_server_unittest.cc +++ b/rtc_base/null_socket_server_unittest.cc @@ -33,10 +33,10 @@ TEST(NullSocketServerTest, WaitAndSet) { thread->PostTask([&ss] { ss.WakeUp(); }); // The process_io will be ignored. const bool process_io = true; - EXPECT_THAT(webrtc::WaitUntil( - [&] { return ss.Wait(SocketServer::kForever, process_io); }, - ::testing::IsTrue(), {.timeout = TimeDelta::Millis(5'000)}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return ss.Wait(SocketServer::kForever, process_io); }, + ::testing::IsTrue(), {.timeout = TimeDelta::Millis(5'000)}), + IsRtcOk()); } TEST(NullSocketServerTest, TestWait) { diff --git a/rtc_base/openssl_adapter.cc b/rtc_base/openssl_adapter.cc index cb5dfc6a85..f804f08da3 100644 --- a/rtc_base/openssl_adapter.cc +++ b/rtc_base/openssl_adapter.cc @@ -184,7 +184,6 @@ static void LogSslError() { namespace webrtc { -using ::webrtc::TimeDelta; bool OpenSSLAdapter::InitializeSSL() { // TODO: https://issues.webrtc.org/issues/339300437 - remove once @@ -206,12 +205,12 @@ OpenSSLAdapter::OpenSSLAdapter(Socket* socket, ssl_session_cache_(ssl_session_cache), ssl_cert_verifier_(ssl_cert_verifier), state_(SSL_NONE), - role_(webrtc::SSL_CLIENT), + role_(SSL_CLIENT), ssl_read_needs_write_(false), ssl_write_needs_read_(false), ssl_(nullptr), ssl_ctx_(nullptr), - ssl_mode_(webrtc::SSL_MODE_TLS), + ssl_mode_(SSL_MODE_TLS), ignore_bad_cert_(false), custom_cert_verifier_status_(false) { // If a factory is used, take a reference on the factory's SSL_CTX. @@ -365,12 +364,12 @@ int OpenSSLAdapter::BeginSSL() { if (!tls_alpn_string.empty()) { SSL_set_alpn_protos( ssl_, reinterpret_cast(tls_alpn_string.data()), - webrtc::dchecked_cast(tls_alpn_string.size())); + dchecked_cast(tls_alpn_string.size())); } } if (!elliptic_curves_.empty()) { - SSL_set1_curves_list(ssl_, webrtc::StrJoin(elliptic_curves_, ":").c_str()); + SSL_set1_curves_list(ssl_, StrJoin(elliptic_curves_, ":").c_str()); } // Now that the initial config is done, transfer ownership of `bio` to the @@ -393,8 +392,7 @@ int OpenSSLAdapter::ContinueSSL() { // Clear the DTLS timer timer_.reset(); - int code = - (role_ == webrtc::SSL_CLIENT) ? SSL_connect(ssl_) : SSL_accept(ssl_); + int code = (role_ == SSL_CLIENT) ? SSL_connect(ssl_) : SSL_accept(ssl_); switch (SSL_get_error(ssl_, code)) { case SSL_ERROR_NONE: if (!SSLPostConnectionCheck(ssl_, ssl_host_name_)) { @@ -480,7 +478,7 @@ int OpenSSLAdapter::DoSslWrite(const void* pv, size_t cb, int* error) { RTC_DCHECK(error != nullptr); ssl_write_needs_read_ = false; - int ret = SSL_write(ssl_, pv, webrtc::checked_cast(cb)); + int ret = SSL_write(ssl_, pv, checked_cast(cb)); *error = SSL_get_error(ssl_, ret); switch (*error) { case SSL_ERROR_NONE: @@ -574,7 +572,7 @@ int OpenSSLAdapter::Send(const void* pv, size_t cb) { pending_data_.SetData(static_cast(pv), cb); // Since we're taking responsibility for sending this data, return its full // size. The user of this class can consider it sent. - return webrtc::dchecked_cast(cb); + return dchecked_cast(cb); } return ret; } @@ -612,7 +610,7 @@ int OpenSSLAdapter::Recv(void* pv, size_t cb, int64_t* timestamp) { } ssl_read_needs_write_ = false; - int code = SSL_read(ssl_, pv, webrtc::checked_cast(cb)); + int code = SSL_read(ssl_, pv, checked_cast(cb)); int error = SSL_get_error(ssl_, code); switch (error) { @@ -944,7 +942,7 @@ SSL_CTX* OpenSSLAdapter::CreateContext(SSLMode mode, bool enable_cache) { : TLS_with_buffers_method()); #else SSL_CTX* ctx = - SSL_CTX_new(mode == webrtc::SSL_MODE_DTLS ? DTLS_method() : TLS_method()); + SSL_CTX_new(mode == SSL_MODE_DTLS ? DTLS_method() : TLS_method()); #endif if (ctx == nullptr) { unsigned long error = ERR_get_error(); // NOLINT: type used by OpenSSL. diff --git a/rtc_base/openssl_digest.cc b/rtc_base/openssl_digest.cc index 96cac43031..ce0c6c1cb6 100644 --- a/rtc_base/openssl_digest.cc +++ b/rtc_base/openssl_digest.cc @@ -59,17 +59,17 @@ size_t OpenSSLDigest::Finish(void* buf, size_t len) { bool OpenSSLDigest::GetDigestEVP(absl::string_view algorithm, const EVP_MD** mdp) { const EVP_MD* md; - if (algorithm == webrtc::DIGEST_MD5) { + if (algorithm == DIGEST_MD5) { md = EVP_md5(); - } else if (algorithm == webrtc::DIGEST_SHA_1) { + } else if (algorithm == DIGEST_SHA_1) { md = EVP_sha1(); - } else if (algorithm == webrtc::DIGEST_SHA_224) { + } else if (algorithm == DIGEST_SHA_224) { md = EVP_sha224(); - } else if (algorithm == webrtc::DIGEST_SHA_256) { + } else if (algorithm == DIGEST_SHA_256) { md = EVP_sha256(); - } else if (algorithm == webrtc::DIGEST_SHA_384) { + } else if (algorithm == DIGEST_SHA_384) { md = EVP_sha384(); - } else if (algorithm == webrtc::DIGEST_SHA_512) { + } else if (algorithm == DIGEST_SHA_512) { md = EVP_sha512(); } else { return false; @@ -87,17 +87,17 @@ bool OpenSSLDigest::GetDigestName(const EVP_MD* md, std::string* algorithm) { int md_type = EVP_MD_type(md); if (md_type == NID_md5) { - *algorithm = webrtc::DIGEST_MD5; + *algorithm = DIGEST_MD5; } else if (md_type == NID_sha1) { - *algorithm = webrtc::DIGEST_SHA_1; + *algorithm = DIGEST_SHA_1; } else if (md_type == NID_sha224) { - *algorithm = webrtc::DIGEST_SHA_224; + *algorithm = DIGEST_SHA_224; } else if (md_type == NID_sha256) { - *algorithm = webrtc::DIGEST_SHA_256; + *algorithm = DIGEST_SHA_256; } else if (md_type == NID_sha384) { - *algorithm = webrtc::DIGEST_SHA_384; + *algorithm = DIGEST_SHA_384; } else if (md_type == NID_sha512) { - *algorithm = webrtc::DIGEST_SHA_512; + *algorithm = DIGEST_SHA_512; } else { algorithm->clear(); return false; diff --git a/rtc_base/openssl_stream_adapter.cc b/rtc_base/openssl_stream_adapter.cc index 654fc46821..3a0a55884e 100644 --- a/rtc_base/openssl_stream_adapter.cc +++ b/rtc_base/openssl_stream_adapter.cc @@ -72,7 +72,6 @@ static constexpr absl::string_view kDtlsSrtpExporterLabel = namespace webrtc { namespace { -using ::webrtc::SafeTask; // SRTP cipher suite table. `internal_name` is used to construct a // colon-separated profile strings which is needed by // SSL_CTX_set_tlsext_use_srtp(). @@ -83,10 +82,10 @@ struct SrtpCipherMapEntry { // This isn't elegant, but it's better than an external reference constexpr SrtpCipherMapEntry kSrtpCipherMap[] = { - {"SRTP_AES128_CM_SHA1_80", webrtc::kSrtpAes128CmSha1_80}, - {"SRTP_AES128_CM_SHA1_32", webrtc::kSrtpAes128CmSha1_32}, - {"SRTP_AEAD_AES_128_GCM", webrtc::kSrtpAeadAes128Gcm}, - {"SRTP_AEAD_AES_256_GCM", webrtc::kSrtpAeadAes256Gcm}}; + {"SRTP_AES128_CM_SHA1_80", kSrtpAes128CmSha1_80}, + {"SRTP_AES128_CM_SHA1_32", kSrtpAes128CmSha1_32}, + {"SRTP_AEAD_AES_128_GCM", kSrtpAeadAes128Gcm}, + {"SRTP_AEAD_AES_256_GCM", kSrtpAeadAes256Gcm}}; #ifdef OPENSSL_IS_BORINGSSL // Enabled by EnableTimeCallbackForTesting. Should never be set in production @@ -94,38 +93,37 @@ constexpr SrtpCipherMapEntry kSrtpCipherMap[] = { bool g_use_time_callback_for_testing = false; // Not used in production code. Actual time should be relative to Jan 1, 1970. void TimeCallbackForTesting(const SSL* ssl, struct timeval* out_clock) { - int64_t time = webrtc::TimeNanos(); - out_clock->tv_sec = time / webrtc::kNumNanosecsPerSec; - out_clock->tv_usec = - (time % webrtc::kNumNanosecsPerSec) / webrtc::kNumNanosecsPerMicrosec; + int64_t time = TimeNanos(); + out_clock->tv_sec = time / kNumNanosecsPerSec; + out_clock->tv_usec = (time % kNumNanosecsPerSec) / kNumNanosecsPerMicrosec; } #endif uint16_t GetMaxVersion(SSLMode ssl_mode, SSLProtocolVersion version) { switch (ssl_mode) { - case webrtc::SSL_MODE_TLS: + case SSL_MODE_TLS: switch (version) { default: - case webrtc::SSL_PROTOCOL_NOT_GIVEN: - case webrtc::SSL_PROTOCOL_TLS_10: - case webrtc::SSL_PROTOCOL_TLS_11: - case webrtc::SSL_PROTOCOL_TLS_12: + case SSL_PROTOCOL_NOT_GIVEN: + case SSL_PROTOCOL_TLS_10: + case SSL_PROTOCOL_TLS_11: + case SSL_PROTOCOL_TLS_12: return TLS1_2_VERSION; - case webrtc::SSL_PROTOCOL_TLS_13: + case SSL_PROTOCOL_TLS_13: #ifdef TLS1_3_VERSION return TLS1_3_VERSION; #else return TLS1_2_VERSION; #endif } - case webrtc::SSL_MODE_DTLS: + case SSL_MODE_DTLS: switch (version) { default: - case webrtc::SSL_PROTOCOL_NOT_GIVEN: - case webrtc::SSL_PROTOCOL_DTLS_10: - case webrtc::SSL_PROTOCOL_DTLS_12: + case SSL_PROTOCOL_NOT_GIVEN: + case SSL_PROTOCOL_DTLS_10: + case SSL_PROTOCOL_DTLS_12: return DTLS1_2_VERSION; - case webrtc::SSL_PROTOCOL_DTLS_13: + case SSL_PROTOCOL_DTLS_13: #ifdef DTLS1_3_VERSION return DTLS1_3_VERSION; #else @@ -219,9 +217,9 @@ static int stream_read(BIO* b, char* out, int outl) { int error; StreamResult result = stream->Read( MakeArrayView(reinterpret_cast(out), outl), read, error); - if (result == webrtc::SR_SUCCESS) { - return webrtc::checked_cast(read); - } else if (result == webrtc::SR_BLOCK) { + if (result == SR_SUCCESS) { + return checked_cast(read); + } else if (result == SR_BLOCK) { BIO_set_retry_read(b); } return -1; @@ -237,16 +235,16 @@ static int stream_write(BIO* b, const char* in, int inl) { int error; StreamResult result = stream->Write( MakeArrayView(reinterpret_cast(in), inl), written, error); - if (result == webrtc::SR_SUCCESS) { - return webrtc::checked_cast(written); - } else if (result == webrtc::SR_BLOCK) { + if (result == SR_SUCCESS) { + return checked_cast(written); + } else if (result == SR_BLOCK) { BIO_set_retry_write(b); } return -1; } static int stream_puts(BIO* b, const char* str) { - return stream_write(b, str, webrtc::checked_cast(strlen(str))); + return stream_write(b, str, checked_cast(strlen(str))); } static long stream_ctrl(BIO* b, int cmd, long num, void* ptr) { @@ -256,7 +254,7 @@ static long stream_ctrl(BIO* b, int cmd, long num, void* ptr) { case BIO_CTRL_EOF: { StreamInterface* stream = static_cast(ptr); // 1 means end-of-stream. - return (stream->GetState() == webrtc::SS_CLOSED) ? 1 : 0; + return (stream->GetState() == SS_CLOSED) ? 1 : 0; } case BIO_CTRL_WPENDING: case BIO_CTRL_PENDING: @@ -295,13 +293,13 @@ OpenSSLStreamAdapter::OpenSSLStreamAdapter( handshake_error_(std::move(handshake_error)), owner_(Thread::Current()), state_(SSL_NONE), - role_(webrtc::SSL_CLIENT), + role_(SSL_CLIENT), ssl_read_needs_write_(false), ssl_write_needs_read_(false), ssl_(nullptr), ssl_ctx_(nullptr), - ssl_mode_(webrtc::SSL_MODE_DTLS), - ssl_max_version_(webrtc::SSL_PROTOCOL_DTLS_12), + ssl_mode_(SSL_MODE_DTLS), + ssl_max_version_(SSL_PROTOCOL_DTLS_12), force_dtls_13_(GetForceDtls13(field_trials)), disable_ssl_group_ids_(field_trials && field_trials->IsEnabled( "WebRTC-DisableSslGroupIds")) { @@ -364,7 +362,7 @@ SSLPeerCertificateDigestError OpenSSLStreamAdapter::SetPeerCertificateDigest( // Post the event asynchronously to unwind the stack. The caller // of ContinueSSL may be the same object listening for these // events and may not be prepared for reentrancy. - PostEvent(webrtc::SE_OPEN | webrtc::SE_READ | webrtc::SE_WRITE, 0); + PostEvent(SE_OPEN | SE_READ | SE_WRITE, 0); } return SSLPeerCertificateDigestError::NONE; } @@ -395,37 +393,37 @@ bool OpenSSLStreamAdapter::GetSslCipherSuite(int* cipher_suite) const { SSLProtocolVersion OpenSSLStreamAdapter::GetSslVersion() const { if (state_ != SSL_CONNECTED) { - return webrtc::SSL_PROTOCOL_NOT_GIVEN; + return SSL_PROTOCOL_NOT_GIVEN; } int ssl_version = SSL_version(ssl_); - if (ssl_mode_ == webrtc::SSL_MODE_DTLS) { + if (ssl_mode_ == SSL_MODE_DTLS) { if (ssl_version == DTLS1_VERSION) { - return webrtc::SSL_PROTOCOL_DTLS_10; + return SSL_PROTOCOL_DTLS_10; } else if (ssl_version == DTLS1_2_VERSION) { - return webrtc::SSL_PROTOCOL_DTLS_12; + return SSL_PROTOCOL_DTLS_12; } #ifdef DTLS1_3_VERSION if (ssl_version == DTLS1_3_VERSION) { - return webrtc::SSL_PROTOCOL_DTLS_13; + return SSL_PROTOCOL_DTLS_13; } #endif } else { if (ssl_version == TLS1_VERSION) { - return webrtc::SSL_PROTOCOL_TLS_10; + return SSL_PROTOCOL_TLS_10; } else if (ssl_version == TLS1_1_VERSION) { - return webrtc::SSL_PROTOCOL_TLS_11; + return SSL_PROTOCOL_TLS_11; } else if (ssl_version == TLS1_2_VERSION) { - return webrtc::SSL_PROTOCOL_TLS_12; + return SSL_PROTOCOL_TLS_12; } #ifdef TLS1_3_VERSION if (ssl_version == TLS1_3_VERSION) { - return webrtc::SSL_PROTOCOL_TLS_13; + return SSL_PROTOCOL_TLS_13; } #endif } - return webrtc::SSL_PROTOCOL_NOT_GIVEN; + return SSL_PROTOCOL_NOT_GIVEN; } bool OpenSSLStreamAdapter::GetSslVersionBytes(int* version) const { @@ -524,7 +522,7 @@ bool OpenSSLStreamAdapter::GetDtlsSrtpCryptoSuite(int* crypto_suite) const { } *crypto_suite = srtp_profile->id; - RTC_DCHECK(!webrtc::SrtpCryptoSuiteToName(*crypto_suite).empty()); + RTC_DCHECK(!SrtpCryptoSuiteToName(*crypto_suite).empty()); return true; } @@ -538,7 +536,7 @@ int OpenSSLStreamAdapter::StartSSL() { return -1; } - if (stream_->GetState() != webrtc::SS_OPEN) { + if (stream_->GetState() != SS_OPEN) { state_ = SSL_WAIT; return 0; } @@ -565,7 +563,7 @@ void OpenSSLStreamAdapter::SetMaxProtocolVersion(SSLProtocolVersion version) { void OpenSSLStreamAdapter::SetInitialRetransmissionTimeout(int timeout_ms) { dtls_handshake_timeout_ms_ = timeout_ms; #ifdef OPENSSL_IS_BORINGSSL - if (ssl_ctx_ != nullptr && ssl_mode_ == webrtc::SSL_MODE_DTLS) { + if (ssl_ctx_ != nullptr && ssl_mode_ == SSL_MODE_DTLS) { // TODO (jonaso, webrtc:367395350): Switch to upcoming // DTLSv1_set_timeout_duration. DTLSv1_set_initial_timeout_duration(ssl_, dtls_handshake_timeout_ms_); @@ -594,29 +592,28 @@ StreamResult OpenSSLStreamAdapter::Write(ArrayView data, return stream_->Write(data, written, error); case SSL_WAIT: case SSL_CONNECTING: - return webrtc::SR_BLOCK; + return SR_BLOCK; case SSL_CONNECTED: if (WaitingToVerifyPeerCertificate()) { - return webrtc::SR_BLOCK; + return SR_BLOCK; } break; case SSL_ERROR: case SSL_CLOSED: default: error = ssl_error_code_; - return webrtc::SR_ERROR; + return SR_ERROR; } // OpenSSL will return an error if we try to write zero bytes if (data.size() == 0) { written = 0; - return webrtc::SR_SUCCESS; + return SR_SUCCESS; } ssl_write_needs_read_ = false; - int code = - SSL_write(ssl_, data.data(), webrtc::checked_cast(data.size())); + int code = SSL_write(ssl_, data.data(), checked_cast(data.size())); int ssl_error = SSL_get_error(ssl_, code); switch (ssl_error) { case SSL_ERROR_NONE: @@ -624,19 +621,19 @@ StreamResult OpenSSLStreamAdapter::Write(ArrayView data, RTC_DCHECK_GT(code, 0); RTC_DCHECK_LE(code, data.size()); written = code; - return webrtc::SR_SUCCESS; + return SR_SUCCESS; case SSL_ERROR_WANT_READ: RTC_DLOG(LS_VERBOSE) << " -- error want read"; ssl_write_needs_read_ = true; - return webrtc::SR_BLOCK; + return SR_BLOCK; case SSL_ERROR_WANT_WRITE: RTC_DLOG(LS_VERBOSE) << " -- error want write"; - return webrtc::SR_BLOCK; + return SR_BLOCK; case SSL_ERROR_ZERO_RETURN: default: Error("SSL_write", (ssl_error ? ssl_error : -1), 0, false); error = ssl_error_code_; - return webrtc::SR_ERROR; + return SR_ERROR; } // not reached } @@ -651,30 +648,29 @@ StreamResult OpenSSLStreamAdapter::Read(ArrayView data, return stream_->Read(data, read, error); case SSL_WAIT: case SSL_CONNECTING: - return webrtc::SR_BLOCK; + return SR_BLOCK; case SSL_CONNECTED: if (WaitingToVerifyPeerCertificate()) { - return webrtc::SR_BLOCK; + return SR_BLOCK; } break; case SSL_CLOSED: - return webrtc::SR_EOS; + return SR_EOS; case SSL_ERROR: default: error = ssl_error_code_; - return webrtc::SR_ERROR; + return SR_ERROR; } // Don't trust OpenSSL with zero byte reads if (data.size() == 0) { read = 0; - return webrtc::SR_SUCCESS; + return SR_SUCCESS; } ssl_read_needs_write_ = false; - const int code = - SSL_read(ssl_, data.data(), webrtc::checked_cast(data.size())); + const int code = SSL_read(ssl_, data.data(), checked_cast(data.size())); const int ssl_error = SSL_get_error(ssl_, code); switch (ssl_error) { @@ -684,33 +680,33 @@ StreamResult OpenSSLStreamAdapter::Read(ArrayView data, RTC_DCHECK_LE(code, data.size()); read = code; - if (ssl_mode_ == webrtc::SSL_MODE_DTLS) { + if (ssl_mode_ == SSL_MODE_DTLS) { // Enforce atomic reads -- this is a short read unsigned int pending = SSL_pending(ssl_); if (pending) { RTC_DLOG(LS_INFO) << " -- short DTLS read. flushing"; FlushInput(pending); - error = webrtc::SSE_MSG_TRUNC; - return webrtc::SR_ERROR; + error = SSE_MSG_TRUNC; + return SR_ERROR; } } - return webrtc::SR_SUCCESS; + return SR_SUCCESS; case SSL_ERROR_WANT_READ: RTC_DLOG(LS_VERBOSE) << " -- error want read"; - return webrtc::SR_BLOCK; + return SR_BLOCK; case SSL_ERROR_WANT_WRITE: RTC_DLOG(LS_VERBOSE) << " -- error want write"; ssl_read_needs_write_ = true; - return webrtc::SR_BLOCK; + return SR_BLOCK; case SSL_ERROR_ZERO_RETURN: RTC_DLOG(LS_VERBOSE) << " -- remote side closed"; Close(); - return webrtc::SR_EOS; + return SR_EOS; default: Error("SSL_read", (ssl_error ? ssl_error : -1), 0, false); error = ssl_error_code_; - return webrtc::SR_ERROR; + return SR_ERROR; } // not reached } @@ -750,14 +746,14 @@ StreamState OpenSSLStreamAdapter::GetState() const { switch (state_) { case SSL_WAIT: case SSL_CONNECTING: - return webrtc::SS_OPENING; + return SS_OPENING; case SSL_CONNECTED: if (WaitingToVerifyPeerCertificate()) { - return webrtc::SS_OPENING; + return SS_OPENING; } - return webrtc::SS_OPEN; + return SS_OPEN; default: - return webrtc::SS_CLOSED; + return SS_CLOSED; } // not reached } @@ -767,11 +763,11 @@ void OpenSSLStreamAdapter::OnEvent(int events, int err) { int events_to_signal = 0; int signal_error = 0; - if ((events & webrtc::SE_OPEN)) { + if ((events & SE_OPEN)) { RTC_DLOG(LS_VERBOSE) << "OpenSSLStreamAdapter::OnEvent SE_OPEN"; if (state_ != SSL_WAIT) { RTC_DCHECK(state_ == SSL_NONE); - events_to_signal |= webrtc::SE_OPEN; + events_to_signal |= SE_OPEN; } else { state_ = SSL_CONNECTING; if (int error = BeginSSL()) { @@ -781,36 +777,36 @@ void OpenSSLStreamAdapter::OnEvent(int events, int err) { } } - if ((events & (webrtc::SE_READ | webrtc::SE_WRITE))) { + if ((events & (SE_READ | SE_WRITE))) { RTC_DLOG(LS_VERBOSE) << "OpenSSLStreamAdapter::OnEvent" - << ((events & webrtc::SE_READ) ? " SE_READ" : "") - << ((events & webrtc::SE_WRITE) ? " SE_WRITE" : ""); + << ((events & SE_READ) ? " SE_READ" : "") + << ((events & SE_WRITE) ? " SE_WRITE" : ""); if (state_ == SSL_NONE) { - events_to_signal |= events & (webrtc::SE_READ | webrtc::SE_WRITE); + events_to_signal |= events & (SE_READ | SE_WRITE); } else if (state_ == SSL_CONNECTING) { if (int error = ContinueSSL()) { Error("ContinueSSL", error, 0, true); return; } } else if (state_ == SSL_CONNECTED) { - if (((events & webrtc::SE_READ) && ssl_write_needs_read_) || - (events & webrtc::SE_WRITE)) { + if (((events & SE_READ) && ssl_write_needs_read_) || + (events & SE_WRITE)) { RTC_DLOG(LS_VERBOSE) << " -- onStreamWriteable"; - events_to_signal |= webrtc::SE_WRITE; + events_to_signal |= SE_WRITE; } - if (((events & webrtc::SE_WRITE) && ssl_read_needs_write_) || - (events & webrtc::SE_READ)) { + if (((events & SE_WRITE) && ssl_read_needs_write_) || + (events & SE_READ)) { RTC_DLOG(LS_VERBOSE) << " -- onStreamReadable"; - events_to_signal |= webrtc::SE_READ; + events_to_signal |= SE_READ; } } } - if ((events & webrtc::SE_CLOSE)) { + if ((events & SE_CLOSE)) { RTC_DLOG(LS_VERBOSE) << "OpenSSLStreamAdapter::OnEvent(SE_CLOSE, " << err << ")"; Cleanup(0); - events_to_signal |= webrtc::SE_CLOSE; + events_to_signal |= SE_CLOSE; // SE_CLOSE is the only event that uses the final parameter to OnEvent(). RTC_DCHECK(signal_error == 0); signal_error = err; @@ -899,7 +895,7 @@ int OpenSSLStreamAdapter::BeginSSL() { SSL_set_mtu(ssl_, dtls_mtu_); #ifdef OPENSSL_IS_BORINGSSL - if (ssl_mode_ == webrtc::SSL_MODE_DTLS) { + if (ssl_mode_ == SSL_MODE_DTLS) { DTLSv1_set_initial_timeout_duration(ssl_, dtls_handshake_timeout_ms_); } @@ -927,8 +923,7 @@ int OpenSSLStreamAdapter::ContinueSSL() { // Clear the DTLS timer timeout_task_.Stop(); - const int code = - (role_ == webrtc::SSL_CLIENT) ? SSL_connect(ssl_) : SSL_accept(ssl_); + const int code = (role_ == SSL_CLIENT) ? SSL_connect(ssl_) : SSL_accept(ssl_); const int ssl_error = SSL_get_error(ssl_, code); switch (ssl_error) { @@ -948,7 +943,7 @@ int OpenSSLStreamAdapter::ContinueSSL() { // The caller of ContinueSSL may be the same object listening for these // events and may not be prepared for reentrancy. // PostEvent(SE_OPEN | SE_READ | SE_WRITE, 0); - FireEvent(webrtc::SE_OPEN | webrtc::SE_READ | webrtc::SE_WRITE, 0); + FireEvent(SE_OPEN | SE_READ | SE_WRITE, 0); } break; case SSL_ERROR_WANT_READ: @@ -995,7 +990,7 @@ void OpenSSLStreamAdapter::Error(absl::string_view context, ssl_error_code_ = err; Cleanup(alert); if (signal) { - FireEvent(webrtc::SE_CLOSE, err); + FireEvent(SE_CLOSE, err); } } @@ -1045,9 +1040,9 @@ SSL_CTX* OpenSSLStreamAdapter::SetupSSLContext() { #ifdef OPENSSL_IS_BORINGSSL // If X509 objects aren't used, we can use these methods to avoid // linking the sizable crypto/x509 code, using CRYPTO_BUFFER instead. - SSL_CTX* ctx = SSL_CTX_new(ssl_mode_ == webrtc::SSL_MODE_DTLS - ? DTLS_with_buffers_method() - : TLS_with_buffers_method()); + SSL_CTX* ctx = + SSL_CTX_new(ssl_mode_ == SSL_MODE_DTLS ? DTLS_with_buffers_method() + : TLS_with_buffers_method()); #else SSL_CTX* ctx = SSL_CTX_new(ssl_mode_ == SSL_MODE_DTLS ? DTLS_method() : TLS_method()); @@ -1057,7 +1052,7 @@ SSL_CTX* OpenSSLStreamAdapter::SetupSSLContext() { } auto min_version = - ssl_mode_ == webrtc::SSL_MODE_DTLS ? DTLS1_2_VERSION : TLS1_2_VERSION; + ssl_mode_ == SSL_MODE_DTLS ? DTLS1_2_VERSION : TLS1_2_VERSION; auto max_version = GetMaxVersion(ssl_mode_, ssl_max_version_); #ifdef DTLS1_3_VERSION if (force_dtls_13_ == kForceDtls13Enabled) { @@ -1348,7 +1343,7 @@ void OpenSSLStreamAdapter::EnableTimeCallbackForTesting() { SSLProtocolVersion OpenSSLStreamAdapter::GetMaxSupportedDTLSProtocolVersion() { #if defined(OPENSSL_IS_BORINGSSL) && defined(DTLS1_3_VERSION) - return webrtc::SSL_PROTOCOL_DTLS_13; + return SSL_PROTOCOL_DTLS_13; #else return SSL_PROTOCOL_DTLS_12; #endif diff --git a/rtc_base/openssl_utility.cc b/rtc_base/openssl_utility.cc index 906127f932..9c6dcf8448 100644 --- a/rtc_base/openssl_utility.cc +++ b/rtc_base/openssl_utility.cc @@ -152,8 +152,8 @@ bool ParseCertificate(CRYPTO_BUFFER* cert_buffer, return false; } if (expiration_time) { - *expiration_time = webrtc::ASN1TimeToSec(CBS_data(¬_after), - CBS_len(¬_after), long_format); + *expiration_time = + ASN1TimeToSec(CBS_data(¬_after), CBS_len(¬_after), long_format); } // subject Name, if (!CBS_get_asn1_element(&tbs_certificate, nullptr, CBS_ASN1_SEQUENCE)) { @@ -248,9 +248,8 @@ bool LoadBuiltinSSLRootCertificates(SSL_CTX* ctx) { for (size_t i = 0; i < arraysize(kSSLCertCertificateList); i++) { const unsigned char* cert_buffer = kSSLCertCertificateList[i]; size_t cert_buffer_len = kSSLCertCertificateSizeList[i]; - X509* cert = - d2i_X509(nullptr, &cert_buffer, - webrtc::checked_cast(cert_buffer_len)); // NOLINT + X509* cert = d2i_X509(nullptr, &cert_buffer, + checked_cast(cert_buffer_len)); // NOLINT if (cert) { int return_value = X509_STORE_add_cert(SSL_CTX_get_cert_store(ctx), cert); if (return_value == 0) { diff --git a/rtc_base/openssl_utility_unittest.cc b/rtc_base/openssl_utility_unittest.cc index 1f0fcd9af0..3008147ac3 100644 --- a/rtc_base/openssl_utility_unittest.cc +++ b/rtc_base/openssl_utility_unittest.cc @@ -187,7 +187,7 @@ SSL* CreateSSLWithPeerCertificate(const unsigned char* cert, size_t cert_len) { const unsigned char* key_ptr = kFakeSSLPrivateKey; EVP_PKEY* key = d2i_PrivateKey( EVP_PKEY_EC, nullptr, &key_ptr, - webrtc::checked_cast(arraysize(kFakeSSLPrivateKey))); // NOLINT + checked_cast(arraysize(kFakeSSLPrivateKey))); // NOLINT RTC_CHECK(key); #ifdef OPENSSL_IS_BORINGSSL diff --git a/rtc_base/operations_chain_unittest.cc b/rtc_base/operations_chain_unittest.cc index ed5cd40e18..9f7ce22758 100644 --- a/rtc_base/operations_chain_unittest.cc +++ b/rtc_base/operations_chain_unittest.cc @@ -415,10 +415,9 @@ TEST(OperationsChainTest, OnChainEmptyCallback) { // Completing the operation empties the chain, invoking the callback. unblock_async_operation_event0.Set(); async_operation_completed_event0->Wait(Event::kForever); - EXPECT_THAT( - webrtc::WaitUntil([&] { return on_empty_callback_counter == 1u; }, + EXPECT_THAT(WaitUntil([&] { return on_empty_callback_counter == 1u; }, ::testing::IsTrue(), {.timeout = kDefaultTimeout}), - webrtc::IsRtcOk()); + IsRtcOk()); // Chain multiple events. Event unblock_async_operation_event1; @@ -430,25 +429,22 @@ TEST(OperationsChainTest, OnChainEmptyCallback) { operation_tracker_proxy.PostAsynchronousOperation( &unblock_async_operation_event2); // Again, the callback is not invoked until the operation has completed. - EXPECT_THAT( - webrtc::WaitUntil([&] { return on_empty_callback_counter == 1u; }, + EXPECT_THAT(WaitUntil([&] { return on_empty_callback_counter == 1u; }, ::testing::IsTrue(), {.timeout = kDefaultTimeout}), - webrtc::IsRtcOk()); + IsRtcOk()); // Upon completing the first event, the chain is still not empty, so the // callback must not be invoked yet. unblock_async_operation_event1.Set(); async_operation_completed_event1->Wait(Event::kForever); - EXPECT_THAT( - webrtc::WaitUntil([&] { return on_empty_callback_counter == 1u; }, + EXPECT_THAT(WaitUntil([&] { return on_empty_callback_counter == 1u; }, ::testing::IsTrue(), {.timeout = kDefaultTimeout}), - webrtc::IsRtcOk()); + IsRtcOk()); // Completing the last event empties the chain, invoking the callback. unblock_async_operation_event2.Set(); async_operation_completed_event2->Wait(Event::kForever); - EXPECT_THAT( - webrtc::WaitUntil([&] { return on_empty_callback_counter == 2u; }, + EXPECT_THAT(WaitUntil([&] { return on_empty_callback_counter == 2u; }, ::testing::IsTrue(), {.timeout = kDefaultTimeout}), - webrtc::IsRtcOk()); + IsRtcOk()); } TEST(OperationsChainTest, diff --git a/rtc_base/physical_socket_server.cc b/rtc_base/physical_socket_server.cc index 4769f41bc0..b18052a0a7 100644 --- a/rtc_base/physical_socket_server.cc +++ b/rtc_base/physical_socket_server.cc @@ -204,7 +204,7 @@ SocketAddress PhysicalSocket::GetLocalAddress() const { int result = ::getsockname(s_, addr, &addrlen); SocketAddress address; if (result >= 0) { - webrtc::SocketAddressFromSockAddrStorage(addr_storage, &address); + SocketAddressFromSockAddrStorage(addr_storage, &address); } else { RTC_LOG(LS_WARNING) << "GetLocalAddress: unable to get local addr, socket=" << s_; @@ -219,7 +219,7 @@ SocketAddress PhysicalSocket::GetRemoteAddress() const { int result = ::getpeername(s_, addr, &addrlen); SocketAddress address; if (result >= 0) { - webrtc::SocketAddressFromSockAddrStorage(addr_storage, &address); + SocketAddressFromSockAddrStorage(addr_storage, &address); } else { RTC_LOG(LS_WARNING) << "GetRemoteAddress: unable to get remote addr, socket=" << s_; @@ -239,8 +239,7 @@ int PhysicalSocket::Bind(const SocketAddress& bind_addr) { // Since the network binder handled binding the socket to the desired // network interface, we don't need to (and shouldn't) include an IP in // the bind() call; bind() just needs to assign a port. - copied_bind_addr.SetIP( - webrtc::GetAnyIP(copied_bind_addr.ipaddr().family())); + copied_bind_addr.SetIP(GetAnyIP(copied_bind_addr.ipaddr().family())); } else if (result == NetworkBindingResult::NOT_IMPLEMENTED) { RTC_LOG(LS_INFO) << "Can't bind socket to network because " "network binding is not implemented for this OS."; @@ -305,7 +304,7 @@ int PhysicalSocket::DoConnect(const SocketAddress& connect_addr) { uint8_t events = DE_READ | DE_WRITE; if (err == 0) { state_ = CS_CONNECTED; - } else if (webrtc::IsBlockingError(GetError())) { + } else if (IsBlockingError(GetError())) { state_ = CS_CONNECTING; events |= DE_CONNECT; } else { @@ -417,7 +416,7 @@ int PhysicalSocket::Send(const void* pv, size_t cb) { // We have seen minidumps where this may be false. RTC_DCHECK(sent <= static_cast(cb)); if ((sent > 0 && sent < static_cast(cb)) || - (sent < 0 && webrtc::IsBlockingError(GetError()))) { + (sent < 0 && IsBlockingError(GetError()))) { EnableEvents(DE_WRITE); } return sent; @@ -442,7 +441,7 @@ int PhysicalSocket::SendTo(const void* buffer, // We have seen minidumps where this may be false. RTC_DCHECK(sent <= static_cast(length)); if ((sent > 0 && sent < static_cast(length)) || - (sent < 0 && webrtc::IsBlockingError(GetError()))) { + (sent < 0 && IsBlockingError(GetError()))) { EnableEvents(DE_WRITE); } return sent; @@ -465,7 +464,7 @@ int PhysicalSocket::Recv(void* buffer, size_t length, int64_t* timestamp) { UpdateLastError(); int error = GetError(); - bool success = (received >= 0) || webrtc::IsBlockingError(error); + bool success = (received >= 0) || IsBlockingError(error); if (udp_ || success) { EnableEvents(DE_READ); } @@ -483,7 +482,7 @@ int PhysicalSocket::RecvFrom(void* buffer, UpdateLastError(); int error = GetError(); - bool success = (received >= 0) || webrtc::IsBlockingError(error); + bool success = (received >= 0) || IsBlockingError(error); if (udp_ || success) { EnableEvents(DE_READ); } @@ -507,7 +506,7 @@ int PhysicalSocket::RecvFrom(ReceiveBuffer& buffer) { } UpdateLastError(); int error = GetError(); - bool success = (received >= 0) || webrtc::IsBlockingError(error); + bool success = (received >= 0) || IsBlockingError(error); if (udp_ || success) { EnableEvents(DE_READ); } @@ -569,7 +568,7 @@ int PhysicalSocket::DoReadFromSocket(void* buffer, } } if (out_addr) { - webrtc::SocketAddressFromSockAddrStorage(addr_storage, out_addr); + SocketAddressFromSockAddrStorage(addr_storage, out_addr); } return received; @@ -616,7 +615,7 @@ Socket* PhysicalSocket::Accept(SocketAddress* out_addr) { if (s == INVALID_SOCKET) return nullptr; if (out_addr != nullptr) - webrtc::SocketAddressFromSockAddrStorage(addr_storage, out_addr); + SocketAddressFromSockAddrStorage(addr_storage, out_addr); return ss_->WrapSocket(s); } diff --git a/rtc_base/physical_socket_server_unittest.cc b/rtc_base/physical_socket_server_unittest.cc index ac1743e0f9..56ec60903a 100644 --- a/rtc_base/physical_socket_server_unittest.cc +++ b/rtc_base/physical_socket_server_unittest.cc @@ -208,13 +208,13 @@ void PhysicalSocketTest::ConnectInternalAcceptError(const IPAddress& loopback) { server_.CreateSocket(loopback.family(), SOCK_STREAM)); sink.Monitor(client1.get()); EXPECT_EQ(Socket::CS_CLOSED, client1->GetState()); - EXPECT_TRUE(webrtc::IsUnspecOrEmptyIP(client1->GetLocalAddress().ipaddr())); + EXPECT_TRUE(IsUnspecOrEmptyIP(client1->GetLocalAddress().ipaddr())); std::unique_ptr client2( server_.CreateSocket(loopback.family(), SOCK_STREAM)); sink.Monitor(client2.get()); EXPECT_EQ(Socket::CS_CLOSED, client2->GetState()); - EXPECT_TRUE(webrtc::IsUnspecOrEmptyIP(client2->GetLocalAddress().ipaddr())); + EXPECT_TRUE(IsUnspecOrEmptyIP(client2->GetLocalAddress().ipaddr())); // Create server and listen. std::unique_ptr server( @@ -225,7 +225,7 @@ void PhysicalSocketTest::ConnectInternalAcceptError(const IPAddress& loopback) { EXPECT_EQ(Socket::CS_CONNECTING, server->GetState()); // Ensure no pending server connections, since we haven't done anything yet. - EXPECT_FALSE(sink.Check(server.get(), webrtc::testing::SSE_READ)); + EXPECT_FALSE(sink.Check(server.get(), testing::SSE_READ)); EXPECT_TRUE(nullptr == server->Accept(&accept_addr)); EXPECT_TRUE(accept_addr.IsNil()); @@ -236,15 +236,14 @@ void PhysicalSocketTest::ConnectInternalAcceptError(const IPAddress& loopback) { // Client is connecting, outcome not yet determined. EXPECT_EQ(Socket::CS_CONNECTING, client1->GetState()); - EXPECT_FALSE(sink.Check(client1.get(), webrtc::testing::SSE_OPEN)); - EXPECT_FALSE(sink.Check(client1.get(), webrtc::testing::SSE_CLOSE)); + EXPECT_FALSE(sink.Check(client1.get(), testing::SSE_OPEN)); + EXPECT_FALSE(sink.Check(client1.get(), testing::SSE_CLOSE)); // Server has pending connection, try to accept it (will fail). EXPECT_THAT( - webrtc::WaitUntil( - [&] { return (sink.Check(server.get(), webrtc::testing::SSE_READ)); }, - ::testing::IsTrue()), - webrtc::IsRtcOk()); + WaitUntil([&] { return (sink.Check(server.get(), testing::SSE_READ)); }, + ::testing::IsTrue()), + IsRtcOk()); // Simulate "::accept" returning an error. SetFailAccept(true); std::unique_ptr accepted(server->Accept(&accept_addr)); @@ -252,7 +251,7 @@ void PhysicalSocketTest::ConnectInternalAcceptError(const IPAddress& loopback) { ASSERT_TRUE(accept_addr.IsNil()); // Ensure no more pending server connections. - EXPECT_FALSE(sink.Check(server.get(), webrtc::testing::SSE_READ)); + EXPECT_FALSE(sink.Check(server.get(), testing::SSE_READ)); EXPECT_TRUE(nullptr == server->Accept(&accept_addr)); EXPECT_TRUE(accept_addr.IsNil()); @@ -263,15 +262,14 @@ void PhysicalSocketTest::ConnectInternalAcceptError(const IPAddress& loopback) { // Client is connecting, outcome not yet determined. EXPECT_EQ(Socket::CS_CONNECTING, client2->GetState()); - EXPECT_FALSE(sink.Check(client2.get(), webrtc::testing::SSE_OPEN)); - EXPECT_FALSE(sink.Check(client2.get(), webrtc::testing::SSE_CLOSE)); + EXPECT_FALSE(sink.Check(client2.get(), testing::SSE_OPEN)); + EXPECT_FALSE(sink.Check(client2.get(), testing::SSE_CLOSE)); // Server has pending connection, try to accept it (will succeed). EXPECT_THAT( - webrtc::WaitUntil( - [&] { return (sink.Check(server.get(), webrtc::testing::SSE_READ)); }, - ::testing::IsTrue()), - webrtc::IsRtcOk()); + WaitUntil([&] { return (sink.Check(server.get(), testing::SSE_READ)); }, + ::testing::IsTrue()), + IsRtcOk()); SetFailAccept(false); std::unique_ptr accepted2(server->Accept(&accept_addr)); ASSERT_TRUE(accepted2); diff --git a/rtc_base/rate_tracker.cc b/rtc_base/rate_tracker.cc index 83481c196d..f6f4cf32b9 100644 --- a/rtc_base/rate_tracker.cc +++ b/rtc_base/rate_tracker.cc @@ -64,7 +64,7 @@ double RateTracker::ComputeRateForInterval( buckets_to_skip = bucket_count_ - current_bucket_; milliseconds_to_skip = 0; available_interval_milliseconds = - webrtc::TimeDiff(current_time, initialization_time_milliseconds_); + TimeDiff(current_time, initialization_time_milliseconds_); // Let one bucket interval pass after initialization before reporting. if (available_interval_milliseconds < bucket_milliseconds_) { return 0.0; @@ -102,7 +102,7 @@ double RateTracker::ComputeTotalRate() const { } return static_cast(total_sample_count_ * 1000) / static_cast( - webrtc::TimeDiff(current_time, initialization_time_milliseconds_)); + TimeDiff(current_time, initialization_time_milliseconds_)); } int64_t RateTracker::TotalSampleCount() const { @@ -139,7 +139,7 @@ void RateTracker::AddSamplesAtTime(int64_t current_time_ms, } int64_t RateTracker::Time() const { - return webrtc::TimeMillis(); + return TimeMillis(); } void RateTracker::EnsureInitialized() { diff --git a/rtc_base/rtc_certificate_generator_unittest.cc b/rtc_base/rtc_certificate_generator_unittest.cc index a4388f566f..06bb5812e4 100644 --- a/rtc_base/rtc_certificate_generator_unittest.cc +++ b/rtc_base/rtc_certificate_generator_unittest.cc @@ -97,10 +97,9 @@ TEST_F(RTCCertificateGeneratorTest, GenerateAsyncECDSA) { // posted to this thread (which is done by `EXPECT_TRUE_WAIT`). EXPECT_FALSE(fixture_.GenerateAsyncCompleted()); EXPECT_FALSE(fixture_.certificate()); - EXPECT_THAT( - webrtc::WaitUntil([&] { return fixture_.GenerateAsyncCompleted(); }, + EXPECT_THAT(WaitUntil([&] { return fixture_.GenerateAsyncCompleted(); }, ::testing::IsTrue(), {.timeout = kGenerationTimeoutMs}), - webrtc::IsRtcOk()); + IsRtcOk()); EXPECT_TRUE(fixture_.certificate()); } @@ -140,10 +139,9 @@ TEST_F(RTCCertificateGeneratorTest, GenerateWithInvalidParamsShouldFail) { fixture_.generator()->GenerateCertificateAsync(invalid_params, std::nullopt, fixture_.OnGenerated()); - EXPECT_THAT( - webrtc::WaitUntil([&] { return fixture_.GenerateAsyncCompleted(); }, + EXPECT_THAT(WaitUntil([&] { return fixture_.GenerateAsyncCompleted(); }, ::testing::IsTrue(), {.timeout = kGenerationTimeoutMs}), - webrtc::IsRtcOk()); + IsRtcOk()); EXPECT_FALSE(fixture_.certificate()); } diff --git a/rtc_base/rtc_certificate_unittest.cc b/rtc_base/rtc_certificate_unittest.cc index 67f7e2cb3a..5233a80d27 100644 --- a/rtc_base/rtc_certificate_unittest.cc +++ b/rtc_base/rtc_certificate_unittest.cc @@ -68,7 +68,7 @@ class RTCCertificateTest : public ::testing::Test { // or later. scoped_refptr GenerateCertificateWithExpires( uint64_t expires_s) const { - RTC_CHECK(webrtc::IsValueInRangeForNumericType(expires_s)); + RTC_CHECK(IsValueInRangeForNumericType(expires_s)); SSLIdentityParams params; params.common_name = kTestCertCommonName; diff --git a/rtc_base/socket_address.cc b/rtc_base/socket_address.cc index 1e7d9418b8..7c08bf42ef 100644 --- a/rtc_base/socket_address.cc +++ b/rtc_base/socket_address.cc @@ -65,11 +65,11 @@ void SocketAddress::Clear() { } bool SocketAddress::IsNil() const { - return hostname_.empty() && webrtc::IPIsUnspec(ip_) && 0 == port_; + return hostname_.empty() && IPIsUnspec(ip_) && 0 == port_; } bool SocketAddress::IsComplete() const { - return (!webrtc::IPIsAny(ip_)) && (0 != port_); + return (!IPIsAny(ip_)) && (0 != port_); } SocketAddress& SocketAddress::operator=(const SocketAddress& addr) { @@ -97,7 +97,7 @@ void SocketAddress::SetIP(const IPAddress& ip) { void SocketAddress::SetIP(absl::string_view hostname) { hostname_ = std::string(hostname); - literal_ = webrtc::IPFromString(hostname, &ip_); + literal_ = IPFromString(hostname, &ip_); if (!literal_) { ip_ = IPAddress(); } @@ -115,7 +115,7 @@ void SocketAddress::SetResolvedIP(const IPAddress& ip) { } void SocketAddress::SetPort(int port) { - port_ = webrtc::dchecked_cast(port); + port_ = dchecked_cast(port); } uint32_t SocketAddress::ip() const { @@ -214,20 +214,20 @@ bool SocketAddress::FromString(absl::string_view str) { } bool SocketAddress::IsAnyIP() const { - return webrtc::IPIsAny(ip_); + return IPIsAny(ip_); } bool SocketAddress::IsLoopbackIP() const { - return webrtc::IPIsLoopback(ip_) || - (webrtc::IPIsAny(ip_) && 0 == strcmp(hostname_.c_str(), "localhost")); + return IPIsLoopback(ip_) || + (IPIsAny(ip_) && 0 == strcmp(hostname_.c_str(), "localhost")); } bool SocketAddress::IsPrivateIP() const { - return webrtc::IPIsPrivate(ip_); + return IPIsPrivate(ip_); } bool SocketAddress::IsUnresolvedIP() const { - return webrtc::IPIsUnspec(ip_) && !literal_ && !hostname_.empty(); + return IPIsUnspec(ip_) && !literal_ && !hostname_.empty(); } bool SocketAddress::operator==(const SocketAddress& addr) const { @@ -240,8 +240,7 @@ bool SocketAddress::operator<(const SocketAddress& addr) const { // We only check hostnames if both IPs are ANY or unspecified. This matches // EqualIPs(). - if ((webrtc::IPIsAny(ip_) || webrtc::IPIsUnspec(ip_)) && - hostname_ != addr.hostname_) + if ((IPIsAny(ip_) || IPIsUnspec(ip_)) && hostname_ != addr.hostname_) return hostname_ < addr.hostname_; return port_ < addr.port_; @@ -249,8 +248,7 @@ bool SocketAddress::operator<(const SocketAddress& addr) const { bool SocketAddress::EqualIPs(const SocketAddress& addr) const { return (ip_ == addr.ip_) && - ((!webrtc::IPIsAny(ip_) && !webrtc::IPIsUnspec(ip_)) || - (hostname_ == addr.hostname_)); + ((!IPIsAny(ip_) && !IPIsUnspec(ip_)) || (hostname_ == addr.hostname_)); } bool SocketAddress::EqualPorts(const SocketAddress& addr) const { @@ -259,7 +257,7 @@ bool SocketAddress::EqualPorts(const SocketAddress& addr) const { size_t SocketAddress::Hash() const { size_t h = 0; - h ^= webrtc::HashIP(ip_); + h ^= HashIP(ip_); h ^= port_ | (port_ << 16); return h; } @@ -271,8 +269,8 @@ void SocketAddress::ToSockAddr(sockaddr_in* saddr) const { return; } saddr->sin_family = AF_INET; - saddr->sin_port = webrtc::HostToNetwork16(port_); - if (webrtc::IPIsAny(ip_)) { + saddr->sin_port = HostToNetwork16(port_); + if (IPIsAny(ip_)) { saddr->sin_addr.s_addr = INADDR_ANY; } else { saddr->sin_addr = ip_.ipv4_address(); @@ -282,8 +280,8 @@ void SocketAddress::ToSockAddr(sockaddr_in* saddr) const { bool SocketAddress::FromSockAddr(const sockaddr_in& saddr) { if (saddr.sin_family != AF_INET) return false; - SetIP(webrtc::NetworkToHost32(saddr.sin_addr.s_addr)); - SetPort(webrtc::NetworkToHost16(saddr.sin_port)); + SetIP(NetworkToHost32(saddr.sin_addr.s_addr)); + SetPort(NetworkToHost16(saddr.sin_port)); literal_ = false; return true; } @@ -297,13 +295,13 @@ static size_t ToSockAddrStorageHelper(sockaddr_storage* addr, if (addr->ss_family == AF_INET6) { sockaddr_in6* saddr = reinterpret_cast(addr); saddr->sin6_addr = ip.ipv6_address(); - saddr->sin6_port = webrtc::HostToNetwork16(port); + saddr->sin6_port = HostToNetwork16(port); saddr->sin6_scope_id = scope_id; return sizeof(sockaddr_in6); } else if (addr->ss_family == AF_INET) { sockaddr_in* saddr = reinterpret_cast(addr); saddr->sin_addr = ip.ipv4_address(); - saddr->sin_port = webrtc::HostToNetwork16(port); + saddr->sin_port = HostToNetwork16(port); return sizeof(sockaddr_in); } return 0; @@ -325,12 +323,12 @@ bool SocketAddressFromSockAddrStorage(const sockaddr_storage& addr, if (addr.ss_family == AF_INET) { const sockaddr_in* saddr = reinterpret_cast(&addr); *out = SocketAddress(IPAddress(saddr->sin_addr), - webrtc::NetworkToHost16(saddr->sin_port)); + NetworkToHost16(saddr->sin_port)); return true; } else if (addr.ss_family == AF_INET6) { const sockaddr_in6* saddr = reinterpret_cast(&addr); *out = SocketAddress(IPAddress(saddr->sin6_addr), - webrtc::NetworkToHost16(saddr->sin6_port)); + NetworkToHost16(saddr->sin6_port)); out->SetScopeID(saddr->sin6_scope_id); return true; } diff --git a/rtc_base/socket_unittest.cc b/rtc_base/socket_unittest.cc index c7fd137505..4f5d219a4f 100644 --- a/rtc_base/socket_unittest.cc +++ b/rtc_base/socket_unittest.cc @@ -52,12 +52,12 @@ namespace webrtc { -using webrtc::testing::SSE_CLOSE; -using webrtc::testing::SSE_ERROR; -using webrtc::testing::SSE_OPEN; -using webrtc::testing::SSE_READ; -using webrtc::testing::SSE_WRITE; -using webrtc::testing::StreamSink; +using testing::SSE_CLOSE; +using testing::SSE_ERROR; +using testing::SSE_OPEN; +using testing::SSE_READ; +using testing::SSE_WRITE; +using testing::StreamSink; // Data size to be used in TcpInternal tests. static const size_t kTcpInternalDataSize = 1024 * 1024; // bytes @@ -253,7 +253,7 @@ void SocketTest::TestSocketSendRecvWithEcnIPV6() { // values on Windows, but an empty address of the same family on Linux/MacOS X. bool IsUnspecOrEmptyIP(const IPAddress& address) { #if !defined(WEBRTC_WIN) - return webrtc::IPIsAny(address); + return IPIsAny(address); #else return address.family() == AF_UNSPEC; #endif @@ -294,10 +294,9 @@ void SocketTest::ConnectInternal(const IPAddress& loopback) { EXPECT_FALSE(sink.Check(client.get(), SSE_CLOSE)); // Server has pending connection, accept it. - EXPECT_THAT( - webrtc::WaitUntil([&] { return (sink.Check(server.get(), SSE_READ)); }, + EXPECT_THAT(WaitUntil([&] { return (sink.Check(server.get(), SSE_READ)); }, ::testing::IsTrue()), - webrtc::IsRtcOk()); + IsRtcOk()); std::unique_ptr accepted(server->Accept(&accept_addr)); ASSERT_TRUE(accepted); EXPECT_FALSE(accept_addr.IsNil()); @@ -309,9 +308,9 @@ void SocketTest::ConnectInternal(const IPAddress& loopback) { EXPECT_EQ(client->GetLocalAddress(), accepted->GetRemoteAddress()); // Connected from client perspective, check the addresses are correct. - EXPECT_THAT(webrtc::WaitUntil([&] { return client->GetState(); }, - ::testing::Eq(Socket::CS_CONNECTED)), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return client->GetState(); }, + ::testing::Eq(Socket::CS_CONNECTED)), + IsRtcOk()); EXPECT_TRUE(sink.Check(client.get(), SSE_OPEN)); EXPECT_FALSE(sink.Check(client.get(), SSE_CLOSE)); EXPECT_EQ(client->GetRemoteAddress(), server->GetLocalAddress()); @@ -348,10 +347,9 @@ void SocketTest::ConnectWithDnsLookupInternal(const IPAddress& loopback, EXPECT_FALSE(sink.Check(client.get(), SSE_CLOSE)); // Server has pending connection, accept it. - EXPECT_THAT( - webrtc::WaitUntil([&] { return (sink.Check(server.get(), SSE_READ)); }, + EXPECT_THAT(WaitUntil([&] { return (sink.Check(server.get(), SSE_READ)); }, ::testing::IsTrue()), - webrtc::IsRtcOk()); + IsRtcOk()); std::unique_ptr accepted(server->Accept(&accept_addr)); ASSERT_TRUE(accepted); EXPECT_FALSE(accept_addr.IsNil()); @@ -363,9 +361,9 @@ void SocketTest::ConnectWithDnsLookupInternal(const IPAddress& loopback, EXPECT_EQ(client->GetLocalAddress(), accepted->GetRemoteAddress()); // Connected from client perspective, check the addresses are correct. - EXPECT_THAT(webrtc::WaitUntil([&] { return client->GetState(); }, - ::testing::Eq(Socket::CS_CONNECTED)), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return client->GetState(); }, + ::testing::Eq(Socket::CS_CONNECTED)), + IsRtcOk()); EXPECT_TRUE(sink.Check(client.get(), SSE_OPEN)); EXPECT_FALSE(sink.Check(client.get(), SSE_CLOSE)); EXPECT_EQ(client->GetRemoteAddress(), server->GetLocalAddress()); @@ -394,9 +392,9 @@ void SocketTest::ConnectFailInternal(const IPAddress& loopback) { EXPECT_EQ(0, client->Connect(bogus_addr)); // Wait for connection to fail (ECONNREFUSED). - EXPECT_THAT(webrtc::WaitUntil([&] { return client->GetState(); }, - ::testing::Eq(Socket::CS_CLOSED)), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return client->GetState(); }, + ::testing::Eq(Socket::CS_CLOSED)), + IsRtcOk()); EXPECT_FALSE(sink.Check(client.get(), SSE_OPEN)); EXPECT_TRUE(sink.Check(client.get(), SSE_ERROR)); EXPECT_TRUE(client->GetRemoteAddress().IsNil()); @@ -437,9 +435,9 @@ void SocketTest::ConnectWithDnsLookupFailInternal(const IPAddress& loopback) { return; } - EXPECT_THAT(webrtc::WaitUntil([&] { return client->GetState(); }, - ::testing::Eq(Socket::CS_CLOSED)), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return client->GetState(); }, + ::testing::Eq(Socket::CS_CLOSED)), + IsRtcOk()); EXPECT_FALSE(sink.Check(client.get(), SSE_OPEN)); EXPECT_TRUE(sink.Check(client.get(), SSE_ERROR)); EXPECT_TRUE(client->GetRemoteAddress().IsNil()); @@ -486,10 +484,9 @@ void SocketTest::ConnectWhileNotClosedInternal(const IPAddress& loopback) { // Accept the original connection. SocketAddress accept_addr; - EXPECT_THAT( - webrtc::WaitUntil([&] { return (sink.Check(server.get(), SSE_READ)); }, + EXPECT_THAT(WaitUntil([&] { return (sink.Check(server.get(), SSE_READ)); }, ::testing::IsTrue()), - webrtc::IsRtcOk()); + IsRtcOk()); std::unique_ptr accepted(server->Accept(&accept_addr)); ASSERT_TRUE(accepted); EXPECT_FALSE(accept_addr.IsNil()); @@ -498,9 +495,9 @@ void SocketTest::ConnectWhileNotClosedInternal(const IPAddress& loopback) { EXPECT_EQ(Socket::CS_CONNECTED, accepted->GetState()); EXPECT_EQ(server->GetLocalAddress(), accepted->GetLocalAddress()); EXPECT_EQ(client->GetLocalAddress(), accepted->GetRemoteAddress()); - EXPECT_THAT(webrtc::WaitUntil([&] { return client->GetState(); }, - ::testing::Eq(Socket::CS_CONNECTED)), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return client->GetState(); }, + ::testing::Eq(Socket::CS_CONNECTED)), + IsRtcOk()); EXPECT_EQ(client->GetRemoteAddress(), server->GetLocalAddress()); EXPECT_EQ(client->GetRemoteAddress(), accepted->GetLocalAddress()); @@ -533,16 +530,15 @@ void SocketTest::ServerCloseDuringConnectInternal(const IPAddress& loopback) { EXPECT_EQ(0, client->Connect(server->GetLocalAddress())); // Close down the server while the socket is in the accept queue. - EXPECT_THAT( - webrtc::WaitUntil([&] { return sink.Check(server.get(), SSE_READ); }, + EXPECT_THAT(WaitUntil([&] { return sink.Check(server.get(), SSE_READ); }, ::testing::IsTrue()), - webrtc::IsRtcOk()); + IsRtcOk()); server->Close(); // This should fail the connection for the client. Clean up. - EXPECT_THAT(webrtc::WaitUntil([&] { return client->GetState(); }, - ::testing::Eq(Socket::CS_CLOSED)), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return client->GetState(); }, + ::testing::Eq(Socket::CS_CLOSED)), + IsRtcOk()); EXPECT_TRUE(sink.Check(client.get(), SSE_ERROR)); client->Close(); } @@ -567,10 +563,9 @@ void SocketTest::ClientCloseDuringConnectInternal(const IPAddress& loopback) { EXPECT_EQ(0, client->Connect(server->GetLocalAddress())); // Close down the client while the socket is in the accept queue. - EXPECT_THAT( - webrtc::WaitUntil([&] { return sink.Check(server.get(), SSE_READ); }, + EXPECT_THAT(WaitUntil([&] { return sink.Check(server.get(), SSE_READ); }, ::testing::IsTrue()), - webrtc::IsRtcOk()); + IsRtcOk()); client->Close(); // The connection should still be able to be accepted. @@ -580,9 +575,9 @@ void SocketTest::ClientCloseDuringConnectInternal(const IPAddress& loopback) { EXPECT_EQ(Socket::CS_CONNECTED, accepted->GetState()); // The accepted socket should then close (possibly with err, timing-related) - EXPECT_THAT(webrtc::WaitUntil([&] { return accepted->GetState(); }, - ::testing::Eq(Socket::CS_CLOSED)), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return accepted->GetState(); }, + ::testing::Eq(Socket::CS_CLOSED)), + IsRtcOk()); EXPECT_TRUE(sink.Check(accepted.get(), SSE_CLOSE) || sink.Check(accepted.get(), SSE_ERROR)); @@ -610,18 +605,17 @@ void SocketTest::ServerCloseInternal(const IPAddress& loopback) { EXPECT_EQ(0, client->Connect(server->GetLocalAddress())); // Accept connection. - EXPECT_THAT( - webrtc::WaitUntil([&] { return (sink.Check(server.get(), SSE_READ)); }, + EXPECT_THAT(WaitUntil([&] { return (sink.Check(server.get(), SSE_READ)); }, ::testing::IsTrue()), - webrtc::IsRtcOk()); + IsRtcOk()); std::unique_ptr accepted(server->Accept(&accept_addr)); ASSERT_TRUE(accepted); sink.Monitor(accepted.get()); // Both sides are now connected. - EXPECT_THAT(webrtc::WaitUntil([&] { return client->GetState(); }, - ::testing::Eq(Socket::CS_CONNECTED)), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return client->GetState(); }, + ::testing::Eq(Socket::CS_CONNECTED)), + IsRtcOk()); EXPECT_TRUE(sink.Check(client.get(), SSE_OPEN)); EXPECT_EQ(client->GetRemoteAddress(), accepted->GetLocalAddress()); EXPECT_EQ(accepted->GetRemoteAddress(), client->GetLocalAddress()); @@ -632,10 +626,9 @@ void SocketTest::ServerCloseInternal(const IPAddress& loopback) { EXPECT_EQ(Socket::CS_CLOSED, accepted->GetState()); // Expect that the client is notified, and has not yet closed. - EXPECT_THAT( - webrtc::WaitUntil([&] { return sink.Check(client.get(), SSE_READ); }, + EXPECT_THAT(WaitUntil([&] { return sink.Check(client.get(), SSE_READ); }, ::testing::IsTrue()), - webrtc::IsRtcOk()); + IsRtcOk()); EXPECT_FALSE(sink.Check(client.get(), SSE_CLOSE)); EXPECT_EQ(Socket::CS_CONNECTED, client->GetState()); @@ -645,9 +638,9 @@ void SocketTest::ServerCloseInternal(const IPAddress& loopback) { EXPECT_EQ('a', buffer[0]); // Now we should close, but the remote address will remain. - EXPECT_THAT(webrtc::WaitUntil([&] { return client->GetState(); }, - ::testing::Eq(Socket::CS_CLOSED)), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return client->GetState(); }, + ::testing::Eq(Socket::CS_CLOSED)), + IsRtcOk()); EXPECT_TRUE(sink.Check(client.get(), SSE_CLOSE)); EXPECT_FALSE(client->GetRemoteAddress().IsAnyIP()); @@ -695,18 +688,17 @@ void SocketTest::CloseInClosedCallbackInternal(const IPAddress& loopback) { EXPECT_EQ(0, client->Connect(server->GetLocalAddress())); // Accept connection. - EXPECT_THAT( - webrtc::WaitUntil([&] { return (sink.Check(server.get(), SSE_READ)); }, + EXPECT_THAT(WaitUntil([&] { return (sink.Check(server.get(), SSE_READ)); }, ::testing::IsTrue()), - webrtc::IsRtcOk()); + IsRtcOk()); std::unique_ptr accepted(server->Accept(&accept_addr)); ASSERT_TRUE(accepted); sink.Monitor(accepted.get()); // Both sides are now connected. - EXPECT_THAT(webrtc::WaitUntil([&] { return client->GetState(); }, - ::testing::Eq(Socket::CS_CONNECTED)), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return client->GetState(); }, + ::testing::Eq(Socket::CS_CONNECTED)), + IsRtcOk()); EXPECT_TRUE(sink.Check(client.get(), SSE_OPEN)); EXPECT_EQ(client->GetRemoteAddress(), accepted->GetLocalAddress()); EXPECT_EQ(accepted->GetRemoteAddress(), client->GetLocalAddress()); @@ -720,9 +712,9 @@ void SocketTest::CloseInClosedCallbackInternal(const IPAddress& loopback) { EXPECT_EQ(Socket::CS_CONNECTED, client->GetState()); // Now we should be closed and invalidated - EXPECT_THAT(webrtc::WaitUntil([&] { return client->GetState(); }, - ::testing::Eq(Socket::CS_CLOSED)), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return client->GetState(); }, + ::testing::Eq(Socket::CS_CLOSED)), + IsRtcOk()); EXPECT_TRUE(sink.Check(client.get(), SSE_CLOSE)); EXPECT_TRUE(Socket::CS_CLOSED == client->GetState()); } @@ -760,9 +752,8 @@ void SocketTest::DeleteInReadCallbackInternal(const IPAddress& loopback) { // event. SocketDeleter deleter(std::move(socket2)); socket1->SignalReadEvent.connect(&deleter, &SocketDeleter::Delete); - EXPECT_THAT( - webrtc::WaitUntil([&] { return deleter.deleted(); }, ::testing::IsTrue()), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return deleter.deleted(); }, ::testing::IsTrue()), + IsRtcOk()); } void SocketTest::SocketServerWaitInternal(const IPAddress& loopback) { @@ -780,10 +771,9 @@ void SocketTest::SocketServerWaitInternal(const IPAddress& loopback) { EXPECT_EQ(0, server->Listen(5)); EXPECT_EQ(0, client->Connect(server->GetLocalAddress())); - EXPECT_THAT( - webrtc::WaitUntil([&] { return (sink.Check(server.get(), SSE_READ)); }, + EXPECT_THAT(WaitUntil([&] { return (sink.Check(server.get(), SSE_READ)); }, ::testing::IsTrue()), - webrtc::IsRtcOk()); + IsRtcOk()); std::unique_ptr accepted(server->Accept(&accept_addr)); ASSERT_TRUE(accepted); @@ -792,9 +782,9 @@ void SocketTest::SocketServerWaitInternal(const IPAddress& loopback) { EXPECT_EQ(server->GetLocalAddress(), accepted->GetLocalAddress()); EXPECT_EQ(client->GetLocalAddress(), accepted->GetRemoteAddress()); - EXPECT_THAT(webrtc::WaitUntil([&] { return client->GetState(); }, - ::testing::Eq(Socket::CS_CONNECTED)), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return client->GetState(); }, + ::testing::Eq(Socket::CS_CONNECTED)), + IsRtcOk()); EXPECT_TRUE(sink.Check(client.get(), SSE_OPEN)); EXPECT_FALSE(sink.Check(client.get(), SSE_CLOSE)); EXPECT_EQ(client->GetRemoteAddress(), server->GetLocalAddress()); @@ -814,10 +804,9 @@ void SocketTest::SocketServerWaitInternal(const IPAddress& loopback) { EXPECT_FALSE(sink.Check(accepted.get(), SSE_READ)); // But should signal when process_io is true. - EXPECT_THAT( - webrtc::WaitUntil([&] { return (sink.Check(accepted.get(), SSE_READ)); }, + EXPECT_THAT(WaitUntil([&] { return (sink.Check(accepted.get(), SSE_READ)); }, ::testing::IsTrue()), - webrtc::IsRtcOk()); + IsRtcOk()); EXPECT_LT(0, accepted->Recv(buf, 1024, nullptr)); } @@ -843,18 +832,17 @@ void SocketTest::TcpInternal(const IPAddress& loopback, EXPECT_EQ(0, receiver->Connect(server->GetLocalAddress())); // Accept connection which will be used for sending. - EXPECT_THAT( - webrtc::WaitUntil([&] { return (sink.Check(server.get(), SSE_READ)); }, + EXPECT_THAT(WaitUntil([&] { return (sink.Check(server.get(), SSE_READ)); }, ::testing::IsTrue()), - webrtc::IsRtcOk()); + IsRtcOk()); std::unique_ptr sender(server->Accept(&accept_addr)); ASSERT_TRUE(sender); sink.Monitor(sender.get()); // Both sides are now connected. - EXPECT_THAT(webrtc::WaitUntil([&] { return receiver->GetState(); }, - ::testing::Eq(Socket::CS_CONNECTED)), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return receiver->GetState(); }, + ::testing::Eq(Socket::CS_CONNECTED)), + IsRtcOk()); EXPECT_TRUE(sink.Check(receiver.get(), SSE_OPEN)); EXPECT_EQ(receiver->GetRemoteAddress(), sender->GetLocalAddress()); EXPECT_EQ(sender->GetRemoteAddress(), receiver->GetLocalAddress()); @@ -907,10 +895,10 @@ void SocketTest::TcpInternal(const IPAddress& loopback, while (recv_buffer.size() < sent_size) { if (!readable) { // Wait until data is available. - EXPECT_THAT(webrtc::WaitUntil( - [&] { return sink.Check(receiver.get(), SSE_READ); }, - ::testing::IsTrue()), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return sink.Check(receiver.get(), SSE_READ); }, + ::testing::IsTrue()), + IsRtcOk()); readable = true; recv_called = false; } @@ -939,10 +927,9 @@ void SocketTest::TcpInternal(const IPAddress& loopback, // Once all that we've sent has been received, expect to be able to send // again. if (!writable) { - ASSERT_THAT( - webrtc::WaitUntil([&] { return sink.Check(sender.get(), SSE_WRITE); }, + ASSERT_THAT(WaitUntil([&] { return sink.Check(sender.get(), SSE_WRITE); }, ::testing::IsTrue()), - webrtc::IsRtcOk()); + IsRtcOk()); writable = true; send_called = false; } @@ -955,9 +942,9 @@ void SocketTest::TcpInternal(const IPAddress& loopback, // Close down. sender->Close(); - EXPECT_THAT(webrtc::WaitUntil([&] { return receiver->GetState(); }, - ::testing::Eq(Socket::CS_CLOSED)), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return receiver->GetState(); }, + ::testing::Eq(Socket::CS_CLOSED)), + IsRtcOk()); EXPECT_TRUE(sink.Check(receiver.get(), SSE_CLOSE)); receiver->Close(); } @@ -982,27 +969,25 @@ void SocketTest::SingleFlowControlCallbackInternal(const IPAddress& loopback) { EXPECT_EQ(0, client->Connect(server->GetLocalAddress())); // Accept connection. - EXPECT_THAT( - webrtc::WaitUntil([&] { return (sink.Check(server.get(), SSE_READ)); }, + EXPECT_THAT(WaitUntil([&] { return (sink.Check(server.get(), SSE_READ)); }, ::testing::IsTrue()), - webrtc::IsRtcOk()); + IsRtcOk()); std::unique_ptr accepted(server->Accept(&accept_addr)); ASSERT_TRUE(accepted); sink.Monitor(accepted.get()); // Both sides are now connected. - EXPECT_THAT(webrtc::WaitUntil([&] { return client->GetState(); }, - ::testing::Eq(Socket::CS_CONNECTED)), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return client->GetState(); }, + ::testing::Eq(Socket::CS_CONNECTED)), + IsRtcOk()); EXPECT_TRUE(sink.Check(client.get(), SSE_OPEN)); EXPECT_EQ(client->GetRemoteAddress(), accepted->GetLocalAddress()); EXPECT_EQ(accepted->GetRemoteAddress(), client->GetLocalAddress()); // Expect a writable callback from the connect. - EXPECT_THAT( - webrtc::WaitUntil([&] { return sink.Check(accepted.get(), SSE_WRITE); }, + EXPECT_THAT(WaitUntil([&] { return sink.Check(accepted.get(), SSE_WRITE); }, ::testing::IsTrue()), - webrtc::IsRtcOk()); + IsRtcOk()); // Fill the socket buffer. char buf[1024 * 16] = {0}; @@ -1012,10 +997,9 @@ void SocketTest::SingleFlowControlCallbackInternal(const IPAddress& loopback) { EXPECT_TRUE(accepted->IsBlocking()); // Wait until data is available. - EXPECT_THAT( - webrtc::WaitUntil([&] { return sink.Check(client.get(), SSE_READ); }, + EXPECT_THAT(WaitUntil([&] { return sink.Check(client.get(), SSE_READ); }, ::testing::IsTrue()), - webrtc::IsRtcOk()); + IsRtcOk()); // Pull data. for (int i = 0; i < sends; ++i) { @@ -1023,10 +1007,9 @@ void SocketTest::SingleFlowControlCallbackInternal(const IPAddress& loopback) { } // Expect at least one additional writable callback. - EXPECT_THAT( - webrtc::WaitUntil([&] { return sink.Check(accepted.get(), SSE_WRITE); }, + EXPECT_THAT(WaitUntil([&] { return sink.Check(accepted.get(), SSE_WRITE); }, ::testing::IsTrue()), - webrtc::IsRtcOk()); + IsRtcOk()); // Adding data in response to the writeable callback shouldn't cause infinite // callbacks. @@ -1046,7 +1029,7 @@ void SocketTest::SingleFlowControlCallbackInternal(const IPAddress& loopback) { } void SocketTest::UdpInternal(const IPAddress& loopback) { - SocketAddress empty = webrtc::EmptySocketAddressWithFamily(loopback.family()); + SocketAddress empty = EmptySocketAddressWithFamily(loopback.family()); // Test basic bind and connect behavior. Socket* socket = socket_factory_->CreateSocket(loopback.family(), SOCK_DGRAM); EXPECT_EQ(Socket::CS_CLOSED, socket->GetState()); @@ -1092,7 +1075,7 @@ void SocketTest::UdpInternal(const IPAddress& loopback) { } void SocketTest::UdpReadyToSend(const IPAddress& loopback) { - SocketAddress empty = webrtc::EmptySocketAddressWithFamily(loopback.family()); + SocketAddress empty = EmptySocketAddressWithFamily(loopback.family()); // RFC 5737 - The blocks 192.0.2.0/24 (TEST-NET-1) ... are provided for use in // documentation. // RFC 3849 - 2001:DB8::/32 as a documentation-only prefix. @@ -1133,9 +1116,9 @@ void SocketTest::UdpReadyToSend(const IPAddress& loopback) { } EXPECT_EQ(expected_error, error); EXPECT_FALSE(client->ready_to_send()); - EXPECT_THAT(webrtc::WaitUntil([&] { return client->ready_to_send(); }, - ::testing::IsTrue()), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return client->ready_to_send(); }, ::testing::IsTrue()), + IsRtcOk()); RTC_LOG(LS_INFO) << "Got SignalReadyToSend"; } @@ -1244,10 +1227,9 @@ void SocketTest::SocketRecvTimestamp(const IPAddress& loopback) { socket->SendTo("foo", 3, address); // Wait until data is available. - EXPECT_THAT( - webrtc::WaitUntil([&] { return sink.Check(socket.get(), SSE_READ); }, + EXPECT_THAT(WaitUntil([&] { return sink.Check(socket.get(), SSE_READ); }, ::testing::IsTrue()), - webrtc::IsRtcOk()); + IsRtcOk()); Buffer buffer; Socket::ReceiveBuffer receive_buffer_1(buffer); ASSERT_GT(socket->RecvFrom(receive_buffer_1), 0); @@ -1258,10 +1240,9 @@ void SocketTest::SocketRecvTimestamp(const IPAddress& loopback) { int64_t send_time_2 = TimeMicros(); socket->SendTo("bar", 3, address); // Wait until data is available. - EXPECT_THAT( - webrtc::WaitUntil([&] { return sink.Check(socket.get(), SSE_READ); }, + EXPECT_THAT(WaitUntil([&] { return sink.Check(socket.get(), SSE_READ); }, ::testing::IsTrue()), - webrtc::IsRtcOk()); + IsRtcOk()); Socket::ReceiveBuffer receive_buffer_2(buffer); ASSERT_GT(socket->RecvFrom(receive_buffer_2), 0); @@ -1274,7 +1255,7 @@ void SocketTest::SocketRecvTimestamp(const IPAddress& loopback) { } void SocketTest::UdpSocketRecvTimestampUseRtcEpoch(const IPAddress& loopback) { - SocketAddress empty = webrtc::EmptySocketAddressWithFamily(loopback.family()); + SocketAddress empty = EmptySocketAddressWithFamily(loopback.family()); std::unique_ptr socket( socket_factory_->CreateSocket(loopback.family(), SOCK_DGRAM)); ASSERT_EQ(socket->Bind(SocketAddress(loopback, 0)), 0); @@ -1311,10 +1292,9 @@ void SocketTest::SocketSendRecvWithEcn(const IPAddress& loopback) { Socket::ReceiveBuffer receive_buffer(buffer); socket->SendTo("foo", 3, address); - EXPECT_THAT( - webrtc::WaitUntil([&] { return sink.Check(socket.get(), SSE_READ); }, + EXPECT_THAT(WaitUntil([&] { return sink.Check(socket.get(), SSE_READ); }, ::testing::IsTrue()), - webrtc::IsRtcOk()); + IsRtcOk()); ASSERT_GT(socket->RecvFrom(receive_buffer), 0); EXPECT_EQ(receive_buffer.ecn, EcnMarking::kNotEct); @@ -1322,28 +1302,25 @@ void SocketTest::SocketSendRecvWithEcn(const IPAddress& loopback) { socket->SetOption(Socket::OPT_RECV_ECN, 1); socket->SendTo("bar", 3, address); - EXPECT_THAT( - webrtc::WaitUntil([&] { return sink.Check(socket.get(), SSE_READ); }, + EXPECT_THAT(WaitUntil([&] { return sink.Check(socket.get(), SSE_READ); }, ::testing::IsTrue()), - webrtc::IsRtcOk()); + IsRtcOk()); ASSERT_GT(socket->RecvFrom(receive_buffer), 0); EXPECT_EQ(receive_buffer.ecn, EcnMarking::kEct1); socket->SetOption(Socket::OPT_SEND_ECN, 2); // Ect(0) socket->SendTo("bar", 3, address); - EXPECT_THAT( - webrtc::WaitUntil([&] { return sink.Check(socket.get(), SSE_READ); }, + EXPECT_THAT(WaitUntil([&] { return sink.Check(socket.get(), SSE_READ); }, ::testing::IsTrue()), - webrtc::IsRtcOk()); + IsRtcOk()); ASSERT_GT(socket->RecvFrom(receive_buffer), 0); EXPECT_EQ(receive_buffer.ecn, EcnMarking::kEct0); socket->SetOption(Socket::OPT_SEND_ECN, 3); // Ce socket->SendTo("bar", 3, address); - EXPECT_THAT( - webrtc::WaitUntil([&] { return sink.Check(socket.get(), SSE_READ); }, + EXPECT_THAT(WaitUntil([&] { return sink.Check(socket.get(), SSE_READ); }, ::testing::IsTrue()), - webrtc::IsRtcOk()); + IsRtcOk()); ASSERT_GT(socket->RecvFrom(receive_buffer), 0); EXPECT_EQ(receive_buffer.ecn, EcnMarking::kCe); } diff --git a/rtc_base/ssl_fingerprint.cc b/rtc_base/ssl_fingerprint.cc index daca55873d..ca15d11bda 100644 --- a/rtc_base/ssl_fingerprint.cc +++ b/rtc_base/ssl_fingerprint.cc @@ -61,7 +61,7 @@ SSLFingerprint* SSLFingerprint::CreateFromRfc4572( std::unique_ptr SSLFingerprint::CreateUniqueFromRfc4572( absl::string_view algorithm, absl::string_view fingerprint) { - if (algorithm.empty() || !webrtc::IsFips180DigestAlgorithm(algorithm)) + if (algorithm.empty() || !IsFips180DigestAlgorithm(algorithm)) return nullptr; if (fingerprint.empty()) diff --git a/rtc_base/ssl_identity.cc b/rtc_base/ssl_identity.cc index 9e16cb1ac9..35930f8fe1 100644 --- a/rtc_base/ssl_identity.cc +++ b/rtc_base/ssl_identity.cc @@ -108,7 +108,7 @@ int64_t ASN1TimeToSec(const unsigned char* s, size_t length, bool long_format) { if (bytes_left != 1) { return -1; } - return webrtc::TmToSeconds(tm); + return TmToSeconds(tm); } ////////////////////////////////////////////////////////////////////// diff --git a/rtc_base/ssl_stream_adapter.cc b/rtc_base/ssl_stream_adapter.cc index 02ee72948b..d02318bc68 100644 --- a/rtc_base/ssl_stream_adapter.cc +++ b/rtc_base/ssl_stream_adapter.cc @@ -86,7 +86,7 @@ bool IsGcmCryptoSuite(int crypto_suite) { std::unique_ptr SSLStreamAdapter::Create( std::unique_ptr stream, - absl::AnyInvocable handshake_error, + absl::AnyInvocable handshake_error, const FieldTrialsView* field_trials) { return std::make_unique( std::move(stream), std::move(handshake_error), field_trials); diff --git a/rtc_base/strings/string_builder_unittest.cc b/rtc_base/strings/string_builder_unittest.cc index 44c6c4b633..d04e0eb4d9 100644 --- a/rtc_base/strings/string_builder_unittest.cc +++ b/rtc_base/strings/string_builder_unittest.cc @@ -20,7 +20,7 @@ namespace webrtc { TEST(SimpleStringBuilder, Limit) { char sb_buf[10]; - webrtc::SimpleStringBuilder sb(sb_buf); + SimpleStringBuilder sb(sb_buf); EXPECT_EQ(0u, strlen(sb.str())); // Test that for a SSB with a buffer size of 10, that we can write 9 chars @@ -31,7 +31,7 @@ TEST(SimpleStringBuilder, Limit) { TEST(SimpleStringBuilder, NumbersAndChars) { char sb_buf[100]; - webrtc::SimpleStringBuilder sb(sb_buf); + SimpleStringBuilder sb(sb_buf); sb << 1 << ':' << 2.1 << ":" << 2.2f << ':' << 78187493520ll << ':' << 78187493520ul; EXPECT_EQ(0, strcmp(sb.str(), "1:2.1:2.2:78187493520:78187493520")); @@ -39,7 +39,7 @@ TEST(SimpleStringBuilder, NumbersAndChars) { TEST(SimpleStringBuilder, Format) { char sb_buf[100]; - webrtc::SimpleStringBuilder sb(sb_buf); + SimpleStringBuilder sb(sb_buf); sb << "Here we go - "; sb.AppendFormat("This is a hex formatted value: 0x%08llx", 3735928559ULL); EXPECT_EQ(0, @@ -49,7 +49,7 @@ TEST(SimpleStringBuilder, Format) { TEST(SimpleStringBuilder, StdString) { char sb_buf[100]; - webrtc::SimpleStringBuilder sb(sb_buf); + SimpleStringBuilder sb(sb_buf); std::string str = "does this work?"; sb << str; EXPECT_EQ(str, sb.str()); @@ -61,7 +61,7 @@ TEST(SimpleStringBuilder, StdString) { TEST(SimpleStringBuilderDeathTest, BufferOverrunConstCharP) { char sb_buf[4]; - webrtc::SimpleStringBuilder sb(sb_buf); + SimpleStringBuilder sb(sb_buf); const char* const msg = "This is just too much"; #if RTC_DCHECK_IS_ON EXPECT_DEATH(sb << msg, ""); @@ -73,7 +73,7 @@ TEST(SimpleStringBuilderDeathTest, BufferOverrunConstCharP) { TEST(SimpleStringBuilderDeathTest, BufferOverrunStdString) { char sb_buf[4]; - webrtc::SimpleStringBuilder sb(sb_buf); + SimpleStringBuilder sb(sb_buf); sb << 12; const std::string msg = "Aw, come on!"; #if RTC_DCHECK_IS_ON @@ -86,7 +86,7 @@ TEST(SimpleStringBuilderDeathTest, BufferOverrunStdString) { TEST(SimpleStringBuilderDeathTest, BufferOverrunInt) { char sb_buf[4]; - webrtc::SimpleStringBuilder sb(sb_buf); + SimpleStringBuilder sb(sb_buf); constexpr int num = -12345; #if RTC_DCHECK_IS_ON EXPECT_DEATH(sb << num, ""); @@ -102,7 +102,7 @@ TEST(SimpleStringBuilderDeathTest, BufferOverrunInt) { TEST(SimpleStringBuilderDeathTest, BufferOverrunDouble) { char sb_buf[5]; - webrtc::SimpleStringBuilder sb(sb_buf); + SimpleStringBuilder sb(sb_buf); constexpr double num = 123.456; #if RTC_DCHECK_IS_ON EXPECT_DEATH(sb << num, ""); @@ -115,7 +115,7 @@ TEST(SimpleStringBuilderDeathTest, BufferOverrunDouble) { TEST(SimpleStringBuilderDeathTest, BufferOverrunConstCharPAlreadyFull) { char sb_buf[4]; - webrtc::SimpleStringBuilder sb(sb_buf); + SimpleStringBuilder sb(sb_buf); sb << 123; const char* const msg = "This is just too much"; #if RTC_DCHECK_IS_ON @@ -128,7 +128,7 @@ TEST(SimpleStringBuilderDeathTest, BufferOverrunConstCharPAlreadyFull) { TEST(SimpleStringBuilderDeathTest, BufferOverrunIntAlreadyFull) { char sb_buf[4]; - webrtc::SimpleStringBuilder sb(sb_buf); + SimpleStringBuilder sb(sb_buf); sb << "xyz"; constexpr int num = -12345; #if RTC_DCHECK_IS_ON @@ -145,7 +145,7 @@ TEST(SimpleStringBuilderDeathTest, BufferOverrunIntAlreadyFull) { // StringBuilder. TEST(StringBuilder, Limit) { - webrtc::StringBuilder sb; + StringBuilder sb; EXPECT_EQ(0u, sb.str().size()); sb << "012345678"; @@ -153,7 +153,7 @@ TEST(StringBuilder, Limit) { } TEST(StringBuilder, NumbersAndChars) { - webrtc::StringBuilder sb; + StringBuilder sb; sb << 1 << ":" << 2.1 << ":" << 2.2f << ":" << 78187493520ll << ":" << 78187493520ul; EXPECT_THAT(sb.str(), @@ -161,21 +161,21 @@ TEST(StringBuilder, NumbersAndChars) { } TEST(StringBuilder, Format) { - webrtc::StringBuilder sb; + StringBuilder sb; sb << "Here we go - "; sb.AppendFormat("This is a hex formatted value: 0x%08llx", 3735928559ULL); EXPECT_EQ(sb.str(), "Here we go - This is a hex formatted value: 0xdeadbeef"); } TEST(StringBuilder, StdString) { - webrtc::StringBuilder sb; + StringBuilder sb; std::string str = "does this work?"; sb << str; EXPECT_EQ(str, sb.str()); } TEST(StringBuilder, Release) { - webrtc::StringBuilder sb; + StringBuilder sb; std::string str = "This string has to be of a moderate length, or we might " "run into problems with small object optimizations."; @@ -190,7 +190,7 @@ TEST(StringBuilder, Release) { } TEST(StringBuilder, Reset) { - webrtc::StringBuilder sb("abc"); + StringBuilder sb("abc"); sb << "def"; EXPECT_EQ("abcdef", sb.str()); sb.Clear(); diff --git a/rtc_base/strong_alias_unittest.cc b/rtc_base/strong_alias_unittest.cc index 711c805e89..c663cad837 100644 --- a/rtc_base/strong_alias_unittest.cc +++ b/rtc_base/strong_alias_unittest.cc @@ -282,7 +282,7 @@ TEST(StrongAliasTest, CanWrapComplexStructures) { TYPED_TEST(StrongAliasTest, CanBeKeysInFlatMap) { using FooAlias = StrongAlias; - webrtc::flat_map map; + flat_map map; FooAlias k1(GetExampleValue(0)); FooAlias k2(GetExampleValue(1)); diff --git a/rtc_base/synchronization/mutex_unittest.cc b/rtc_base/synchronization/mutex_unittest.cc index 06938a21f2..cafd24cf7e 100644 --- a/rtc_base/synchronization/mutex_unittest.cc +++ b/rtc_base/synchronization/mutex_unittest.cc @@ -30,8 +30,6 @@ namespace webrtc { namespace { -using ::webrtc::Event; -using ::webrtc::Thread; constexpr int kNumThreads = 16; diff --git a/rtc_base/synchronization/sequence_checker_internal.cc b/rtc_base/synchronization/sequence_checker_internal.cc index 007acacd14..3b0a0b123d 100644 --- a/rtc_base/synchronization/sequence_checker_internal.cc +++ b/rtc_base/synchronization/sequence_checker_internal.cc @@ -56,7 +56,7 @@ void SequenceCheckerImpl::Detach() { #if RTC_DCHECK_IS_ON std::string SequenceCheckerImpl::ExpectationToString() const { const TaskQueueBase* const current_queue = TaskQueueBase::Current(); - const webrtc::PlatformThreadRef current_thread = webrtc::CurrentThreadRef(); + const PlatformThreadRef current_thread = CurrentThreadRef(); MutexLock scoped_lock(&lock_); if (!attached_) return "Checker currently not attached."; @@ -68,7 +68,7 @@ std::string SequenceCheckerImpl::ExpectationToString() const { // # Actual: TQ: 0x7fa8f0604190 SysQ: 0x7fa8f0604a30 Thread: 0x700006f1a000 // TaskQueue doesn't match - webrtc::StringBuilder message; + StringBuilder message; message.AppendFormat( "# Expected: TQ: %p Thread: %p\n" "# Actual: TQ: %p Thread: %p\n", @@ -77,7 +77,7 @@ std::string SequenceCheckerImpl::ExpectationToString() const { if ((valid_queue_ || current_queue) && valid_queue_ != current_queue) { message << "TaskQueue doesn't match\n"; - } else if (!webrtc::IsThreadRefEqual(valid_thread_, current_thread)) { + } else if (!IsThreadRefEqual(valid_thread_, current_thread)) { message << "Threads don't match\n"; } diff --git a/rtc_base/task_queue_stdlib_unittest.cc b/rtc_base/task_queue_stdlib_unittest.cc index c090b5e497..ca27f82219 100644 --- a/rtc_base/task_queue_stdlib_unittest.cc +++ b/rtc_base/task_queue_stdlib_unittest.cc @@ -25,7 +25,7 @@ namespace webrtc { namespace { std::unique_ptr CreateTaskQueueFactory( - const webrtc::FieldTrialsView*) { + const FieldTrialsView*) { return CreateTaskQueueStdlibFactory(); } diff --git a/rtc_base/test_client.cc b/rtc_base/test_client.cc index 3798c595d8..a204d1c80e 100644 --- a/rtc_base/test_client.cc +++ b/rtc_base/test_client.cc @@ -138,7 +138,7 @@ void TestClient::AdvanceTime(int ms) { // advance time. Otherwise, ProcessMessages will work. if (fake_clock_) { for (int64_t start = TimeMillis(); TimeMillis() < start + ms;) { - fake_clock_->AdvanceTime(webrtc::TimeDelta::Millis(1)); + fake_clock_->AdvanceTime(TimeDelta::Millis(1)); }; } else { Thread::Current()->ProcessMessages(1); diff --git a/rtc_base/test_client_unittest.cc b/rtc_base/test_client_unittest.cc index c585a8e532..4a87ae36d4 100644 --- a/rtc_base/test_client_unittest.cc +++ b/rtc_base/test_client_unittest.cc @@ -57,7 +57,7 @@ void TestUdpInternal(const SocketAddress& loopback) { void TestTcpInternal(const SocketAddress& loopback) { PhysicalSocketServer socket_server; AutoSocketServerThread main_thread(&socket_server); - webrtc::TestEchoServer server(&main_thread, loopback); + TestEchoServer server(&main_thread, loopback); Socket* socket = socket_server.CreateSocket(loopback.family(), SOCK_STREAM); std::unique_ptr tcp_socket = absl::WrapUnique( diff --git a/rtc_base/thread.cc b/rtc_base/thread.cc index 2491335c96..6599615233 100644 --- a/rtc_base/thread.cc +++ b/rtc_base/thread.cc @@ -88,8 +88,6 @@ class ScopedAutoReleasePool { namespace webrtc { -using ::webrtc::MutexLock; -using ::webrtc::TimeDelta; ThreadManager* ThreadManager::Instance() { static ThreadManager* const thread_manager = new ThreadManager(); @@ -499,7 +497,7 @@ void Thread::PostDelayedTaskImpl(absl::AnyInvocable task, // Add to the priority queue. Gets sorted soonest first. // Signal for the multiplexer to return. - int64_t delay_ms = delay.RoundUpTo(webrtc::TimeDelta::Millis(1)).ms(); + int64_t delay_ms = delay.RoundUpTo(TimeDelta::Millis(1)).ms(); int64_t run_time_ms = TimeAfter(delay_ms); { MutexLock lock(&mutex_); @@ -523,7 +521,7 @@ int Thread::GetDelay() { return 0; if (!delayed_messages_.empty()) { - int delay = webrtc::TimeUntil(delayed_messages_.top().run_time_ms); + int delay = TimeUntil(delayed_messages_.top().run_time_ms); if (delay < 0) delay = 0; return delay; @@ -535,10 +533,10 @@ int Thread::GetDelay() { void Thread::Dispatch(absl::AnyInvocable task) { TRACE_EVENT0("webrtc", "Thread::Dispatch"); RTC_DCHECK_RUN_ON(this); - int64_t start_time = webrtc::TimeMillis(); + int64_t start_time = TimeMillis(); std::move(task)(); - int64_t end_time = webrtc::TimeMillis(); - int64_t diff = webrtc::TimeDiff(end_time, start_time); + int64_t end_time = TimeMillis(); + int64_t diff = TimeDiff(end_time, start_time); if (diff >= dispatch_warning_ms_) { RTC_LOG(LS_INFO) << "Message to " << name() << " took " << diff << "ms to dispatch."; @@ -850,9 +848,9 @@ bool Thread::ProcessMessages(int cmsLoop) { // Using ProcessMessages with a custom clock for testing and a time greater // than 0 doesn't work, since it's not guaranteed to advance the custom // clock's time, and may get stuck in an infinite loop. - RTC_DCHECK(webrtc::GetClockForTesting() == nullptr || cmsLoop == 0 || + RTC_DCHECK(GetClockForTesting() == nullptr || cmsLoop == 0 || cmsLoop == kForever); - int64_t msEnd = (kForever == cmsLoop) ? 0 : webrtc::TimeAfter(cmsLoop); + int64_t msEnd = (kForever == cmsLoop) ? 0 : TimeAfter(cmsLoop); int cmsNext = cmsLoop; while (true) { @@ -865,7 +863,7 @@ bool Thread::ProcessMessages(int cmsLoop) { Dispatch(std::move(task)); if (cmsLoop != kForever) { - cmsNext = static_cast(webrtc::TimeUntil(msEnd)); + cmsNext = static_cast(TimeUntil(msEnd)); if (cmsNext < 0) return true; } diff --git a/rtc_base/thread_unittest.cc b/rtc_base/thread_unittest.cc index 9c417fe75a..24e7106cf5 100644 --- a/rtc_base/thread_unittest.cc +++ b/rtc_base/thread_unittest.cc @@ -53,7 +53,6 @@ namespace webrtc { namespace { using ::testing::ElementsAre; -using ::webrtc::TimeDelta; // Generates a sequence of numbers (collaboratively). class TestGenerator { @@ -206,7 +205,7 @@ TEST(ThreadTest, CountBlockingCalls) { // (thread_unittest.cc:262): Blocking TestBody: total=2 (actual=1, could=1) RTC_LOG_THREAD_BLOCK_COUNT(); #if RTC_DCHECK_IS_ON - webrtc::Thread::ScopedCountBlockingCalls blocked_calls( + Thread::ScopedCountBlockingCalls blocked_calls( [&](uint32_t actual_block, uint32_t could_block) { EXPECT_EQ(1u, actual_block); EXPECT_EQ(1u, could_block); @@ -242,10 +241,10 @@ TEST(ThreadTest, CountBlockingCalls) { #if RTC_DCHECK_IS_ON TEST(ThreadTest, CountBlockingCallsOneCallback) { - webrtc::AutoThread current; + AutoThread current; bool was_called_back = false; { - webrtc::Thread::ScopedCountBlockingCalls blocked_calls( + Thread::ScopedCountBlockingCalls blocked_calls( [&](uint32_t actual_block, uint32_t could_block) { was_called_back = true; }); @@ -255,10 +254,10 @@ TEST(ThreadTest, CountBlockingCallsOneCallback) { } TEST(ThreadTest, CountBlockingCallsSkipCallback) { - webrtc::AutoThread current; + AutoThread current; bool was_called_back = false; { - webrtc::Thread::ScopedCountBlockingCalls blocked_calls( + Thread::ScopedCountBlockingCalls blocked_calls( [&](uint32_t actual_block, uint32_t could_block) { was_called_back = true; }); @@ -309,7 +308,7 @@ TEST(ThreadTest, Wrap) { #if (!defined(NDEBUG) || RTC_DCHECK_IS_ON) TEST(ThreadTest, InvokeToThreadAllowedReturnsTrueWithoutPolicies) { - webrtc::AutoThread main_thread; + AutoThread main_thread; // Create and start the thread. auto thread1 = Thread::CreateWithSocketServer(); auto thread2 = Thread::CreateWithSocketServer(); @@ -320,7 +319,7 @@ TEST(ThreadTest, InvokeToThreadAllowedReturnsTrueWithoutPolicies) { } TEST(ThreadTest, InvokeAllowedWhenThreadsAdded) { - webrtc::AutoThread main_thread; + AutoThread main_thread; // Create and start the thread. auto thread1 = Thread::CreateWithSocketServer(); auto thread2 = Thread::CreateWithSocketServer(); @@ -339,7 +338,7 @@ TEST(ThreadTest, InvokeAllowedWhenThreadsAdded) { } TEST(ThreadTest, InvokesDisallowedWhenDisallowAllInvokes) { - webrtc::AutoThread main_thread; + AutoThread main_thread; // Create and start the thread. auto thread1 = Thread::CreateWithSocketServer(); auto thread2 = Thread::CreateWithSocketServer(); @@ -469,9 +468,9 @@ TEST(ThreadTest, ThreeThreadsBlockingCall) { SetAndInvokeSet(&async_invoked, thread2, out); }); - EXPECT_THAT(webrtc::WaitUntil([&] { return async_invoked.Get(); }, - ::testing::IsTrue()), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return async_invoked.Get(); }, ::testing::IsTrue()), + IsRtcOk()); } }; @@ -486,9 +485,9 @@ TEST(ThreadTest, ThreeThreadsBlockingCall) { }); EXPECT_FALSE(thread_a_called.Get()); - EXPECT_THAT(webrtc::WaitUntil([&] { return thread_a_called.Get(); }, - ::testing::IsTrue()), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return thread_a_called.Get(); }, ::testing::IsTrue()), + IsRtcOk()); } static void DelayedPostsWithIdenticalTimesAreProcessedInFifoOrder( @@ -858,7 +857,6 @@ std::unique_ptr CreateDefaultThreadFactory( return std::make_unique(); } -using ::webrtc::TaskQueueTest; INSTANTIATE_TEST_SUITE_P(RtcThread, TaskQueueTest, diff --git a/rtc_base/unique_id_generator.cc b/rtc_base/unique_id_generator.cc index f817decb99..6bd47d32ed 100644 --- a/rtc_base/unique_id_generator.cc +++ b/rtc_base/unique_id_generator.cc @@ -60,7 +60,7 @@ bool UniqueStringGenerator::AddKnownId(absl::string_view value) { // TODO(webrtc:13579): remove string copy here once absl::string_view version // of StringToNumber is available. std::optional int_value = - webrtc::StringToNumber(std::string(value)); + StringToNumber(std::string(value)); // The underlying generator works for uint32_t values, so if the provided // value is not a uint32_t it will never be generated anyway. if (int_value.has_value()) { diff --git a/rtc_base/unique_id_generator_unittest.cc b/rtc_base/unique_id_generator_unittest.cc index 85ed5d4715..661ba58bdf 100644 --- a/rtc_base/unique_id_generator_unittest.cc +++ b/rtc_base/unique_id_generator_unittest.cc @@ -173,7 +173,7 @@ TYPED_TEST(UniqueIdGeneratorTest, // Tests that it's OK to construct the generator in one execution environment // (thread/task queue) but use it in another. TEST(UniqueNumberGenerator, UsedOnSecondaryThread) { - const auto* current_tq = webrtc::TaskQueueBase::Current(); + const auto* current_tq = TaskQueueBase::Current(); // Construct the generator before `fake_task_queue` to ensure that it is // constructed in a different execution environment than what // `fake_task_queue` will represent. @@ -181,7 +181,7 @@ TEST(UniqueNumberGenerator, UsedOnSecondaryThread) { FakeTaskQueue fake_task_queue; // Sanity check to make sure we're in a different runtime environment. - ASSERT_NE(current_tq, webrtc::TaskQueueBase::Current()); + ASSERT_NE(current_tq, TaskQueueBase::Current()); // Generating an id should be fine in this context. generator.Generate(); diff --git a/rtc_base/virtual_socket_server.cc b/rtc_base/virtual_socket_server.cc index 9fad9dfe27..20ae7890c4 100644 --- a/rtc_base/virtual_socket_server.cc +++ b/rtc_base/virtual_socket_server.cc @@ -31,9 +31,6 @@ namespace webrtc { -using ::webrtc::MutexLock; -using ::webrtc::TaskQueueBase; -using ::webrtc::TimeDelta; #if defined(WEBRTC_WIN) const in_addr kInitialNextIPv4 = {{{0x01, 0, 0, 0}}}; @@ -550,7 +547,7 @@ int VirtualSocket::SendUdp(const void* pv, // If we have not been assigned a local port, then get one. if (local_addr_.IsNil()) { local_addr_ = server_->AssignBindAddress( - webrtc::EmptySocketAddressWithFamily(addr.ipaddr().family())); + EmptySocketAddressWithFamily(addr.ipaddr().family())); int result = server_->Bind(this, local_addr_); if (result != 0) { local_addr_.Clear(); @@ -681,8 +678,7 @@ VirtualSocketServer::~VirtualSocketServer() { IPAddress VirtualSocketServer::GetNextIP(int family) { if (family == AF_INET) { IPAddress next_ip(next_ipv4_); - next_ipv4_.s_addr = - webrtc::HostToNetwork32(webrtc::NetworkToHost32(next_ipv4_.s_addr) + 1); + next_ipv4_.s_addr = HostToNetwork32(NetworkToHost32(next_ipv4_.s_addr) + 1); return next_ip; } else if (family == AF_INET6) { IPAddress next_ip(next_ipv6_); @@ -791,7 +787,7 @@ int VirtualSocketServer::Bind(VirtualSocket* socket, const SocketAddress& addr) { RTC_DCHECK(nullptr != socket); // Address must be completely specified at this point - RTC_DCHECK(!webrtc::IPIsUnspec(addr.ipaddr())); + RTC_DCHECK(!IPIsUnspec(addr.ipaddr())); RTC_DCHECK(addr.port() != 0); // Normalize the address (turns v6-mapped addresses into v4-addresses). @@ -803,7 +799,7 @@ int VirtualSocketServer::Bind(VirtualSocket* socket, SocketAddress VirtualSocketServer::AssignBindAddress( const SocketAddress& app_addr) { - RTC_DCHECK(!webrtc::IPIsUnspec(app_addr.ipaddr())); + RTC_DCHECK(!IPIsUnspec(app_addr.ipaddr())); // Normalize the IP. SocketAddress addr; @@ -840,12 +836,12 @@ VirtualSocket* VirtualSocketServer::LookupBinding(const SocketAddress& addr) { } IPAddress default_ip = GetDefaultSourceAddress(addr.ipaddr().family()); - if (!webrtc::IPIsUnspec(default_ip) && addr.ipaddr() == default_ip) { + if (!IPIsUnspec(default_ip) && addr.ipaddr() == default_ip) { // If we can't find a binding for the packet which is sent to the interface // corresponding to the default route, it should match a binding with the // correct port to the any address. SocketAddress sock_addr = - webrtc::EmptySocketAddressWithFamily(addr.ipaddr().family()); + EmptySocketAddressWithFamily(addr.ipaddr().family()); sock_addr.SetPort(addr.port()); return LookupBinding(sock_addr); } @@ -1096,7 +1092,7 @@ void VirtualSocketServer::AddPacketToNetwork(VirtualSocket* sender, // route. SocketAddress sender_addr = sender->GetLocalAddress(); IPAddress default_ip = GetDefaultSourceAddress(sender_addr.ipaddr().family()); - if (sender_addr.IsAnyIP() && !webrtc::IPIsUnspec(default_ip)) { + if (sender_addr.IsAnyIP() && !IPIsUnspec(default_ip)) { sender_addr.SetIP(default_ip); } @@ -1285,15 +1281,13 @@ bool VirtualSocketServer::CanInteractWith(VirtualSocket* local, // If ip1 is IPv4 and ip2 is :: and ip2 is not IPV6_V6ONLY. int remote_v6_only = 0; remote->GetOption(Socket::OPT_IPV6_V6ONLY, &remote_v6_only); - if (local_ip.family() == AF_INET && !remote_v6_only && - webrtc::IPIsAny(remote_ip)) { + if (local_ip.family() == AF_INET && !remote_v6_only && IPIsAny(remote_ip)) { return true; } // Same check, backwards. int local_v6_only = 0; local->GetOption(Socket::OPT_IPV6_V6ONLY, &local_v6_only); - if (remote_ip.family() == AF_INET && !local_v6_only && - webrtc::IPIsAny(local_ip)) { + if (remote_ip.family() == AF_INET && !local_v6_only && IPIsAny(local_ip)) { return true; } @@ -1319,7 +1313,7 @@ IPAddress VirtualSocketServer::GetDefaultSourceAddress(int family) { return IPAddress(); } void VirtualSocketServer::SetDefaultSourceAddress(const IPAddress& from_addr) { - RTC_DCHECK(!webrtc::IPIsAny(from_addr)); + RTC_DCHECK(!IPIsAny(from_addr)); if (from_addr.family() == AF_INET) { default_source_address_v4_ = from_addr; } else if (from_addr.family() == AF_INET6) { diff --git a/rtc_base/virtual_socket_unittest.cc b/rtc_base/virtual_socket_unittest.cc index c05f8d87c7..c6237303a9 100644 --- a/rtc_base/virtual_socket_unittest.cc +++ b/rtc_base/virtual_socket_unittest.cc @@ -42,24 +42,24 @@ namespace webrtc { namespace { -using ::webrtc::testing::SSE_CLOSE; -using ::webrtc::testing::SSE_ERROR; -using ::webrtc::testing::SSE_OPEN; -using ::webrtc::testing::SSE_READ; -using ::webrtc::testing::SSE_WRITE; -using ::webrtc::testing::StreamSink; +using testing::SSE_CLOSE; +using testing::SSE_ERROR; +using testing::SSE_OPEN; +using testing::SSE_READ; +using testing::SSE_WRITE; +using testing::StreamSink; // Sends at a constant rate but with random packet sizes. struct Sender { - Sender(webrtc::Thread* th, webrtc::Socket* s, uint32_t rt) + Sender(Thread* th, Socket* s, uint32_t rt) : thread(th), - socket(std::make_unique(s)), + socket(std::make_unique(s)), rate(rt), count(0) { - last_send = webrtc::TimeMillis(); + last_send = TimeMillis(); periodic = RepeatingTaskHandle::DelayedStart(thread, NextDelay(), [this] { - int64_t cur_time = webrtc::TimeMillis(); + int64_t cur_time = TimeMillis(); int64_t delay = cur_time - last_send; uint32_t size = std::clamp(rate * delay / 1000, sizeof(uint32_t), 4096); @@ -77,8 +77,8 @@ struct Sender { return TimeDelta::Seconds(1) * size / rate; } - webrtc::Thread* thread; - std::unique_ptr socket; + Thread* thread; + std::unique_ptr socket; AsyncSocketPacketOptions options; RepeatingTaskHandle periodic; uint32_t rate; // bytes per second @@ -88,9 +88,9 @@ struct Sender { }; struct Receiver : public sigslot::has_slots<> { - Receiver(webrtc::Thread* th, webrtc::Socket* s, uint32_t bw) + Receiver(Thread* th, Socket* s, uint32_t bw) : thread(th), - socket(std::make_unique(s)), + socket(std::make_unique(s)), bandwidth(bw), count(0), sec_count(0), @@ -115,8 +115,7 @@ struct Receiver : public sigslot::has_slots<> { ~Receiver() override { periodic.Stop(); } - void OnReadPacket(webrtc::AsyncPacketSocket* s, - const ReceivedIpPacket& packet) { + void OnReadPacket(AsyncPacketSocket* s, const ReceivedIpPacket& packet) { ASSERT_EQ(socket.get(), s); ASSERT_GE(packet.payload().size(), 4U); @@ -125,15 +124,15 @@ struct Receiver : public sigslot::has_slots<> { uint32_t send_time = *reinterpret_cast(packet.payload().data()); - uint32_t recv_time = webrtc::TimeMillis(); + uint32_t recv_time = TimeMillis(); uint32_t delay = recv_time - send_time; sum += delay; sum_sq += delay * delay; samples += 1; } - webrtc::Thread* thread; - std::unique_ptr socket; + Thread* thread; + std::unique_ptr socket; uint32_t bandwidth; RepeatingTaskHandle periodic; size_t count; @@ -149,14 +148,14 @@ class VirtualSocketServerTest : public ::testing::Test { VirtualSocketServerTest() : ss_(&fake_clock_), thread_(&ss_), - kIPv4AnyAddress(webrtc::IPAddress(INADDR_ANY), 0), - kIPv6AnyAddress(webrtc::IPAddress(in6addr_any), 0) {} + kIPv4AnyAddress(IPAddress(INADDR_ANY), 0), + kIPv6AnyAddress(IPAddress(in6addr_any), 0) {} - void CheckPortIncrementalization(const webrtc::SocketAddress& post, - const webrtc::SocketAddress& pre) { + void CheckPortIncrementalization(const SocketAddress& post, + const SocketAddress& pre) { EXPECT_EQ(post.port(), pre.port() + 1); - webrtc::IPAddress post_ip = post.ipaddr(); - webrtc::IPAddress pre_ip = pre.ipaddr(); + IPAddress post_ip = post.ipaddr(); + IPAddress pre_ip = pre.ipaddr(); EXPECT_EQ(pre_ip.family(), post_ip.family()); if (post_ip.family() == AF_INET) { in_addr pre_ipv4 = pre_ip.ipv4_address(); @@ -174,79 +173,72 @@ class VirtualSocketServerTest : public ::testing::Test { // Test a client can bind to the any address, and all sent packets will have // the default source address. Also, it can receive packets sent to the // default address. - void TestDefaultSourceAddress(const webrtc::IPAddress& default_address) { + void TestDefaultSourceAddress(const IPAddress& default_address) { ss_.SetDefaultSourceAddress(default_address); // Create client1 bound to the any address. - webrtc::Socket* socket = - ss_.CreateSocket(default_address.family(), SOCK_DGRAM); - socket->Bind( - webrtc::EmptySocketAddressWithFamily(default_address.family())); - webrtc::SocketAddress client1_any_addr = socket->GetLocalAddress(); + Socket* socket = ss_.CreateSocket(default_address.family(), SOCK_DGRAM); + socket->Bind(EmptySocketAddressWithFamily(default_address.family())); + SocketAddress client1_any_addr = socket->GetLocalAddress(); EXPECT_TRUE(client1_any_addr.IsAnyIP()); - auto client1 = std::make_unique( - std::make_unique(socket), &fake_clock_); + auto client1 = std::make_unique( + std::make_unique(socket), &fake_clock_); // Create client2 bound to the address route. - webrtc::Socket* socket2 = - ss_.CreateSocket(default_address.family(), SOCK_DGRAM); - socket2->Bind(webrtc::SocketAddress(default_address, 0)); - webrtc::SocketAddress client2_addr = socket2->GetLocalAddress(); + Socket* socket2 = ss_.CreateSocket(default_address.family(), SOCK_DGRAM); + socket2->Bind(SocketAddress(default_address, 0)); + SocketAddress client2_addr = socket2->GetLocalAddress(); EXPECT_FALSE(client2_addr.IsAnyIP()); - auto client2 = std::make_unique( - std::make_unique(socket2), &fake_clock_); + auto client2 = std::make_unique( + std::make_unique(socket2), &fake_clock_); // Client1 sends to client2, client2 should see the default address as // client1's address. - webrtc::SocketAddress client1_addr; + SocketAddress client1_addr; EXPECT_EQ(6, client1->SendTo("bizbaz", 6, client2_addr)); EXPECT_TRUE(client2->CheckNextPacket("bizbaz", 6, &client1_addr)); EXPECT_EQ(client1_addr, - webrtc::SocketAddress(default_address, client1_any_addr.port())); + SocketAddress(default_address, client1_any_addr.port())); // Client2 can send back to client1's default address. EXPECT_EQ(3, client2->SendTo("foo", 3, client1_addr)); EXPECT_TRUE(client1->CheckNextPacket("foo", 3, &client2_addr)); } - void BasicTest(const webrtc::SocketAddress& initial_addr) { - webrtc::Socket* socket = - ss_.CreateSocket(initial_addr.family(), SOCK_DGRAM); + void BasicTest(const SocketAddress& initial_addr) { + Socket* socket = ss_.CreateSocket(initial_addr.family(), SOCK_DGRAM); socket->Bind(initial_addr); - webrtc::SocketAddress server_addr = socket->GetLocalAddress(); + SocketAddress server_addr = socket->GetLocalAddress(); // Make sure VSS didn't switch families on us. EXPECT_EQ(server_addr.family(), initial_addr.family()); - auto client1 = std::make_unique( - std::make_unique(socket), &fake_clock_); - webrtc::Socket* socket2 = - ss_.CreateSocket(initial_addr.family(), SOCK_DGRAM); - auto client2 = std::make_unique( - std::make_unique(socket2), &fake_clock_); + auto client1 = std::make_unique( + std::make_unique(socket), &fake_clock_); + Socket* socket2 = ss_.CreateSocket(initial_addr.family(), SOCK_DGRAM); + auto client2 = std::make_unique( + std::make_unique(socket2), &fake_clock_); - webrtc::SocketAddress client2_addr; + SocketAddress client2_addr; EXPECT_EQ(3, client2->SendTo("foo", 3, server_addr)); EXPECT_TRUE(client1->CheckNextPacket("foo", 3, &client2_addr)); - webrtc::SocketAddress client1_addr; + SocketAddress client1_addr; EXPECT_EQ(6, client1->SendTo("bizbaz", 6, client2_addr)); EXPECT_TRUE(client2->CheckNextPacket("bizbaz", 6, &client1_addr)); EXPECT_EQ(client1_addr, server_addr); - webrtc::SocketAddress empty = - webrtc::EmptySocketAddressWithFamily(initial_addr.family()); + SocketAddress empty = EmptySocketAddressWithFamily(initial_addr.family()); for (int i = 0; i < 10; i++) { - client2 = std::make_unique( - absl::WrapUnique(webrtc::AsyncUDPSocket::Create(&ss_, empty)), - &fake_clock_); + client2 = std::make_unique( + absl::WrapUnique(AsyncUDPSocket::Create(&ss_, empty)), &fake_clock_); - webrtc::SocketAddress next_client2_addr; + SocketAddress next_client2_addr; EXPECT_EQ(3, client2->SendTo("foo", 3, server_addr)); EXPECT_TRUE(client1->CheckNextPacket("foo", 3, &next_client2_addr)); CheckPortIncrementalization(next_client2_addr, client2_addr); // EXPECT_EQ(next_client2_addr.port(), client2_addr.port() + 1); - webrtc::SocketAddress server_addr2; + SocketAddress server_addr2; EXPECT_EQ(6, client1->SendTo("bizbaz", 6, next_client2_addr)); EXPECT_TRUE(client2->CheckNextPacket("bizbaz", 6, &server_addr2)); EXPECT_EQ(server_addr2, server_addr); @@ -256,28 +248,28 @@ class VirtualSocketServerTest : public ::testing::Test { } // initial_addr should be made from either INADDR_ANY or in6addr_any. - void ConnectTest(const webrtc::SocketAddress& initial_addr) { + void ConnectTest(const SocketAddress& initial_addr) { StreamSink sink; - webrtc::SocketAddress accept_addr; - const webrtc::SocketAddress kEmptyAddr = - webrtc::EmptySocketAddressWithFamily(initial_addr.family()); + SocketAddress accept_addr; + const SocketAddress kEmptyAddr = + EmptySocketAddressWithFamily(initial_addr.family()); // Create client - std::unique_ptr client = + std::unique_ptr client = absl::WrapUnique(ss_.CreateSocket(initial_addr.family(), SOCK_STREAM)); sink.Monitor(client.get()); - EXPECT_EQ(client->GetState(), webrtc::Socket::CS_CLOSED); + EXPECT_EQ(client->GetState(), Socket::CS_CLOSED); EXPECT_TRUE(client->GetLocalAddress().IsNil()); // Create server - std::unique_ptr server = + std::unique_ptr server = absl::WrapUnique(ss_.CreateSocket(initial_addr.family(), SOCK_STREAM)); sink.Monitor(server.get()); EXPECT_NE(0, server->Listen(5)); // Bind required EXPECT_EQ(0, server->Bind(initial_addr)); EXPECT_EQ(server->GetLocalAddress().family(), initial_addr.family()); EXPECT_EQ(0, server->Listen(5)); - EXPECT_EQ(server->GetState(), webrtc::Socket::CS_CONNECTING); + EXPECT_EQ(server->GetState(), Socket::CS_CONNECTING); // No pending server connections EXPECT_FALSE(sink.Check(server.get(), SSE_READ)); @@ -291,53 +283,53 @@ class VirtualSocketServerTest : public ::testing::Test { EXPECT_NE(client->GetLocalAddress(), server->GetLocalAddress()); // Client is connecting - EXPECT_EQ(client->GetState(), webrtc::Socket::CS_CONNECTING); + EXPECT_EQ(client->GetState(), Socket::CS_CONNECTING); EXPECT_FALSE(sink.Check(client.get(), SSE_OPEN)); EXPECT_FALSE(sink.Check(client.get(), SSE_CLOSE)); ss_.ProcessMessagesUntilIdle(); // Client still connecting - EXPECT_EQ(client->GetState(), webrtc::Socket::CS_CONNECTING); + EXPECT_EQ(client->GetState(), Socket::CS_CONNECTING); EXPECT_FALSE(sink.Check(client.get(), SSE_OPEN)); EXPECT_FALSE(sink.Check(client.get(), SSE_CLOSE)); // Server has pending connection EXPECT_TRUE(sink.Check(server.get(), SSE_READ)); - std::unique_ptr accepted = + std::unique_ptr accepted = absl::WrapUnique(server->Accept(&accept_addr)); EXPECT_TRUE(nullptr != accepted); EXPECT_NE(accept_addr, kEmptyAddr); EXPECT_EQ(accepted->GetRemoteAddress(), accept_addr); - EXPECT_EQ(accepted->GetState(), webrtc::Socket::CS_CONNECTED); + EXPECT_EQ(accepted->GetState(), Socket::CS_CONNECTED); EXPECT_EQ(accepted->GetLocalAddress(), server->GetLocalAddress()); EXPECT_EQ(accepted->GetRemoteAddress(), client->GetLocalAddress()); ss_.ProcessMessagesUntilIdle(); // Client has connected - EXPECT_EQ(client->GetState(), webrtc::Socket::CS_CONNECTED); + EXPECT_EQ(client->GetState(), Socket::CS_CONNECTED); EXPECT_TRUE(sink.Check(client.get(), SSE_OPEN)); EXPECT_FALSE(sink.Check(client.get(), SSE_CLOSE)); EXPECT_EQ(client->GetRemoteAddress(), server->GetLocalAddress()); EXPECT_EQ(client->GetRemoteAddress(), accepted->GetLocalAddress()); } - void ConnectToNonListenerTest(const webrtc::SocketAddress& initial_addr) { + void ConnectToNonListenerTest(const SocketAddress& initial_addr) { StreamSink sink; - webrtc::SocketAddress accept_addr; - const webrtc::SocketAddress nil_addr; - const webrtc::SocketAddress empty_addr = - webrtc::EmptySocketAddressWithFamily(initial_addr.family()); + SocketAddress accept_addr; + const SocketAddress nil_addr; + const SocketAddress empty_addr = + EmptySocketAddressWithFamily(initial_addr.family()); // Create client - std::unique_ptr client = + std::unique_ptr client = absl::WrapUnique(ss_.CreateSocket(initial_addr.family(), SOCK_STREAM)); sink.Monitor(client.get()); // Create server - std::unique_ptr server = + std::unique_ptr server = absl::WrapUnique(ss_.CreateSocket(initial_addr.family(), SOCK_STREAM)); sink.Monitor(server.get()); EXPECT_EQ(0, server->Bind(initial_addr)); @@ -353,23 +345,23 @@ class VirtualSocketServerTest : public ::testing::Test { EXPECT_EQ(accept_addr, nil_addr); // Connection failed - EXPECT_EQ(client->GetState(), webrtc::Socket::CS_CLOSED); + EXPECT_EQ(client->GetState(), Socket::CS_CLOSED); EXPECT_FALSE(sink.Check(client.get(), SSE_OPEN)); EXPECT_TRUE(sink.Check(client.get(), SSE_ERROR)); EXPECT_EQ(client->GetRemoteAddress(), nil_addr); } - void CloseDuringConnectTest(const webrtc::SocketAddress& initial_addr) { + void CloseDuringConnectTest(const SocketAddress& initial_addr) { StreamSink sink; - webrtc::SocketAddress accept_addr; - const webrtc::SocketAddress empty_addr = - webrtc::EmptySocketAddressWithFamily(initial_addr.family()); + SocketAddress accept_addr; + const SocketAddress empty_addr = + EmptySocketAddressWithFamily(initial_addr.family()); // Create client and server - std::unique_ptr client( + std::unique_ptr client( ss_.CreateSocket(initial_addr.family(), SOCK_STREAM)); sink.Monitor(client.get()); - std::unique_ptr server( + std::unique_ptr server( ss_.CreateSocket(initial_addr.family(), SOCK_STREAM)); sink.Monitor(server.get()); @@ -387,7 +379,7 @@ class VirtualSocketServerTest : public ::testing::Test { ss_.ProcessMessagesUntilIdle(); // Result: connection failed - EXPECT_EQ(client->GetState(), webrtc::Socket::CS_CLOSED); + EXPECT_EQ(client->GetState(), Socket::CS_CLOSED); EXPECT_TRUE(sink.Check(client.get(), SSE_ERROR)); server.reset(ss_.CreateSocket(initial_addr.family(), SOCK_STREAM)); @@ -409,7 +401,7 @@ class VirtualSocketServerTest : public ::testing::Test { ss_.ProcessMessagesUntilIdle(); // Result: connection failed - EXPECT_EQ(client->GetState(), webrtc::Socket::CS_CLOSED); + EXPECT_EQ(client->GetState(), Socket::CS_CLOSED); EXPECT_TRUE(sink.Check(client.get(), SSE_ERROR)); // New server @@ -427,37 +419,37 @@ class VirtualSocketServerTest : public ::testing::Test { // Server accepts connection EXPECT_TRUE(sink.Check(server.get(), SSE_READ)); - std::unique_ptr accepted(server->Accept(&accept_addr)); + std::unique_ptr accepted(server->Accept(&accept_addr)); ASSERT_TRUE(nullptr != accepted.get()); sink.Monitor(accepted.get()); // Client closes before connection complets - EXPECT_EQ(accepted->GetState(), webrtc::Socket::CS_CONNECTED); + EXPECT_EQ(accepted->GetState(), Socket::CS_CONNECTED); // Connected message has not been processed yet. - EXPECT_EQ(client->GetState(), webrtc::Socket::CS_CONNECTING); + EXPECT_EQ(client->GetState(), Socket::CS_CONNECTING); client->Close(); ss_.ProcessMessagesUntilIdle(); // Result: accepted socket closes - EXPECT_EQ(accepted->GetState(), webrtc::Socket::CS_CLOSED); + EXPECT_EQ(accepted->GetState(), Socket::CS_CLOSED); EXPECT_TRUE(sink.Check(accepted.get(), SSE_CLOSE)); EXPECT_FALSE(sink.Check(client.get(), SSE_CLOSE)); } - void CloseTest(const webrtc::SocketAddress& initial_addr) { + void CloseTest(const SocketAddress& initial_addr) { StreamSink sink; - const webrtc::SocketAddress kEmptyAddr; + const SocketAddress kEmptyAddr; // Create clients - std::unique_ptr a = + std::unique_ptr a = absl::WrapUnique(ss_.CreateSocket(initial_addr.family(), SOCK_STREAM)); sink.Monitor(a.get()); a->Bind(initial_addr); EXPECT_EQ(a->GetLocalAddress().family(), initial_addr.family()); - std::unique_ptr b = + std::unique_ptr b = absl::WrapUnique(ss_.CreateSocket(initial_addr.family(), SOCK_STREAM)); sink.Monitor(b.get()); b->Bind(initial_addr); @@ -469,11 +461,11 @@ class VirtualSocketServerTest : public ::testing::Test { ss_.ProcessMessagesUntilIdle(); EXPECT_TRUE(sink.Check(a.get(), SSE_OPEN)); - EXPECT_EQ(a->GetState(), webrtc::Socket::CS_CONNECTED); + EXPECT_EQ(a->GetState(), Socket::CS_CONNECTED); EXPECT_EQ(a->GetRemoteAddress(), b->GetLocalAddress()); EXPECT_TRUE(sink.Check(b.get(), SSE_OPEN)); - EXPECT_EQ(b->GetState(), webrtc::Socket::CS_CONNECTED); + EXPECT_EQ(b->GetState(), Socket::CS_CONNECTED); EXPECT_EQ(b->GetRemoteAddress(), a->GetLocalAddress()); EXPECT_EQ(1, a->Send("a", 1)); @@ -487,27 +479,27 @@ class VirtualSocketServerTest : public ::testing::Test { EXPECT_EQ(-1, b->Recv(buffer, 10, nullptr)); EXPECT_TRUE(sink.Check(a.get(), SSE_CLOSE)); - EXPECT_EQ(a->GetState(), webrtc::Socket::CS_CLOSED); + EXPECT_EQ(a->GetState(), Socket::CS_CLOSED); EXPECT_EQ(a->GetRemoteAddress(), kEmptyAddr); // No signal for Closer EXPECT_FALSE(sink.Check(b.get(), SSE_CLOSE)); - EXPECT_EQ(b->GetState(), webrtc::Socket::CS_CLOSED); + EXPECT_EQ(b->GetState(), Socket::CS_CLOSED); EXPECT_EQ(b->GetRemoteAddress(), kEmptyAddr); } - void TcpSendTest(const webrtc::SocketAddress& initial_addr) { + void TcpSendTest(const SocketAddress& initial_addr) { StreamSink sink; - const webrtc::SocketAddress kEmptyAddr; + const SocketAddress kEmptyAddr; // Connect two sockets - std::unique_ptr a = + std::unique_ptr a = absl::WrapUnique(ss_.CreateSocket(initial_addr.family(), SOCK_STREAM)); sink.Monitor(a.get()); a->Bind(initial_addr); EXPECT_EQ(a->GetLocalAddress().family(), initial_addr.family()); - std::unique_ptr b = + std::unique_ptr b = absl::WrapUnique(ss_.CreateSocket(initial_addr.family(), SOCK_STREAM)); sink.Monitor(b.get()); b->Bind(initial_addr); @@ -619,13 +611,13 @@ class VirtualSocketServerTest : public ::testing::Test { EXPECT_EQ(0, memcmp(recv_buffer, send_buffer, kDataSize)); } - void TcpSendsPacketsInOrderTest(const webrtc::SocketAddress& initial_addr) { - const webrtc::SocketAddress kEmptyAddr; + void TcpSendsPacketsInOrderTest(const SocketAddress& initial_addr) { + const SocketAddress kEmptyAddr; // Connect two sockets - std::unique_ptr a = + std::unique_ptr a = absl::WrapUnique(ss_.CreateSocket(initial_addr.family(), SOCK_STREAM)); - std::unique_ptr b = + std::unique_ptr b = absl::WrapUnique(ss_.CreateSocket(initial_addr.family(), SOCK_STREAM)); a->Bind(initial_addr); EXPECT_EQ(a->GetLocalAddress().family(), initial_addr.family()); @@ -676,11 +668,9 @@ class VirtualSocketServerTest : public ::testing::Test { // It is important that initial_addr's port has to be 0 such that the // incremental port behavior could ensure the 2 Binds result in different // address. - void BandwidthTest(const webrtc::SocketAddress& initial_addr) { - webrtc::Socket* send_socket = - ss_.CreateSocket(initial_addr.family(), SOCK_DGRAM); - webrtc::Socket* recv_socket = - ss_.CreateSocket(initial_addr.family(), SOCK_DGRAM); + void BandwidthTest(const SocketAddress& initial_addr) { + Socket* send_socket = ss_.CreateSocket(initial_addr.family(), SOCK_DGRAM); + Socket* recv_socket = ss_.CreateSocket(initial_addr.family(), SOCK_DGRAM); ASSERT_EQ(0, send_socket->Bind(initial_addr)); ASSERT_EQ(0, recv_socket->Bind(initial_addr)); EXPECT_EQ(send_socket->GetLocalAddress().family(), initial_addr.family()); @@ -690,7 +680,7 @@ class VirtualSocketServerTest : public ::testing::Test { uint32_t bandwidth = 64 * 1024; ss_.set_bandwidth(bandwidth); - webrtc::Thread* pthMain = webrtc::Thread::Current(); + Thread* pthMain = Thread::Current(); Sender sender(pthMain, send_socket, 80 * 1024); Receiver receiver(pthMain, recv_socket, bandwidth); @@ -710,7 +700,7 @@ class VirtualSocketServerTest : public ::testing::Test { // It is important that initial_addr's port has to be 0 such that the // incremental port behavior could ensure the 2 Binds result in different // address. - void DelayTest(const webrtc::SocketAddress& initial_addr) { + void DelayTest(const SocketAddress& initial_addr) { time_t seed = ::time(nullptr); RTC_LOG(LS_VERBOSE) << "seed = " << seed; srand(static_cast(seed)); @@ -722,17 +712,15 @@ class VirtualSocketServerTest : public ::testing::Test { ss_.set_delay_stddev(stddev); ss_.UpdateDelayDistribution(); - webrtc::Socket* send_socket = - ss_.CreateSocket(initial_addr.family(), SOCK_DGRAM); - webrtc::Socket* recv_socket = - ss_.CreateSocket(initial_addr.family(), SOCK_DGRAM); + Socket* send_socket = ss_.CreateSocket(initial_addr.family(), SOCK_DGRAM); + Socket* recv_socket = ss_.CreateSocket(initial_addr.family(), SOCK_DGRAM); ASSERT_EQ(0, send_socket->Bind(initial_addr)); ASSERT_EQ(0, recv_socket->Bind(initial_addr)); EXPECT_EQ(send_socket->GetLocalAddress().family(), initial_addr.family()); EXPECT_EQ(recv_socket->GetLocalAddress().family(), initial_addr.family()); ASSERT_EQ(0, send_socket->Connect(recv_socket->GetLocalAddress())); - webrtc::Thread* pthMain = webrtc::Thread::Current(); + Thread* pthMain = Thread::Current(); // Avg packet size is 2K, so at 200KB/s for 10s, we should see about // 1000 packets, which is necessary to get a good distribution. Sender sender(pthMain, send_socket, 100 * 2 * 1024); @@ -767,24 +755,24 @@ class VirtualSocketServerTest : public ::testing::Test { // Test cross-family communication between a client bound to client_addr and a // server bound to server_addr. shouldSucceed indicates if communication is // expected to work or not. - void CrossFamilyConnectionTest(const webrtc::SocketAddress& client_addr, - const webrtc::SocketAddress& server_addr, + void CrossFamilyConnectionTest(const SocketAddress& client_addr, + const SocketAddress& server_addr, bool shouldSucceed) { StreamSink sink; - webrtc::SocketAddress accept_address; - const webrtc::SocketAddress kEmptyAddr; + SocketAddress accept_address; + const SocketAddress kEmptyAddr; // Client gets a IPv4 address - std::unique_ptr client = + std::unique_ptr client = absl::WrapUnique(ss_.CreateSocket(client_addr.family(), SOCK_STREAM)); sink.Monitor(client.get()); - EXPECT_EQ(client->GetState(), webrtc::Socket::CS_CLOSED); + EXPECT_EQ(client->GetState(), Socket::CS_CLOSED); EXPECT_EQ(client->GetLocalAddress(), kEmptyAddr); client->Bind(client_addr); // Server gets a non-mapped non-any IPv6 address. // IPv4 sockets should not be able to connect to this. - std::unique_ptr server = + std::unique_ptr server = absl::WrapUnique(ss_.CreateSocket(server_addr.family(), SOCK_STREAM)); sink.Monitor(server.get()); server->Bind(server_addr); @@ -794,7 +782,7 @@ class VirtualSocketServerTest : public ::testing::Test { EXPECT_EQ(0, client->Connect(server->GetLocalAddress())); ss_.ProcessMessagesUntilIdle(); EXPECT_TRUE(sink.Check(server.get(), SSE_READ)); - std::unique_ptr accepted = + std::unique_ptr accepted = absl::WrapUnique(server->Accept(&accept_address)); EXPECT_TRUE(nullptr != accepted); EXPECT_NE(kEmptyAddr, accept_address); @@ -809,7 +797,7 @@ class VirtualSocketServerTest : public ::testing::Test { EXPECT_FALSE(sink.Check(server.get(), SSE_READ)); EXPECT_TRUE(nullptr == server->Accept(&accept_address)); EXPECT_EQ(accept_address, kEmptyAddr); - EXPECT_EQ(client->GetState(), webrtc::Socket::CS_CLOSED); + EXPECT_EQ(client->GetState(), Socket::CS_CLOSED); EXPECT_FALSE(sink.Check(client.get(), SSE_OPEN)); EXPECT_EQ(client->GetRemoteAddress(), kEmptyAddr); } @@ -818,25 +806,25 @@ class VirtualSocketServerTest : public ::testing::Test { // Test cross-family datagram sending between a client bound to client_addr // and a server bound to server_addr. shouldSucceed indicates if sending is // expected to succeed or not. - void CrossFamilyDatagramTest(const webrtc::SocketAddress& client_addr, - const webrtc::SocketAddress& server_addr, + void CrossFamilyDatagramTest(const SocketAddress& client_addr, + const SocketAddress& server_addr, bool shouldSucceed) { - webrtc::Socket* socket = ss_.CreateSocket(AF_INET, SOCK_DGRAM); + Socket* socket = ss_.CreateSocket(AF_INET, SOCK_DGRAM); socket->Bind(server_addr); - webrtc::SocketAddress bound_server_addr = socket->GetLocalAddress(); - auto client1 = std::make_unique( - std::make_unique(socket), &fake_clock_); + SocketAddress bound_server_addr = socket->GetLocalAddress(); + auto client1 = std::make_unique( + std::make_unique(socket), &fake_clock_); - webrtc::Socket* socket2 = ss_.CreateSocket(AF_INET, SOCK_DGRAM); + Socket* socket2 = ss_.CreateSocket(AF_INET, SOCK_DGRAM); socket2->Bind(client_addr); - auto client2 = std::make_unique( - std::make_unique(socket2), &fake_clock_); - webrtc::SocketAddress client2_addr; + auto client2 = std::make_unique( + std::make_unique(socket2), &fake_clock_); + SocketAddress client2_addr; if (shouldSucceed) { EXPECT_EQ(3, client2->SendTo("foo", 3, bound_server_addr)); EXPECT_TRUE(client1->CheckNextPacket("foo", 3, &client2_addr)); - webrtc::SocketAddress client1_addr; + SocketAddress client1_addr; EXPECT_EQ(6, client1->SendTo("bizbaz", 6, client2_addr)); EXPECT_TRUE(client2->CheckNextPacket("bizbaz", 6, &client1_addr)); EXPECT_EQ(client1_addr, bound_server_addr); @@ -847,32 +835,32 @@ class VirtualSocketServerTest : public ::testing::Test { } protected: - webrtc::ScopedFakeClock fake_clock_; - webrtc::VirtualSocketServer ss_; - webrtc::AutoSocketServerThread thread_; - const webrtc::SocketAddress kIPv4AnyAddress; - const webrtc::SocketAddress kIPv6AnyAddress; + ScopedFakeClock fake_clock_; + VirtualSocketServer ss_; + AutoSocketServerThread thread_; + const SocketAddress kIPv4AnyAddress; + const SocketAddress kIPv6AnyAddress; }; TEST_F(VirtualSocketServerTest, basic_v4) { - webrtc::SocketAddress ipv4_test_addr(webrtc::IPAddress(INADDR_ANY), 5000); + SocketAddress ipv4_test_addr(IPAddress(INADDR_ANY), 5000); BasicTest(ipv4_test_addr); } TEST_F(VirtualSocketServerTest, basic_v6) { - webrtc::SocketAddress ipv6_test_addr(webrtc::IPAddress(in6addr_any), 5000); + SocketAddress ipv6_test_addr(IPAddress(in6addr_any), 5000); BasicTest(ipv6_test_addr); } TEST_F(VirtualSocketServerTest, TestDefaultRoute_v4) { - webrtc::IPAddress ipv4_default_addr(0x01020304); + IPAddress ipv4_default_addr(0x01020304); TestDefaultSourceAddress(ipv4_default_addr); } TEST_F(VirtualSocketServerTest, TestDefaultRoute_v6) { - webrtc::IPAddress ipv6_default_addr; - EXPECT_TRUE(webrtc::IPFromString("2401:fa00:4:1000:be30:5bff:fee5:c3", - &ipv6_default_addr)); + IPAddress ipv6_default_addr; + EXPECT_TRUE( + IPFromString("2401:fa00:4:1000:be30:5bff:fee5:c3", &ipv6_default_addr)); TestDefaultSourceAddress(ipv6_default_addr); } @@ -942,105 +930,100 @@ TEST_F(VirtualSocketServerTest, delay_v6) { // Works, receiving socket sees 127.0.0.2. TEST_F(VirtualSocketServerTest, CanConnectFromMappedIPv6ToIPv4Any) { - CrossFamilyConnectionTest(webrtc::SocketAddress("::ffff:127.0.0.2", 0), - webrtc::SocketAddress("0.0.0.0", 5000), true); + CrossFamilyConnectionTest(SocketAddress("::ffff:127.0.0.2", 0), + SocketAddress("0.0.0.0", 5000), true); } // Fails. TEST_F(VirtualSocketServerTest, CantConnectFromUnMappedIPv6ToIPv4Any) { - CrossFamilyConnectionTest(webrtc::SocketAddress("::2", 0), - webrtc::SocketAddress("0.0.0.0", 5000), false); + CrossFamilyConnectionTest(SocketAddress("::2", 0), + SocketAddress("0.0.0.0", 5000), false); } // Fails. TEST_F(VirtualSocketServerTest, CantConnectFromUnMappedIPv6ToMappedIPv6) { - CrossFamilyConnectionTest(webrtc::SocketAddress("::2", 0), - webrtc::SocketAddress("::ffff:127.0.0.1", 5000), - false); + CrossFamilyConnectionTest(SocketAddress("::2", 0), + SocketAddress("::ffff:127.0.0.1", 5000), false); } // Works. receiving socket sees ::ffff:127.0.0.2. TEST_F(VirtualSocketServerTest, CanConnectFromIPv4ToIPv6Any) { - CrossFamilyConnectionTest(webrtc::SocketAddress("127.0.0.2", 0), - webrtc::SocketAddress("::", 5000), true); + CrossFamilyConnectionTest(SocketAddress("127.0.0.2", 0), + SocketAddress("::", 5000), true); } // Fails. TEST_F(VirtualSocketServerTest, CantConnectFromIPv4ToUnMappedIPv6) { - CrossFamilyConnectionTest(webrtc::SocketAddress("127.0.0.2", 0), - webrtc::SocketAddress("::1", 5000), false); + CrossFamilyConnectionTest(SocketAddress("127.0.0.2", 0), + SocketAddress("::1", 5000), false); } // Works. Receiving socket sees ::ffff:127.0.0.1. TEST_F(VirtualSocketServerTest, CanConnectFromIPv4ToMappedIPv6) { - CrossFamilyConnectionTest(webrtc::SocketAddress("127.0.0.1", 0), - webrtc::SocketAddress("::ffff:127.0.0.2", 5000), - true); + CrossFamilyConnectionTest(SocketAddress("127.0.0.1", 0), + SocketAddress("::ffff:127.0.0.2", 5000), true); } // Works, receiving socket sees a result from GetNextIP. TEST_F(VirtualSocketServerTest, CanConnectFromUnboundIPv6ToIPv4Any) { - CrossFamilyConnectionTest(webrtc::SocketAddress("::", 0), - webrtc::SocketAddress("0.0.0.0", 5000), true); + CrossFamilyConnectionTest(SocketAddress("::", 0), + SocketAddress("0.0.0.0", 5000), true); } // Works, receiving socket sees whatever GetNextIP gave the client. TEST_F(VirtualSocketServerTest, CanConnectFromUnboundIPv4ToIPv6Any) { - CrossFamilyConnectionTest(webrtc::SocketAddress("0.0.0.0", 0), - webrtc::SocketAddress("::", 5000), true); + CrossFamilyConnectionTest(SocketAddress("0.0.0.0", 0), + SocketAddress("::", 5000), true); } TEST_F(VirtualSocketServerTest, CanSendDatagramFromUnboundIPv4ToIPv6Any) { - CrossFamilyDatagramTest(webrtc::SocketAddress("0.0.0.0", 0), - webrtc::SocketAddress("::", 5000), true); + CrossFamilyDatagramTest(SocketAddress("0.0.0.0", 0), + SocketAddress("::", 5000), true); } TEST_F(VirtualSocketServerTest, CanSendDatagramFromMappedIPv6ToIPv4Any) { - CrossFamilyDatagramTest(webrtc::SocketAddress("::ffff:127.0.0.1", 0), - webrtc::SocketAddress("0.0.0.0", 5000), true); + CrossFamilyDatagramTest(SocketAddress("::ffff:127.0.0.1", 0), + SocketAddress("0.0.0.0", 5000), true); } TEST_F(VirtualSocketServerTest, CantSendDatagramFromUnMappedIPv6ToIPv4Any) { - CrossFamilyDatagramTest(webrtc::SocketAddress("::2", 0), - webrtc::SocketAddress("0.0.0.0", 5000), false); + CrossFamilyDatagramTest(SocketAddress("::2", 0), + SocketAddress("0.0.0.0", 5000), false); } TEST_F(VirtualSocketServerTest, CantSendDatagramFromUnMappedIPv6ToMappedIPv6) { - CrossFamilyDatagramTest(webrtc::SocketAddress("::2", 0), - webrtc::SocketAddress("::ffff:127.0.0.1", 5000), - false); + CrossFamilyDatagramTest(SocketAddress("::2", 0), + SocketAddress("::ffff:127.0.0.1", 5000), false); } TEST_F(VirtualSocketServerTest, CanSendDatagramFromIPv4ToIPv6Any) { - CrossFamilyDatagramTest(webrtc::SocketAddress("127.0.0.2", 0), - webrtc::SocketAddress("::", 5000), true); + CrossFamilyDatagramTest(SocketAddress("127.0.0.2", 0), + SocketAddress("::", 5000), true); } TEST_F(VirtualSocketServerTest, CantSendDatagramFromIPv4ToUnMappedIPv6) { - CrossFamilyDatagramTest(webrtc::SocketAddress("127.0.0.2", 0), - webrtc::SocketAddress("::1", 5000), false); + CrossFamilyDatagramTest(SocketAddress("127.0.0.2", 0), + SocketAddress("::1", 5000), false); } TEST_F(VirtualSocketServerTest, CanSendDatagramFromIPv4ToMappedIPv6) { - CrossFamilyDatagramTest(webrtc::SocketAddress("127.0.0.1", 0), - webrtc::SocketAddress("::ffff:127.0.0.2", 5000), - true); + CrossFamilyDatagramTest(SocketAddress("127.0.0.1", 0), + SocketAddress("::ffff:127.0.0.2", 5000), true); } TEST_F(VirtualSocketServerTest, CanSendDatagramFromUnboundIPv6ToIPv4Any) { - CrossFamilyDatagramTest(webrtc::SocketAddress("::", 0), - webrtc::SocketAddress("0.0.0.0", 5000), true); + CrossFamilyDatagramTest(SocketAddress("::", 0), + SocketAddress("0.0.0.0", 5000), true); } TEST_F(VirtualSocketServerTest, SetSendingBlockedWithUdpSocket) { - webrtc::Socket* socket1 = - ss_.CreateSocket(kIPv4AnyAddress.family(), SOCK_DGRAM); - std::unique_ptr socket2 = + Socket* socket1 = ss_.CreateSocket(kIPv4AnyAddress.family(), SOCK_DGRAM); + std::unique_ptr socket2 = absl::WrapUnique(ss_.CreateSocket(kIPv4AnyAddress.family(), SOCK_DGRAM)); socket1->Bind(kIPv4AnyAddress); socket2->Bind(kIPv4AnyAddress); - auto client1 = std::make_unique( - std::make_unique(socket1), &fake_clock_); + auto client1 = std::make_unique( + std::make_unique(socket1), &fake_clock_); ss_.SetSendingBlocked(true); EXPECT_EQ(-1, client1->SendTo("foo", 3, socket2->GetLocalAddress())); @@ -1058,9 +1041,9 @@ TEST_F(VirtualSocketServerTest, SetSendingBlockedWithTcpSocket) { ss_.set_recv_buffer_capacity(kBufferSize); StreamSink sink; - std::unique_ptr socket1 = + std::unique_ptr socket1 = absl::WrapUnique(ss_.CreateSocket(kIPv4AnyAddress.family(), SOCK_STREAM)); - std::unique_ptr socket2 = + std::unique_ptr socket2 = absl::WrapUnique(ss_.CreateSocket(kIPv4AnyAddress.family(), SOCK_STREAM)); sink.Monitor(socket1.get()); sink.Monitor(socket2.get()); @@ -1107,9 +1090,9 @@ TEST_F(VirtualSocketServerTest, CreatesStandardDistribution) { ASSERT_LT(0u, kTestSamples[sidx]); const uint32_t kStdDev = static_cast(kTestDev[didx] * kTestMean[midx]); - std::unique_ptr f = - webrtc::VirtualSocketServer::CreateDistribution( - kTestMean[midx], kStdDev, kTestSamples[sidx]); + std::unique_ptr f = + VirtualSocketServer::CreateDistribution(kTestMean[midx], kStdDev, + kTestSamples[sidx]); ASSERT_TRUE(nullptr != f.get()); ASSERT_EQ(kTestSamples[sidx], f->size()); double sum = 0; From 94adaee01aedd003641abd0e83d88373084d0102 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Fri, 9 May 2025 10:35:15 +0000 Subject: [PATCH 104/292] [cleanup] Fix redundant webrtc name specifier This CL was uploaded by git cl split. R=hta@webrtc.org No-IWYU: LSC Bug: webrtc:42232595 Change-Id: I4dd648eb7ca130869240d63855d73feee5d0db74 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390562 Auto-Submit: Evan Shrubsole Reviewed-by: Tomas Gunnarsson Reviewed-by: Harald Alvestrand Commit-Queue: Tomas Gunnarsson Cr-Commit-Position: refs/heads/main@{#44576} --- call/call_perf_tests.cc | 8 +- call/call_unittest.cc | 18 +- call/payload_type_picker.cc | 2 +- call/rampup_tests.cc | 10 +- call/receive_time_calculator_unittest.cc | 2 +- p2p/base/async_stun_tcp_socket.cc | 4 +- p2p/base/basic_async_resolver_factory.cc | 10 +- p2p/base/basic_ice_controller.cc | 29 +- p2p/base/connection.cc | 94 +- p2p/base/ice_controller_interface.cc | 2 +- p2p/base/ice_credentials_iterator.cc | 5 +- p2p/base/ice_transport_internal.cc | 5 +- p2p/base/p2p_transport_channel.cc | 62 +- p2p/base/packet_transport_internal.cc | 4 +- p2p/base/port.cc | 53 +- p2p/base/port_allocator.cc | 13 +- p2p/base/pseudo_tcp.cc | 39 +- p2p/base/regathering_controller_unittest.cc | 2 +- p2p/base/stun_port.cc | 28 +- p2p/base/stun_request.cc | 10 +- p2p/base/stun_request_unittest.cc | 9 +- p2p/base/tcp_port.cc | 24 +- p2p/base/turn_port.cc | 109 +- p2p/base/turn_port_unittest.cc | 617 ++++---- p2p/client/basic_port_allocator.cc | 121 +- p2p/client/basic_port_allocator_unittest.cc | 1284 ++++++++--------- p2p/dtls/dtls_stun_piggyback_controller.cc | 3 +- p2p/dtls/dtls_transport.cc | 284 ++-- p2p/dtls/dtls_utils.cc | 14 +- p2p/test/nat_server.cc | 19 +- p2p/test/nat_socket_factory.cc | 9 +- p2p/test/test_stun_server.cc | 6 +- p2p/test/turn_server.cc | 9 +- pc/audio_rtp_receiver_unittest.cc | 2 +- pc/channel.cc | 76 +- pc/data_channel_controller_unittest.cc | 2 +- pc/data_channel_integrationtest.cc | 6 +- pc/data_channel_unittest.cc | 4 +- pc/jsep_transport.cc | 7 +- pc/jsep_transport_controller_unittest.cc | 94 +- pc/jsep_transport_unittest.cc | 67 +- pc/legacy_stats_collector.cc | 8 +- pc/legacy_stats_collector_unittest.cc | 8 +- pc/media_session.cc | 146 +- pc/peer_connection_bundle_unittest.cc | 19 +- pc/peer_connection_crypto_unittest.cc | 2 +- pc/peer_connection_data_channel_unittest.cc | 6 +- ...er_connection_encodings_integrationtest.cc | 139 +- pc/peer_connection_factory.cc | 12 +- pc/peer_connection_factory_unittest.cc | 28 +- pc/peer_connection_field_trial_tests.cc | 10 +- pc/peer_connection_histogram_unittest.cc | 3 +- pc/peer_connection_ice_unittest.cc | 52 +- pc/peer_connection_integrationtest.cc | 72 +- pc/peer_connection_interface_unittest.cc | 84 +- pc/peer_connection_jsep_unittest.cc | 180 ++- pc/peer_connection_media_unittest.cc | 135 +- pc/peer_connection_rampup_tests.cc | 6 +- pc/peer_connection_rtp_unittest.cc | 156 +- pc/peer_connection_signaling_unittest.cc | 35 +- pc/peer_connection_simulcast_unittest.cc | 11 +- pc/peer_connection_svc_integrationtest.cc | 9 +- pc/peer_connection_wrapper.cc | 4 +- pc/rtc_stats_collector.cc | 92 +- pc/rtc_stats_collector_unittest.cc | 213 ++- pc/rtc_stats_integrationtest.cc | 20 +- pc/rtcp_mux_filter.cc | 18 +- pc/rtp_parameters_conversion.cc | 5 +- pc/rtp_parameters_conversion_unittest.cc | 4 +- pc/rtp_sender.cc | 8 +- pc/rtp_transceiver.cc | 25 +- pc/rtp_transceiver_unittest.cc | 47 +- pc/rtp_transmission_manager.cc | 81 +- pc/rtp_transport.cc | 2 +- pc/rtp_transport_unittest.cc | 4 +- pc/scenario_tests/goog_cc_test.cc | 2 +- pc/sdp_munging_detector.cc | 12 +- pc/sdp_munging_detector_unittest.cc | 4 +- pc/sdp_offer_answer.cc | 145 +- pc/sdp_offer_answer_unittest.cc | 27 +- pc/simulcast_sdp_serializer.cc | 6 +- pc/srtp_session.cc | 17 +- pc/srtp_session_unittest.cc | 119 +- pc/test/svc_e2e_tests.cc | 12 +- pc/track_media_info_map.cc | 12 +- pc/track_media_info_map_unittest.cc | 12 +- test/peer_scenario/peer_scenario_client.cc | 9 +- test/peer_scenario/scenario_connection.cc | 8 +- test/peer_scenario/tests/bwe_ramp_up_test.cc | 13 +- test/peer_scenario/tests/l4s_test.cc | 3 +- 90 files changed, 2461 insertions(+), 2750 deletions(-) diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc index 17e988e76c..c7d2e17d8b 100644 --- a/call/call_perf_tests.cc +++ b/call/call_perf_tests.cc @@ -90,9 +90,9 @@ using webrtc::test::DriftingClock; namespace webrtc { namespace { -using ::webrtc::test::GetGlobalMetricsLogger; -using ::webrtc::test::ImprovementDirection; -using ::webrtc::test::Unit; +using test::GetGlobalMetricsLogger; +using test::ImprovementDirection; +using test::Unit; enum : int { // The first valid value is 1. kTransportSequenceNumberExtensionId = 1, @@ -666,7 +666,7 @@ TEST_F(CallPerfTest, MAYBE_KeepsHighBitrateWhenReconfiguringSender) { const FieldTrialsView& /*field_trials*/, int frame_width, int frame_height, - const webrtc::VideoEncoderConfig& encoder_config) override { + const VideoEncoderConfig& encoder_config) override { std::vector streams = test::CreateVideoStreams(frame_width, frame_height, encoder_config); streams[0].min_bitrate_bps = 50000; diff --git a/call/call_unittest.cc b/call/call_unittest.cc index d891853dcc..32f41fece0 100644 --- a/call/call_unittest.cc +++ b/call/call_unittest.cc @@ -53,16 +53,16 @@ namespace webrtc { namespace { +using test::FakeEncoder; +using test::FunctionVideoEncoderFactory; +using test::MockAudioDeviceModule; +using test::MockAudioMixer; +using test::MockAudioProcessing; +using test::RunLoop; using ::testing::_; using ::testing::MockFunction; using ::testing::NiceMock; using ::testing::StrictMock; -using ::webrtc::test::FakeEncoder; -using ::webrtc::test::FunctionVideoEncoderFactory; -using ::webrtc::test::MockAudioDeviceModule; -using ::webrtc::test::MockAudioMixer; -using ::webrtc::test::MockAudioProcessing; -using ::webrtc::test::RunLoop; struct CallHelper { explicit CallHelper(bool use_null_audio_processing) { @@ -123,8 +123,7 @@ TEST(CallTest, CreateDestroy_AudioReceiveStream) { MockTransport rtcp_send_transport; config.rtp.remote_ssrc = 42; config.rtcp_send_transport = &rtcp_send_transport; - config.decoder_factory = - make_ref_counted(); + config.decoder_factory = make_ref_counted(); AudioReceiveStreamInterface* stream = call->CreateAudioReceiveStream(config); EXPECT_NE(stream, nullptr); @@ -163,8 +162,7 @@ TEST(CallTest, CreateDestroy_AudioReceiveStreams) { AudioReceiveStreamInterface::Config config; MockTransport rtcp_send_transport; config.rtcp_send_transport = &rtcp_send_transport; - config.decoder_factory = - make_ref_counted(); + config.decoder_factory = make_ref_counted(); std::list streams; for (int i = 0; i < 2; ++i) { for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) { diff --git a/call/payload_type_picker.cc b/call/payload_type_picker.cc index cee6d46094..c5bfa71269 100644 --- a/call/payload_type_picker.cc +++ b/call/payload_type_picker.cc @@ -45,7 +45,7 @@ static const int kLastDynamicPayloadTypeUpperRange = 127; // is only a temporary measure. struct MapTableEntry { - webrtc::SdpAudioFormat format; + SdpAudioFormat format; int payload_type; }; diff --git a/call/rampup_tests.cc b/call/rampup_tests.cc index 5b121f01b5..3a18e8c6d8 100644 --- a/call/rampup_tests.cc +++ b/call/rampup_tests.cc @@ -61,9 +61,9 @@ ABSL_FLAG(std::string, namespace webrtc { namespace { -using ::webrtc::test::GetGlobalMetricsLogger; -using ::webrtc::test::ImprovementDirection; -using ::webrtc::test::Unit; +using test::GetGlobalMetricsLogger; +using test::ImprovementDirection; +using test::Unit; constexpr TimeDelta kPollInterval = TimeDelta::Millis(20); static const int kExpectedHighVideoBitrateBps = 80000; @@ -450,7 +450,7 @@ void RampUpDownUpTester::PollStats() { int transmit_bitrate_bps = 0; bool suspended = false; if (num_video_streams_ > 0 && send_stream_) { - webrtc::VideoSendStream::Stats stats = send_stream_->GetStats(); + VideoSendStream::Stats stats = send_stream_->GetStats(); for (const auto& it : stats.substreams) { transmit_bitrate_bps += it.second.total_bitrate_bps; } @@ -503,7 +503,7 @@ int RampUpDownUpTester::GetExpectedHighBitrate() const { size_t RampUpDownUpTester::GetFecBytes() const { size_t flex_fec_bytes = 0; if (num_flexfec_streams_ > 0) { - webrtc::VideoSendStream::Stats stats = send_stream_->GetStats(); + VideoSendStream::Stats stats = send_stream_->GetStats(); for (const auto& kv : stats.substreams) flex_fec_bytes += kv.second.rtp_stats.fec.TotalBytes(); } diff --git a/call/receive_time_calculator_unittest.cc b/call/receive_time_calculator_unittest.cc index 0dd8e45545..61275f884e 100644 --- a/call/receive_time_calculator_unittest.cc +++ b/call/receive_time_calculator_unittest.cc @@ -169,7 +169,7 @@ class EmulatedNonMonotoneousClock : public EmulatedClock { }; TEST(ClockRepair, NoClockDrift) { - webrtc::test::ScopedKeyValueConfig field_trials; + test::ScopedKeyValueConfig field_trials; const int kSeeds = 10; const int kFirstSeed = 1; const int64_t kRuntimeUs = 10 * kNumMicrosecsPerSec; diff --git a/p2p/base/async_stun_tcp_socket.cc b/p2p/base/async_stun_tcp_socket.cc index 8eb8c3c1cf..9a1c906619 100644 --- a/p2p/base/async_stun_tcp_socket.cc +++ b/p2p/base/async_stun_tcp_socket.cc @@ -140,9 +140,9 @@ size_t AsyncStunTCPSocket::GetExpectedLength(const void* data, int* pad_bytes) { *pad_bytes = 0; PacketLength pkt_len = - webrtc::GetBE16(static_cast(data) + kPacketLenOffset); + GetBE16(static_cast(data) + kPacketLenOffset); size_t expected_pkt_len; - uint16_t msg_type = webrtc::GetBE16(data); + uint16_t msg_type = GetBE16(data); if (IsStunMessage(msg_type)) { // STUN message. expected_pkt_len = kStunHeaderSize + pkt_len; diff --git a/p2p/base/basic_async_resolver_factory.cc b/p2p/base/basic_async_resolver_factory.cc index e66e8fb6e9..260fff3c00 100644 --- a/p2p/base/basic_async_resolver_factory.cc +++ b/p2p/base/basic_async_resolver_factory.cc @@ -20,26 +20,26 @@ namespace webrtc { -std::unique_ptr +std::unique_ptr BasicAsyncDnsResolverFactory::Create() { return std::make_unique(); } -std::unique_ptr +std::unique_ptr BasicAsyncDnsResolverFactory::CreateAndResolve( const SocketAddress& addr, absl::AnyInvocable callback) { - std::unique_ptr resolver = Create(); + std::unique_ptr resolver = Create(); resolver->Start(addr, std::move(callback)); return resolver; } -std::unique_ptr +std::unique_ptr BasicAsyncDnsResolverFactory::CreateAndResolve( const SocketAddress& addr, int family, absl::AnyInvocable callback) { - std::unique_ptr resolver = Create(); + std::unique_ptr resolver = Create(); resolver->Start(addr, family, std::move(callback)); return resolver; } diff --git a/p2p/base/basic_ice_controller.cc b/p2p/base/basic_ice_controller.cc index 713a154c99..cea38d9270 100644 --- a/p2p/base/basic_ice_controller.cc +++ b/p2p/base/basic_ice_controller.cc @@ -116,7 +116,7 @@ void BasicIceController::OnConnectionDestroyed(const Connection* connection) { } bool BasicIceController::HasPingableConnection() const { - int64_t now = webrtc::TimeMillis(); + int64_t now = TimeMillis(); return absl::c_any_of(connections_, [this, now](const Connection* c) { return IsPingable(c, now); }); @@ -130,14 +130,14 @@ IceControllerInterface::PingResult BasicIceController::SelectConnectionToPing( bool need_more_pings_at_weak_interval = absl::c_any_of(connections_, [](const webrtc::Connection* conn) { return conn->active() && - conn->num_pings_sent() < webrtc::MIN_PINGS_AT_WEAK_PING_INTERVAL; + conn->num_pings_sent() < MIN_PINGS_AT_WEAK_PING_INTERVAL; }); int ping_interval = (weak() || need_more_pings_at_weak_interval) ? weak_ping_interval() : strong_ping_interval(); const Connection* conn = nullptr; - if (webrtc::TimeMillis() >= last_ping_sent_ms + ping_interval) { + if (TimeMillis() >= last_ping_sent_ms + ping_interval) { conn = FindNextPingableConnection(); } PingResult res(conn, std::min(ping_interval, check_receiving_interval())); @@ -152,7 +152,7 @@ void BasicIceController::MarkConnectionPinged(const Connection* conn) { // Returns the next pingable connection to ping. const Connection* BasicIceController::FindNextPingableConnection() { - int64_t now = webrtc::TimeMillis(); + int64_t now = TimeMillis(); // Rule 1: Selected connection takes priority over non-selected ones. if (selected_connection_ && selected_connection_->connected() && @@ -274,15 +274,14 @@ int BasicIceController::CalculateActiveWritablePingInterval( int64_t now) const { // Ping each connection at a higher rate at least // MIN_PINGS_AT_WEAK_PING_INTERVAL times. - if (conn->num_pings_sent() < webrtc::MIN_PINGS_AT_WEAK_PING_INTERVAL) { + if (conn->num_pings_sent() < MIN_PINGS_AT_WEAK_PING_INTERVAL) { return weak_ping_interval(); } int stable_interval = config_.stable_writable_connection_ping_interval_or_default(); - int weak_or_stablizing_interval = - std::min(stable_interval, - webrtc::WEAK_OR_STABILIZING_WRITABLE_CONNECTION_PING_INTERVAL); + int weak_or_stablizing_interval = std::min( + stable_interval, WEAK_OR_STABILIZING_WRITABLE_CONNECTION_PING_INTERVAL); // If the channel is weak or the connection is not stable yet, use the // weak_or_stablizing_interval. return (!weak() && conn->stable(now)) ? stable_interval @@ -453,7 +452,7 @@ BasicIceController::HandleInitialSelectDampening( return {new_connection, std::nullopt}; } - int64_t now = webrtc::TimeMillis(); + int64_t now = TimeMillis(); int64_t max_delay = 0; if (new_connection->last_ping_received() > 0 && field_trials_->initial_select_dampening_ping_received.has_value()) { @@ -521,7 +520,7 @@ IceControllerInterface::SwitchResult BasicIceController::ShouldSwitchConnection( bool missed_receiving_unchanged_threshold = false; std::optional receiving_unchanged_threshold( - webrtc::TimeMillis() - config_.receiving_switching_delay_or_default()); + TimeMillis() - config_.receiving_switching_delay_or_default()); int cmp = CompareConnections(selected_connection_, new_connection, receiving_unchanged_threshold, &missed_receiving_unchanged_threshold); @@ -571,7 +570,7 @@ BasicIceController::SortAndSwitchConnection(IceSwitchReason reason) { RTC_LOG(LS_VERBOSE) << "Sorting " << connections_.size() << " available connections due to: " - << webrtc::IceSwitchReasonToString(reason); + << IceSwitchReasonToString(reason); for (size_t i = 0; i < connections_.size(); ++i) { RTC_LOG(LS_VERBOSE) << connections_[i]->ToString(); } @@ -737,7 +736,7 @@ int BasicIceController::CompareConnections( return state_cmp; } - if (ice_role_func_() == webrtc::ICEROLE_CONTROLLED) { + if (ice_role_func_() == ICEROLE_CONTROLLED) { // Compare the connections based on the nomination states and the last data // received time if this is on the controlled side. if (a->remote_nomination() > b->remote_nomination()) { @@ -827,7 +826,7 @@ std::vector BasicIceController::PruneConnections() { auto best_connection_by_network = GetBestConnectionByNetwork(); for (const Connection* conn : connections_) { const Connection* best_conn = selected_connection_; - if (!webrtc::IPIsAny(conn->network()->GetBestIP())) { + if (!IPIsAny(conn->network()->GetBestIP())) { // If the connection is bound to a specific network interface (not an // "any address" network), compare it against the best connection for // that network interface rather than the best connection overall. This @@ -853,7 +852,7 @@ bool BasicIceController::GetUseCandidateAttr(const Connection* conn, // TODO(honghaiz): Implement regular nomination. return false; case NominationMode::AGGRESSIVE: - if (remote_ice_mode == webrtc::ICEMODE_LITE) { + if (remote_ice_mode == ICEMODE_LITE) { return GetUseCandidateAttr(conn, NominationMode::REGULAR, remote_ice_mode); } @@ -869,7 +868,7 @@ bool BasicIceController::GetUseCandidateAttr(const Connection* conn, // b.1) `conn` is the selected_connection AND // b.2) `conn` is writable. bool selected = conn == selected_connection_; - if (remote_ice_mode == webrtc::ICEMODE_LITE) { + if (remote_ice_mode == ICEMODE_LITE) { return selected && conn->writable(); } bool better_than_selected = diff --git a/p2p/base/connection.cc b/p2p/base/connection.cc index e2ccfaed53..497df771a6 100644 --- a/p2p/base/connection.cc +++ b/p2p/base/connection.cc @@ -64,8 +64,6 @@ namespace webrtc { namespace { -using webrtc::IceCandidateType; - // Determines whether we have seen at least the given maximum number of // pings fail to have a response. inline bool TooManyFailures( @@ -111,13 +109,13 @@ IceCandidateType GetRtcEventLogCandidateType(const Candidate& c) { } IceCandidatePairProtocol GetProtocolByString(absl::string_view protocol) { - if (protocol == webrtc::UDP_PROTOCOL_NAME) { + if (protocol == UDP_PROTOCOL_NAME) { return IceCandidatePairProtocol::kUdp; - } else if (protocol == webrtc::TCP_PROTOCOL_NAME) { + } else if (protocol == TCP_PROTOCOL_NAME) { return IceCandidatePairProtocol::kTcp; - } else if (protocol == webrtc::SSLTCP_PROTOCOL_NAME) { + } else if (protocol == SSLTCP_PROTOCOL_NAME) { return IceCandidatePairProtocol::kSsltcp; - } else if (protocol == webrtc::TLS_PROTOCOL_NAME) { + } else if (protocol == TLS_PROTOCOL_NAME) { return IceCandidatePairProtocol::kTls; } return IceCandidatePairProtocol::kUnknown; @@ -134,19 +132,19 @@ IceCandidatePairAddressFamily GetAddressFamilyByInt(int address_family) { IceCandidateNetworkType ConvertNetworkType(AdapterType type) { switch (type) { - case webrtc::ADAPTER_TYPE_ETHERNET: + case ADAPTER_TYPE_ETHERNET: return IceCandidateNetworkType::kEthernet; - case webrtc::ADAPTER_TYPE_LOOPBACK: + case ADAPTER_TYPE_LOOPBACK: return IceCandidateNetworkType::kLoopback; - case webrtc::ADAPTER_TYPE_WIFI: + case ADAPTER_TYPE_WIFI: return IceCandidateNetworkType::kWifi; - case webrtc::ADAPTER_TYPE_VPN: + case ADAPTER_TYPE_VPN: return IceCandidateNetworkType::kVpn; - case webrtc::ADAPTER_TYPE_CELLULAR: - case webrtc::ADAPTER_TYPE_CELLULAR_2G: - case webrtc::ADAPTER_TYPE_CELLULAR_3G: - case webrtc::ADAPTER_TYPE_CELLULAR_4G: - case webrtc::ADAPTER_TYPE_CELLULAR_5G: + case ADAPTER_TYPE_CELLULAR: + case ADAPTER_TYPE_CELLULAR_2G: + case ADAPTER_TYPE_CELLULAR_3G: + case ADAPTER_TYPE_CELLULAR_4G: + case ADAPTER_TYPE_CELLULAR_5G: return IceCandidateNetworkType::kCellular; default: return IceCandidateNetworkType::kUnknown; @@ -225,14 +223,14 @@ void Connection::ConnectionRequest::OnSent() { } int Connection::ConnectionRequest::resend_delay() { - return webrtc::CONNECTION_RESPONSE_TIMEOUT; + return CONNECTION_RESPONSE_TIMEOUT; } Connection::Connection(WeakPtr port, size_t index, const Candidate& remote_candidate) : network_thread_(port->thread()), - id_(webrtc::CreateRandomId()), + id_(CreateRandomId()), port_(std::move(port)), local_candidate_(port_->Candidates()[index]), remote_candidate_(remote_candidate), @@ -253,9 +251,8 @@ Connection::Connection(WeakPtr port, last_data_received_(0), last_ping_response_received_(0), state_(IceCandidatePairState::WAITING), - time_created_ms_(webrtc::TimeMillis()), - delta_internal_unix_epoch_ms_(webrtc::TimeUTCMillis() - - webrtc::TimeMillis()), + time_created_ms_(TimeMillis()), + delta_internal_unix_epoch_ms_(TimeUTCMillis() - TimeMillis()), field_trials_(&kDefaultFieldTrials), rtt_estimate_(DEFAULT_RTT_ESTIMATE_HALF_TIME_MS) { RTC_DCHECK_RUN_ON(network_thread_); @@ -304,10 +301,10 @@ uint64_t Connection::priority() const { // controlled agent. // pair priority = 2^32*MIN(G,D) + 2*MAX(G,D) + (G>D?1:0) IceRole role = port_->GetIceRole(); - if (role != webrtc::ICEROLE_UNKNOWN) { + if (role != ICEROLE_UNKNOWN) { uint32_t g = 0; uint32_t d = 0; - if (role == webrtc::ICEROLE_CONTROLLING) { + if (role == ICEROLE_CONTROLLING) { g = local_candidate().priority(); d = remote_candidate_.priority(); } else { @@ -404,7 +401,7 @@ bool Connection::nominated() const { int Connection::unwritable_timeout() const { RTC_DCHECK_RUN_ON(network_thread_); - return unwritable_timeout_.value_or(webrtc::CONNECTION_WRITE_CONNECT_TIMEOUT); + return unwritable_timeout_.value_or(CONNECTION_WRITE_CONNECT_TIMEOUT); } void Connection::set_unwritable_timeout(const std::optional& value_ms) { @@ -414,8 +411,7 @@ void Connection::set_unwritable_timeout(const std::optional& value_ms) { int Connection::unwritable_min_checks() const { RTC_DCHECK_RUN_ON(network_thread_); - return unwritable_min_checks_.value_or( - webrtc::CONNECTION_WRITE_CONNECT_FAILURES); + return unwritable_min_checks_.value_or(CONNECTION_WRITE_CONNECT_FAILURES); } void Connection::set_unwritable_min_checks(const std::optional& value) { @@ -425,7 +421,7 @@ void Connection::set_unwritable_min_checks(const std::optional& value) { int Connection::inactive_timeout() const { RTC_DCHECK_RUN_ON(network_thread_); - return inactive_timeout_.value_or(webrtc::CONNECTION_WRITE_TIMEOUT); + return inactive_timeout_.value_or(CONNECTION_WRITE_TIMEOUT); } void Connection::set_inactive_timeout(const std::optional& value) { @@ -435,7 +431,7 @@ void Connection::set_inactive_timeout(const std::optional& value) { int Connection::receiving_timeout() const { RTC_DCHECK_RUN_ON(network_thread_); - return receiving_timeout_.value_or(webrtc::WEAK_CONNECTION_RECEIVE_TIMEOUT); + return receiving_timeout_.value_or(WEAK_CONNECTION_RECEIVE_TIMEOUT); } void Connection::set_receiving_timeout( @@ -468,8 +464,7 @@ void Connection::OnSendStunPacket(const void* data, } void Connection::RegisterReceivedPacketCallback( - absl::AnyInvocable + absl::AnyInvocable received_packet_callback) { RTC_DCHECK_RUN_ON(network_thread_); RTC_CHECK(!received_packet_callback_); @@ -496,7 +491,7 @@ void Connection::OnReadPacket(const ReceivedIpPacket& packet) { packet.payload().size(), addr, &msg, &remote_ufrag)) { // The packet did not parse as a valid STUN message // This is a data packet, pass it along. - last_data_received_ = webrtc::TimeMillis(); + last_data_received_ = TimeMillis(); UpdateReceiving(last_data_received_); recv_rate_tracker_.AddSamples(packet.payload().size()); stats_.packets_received++; @@ -642,7 +637,7 @@ void Connection::HandleStunBindingOrGoogPingRequest(IceMessage* msg) { if (field_trials_->extra_ice_ping && last_ping_response_received_ == 0) { if (local_candidate().is_relay() || local_candidate().is_prflx() || remote_candidate().is_relay() || remote_candidate().is_prflx()) { - const int64_t now = webrtc::TimeMillis(); + const int64_t now = TimeMillis(); if (last_ping_sent_ + kMinExtraPingDelayMs <= now) { RTC_LOG(LS_INFO) << ToString() << "WebRTC-ExtraICEPing/Sending extra ping" @@ -696,7 +691,7 @@ void Connection::HandleStunBindingOrGoogPingRequest(IceMessage* msg) { set_write_state(STATE_WRITE_INIT); } - if (port_->GetIceRole() == webrtc::ICEROLE_CONTROLLED) { + if (port_->GetIceRole() == ICEROLE_CONTROLLED) { const StunUInt32Attribute* nomination_attr = msg->GetUInt32(STUN_ATTR_NOMINATION); uint32_t nomination = 0; @@ -762,7 +757,7 @@ void Connection::SendStunBindingResponse(const StunMessage* message) { response.AddAttribute(std::make_unique( STUN_ATTR_RETRANSMIT_COUNT, retransmit_attr->value())); - if (retransmit_attr->value() > webrtc::CONNECTION_WRITE_CONNECT_FAILURES) { + if (retransmit_attr->value() > CONNECTION_WRITE_CONNECT_FAILURES) { RTC_LOG(LS_INFO) << ToString() << ": Received a remote ping with high retransmit count: " @@ -977,7 +972,7 @@ void Connection::UpdateState(int64_t now) { return; // Computes our estimate of the RTT given the current estimate. - int rtt = webrtc::SafeClamp(2 * rtt_, MINIMUM_RTT, MAXIMUM_RTT); + int rtt = SafeClamp(2 * rtt_, MINIMUM_RTT, MAXIMUM_RTT); if (RTC_LOG_CHECK_LEVEL(LS_VERBOSE)) { std::string pings; @@ -1104,14 +1099,13 @@ std::unique_ptr Connection::BuildPingRequest( // Adding ICE_CONTROLLED or ICE_CONTROLLING attribute based on the role. IceRole ice_role = port_->GetIceRole(); - RTC_DCHECK(ice_role == webrtc::ICEROLE_CONTROLLING || - ice_role == webrtc::ICEROLE_CONTROLLED); + RTC_DCHECK(ice_role == ICEROLE_CONTROLLING || ice_role == ICEROLE_CONTROLLED); message->AddAttribute(std::make_unique( - ice_role == webrtc::ICEROLE_CONTROLLING ? STUN_ATTR_ICE_CONTROLLING - : STUN_ATTR_ICE_CONTROLLED, + ice_role == ICEROLE_CONTROLLING ? STUN_ATTR_ICE_CONTROLLING + : STUN_ATTR_ICE_CONTROLLED, port_->IceTiebreaker())); - if (ice_role == webrtc::ICEROLE_CONTROLLING) { + if (ice_role == ICEROLE_CONTROLLING) { // We should have either USE_CANDIDATE attribute or ICE_NOMINATION // attribute but not both. That was enforced in p2ptransportchannel. if (use_candidate_attr()) { @@ -1181,7 +1175,7 @@ int64_t Connection::last_ping_received() const { void Connection::ReceivedPing(const std::optional& request_id) { RTC_DCHECK_RUN_ON(network_thread_); - last_ping_received_ = webrtc::TimeMillis(); + last_ping_received_ = TimeMillis(); last_ping_id_received_ = request_id; UpdateReceiving(last_ping_received_); } @@ -1203,7 +1197,7 @@ void Connection::HandlePiggybackCheckAcknowledgementIfAny(StunMessage* msg) { RTC_LOG_V(sev) << ToString() << ": Received piggyback STUN ping response, id=" << hex_encode(request_id); - const int64_t rtt = webrtc::TimeMillis() - iter->sent_time; + const int64_t rtt = TimeMillis() - iter->sent_time; ReceivedPingResponse(rtt, request_id, iter->nomination); } } @@ -1234,7 +1228,7 @@ void Connection::ReceivedPingResponse( acked_nomination_ = nomination.value(); } - int64_t now = webrtc::TimeMillis(); + int64_t now = TimeMillis(); total_round_trip_time_ms_ += rtt; current_round_trip_time_ms_ = static_cast(rtt); rtt_estimate_.AddSample(now, rtt); @@ -1299,7 +1293,7 @@ bool Connection::dead(int64_t now) const { // locally inactive (pruned) connection. This also allows the local agent to // ping with longer interval than 30s as long as it shorter than // `dead_connection_timeout_ms`. - if (now <= (last_received() + webrtc::DEAD_CONNECTION_RECEIVE_TIMEOUT)) { + if (now <= (last_received() + DEAD_CONNECTION_RECEIVE_TIMEOUT)) { // Not dead since we have received the last 30s. return false; } @@ -1307,7 +1301,7 @@ bool Connection::dead(int64_t now) const { // Outstanding pings: let it live until the ping is unreplied for // DEAD_CONNECTION_RECEIVE_TIMEOUT. return now > (pings_since_last_response_[0].sent_time + - webrtc::DEAD_CONNECTION_RECEIVE_TIMEOUT); + DEAD_CONNECTION_RECEIVE_TIMEOUT); } // No outstanding pings: let it live until @@ -1327,7 +1321,7 @@ bool Connection::dead(int64_t now) const { // keep it around for at least MIN_CONNECTION_LIFETIME to prevent connections // from being pruned too quickly during a network change event when two // networks would be up simultaneously but only for a brief period. - return now > (time_created_ms_ + webrtc::MIN_CONNECTION_LIFETIME); + return now > (time_created_ms_ + MIN_CONNECTION_LIFETIME); } int Connection::rtt() const { @@ -1344,7 +1338,7 @@ bool Connection::stable(int64_t now) const { } std::string Connection::ToDebugId() const { - return webrtc::ToHex(reinterpret_cast(this)); + return ToHex(reinterpret_cast(this)); } uint32_t Connection::ComputeNetworkCost() const { @@ -1682,9 +1676,9 @@ uint32_t Connection::prflx_priority() const { // (2^8)*(local preference) + // (2^0)*(256 - component ID) IcePriorityValue type_preference = - (local_candidate_.protocol() == webrtc::TCP_PROTOCOL_NAME) - ? webrtc::ICE_TYPE_PREFERENCE_PRFLX_TCP - : webrtc::ICE_TYPE_PREFERENCE_PRFLX; + (local_candidate_.protocol() == TCP_PROTOCOL_NAME) + ? ICE_TYPE_PREFERENCE_PRFLX_TCP + : ICE_TYPE_PREFERENCE_PRFLX; return type_preference << 24 | (local_candidate_.priority() & 0x00FFFFFF); } @@ -1872,7 +1866,7 @@ int ProxyConnection::Send(const void* data, stats_.sent_total_packets++; int sent = port_->SendTo(data, size, remote_candidate_.address(), options, true); - int64_t now = webrtc::TimeMillis(); + int64_t now = TimeMillis(); if (sent <= 0) { RTC_DCHECK(sent < 0); error_ = port_->GetError(); diff --git a/p2p/base/ice_controller_interface.cc b/p2p/base/ice_controller_interface.cc index 9c8ad0bfbd..b4360b9cd7 100644 --- a/p2p/base/ice_controller_interface.cc +++ b/p2p/base/ice_controller_interface.cc @@ -17,7 +17,7 @@ namespace webrtc { std::string IceRecheckEvent::ToString() const { - std::string str = webrtc::IceSwitchReasonToString(reason); + std::string str = IceSwitchReasonToString(reason); if (recheck_delay_ms) { str += " (after delay: " + std::to_string(recheck_delay_ms) + ")"; } diff --git a/p2p/base/ice_credentials_iterator.cc b/p2p/base/ice_credentials_iterator.cc index 1dc7d41762..d02464153d 100644 --- a/p2p/base/ice_credentials_iterator.cc +++ b/p2p/base/ice_credentials_iterator.cc @@ -25,9 +25,8 @@ IceCredentialsIterator::IceCredentialsIterator( IceCredentialsIterator::~IceCredentialsIterator() = default; IceParameters IceCredentialsIterator::CreateRandomIceCredentials() { - return IceParameters(webrtc::CreateRandomString(webrtc::ICE_UFRAG_LENGTH), - webrtc::CreateRandomString(webrtc::ICE_PWD_LENGTH), - false); + return IceParameters(CreateRandomString(ICE_UFRAG_LENGTH), + CreateRandomString(ICE_PWD_LENGTH), false); } IceParameters IceCredentialsIterator::GetIceCredentials() { diff --git a/p2p/base/ice_transport_internal.cc b/p2p/base/ice_transport_internal.cc index 601da19971..6b0011b8b3 100644 --- a/p2p/base/ice_transport_internal.cc +++ b/p2p/base/ice_transport_internal.cc @@ -54,9 +54,6 @@ ContinualGatheringPolicy GetContinualGatheringPolicy( } // namespace -using webrtc::RTCError; -using webrtc::RTCErrorType; - RTCError VerifyCandidate(const Candidate& cand) { // No address zero. if (cand.address().IsNil() || cand.address().IsAnyIP()) { @@ -245,7 +242,7 @@ void IceTransportInternal::SetRemoteIceCredentials(absl::string_view ice_ufrag, void IceTransportInternal::AddGatheringStateCallback( const void* removal_tag, - absl::AnyInvocable callback) { + absl::AnyInvocable callback) { gathering_state_callback_list_.AddReceiver(removal_tag, std::move(callback)); } void IceTransportInternal::RemoveGatheringStateCallback( diff --git a/p2p/base/p2p_transport_channel.cc b/p2p/base/p2p_transport_channel.cc index 4b1b5e7094..93ab452f50 100644 --- a/p2p/base/p2p_transport_channel.cc +++ b/p2p/base/p2p_transport_channel.cc @@ -181,13 +181,13 @@ P2PTransportChannel::P2PTransportChannel( network_thread_(Thread::Current()), incoming_only_(false), error_(0), - remote_ice_mode_(webrtc::ICEMODE_FULL), - ice_role_(webrtc::ICEROLE_UNKNOWN), - gathering_state_(webrtc::kIceGatheringNew), + remote_ice_mode_(ICEMODE_FULL), + ice_role_(ICEROLE_UNKNOWN), + gathering_state_(kIceGatheringNew), weak_ping_interval_(GetWeakPingIntervalInFieldTrial(field_trials)), config_(RECEIVING_TIMEOUT, BACKUP_CONNECTION_PING_INTERVAL, - webrtc::GATHER_ONCE /* continual_gathering_policy */, + GATHER_ONCE /* continual_gathering_policy */, false /* prioritize_most_likely_candidate_pairs */, STRONG_AND_STABLE_WRITABLE_CONNECTION_PING_INTERVAL, true /* presume_writable_when_fully_relayed */, @@ -299,7 +299,7 @@ void P2PTransportChannel::AddConnection(Connection* connection) { [this](const StunByteStringAttribute* delta) { return GoogDeltaReceived(delta); }, - [this](webrtc::RTCErrorOr delta_ack) { + [this](RTCErrorOr delta_ack) { GoogDeltaAckReceived(std::move(delta_ack)); }); if (!dtls_stun_piggyback_callbacks_.empty()) { @@ -587,7 +587,7 @@ void P2PTransportChannel::SetIceConfig(const IceConfig& config) { config_.surface_ice_candidates_on_ice_transport_type_changed = config.surface_ice_candidates_on_ice_transport_type_changed; if (config_.surface_ice_candidates_on_ice_transport_type_changed && - config_.continual_gathering_policy != webrtc::GATHER_CONTINUALLY) { + config_.continual_gathering_policy != GATHER_CONTINUALLY) { RTC_LOG(LS_WARNING) << "surface_ice_candidates_on_ice_transport_type_changed is " "ineffective since we do not gather continually."; @@ -847,8 +847,8 @@ void P2PTransportChannel::MaybeStartGathering() { IceCredentialsChanged(allocator_sessions_.back()->ice_ufrag(), allocator_sessions_.back()->ice_pwd(), ice_parameters_.ufrag, ice_parameters_.pwd)) { - if (gathering_state_ != webrtc::kIceGatheringGathering) { - gathering_state_ = webrtc::kIceGatheringGathering; + if (gathering_state_ != kIceGatheringGathering) { + gathering_state_ = kIceGatheringGathering; SendGatheringStateEvent(); } @@ -960,7 +960,7 @@ void P2PTransportChannel::OnCandidatesAllocationDone( "gathering so not changing gathering state."; return; } - gathering_state_ = webrtc::kIceGatheringComplete; + gathering_state_ = kIceGatheringComplete; RTC_LOG(LS_INFO) << "P2PTransportChannel: " << transport_name() << ", component " << component() << " gathering complete"; SendGatheringStateEvent(); @@ -1000,7 +1000,7 @@ void P2PTransportChannel::OnUnknownAddress(PortInterface* port, const Candidate* candidate = nullptr; for (const Candidate& c : remote_candidates_) { if (c.username() == remote_username && c.address() == address && - c.protocol() == webrtc::ProtoToString(proto)) { + c.protocol() == ProtoToString(proto)) { candidate = &c; break; } @@ -1052,13 +1052,12 @@ void P2PTransportChannel::OnUnknownAddress(PortInterface* port, // If the source transport address of the request does not match any // existing remote candidates, it represents a new peer reflexive remote // candidate. - remote_candidate = - Candidate(component(), webrtc::ProtoToString(proto), address, - remote_candidate_priority, remote_username, remote_password, - IceCandidateType::kPrflx, remote_generation, "", network_id, - network_cost); - if (proto == webrtc::PROTO_TCP) { - remote_candidate.set_tcptype(webrtc::TCPTYPE_ACTIVE_STR); + remote_candidate = Candidate( + component(), ProtoToString(proto), address, remote_candidate_priority, + remote_username, remote_password, IceCandidateType::kPrflx, + remote_generation, "", network_id, network_cost); + if (proto == PROTO_TCP) { + remote_candidate.set_tcptype(TCPTYPE_ACTIVE_STR); } // From RFC 5245, section-7.2.1.3: @@ -1148,7 +1147,7 @@ const IceParameters* P2PTransportChannel::FindRemoteIceFromUfrag( void P2PTransportChannel::OnNominated(Connection* conn) { RTC_DCHECK_RUN_ON(network_thread_); - RTC_DCHECK(ice_role_ == webrtc::ICEROLE_CONTROLLED); + RTC_DCHECK(ice_role_ == ICEROLE_CONTROLLED); if (selected_connection_ == conn) { return; @@ -1228,10 +1227,8 @@ void P2PTransportChannel::AddRemoteCandidate(const Candidate& candidate) { if (new_remote_candidate.address().IsUnresolvedIP()) { // Don't do DNS lookups if the IceTransportPolicy is "none" or "relay". - bool sharing_host = - ((allocator_->candidate_filter() & webrtc::CF_HOST) != 0); - bool sharing_stun = - ((allocator_->candidate_filter() & webrtc::CF_REFLEXIVE) != 0); + bool sharing_host = ((allocator_->candidate_filter() & CF_HOST) != 0); + bool sharing_stun = ((allocator_->candidate_filter() & CF_REFLEXIVE) != 0); if (sharing_host || sharing_stun) { ResolveHostnameCandidate(new_remote_candidate); } @@ -1401,7 +1398,7 @@ bool P2PTransportChannel::CreateConnection(PortInterface* port, (port_type == IceCandidateType::kRelay || remote_candidate.is_relay())) { RTC_LOG(LS_INFO) << ToString() << ": skip creating connection " - << webrtc::IceCandidateTypeToString(port_type) << " to " + << IceCandidateTypeToString(port_type) << " to " << remote_candidate.type_name(); return false; } @@ -1632,7 +1629,7 @@ DiffServCodePoint P2PTransportChannel::DefaultDscpValue() const { RTC_DCHECK_RUN_ON(network_thread_); OptionMap::const_iterator it = options_.find(Socket::OPT_DSCP); if (it == options_.end()) { - return webrtc::DSCP_NO_CHANGE; + return DSCP_NO_CHANGE; } return static_cast(it->second); } @@ -1656,7 +1653,7 @@ void P2PTransportChannel::RemoveConnectionForTest(Connection* connection) { // Monitor connection states. void P2PTransportChannel::UpdateConnectionStates() { RTC_DCHECK_RUN_ON(network_thread_); - int64_t now = webrtc::TimeMillis(); + int64_t now = TimeMillis(); // We need to copy the list of connections since some may delete themselves // when we call UpdateState. @@ -1720,7 +1717,7 @@ void P2PTransportChannel::UpdateState() { bool P2PTransportChannel::AllowedToPruneConnections() const { RTC_DCHECK_RUN_ON(network_thread_); - return ice_role_ == webrtc::ICEROLE_CONTROLLING || + return ice_role_ == ICEROLE_CONTROLLING || (selected_connection_ && selected_connection_->nominated()); } @@ -1751,7 +1748,7 @@ NetworkRoute P2PTransportChannel::ConfigureNetworkRoute( .last_sent_packet_id = last_sent_packet_id_, .packet_overhead = conn->local_candidate().address().ipaddr().overhead() + - webrtc::GetProtocolOverhead(conn->local_candidate().protocol())}; + GetProtocolOverhead(conn->local_candidate().protocol())}; } void P2PTransportChannel::SwitchSelectedConnection( @@ -1800,7 +1797,7 @@ void P2PTransportChannel::SwitchSelectedConnectionInternal( RTC_LOG(LS_INFO) << ToString() << ": No selected connection"; } - if (conn != nullptr && ice_role_ == webrtc::ICEROLE_CONTROLLING && + if (conn != nullptr && ice_role_ == ICEROLE_CONTROLLING && ((ice_field_trials_.send_ping_on_switch_ice_controlling && old_selected_connection != nullptr) || ice_field_trials_.send_ping_on_selected_ice_controlling)) { @@ -1819,7 +1816,7 @@ void P2PTransportChannel::SwitchSelectedConnectionInternal( if (old_selected_connection) { pair_change.estimated_disconnected_time_ms = - ComputeEstimatedDisconnectedTimeMs(webrtc::TimeMillis(), + ComputeEstimatedDisconnectedTimeMs(TimeMillis(), old_selected_connection); } else { pair_change.estimated_disconnected_time_ms = 0; @@ -2020,7 +2017,7 @@ void P2PTransportChannel::PingConnection(Connection* conn) { RTC_DCHECK_RUN_ON(network_thread_); bool use_candidate_attr = false; uint32_t nomination = 0; - if (ice_role_ == webrtc::ICEROLE_CONTROLLING) { + if (ice_role_ == ICEROLE_CONTROLLING) { bool renomination_supported = ice_parameters_.renomination && !remote_ice_parameters_.empty() && remote_ice_parameters_.back().renomination; @@ -2032,7 +2029,7 @@ void P2PTransportChannel::PingConnection(Connection* conn) { } conn->set_nomination(nomination); conn->set_use_candidate_attr(use_candidate_attr); - last_ping_sent_ms_ = webrtc::TimeMillis(); + last_ping_sent_ms_ = TimeMillis(); conn->Ping(last_ping_sent_ms_, stun_dict_writer_.CreateDelta()); } @@ -2202,8 +2199,7 @@ void P2PTransportChannel::OnReadPacket(Connection* connection, // May need to switch the sending connection based on the receiving media // path if this is the controlled side. - if (ice_role_ == webrtc::ICEROLE_CONTROLLED && - connection != selected_connection_) { + if (ice_role_ == ICEROLE_CONTROLLED && connection != selected_connection_) { ice_controller_->OnImmediateSwitchRequest(IceSwitchReason::DATA_RECEIVED, connection); } diff --git a/p2p/base/packet_transport_internal.cc b/p2p/base/packet_transport_internal.cc index 5f8017d8ab..7670f01d72 100644 --- a/p2p/base/packet_transport_internal.cc +++ b/p2p/base/packet_transport_internal.cc @@ -37,8 +37,8 @@ std::optional PacketTransportInternal::network_route() const { void PacketTransportInternal::RegisterReceivedPacketCallback( void* id, - absl::AnyInvocable callback) { + absl::AnyInvocable + callback) { RTC_DCHECK_RUN_ON(&network_checker_); received_packet_callback_list_.AddReceiver(id, std::move(callback)); } diff --git a/p2p/base/port.cc b/p2p/base/port.cc index 5c78a04092..e7feb8eecf 100644 --- a/p2p/base/port.cc +++ b/p2p/base/port.cc @@ -76,20 +76,20 @@ PacketInfoProtocolType ConvertProtocolTypeToPacketInfoProtocolType( // The delay before we begin checking if this port is useless. We set // it to a little higher than a total STUN timeout. -const int kPortTimeoutDelay = webrtc::STUN_TOTAL_TIMEOUT + 5000; +const int kPortTimeoutDelay = STUN_TOTAL_TIMEOUT + 5000; } // namespace -static const char* const PROTO_NAMES[] = { - webrtc::UDP_PROTOCOL_NAME, webrtc::TCP_PROTOCOL_NAME, - webrtc::SSLTCP_PROTOCOL_NAME, webrtc::TLS_PROTOCOL_NAME}; +static const char* const PROTO_NAMES[] = {UDP_PROTOCOL_NAME, TCP_PROTOCOL_NAME, + SSLTCP_PROTOCOL_NAME, + TLS_PROTOCOL_NAME}; const char* ProtoToString(ProtocolType proto) { return PROTO_NAMES[proto]; } std::optional StringToProto(absl::string_view proto_name) { - for (size_t i = 0; i <= webrtc::PROTO_LAST; ++i) { + for (size_t i = 0; i <= PROTO_LAST; ++i) { if (absl::EqualsIgnoreCase(PROTO_NAMES[i], proto_name)) { return static_cast(i); } @@ -125,7 +125,7 @@ Port::Port(const PortParametersRef& args, password_(args.ice_password), timeout_delay_(kPortTimeoutDelay), enable_port_packets_(false), - ice_role_(webrtc::ICEROLE_UNKNOWN), + ice_role_(ICEROLE_UNKNOWN), tiebreaker_(0), shared_socket_(shared_socket), network_cost_(args.network->GetCost(env_.field_trials())), @@ -137,8 +137,8 @@ Port::Port(const PortParametersRef& args, // we should just create one. if (ice_username_fragment_.empty()) { RTC_DCHECK(password_.empty()); - ice_username_fragment_ = webrtc::CreateRandomString(ICE_UFRAG_LENGTH); - password_ = webrtc::CreateRandomString(ICE_PWD_LENGTH); + ice_username_fragment_ = CreateRandomString(ICE_UFRAG_LENGTH); + password_ = CreateRandomString(ICE_PWD_LENGTH); } network_->SignalTypeChanged.connect(this, &Port::OnNetworkTypeChanged); @@ -347,7 +347,7 @@ void Port::OnReadPacket(const ReceivedIpPacket& packet, ProtocolType proto) { // STUN message handled already } else if (msg->type() == STUN_BINDING_REQUEST) { RTC_LOG(LS_INFO) << "Received " << StunMethodToString(msg->type()) - << " id=" << webrtc::hex_encode(msg->transaction_id()) + << " id=" << hex_encode(msg->transaction_id()) << " from unknown address " << addr.ToSensitiveString(); // We need to signal an unknown address before we handle any role conflict // below. Otherwise there would be no candidate pair and TURN entry created @@ -569,7 +569,7 @@ bool Port::IsCompatibleAddress(const SocketAddress& addr) { } // Link-local IPv6 ports can only connect to other link-local IPv6 ports. if (ip.family() == AF_INET6 && - (webrtc::IPIsLinkLocal(ip) != webrtc::IPIsLinkLocal(addr.ipaddr()))) { + (IPIsLinkLocal(ip) != IPIsLinkLocal(addr.ipaddr()))) { return false; } return true; @@ -577,7 +577,7 @@ bool Port::IsCompatibleAddress(const SocketAddress& addr) { DiffServCodePoint Port::StunDscpValue() const { // By default, inherit from whatever the MediaChannel sends. - return webrtc::DSCP_NO_CHANGE; + return DSCP_NO_CHANGE; } void Port::DestroyAllConnections() { @@ -630,12 +630,12 @@ bool Port::MaybeIceRoleConflict(const SocketAddress& addr, RTC_DCHECK_RUN_ON(thread_); // Validate ICE_CONTROLLING or ICE_CONTROLLED attributes. bool ret = true; - IceRole remote_ice_role = webrtc::ICEROLE_UNKNOWN; + IceRole remote_ice_role = ICEROLE_UNKNOWN; uint64_t remote_tiebreaker = 0; const StunUInt64Attribute* stun_attr = stun_msg->GetUInt64(STUN_ATTR_ICE_CONTROLLING); if (stun_attr) { - remote_ice_role = webrtc::ICEROLE_CONTROLLING; + remote_ice_role = ICEROLE_CONTROLLING; remote_tiebreaker = stun_attr->value(); } @@ -643,7 +643,7 @@ bool Port::MaybeIceRoleConflict(const SocketAddress& addr, // tie breaker value received in the ping message matches port // tiebreaker value this must be a loopback call. // We will treat this as valid scenario. - if (remote_ice_role == webrtc::ICEROLE_CONTROLLING && + if (remote_ice_role == ICEROLE_CONTROLLING && username_fragment() == remote_ufrag && remote_tiebreaker == IceTiebreaker()) { return true; @@ -651,13 +651,13 @@ bool Port::MaybeIceRoleConflict(const SocketAddress& addr, stun_attr = stun_msg->GetUInt64(STUN_ATTR_ICE_CONTROLLED); if (stun_attr) { - remote_ice_role = webrtc::ICEROLE_CONTROLLED; + remote_ice_role = ICEROLE_CONTROLLED; remote_tiebreaker = stun_attr->value(); } switch (ice_role_) { - case webrtc::ICEROLE_CONTROLLING: - if (webrtc::ICEROLE_CONTROLLING == remote_ice_role) { + case ICEROLE_CONTROLLING: + if (ICEROLE_CONTROLLING == remote_ice_role) { if (remote_tiebreaker >= tiebreaker_) { SignalRoleConflict(this); } else { @@ -668,8 +668,8 @@ bool Port::MaybeIceRoleConflict(const SocketAddress& addr, } } break; - case webrtc::ICEROLE_CONTROLLED: - if (webrtc::ICEROLE_CONTROLLED == remote_ice_role) { + case ICEROLE_CONTROLLED: + if (ICEROLE_CONTROLLED == remote_ice_role) { if (remote_tiebreaker < tiebreaker_) { SignalRoleConflict(this); } else { @@ -824,10 +824,10 @@ void Port::PostDestroyIfDead(bool delayed) { void Port::DestroyIfDead() { RTC_DCHECK_RUN_ON(thread_); - bool dead = (state_ == State::INIT || state_ == State::PRUNED) && - connections_.empty() && - webrtc::TimeMillis() - last_time_all_connections_removed_ >= - timeout_delay_; + bool dead = + (state_ == State::INIT || state_ == State::PRUNED) && + connections_.empty() && + TimeMillis() - last_time_all_connections_removed_ >= timeout_delay_; if (dead) { Destroy(); } @@ -849,10 +849,9 @@ void Port::OnNetworkTypeChanged(const ::webrtc::Network* network) { std::string Port::ToString() const { StringBuilder ss; - ss << "Port[" << webrtc::ToHex(reinterpret_cast(this)) << ":" + ss << "Port[" << ToHex(reinterpret_cast(this)) << ":" << content_name_ << ":" << component_ << ":" << generation_ << ":" - << webrtc::IceCandidateTypeToString(type_) << ":" << network_->ToString() - << "]"; + << IceCandidateTypeToString(type_) << ":" << network_->ToString() << "]"; return ss.Release(); } @@ -897,7 +896,7 @@ bool Port::OnConnectionDestroyed(Connection* conn) { // fails and is removed before kPortTimeoutDelay, then this message will // not cause the Port to be destroyed. if (connections_.empty()) { - last_time_all_connections_removed_ = webrtc::TimeMillis(); + last_time_all_connections_removed_ = TimeMillis(); PostDestroyIfDead(/*delayed=*/true); } diff --git a/p2p/base/port_allocator.cc b/p2p/base/port_allocator.cc index c7a40d9074..0e7c1ad1df 100644 --- a/p2p/base/port_allocator.cc +++ b/p2p/base/port_allocator.cc @@ -58,12 +58,11 @@ RelayServerConfig::RelayServerConfig(absl::string_view address, absl::string_view password, ProtocolType proto, bool secure) - : RelayServerConfig( - address, - port, - username, - password, - (proto == webrtc::PROTO_TCP && secure ? webrtc::PROTO_TLS : proto)) {} + : RelayServerConfig(address, + port, + username, + password, + (proto == PROTO_TCP && secure ? PROTO_TLS : proto)) {} RelayServerConfig::RelayServerConfig(const RelayServerConfig&) = default; @@ -138,7 +137,7 @@ bool PortAllocator::SetConfiguration( TurnCustomizer* turn_customizer, const std::optional& stun_candidate_keepalive_interval) { PortPrunePolicy turn_port_prune_policy = - prune_turn_ports ? webrtc::PRUNE_BASED_ON_PRIORITY : webrtc::NO_PRUNE; + prune_turn_ports ? PRUNE_BASED_ON_PRIORITY : NO_PRUNE; return SetConfiguration(stun_servers, turn_servers, candidate_pool_size, turn_port_prune_policy, turn_customizer, stun_candidate_keepalive_interval); diff --git a/p2p/base/pseudo_tcp.cc b/p2p/base/pseudo_tcp.cc index 6e2e1e5eac..bef1a51dbf 100644 --- a/p2p/base/pseudo_tcp.cc +++ b/p2p/base/pseudo_tcp.cc @@ -144,19 +144,19 @@ const uint32_t IDLE_TIMEOUT = 90 * 1000; // 90 seconds; ////////////////////////////////////////////////////////////////////// inline void long_to_bytes(uint32_t val, void* buf) { - *static_cast(buf) = webrtc::HostToNetwork32(val); + *static_cast(buf) = HostToNetwork32(val); } inline void short_to_bytes(uint16_t val, void* buf) { - *static_cast(buf) = webrtc::HostToNetwork16(val); + *static_cast(buf) = HostToNetwork16(val); } inline uint32_t bytes_to_long(const void* buf) { - return webrtc::NetworkToHost32(*static_cast(buf)); + return NetworkToHost32(*static_cast(buf)); } inline uint16_t bytes_to_short(const void* buf) { - return webrtc::NetworkToHost16(*static_cast(buf)); + return NetworkToHost16(*static_cast(buf)); } ////////////////////////////////////////////////////////////////////// @@ -205,7 +205,7 @@ uint32_t PseudoTcp::Now() { #if 0 // Use this to synchronize timers with logging timestamps (easier debug) return static_cast(webrtc::TimeSince(StartTime())); #else - return webrtc::Time32(); + return Time32(); #endif } @@ -288,7 +288,7 @@ void PseudoTcp::NotifyClock(uint32_t now) { return; // Check if it's time to retransmit a segment - if (m_rto_base && (webrtc::TimeDiff32(m_rto_base + m_rx_rto, now) <= 0)) { + if (m_rto_base && (TimeDiff32(m_rto_base + m_rx_rto, now) <= 0)) { if (m_slist.empty()) { RTC_DCHECK_NOTREACHED(); } else { @@ -319,9 +319,8 @@ void PseudoTcp::NotifyClock(uint32_t now) { } // Check if it's time to probe closed windows - if ((m_snd_wnd == 0) && - (webrtc::TimeDiff32(m_lastsend + m_rx_rto, now) <= 0)) { - if (webrtc::TimeDiff32(now, m_lastrecv) >= 15000) { + if ((m_snd_wnd == 0) && (TimeDiff32(m_lastsend + m_rx_rto, now) <= 0)) { + if (TimeDiff32(now, m_lastrecv) >= 15000) { closedown(ECONNABORTED); return; } @@ -335,7 +334,7 @@ void PseudoTcp::NotifyClock(uint32_t now) { } // Check if it's time to send delayed acks - if (m_t_ack && (webrtc::TimeDiff32(m_t_ack + m_ack_delay, now) <= 0)) { + if (m_t_ack && (TimeDiff32(m_t_ack + m_ack_delay, now) <= 0)) { packet(m_snd_nxt, 0, 0, 0); } @@ -612,16 +611,16 @@ bool PseudoTcp::clock_check(uint32_t now, long& nTimeout) { nTimeout = DEFAULT_TIMEOUT; if (m_t_ack) { - nTimeout = std::min( - nTimeout, webrtc::TimeDiff32(m_t_ack + m_ack_delay, now)); + nTimeout = + std::min(nTimeout, TimeDiff32(m_t_ack + m_ack_delay, now)); } if (m_rto_base) { - nTimeout = std::min( - nTimeout, webrtc::TimeDiff32(m_rto_base + m_rx_rto, now)); + nTimeout = + std::min(nTimeout, TimeDiff32(m_rto_base + m_rx_rto, now)); } if (m_snd_wnd == 0) { - nTimeout = std::min( - nTimeout, webrtc::TimeDiff32(m_lastsend + m_rx_rto, now)); + nTimeout = + std::min(nTimeout, TimeDiff32(m_lastsend + m_rx_rto, now)); } #if PSEUDO_KEEPALIVE if (m_state == TCP_ESTABLISHED) { @@ -703,7 +702,7 @@ bool PseudoTcp::process(Segment& seg) { if ((seg.ack > m_snd_una) && (seg.ack <= m_snd_nxt)) { // Calculate round-trip time if (seg.tsecr) { - int32_t rtt = webrtc::TimeDiff32(now, seg.tsecr); + int32_t rtt = TimeDiff32(now, seg.tsecr); if (rtt >= 0) { if (m_rx_srtt == 0) { m_rx_srtt = rtt; @@ -716,8 +715,8 @@ bool PseudoTcp::process(Segment& seg) { m_rx_rttvar = (3 * m_rx_rttvar + abs_err) / 4; m_rx_srtt = (7 * m_rx_srtt + rtt) / 8; } - m_rx_rto = webrtc::SafeClamp( - m_rx_srtt + webrtc::SafeMax(1, 4 * m_rx_rttvar), MIN_RTO, MAX_RTO); + m_rx_rto = SafeClamp(m_rx_srtt + SafeMax(1, 4 * m_rx_rttvar), MIN_RTO, + MAX_RTO); #if _DEBUGMSG >= _DBG_VERBOSE RTC_LOG(LS_INFO) << "rtt: " << rtt << " srtt: " << m_rx_srtt << " rto: " << m_rx_rto; @@ -1049,7 +1048,7 @@ bool PseudoTcp::transmit(const SList::iterator& seg, uint32_t now) { void PseudoTcp::attemptSend(SendFlags sflags) { uint32_t now = Now(); - if (webrtc::TimeDiff32(now, m_lastsend) > static_cast(m_rx_rto)) { + if (TimeDiff32(now, m_lastsend) > static_cast(m_rx_rto)) { m_cwnd = m_mss; } diff --git a/p2p/base/regathering_controller_unittest.cc b/p2p/base/regathering_controller_unittest.cc index 9b1c216db4..33ead46919 100644 --- a/p2p/base/regathering_controller_unittest.cc +++ b/p2p/base/regathering_controller_unittest.cc @@ -75,7 +75,7 @@ class RegatheringControllerTest : public ::testing::Test, std::vector turn_servers(1, turn_server); allocator_->set_flags(kOnlyLocalPorts); allocator_->SetConfiguration(stun_servers, turn_servers, 0 /* pool size */, - webrtc::NO_PRUNE); + NO_PRUNE); allocator_session_ = allocator_->CreateSession( "test", ICE_CANDIDATE_COMPONENT_RTP, kIceUfrag, kIcePwd); // The gathering will take place on the current thread and the following diff --git a/p2p/base/stun_port.cc b/p2p/base/stun_port.cc index 63105bec4b..f5df2d830c 100644 --- a/p2p/base/stun_port.cc +++ b/p2p/base/stun_port.cc @@ -83,7 +83,7 @@ class StunBindingRequest : public StunRequest { } // The keep-alive requests will be stopped after its lifetime has passed. - if (WithinLifetime(webrtc::TimeMillis())) { + if (WithinLifetime(TimeMillis())) { port_->request_manager_.SendDelayed( new StunBindingRequest(port_, server_addr_, start_time_), port_->stun_keepalive_delay()); @@ -106,9 +106,8 @@ class StunBindingRequest : public StunRequest { attr ? attr->reason() : "STUN binding response with no error code attribute."); - int64_t now = webrtc::TimeMillis(); - if (WithinLifetime(now) && - webrtc::TimeDiff(now, start_time_) < RETRY_TIMEOUT) { + int64_t now = TimeMillis(); + if (WithinLifetime(now) && TimeDiff(now, start_time_) < RETRY_TIMEOUT) { port_->request_manager_.SendDelayed( new StunBindingRequest(port_, server_addr_, start_time_), port_->stun_keepalive_delay()); @@ -128,7 +127,7 @@ class StunBindingRequest : public StunRequest { // lifetime means infinite). bool WithinLifetime(int64_t now) const { int lifetime = port_->stun_keepalive_lifetime(); - return lifetime < 0 || webrtc::TimeDiff(now, start_time_) <= lifetime; + return lifetime < 0 || TimeDiff(now, start_time_) <= lifetime; } UDPPort* port_; @@ -188,7 +187,7 @@ UDPPort::UDPPort(const PortParametersRef& args, error_(0), ready_(false), stun_keepalive_delay_(STUN_KEEPALIVE_INTERVAL), - dscp_(webrtc::DSCP_NO_CHANGE), + dscp_(DSCP_NO_CHANGE), emit_local_for_anyaddress_(emit_local_for_anyaddress) {} UDPPort::UDPPort(const PortParametersRef& args, @@ -206,7 +205,7 @@ UDPPort::UDPPort(const PortParametersRef& args, error_(0), ready_(false), stun_keepalive_delay_(STUN_KEEPALIVE_INTERVAL), - dscp_(webrtc::DSCP_NO_CHANGE), + dscp_(DSCP_NO_CHANGE), emit_local_for_anyaddress_(emit_local_for_anyaddress) {} bool UDPPort::Init() { @@ -349,11 +348,11 @@ bool UDPPort::HandleIncomingPacket(AsyncPacketSocket* socket, } bool UDPPort::SupportsProtocol(absl::string_view protocol) const { - return protocol == webrtc::UDP_PROTOCOL_NAME; + return protocol == UDP_PROTOCOL_NAME; } ProtocolType UDPPort::GetProtocol() const { - return webrtc::PROTO_UDP; + return PROTO_UDP; } void UDPPort::GetStunStats(std::optional* stats) { @@ -376,7 +375,7 @@ void UDPPort::OnLocalAddressReady(AsyncPacketSocket* /* socket */, // least the port is listening. MaybeSetDefaultLocalAddress(&addr); - AddAddress(addr, addr, SocketAddress(), webrtc::UDP_PROTOCOL_NAME, "", "", + AddAddress(addr, addr, SocketAddress(), UDP_PROTOCOL_NAME, "", "", IceCandidateType::kHost, ICE_TYPE_PREFERENCE_HOST, 0, "", false); MaybePrepareStunCandidate(); } @@ -405,7 +404,7 @@ void UDPPort::OnReadPacket(AsyncPacketSocket* socket, if (Connection* conn = GetConnection(packet.source_address())) { conn->OnReadPacket(packet); } else { - Port::OnReadPacket(packet, webrtc::PROTO_UDP); + Port::OnReadPacket(packet, PROTO_UDP); } } @@ -477,7 +476,7 @@ void UDPPort::SendStunBindingRequest(const SocketAddress& stun_addr) { // Check if `server_addr_` is compatible with the port's ip. if (IsCompatibleAddress(stun_addr)) { request_manager_.Send( - new StunBindingRequest(this, stun_addr, webrtc::TimeMillis())); + new StunBindingRequest(this, stun_addr, TimeMillis())); } else { // Since we can't send stun messages to the server, we should mark this // port ready. This is not an error but similar to ignoring @@ -532,15 +531,14 @@ void UDPPort::OnStunBindingRequestSucceeded( SocketAddress related_address = socket_->GetLocalAddress(); // If we can't stamp the related address correctly, empty it to avoid leak. if (!MaybeSetDefaultLocalAddress(&related_address)) { - related_address = - webrtc::EmptySocketAddressWithFamily(related_address.family()); + related_address = EmptySocketAddressWithFamily(related_address.family()); } StringBuilder url; url << "stun:" << stun_server_addr.hostname() << ":" << stun_server_addr.port(); AddAddress(stun_reflected_addr, socket_->GetLocalAddress(), related_address, - webrtc::UDP_PROTOCOL_NAME, "", "", IceCandidateType::kSrflx, + UDP_PROTOCOL_NAME, "", "", IceCandidateType::kSrflx, ICE_TYPE_PREFERENCE_SRFLX, 0, url.str(), false); } MaybeSetPortCompleteOrError(); diff --git a/p2p/base/stun_request.cc b/p2p/base/stun_request.cc index 952590e471..1ff8389ff1 100644 --- a/p2p/base/stun_request.cc +++ b/p2p/base/stun_request.cc @@ -33,7 +33,6 @@ #include "rtc_base/time_utils.h" // For TimeMillis namespace webrtc { -using ::webrtc::SafeTask; // RFC 5389 says SHOULD be 500ms. // For years, this was 100ms, but for networks that @@ -54,7 +53,7 @@ const int STUN_MAX_RTO = 8000; // milliseconds, or 5 doublings StunRequestManager::StunRequestManager( TaskQueueBase* thread, - std::function send_packet) + std::function send_packet) : thread_(thread), send_packet_(std::move(send_packet)) {} StunRequestManager::~StunRequestManager() = default; @@ -215,8 +214,7 @@ bool StunRequestManager::CheckResponse(const char* data, size_t size) { MakeArrayView(reinterpret_cast(data), size)); std::unique_ptr response(iter->second->msg_->CreateNew()); if (!response->Read(&buf)) { - RTC_LOG(LS_WARNING) << "Failed to read STUN response " - << webrtc::hex_encode(id); + RTC_LOG(LS_WARNING) << "Failed to read STUN response " << hex_encode(id); return false; } @@ -268,7 +266,7 @@ const StunMessage* StunRequest::msg() const { int StunRequest::Elapsed() const { RTC_DCHECK_RUN_ON(network_thread()); - return static_cast(webrtc::TimeMillis() - tstamp_); + return static_cast(TimeMillis() - tstamp_); } void StunRequest::SendInternal() { @@ -279,7 +277,7 @@ void StunRequest::SendInternal() { return; } - tstamp_ = webrtc::TimeMillis(); + tstamp_ = TimeMillis(); ByteBufferWriter buf; msg_->Write(&buf); diff --git a/p2p/base/stun_request_unittest.cc b/p2p/base/stun_request_unittest.cc index f1745d490a..14f497ab5c 100644 --- a/p2p/base/stun_request_unittest.cc +++ b/p2p/base/stun_request_unittest.cc @@ -156,15 +156,14 @@ TEST_F(StunRequestTest, TestBackoff) { std::unique_ptr res = request->CreateResponseMessage(STUN_BINDING_RESPONSE); - int64_t start = webrtc::TimeMillis(); + int64_t start = TimeMillis(); manager_.Send(request); for (int i = 0; i < 9; ++i) { - EXPECT_THAT( - webrtc::WaitUntil([&] { return request_count_; }, Ne(i), + EXPECT_THAT(WaitUntil([&] { return request_count_; }, Ne(i), {.timeout = TimeDelta::Millis(STUN_TOTAL_TIMEOUT), .clock = &fake_clock}), - webrtc::IsRtcOk()); - int64_t elapsed = webrtc::TimeMillis() - start; + IsRtcOk()); + int64_t elapsed = TimeMillis() - start; RTC_DLOG(LS_INFO) << "STUN request #" << (i + 1) << " sent at " << elapsed << " ms"; EXPECT_EQ(TotalDelay(i), elapsed); diff --git a/p2p/base/tcp_port.cc b/p2p/base/tcp_port.cc index 7b3b9fddbf..5717f0a2d0 100644 --- a/p2p/base/tcp_port.cc +++ b/p2p/base/tcp_port.cc @@ -102,9 +102,6 @@ #include "rtc_base/weak_ptr.h" namespace webrtc { -using ::webrtc::IceCandidateType; -using ::webrtc::SafeTask; -using ::webrtc::TimeDelta; TCPPort::TCPPort(const PortParametersRef& args, uint16_t min_port, @@ -150,7 +147,7 @@ Connection* TCPPort::CreateConnection(const Candidate& address, return NULL; // We don't know how to act as an ssl server yet - if ((address.protocol() == webrtc::SSLTCP_PROTOCOL_NAME) && + if ((address.protocol() == SSLTCP_PROTOCOL_NAME) && (origin == ORIGIN_THIS_PORT)) { return NULL; } @@ -182,7 +179,7 @@ void TCPPort::PrepareAddress() { << static_cast(listen_socket_->GetState()); AddAddress( listen_socket_->GetLocalAddress(), listen_socket_->GetLocalAddress(), - SocketAddress(), webrtc::TCP_PROTOCOL_NAME, "", TCPTYPE_PASSIVE_STR, + SocketAddress(), TCP_PROTOCOL_NAME, "", TCPTYPE_PASSIVE_STR, IceCandidateType::kHost, ICE_TYPE_PREFERENCE_HOST_TCP, 0, "", true); } else { RTC_LOG(LS_INFO) << ToString() @@ -197,7 +194,7 @@ void TCPPort::PrepareAddress() { // see what IP we get. But that may be overkill. AddAddress(SocketAddress(Network()->GetBestIP(), DISCARD_PORT), SocketAddress(Network()->GetBestIP(), 0), SocketAddress(), - webrtc::TCP_PROTOCOL_NAME, "", TCPTYPE_ACTIVE_STR, + TCP_PROTOCOL_NAME, "", TCPTYPE_ACTIVE_STR, IceCandidateType::kHost, ICE_TYPE_PREFERENCE_HOST_TCP, 0, "", true); } @@ -272,12 +269,11 @@ int TCPPort::GetError() { } bool TCPPort::SupportsProtocol(absl::string_view protocol) const { - return protocol == webrtc::TCP_PROTOCOL_NAME || - protocol == webrtc::SSLTCP_PROTOCOL_NAME; + return protocol == TCP_PROTOCOL_NAME || protocol == SSLTCP_PROTOCOL_NAME; } ProtocolType TCPPort::GetProtocol() const { - return webrtc::PROTO_TCP; + return PROTO_TCP; } void TCPPort::OnNewConnection(AsyncListenSocket* socket, @@ -332,7 +328,7 @@ AsyncPacketSocket* TCPPort::GetIncoming(const SocketAddress& addr, void TCPPort::OnReadPacket(AsyncPacketSocket* socket, const ReceivedIpPacket& packet) { - Port::OnReadPacket(packet, webrtc::PROTO_TCP); + Port::OnReadPacket(packet, PROTO_TCP); } void TCPPort::OnSentPacket(AsyncPacketSocket* socket, @@ -360,7 +356,7 @@ TCPConnection::TCPConnection(WeakPtr tcp_port, reconnection_timeout_(CONNECTION_WRITE_CONNECT_TIMEOUT) { RTC_DCHECK_RUN_ON(network_thread_); RTC_DCHECK_EQ(port()->GetProtocol(), - webrtc::PROTO_TCP); // Needs to be TCPPort. + PROTO_TCP); // Needs to be TCPPort. SignalDestroyed.connect(this, &TCPConnection::OnDestroyed); @@ -413,7 +409,7 @@ int TCPConnection::Send(const void* data, tcp_port()->CopyPortInformationToPacketInfo( &modified_options.info_signaled_after_sent); int sent = socket_->Send(data, size, modified_options); - int64_t now = webrtc::TimeMillis(); + int64_t now = TimeMillis(); if (sent < 0) { stats_.sent_discarded_packets++; error_ = socket_->GetError(); @@ -487,7 +483,7 @@ void TCPConnection::OnConnect(AsyncPacketSocket* socket) { << ", rather than an address associated with network:" << port_->Network()->ToString() << ". Still allowing it since it's localhost."; - } else if (webrtc::IPIsAny(port_->Network()->GetBestIP())) { + } else if (IPIsAny(port_->Network()->GetBestIP())) { RTC_LOG(LS_WARNING) << "Socket is bound to the address:" << socket_address.ipaddr().ToSensitiveString() @@ -592,7 +588,7 @@ void TCPConnection::OnDestroyed(Connection* c) { void TCPConnection::CreateOutgoingTcpSocket() { RTC_DCHECK(outgoing_); - int opts = (remote_candidate().protocol() == webrtc::SSLTCP_PROTOCOL_NAME) + int opts = (remote_candidate().protocol() == SSLTCP_PROTOCOL_NAME) ? PacketSocketFactory::OPT_TLS_FAKE : 0; diff --git a/p2p/base/turn_port.cc b/p2p/base/turn_port.cc index e0ef1c8384..9395fe5ee9 100644 --- a/p2p/base/turn_port.cc +++ b/p2p/base/turn_port.cc @@ -87,12 +87,12 @@ inline bool IsTurnChannelData(uint16_t msg_type) { static int GetRelayPreference(ProtocolType proto) { switch (proto) { - case webrtc::PROTO_TCP: + case PROTO_TCP: return ICE_TYPE_PREFERENCE_RELAY_TCP; - case webrtc::PROTO_TLS: + case PROTO_TLS: return ICE_TYPE_PREFERENCE_RELAY_TLS; default: - RTC_DCHECK(proto == webrtc::PROTO_UDP); + RTC_DCHECK(proto == PROTO_UDP); return ICE_TYPE_PREFERENCE_RELAY_UDP; } } @@ -240,7 +240,7 @@ TurnPort::TurnPort(const PortParametersRef& args, credentials_(credentials), socket_(socket), error_(0), - stun_dscp_value_(webrtc::DSCP_NO_CHANGE), + stun_dscp_value_(DSCP_NO_CHANGE), request_manager_( args.network_thread, [this](const void* data, size_t size, StunRequest* request) { @@ -271,7 +271,7 @@ TurnPort::TurnPort(const PortParametersRef& args, credentials_(credentials), socket_(nullptr), error_(0), - stun_dscp_value_(webrtc::DSCP_NO_CHANGE), + stun_dscp_value_(DSCP_NO_CHANGE), request_manager_( args.network_thread, [this](const void* data, size_t size, StunRequest* request) { @@ -397,7 +397,7 @@ void TurnPort::PrepareAddress() { "Failed to create TURN client socket."); return; } - if (server_address_.proto == webrtc::PROTO_UDP) { + if (server_address_.proto == PROTO_UDP) { // If its UDP, send AllocateRequest now. // For TCP and TLS AllcateRequest will be sent by OnSocketConnect. SendRequest(new TurnAllocateRequest(this), 0); @@ -408,16 +408,16 @@ void TurnPort::PrepareAddress() { bool TurnPort::CreateTurnClientSocket() { RTC_DCHECK(!socket_ || SharedSocket()); - if (server_address_.proto == webrtc::PROTO_UDP && !SharedSocket()) { + if (server_address_.proto == PROTO_UDP && !SharedSocket()) { socket_ = socket_factory()->CreateUdpSocket( SocketAddress(Network()->GetBestIP(), 0), min_port(), max_port()); - } else if (server_address_.proto == webrtc::PROTO_TCP || - server_address_.proto == webrtc::PROTO_TLS) { + } else if (server_address_.proto == PROTO_TCP || + server_address_.proto == PROTO_TLS) { RTC_DCHECK(!SharedSocket()); int opts = PacketSocketFactory::OPT_STUN; // Apply server address TLS and insecure bits to options. - if (server_address_.proto == webrtc::PROTO_TLS) { + if (server_address_.proto == PROTO_TLS) { if (tls_cert_policy_ == TlsCertPolicy::TLS_CERT_POLICY_INSECURE_NO_CHECK) { opts |= PacketSocketFactory::OPT_TLS_INSECURE; @@ -461,8 +461,8 @@ bool TurnPort::CreateTurnClientSocket() { // TCP port is ready to send stun requests after the socket is connected, // while UDP port is ready to do so once the socket is created. - if (server_address_.proto == webrtc::PROTO_TCP || - server_address_.proto == webrtc::PROTO_TLS) { + if (server_address_.proto == PROTO_TCP || + server_address_.proto == PROTO_TLS) { socket_->SignalConnect.connect(this, &TurnPort::OnSocketConnect); socket_->SubscribeCloseEvent( this, [this](AsyncPacketSocket* s, int err) { OnSocketClose(s, err); }); @@ -475,8 +475,8 @@ bool TurnPort::CreateTurnClientSocket() { void TurnPort::OnSocketConnect(AsyncPacketSocket* socket) { // This slot should only be invoked if we're using a connection-oriented // protocol. - RTC_DCHECK(server_address_.proto == webrtc::PROTO_TCP || - server_address_.proto == webrtc::PROTO_TLS); + RTC_DCHECK(server_address_.proto == PROTO_TCP || + server_address_.proto == PROTO_TLS); // Do not use this port if the socket bound to an address not associated with // the desired network interface. This is seen in Chrome, where TCP sockets @@ -503,7 +503,7 @@ void TurnPort::OnSocketConnect(AsyncPacketSocket* socket) { << ", rather than an address associated with network:" << Network()->ToString() << ". Still allowing it since it's localhost."; - } else if (webrtc::IPIsAny(Network()->GetBestIP())) { + } else if (IPIsAny(Network()->GetBestIP())) { RTC_LOG(LS_WARNING) << ToString() << ": Socket is bound to the address:" << socket_address.ToSensitiveNameAndAddressString() @@ -734,7 +734,7 @@ bool TurnPort::HandleIncomingPacket(AsyncPacketSocket* socket, // Check the message type, to see if is a Channel Data message. // The message will either be channel data, a TURN data indication, or // a response to a previous request. - uint16_t msg_type = webrtc::GetBE16(packet.payload().data()); + uint16_t msg_type = GetBE16(packet.payload().data()); if (IsTurnChannelData(msg_type)) { HandleChannelData(msg_type, data, size, packet_time_us); return true; @@ -776,7 +776,7 @@ void TurnPort::OnReadyToSend(AsyncPacketSocket* socket) { bool TurnPort::SupportsProtocol(absl::string_view protocol) const { // Turn port only connects to UDP candidates. - return protocol == webrtc::UDP_PROTOCOL_NAME; + return protocol == UDP_PROTOCOL_NAME; } // Update current server address port with the alternate server address port. @@ -831,9 +831,8 @@ void TurnPort::ResolveTurnAddress(const SocketAddress& address) { // assuming socket layer will resolve the hostname through a HTTP proxy (if // any). auto& result = resolver_->result(); - if (result.GetError() != 0 && - (server_address_.proto == webrtc::PROTO_TCP || - server_address_.proto == webrtc::PROTO_TLS)) { + if (result.GetError() != 0 && (server_address_.proto == PROTO_TCP || + server_address_.proto == PROTO_TLS)) { if (!CreateTurnClientSocket()) { OnAllocateError(STUN_ERROR_SERVER_NOT_REACHABLE, "TURN host lookup received error."); @@ -893,7 +892,7 @@ void TurnPort::OnAllocateSuccess(const SocketAddress& address, AddAddress(address, // Candidate address. address, // Base address. related_address, // Related address. - webrtc::UDP_PROTOCOL_NAME, + UDP_PROTOCOL_NAME, ProtoToString(server_address_.proto), // The first hop protocol. "", // TCP candidate type, empty for turn candidates. IceCandidateType::kRelay, @@ -909,7 +908,7 @@ void TurnPort::OnAllocateError(int error_code, absl::string_view reason) { SafeTask(task_safety_.flag(), [this] { SignalPortError(this); })); std::string address = GetLocalAddress().HostAsSensitiveURIString(); int port = GetLocalAddress().port(); - if (server_address_.proto == webrtc::PROTO_TCP && + if (server_address_.proto == PROTO_TCP && server_address_.address.IsPrivateIP()) { address.clear(); port = 0; @@ -951,10 +950,9 @@ void TurnPort::Release() { void TurnPort::Close() { if (!ready()) { - OnAllocateError(STUN_ERROR_SERVER_NOT_REACHABLE, - GetProtocol() != webrtc::PROTO_UDP - ? "Failed to establish connection" - : ""); + OnAllocateError( + STUN_ERROR_SERVER_NOT_REACHABLE, + GetProtocol() != PROTO_UDP ? "Failed to establish connection" : ""); } request_manager_.Clear(); // Stop the port from creating new connections. @@ -981,7 +979,7 @@ bool TurnPort::AllowedTurnPort(int port) { } void TurnPort::TryAlternateServer() { - if (server_address().proto == webrtc::PROTO_UDP) { + if (server_address().proto == PROTO_UDP) { // Send another allocate request to alternate server, with the received // realm and nonce values. SendRequest(new TurnAllocateRequest(this), 0); @@ -989,8 +987,8 @@ void TurnPort::TryAlternateServer() { // Since it's TCP, we have to delete the connected socket and reconnect // with the alternate server. PrepareAddress will send stun binding once // the new socket is connected. - RTC_DCHECK(server_address().proto == webrtc::PROTO_TCP || - server_address().proto == webrtc::PROTO_TLS); + RTC_DCHECK(server_address().proto == PROTO_TCP || + server_address().proto == PROTO_TLS); RTC_DCHECK(!SharedSocket()); delete socket_; socket_ = nullptr; @@ -1046,8 +1044,7 @@ void TurnPort::HandleDataIndication(const char* data, // TODO(bugs.webrtc.org/14870): rebuild DispatchPacket to take an // ArrayView DispatchPacket(reinterpret_cast(data_attr->array_view().data()), - data_attr->length(), ext_addr, webrtc::PROTO_UDP, - packet_time_us); + data_attr->length(), ext_addr, PROTO_UDP, packet_time_us); } void TurnPort::HandleChannelData(uint16_t channel_id, @@ -1069,7 +1066,7 @@ void TurnPort::HandleChannelData(uint16_t channel_id, // +-------------------------------+ // Extract header fields from the message. - uint16_t len = webrtc::GetBE16(data + 2); + uint16_t len = GetBE16(data + 2); if (len > size - TURN_CHANNEL_HEADER_SIZE) { RTC_LOG(LS_WARNING) << ToString() << ": Received TURN channel data message with " @@ -1089,7 +1086,7 @@ void TurnPort::HandleChannelData(uint16_t channel_id, } DispatchPacket(data + TURN_CHANNEL_HEADER_SIZE, len, entry->address(), - webrtc::PROTO_UDP, packet_time_us); + PROTO_UDP, packet_time_us); } void TurnPort::DispatchPacket(const char* data, @@ -1273,14 +1270,14 @@ std::string TurnPort::ReconstructServerUrl() { std::string scheme = "turn"; std::string transport = "tcp"; switch (server_address_.proto) { - case webrtc::PROTO_SSLTCP: - case webrtc::PROTO_TLS: + case PROTO_SSLTCP: + case PROTO_TLS: scheme = "turns"; break; - case webrtc::PROTO_UDP: + case PROTO_UDP: transport = "udp"; break; - case webrtc::PROTO_TCP: + case PROTO_TCP: break; } StringBuilder url; @@ -1336,15 +1333,15 @@ TurnAllocateRequest::TurnAllocateRequest(TurnPort* port) } void TurnAllocateRequest::OnSent() { - RTC_LOG(LS_INFO) << port_->ToString() << ": TURN allocate request sent, id=" - << webrtc::hex_encode(id()); + RTC_LOG(LS_INFO) << port_->ToString() + << ": TURN allocate request sent, id=" << hex_encode(id()); StunRequest::OnSent(); } void TurnAllocateRequest::OnResponse(StunMessage* response) { RTC_LOG(LS_INFO) << port_->ToString() << ": TURN allocate requested successfully, id=" - << webrtc::hex_encode(id()) + << hex_encode(id()) << ", code=0" // Makes logging easier to parse. ", rtt=" << Elapsed(); @@ -1390,7 +1387,7 @@ void TurnAllocateRequest::OnErrorResponse(StunMessage* response) { RTC_LOG(LS_INFO) << port_->ToString() << ": Received TURN allocate error response, id=" - << webrtc::hex_encode(id()) << ", code=" << error_code + << hex_encode(id()) << ", code=" << error_code << ", rtt=" << Elapsed(); switch (error_code) { @@ -1410,7 +1407,7 @@ void TurnAllocateRequest::OnErrorResponse(StunMessage* response) { default: RTC_LOG(LS_WARNING) << port_->ToString() << ": Received TURN allocate error response, id=" - << webrtc::hex_encode(id()) << ", code=" << error_code + << hex_encode(id()) << ", code=" << error_code << ", rtt=" << Elapsed(); const StunErrorCodeAttribute* attr = response->GetErrorCode(); port_->OnAllocateError(error_code, attr ? attr->reason() : ""); @@ -1419,7 +1416,7 @@ void TurnAllocateRequest::OnErrorResponse(StunMessage* response) { void TurnAllocateRequest::OnTimeout() { RTC_LOG(LS_WARNING) << port_->ToString() << ": TURN allocate request " - << webrtc::hex_encode(id()) << " timeout"; + << hex_encode(id()) << " timeout"; port_->OnAllocateRequestTimeout(); } @@ -1526,15 +1523,15 @@ TurnRefreshRequest::TurnRefreshRequest(TurnPort* port, int lifetime /*= -1*/) } void TurnRefreshRequest::OnSent() { - RTC_LOG(LS_INFO) << port_->ToString() << ": TURN refresh request sent, id=" - << webrtc::hex_encode(id()); + RTC_LOG(LS_INFO) << port_->ToString() + << ": TURN refresh request sent, id=" << hex_encode(id()); StunRequest::OnSent(); } void TurnRefreshRequest::OnResponse(StunMessage* response) { RTC_LOG(LS_INFO) << port_->ToString() << ": TURN refresh requested successfully, id=" - << webrtc::hex_encode(id()) + << hex_encode(id()) << ", code=0" // Makes logging easier to parse. ", rtt=" << Elapsed(); @@ -1576,7 +1573,7 @@ void TurnRefreshRequest::OnErrorResponse(StunMessage* response) { } else { RTC_LOG(LS_WARNING) << port_->ToString() << ": Received TURN refresh error response, id=" - << webrtc::hex_encode(id()) << ", code=" << error_code + << hex_encode(id()) << ", code=" << error_code << ", rtt=" << Elapsed(); port_->OnRefreshError(); if (port_->callbacks_for_test_) { @@ -1587,7 +1584,7 @@ void TurnRefreshRequest::OnErrorResponse(StunMessage* response) { void TurnRefreshRequest::OnTimeout() { RTC_LOG(LS_WARNING) << port_->ToString() << ": TURN refresh timeout " - << webrtc::hex_encode(id()); + << hex_encode(id()); port_->OnRefreshError(); } @@ -1624,14 +1621,14 @@ TurnCreatePermissionRequest::~TurnCreatePermissionRequest() { void TurnCreatePermissionRequest::OnSent() { RTC_LOG(LS_INFO) << port_->ToString() << ": TURN create permission request sent, id=" - << webrtc::hex_encode(id()); + << hex_encode(id()); StunRequest::OnSent(); } void TurnCreatePermissionRequest::OnResponse(StunMessage* response) { RTC_LOG(LS_INFO) << port_->ToString() << ": TURN permission requested successfully, id=" - << webrtc::hex_encode(id()) + << hex_encode(id()) << ", code=0" // Makes logging easier to parse. ", rtt=" << Elapsed(); @@ -1645,7 +1642,7 @@ void TurnCreatePermissionRequest::OnErrorResponse(StunMessage* response) { int error_code = response->GetErrorCodeValue(); RTC_LOG(LS_WARNING) << port_->ToString() << ": Received TURN create permission error response, id=" - << webrtc::hex_encode(id()) << ", code=" << error_code + << hex_encode(id()) << ", code=" << error_code << ", rtt=" << Elapsed(); if (entry_) { entry_->OnCreatePermissionError(response, error_code); @@ -1655,7 +1652,7 @@ void TurnCreatePermissionRequest::OnErrorResponse(StunMessage* response) { void TurnCreatePermissionRequest::OnTimeout() { RTC_LOG(LS_WARNING) << port_->ToString() << ": TURN create permission timeout " - << webrtc::hex_encode(id()); + << hex_encode(id()); if (entry_) { entry_->OnCreatePermissionTimeout(); } @@ -1696,14 +1693,14 @@ TurnChannelBindRequest::~TurnChannelBindRequest() { void TurnChannelBindRequest::OnSent() { RTC_LOG(LS_INFO) << port_->ToString() << ": TURN channel bind request sent, id=" - << webrtc::hex_encode(id()); + << hex_encode(id()); StunRequest::OnSent(); } void TurnChannelBindRequest::OnResponse(StunMessage* response) { RTC_LOG(LS_INFO) << port_->ToString() << ": TURN channel bind requested successfully, id=" - << webrtc::hex_encode(id()) + << hex_encode(id()) << ", code=0" // Makes logging easier to parse. ", rtt=" << Elapsed(); @@ -1725,7 +1722,7 @@ void TurnChannelBindRequest::OnErrorResponse(StunMessage* response) { int error_code = response->GetErrorCodeValue(); RTC_LOG(LS_WARNING) << port_->ToString() << ": Received TURN channel bind error response, id=" - << webrtc::hex_encode(id()) << ", code=" << error_code + << hex_encode(id()) << ", code=" << error_code << ", rtt=" << Elapsed(); if (entry_) { entry_->OnChannelBindError(response, error_code); @@ -1734,7 +1731,7 @@ void TurnChannelBindRequest::OnErrorResponse(StunMessage* response) { void TurnChannelBindRequest::OnTimeout() { RTC_LOG(LS_WARNING) << port_->ToString() << ": TURN channel bind timeout " - << webrtc::hex_encode(id()); + << hex_encode(id()); if (entry_) { entry_->OnChannelBindTimeout(); } diff --git a/p2p/base/turn_port_unittest.cc b/p2p/base/turn_port_unittest.cc index 8f6403c966..dea2713cfc 100644 --- a/p2p/base/turn_port_unittest.cc +++ b/p2p/base/turn_port_unittest.cc @@ -326,7 +326,7 @@ class TurnPortTest : public ::testing::Test, turn_port_->SetIceTiebreaker(kTiebreakerDefault); ConnectSignals(); - if (server_address.proto == webrtc::PROTO_TLS) { + if (server_address.proto == PROTO_TLS) { // The test TURN server has a self-signed certificate so will not pass // the normal client validation. Instruct the client to ignore certificate // errors for testing only. @@ -339,7 +339,7 @@ class TurnPortTest : public ::testing::Test, void CreateSharedTurnPort(absl::string_view username, absl::string_view password, const ProtocolAddress& server_address) { - RTC_CHECK(server_address.proto == webrtc::PROTO_UDP); + RTC_CHECK(server_address.proto == PROTO_UDP); if (!socket_) { socket_.reset(socket_factory()->CreateUdpSocket( @@ -403,33 +403,33 @@ class TurnPortTest : public ::testing::Test, // turn_port_ should have been created. ASSERT_TRUE(turn_port_ != nullptr); turn_port_->PrepareAddress(); - ASSERT_THAT(webrtc::WaitUntil([&] { return turn_ready_; }, IsTrue(), - {.timeout = TimeDelta::Millis( - TimeToGetTurnCandidate(protocol_type)), - .clock = &fake_clock_}), - webrtc::IsRtcOk()); + ASSERT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(), + {.timeout = TimeDelta::Millis( + TimeToGetTurnCandidate(protocol_type)), + .clock = &fake_clock_}), + IsRtcOk()); CreateUdpPort(); udp_port_->PrepareAddress(); - ASSERT_THAT(webrtc::WaitUntil([&] { return udp_ready_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kSimulatedRtt), - .clock = &fake_clock_}), - webrtc::IsRtcOk()); + ASSERT_THAT(WaitUntil([&] { return udp_ready_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kSimulatedRtt), + .clock = &fake_clock_}), + IsRtcOk()); } // Returns the fake clock time to establish a connection over the given // protocol. int TimeToConnect(ProtocolType protocol_type) { switch (protocol_type) { - case webrtc::PROTO_TCP: + case PROTO_TCP: // The virtual socket server will delay by a fixed half a round trip // for a TCP connection. return kSimulatedRtt / 2; - case webrtc::PROTO_TLS: + case PROTO_TLS: // TLS operates over TCP and additionally has a round of HELLO for // negotiating ciphers and a round for exchanging certificates. - return 2 * kSimulatedRtt + TimeToConnect(webrtc::PROTO_TCP); - case webrtc::PROTO_UDP: + return 2 * kSimulatedRtt + TimeToConnect(PROTO_TCP); + case PROTO_UDP: default: // UDP requires no round trips to set up the connection. return 0; @@ -478,10 +478,10 @@ class TurnPortTest : public ::testing::Test, void TestTurnAllocateSucceeds(unsigned int timeout) { ASSERT_TRUE(turn_port_); turn_port_->PrepareAddress(); - EXPECT_THAT(webrtc::WaitUntil([&] { return turn_ready_; }, IsTrue(), - {.timeout = TimeDelta::Millis(timeout), - .clock = &fake_clock_}), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(), + {.timeout = TimeDelta::Millis(timeout), + .clock = &fake_clock_}), + IsRtcOk()); ASSERT_EQ(1U, turn_port_->Candidates().size()); EXPECT_EQ(kTurnUdpExtAddr.ipaddr(), turn_port_->Candidates()[0].address().ipaddr()); @@ -492,11 +492,11 @@ class TurnPortTest : public ::testing::Test, absl::string_view expected_url) { ASSERT_TRUE(turn_port_); turn_port_->PrepareAddress(); - ASSERT_THAT(webrtc::WaitUntil([&] { return turn_ready_; }, IsTrue(), - {.timeout = TimeDelta::Millis( - TimeToGetTurnCandidate(protocol_type)), - .clock = &fake_clock_}), - webrtc::IsRtcOk()); + ASSERT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(), + {.timeout = TimeDelta::Millis( + TimeToGetTurnCandidate(protocol_type)), + .clock = &fake_clock_}), + IsRtcOk()); ASSERT_EQ(1U, turn_port_->Candidates().size()); EXPECT_EQ(turn_port_->Candidates()[0].url(), expected_url); } @@ -517,12 +517,11 @@ class TurnPortTest : public ::testing::Test, const SocketAddress old_addr = turn_port_->server_address().address; turn_port_->PrepareAddress(); - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_ready_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(), {.timeout = TimeDelta::Millis( TimeToGetAlternateTurnCandidate(protocol_type)), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); // Retrieve the address again, the turn port's address should be // changed. const SocketAddress new_addr = turn_port_->server_address().address; @@ -545,12 +544,11 @@ class TurnPortTest : public ::testing::Test, turn_port_->PrepareAddress(); // Need time to connect to TURN server, send Allocate request and receive // redirect notice. - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_error_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(), {.timeout = TimeDelta::Millis( kSimulatedRtt + TimeToConnect(protocol_type)), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); } void TestTurnAlternateServerPingPong(ProtocolType protocol_type) { @@ -567,12 +565,11 @@ class TurnPortTest : public ::testing::Test, ProtocolAddress(kTurnIntAddr, protocol_type)); turn_port_->PrepareAddress(); - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_error_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(), {.timeout = TimeDelta::Millis( TimeToGetAlternateTurnCandidate(protocol_type)), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); ASSERT_EQ(0U, turn_port_->Candidates().size()); SocketAddress address; // Verify that we have exhausted all alternate servers instead of @@ -594,12 +591,11 @@ class TurnPortTest : public ::testing::Test, ProtocolAddress(kTurnIntAddr, protocol_type)); turn_port_->PrepareAddress(); - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_error_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(), {.timeout = TimeDelta::Millis( TimeToGetAlternateTurnCandidate(protocol_type)), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); ASSERT_EQ(0U, turn_port_->Candidates().size()); } @@ -616,17 +612,16 @@ class TurnPortTest : public ::testing::Test, // Pick an unusual address in the 127.0.0.0/8 range to make sure more than // 127.0.0.1 is covered. SocketAddress loopback_address(ipv6 ? "::1" : "127.1.2.3", - webrtc::TURN_SERVER_PORT); + TURN_SERVER_PORT); redirect_addresses.push_back(loopback_address); // Make a socket and bind it to the local port, to make extra sure no // packet is sent to this address. std::unique_ptr loopback_socket(ss_->CreateSocket( - AF_INET, - protocol_type == webrtc::PROTO_UDP ? SOCK_DGRAM : SOCK_STREAM)); + AF_INET, protocol_type == PROTO_UDP ? SOCK_DGRAM : SOCK_STREAM)); ASSERT_NE(nullptr, loopback_socket.get()); ASSERT_EQ(0, loopback_socket->Bind(loopback_address)); - if (protocol_type == webrtc::PROTO_TCP) { + if (protocol_type == PROTO_TCP) { ASSERT_EQ(0, loopback_socket->Listen(1)); } @@ -638,17 +633,17 @@ class TurnPortTest : public ::testing::Test, ProtocolAddress(server_address, protocol_type)); turn_port_->PrepareAddress(); - EXPECT_THAT(webrtc::WaitUntil([&] { return turn_error_; }, IsTrue(), - {.timeout = TimeDelta::Millis( - TimeToGetTurnCandidate(protocol_type)), - .clock = &fake_clock_}), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(), + {.timeout = TimeDelta::Millis( + TimeToGetTurnCandidate(protocol_type)), + .clock = &fake_clock_}), + IsRtcOk()); // Wait for some extra time, and make sure no packets were received on the // loopback port we created (or in the case of TCP, no connection attempt // occurred). SIMULATED_WAIT(false, kSimulatedRtt, fake_clock_); - if (protocol_type == webrtc::PROTO_UDP) { + if (protocol_type == PROTO_UDP) { char buf[1]; EXPECT_EQ(-1, loopback_socket->Recv(&buf, 1, nullptr)); } else { @@ -676,37 +671,35 @@ class TurnPortTest : public ::testing::Test, Connection* conn2 = turn_port_->CreateConnection(udp_port_->Candidates()[0], Port::ORIGIN_MESSAGE); ASSERT_TRUE(conn2 != NULL); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return turn_create_permission_success_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kSimulatedRtt), - .clock = &fake_clock_}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return turn_create_permission_success_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kSimulatedRtt), + .clock = &fake_clock_}), + IsRtcOk()); conn2->Ping(0); // Two hops from TURN port to UDP port through TURN server, thus two RTTs. - EXPECT_THAT( - webrtc::WaitUntil([&] { return conn2->write_state(); }, + EXPECT_THAT(WaitUntil([&] { return conn2->write_state(); }, Eq(Connection::STATE_WRITABLE), {.timeout = TimeDelta::Millis(kSimulatedRtt * 2), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); EXPECT_TRUE(conn1->receiving()); EXPECT_TRUE(conn2->receiving()); EXPECT_EQ(Connection::STATE_WRITE_INIT, conn1->write_state()); // Send another ping from UDP to TURN. conn1->Ping(0); - EXPECT_THAT( - webrtc::WaitUntil([&] { return conn1->write_state(); }, + EXPECT_THAT(WaitUntil([&] { return conn1->write_state(); }, Eq(Connection::STATE_WRITABLE), {.timeout = TimeDelta::Millis(kSimulatedRtt * 2), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); EXPECT_TRUE(conn2->receiving()); } void TestDestroyTurnConnection() { - PrepareTurnAndUdpPorts(webrtc::PROTO_UDP); + PrepareTurnAndUdpPorts(PROTO_UDP); // Create connections on both ends. Connection* conn1 = udp_port_->CreateConnection(turn_port_->Candidates()[0], @@ -719,19 +712,18 @@ class TurnPortTest : public ::testing::Test, turn_port_->set_timeout_delay(10 * 60 * 1000); ASSERT_TRUE(conn2 != NULL); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return turn_create_permission_success_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kSimulatedRtt), - .clock = &fake_clock_}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return turn_create_permission_success_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kSimulatedRtt), + .clock = &fake_clock_}), + IsRtcOk()); // Make sure turn connection can receive. conn1->Ping(0); - EXPECT_THAT( - webrtc::WaitUntil([&] { return conn1->write_state(); }, + EXPECT_THAT(WaitUntil([&] { return conn1->write_state(); }, Eq(Connection::STATE_WRITABLE), {.timeout = TimeDelta::Millis(kSimulatedRtt * 2), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); EXPECT_FALSE(turn_unknown_address_); // Destroy the connection on the TURN port. The TurnEntry still exists, so @@ -739,11 +731,10 @@ class TurnPortTest : public ::testing::Test, turn_port_->DestroyConnection(conn2); conn1->Ping(0); - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_unknown_address_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_unknown_address_; }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); // Wait for TurnEntry to expire. Timeout is 5 minutes. // Expect that it still processes an incoming ping and signals the @@ -775,20 +766,19 @@ class TurnPortTest : public ::testing::Test, conn1->set_remote_password_for_test(pwd); conn1->Ping(0); - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_unknown_address_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_unknown_address_; }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); // If the connection is created again, it will start to receive pings. conn2 = turn_port_->CreateConnection(udp_port_->Candidates()[0], Port::ORIGIN_MESSAGE); conn1->Ping(0); - EXPECT_THAT(webrtc::WaitUntil([&] { return conn2->receiving(); }, IsTrue(), - {.timeout = TimeDelta::Millis(kSimulatedRtt), - .clock = &fake_clock_}), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return conn2->receiving(); }, IsTrue(), + {.timeout = TimeDelta::Millis(kSimulatedRtt), + .clock = &fake_clock_}), + IsRtcOk()); } void TestTurnSendData(ProtocolType protocol_type) { @@ -816,19 +806,17 @@ class TurnPortTest : public ::testing::Test, conn2->SignalDestroyed.connect(this, &TurnPortTest::OnConnectionSignalDestroyed); conn1->Ping(0); - EXPECT_THAT( - webrtc::WaitUntil([&] { return conn1->write_state(); }, + EXPECT_THAT(WaitUntil([&] { return conn1->write_state(); }, Eq(Connection::STATE_WRITABLE), {.timeout = TimeDelta::Millis(kSimulatedRtt * 2), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); conn2->Ping(0); - EXPECT_THAT( - webrtc::WaitUntil([&] { return conn2->write_state(); }, + EXPECT_THAT(WaitUntil([&] { return conn2->write_state(); }, Eq(Connection::STATE_WRITABLE), {.timeout = TimeDelta::Millis(kSimulatedRtt * 2), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); // Send some data. size_t num_packets = 256; @@ -857,11 +845,11 @@ class TurnPortTest : public ::testing::Test, PrepareTurnAndUdpPorts(protocol_type); turn_port_.reset(); EXPECT_THAT( - webrtc::WaitUntil( - [&] { return turn_server_.server()->allocations().size(); }, Eq(0U), - {.timeout = TimeDelta::Millis(kSimulatedRtt), - .clock = &fake_clock_}), - webrtc::IsRtcOk()); + WaitUntil([&] { return turn_server_.server()->allocations().size(); }, + Eq(0U), + {.timeout = TimeDelta::Millis(kSimulatedRtt), + .clock = &fake_clock_}), + IsRtcOk()); } // Test that the TURN allocation is released by sending a refresh request @@ -892,19 +880,17 @@ class TurnPortTest : public ::testing::Test, &TurnPortTest::OnConnectionSignalDestroyed); conn1->Ping(0); - EXPECT_THAT( - webrtc::WaitUntil([&] { return conn1->write_state(); }, + EXPECT_THAT(WaitUntil([&] { return conn1->write_state(); }, Eq(Connection::STATE_WRITABLE), {.timeout = TimeDelta::Millis(kSimulatedRtt * 2), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); conn2->Ping(0); - EXPECT_THAT( - webrtc::WaitUntil([&] { return conn2->write_state(); }, + EXPECT_THAT(WaitUntil([&] { return conn2->write_state(); }, Eq(Connection::STATE_WRITABLE), {.timeout = TimeDelta::Millis(kSimulatedRtt * 2), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); // Send some data from Udp to TurnPort. unsigned char buf[256] = {0}; @@ -915,10 +901,10 @@ class TurnPortTest : public ::testing::Test, turn_port_->Release(); // Wait for the TurnPort to signal closed. - ASSERT_THAT(webrtc::WaitUntil([&] { return turn_port_closed_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kSimulatedRtt), - .clock = &fake_clock_}), - webrtc::IsRtcOk()); + ASSERT_THAT(WaitUntil([&] { return turn_port_closed_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kSimulatedRtt), + .clock = &fake_clock_}), + IsRtcOk()); // But the data should have arrived first. ASSERT_EQ(1ul, turn_packets_.size()); @@ -971,32 +957,29 @@ TEST_F(TurnPortTest, TestTurnPortType) { // gathering the candidates. TEST_F(TurnPortTest, TestReconstructedServerUrlForUdpIPv4) { CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr); - TestReconstructedServerUrl(webrtc::PROTO_UDP, - "turn:99.99.99.3:3478?transport=udp"); + TestReconstructedServerUrl(PROTO_UDP, "turn:99.99.99.3:3478?transport=udp"); } TEST_F(TurnPortTest, TestReconstructedServerUrlForUdpIPv6) { - turn_server_.AddInternalSocket(kTurnUdpIPv6IntAddr, webrtc::PROTO_UDP); + turn_server_.AddInternalSocket(kTurnUdpIPv6IntAddr, PROTO_UDP); CreateTurnPort(kLocalIPv6Addr, kTurnUsername, kTurnPassword, kTurnUdpIPv6ProtoAddr); // Should add [] around the IPv6. TestReconstructedServerUrl( - webrtc::PROTO_UDP, + PROTO_UDP, "turn:[2400:4030:1:2c00:be30:abcd:efab:cdef]:3478?transport=udp"); } TEST_F(TurnPortTest, TestReconstructedServerUrlForTcp) { - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TCP); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP); CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTcpProtoAddr); - TestReconstructedServerUrl(webrtc::PROTO_TCP, - "turn:99.99.99.4:3478?transport=tcp"); + TestReconstructedServerUrl(PROTO_TCP, "turn:99.99.99.4:3478?transport=tcp"); } TEST_F(TurnPortTest, TestReconstructedServerUrlForTls) { - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TLS); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TLS); CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTlsProtoAddr); - TestReconstructedServerUrl(webrtc::PROTO_TLS, - "turns:99.99.99.4:3478?transport=tcp"); + TestReconstructedServerUrl(PROTO_TLS, "turns:99.99.99.4:3478?transport=tcp"); } TEST_F(TurnPortTest, TestReconstructedServerUrlForHostname) { @@ -1006,10 +989,9 @@ TEST_F(TurnPortTest, TestReconstructedServerUrlForHostname) { // As VSS doesn't provide DNS resolution, name resolve will fail, // the error will be set and contain the url. turn_port_->PrepareAddress(); - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_error_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(), {.timeout = TimeDelta::Millis(kResolverTimeout)}), - webrtc::IsRtcOk()); + IsRtcOk()); std::string server_url = "turn:" + kTurnInvalidAddr.ToString() + "?transport=udp"; ASSERT_EQ(error_event_.url, server_url); @@ -1064,18 +1046,16 @@ TEST_F(TurnPortTest, TestTurnAllocateWithoutLoggingId) { TEST_F(TurnPortTest, TestTurnBadCredentials) { CreateTurnPort(kTurnUsername, "bad", kTurnUdpProtoAddr); turn_port_->PrepareAddress(); - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_error_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt * 3), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); ASSERT_EQ(0U, turn_port_->Candidates().size()); - EXPECT_THAT( - webrtc::WaitUntil([&] { return error_event_.error_code; }, + EXPECT_THAT(WaitUntil([&] { return error_event_.error_code; }, Eq(STUN_ERROR_UNAUTHORIZED), {.timeout = TimeDelta::Millis(kSimulatedRtt * 3), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); EXPECT_EQ(error_event_.error_text, "Unauthorized"); } @@ -1084,11 +1064,10 @@ TEST_F(TurnPortTest, TestTurnBadCredentials) { TEST_F(TurnPortTest, TestServerAddressFamilyMismatch) { CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpIPv6ProtoAddr); turn_port_->PrepareAddress(); - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_error_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt * 3), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); ASSERT_EQ(0U, turn_port_->Candidates().size()); EXPECT_EQ(0, error_event_.error_code); } @@ -1099,18 +1078,17 @@ TEST_F(TurnPortTest, TestServerAddressFamilyMismatch6) { CreateTurnPort(kLocalIPv6Addr, kTurnUsername, kTurnPassword, kTurnUdpProtoAddr); turn_port_->PrepareAddress(); - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_error_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt * 3), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); ASSERT_EQ(0U, turn_port_->Candidates().size()); EXPECT_EQ(0, error_event_.error_code); } // Testing a normal UDP allocation using TCP connection. TEST_F(TurnPortTest, TestTurnTcpAllocate) { - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TCP); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP); CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTcpProtoAddr); EXPECT_EQ(0, turn_port_->SetOption(Socket::OPT_SNDBUF, 10 * 1024)); TestTurnAllocateSucceeds(kSimulatedRtt * 3); @@ -1125,7 +1103,7 @@ TEST_F(TurnPortTest, TestTurnTcpAllocationWhenProxyChangesAddressToLocalHost) { // kLocalAddr, it will end up using localhost instead. ss_->SetAlternativeLocalAddress(kLocalAddr1.ipaddr(), local_address.ipaddr()); - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TCP); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP); CreateTurnPort(kLocalAddr1, kTurnUsername, kTurnPassword, kTurnTcpProtoAddr); EXPECT_EQ(0, turn_port_->SetOption(Socket::OPT_SNDBUF, 10 * 1024)); TestTurnAllocateSucceeds(kSimulatedRtt * 3); @@ -1148,7 +1126,7 @@ TEST_F(TurnPortTest, ss_->SetAlternativeLocalAddress(kLocalAddr1.ipaddr(), kLocalAddr2.ipaddr()); // Set up TURN server to use TCP (this logic only exists for TCP). - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TCP); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP); // Create TURN port and tell it to start allocation. CreateTurnPort(kLocalAddr1, kTurnUsername, kTurnPassword, kTurnTcpProtoAddr); @@ -1156,15 +1134,15 @@ TEST_F(TurnPortTest, // Shouldn't take more than 1 RTT to realize the bound address isn't the one // expected. - EXPECT_THAT(webrtc::WaitUntil([&] { return turn_error_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kSimulatedRtt), - .clock = &fake_clock_}), - webrtc::IsRtcOk()); - EXPECT_THAT(webrtc::WaitUntil([&] { return error_event_.error_code; }, - Eq(STUN_ERROR_SERVER_NOT_REACHABLE), - {.timeout = TimeDelta::Millis(kSimulatedRtt), - .clock = &fake_clock_}), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kSimulatedRtt), + .clock = &fake_clock_}), + IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return error_event_.error_code; }, + Eq(STUN_ERROR_SERVER_NOT_REACHABLE), + {.timeout = TimeDelta::Millis(kSimulatedRtt), + .clock = &fake_clock_}), + IsRtcOk()); EXPECT_NE(error_event_.error_text.find('.'), std::string::npos); EXPECT_NE(error_event_.address.find(kLocalAddr2.HostAsSensitiveURIString()), std::string::npos); @@ -1187,7 +1165,7 @@ TEST_F(TurnPortTest, TurnTcpAllocationNotDiscardedIfNotBoundToBestIP) { ASSERT_EQ(kLocalAddr1.ipaddr(), network->GetBestIP()); // Set up TURN server to use TCP (this logic only exists for TCP). - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TCP); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP); // Create TURN port using our special Network, and tell it to start // allocation. @@ -1196,11 +1174,10 @@ TEST_F(TurnPortTest, TurnTcpAllocationNotDiscardedIfNotBoundToBestIP) { turn_port_->PrepareAddress(); // Candidate should be gathered as normally. - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_ready_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt * 3), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); ASSERT_EQ(1U, turn_port_->Candidates().size()); // Verify that the socket actually used the alternate address, otherwise this @@ -1213,48 +1190,45 @@ TEST_F(TurnPortTest, TurnTcpAllocationNotDiscardedIfNotBoundToBestIP) { // between webrtc::IPAddress and webrtc::InterfaceAddress. TEST_F(TurnPortTest, TCPPortNotDiscardedIfBoundToTemporaryIP) { networks_.emplace_back("unittest", "unittest", kLocalIPv6Addr.ipaddr(), 32); - networks_.back().AddIP(InterfaceAddress(kLocalIPv6Addr.ipaddr(), - webrtc::IPV6_ADDRESS_FLAG_TEMPORARY)); + networks_.back().AddIP( + InterfaceAddress(kLocalIPv6Addr.ipaddr(), IPV6_ADDRESS_FLAG_TEMPORARY)); // Set up TURN server to use TCP (this logic only exists for TCP). - turn_server_.AddInternalSocket(kTurnIPv6IntAddr, webrtc::PROTO_TCP); + turn_server_.AddInternalSocket(kTurnIPv6IntAddr, PROTO_TCP); // Create TURN port using our special Network, and tell it to start // allocation. - CreateTurnPortWithNetwork( - &networks_.back(), kTurnUsername, kTurnPassword, - ProtocolAddress(kTurnIPv6IntAddr, webrtc::PROTO_TCP)); + CreateTurnPortWithNetwork(&networks_.back(), kTurnUsername, kTurnPassword, + ProtocolAddress(kTurnIPv6IntAddr, PROTO_TCP)); turn_port_->PrepareAddress(); // Candidate should be gathered as normally. - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_ready_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt * 3), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); ASSERT_EQ(1U, turn_port_->Candidates().size()); } // Testing turn port will attempt to create TCP socket on address resolution // failure. TEST_F(TurnPortTest, TestTurnTcpOnAddressResolveFailure) { - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TCP); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP); CreateTurnPort(kTurnUsername, kTurnPassword, - ProtocolAddress(kTurnInvalidAddr, webrtc::PROTO_TCP)); + ProtocolAddress(kTurnInvalidAddr, PROTO_TCP)); turn_port_->PrepareAddress(); - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_error_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(), {.timeout = TimeDelta::Millis(kResolverTimeout)}), - webrtc::IsRtcOk()); + IsRtcOk()); // As VSS doesn't provide DNS resolution, name resolve will fail. TurnPort // will proceed in creating a TCP socket which will fail as there is no // server on the above domain and error will be set to SOCKET_ERROR. EXPECT_EQ(SOCKET_ERROR, turn_port_->error()); - EXPECT_THAT(webrtc::WaitUntil([&] { return error_event_.error_code; }, - Eq(STUN_ERROR_SERVER_NOT_REACHABLE), - {.timeout = TimeDelta::Millis(kSimulatedRtt), - .clock = &fake_clock_}), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return error_event_.error_code; }, + Eq(STUN_ERROR_SERVER_NOT_REACHABLE), + {.timeout = TimeDelta::Millis(kSimulatedRtt), + .clock = &fake_clock_}), + IsRtcOk()); std::string server_url = "turn:" + kTurnInvalidAddr.ToString() + "?transport=tcp"; ASSERT_EQ(error_event_.url, server_url); @@ -1263,14 +1237,13 @@ TEST_F(TurnPortTest, TestTurnTcpOnAddressResolveFailure) { // Testing turn port will attempt to create TLS socket on address resolution // failure. TEST_F(TurnPortTest, TestTurnTlsOnAddressResolveFailure) { - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TLS); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TLS); CreateTurnPort(kTurnUsername, kTurnPassword, - ProtocolAddress(kTurnInvalidAddr, webrtc::PROTO_TLS)); + ProtocolAddress(kTurnInvalidAddr, PROTO_TLS)); turn_port_->PrepareAddress(); - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_error_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(), {.timeout = TimeDelta::Millis(kResolverTimeout)}), - webrtc::IsRtcOk()); + IsRtcOk()); EXPECT_EQ(SOCKET_ERROR, turn_port_->error()); } @@ -1278,12 +1251,11 @@ TEST_F(TurnPortTest, TestTurnTlsOnAddressResolveFailure) { // and return allocate failure. TEST_F(TurnPortTest, TestTurnUdpOnAddressResolveFailure) { CreateTurnPort(kTurnUsername, kTurnPassword, - ProtocolAddress(kTurnInvalidAddr, webrtc::PROTO_UDP)); + ProtocolAddress(kTurnInvalidAddr, PROTO_UDP)); turn_port_->PrepareAddress(); - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_error_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(), {.timeout = TimeDelta::Millis(kResolverTimeout)}), - webrtc::IsRtcOk()); + IsRtcOk()); // Error from turn port will not be socket error. EXPECT_NE(SOCKET_ERROR, turn_port_->error()); } @@ -1292,11 +1264,10 @@ TEST_F(TurnPortTest, TestTurnUdpOnAddressResolveFailure) { TEST_F(TurnPortTest, TestTurnAllocateBadPassword) { CreateTurnPort(kTurnUsername, "bad", kTurnUdpProtoAddr); turn_port_->PrepareAddress(); - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_error_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt * 2), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); ASSERT_EQ(0U, turn_port_->Candidates().size()); } @@ -1306,11 +1277,10 @@ TEST_F(TurnPortTest, TestTurnAllocateNonceResetAfterAllocateMismatch) { // Do a normal allocation first. CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr); turn_port_->PrepareAddress(); - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_ready_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt * 2), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); SocketAddress first_addr(turn_port_->socket()->GetLocalAddress()); // Destroy the turnport while keeping the drop probability to 1 to // suppress the release of the allocation at the server. @@ -1328,7 +1298,7 @@ TEST_F(TurnPortTest, TestTurnAllocateNonceResetAfterAllocateMismatch) { // using timestamp `ts_before` but then get an allocate mismatch error and // receive an even newer nonce based on the system clock. `ts_before` is // chosen so that the two NONCEs generated by the server will be different. - int64_t ts_before = webrtc::TimeMillis() - 1; + int64_t ts_before = TimeMillis() - 1; std::string first_nonce = turn_server_.server()->SetTimestampForNextNonce(ts_before); turn_port_->PrepareAddress(); @@ -1336,11 +1306,10 @@ TEST_F(TurnPortTest, TestTurnAllocateNonceResetAfterAllocateMismatch) { // Four round trips; first we'll get "stale nonce", then // "allocate mismatch", then "stale nonce" again, then finally it will // succeed. - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_ready_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt * 4), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); EXPECT_NE(first_nonce, turn_port_->nonce()); } @@ -1350,11 +1319,10 @@ TEST_F(TurnPortTest, TestTurnAllocateMismatch) { // Do a normal allocation first. CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr); turn_port_->PrepareAddress(); - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_ready_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt * 2), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); SocketAddress first_addr(turn_port_->socket()->GetLocalAddress()); // Clear connected_ flag on turnport to suppress the release of @@ -1374,11 +1342,10 @@ TEST_F(TurnPortTest, TestTurnAllocateMismatch) { // Four round trips; first we'll get "stale nonce", then // "allocate mismatch", then "stale nonce" again, then finally it will // succeed. - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_ready_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt * 4), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); // Verifies that the new port has a different address now. EXPECT_NE(first_addr, turn_port_->socket()->GetLocalAddress()); @@ -1386,10 +1353,9 @@ TEST_F(TurnPortTest, TestTurnAllocateMismatch) { // Verify that all packets received from the shared socket are ignored. std::string test_packet = "Test packet"; EXPECT_FALSE(turn_port_->HandleIncomingPacket( - socket_.get(), - ReceivedIpPacket::CreateFromLegacy( - test_packet.data(), test_packet.size(), webrtc::TimeMicros(), - SocketAddress(kTurnUdpExtAddr.ipaddr(), 0)))); + socket_.get(), ReceivedIpPacket::CreateFromLegacy( + test_packet.data(), test_packet.size(), TimeMicros(), + SocketAddress(kTurnUdpExtAddr.ipaddr(), 0)))); } // Tests that a shared-socket-TurnPort creates its own socket after @@ -1398,11 +1364,10 @@ TEST_F(TurnPortTest, TestSharedSocketAllocateMismatch) { // Do a normal allocation first. CreateSharedTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr); turn_port_->PrepareAddress(); - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_ready_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt * 2), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); SocketAddress first_addr(turn_port_->socket()->GetLocalAddress()); // Clear connected_ flag on turnport to suppress the release of @@ -1418,11 +1383,10 @@ TEST_F(TurnPortTest, TestSharedSocketAllocateMismatch) { turn_port_->PrepareAddress(); // Extra 2 round trips due to allocate mismatch. - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_ready_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt * 4), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); // Verifies that the new port has a different address now. EXPECT_NE(first_addr, turn_port_->socket()->GetLocalAddress()); @@ -1430,16 +1394,15 @@ TEST_F(TurnPortTest, TestSharedSocketAllocateMismatch) { } TEST_F(TurnPortTest, TestTurnTcpAllocateMismatch) { - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TCP); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP); CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTcpProtoAddr); // Do a normal allocation first. turn_port_->PrepareAddress(); - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_ready_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt * 3), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); SocketAddress first_addr(turn_port_->socket()->GetLocalAddress()); // Clear connected_ flag on turnport to suppress the release of @@ -1457,11 +1420,10 @@ TEST_F(TurnPortTest, TestTurnTcpAllocateMismatch) { EXPECT_EQ(first_addr, turn_port_->socket()->GetLocalAddress()); // Extra 2 round trips due to allocate mismatch. - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_ready_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt * 5), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); // Verifies that the new port has a different address now. EXPECT_NE(first_addr, turn_port_->socket()->GetLocalAddress()); @@ -1469,7 +1431,7 @@ TEST_F(TurnPortTest, TestTurnTcpAllocateMismatch) { TEST_F(TurnPortTest, TestRefreshRequestGetsErrorResponse) { CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr); - PrepareTurnAndUdpPorts(webrtc::PROTO_UDP); + PrepareTurnAndUdpPorts(PROTO_UDP); turn_port_->CreateConnection(udp_port_->Candidates()[0], Port::ORIGIN_MESSAGE); // Set bad credentials. @@ -1480,17 +1442,16 @@ TEST_F(TurnPortTest, TestRefreshRequestGetsErrorResponse) { // When this succeeds, it will schedule a new RefreshRequest with the bad // credential. turn_port_->request_manager().FlushForTest(TURN_REFRESH_REQUEST); - EXPECT_THAT(webrtc::WaitUntil([&] { return turn_refresh_success_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kSimulatedRtt), - .clock = &fake_clock_}), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return turn_refresh_success_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kSimulatedRtt), + .clock = &fake_clock_}), + IsRtcOk()); // Flush it again, it will receive a bad response. turn_port_->request_manager().FlushForTest(TURN_REFRESH_REQUEST); - EXPECT_THAT( - webrtc::WaitUntil( - [&] { return !turn_refresh_success_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kSimulatedRtt), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return !turn_refresh_success_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kSimulatedRtt), + .clock = &fake_clock_}), + IsRtcOk()); EXPECT_FALSE(turn_port_->connected()); EXPECT_TRUE(CheckAllConnectionsFailedAndPruned()); EXPECT_FALSE(turn_port_->HasRequests()); @@ -1500,7 +1461,7 @@ TEST_F(TurnPortTest, TestRefreshRequestGetsErrorResponse) { // closed. TEST_F(TurnPortTest, TestStopProcessingPacketsAfterClosed) { CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr); - PrepareTurnAndUdpPorts(webrtc::PROTO_UDP); + PrepareTurnAndUdpPorts(PROTO_UDP); Connection* conn1 = turn_port_->CreateConnection(udp_port_->Candidates()[0], Port::ORIGIN_MESSAGE); Connection* conn2 = udp_port_->CreateConnection(turn_port_->Candidates()[0], @@ -1509,12 +1470,11 @@ TEST_F(TurnPortTest, TestStopProcessingPacketsAfterClosed) { ASSERT_TRUE(conn2 != NULL); // Make sure conn2 is writable. conn2->Ping(0); - EXPECT_THAT( - webrtc::WaitUntil([&] { return conn2->write_state(); }, + EXPECT_THAT(WaitUntil([&] { return conn2->write_state(); }, Eq(Connection::STATE_WRITABLE), {.timeout = TimeDelta::Millis(kSimulatedRtt * 2), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); turn_port_->CloseForTest(); SIMULATED_WAIT(false, kSimulatedRtt, fake_clock_); @@ -1528,9 +1488,9 @@ TEST_F(TurnPortTest, TestStopProcessingPacketsAfterClosed) { // Test that CreateConnection will return null if port becomes disconnected. TEST_F(TurnPortTest, TestCreateConnectionWhenSocketClosed) { - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TCP); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP); CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTcpProtoAddr); - PrepareTurnAndUdpPorts(webrtc::PROTO_TCP); + PrepareTurnAndUdpPorts(PROTO_TCP); // Create a connection. Connection* conn1 = turn_port_->CreateConnection(udp_port_->Candidates()[0], Port::ORIGIN_MESSAGE); @@ -1546,97 +1506,97 @@ TEST_F(TurnPortTest, TestCreateConnectionWhenSocketClosed) { // Tests that when a TCP socket is closed, the respective TURN connection will // be destroyed. TEST_F(TurnPortTest, TestSocketCloseWillDestroyConnection) { - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TCP); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP); CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTcpProtoAddr); - PrepareTurnAndUdpPorts(webrtc::PROTO_TCP); + PrepareTurnAndUdpPorts(PROTO_TCP); Connection* conn = turn_port_->CreateConnection(udp_port_->Candidates()[0], Port::ORIGIN_MESSAGE); EXPECT_NE(nullptr, conn); EXPECT_TRUE(!turn_port_->connections().empty()); turn_port_->socket()->NotifyClosedForTest(1); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return turn_port_->connections().empty(); }, IsTrue(), - {.timeout = TimeDelta::Millis(kConnectionDestructionDelay), - .clock = &fake_clock_}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return turn_port_->connections().empty(); }, IsTrue(), + {.timeout = TimeDelta::Millis(kConnectionDestructionDelay), + .clock = &fake_clock_}), + IsRtcOk()); } // Test try-alternate-server feature. TEST_F(TurnPortTest, TestTurnAlternateServerUDP) { - TestTurnAlternateServer(webrtc::PROTO_UDP); + TestTurnAlternateServer(PROTO_UDP); } TEST_F(TurnPortTest, TestTurnAlternateServerTCP) { - TestTurnAlternateServer(webrtc::PROTO_TCP); + TestTurnAlternateServer(PROTO_TCP); } TEST_F(TurnPortTest, TestTurnAlternateServerTLS) { - TestTurnAlternateServer(webrtc::PROTO_TLS); + TestTurnAlternateServer(PROTO_TLS); } // Test that we fail when we redirect to an address different from // current IP family. TEST_F(TurnPortTest, TestTurnAlternateServerV4toV6UDP) { - TestTurnAlternateServerV4toV6(webrtc::PROTO_UDP); + TestTurnAlternateServerV4toV6(PROTO_UDP); } TEST_F(TurnPortTest, TestTurnAlternateServerV4toV6TCP) { - TestTurnAlternateServerV4toV6(webrtc::PROTO_TCP); + TestTurnAlternateServerV4toV6(PROTO_TCP); } TEST_F(TurnPortTest, TestTurnAlternateServerV4toV6TLS) { - TestTurnAlternateServerV4toV6(webrtc::PROTO_TLS); + TestTurnAlternateServerV4toV6(PROTO_TLS); } // Test try-alternate-server catches the case of pingpong. TEST_F(TurnPortTest, TestTurnAlternateServerPingPongUDP) { - TestTurnAlternateServerPingPong(webrtc::PROTO_UDP); + TestTurnAlternateServerPingPong(PROTO_UDP); } TEST_F(TurnPortTest, TestTurnAlternateServerPingPongTCP) { - TestTurnAlternateServerPingPong(webrtc::PROTO_TCP); + TestTurnAlternateServerPingPong(PROTO_TCP); } TEST_F(TurnPortTest, TestTurnAlternateServerPingPongTLS) { - TestTurnAlternateServerPingPong(webrtc::PROTO_TLS); + TestTurnAlternateServerPingPong(PROTO_TLS); } // Test try-alternate-server catch the case of repeated server. TEST_F(TurnPortTest, TestTurnAlternateServerDetectRepetitionUDP) { - TestTurnAlternateServerDetectRepetition(webrtc::PROTO_UDP); + TestTurnAlternateServerDetectRepetition(PROTO_UDP); } TEST_F(TurnPortTest, TestTurnAlternateServerDetectRepetitionTCP) { - TestTurnAlternateServerDetectRepetition(webrtc::PROTO_TCP); + TestTurnAlternateServerDetectRepetition(PROTO_TCP); } TEST_F(TurnPortTest, TestTurnAlternateServerDetectRepetitionTLS) { - TestTurnAlternateServerDetectRepetition(webrtc::PROTO_TCP); + TestTurnAlternateServerDetectRepetition(PROTO_TCP); } // Test catching the case of a redirect to loopback. TEST_F(TurnPortTest, TestTurnAlternateServerLoopbackUdpIpv4) { - TestTurnAlternateServerLoopback(webrtc::PROTO_UDP, false); + TestTurnAlternateServerLoopback(PROTO_UDP, false); } TEST_F(TurnPortTest, TestTurnAlternateServerLoopbackUdpIpv6) { - TestTurnAlternateServerLoopback(webrtc::PROTO_UDP, true); + TestTurnAlternateServerLoopback(PROTO_UDP, true); } TEST_F(TurnPortTest, TestTurnAlternateServerLoopbackTcpIpv4) { - TestTurnAlternateServerLoopback(webrtc::PROTO_TCP, false); + TestTurnAlternateServerLoopback(PROTO_TCP, false); } TEST_F(TurnPortTest, TestTurnAlternateServerLoopbackTcpIpv6) { - TestTurnAlternateServerLoopback(webrtc::PROTO_TCP, true); + TestTurnAlternateServerLoopback(PROTO_TCP, true); } TEST_F(TurnPortTest, TestTurnAlternateServerLoopbackTlsIpv4) { - TestTurnAlternateServerLoopback(webrtc::PROTO_TLS, false); + TestTurnAlternateServerLoopback(PROTO_TLS, false); } TEST_F(TurnPortTest, TestTurnAlternateServerLoopbackTlsIpv6) { - TestTurnAlternateServerLoopback(webrtc::PROTO_TLS, true); + TestTurnAlternateServerLoopback(PROTO_TLS, true); } // Do a TURN allocation and try to send a packet to it from the outside. @@ -1645,27 +1605,27 @@ TEST_F(TurnPortTest, TestTurnAlternateServerLoopbackTlsIpv6) { // outside. It should now work as well. TEST_F(TurnPortTest, TestTurnConnection) { CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr); - TestTurnConnection(webrtc::PROTO_UDP); + TestTurnConnection(PROTO_UDP); } // Similar to above, except that this test will use the shared socket. TEST_F(TurnPortTest, TestTurnConnectionUsingSharedSocket) { CreateSharedTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr); - TestTurnConnection(webrtc::PROTO_UDP); + TestTurnConnection(PROTO_UDP); } // Test that we can establish a TCP connection with TURN server. TEST_F(TurnPortTest, TestTurnTcpConnection) { - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TCP); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP); CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTcpProtoAddr); - TestTurnConnection(webrtc::PROTO_TCP); + TestTurnConnection(PROTO_TCP); } // Test that we can establish a TLS connection with TURN server. TEST_F(TurnPortTest, TestTurnTlsConnection) { - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TLS); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TLS); CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTlsProtoAddr); - TestTurnConnection(webrtc::PROTO_TLS); + TestTurnConnection(PROTO_TLS); } // Test that if a connection on a TURN port is destroyed, the TURN port can @@ -1688,7 +1648,7 @@ TEST_F(TurnPortTest, TestDestroyTurnConnectionUsingSharedSocket) { TEST_F(TurnPortTest, TestTurnConnectionUsingOTUNonce) { turn_server_.set_enable_otu_nonce(true); CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr); - TestTurnConnection(webrtc::PROTO_UDP); + TestTurnConnection(PROTO_UDP); } // Test that CreatePermissionRequest will be scheduled after the success @@ -1696,16 +1656,16 @@ TEST_F(TurnPortTest, TestTurnConnectionUsingOTUNonce) { // ErrorResponse if the ufrag and pwd are incorrect. TEST_F(TurnPortTest, TestRefreshCreatePermissionRequest) { CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr); - PrepareTurnAndUdpPorts(webrtc::PROTO_UDP); + PrepareTurnAndUdpPorts(PROTO_UDP); Connection* conn = turn_port_->CreateConnection(udp_port_->Candidates()[0], Port::ORIGIN_MESSAGE); ASSERT_TRUE(conn != NULL); EXPECT_THAT( - webrtc::WaitUntil( + WaitUntil( [&] { return turn_create_permission_success_; }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); turn_create_permission_success_ = false; // A create-permission-request should be pending. // After the next create-permission-response is received, it will schedule @@ -1714,23 +1674,23 @@ TEST_F(TurnPortTest, TestRefreshCreatePermissionRequest) { turn_port_->set_credentials(bad_credentials); turn_port_->request_manager().FlushForTest(kAllRequestsForTest); EXPECT_THAT( - webrtc::WaitUntil( + WaitUntil( [&] { return turn_create_permission_success_; }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); // Flush the requests again; the create-permission-request will fail. turn_port_->request_manager().FlushForTest(kAllRequestsForTest); EXPECT_THAT( - webrtc::WaitUntil( + WaitUntil( [&] { return !turn_create_permission_success_; }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); EXPECT_TRUE(CheckConnectionFailedAndPruned(conn)); } TEST_F(TurnPortTest, TestChannelBindGetErrorResponse) { CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr); - PrepareTurnAndUdpPorts(webrtc::PROTO_UDP); + PrepareTurnAndUdpPorts(PROTO_UDP); Connection* conn1 = turn_port_->CreateConnection(udp_port_->Candidates()[0], Port::ORIGIN_MESSAGE); ASSERT_TRUE(conn1 != nullptr); @@ -1739,11 +1699,10 @@ TEST_F(TurnPortTest, TestChannelBindGetErrorResponse) { ASSERT_TRUE(conn2 != nullptr); conn1->Ping(0); - EXPECT_THAT( - webrtc::WaitUntil([&] { return conn1->writable(); }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return conn1->writable(); }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt * 2), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); // Tell the TURN server to reject all bind requests from now on. turn_server_.server()->set_reject_bind_requests(true); @@ -1752,10 +1711,10 @@ TEST_F(TurnPortTest, TestChannelBindGetErrorResponse) { conn1->Send(data.data(), data.length(), options); EXPECT_THAT( - webrtc::WaitUntil( + WaitUntil( [&] { return CheckConnectionFailedAndPruned(conn1); }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); // Verify that packets are allowed to be sent after a bind request error. // They'll just use a send indication instead. @@ -1767,10 +1726,10 @@ TEST_F(TurnPortTest, TestChannelBindGetErrorResponse) { Buffer(packet.payload().data(), packet.payload().size())); }); conn1->Send(data.data(), data.length(), options); - EXPECT_THAT(webrtc::WaitUntil([&] { return !udp_packets_.empty(); }, IsTrue(), - {.timeout = TimeDelta::Millis(kSimulatedRtt), - .clock = &fake_clock_}), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return !udp_packets_.empty(); }, IsTrue(), + {.timeout = TimeDelta::Millis(kSimulatedRtt), + .clock = &fake_clock_}), + IsRtcOk()); conn2->DeregisterReceivedPacketCallback(); } @@ -1778,41 +1737,38 @@ TEST_F(TurnPortTest, TestChannelBindGetErrorResponse) { TEST_F(TurnPortTest, TestTurnSendDataTurnUdpToUdp) { // Create ports and prepare addresses. CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr); - TestTurnSendData(webrtc::PROTO_UDP); - EXPECT_EQ(webrtc::UDP_PROTOCOL_NAME, - turn_port_->Candidates()[0].relay_protocol()); + TestTurnSendData(PROTO_UDP); + EXPECT_EQ(UDP_PROTOCOL_NAME, turn_port_->Candidates()[0].relay_protocol()); } // Do a TURN allocation, establish a TCP connection, and send some data. TEST_F(TurnPortTest, TestTurnSendDataTurnTcpToUdp) { - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TCP); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP); // Create ports and prepare addresses. CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTcpProtoAddr); - TestTurnSendData(webrtc::PROTO_TCP); - EXPECT_EQ(webrtc::TCP_PROTOCOL_NAME, - turn_port_->Candidates()[0].relay_protocol()); + TestTurnSendData(PROTO_TCP); + EXPECT_EQ(TCP_PROTOCOL_NAME, turn_port_->Candidates()[0].relay_protocol()); } // Do a TURN allocation, establish a TLS connection, and send some data. TEST_F(TurnPortTest, TestTurnSendDataTurnTlsToUdp) { - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TLS); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TLS); CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTlsProtoAddr); - TestTurnSendData(webrtc::PROTO_TLS); - EXPECT_EQ(webrtc::TLS_PROTOCOL_NAME, - turn_port_->Candidates()[0].relay_protocol()); + TestTurnSendData(PROTO_TLS); + EXPECT_EQ(TLS_PROTOCOL_NAME, turn_port_->Candidates()[0].relay_protocol()); } // Test TURN fails to make a connection from IPv6 address to a server which has // IPv4 address. TEST_F(TurnPortTest, TestTurnLocalIPv6AddressServerIPv4) { - turn_server_.AddInternalSocket(kTurnUdpIPv6IntAddr, webrtc::PROTO_UDP); + turn_server_.AddInternalSocket(kTurnUdpIPv6IntAddr, PROTO_UDP); CreateTurnPort(kLocalIPv6Addr, kTurnUsername, kTurnPassword, kTurnUdpProtoAddr); turn_port_->PrepareAddress(); - ASSERT_THAT(webrtc::WaitUntil([&] { return turn_error_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kSimulatedRtt), - .clock = &fake_clock_}), - webrtc::IsRtcOk()); + ASSERT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kSimulatedRtt), + .clock = &fake_clock_}), + IsRtcOk()); EXPECT_TRUE(turn_port_->Candidates().empty()); } @@ -1820,7 +1776,7 @@ TEST_F(TurnPortTest, TestTurnLocalIPv6AddressServerIPv4) { // IPv6 intenal address. But in this test external address is a IPv4 address, // hence allocated address will be a IPv4 address. TEST_F(TurnPortTest, TestTurnLocalIPv6AddressServerIPv6ExtenalIPv4) { - turn_server_.AddInternalSocket(kTurnUdpIPv6IntAddr, webrtc::PROTO_UDP); + turn_server_.AddInternalSocket(kTurnUdpIPv6IntAddr, PROTO_UDP); CreateTurnPort(kLocalIPv6Addr, kTurnUsername, kTurnPassword, kTurnUdpIPv6ProtoAddr); TestTurnAllocateSucceeds(kSimulatedRtt * 2); @@ -1831,16 +1787,15 @@ TEST_F(TurnPortTest, TestTurnLocalIPv6AddressServerIPv6ExtenalIPv4) { // its local candidate will still be an IPv4 address and it can only create // connections with IPv4 remote candidates. TEST_F(TurnPortTest, TestCandidateAddressFamilyMatch) { - turn_server_.AddInternalSocket(kTurnUdpIPv6IntAddr, webrtc::PROTO_UDP); + turn_server_.AddInternalSocket(kTurnUdpIPv6IntAddr, PROTO_UDP); CreateTurnPort(kLocalIPv6Addr, kTurnUsername, kTurnPassword, kTurnUdpIPv6ProtoAddr); turn_port_->PrepareAddress(); - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_ready_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt * 2), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); ASSERT_EQ(1U, turn_port_->Candidates().size()); // Create an IPv4 candidate. It will match the TURN candidate. @@ -1862,22 +1817,21 @@ TEST_F(TurnPortTest, TestCandidateAddressFamilyMatch) { // Test that a CreatePermission failure will result in the connection being // pruned and failed. TEST_F(TurnPortTest, TestConnectionFailedAndPrunedOnCreatePermissionFailure) { - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TCP); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP); turn_server_.server()->set_reject_private_addresses(true); CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTcpProtoAddr); turn_port_->PrepareAddress(); - EXPECT_THAT( - webrtc::WaitUntil([&] { return turn_ready_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return turn_ready_; }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt * 3), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); CreateUdpPort(SocketAddress("10.0.0.10", 0)); udp_port_->PrepareAddress(); - EXPECT_THAT(webrtc::WaitUntil([&] { return udp_ready_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kSimulatedRtt), - .clock = &fake_clock_}), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return udp_ready_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kSimulatedRtt), + .clock = &fake_clock_}), + IsRtcOk()); // Create a connection. TestConnectionWrapper conn(turn_port_->CreateConnection( udp_port_->Candidates()[0], Port::ORIGIN_MESSAGE)); @@ -1886,16 +1840,16 @@ TEST_F(TurnPortTest, TestConnectionFailedAndPrunedOnCreatePermissionFailure) { // Asynchronously, CreatePermission request should be sent and fail, which // will make the connection pruned and failed. EXPECT_THAT( - webrtc::WaitUntil( + WaitUntil( [&] { return CheckConnectionFailedAndPruned(conn.connection()); }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); EXPECT_THAT( - webrtc::WaitUntil( + WaitUntil( [&] { return !turn_create_permission_success_; }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); // Check that the connection is not deleted asynchronously. SIMULATED_WAIT(conn.connection() == nullptr, kConnectionDestructionDelay, fake_clock_); @@ -1905,38 +1859,38 @@ TEST_F(TurnPortTest, TestConnectionFailedAndPrunedOnCreatePermissionFailure) { // Test that a TURN allocation is released when the port is closed. TEST_F(TurnPortTest, TestTurnReleaseAllocation) { CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr); - TestTurnReleaseAllocation(webrtc::PROTO_UDP); + TestTurnReleaseAllocation(PROTO_UDP); } // Test that a TURN TCP allocation is released when the port is closed. TEST_F(TurnPortTest, TestTurnTCPReleaseAllocation) { - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TCP); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP); CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTcpProtoAddr); - TestTurnReleaseAllocation(webrtc::PROTO_TCP); + TestTurnReleaseAllocation(PROTO_TCP); } TEST_F(TurnPortTest, TestTurnTLSReleaseAllocation) { - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TLS); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TLS); CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTlsProtoAddr); - TestTurnReleaseAllocation(webrtc::PROTO_TLS); + TestTurnReleaseAllocation(PROTO_TLS); } TEST_F(TurnPortTest, TestTurnUDPGracefulReleaseAllocation) { - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_UDP); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_UDP); CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr); - TestTurnGracefulReleaseAllocation(webrtc::PROTO_UDP); + TestTurnGracefulReleaseAllocation(PROTO_UDP); } TEST_F(TurnPortTest, TestTurnTCPGracefulReleaseAllocation) { - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TCP); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP); CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTcpProtoAddr); - TestTurnGracefulReleaseAllocation(webrtc::PROTO_TCP); + TestTurnGracefulReleaseAllocation(PROTO_TCP); } TEST_F(TurnPortTest, TestTurnTLSGracefulReleaseAllocation) { - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TLS); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TLS); CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTlsProtoAddr); - TestTurnGracefulReleaseAllocation(webrtc::PROTO_TLS); + TestTurnGracefulReleaseAllocation(PROTO_TLS); } // Test that nothing bad happens if we try to create a connection to the same @@ -1944,7 +1898,7 @@ TEST_F(TurnPortTest, TestTurnTLSGracefulReleaseAllocation) { // DCHECK. TEST_F(TurnPortTest, CanCreateTwoConnectionsToSameAddress) { CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr); - PrepareTurnAndUdpPorts(webrtc::PROTO_UDP); + PrepareTurnAndUdpPorts(PROTO_UDP); Connection* conn1 = turn_port_->CreateConnection(udp_port_->Candidates()[0], Port::ORIGIN_MESSAGE); Connection* conn2 = turn_port_->CreateConnection(udp_port_->Candidates()[0], @@ -1957,23 +1911,22 @@ TEST_F(TurnPortTest, CanCreateTwoConnectionsToSameAddress) { #if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID) TEST_F(TurnPortTest, TestResolverShutdown) { - turn_server_.AddInternalSocket(kTurnUdpIPv6IntAddr, webrtc::PROTO_UDP); + turn_server_.AddInternalSocket(kTurnUdpIPv6IntAddr, PROTO_UDP); int last_fd_count = GetFDCount(); // Need to supply unresolved address to kick off resolver. CreateTurnPort(kLocalIPv6Addr, kTurnUsername, kTurnPassword, - ProtocolAddress(kTurnInvalidAddr, webrtc::PROTO_UDP)); + ProtocolAddress(kTurnInvalidAddr, PROTO_UDP)); turn_port_->PrepareAddress(); - ASSERT_THAT( - webrtc::WaitUntil([&] { return turn_error_; }, IsTrue(), + ASSERT_THAT(WaitUntil([&] { return turn_error_; }, IsTrue(), {.timeout = TimeDelta::Millis(kResolverTimeout)}), - webrtc::IsRtcOk()); + IsRtcOk()); EXPECT_TRUE(turn_port_->Candidates().empty()); turn_port_.reset(); Thread::Current()->PostTask([this] { test_finish_ = true; }); // Waiting for above message to be processed. - ASSERT_THAT(webrtc::WaitUntil([&] { return test_finish_; }, IsTrue(), - {.clock = &fake_clock_}), - webrtc::IsRtcOk()); + ASSERT_THAT(WaitUntil([&] { return test_finish_; }, IsTrue(), + {.clock = &fake_clock_}), + IsRtcOk()); EXPECT_EQ(last_fd_count, GetFDCount()); } #endif @@ -2029,14 +1982,13 @@ TEST_F(TurnPortTest, TestTurnCustomizerCount) { &observer_message_counter, &observer_channel_data_counter, &observer_attr_counter)); - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TLS); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TLS); turn_customizer_.reset(customizer); turn_server_.server()->SetStunMessageObserver(std::move(validator)); CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTlsProtoAddr); - TestTurnSendData(webrtc::PROTO_TLS); - EXPECT_EQ(webrtc::TLS_PROTOCOL_NAME, - turn_port_->Candidates()[0].relay_protocol()); + TestTurnSendData(PROTO_TLS); + EXPECT_EQ(TLS_PROTOCOL_NAME, turn_port_->Candidates()[0].relay_protocol()); // There should have been at least turn_packets_.size() calls to `customizer`. EXPECT_GE(customizer->modify_cnt_ + customizer->allow_channel_data_cnt_, @@ -2060,14 +2012,13 @@ TEST_F(TurnPortTest, TestTurnCustomizerDisallowChannelData) { &observer_message_counter, &observer_channel_data_counter, &observer_attr_counter)); customizer->allow_channel_data_ = false; - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TLS); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TLS); turn_customizer_.reset(customizer); turn_server_.server()->SetStunMessageObserver(std::move(validator)); CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTlsProtoAddr); - TestTurnSendData(webrtc::PROTO_TLS); - EXPECT_EQ(webrtc::TLS_PROTOCOL_NAME, - turn_port_->Candidates()[0].relay_protocol()); + TestTurnSendData(PROTO_TLS); + EXPECT_EQ(TLS_PROTOCOL_NAME, turn_port_->Candidates()[0].relay_protocol()); // There should have been at least turn_packets_.size() calls to `customizer`. EXPECT_GE(customizer->modify_cnt_, turn_packets_.size()); @@ -2091,14 +2042,13 @@ TEST_F(TurnPortTest, TestTurnCustomizerAddAttribute) { &observer_attr_counter)); customizer->allow_channel_data_ = false; customizer->add_counter_ = true; - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TLS); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TLS); turn_customizer_.reset(customizer); turn_server_.server()->SetStunMessageObserver(std::move(validator)); CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTlsProtoAddr); - TestTurnSendData(webrtc::PROTO_TLS); - EXPECT_EQ(webrtc::TLS_PROTOCOL_NAME, - turn_port_->Candidates()[0].relay_protocol()); + TestTurnSendData(PROTO_TLS); + EXPECT_EQ(TLS_PROTOCOL_NAME, turn_port_->Candidates()[0].relay_protocol()); // There should have been at least turn_packets_.size() calls to `customizer`. EXPECT_GE(customizer->modify_cnt_, turn_packets_.size()); @@ -2148,7 +2098,7 @@ TEST_F(TurnPortTest, TestTurnDangerousServerPermits443) { } TEST_F(TurnPortTest, TestTurnDangerousAlternateServer) { - const ProtocolType protocol_type = webrtc::PROTO_TCP; + const ProtocolType protocol_type = PROTO_TCP; std::vector redirect_addresses; redirect_addresses.push_back(kTurnDangerousAddr); @@ -2165,12 +2115,11 @@ TEST_F(TurnPortTest, TestTurnDangerousAlternateServer) { turn_port_->PrepareAddress(); // This should result in an error event. - EXPECT_THAT( - webrtc::WaitUntil([&] { return error_event_.error_code; }, Ne(0), + EXPECT_THAT(WaitUntil([&] { return error_event_.error_code; }, Ne(0), {.timeout = TimeDelta::Millis( TimeToGetAlternateTurnCandidate(protocol_type)), .clock = &fake_clock_}), - webrtc::IsRtcOk()); + IsRtcOk()); // but should NOT result in the port turning ready, and no candidates // should be gathered. EXPECT_FALSE(turn_ready_); @@ -2214,7 +2163,7 @@ TEST_F(TurnPortWithMockDnsResolverTest, TestHostnameResolved) { // Test an allocation from a TURN server specified by a hostname on an IPv6 // network. TEST_F(TurnPortWithMockDnsResolverTest, TestHostnameResolvedIPv6Network) { - turn_server_.AddInternalSocket(kTurnUdpIPv6IntAddr, webrtc::PROTO_UDP); + turn_server_.AddInternalSocket(kTurnUdpIPv6IntAddr, PROTO_UDP); CreateTurnPort(kLocalIPv6Addr, kTurnUsername, kTurnPassword, kTurnPortValidHostnameProtoAddr); SetDnsResolverExpectations( diff --git a/p2p/client/basic_port_allocator.cc b/p2p/client/basic_port_allocator.cc index ed9db392c4..3e04332aa4 100644 --- a/p2p/client/basic_port_allocator.cc +++ b/p2p/client/basic_port_allocator.cc @@ -57,10 +57,6 @@ namespace webrtc { namespace { -using ::webrtc::CreateRandomId; -using ::webrtc::IceCandidateType; -using ::webrtc::SafeTask; -using ::webrtc::TimeDelta; const int PHASE_UDP = 0; const int PHASE_RELAY = 1; @@ -71,12 +67,12 @@ const int kNumPhases = 3; // Gets protocol priority: UDP > TCP > SSLTCP == TLS. int GetProtocolPriority(ProtocolType protocol) { switch (protocol) { - case webrtc::PROTO_UDP: + case PROTO_UDP: return 2; - case webrtc::PROTO_TCP: + case PROTO_TCP: return 1; - case webrtc::PROTO_SSLTCP: - case webrtc::PROTO_TLS: + case PROTO_SSLTCP: + case PROTO_TLS: return 0; default: RTC_DCHECK_NOTREACHED(); @@ -143,15 +139,15 @@ bool IsAllowedByCandidateFilter(const Candidate& c, uint32_t filter) { } if (c.is_relay()) { - return ((filter & webrtc::CF_RELAY) != 0); + return ((filter & CF_RELAY) != 0); } if (c.is_stun()) { - return ((filter & webrtc::CF_REFLEXIVE) != 0); + return ((filter & CF_REFLEXIVE) != 0); } if (c.is_local()) { - if ((filter & webrtc::CF_REFLEXIVE) && !c.address().IsPrivateIP()) { + if ((filter & CF_REFLEXIVE) && !c.address().IsPrivateIP()) { // We allow host candidates if the filter allows server-reflexive // candidates and the candidate is a public IP. Because we don't generate // server-reflexive candidates if they have the same IP as the host @@ -161,7 +157,7 @@ bool IsAllowedByCandidateFilter(const Candidate& c, uint32_t filter) { return true; } - return ((filter & webrtc::CF_HOST) != 0); + return ((filter & CF_HOST) != 0); } return false; @@ -178,8 +174,8 @@ std::string NetworksToString(const std::vector& networks) { } // namespace const uint32_t DISABLE_ALL_PHASES = - webrtc::PORTALLOCATOR_DISABLE_UDP | webrtc::PORTALLOCATOR_DISABLE_TCP | - webrtc::PORTALLOCATOR_DISABLE_STUN | webrtc::PORTALLOCATOR_DISABLE_RELAY; + PORTALLOCATOR_DISABLE_UDP | PORTALLOCATOR_DISABLE_TCP | + PORTALLOCATOR_DISABLE_STUN | PORTALLOCATOR_DISABLE_RELAY; BasicPortAllocator::BasicPortAllocator( const Environment& env, @@ -194,7 +190,7 @@ BasicPortAllocator::BasicPortAllocator( RTC_CHECK(socket_factory_); RTC_DCHECK(network_manager_); SetConfiguration(ServerAddresses(), std::vector(), 0, - webrtc::NO_PRUNE, turn_customizer); + NO_PRUNE, turn_customizer); } BasicPortAllocator::~BasicPortAllocator() { @@ -217,10 +213,10 @@ int BasicPortAllocator::GetNetworkIgnoreMask() const { int mask = network_ignore_mask_; switch (vpn_preference_) { case VpnPreference::kOnlyUseVpn: - mask |= ~static_cast(webrtc::ADAPTER_TYPE_VPN); + mask |= ~static_cast(ADAPTER_TYPE_VPN); break; case VpnPreference::kNeverUseVpn: - mask |= static_cast(webrtc::ADAPTER_TYPE_VPN); + mask |= static_cast(ADAPTER_TYPE_VPN); break; default: break; @@ -504,7 +500,7 @@ void BasicPortAllocatorSession::SetStunKeepaliveIntervalForReadyPorts( // IceCandidateType::kHost but uses the protocol PROTO_TCP. if (port->Type() == IceCandidateType::kSrflx || (port->Type() == IceCandidateType::kHost && - port->GetProtocol() == webrtc::PROTO_UDP)) { + port->GetProtocol() == PROTO_UDP)) { static_cast(port)->set_stun_keepalive_delay( stun_keepalive_interval); } @@ -679,13 +675,13 @@ std::vector BasicPortAllocatorSession::GetNetworks() { // been passed in. if (network_manager->enumeration_permission() == NetworkManager::ENUMERATION_BLOCKED) { - set_flags(flags() | webrtc::PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION); + set_flags(flags() | PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION); } // If the adapter enumeration is disabled, we'll just bind to any address // instead of specific NIC. This is to ensure the same routing for http // traffic by OS is also used here to avoid any local or public IP leakage // during stun process. - if (flags() & webrtc::PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION) { + if (flags() & PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION) { networks = network_manager->GetAnyAddressNetworks(); } else { networks = network_manager->GetNetworks(); @@ -694,7 +690,7 @@ std::vector BasicPortAllocatorSession::GetNetworks() { // the OS. Or, if the PORTALLOCATOR_ENABLE_ANY_ADDRESS_PORTS flag is // set, we'll use ANY address candidates either way. if (networks.empty() || - (flags() & webrtc::PORTALLOCATOR_ENABLE_ANY_ADDRESS_PORTS)) { + (flags() & PORTALLOCATOR_ENABLE_ANY_ADDRESS_PORTS)) { std::vector any_address_networks = network_manager->GetAnyAddressNetworks(); networks.insert(networks.end(), any_address_networks.begin(), @@ -706,10 +702,10 @@ std::vector BasicPortAllocatorSession::GetNetworks() { } } // Filter out link-local networks if needed. - if (flags() & webrtc::PORTALLOCATOR_DISABLE_LINK_LOCAL_NETWORKS) { + if (flags() & PORTALLOCATOR_DISABLE_LINK_LOCAL_NETWORKS) { NetworkFilter link_local_filter( [](const webrtc::Network* network) { - return webrtc::IPIsLinkLocal(network->prefix()); + return IPIsLinkLocal(network->prefix()); }, "link-local"); FilterNetworks(&networks, link_local_filter); @@ -722,14 +718,14 @@ std::vector BasicPortAllocatorSession::GetNetworks() { }, "ignored"); FilterNetworks(&networks, ignored_filter); - if (flags() & webrtc::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS) { - uint16_t lowest_cost = webrtc::kNetworkCostMax; + if (flags() & PORTALLOCATOR_DISABLE_COSTLY_NETWORKS) { + uint16_t lowest_cost = kNetworkCostMax; for (const Network* network : networks) { // Don't determine the lowest cost from a link-local network. // On iOS, a device connected to the computer will get a link-local // network for communicating with the computer, however this network can't // be used to connect to a peer outside the network. - if (webrtc::IPIsLinkLocal(network->GetBestIP())) { + if (IPIsLinkLocal(network->GetBestIP())) { continue; } lowest_cost = std::min( @@ -738,7 +734,7 @@ std::vector BasicPortAllocatorSession::GetNetworks() { NetworkFilter costly_filter( [lowest_cost, this](const Network* network) { return network->GetCost(allocator()->env().field_trials()) > - lowest_cost + webrtc::kNetworkCostLow; + lowest_cost + kNetworkCostLow; }, "costly"); FilterNetworks(&networks, costly_filter); @@ -771,10 +767,9 @@ std::vector BasicPortAllocatorSession::SelectIPv6Networks( // Adapter types are placed in priority order. Cellular type is an alias of // cellular, 2G..5G types. std::vector adapter_types = { - webrtc::ADAPTER_TYPE_ETHERNET, webrtc::ADAPTER_TYPE_LOOPBACK, - webrtc::ADAPTER_TYPE_WIFI, webrtc::ADAPTER_TYPE_CELLULAR, - webrtc::ADAPTER_TYPE_VPN, webrtc::ADAPTER_TYPE_UNKNOWN, - webrtc::ADAPTER_TYPE_ANY}; + ADAPTER_TYPE_ETHERNET, ADAPTER_TYPE_LOOPBACK, ADAPTER_TYPE_WIFI, + ADAPTER_TYPE_CELLULAR, ADAPTER_TYPE_VPN, ADAPTER_TYPE_UNKNOWN, + ADAPTER_TYPE_ANY}; int adapter_types_cnt = adapter_types.size(); std::vector selected_networks; int adapter_types_pos = 0; @@ -786,7 +781,7 @@ std::vector BasicPortAllocatorSession::SelectIPv6Networks( if (adapter_types[adapter_types_pos % adapter_types_cnt] == all_ipv6_networks[network_pos]->type() || (adapter_types[adapter_types_pos % adapter_types_cnt] == - webrtc::ADAPTER_TYPE_CELLULAR && + ADAPTER_TYPE_CELLULAR && all_ipv6_networks[network_pos]->IsCellular())) { selected_networks.push_back(all_ipv6_networks[network_pos]); all_ipv6_networks.erase(all_ipv6_networks.begin() + network_pos); @@ -825,18 +820,18 @@ void BasicPortAllocatorSession::DoAllocate(bool disable_equivalent) { if (!config || config->relays.empty()) { // No relay ports specified in this config. - sequence_flags |= webrtc::PORTALLOCATOR_DISABLE_RELAY; + sequence_flags |= PORTALLOCATOR_DISABLE_RELAY; } - if (!(sequence_flags & webrtc::PORTALLOCATOR_ENABLE_IPV6) && + if (!(sequence_flags & PORTALLOCATOR_ENABLE_IPV6) && networks[i]->GetBestIP().family() == AF_INET6) { // Skip IPv6 networks unless the flag's been set. continue; } - if (!(sequence_flags & webrtc::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI) && + if (!(sequence_flags & PORTALLOCATOR_ENABLE_IPV6_ON_WIFI) && networks[i]->GetBestIP().family() == AF_INET6 && - networks[i]->type() == webrtc::ADAPTER_TYPE_WIFI) { + networks[i]->type() == ADAPTER_TYPE_WIFI) { // Skip IPv6 Wi-Fi networks unless the flag's been set. continue; } @@ -929,7 +924,7 @@ void BasicPortAllocatorSession::AddAllocatedPort(Port* port, port->set_component(component()); port->set_generation(generation()); port->set_send_retransmit_count_attribute( - (flags() & webrtc::PORTALLOCATOR_ENABLE_STUN_RETRANSMIT_ATTRIBUTE) != 0); + (flags() & PORTALLOCATOR_ENABLE_STUN_RETRANSMIT_ATTRIBUTE) != 0); PortData data(port, seq); ports_.push_back(data); @@ -985,9 +980,9 @@ void BasicPortAllocatorSession::OnCandidateReady(Port* port, data->set_has_pairable_candidate(true); if (port->Type() == IceCandidateType::kRelay) { - if (turn_port_prune_policy_ == webrtc::KEEP_FIRST_READY) { + if (turn_port_prune_policy_ == KEEP_FIRST_READY) { pruned = PruneNewlyPairableTurnPort(data); - } else if (turn_port_prune_policy_ == webrtc::PRUNE_BASED_ON_PRIORITY) { + } else if (turn_port_prune_policy_ == PRUNE_BASED_ON_PRIORITY) { pruned = PruneTurnPorts(port); } } @@ -1160,8 +1155,8 @@ bool BasicPortAllocatorSession::CandidatePairable(const Candidate& c, // both device enumeration and host candidates being disabled. bool network_enumeration_disabled = c.address().IsAnyIP(); bool can_ping_from_candidate = - (port->SharedSocket() || c.protocol() == webrtc::TCP_PROTOCOL_NAME); - bool host_candidates_disabled = !(candidate_filter_ & webrtc::CF_HOST); + (port->SharedSocket() || c.protocol() == TCP_PROTOCOL_NAME); + bool host_candidates_disabled = !(candidate_filter_ & CF_HOST); return candidate_signalable || (network_enumeration_disabled && can_ping_from_candidate && @@ -1281,7 +1276,7 @@ AllocationSequence::AllocationSequence( std::move(port_allocation_complete_callback)) {} void AllocationSequence::Init() { - if (IsFlagSet(webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET)) { + if (IsFlagSet(PORTALLOCATOR_ENABLE_SHARED_SOCKET)) { udp_socket_.reset(session_->socket_factory()->CreateUdpSocket( SocketAddress(network_->GetBestIP(), 0), session_->allocator()->min_port(), session_->allocator()->max_port())); @@ -1343,22 +1338,22 @@ void AllocationSequence::DisableEquivalentPhases(const Network* network, if (absl::c_any_of(session_->ports_, [this](const BasicPortAllocatorSession::PortData& p) { return !p.pruned() && p.port()->Network() == network_ && - p.port()->GetProtocol() == webrtc::PROTO_UDP && + p.port()->GetProtocol() == PROTO_UDP && p.port()->Type() == IceCandidateType::kHost && !p.error(); })) { - *flags |= webrtc::PORTALLOCATOR_DISABLE_UDP; + *flags |= PORTALLOCATOR_DISABLE_UDP; } // Similarly we need to check both the protocol used by an existing Port and // its type. if (absl::c_any_of(session_->ports_, [this](const BasicPortAllocatorSession::PortData& p) { return !p.pruned() && p.port()->Network() == network_ && - p.port()->GetProtocol() == webrtc::PROTO_TCP && + p.port()->GetProtocol() == PROTO_TCP && p.port()->Type() == IceCandidateType::kHost && !p.error(); })) { - *flags |= webrtc::PORTALLOCATOR_DISABLE_TCP; + *flags |= PORTALLOCATOR_DISABLE_TCP; } if (config_ && config) { @@ -1368,9 +1363,9 @@ void AllocationSequence::DisableEquivalentPhases(const Network* network, // 2. We will regather host candidates, hence possibly inducing new NAT // bindings. if (config_->StunServers() == config->StunServers() && - (*flags & webrtc::PORTALLOCATOR_DISABLE_UDP)) { + (*flags & PORTALLOCATOR_DISABLE_UDP)) { // Already got this STUN servers covered. - *flags |= webrtc::PORTALLOCATOR_DISABLE_STUN; + *flags |= PORTALLOCATOR_DISABLE_STUN; } if (!config_->relays.empty()) { // Already got relays covered. @@ -1378,7 +1373,7 @@ void AllocationSequence::DisableEquivalentPhases(const Network* network, // were to be given one, but that never happens in our codebase. Should // probably get rid of the list in PortConfiguration and just keep a // single relay server in each one. - *flags |= webrtc::PORTALLOCATOR_DISABLE_RELAY; + *flags |= PORTALLOCATOR_DISABLE_RELAY; } } } @@ -1447,7 +1442,7 @@ void AllocationSequence::Process(int epoch) { } void AllocationSequence::CreateUDPPorts() { - if (IsFlagSet(webrtc::PORTALLOCATOR_DISABLE_UDP)) { + if (IsFlagSet(PORTALLOCATOR_DISABLE_UDP)) { RTC_LOG(LS_VERBOSE) << "AllocationSequence: UDP ports disabled, skipping."; return; } @@ -1456,8 +1451,8 @@ void AllocationSequence::CreateUDPPorts() { // is enabled completely. std::unique_ptr port; bool emit_local_candidate_for_anyaddress = - !IsFlagSet(webrtc::PORTALLOCATOR_DISABLE_DEFAULT_LOCAL_CANDIDATE); - if (IsFlagSet(webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET) && udp_socket_) { + !IsFlagSet(PORTALLOCATOR_DISABLE_DEFAULT_LOCAL_CANDIDATE); + if (IsFlagSet(PORTALLOCATOR_ENABLE_SHARED_SOCKET) && udp_socket_) { port = UDPPort::Create( {.env = session_->allocator()->env(), .network_thread = session_->network_thread(), @@ -1484,13 +1479,13 @@ void AllocationSequence::CreateUDPPorts() { port->SetIceTiebreaker(session_->allocator()->ice_tiebreaker()); // If shared socket is enabled, STUN candidate will be allocated by the // UDPPort. - if (IsFlagSet(webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET)) { + if (IsFlagSet(PORTALLOCATOR_ENABLE_SHARED_SOCKET)) { udp_port_ = port.get(); port->SubscribePortDestroyed( [this](PortInterface* port) { OnPortDestroyed(port); }); // If STUN is not disabled, setting stun server address to port. - if (!IsFlagSet(webrtc::PORTALLOCATOR_DISABLE_STUN)) { + if (!IsFlagSet(PORTALLOCATOR_DISABLE_STUN)) { if (config_ && !config_->StunServers().empty()) { RTC_LOG(LS_INFO) << "AllocationSequence: UDPPort will be handling the " @@ -1505,7 +1500,7 @@ void AllocationSequence::CreateUDPPorts() { } void AllocationSequence::CreateTCPPorts() { - if (IsFlagSet(webrtc::PORTALLOCATOR_DISABLE_TCP)) { + if (IsFlagSet(PORTALLOCATOR_DISABLE_TCP)) { RTC_LOG(LS_VERBOSE) << "AllocationSequence: TCP ports disabled, skipping."; return; } @@ -1528,12 +1523,12 @@ void AllocationSequence::CreateTCPPorts() { } void AllocationSequence::CreateStunPorts() { - if (IsFlagSet(webrtc::PORTALLOCATOR_DISABLE_STUN)) { + if (IsFlagSet(PORTALLOCATOR_DISABLE_STUN)) { RTC_LOG(LS_VERBOSE) << "AllocationSequence: STUN ports disabled, skipping."; return; } - if (IsFlagSet(webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET)) { + if (IsFlagSet(PORTALLOCATOR_ENABLE_SHARED_SOCKET)) { return; } @@ -1562,7 +1557,7 @@ void AllocationSequence::CreateStunPorts() { } void AllocationSequence::CreateRelayPorts() { - if (IsFlagSet(webrtc::PORTALLOCATOR_DISABLE_RELAY)) { + if (IsFlagSet(PORTALLOCATOR_DISABLE_RELAY)) { RTC_LOG(LS_VERBOSE) << "AllocationSequence: Relay ports disabled, skipping."; return; @@ -1593,8 +1588,8 @@ void AllocationSequence::CreateTurnPort(const RelayServerConfig& config, for (relay_port = config.ports.begin(); relay_port != config.ports.end(); ++relay_port) { // Skip UDP connections to relay servers if it's disallowed. - if (IsFlagSet(webrtc::PORTALLOCATOR_DISABLE_UDP_RELAY) && - relay_port->proto == webrtc::PROTO_UDP) { + if (IsFlagSet(PORTALLOCATOR_DISABLE_UDP_RELAY) && + relay_port->proto == PROTO_UDP) { continue; } @@ -1627,8 +1622,8 @@ void AllocationSequence::CreateTurnPort(const RelayServerConfig& config, // don't pass shared socket for ports which will create TCP sockets. // TODO(mallinath) - Enable shared socket mode for TURN ports. Disabled // due to webrtc bug https://code.google.com/p/webrtc/issues/detail?id=3537 - if (IsFlagSet(webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET) && - relay_port->proto == webrtc::PROTO_UDP && udp_socket_) { + if (IsFlagSet(PORTALLOCATOR_ENABLE_SHARED_SOCKET) && + relay_port->proto == PROTO_UDP && udp_socket_) { port = session_->allocator()->relay_port_factory()->Create( args, udp_socket_.get()); @@ -1737,7 +1732,7 @@ ServerAddresses PortConfiguration::StunServers() { // Every UDP TURN server should also be used as a STUN server if // use_turn_server_as_stun_server is not disabled or the stun servers are // empty. - ServerAddresses turn_servers = GetRelayServerAddresses(webrtc::PROTO_UDP); + ServerAddresses turn_servers = GetRelayServerAddresses(PROTO_UDP); for (const SocketAddress& turn_server : turn_servers) { if (stun_servers.find(turn_server) == stun_servers.end()) { stun_servers.insert(turn_server); diff --git a/p2p/client/basic_port_allocator_unittest.cc b/p2p/client/basic_port_allocator_unittest.cc index 5cf1b6e5f1..7c4f834f95 100644 --- a/p2p/client/basic_port_allocator_unittest.cc +++ b/p2p/client/basic_port_allocator_unittest.cc @@ -172,11 +172,11 @@ class BasicPortAllocatorTestBase : public ::testing::Test, kTurnUdpExtAddr), candidate_allocation_done_(false) { allocator_.emplace(env_, &network_manager_, &socket_factory_); - allocator_->SetConfiguration({kStunAddr}, {}, 0, webrtc::NO_PRUNE, nullptr); + allocator_->SetConfiguration({kStunAddr}, {}, 0, NO_PRUNE, nullptr); allocator_->Initialize(); - allocator_->set_step_delay(webrtc::kMinimumStepDelay); - webrtc::metrics::Reset(); + allocator_->set_step_delay(kMinimumStepDelay); + metrics::Reset(); } void AddInterface(const SocketAddress& addr) { @@ -212,7 +212,7 @@ class BasicPortAllocatorTestBase : public ::testing::Test, void ResetWithNoServersOrNat() { allocator_.emplace(env_, &network_manager_, &socket_factory_); allocator_->Initialize(); - allocator_->set_step_delay(webrtc::kMinimumStepDelay); + allocator_->set_step_delay(kMinimumStepDelay); } // Endpoint is behind a NAT, with STUN specified. void ResetWithStunServerAndNat(const SocketAddress& stun_server) { @@ -236,10 +236,10 @@ class BasicPortAllocatorTestBase : public ::testing::Test, turn_server.credentials = credentials; if (!udp_turn.IsNil()) { - turn_server.ports.push_back(ProtocolAddress(udp_turn, webrtc::PROTO_UDP)); + turn_server.ports.push_back(ProtocolAddress(udp_turn, PROTO_UDP)); } if (!tcp_turn.IsNil()) { - turn_server.ports.push_back(ProtocolAddress(tcp_turn, webrtc::PROTO_TCP)); + turn_server.ports.push_back(ProtocolAddress(tcp_turn, PROTO_TCP)); } return turn_server; } @@ -307,7 +307,7 @@ class BasicPortAllocatorTestBase : public ::testing::Test, const SocketAddress& pattern) { return address.ipaddr() == pattern.ipaddr() && ((pattern.port() == 0 && - (address.port() != 0 || webrtc::IPIsAny(address.ipaddr()))) || + (address.port() != 0 || IPIsAny(address.ipaddr()))) || (pattern.port() != 0 && address.port() == pattern.port())); } @@ -489,11 +489,10 @@ class BasicPortAllocatorTestBase : public ::testing::Test, stun_servers.insert(stun_server); } allocator_.emplace(env_, &network_manager_, nat_socket_factory_.get()); - allocator_->SetConfiguration(stun_servers, {}, 0, webrtc::NO_PRUNE, - nullptr); + allocator_->SetConfiguration(stun_servers, {}, 0, NO_PRUNE, nullptr); allocator_->Initialize(); - allocator_->set_step_delay(webrtc::kMinimumStepDelay); + allocator_->set_step_delay(kMinimumStepDelay); } Environment env_ = CreateEnvironment(); @@ -542,15 +541,15 @@ class BasicPortAllocatorTest : public FakeClockBase, ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); } session_->set_flags(session_->flags() | - webrtc::PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION | - webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET); + PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION | + PORTALLOCATOR_ENABLE_SHARED_SOCKET); allocator().set_allow_tcp_listen(false); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); uint32_t total_candidates = 0; if (!host_candidate_addr.IsNil()) { @@ -561,7 +560,7 @@ class BasicPortAllocatorTest : public FakeClockBase, if (!stun_candidate_addr.IsNil()) { SocketAddress related_address(host_candidate_addr, 0); if (host_candidate_addr.IsNil()) { - related_address.SetIP(webrtc::GetAnyIP(stun_candidate_addr.family())); + related_address.SetIP(GetAnyIP(stun_candidate_addr.family())); } EXPECT_TRUE(HasCandidateWithRelatedAddr( candidates_, IceCandidateType::kSrflx, "udp", @@ -588,7 +587,7 @@ class BasicPortAllocatorTest : public FakeClockBase, } void TestIPv6TurnPortPrunesIPv4TurnPort() { - turn_server_.AddInternalSocket(kTurnUdpIntIPv6Addr, webrtc::PROTO_UDP); + turn_server_.AddInternalSocket(kTurnUdpIntIPv6Addr, PROTO_UDP); // Add two IP addresses on the same interface. AddInterface(kClientAddr, "net1"); AddInterface(kClientIPv6Addr, "net1"); @@ -596,32 +595,32 @@ class BasicPortAllocatorTest : public FakeClockBase, allocator_->Initialize(); allocator_->SetConfiguration(allocator_->stun_servers(), allocator_->turn_servers(), 0, - webrtc::PRUNE_BASED_ON_PRIORITY); + PRUNE_BASED_ON_PRIORITY); AddTurnServers(kTurnUdpIntIPv6Addr, SocketAddress()); AddTurnServers(kTurnUdpIntAddr, SocketAddress()); - allocator_->set_step_delay(webrtc::kMinimumStepDelay); + allocator_->set_step_delay(kMinimumStepDelay); allocator_->set_flags( - allocator().flags() | webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET | - webrtc::PORTALLOCATOR_ENABLE_IPV6 | webrtc::PORTALLOCATOR_DISABLE_TCP); + allocator().flags() | PORTALLOCATOR_ENABLE_SHARED_SOCKET | + PORTALLOCATOR_ENABLE_IPV6 | PORTALLOCATOR_DISABLE_TCP); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); // Three ports (one IPv4 STUN, one IPv6 STUN and one TURN) will be ready. EXPECT_EQ(3U, session_->ReadyPorts().size()); EXPECT_EQ(3U, ports_.size()); - EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kHost, webrtc::PROTO_UDP, - kClientAddr)); - EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kHost, webrtc::PROTO_UDP, + EXPECT_EQ( + 1, CountPorts(ports_, IceCandidateType::kHost, PROTO_UDP, kClientAddr)); + EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kHost, PROTO_UDP, kClientIPv6Addr)); - EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kRelay, webrtc::PROTO_UDP, + EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kRelay, PROTO_UDP, kClientIPv6Addr)); - EXPECT_EQ(0, CountPorts(ports_, IceCandidateType::kRelay, webrtc::PROTO_UDP, + EXPECT_EQ(0, CountPorts(ports_, IceCandidateType::kRelay, PROTO_UDP, kClientAddr)); // Now that we remove candidates when a TURN port is pruned, there will be @@ -638,38 +637,38 @@ class BasicPortAllocatorTest : public FakeClockBase, void TestTurnPortPrunesWithUdpAndTcpPorts(PortPrunePolicy prune_policy, bool tcp_pruned) { - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TCP); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP); AddInterface(kClientAddr); allocator_.emplace(env_, &network_manager_, &socket_factory_); allocator_->Initialize(); allocator_->SetConfiguration(allocator_->stun_servers(), allocator_->turn_servers(), 0, prune_policy); AddTurnServers(kTurnUdpIntAddr, kTurnTcpIntAddr); - allocator_->set_step_delay(webrtc::kMinimumStepDelay); + allocator_->set_step_delay(kMinimumStepDelay); allocator_->set_flags(allocator().flags() | - webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET | - webrtc::PORTALLOCATOR_DISABLE_TCP); + PORTALLOCATOR_ENABLE_SHARED_SOCKET | + PORTALLOCATOR_DISABLE_TCP); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); // Only 2 ports (one STUN and one TURN) are actually being used. EXPECT_EQ(2U, session_->ReadyPorts().size()); // We have verified that each port, when it is added to `ports_`, it is // found in `ready_ports`, and when it is pruned, it is not found in // `ready_ports`, so we only need to verify the content in one of them. EXPECT_EQ(2U, ports_.size()); - EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kHost, webrtc::PROTO_UDP, - kClientAddr)); + EXPECT_EQ( + 1, CountPorts(ports_, IceCandidateType::kHost, PROTO_UDP, kClientAddr)); int num_udp_ports = tcp_pruned ? 1 : 0; EXPECT_EQ(num_udp_ports, CountPorts(ports_, IceCandidateType::kRelay, - webrtc::PROTO_UDP, kClientAddr)); + PROTO_UDP, kClientAddr)); EXPECT_EQ(1 - num_udp_ports, CountPorts(ports_, IceCandidateType::kRelay, - webrtc::PROTO_TCP, kClientAddr)); + PROTO_TCP, kClientAddr)); // Now that we remove candidates when a TURN port is pruned, `candidates_` // should only contains two candidates regardless whether the TCP TURN port @@ -689,9 +688,9 @@ class BasicPortAllocatorTest : public FakeClockBase, } void TestEachInterfaceHasItsOwnTurnPorts() { - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TCP); - turn_server_.AddInternalSocket(kTurnUdpIntIPv6Addr, webrtc::PROTO_UDP); - turn_server_.AddInternalSocket(kTurnTcpIntIPv6Addr, webrtc::PROTO_TCP); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP); + turn_server_.AddInternalSocket(kTurnUdpIntIPv6Addr, PROTO_UDP); + turn_server_.AddInternalSocket(kTurnTcpIntIPv6Addr, PROTO_TCP); // Add two interfaces both having IPv4 and IPv6 addresses. AddInterface(kClientAddr, "net1", ADAPTER_TYPE_WIFI); AddInterface(kClientIPv6Addr, "net1", ADAPTER_TYPE_WIFI); @@ -701,46 +700,45 @@ class BasicPortAllocatorTest : public FakeClockBase, allocator_->Initialize(); allocator_->SetConfiguration(allocator_->stun_servers(), allocator_->turn_servers(), 0, - webrtc::PRUNE_BASED_ON_PRIORITY); + PRUNE_BASED_ON_PRIORITY); // Have both UDP/TCP and IPv4/IPv6 TURN ports. AddTurnServers(kTurnUdpIntAddr, kTurnTcpIntAddr); AddTurnServers(kTurnUdpIntIPv6Addr, kTurnTcpIntIPv6Addr); - allocator_->set_step_delay(webrtc::kMinimumStepDelay); - allocator_->set_flags(allocator().flags() | - webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET | - webrtc::PORTALLOCATOR_ENABLE_IPV6 | - webrtc::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI); + allocator_->set_step_delay(kMinimumStepDelay); + allocator_->set_flags( + allocator().flags() | PORTALLOCATOR_ENABLE_SHARED_SOCKET | + PORTALLOCATOR_ENABLE_IPV6 | PORTALLOCATOR_ENABLE_IPV6_ON_WIFI); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); // 10 ports (4 STUN and 1 TURN ports on each interface) will be ready to // use. EXPECT_EQ(10U, session_->ReadyPorts().size()); EXPECT_EQ(10U, ports_.size()); - EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kHost, webrtc::PROTO_UDP, - kClientAddr)); - EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kHost, webrtc::PROTO_UDP, + EXPECT_EQ( + 1, CountPorts(ports_, IceCandidateType::kHost, PROTO_UDP, kClientAddr)); + EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kHost, PROTO_UDP, kClientAddr2)); - EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kHost, webrtc::PROTO_UDP, + EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kHost, PROTO_UDP, kClientIPv6Addr)); - EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kHost, webrtc::PROTO_UDP, + EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kHost, PROTO_UDP, kClientIPv6Addr2)); - EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kHost, webrtc::PROTO_TCP, - kClientAddr)); - EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kHost, webrtc::PROTO_TCP, + EXPECT_EQ( + 1, CountPorts(ports_, IceCandidateType::kHost, PROTO_TCP, kClientAddr)); + EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kHost, PROTO_TCP, kClientAddr2)); - EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kHost, webrtc::PROTO_TCP, + EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kHost, PROTO_TCP, kClientIPv6Addr)); - EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kHost, webrtc::PROTO_TCP, + EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kHost, PROTO_TCP, kClientIPv6Addr2)); - EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kRelay, webrtc::PROTO_UDP, + EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kRelay, PROTO_UDP, kClientIPv6Addr)); - EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kRelay, webrtc::PROTO_UDP, + EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kRelay, PROTO_UDP, kClientIPv6Addr2)); // Now that we remove candidates when TURN ports are pruned, there will be @@ -793,15 +791,14 @@ TEST_F(BasicPortAllocatorTest, TestIgnoreOnlyLoopbackNetworkByDefault) { AddInterface(SocketAddress(IPAddress(0x12345604U), 0), "test_lo", ADAPTER_TYPE_LOOPBACK); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); - session_->set_flags(webrtc::PORTALLOCATOR_DISABLE_STUN | - webrtc::PORTALLOCATOR_DISABLE_RELAY | - webrtc::PORTALLOCATOR_DISABLE_TCP); + session_->set_flags(PORTALLOCATOR_DISABLE_STUN | PORTALLOCATOR_DISABLE_RELAY | + PORTALLOCATOR_DISABLE_TCP); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(4U, candidates_.size()); for (const Candidate& candidate : candidates_) { EXPECT_LT(candidate.address().ip(), 0x12345604U); @@ -818,15 +815,14 @@ TEST_F(BasicPortAllocatorTest, TestIgnoreNetworksAccordingToIgnoreMask) { allocator_->SetNetworkIgnoreMask(ADAPTER_TYPE_ETHERNET | ADAPTER_TYPE_LOOPBACK | ADAPTER_TYPE_WIFI); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); - session_->set_flags(webrtc::PORTALLOCATOR_DISABLE_STUN | - webrtc::PORTALLOCATOR_DISABLE_RELAY | - webrtc::PORTALLOCATOR_DISABLE_TCP); + session_->set_flags(PORTALLOCATOR_DISABLE_STUN | PORTALLOCATOR_DISABLE_RELAY | + PORTALLOCATOR_DISABLE_TCP); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(1U, candidates_.size()); EXPECT_EQ(0x12345602U, candidates_[0].address().ip()); } @@ -840,17 +836,16 @@ TEST_F(BasicPortAllocatorTest, AddInterface(wifi, "test_wlan0", ADAPTER_TYPE_WIFI); AddInterface(cell, "test_cell0", ADAPTER_TYPE_CELLULAR); // Disable all but UDP candidates to make the test simpler. - allocator().set_flags(webrtc::PORTALLOCATOR_DISABLE_STUN | - webrtc::PORTALLOCATOR_DISABLE_RELAY | - webrtc::PORTALLOCATOR_DISABLE_TCP | - webrtc::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS); + allocator().set_flags( + PORTALLOCATOR_DISABLE_STUN | PORTALLOCATOR_DISABLE_RELAY | + PORTALLOCATOR_DISABLE_TCP | PORTALLOCATOR_DISABLE_COSTLY_NETWORKS); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); // Should only get one Wi-Fi candidate. EXPECT_EQ(1U, candidates_.size()); EXPECT_TRUE(HasCandidate(candidates_, IceCandidateType::kHost, "udp", wifi)); @@ -869,17 +864,16 @@ TEST_F(BasicPortAllocatorTest, AddInterface(unknown1, "test_unknown0", ADAPTER_TYPE_UNKNOWN); AddInterface(unknown2, "test_unknown1", ADAPTER_TYPE_UNKNOWN); // Disable all but UDP candidates to make the test simpler. - allocator().set_flags(webrtc::PORTALLOCATOR_DISABLE_STUN | - webrtc::PORTALLOCATOR_DISABLE_RELAY | - webrtc::PORTALLOCATOR_DISABLE_TCP | - webrtc::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS); + allocator().set_flags( + PORTALLOCATOR_DISABLE_STUN | PORTALLOCATOR_DISABLE_RELAY | + PORTALLOCATOR_DISABLE_TCP | PORTALLOCATOR_DISABLE_COSTLY_NETWORKS); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); // Should only get two candidates, none of which is cell. EXPECT_EQ(2U, candidates_.size()); EXPECT_TRUE( @@ -902,17 +896,16 @@ TEST_F(BasicPortAllocatorTest, AddInterface(unknown1, "test_unknown0", ADAPTER_TYPE_UNKNOWN); AddInterface(unknown2, "test_unknown1", ADAPTER_TYPE_UNKNOWN); // Disable all but UDP candidates to make the test simpler. - allocator().set_flags(webrtc::PORTALLOCATOR_DISABLE_STUN | - webrtc::PORTALLOCATOR_DISABLE_RELAY | - webrtc::PORTALLOCATOR_DISABLE_TCP | - webrtc::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS); + allocator().set_flags( + PORTALLOCATOR_DISABLE_STUN | PORTALLOCATOR_DISABLE_RELAY | + PORTALLOCATOR_DISABLE_TCP | PORTALLOCATOR_DISABLE_COSTLY_NETWORKS); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); // Should only get one Wi-Fi candidate. EXPECT_EQ(1U, candidates_.size()); EXPECT_TRUE(HasCandidate(candidates_, IceCandidateType::kHost, "udp", wifi)); @@ -926,17 +919,16 @@ TEST_F(BasicPortAllocatorTest, SocketAddress cellular(IPAddress(0x12345601U), 0); AddInterface(cellular, "test_cell0", ADAPTER_TYPE_CELLULAR); // Disable all but UDP candidates to make the test simpler. - allocator().set_flags(webrtc::PORTALLOCATOR_DISABLE_STUN | - webrtc::PORTALLOCATOR_DISABLE_RELAY | - webrtc::PORTALLOCATOR_DISABLE_TCP | - webrtc::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS); + allocator().set_flags( + PORTALLOCATOR_DISABLE_STUN | PORTALLOCATOR_DISABLE_RELAY | + PORTALLOCATOR_DISABLE_TCP | PORTALLOCATOR_DISABLE_COSTLY_NETWORKS); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); // Make sure we got the cell candidate. EXPECT_EQ(1U, candidates_.size()); EXPECT_TRUE( @@ -953,17 +945,16 @@ TEST_F(BasicPortAllocatorTest, AddInterface(wifi_link_local, "test_wlan0", ADAPTER_TYPE_WIFI); AddInterface(cellular, "test_cell0", ADAPTER_TYPE_CELLULAR); - allocator().set_flags(webrtc::PORTALLOCATOR_DISABLE_STUN | - webrtc::PORTALLOCATOR_DISABLE_RELAY | - webrtc::PORTALLOCATOR_DISABLE_TCP | - webrtc::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS); + allocator().set_flags( + PORTALLOCATOR_DISABLE_STUN | PORTALLOCATOR_DISABLE_RELAY | + PORTALLOCATOR_DISABLE_TCP | PORTALLOCATOR_DISABLE_COSTLY_NETWORKS); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); // Make sure we got both wifi and cell candidates. EXPECT_EQ(2U, candidates_.size()); EXPECT_TRUE(HasCandidate(candidates_, IceCandidateType::kHost, "udp", @@ -985,17 +976,16 @@ TEST_F(BasicPortAllocatorTest, AddInterface(wifi_link_local, "test_wlan1", ADAPTER_TYPE_WIFI); AddInterface(cellular, "test_cell0", ADAPTER_TYPE_CELLULAR); - allocator().set_flags(webrtc::PORTALLOCATOR_DISABLE_STUN | - webrtc::PORTALLOCATOR_DISABLE_RELAY | - webrtc::PORTALLOCATOR_DISABLE_TCP | - webrtc::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS); + allocator().set_flags( + PORTALLOCATOR_DISABLE_STUN | PORTALLOCATOR_DISABLE_RELAY | + PORTALLOCATOR_DISABLE_TCP | PORTALLOCATOR_DISABLE_COSTLY_NETWORKS); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); // Make sure we got only wifi candidates. EXPECT_EQ(2U, candidates_.size()); EXPECT_TRUE(HasCandidate(candidates_, IceCandidateType::kHost, "udp", wifi)); @@ -1011,16 +1001,16 @@ TEST_F(BasicPortAllocatorTest, EthernetIsNotFilteredOutWhenCostlyNetworksDisabledAndVpnPresent) { AddInterface(kClientAddr, "eth0", ADAPTER_TYPE_ETHERNET); AddInterface(kClientAddr2, "tap0", ADAPTER_TYPE_VPN); - allocator().set_flags(webrtc::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS | - webrtc::PORTALLOCATOR_DISABLE_RELAY | - webrtc::PORTALLOCATOR_DISABLE_TCP); + allocator().set_flags(PORTALLOCATOR_DISABLE_COSTLY_NETWORKS | + PORTALLOCATOR_DISABLE_RELAY | + PORTALLOCATOR_DISABLE_TCP); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); // The VPN tap0 network should be filtered out as a costly network, and we // should have a UDP port and a STUN port from the Ethernet eth0. ASSERT_EQ(2U, ports_.size()); @@ -1038,17 +1028,17 @@ TEST_F(BasicPortAllocatorTest, MaxIpv6NetworksLimitEnforced) { AddInterface(kClientIPv6Addr3, "eth2", ADAPTER_TYPE_ETHERNET); // To simplify the test, only gather UDP host candidates. - allocator().set_flags( - webrtc::PORTALLOCATOR_ENABLE_IPV6 | webrtc::PORTALLOCATOR_DISABLE_TCP | - webrtc::PORTALLOCATOR_DISABLE_STUN | webrtc::PORTALLOCATOR_DISABLE_RELAY); + allocator().set_flags(PORTALLOCATOR_ENABLE_IPV6 | PORTALLOCATOR_DISABLE_TCP | + PORTALLOCATOR_DISABLE_STUN | + PORTALLOCATOR_DISABLE_RELAY); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(2U, candidates_.size()); // Ensure the expected two interfaces (eth0 and eth1) were used. EXPECT_TRUE(HasCandidate(candidates_, IceCandidateType::kHost, "udp", @@ -1068,17 +1058,17 @@ TEST_F(BasicPortAllocatorTest, MaxIpv6NetworksLimitDoesNotImpactIpv4Networks) { AddInterface(kClientAddr2, "eth3", ADAPTER_TYPE_ETHERNET); // To simplify the test, only gather UDP host candidates. - allocator().set_flags( - webrtc::PORTALLOCATOR_ENABLE_IPV6 | webrtc::PORTALLOCATOR_DISABLE_TCP | - webrtc::PORTALLOCATOR_DISABLE_STUN | webrtc::PORTALLOCATOR_DISABLE_RELAY); + allocator().set_flags(PORTALLOCATOR_ENABLE_IPV6 | PORTALLOCATOR_DISABLE_TCP | + PORTALLOCATOR_DISABLE_STUN | + PORTALLOCATOR_DISABLE_RELAY); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(3U, candidates_.size()); // Ensure that only one IPv6 interface was used, but both IPv4 interfaces // were used. @@ -1095,15 +1085,14 @@ TEST_F(BasicPortAllocatorTest, TestLoopbackNetworkInterface) { AddInterface(kLoopbackAddr, "test_loopback", ADAPTER_TYPE_LOOPBACK); allocator_->SetNetworkIgnoreMask(0); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); - session_->set_flags(webrtc::PORTALLOCATOR_DISABLE_STUN | - webrtc::PORTALLOCATOR_DISABLE_RELAY | - webrtc::PORTALLOCATOR_DISABLE_TCP); + session_->set_flags(PORTALLOCATOR_DISABLE_STUN | PORTALLOCATOR_DISABLE_RELAY | + PORTALLOCATOR_DISABLE_TCP); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(1U, candidates_.size()); } @@ -1112,11 +1101,11 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsWithMinimumStepDelay) { AddInterface(kClientAddr); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(3U, candidates_.size()); EXPECT_EQ(3U, ports_.size()); EXPECT_TRUE( @@ -1135,11 +1124,11 @@ TEST_F(BasicPortAllocatorTest, TestSameNetworkDownAndUpWhenSessionNotStopped) { AddInterface(kClientAddr, if_name); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(3U, candidates_.size()); EXPECT_EQ(3U, ports_.size()); candidate_allocation_done_ = false; @@ -1162,11 +1151,11 @@ TEST_F(BasicPortAllocatorTest, TestSameNetworkDownAndUpWhenSessionNotStopped) { fss_->set_tcp_sockets_enabled(true); fss_->set_udp_sockets_enabled(true); AddInterface(kClientAddr, if_name); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(3U, candidates_.size()); EXPECT_EQ(3U, ports_.size()); } @@ -1179,11 +1168,11 @@ TEST_F(BasicPortAllocatorTest, TestSameNetworkDownAndUpWhenSessionStopped) { AddInterface(kClientAddr, if_name); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(3U, candidates_.size()); EXPECT_EQ(3U, ports_.size()); session_->StopGettingPorts(); @@ -1220,11 +1209,11 @@ TEST_F(BasicPortAllocatorTest, CandidatesRegatheredAfterBindingFails) { fss_->set_udp_sockets_enabled(false); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); // Make sure we actually prevented candidates from being gathered (other than // a single TCP active candidate, since that doesn't require creating a // socket). @@ -1238,11 +1227,11 @@ TEST_F(BasicPortAllocatorTest, CandidatesRegatheredAfterBindingFails) { fss_->set_tcp_sockets_enabled(true); fss_->set_udp_sockets_enabled(true); AddInterface(kClientAddr, if_name); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); // Should get UDP and TCP candidate. ASSERT_EQ(2U, candidates_.size()); EXPECT_TRUE( @@ -1257,21 +1246,21 @@ TEST_F(BasicPortAllocatorTest, CandidatesRegatheredAfterBindingFails) { // Verify candidates with default step delay of 1sec. TEST_F(BasicPortAllocatorTest, TestGetAllPortsWithOneSecondStepDelay) { AddInterface(kClientAddr); - allocator_->set_step_delay(webrtc::kDefaultStepDelay); + allocator_->set_step_delay(kDefaultStepDelay); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil([&] { return candidates_.size(); }, Eq(2U), - {.clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT(WaitUntil([&] { return candidates_.size(); }, Eq(2U), + {.clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(2U, ports_.size()); - ASSERT_THAT(webrtc::WaitUntil([&] { return candidates_.size(); }, Eq(3U), - {.clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT(WaitUntil([&] { return candidates_.size(); }, Eq(3U), + {.clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(3U, ports_.size()); - ASSERT_THAT(webrtc::WaitUntil([&] { return candidates_.size(); }, Eq(3U), - {.clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT(WaitUntil([&] { return candidates_.size(); }, Eq(3U), + {.clock = &fake_clock}), + IsRtcOk()); EXPECT_TRUE( HasCandidate(candidates_, IceCandidateType::kHost, "tcp", kClientAddr)); EXPECT_EQ(3U, ports_.size()); @@ -1284,11 +1273,11 @@ TEST_F(BasicPortAllocatorTest, TestSetupVideoRtpPortsWithNormalSendBuffers) { AddInterface(kClientAddr); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP, CN_VIDEO)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(3U, candidates_.size()); // If we Stop gathering now, we shouldn't get a second "done" callback. session_->StopGettingPorts(); @@ -1303,18 +1292,18 @@ TEST_F(BasicPortAllocatorTest, TestStopGetAllPorts) { AddInterface(kClientAddr); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidates_.size(); }, Eq(2U), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return candidates_.size(); }, Eq(2U), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(2U, ports_.size()); session_->StopGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); } // Test that we restrict client ports appropriately when a port range is set. @@ -1330,11 +1319,11 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsPortRange) { EXPECT_TRUE(SetPortRange(kMinPort, kMaxPort)); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(3U, candidates_.size()); EXPECT_EQ(3U, ports_.size()); @@ -1355,26 +1344,26 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoAdapters) { // Default config uses GTURN and no NAT, so replace that with the // desired setup (NAT, STUN server, TURN server, UDP/TCP). ResetWithStunServerAndNat(kStunAddr); - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TCP); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP); AddTurnServers(kTurnUdpIntAddr, kTurnTcpIntAddr); AddTurnServers(kTurnUdpIntIPv6Addr, kTurnTcpIntIPv6Addr); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(4U, ports_.size()); - EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kSrflx, webrtc::PROTO_UDP, - kAnyAddr)); - EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kHost, webrtc::PROTO_TCP, - kAnyAddr)); + EXPECT_EQ(1, + CountPorts(ports_, IceCandidateType::kSrflx, PROTO_UDP, kAnyAddr)); + EXPECT_EQ(1, + CountPorts(ports_, IceCandidateType::kHost, PROTO_TCP, kAnyAddr)); // Two TURN ports, using UDP/TCP for the first hop to the TURN server. - EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kRelay, webrtc::PROTO_UDP, - kAnyAddr)); - EXPECT_EQ(1, CountPorts(ports_, IceCandidateType::kRelay, webrtc::PROTO_TCP, - kAnyAddr)); + EXPECT_EQ(1, + CountPorts(ports_, IceCandidateType::kRelay, PROTO_UDP, kAnyAddr)); + EXPECT_EQ(1, + CountPorts(ports_, IceCandidateType::kRelay, PROTO_TCP, kAnyAddr)); // The "any" address port should be in the signaled ready ports, but the host // candidate for it is useless and shouldn't be signaled. So we only have // STUN/TURN candidates. @@ -1395,7 +1384,7 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoAdapters) { TEST_F(BasicPortAllocatorTest, TestDisableAdapterEnumerationWithoutNatRelayTransportOnly) { ResetWithStunServerNoNat(kStunAddr); - allocator().SetCandidateFilter(webrtc::CF_RELAY); + allocator().SetCandidateFilter(CF_RELAY); // Expect to see no ports and no candidates. CheckDisableAdapterEnumeration(0U, IPAddress(), IPAddress(), IPAddress(), IPAddress()); @@ -1415,7 +1404,7 @@ TEST_F(BasicPortAllocatorTest, // address set and we have no IPv6 STUN server, there should be no IPv6 // candidates. ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); - session_->set_flags(webrtc::PORTALLOCATOR_ENABLE_IPV6); + session_->set_flags(PORTALLOCATOR_ENABLE_IPV6); // Expect to see 3 ports for IPv4: HOST/STUN, TURN/UDP and TCP ports, 2 ports // for IPv6: HOST, and TCP. Only IPv4 candidates: a default private, STUN and @@ -1428,7 +1417,7 @@ TEST_F(BasicPortAllocatorTest, // Test that we should get a default private, STUN, TURN/UDP and TURN/TCP // candidates when both TURN/UDP and TURN/TCP servers are specified. TEST_F(BasicPortAllocatorTest, TestDisableAdapterEnumerationBehindNatWithTcp) { - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TCP); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP); AddInterface(kPrivateAddr); ResetWithStunServerAndNat(kStunAddr); AddTurnServers(kTurnUdpIntAddr, kTurnTcpIntAddr); @@ -1456,7 +1445,7 @@ TEST_F(BasicPortAllocatorTest, TestDisableAdapterEnumerationWithoutNatLocalhostCandidateDisabled) { ResetWithStunServerNoNat(kStunAddr); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); - session_->set_flags(webrtc::PORTALLOCATOR_DISABLE_DEFAULT_LOCAL_CANDIDATE); + session_->set_flags(PORTALLOCATOR_DISABLE_DEFAULT_LOCAL_CANDIDATE); // Expect to see 2 ports: STUN and TCP ports, localhost candidate and STUN // candidate. CheckDisableAdapterEnumeration(2U, IPAddress(), IPAddress(), IPAddress(), @@ -1473,7 +1462,7 @@ TEST_F(BasicPortAllocatorTest, ResetWithStunServerNoNat(kStunAddr); AddInterfaceAsDefaultSourceAddresss(kClientAddr); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); - session_->set_flags(webrtc::PORTALLOCATOR_DISABLE_DEFAULT_LOCAL_CANDIDATE); + session_->set_flags(PORTALLOCATOR_DISABLE_DEFAULT_LOCAL_CANDIDATE); // Expect to see 2 ports: STUN and TCP ports, localhost candidate and STUN // candidate. CheckDisableAdapterEnumeration(2U, IPAddress(), kClientAddr.ipaddr(), @@ -1487,7 +1476,7 @@ TEST_F(BasicPortAllocatorTest, TestDisableAdapterEnumerationWithNatLocalhostCandidateDisabled) { ResetWithStunServerAndNat(kStunAddr); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); - session_->set_flags(webrtc::PORTALLOCATOR_DISABLE_DEFAULT_LOCAL_CANDIDATE); + session_->set_flags(PORTALLOCATOR_DISABLE_DEFAULT_LOCAL_CANDIDATE); // Expect to see 2 ports: STUN and TCP ports, and single STUN candidate. CheckDisableAdapterEnumeration(2U, IPAddress(), kNatUdpAddr.ipaddr(), IPAddress(), IPAddress()); @@ -1496,22 +1485,21 @@ TEST_F(BasicPortAllocatorTest, // Test that we disable relay over UDP, and only TCP is used when connecting to // the relay server. TEST_F(BasicPortAllocatorTest, TestDisableUdpTurn) { - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TCP); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP); AddInterface(kClientAddr); ResetWithStunServerAndNat(kStunAddr); AddTurnServers(kTurnUdpIntAddr, kTurnTcpIntAddr); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); - session_->set_flags(webrtc::PORTALLOCATOR_DISABLE_UDP_RELAY | - webrtc::PORTALLOCATOR_DISABLE_UDP | - webrtc::PORTALLOCATOR_DISABLE_STUN | - webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET); + session_->set_flags(PORTALLOCATOR_DISABLE_UDP_RELAY | + PORTALLOCATOR_DISABLE_UDP | PORTALLOCATOR_DISABLE_STUN | + PORTALLOCATOR_ENABLE_SHARED_SOCKET); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); // Expect to see 2 ports and 2 candidates - TURN/TCP and TCP ports, TCP and // TURN/TCP candidates. @@ -1521,7 +1509,7 @@ TEST_F(BasicPortAllocatorTest, TestDisableUdpTurn) { EXPECT_TRUE(FindCandidate(candidates_, IceCandidateType::kRelay, "udp", kTurnUdpExtAddr, &turn_candidate)); // The TURN candidate should use TCP to contact the TURN server. - EXPECT_EQ(webrtc::TCP_PROTOCOL_NAME, turn_candidate.relay_protocol()); + EXPECT_EQ(TCP_PROTOCOL_NAME, turn_candidate.relay_protocol()); EXPECT_TRUE( HasCandidate(candidates_, IceCandidateType::kHost, "tcp", kClientAddr)); } @@ -1531,13 +1519,12 @@ TEST_F(BasicPortAllocatorTest, TestDisableUdpTurn) { TEST_F(BasicPortAllocatorTest, TestDisableAllPorts) { AddInterface(kClientAddr); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); - session_->set_flags( - webrtc::PORTALLOCATOR_DISABLE_UDP | webrtc::PORTALLOCATOR_DISABLE_STUN | - webrtc::PORTALLOCATOR_DISABLE_RELAY | webrtc::PORTALLOCATOR_DISABLE_TCP); + session_->set_flags(PORTALLOCATOR_DISABLE_UDP | PORTALLOCATOR_DISABLE_STUN | + PORTALLOCATOR_DISABLE_RELAY | PORTALLOCATOR_DISABLE_TCP); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil([&] { return candidate_allocation_done_; }, - IsTrue(), {.clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(0U, candidates_.size()); } @@ -1547,11 +1534,11 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoUdpSockets) { fss_->set_udp_sockets_enabled(false); ASSERT_TRUE(CreateSession(1)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(1U, candidates_.size()); EXPECT_EQ(1U, ports_.size()); EXPECT_TRUE( @@ -1567,11 +1554,11 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoUdpSocketsNoTcpListen) { fss_->set_tcp_listen_enabled(false); ASSERT_TRUE(CreateSession(1)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(1U, candidates_.size()); EXPECT_EQ(1U, ports_.size()); EXPECT_TRUE( @@ -1594,15 +1581,15 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoSockets) { // Testing STUN timeout. TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoUdpAllowed) { - fss_->AddRule(false, webrtc::FP_UDP, webrtc::FD_ANY, kClientAddr); + fss_->AddRule(false, FP_UDP, FD_ANY, kClientAddr); AddInterface(kClientAddr); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidates_.size(); }, Eq(2U), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidates_.size(); }, Eq(2U), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(2U, ports_.size()); EXPECT_TRUE( HasCandidate(candidates_, IceCandidateType::kHost, "udp", kClientAddr)); @@ -1610,11 +1597,10 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoUdpAllowed) { HasCandidate(candidates_, IceCandidateType::kHost, "tcp", kClientAddr)); // We wait at least for a full STUN timeout, which // webrtc::STUN_TOTAL_TIMEOUT seconds. - EXPECT_THAT( - webrtc::WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + EXPECT_THAT(WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), {.timeout = TimeDelta::Millis(STUN_TOTAL_TIMEOUT), .clock = &fake_clock}), - webrtc::IsRtcOk()); + IsRtcOk()); // No additional (STUN) candidates. EXPECT_EQ(2U, candidates_.size()); } @@ -1624,16 +1610,15 @@ TEST_F(BasicPortAllocatorTest, TestCandidatePriorityOfMultipleInterfaces) { AddInterface(kClientAddr2); // Allocating only host UDP ports. This is done purely for testing // convenience. - allocator().set_flags(webrtc::PORTALLOCATOR_DISABLE_TCP | - webrtc::PORTALLOCATOR_DISABLE_STUN | - webrtc::PORTALLOCATOR_DISABLE_RELAY); + allocator().set_flags(PORTALLOCATOR_DISABLE_TCP | PORTALLOCATOR_DISABLE_STUN | + PORTALLOCATOR_DISABLE_RELAY); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); ASSERT_EQ(2U, candidates_.size()); EXPECT_EQ(2U, ports_.size()); // Candidates priorities should be different. @@ -1645,11 +1630,11 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsRestarts) { AddInterface(kClientAddr); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(3U, candidates_.size()); EXPECT_EQ(3U, ports_.size()); // TODO(deadbeef): Extend this to verify ICE restart. @@ -1663,15 +1648,15 @@ TEST_F(BasicPortAllocatorTest, TestSessionUsesOwnCandidateFilter) { AddInterface(kClientAddr); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); // Set candidate filter *after* creating the session. Should have no effect. - allocator().SetCandidateFilter(webrtc::CF_RELAY); + allocator().SetCandidateFilter(CF_RELAY); session_->StartGettingPorts(); // 7 candidates and 4 ports is what we would normally get (see the // TestGetAllPorts* tests). - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(3U, candidates_.size()); EXPECT_EQ(3U, ports_.size()); } @@ -1685,36 +1670,35 @@ TEST_F(BasicPortAllocatorTest, TestCandidateFilterWithRelayOnly) { AddInterface(kClientAddr); // GTURN is not configured here. ResetWithTurnServersNoNat(kTurnUdpIntAddr, SocketAddress()); - allocator().SetCandidateFilter(webrtc::CF_RELAY); + allocator().SetCandidateFilter(CF_RELAY); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_TRUE(HasCandidate(candidates_, IceCandidateType::kRelay, "udp", SocketAddress(kTurnUdpExtAddr.ipaddr(), 0))); EXPECT_EQ(1U, candidates_.size()); EXPECT_EQ(1U, ports_.size()); // Only Relay port will be in ready state. EXPECT_TRUE(candidates_[0].is_relay()); - EXPECT_EQ( - candidates_[0].related_address(), - webrtc::EmptySocketAddressWithFamily(candidates_[0].address().family())); + EXPECT_EQ(candidates_[0].related_address(), + EmptySocketAddressWithFamily(candidates_[0].address().family())); } TEST_F(BasicPortAllocatorTest, TestCandidateFilterWithHostOnly) { AddInterface(kClientAddr); - allocator().set_flags(webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET); - allocator().SetCandidateFilter(webrtc::CF_HOST); + allocator().set_flags(PORTALLOCATOR_ENABLE_SHARED_SOCKET); + allocator().SetCandidateFilter(CF_HOST); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(2U, candidates_.size()); // Host UDP/TCP candidates only. EXPECT_EQ(2U, ports_.size()); // UDP/TCP ports only. for (const Candidate& candidate : candidates_) { @@ -1727,37 +1711,36 @@ TEST_F(BasicPortAllocatorTest, TestCandidateFilterWithReflexiveOnly) { AddInterface(kPrivateAddr); ResetWithStunServerAndNat(kStunAddr); - allocator().set_flags(webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET); - allocator().SetCandidateFilter(webrtc::CF_REFLEXIVE); + allocator().set_flags(PORTALLOCATOR_ENABLE_SHARED_SOCKET); + allocator().SetCandidateFilter(CF_REFLEXIVE); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); // Host is behind NAT, no private address will be exposed. Hence only UDP // port with STUN candidate will be sent outside. EXPECT_EQ(1U, candidates_.size()); // Only STUN candidate. EXPECT_EQ(1U, ports_.size()); // Only UDP port will be in ready state. EXPECT_TRUE(candidates_[0].is_stun()); - EXPECT_EQ( - candidates_[0].related_address(), - webrtc::EmptySocketAddressWithFamily(candidates_[0].address().family())); + EXPECT_EQ(candidates_[0].related_address(), + EmptySocketAddressWithFamily(candidates_[0].address().family())); } // Host is not behind the NAT. TEST_F(BasicPortAllocatorTest, TestCandidateFilterWithReflexiveOnlyAndNoNAT) { AddInterface(kClientAddr); - allocator().set_flags(webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET); - allocator().SetCandidateFilter(webrtc::CF_REFLEXIVE); + allocator().set_flags(PORTALLOCATOR_ENABLE_SHARED_SOCKET); + allocator().SetCandidateFilter(CF_REFLEXIVE); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); // Host has a public address, both UDP and TCP candidates will be exposed. EXPECT_EQ(2U, candidates_.size()); // Local UDP + TCP candidate. EXPECT_EQ(2U, ports_.size()); // UDP and TCP ports will be in ready state. @@ -1771,11 +1754,11 @@ TEST_F(BasicPortAllocatorTest, TestEnableSharedUfrag) { AddInterface(kClientAddr); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(3U, candidates_.size()); EXPECT_TRUE( HasCandidate(candidates_, IceCandidateType::kHost, "udp", kClientAddr)); @@ -1797,22 +1780,22 @@ TEST_F(BasicPortAllocatorTest, TestEnableSharedUfrag) { TEST_F(BasicPortAllocatorTest, TestSharedSocketWithoutNat) { AddInterface(kClientAddr); allocator_->set_flags(allocator().flags() | - webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET); + PORTALLOCATOR_ENABLE_SHARED_SOCKET); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidates_.size(); }, Eq(2U), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return candidates_.size(); }, Eq(2U), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(2U, ports_.size()); EXPECT_TRUE( HasCandidate(candidates_, IceCandidateType::kHost, "udp", kClientAddr)); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); } // Test that when PORTALLOCATOR_ENABLE_SHARED_SOCKET is enabled only one port @@ -1823,49 +1806,49 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketWithNat) { ResetWithStunServerAndNat(kStunAddr); allocator_->set_flags(allocator().flags() | - webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET); + PORTALLOCATOR_ENABLE_SHARED_SOCKET); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidates_.size(); }, Eq(3U), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return candidates_.size(); }, Eq(3U), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); ASSERT_EQ(2U, ports_.size()); EXPECT_TRUE( HasCandidate(candidates_, IceCandidateType::kHost, "udp", kClientAddr)); EXPECT_TRUE(HasCandidate(candidates_, IceCandidateType::kSrflx, "udp", SocketAddress(kNatUdpAddr.ipaddr(), 0))); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(3U, candidates_.size()); } // Test TURN port in shared socket mode with UDP and TCP TURN server addresses. TEST_F(BasicPortAllocatorTest, TestSharedSocketWithoutNatUsingTurn) { - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TCP); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP); AddInterface(kClientAddr); allocator_.emplace(env_, &network_manager_, &socket_factory_); allocator_->Initialize(); AddTurnServers(kTurnUdpIntAddr, kTurnTcpIntAddr); - allocator_->set_step_delay(webrtc::kMinimumStepDelay); + allocator_->set_step_delay(kMinimumStepDelay); allocator_->set_flags(allocator().flags() | - webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET | - webrtc::PORTALLOCATOR_DISABLE_TCP); + PORTALLOCATOR_ENABLE_SHARED_SOCKET | + PORTALLOCATOR_DISABLE_TCP); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); ASSERT_EQ(3U, candidates_.size()); ASSERT_EQ(3U, ports_.size()); EXPECT_TRUE( @@ -1885,7 +1868,7 @@ TEST_F(BasicPortAllocatorTest, virtual_socket_server()->SetDelayOnAddress(kTurnUdpIntAddr, 200); virtual_socket_server()->SetDelayOnAddress(kTurnTcpIntAddr, 100); - TestTurnPortPrunesWithUdpAndTcpPorts(webrtc::PRUNE_BASED_ON_PRIORITY, + TestTurnPortPrunesWithUdpAndTcpPorts(PRUNE_BASED_ON_PRIORITY, true /* tcp_pruned */); } @@ -1898,7 +1881,7 @@ TEST_F(BasicPortAllocatorTest, virtual_socket_server()->SetDelayOnAddress(kTurnUdpIntAddr, 100); virtual_socket_server()->SetDelayOnAddress(kTurnTcpIntAddr, 200); - TestTurnPortPrunesWithUdpAndTcpPorts(webrtc::PRUNE_BASED_ON_PRIORITY, + TestTurnPortPrunesWithUdpAndTcpPorts(PRUNE_BASED_ON_PRIORITY, true /* tcp_pruned */); } @@ -1910,8 +1893,7 @@ TEST_F(BasicPortAllocatorTest, virtual_socket_server()->SetDelayOnAddress(kTurnUdpIntAddr, 100); virtual_socket_server()->SetDelayOnAddress(kTurnTcpIntAddr, 200); - TestTurnPortPrunesWithUdpAndTcpPorts(webrtc::KEEP_FIRST_READY, - true /* tcp_pruned */); + TestTurnPortPrunesWithUdpAndTcpPorts(KEEP_FIRST_READY, true /* tcp_pruned */); } // Test that if turn_port_prune policy is KEEP_FIRST_READY, the first ready port @@ -1922,7 +1904,7 @@ TEST_F(BasicPortAllocatorTest, virtual_socket_server()->SetDelayOnAddress(kTurnUdpIntAddr, 200); virtual_socket_server()->SetDelayOnAddress(kTurnTcpIntAddr, 100); - TestTurnPortPrunesWithUdpAndTcpPorts(webrtc::KEEP_FIRST_READY, + TestTurnPortPrunesWithUdpAndTcpPorts(KEEP_FIRST_READY, false /* tcp_pruned */); } @@ -1984,8 +1966,7 @@ TEST_F(BasicPortAllocatorTestWithRealClock, // This test relies on a real query for "localhost", so it won't work on an // IPv6-only machine. MAYBE_SKIP_IPV4; - turn_server_.AddInternalSocket(SocketAddress("127.0.0.1", 3478), - webrtc::PROTO_UDP); + turn_server_.AddInternalSocket(SocketAddress("127.0.0.1", 3478), PROTO_UDP); AddInterface(kClientAddr); allocator_.emplace(env_, &network_manager_, &socket_factory_); allocator_->Initialize(); @@ -1993,21 +1974,21 @@ TEST_F(BasicPortAllocatorTestWithRealClock, RelayCredentials credentials(kTurnUsername, kTurnPassword); turn_server.credentials = credentials; turn_server.ports.push_back( - ProtocolAddress(SocketAddress("localhost", 3478), webrtc::PROTO_UDP)); + ProtocolAddress(SocketAddress("localhost", 3478), PROTO_UDP)); allocator_->AddTurnServerForTesting(turn_server); - allocator_->set_step_delay(webrtc::kMinimumStepDelay); + allocator_->set_step_delay(kMinimumStepDelay); allocator_->set_flags(allocator().flags() | - webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET | - webrtc::PORTALLOCATOR_DISABLE_TCP); + PORTALLOCATOR_ENABLE_SHARED_SOCKET | + PORTALLOCATOR_DISABLE_TCP); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return ports_.size(); }, Eq(2U), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout)}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return ports_.size(); }, Eq(2U), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout)}), + IsRtcOk()); } // Test that when PORTALLOCATOR_ENABLE_SHARED_SOCKET is enabled only one port @@ -2020,17 +2001,17 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketWithNatUsingTurn) { AddTurnServers(kTurnUdpIntAddr, SocketAddress()); allocator_->set_flags(allocator().flags() | - webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET | - webrtc::PORTALLOCATOR_DISABLE_TCP); + PORTALLOCATOR_ENABLE_SHARED_SOCKET | + PORTALLOCATOR_DISABLE_TCP); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(3U, candidates_.size()); ASSERT_EQ(2U, ports_.size()); EXPECT_TRUE( @@ -2039,11 +2020,11 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketWithNatUsingTurn) { SocketAddress(kNatUdpAddr.ipaddr(), 0))); EXPECT_TRUE(HasCandidate(candidates_, IceCandidateType::kRelay, "udp", SocketAddress(kTurnUdpExtAddr.ipaddr(), 0))); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); // Local port will be created first and then TURN port. // TODO(deadbeef): This isn't something the BasicPortAllocator API contract // guarantees... @@ -2066,17 +2047,17 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketWithNatUsingTurnAsStun) { // webrtc issue 3537. allocator_->set_step_delay(0); allocator_->set_flags(allocator().flags() | - webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET | - webrtc::PORTALLOCATOR_DISABLE_TCP); + PORTALLOCATOR_ENABLE_SHARED_SOCKET | + PORTALLOCATOR_DISABLE_TCP); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(3U, candidates_.size()); EXPECT_TRUE( HasCandidate(candidates_, IceCandidateType::kHost, "udp", kClientAddr)); @@ -2099,23 +2080,23 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketWithNatUsingTurnAsStun) { // a UDP STUN server, as this could leak our IP address. Thus we should only // expect two ports, a UDPPort and TurnPort. TEST_F(BasicPortAllocatorTest, TestSharedSocketWithNatUsingTurnTcpOnly) { - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TCP); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP); AddInterface(kClientAddr); ResetWithStunServerAndNat(SocketAddress()); AddTurnServers(SocketAddress(), kTurnTcpIntAddr); allocator_->set_flags(allocator().flags() | - webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET | - webrtc::PORTALLOCATOR_DISABLE_TCP); + PORTALLOCATOR_ENABLE_SHARED_SOCKET | + PORTALLOCATOR_DISABLE_TCP); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(2U, candidates_.size()); ASSERT_EQ(2U, ports_.size()); EXPECT_TRUE( @@ -2137,17 +2118,16 @@ TEST_F(BasicPortAllocatorTest, TestNonSharedSocketWithNatUsingTurnAsStun) { ResetWithStunServerAndNat(SocketAddress()); AddTurnServers(kTurnUdpIntAddr, SocketAddress()); - allocator_->set_flags(allocator().flags() | - webrtc::PORTALLOCATOR_DISABLE_TCP); + allocator_->set_flags(allocator().flags() | PORTALLOCATOR_DISABLE_TCP); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(3U, candidates_.size()); ASSERT_EQ(3U, ports_.size()); EXPECT_TRUE( @@ -2180,17 +2160,17 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketWithNatUsingTurnAndStun) { AddTurnServers(kTurnUdpIntAddr, SocketAddress()); allocator_->set_flags(allocator().flags() | - webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET | - webrtc::PORTALLOCATOR_DISABLE_TCP); + PORTALLOCATOR_ENABLE_SHARED_SOCKET | + PORTALLOCATOR_DISABLE_TCP); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidates_.size(); }, Eq(3U), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return candidates_.size(); }, Eq(3U), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_TRUE( HasCandidate(candidates_, IceCandidateType::kHost, "udp", kClientAddr)); Candidate stun_candidate; @@ -2209,28 +2189,26 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketWithNatUsingTurnAndStun) { // and fail to generate STUN candidate, local UDP candidate is generated // properly. TEST_F(BasicPortAllocatorTest, TestSharedSocketNoUdpAllowed) { - allocator().set_flags(allocator().flags() | - webrtc::PORTALLOCATOR_DISABLE_RELAY | - webrtc::PORTALLOCATOR_DISABLE_TCP | - webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET); - fss_->AddRule(false, webrtc::FP_UDP, webrtc::FD_ANY, kClientAddr); + allocator().set_flags(allocator().flags() | PORTALLOCATOR_DISABLE_RELAY | + PORTALLOCATOR_DISABLE_TCP | + PORTALLOCATOR_ENABLE_SHARED_SOCKET); + fss_->AddRule(false, FP_UDP, FD_ANY, kClientAddr); AddInterface(kClientAddr); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return ports_.size(); }, Eq(1U), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return ports_.size(); }, Eq(1U), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(1U, candidates_.size()); EXPECT_TRUE( HasCandidate(candidates_, IceCandidateType::kHost, "udp", kClientAddr)); // STUN timeout is 9.5sec. We need to wait to get candidate done signal. - EXPECT_THAT( - webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kStunTimeoutMs), .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kStunTimeoutMs), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(1U, candidates_.size()); } @@ -2242,44 +2220,40 @@ TEST_F(BasicPortAllocatorTest, TestNetworkPermissionBlocked) { IPAddress()); network_manager_.set_enumeration_permission( NetworkManager::ENUMERATION_BLOCKED); - allocator().set_flags(allocator().flags() | - webrtc::PORTALLOCATOR_DISABLE_RELAY | - webrtc::PORTALLOCATOR_DISABLE_TCP | - webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET); - EXPECT_EQ(0U, allocator_->flags() & - webrtc::PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION); + allocator().set_flags(allocator().flags() | PORTALLOCATOR_DISABLE_RELAY | + PORTALLOCATOR_DISABLE_TCP | + PORTALLOCATOR_ENABLE_SHARED_SOCKET); + EXPECT_EQ(0U, + allocator_->flags() & PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); - EXPECT_EQ(0U, session_->flags() & - webrtc::PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION); + EXPECT_EQ(0U, session_->flags() & PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return ports_.size(); }, Eq(1U), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return ports_.size(); }, Eq(1U), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(1U, candidates_.size()); EXPECT_TRUE( HasCandidate(candidates_, IceCandidateType::kHost, "udp", kPrivateAddr)); - EXPECT_NE(0U, session_->flags() & - webrtc::PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION); + EXPECT_NE(0U, session_->flags() & PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION); } // This test verifies allocator can use IPv6 addresses along with IPv4. TEST_F(BasicPortAllocatorTest, TestEnableIPv6Addresses) { - allocator().set_flags(allocator().flags() | - webrtc::PORTALLOCATOR_DISABLE_RELAY | - webrtc::PORTALLOCATOR_ENABLE_IPV6 | - webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET); + allocator().set_flags(allocator().flags() | PORTALLOCATOR_DISABLE_RELAY | + PORTALLOCATOR_ENABLE_IPV6 | + PORTALLOCATOR_ENABLE_SHARED_SOCKET); AddInterface(kClientIPv6Addr); AddInterface(kClientAddr); - allocator_->set_step_delay(webrtc::kMinimumStepDelay); + allocator_->set_step_delay(kMinimumStepDelay); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(4U, ports_.size()); EXPECT_EQ(4U, candidates_.size()); EXPECT_TRUE(HasCandidate(candidates_, IceCandidateType::kHost, "udp", @@ -2294,21 +2268,21 @@ TEST_F(BasicPortAllocatorTest, TestEnableIPv6Addresses) { TEST_F(BasicPortAllocatorTest, TestStopGettingPorts) { AddInterface(kClientAddr); - allocator_->set_step_delay(webrtc::kDefaultStepDelay); + allocator_->set_step_delay(kDefaultStepDelay); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil([&] { return candidates_.size(); }, Eq(2U), - {.clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT(WaitUntil([&] { return candidates_.size(); }, Eq(2U), + {.clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(2U, ports_.size()); session_->StopGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil([&] { return candidate_allocation_done_; }, - IsTrue(), {.clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.clock = &fake_clock}), + IsRtcOk()); // After stopping getting ports, adding a new interface will not start // getting ports again. - allocator_->set_step_delay(webrtc::kMinimumStepDelay); + allocator_->set_step_delay(kMinimumStepDelay); candidates_.clear(); ports_.clear(); candidate_allocation_done_ = false; @@ -2320,34 +2294,34 @@ TEST_F(BasicPortAllocatorTest, TestStopGettingPorts) { TEST_F(BasicPortAllocatorTest, TestClearGettingPorts) { AddInterface(kClientAddr); - allocator_->set_step_delay(webrtc::kDefaultStepDelay); + allocator_->set_step_delay(kDefaultStepDelay); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil([&] { return candidates_.size(); }, Eq(2U), - {.clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT(WaitUntil([&] { return candidates_.size(); }, Eq(2U), + {.clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(2U, ports_.size()); session_->ClearGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil([&] { return candidate_allocation_done_; }, - IsTrue(), {.clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.clock = &fake_clock}), + IsRtcOk()); // After clearing getting ports, adding a new interface will start getting // ports again. - allocator_->set_step_delay(webrtc::kMinimumStepDelay); + allocator_->set_step_delay(kMinimumStepDelay); candidates_.clear(); ports_.clear(); candidate_allocation_done_ = false; network_manager_.AddInterface(kClientAddr2); - ASSERT_THAT(webrtc::WaitUntil([&] { return candidates_.size(); }, Eq(2U), - {.clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT(WaitUntil([&] { return candidates_.size(); }, Eq(2U), + {.clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(2U, ports_.size()); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); } // Test that the ports and candidates are updated with new ufrag/pwd/etc. when @@ -2356,16 +2330,15 @@ TEST_F(BasicPortAllocatorTest, TestTransportInformationUpdated) { AddInterface(kClientAddr); int pool_size = 1; allocator_->SetConfiguration(allocator_->stun_servers(), - allocator_->turn_servers(), pool_size, - webrtc::NO_PRUNE); + allocator_->turn_servers(), pool_size, NO_PRUNE); const PortAllocatorSession* peeked_session = allocator_->GetPooledSession(); ASSERT_NE(nullptr, peeked_session); EXPECT_THAT( - webrtc::WaitUntil( - [&] { return peeked_session->CandidatesAllocationDone(); }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + WaitUntil([&] { return peeked_session->CandidatesAllocationDone(); }, + IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); // Expect that when TakePooledSession is called, // UpdateTransportInformationInternal will be called and the // BasicPortAllocatorSession will update the ufrag/pwd of ports and @@ -2397,19 +2370,18 @@ TEST_F(BasicPortAllocatorTest, TestSetCandidateFilterAfterCandidatesGathered) { AddInterface(kClientAddr); int pool_size = 1; allocator_->SetConfiguration(allocator_->stun_servers(), - allocator_->turn_servers(), pool_size, - webrtc::NO_PRUNE); + allocator_->turn_servers(), pool_size, NO_PRUNE); const PortAllocatorSession* peeked_session = allocator_->GetPooledSession(); ASSERT_NE(nullptr, peeked_session); EXPECT_THAT( - webrtc::WaitUntil( - [&] { return peeked_session->CandidatesAllocationDone(); }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + WaitUntil([&] { return peeked_session->CandidatesAllocationDone(); }, + IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); size_t initial_candidates_size = peeked_session->ReadyCandidates().size(); size_t initial_ports_size = peeked_session->ReadyPorts().size(); - allocator_->SetCandidateFilter(webrtc::CF_RELAY); + allocator_->SetCandidateFilter(CF_RELAY); // Assume that when TakePooledSession is called, the candidate filter will be // applied to the pooled session. This is tested by PortAllocatorTest. session_ = @@ -2428,8 +2400,8 @@ TEST_F(BasicPortAllocatorTest, TestSetCandidateFilterAfterCandidatesGathered) { // Expect only relay candidates now that the filter is applied. EXPECT_TRUE(candidate.is_relay()); // Expect that the raddr is emptied due to the CF_RELAY filter. - EXPECT_EQ(candidate.related_address(), webrtc::EmptySocketAddressWithFamily( - candidate.address().family())); + EXPECT_EQ(candidate.related_address(), + EmptySocketAddressWithFamily(candidate.address().family())); } } @@ -2447,47 +2419,47 @@ TEST_F(BasicPortAllocatorTest, AddTurnServers(kTurnUdpIntAddr, SocketAddress()); allocator_->set_flags(allocator().flags() | - webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET | - webrtc::PORTALLOCATOR_DISABLE_TCP); + PORTALLOCATOR_ENABLE_SHARED_SOCKET | + PORTALLOCATOR_DISABLE_TCP); - allocator_->SetCandidateFilter(webrtc::CF_NONE); + allocator_->SetCandidateFilter(CF_NONE); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_TRUE(candidates_.empty()); EXPECT_TRUE(ports_.empty()); // Surface the relay candidate previously gathered but not signaled. - session_->SetCandidateFilter(webrtc::CF_RELAY); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidates_.size(); }, Eq(1u), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + session_->SetCandidateFilter(CF_RELAY); + ASSERT_THAT( + WaitUntil([&] { return candidates_.size(); }, Eq(1u), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_TRUE(candidates_.back().is_relay()); EXPECT_EQ(1u, ports_.size()); // Surface the srflx candidate previously gathered but not signaled. - session_->SetCandidateFilter(webrtc::CF_RELAY | webrtc::CF_REFLEXIVE); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidates_.size(); }, Eq(2u), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + session_->SetCandidateFilter(CF_RELAY | CF_REFLEXIVE); + ASSERT_THAT( + WaitUntil([&] { return candidates_.size(); }, Eq(2u), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_TRUE(candidates_.back().is_stun()); EXPECT_EQ(2u, ports_.size()); // Surface the srflx candidate previously gathered but not signaled. - session_->SetCandidateFilter(webrtc::CF_ALL); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidates_.size(); }, Eq(3u), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + session_->SetCandidateFilter(CF_ALL); + ASSERT_THAT( + WaitUntil([&] { return candidates_.size(); }, Eq(3u), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_TRUE(candidates_.back().is_local()); EXPECT_EQ(2u, ports_.size()); } @@ -2509,47 +2481,47 @@ TEST_F( AddTurnServers(kTurnUdpIntAddr, SocketAddress()); allocator_->set_flags(allocator().flags() | - webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET | - webrtc::PORTALLOCATOR_DISABLE_TCP); + PORTALLOCATOR_ENABLE_SHARED_SOCKET | + PORTALLOCATOR_DISABLE_TCP); - allocator_->SetCandidateFilter(webrtc::CF_NONE); + allocator_->SetCandidateFilter(CF_NONE); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_TRUE(candidates_.empty()); EXPECT_TRUE(ports_.empty()); // Surface the relay candidate previously gathered but not signaled. - session_->SetCandidateFilter(webrtc::CF_RELAY); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidates_.size(); }, Eq(1u), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + session_->SetCandidateFilter(CF_RELAY); + EXPECT_THAT( + WaitUntil([&] { return candidates_.size(); }, Eq(1u), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_TRUE(candidates_.back().is_relay()); EXPECT_EQ(1u, ports_.size()); // Surface the srflx candidate previously gathered but not signaled. - session_->SetCandidateFilter(webrtc::CF_REFLEXIVE); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidates_.size(); }, Eq(2u), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + session_->SetCandidateFilter(CF_REFLEXIVE); + EXPECT_THAT( + WaitUntil([&] { return candidates_.size(); }, Eq(2u), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_TRUE(candidates_.back().is_stun()); EXPECT_EQ(2u, ports_.size()); // Surface the host candidate previously gathered but not signaled. - session_->SetCandidateFilter(webrtc::CF_HOST); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidates_.size(); }, Eq(3u), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + session_->SetCandidateFilter(CF_HOST); + EXPECT_THAT( + WaitUntil([&] { return candidates_.size(); }, Eq(3u), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_TRUE(candidates_.back().is_local()); // We use a shared socket and webrtc::UDPPort handles the srflx candidate. EXPECT_EQ(2u, ports_.size()); @@ -2566,17 +2538,17 @@ TEST_F(BasicPortAllocatorTest, AddTurnServers(kTurnUdpIntAddr, SocketAddress()); allocator_->set_flags(allocator().flags() | - webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET | - webrtc::PORTALLOCATOR_DISABLE_TCP); + PORTALLOCATOR_ENABLE_SHARED_SOCKET | + PORTALLOCATOR_DISABLE_TCP); - allocator_->SetCandidateFilter(webrtc::CF_NONE); + allocator_->SetCandidateFilter(CF_NONE); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); auto test_invariants = [this]() { EXPECT_TRUE(candidates_.empty()); EXPECT_TRUE(ports_.empty()); @@ -2586,15 +2558,15 @@ TEST_F(BasicPortAllocatorTest, session_->StopGettingPorts(); - session_->SetCandidateFilter(webrtc::CF_RELAY); + session_->SetCandidateFilter(CF_RELAY); SIMULATED_WAIT(false, kDefaultAllocationTimeout, fake_clock); test_invariants(); - session_->SetCandidateFilter(webrtc::CF_RELAY | webrtc::CF_REFLEXIVE); + session_->SetCandidateFilter(CF_RELAY | CF_REFLEXIVE); SIMULATED_WAIT(false, kDefaultAllocationTimeout, fake_clock); test_invariants(); - session_->SetCandidateFilter(webrtc::CF_ALL); + session_->SetCandidateFilter(CF_ALL); SIMULATED_WAIT(false, kDefaultAllocationTimeout, fake_clock); test_invariants(); } @@ -2603,17 +2575,17 @@ TEST_F(BasicPortAllocatorTest, SetStunKeepaliveIntervalForPorts) { const int pool_size = 1; const int expected_stun_keepalive_interval = 123; AddInterface(kClientAddr); - allocator_->SetConfiguration( - allocator_->stun_servers(), allocator_->turn_servers(), pool_size, - webrtc::NO_PRUNE, nullptr, expected_stun_keepalive_interval); + allocator_->SetConfiguration(allocator_->stun_servers(), + allocator_->turn_servers(), pool_size, NO_PRUNE, + nullptr, expected_stun_keepalive_interval); auto* pooled_session = allocator_->GetPooledSession(); ASSERT_NE(nullptr, pooled_session); EXPECT_THAT( - webrtc::WaitUntil( - [&] { return pooled_session->CandidatesAllocationDone(); }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + WaitUntil([&] { return pooled_session->CandidatesAllocationDone(); }, + IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); CheckStunKeepaliveIntervalOfAllReadyPorts(pooled_session, expected_stun_keepalive_interval); } @@ -2622,21 +2594,21 @@ TEST_F(BasicPortAllocatorTest, ChangeStunKeepaliveIntervalForPortsAfterInitialConfig) { const int pool_size = 1; AddInterface(kClientAddr); - allocator_->SetConfiguration( - allocator_->stun_servers(), allocator_->turn_servers(), pool_size, - webrtc::NO_PRUNE, nullptr, 123 /* stun keepalive interval */); + allocator_->SetConfiguration(allocator_->stun_servers(), + allocator_->turn_servers(), pool_size, NO_PRUNE, + nullptr, 123 /* stun keepalive interval */); auto* pooled_session = allocator_->GetPooledSession(); ASSERT_NE(nullptr, pooled_session); EXPECT_THAT( - webrtc::WaitUntil( - [&] { return pooled_session->CandidatesAllocationDone(); }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + WaitUntil([&] { return pooled_session->CandidatesAllocationDone(); }, + IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); const int expected_stun_keepalive_interval = 321; - allocator_->SetConfiguration( - allocator_->stun_servers(), allocator_->turn_servers(), pool_size, - webrtc::NO_PRUNE, nullptr, expected_stun_keepalive_interval); + allocator_->SetConfiguration(allocator_->stun_servers(), + allocator_->turn_servers(), pool_size, NO_PRUNE, + nullptr, expected_stun_keepalive_interval); CheckStunKeepaliveIntervalOfAllReadyPorts(pooled_session, expected_stun_keepalive_interval); } @@ -2647,17 +2619,17 @@ TEST_F(BasicPortAllocatorTest, const int expected_stun_keepalive_interval = 123; AddInterface(kClientAddr); allocator_->set_flags(allocator().flags() | - webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET); - allocator_->SetConfiguration( - allocator_->stun_servers(), allocator_->turn_servers(), pool_size, - webrtc::NO_PRUNE, nullptr, expected_stun_keepalive_interval); + PORTALLOCATOR_ENABLE_SHARED_SOCKET); + allocator_->SetConfiguration(allocator_->stun_servers(), + allocator_->turn_servers(), pool_size, NO_PRUNE, + nullptr, expected_stun_keepalive_interval); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); CheckStunKeepaliveIntervalOfAllReadyPorts(session_.get(), expected_stun_keepalive_interval); } @@ -2668,17 +2640,17 @@ TEST_F(BasicPortAllocatorTest, const int expected_stun_keepalive_interval = 123; AddInterface(kClientAddr); allocator_->set_flags(allocator().flags() & - ~(webrtc::PORTALLOCATOR_ENABLE_SHARED_SOCKET)); - allocator_->SetConfiguration( - allocator_->stun_servers(), allocator_->turn_servers(), pool_size, - webrtc::NO_PRUNE, nullptr, expected_stun_keepalive_interval); + ~(PORTALLOCATOR_ENABLE_SHARED_SOCKET)); + allocator_->SetConfiguration(allocator_->stun_servers(), + allocator_->turn_servers(), pool_size, NO_PRUNE, + nullptr, expected_stun_keepalive_interval); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); CheckStunKeepaliveIntervalOfAllReadyPorts(session_.get(), expected_stun_keepalive_interval); } @@ -2690,7 +2662,7 @@ TEST_F(BasicPortAllocatorTest, HostCandidateAddressIsReplacedByHostname) { // Default config uses GTURN and no NAT, so replace that with the // desired setup (NAT, STUN server, TURN server, UDP/TCP). ResetWithStunServerAndNat(kStunAddr); - turn_server_.AddInternalSocket(kTurnTcpIntAddr, webrtc::PROTO_TCP); + turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP); AddTurnServers(kTurnUdpIntAddr, kTurnTcpIntAddr); AddTurnServers(kTurnUdpIntIPv6Addr, kTurnTcpIntIPv6Addr); @@ -2700,11 +2672,11 @@ TEST_F(BasicPortAllocatorTest, HostCandidateAddressIsReplacedByHostname) { AddInterface(kClientAddr); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(5u, candidates_.size()); int num_host_udp_candidates = 0; int num_host_tcp_candidates = 0; @@ -2716,7 +2688,7 @@ TEST_F(BasicPortAllocatorTest, HostCandidateAddressIsReplacedByHostname) { if (candidate.is_local()) { EXPECT_FALSE(candidate.address().hostname().empty()); EXPECT_TRUE(raddr.IsNil()); - if (candidate.protocol() == webrtc::UDP_PROTOCOL_NAME) { + if (candidate.protocol() == UDP_PROTOCOL_NAME) { ++num_host_udp_candidates; } else { ++num_host_tcp_candidates; @@ -2724,7 +2696,7 @@ TEST_F(BasicPortAllocatorTest, HostCandidateAddressIsReplacedByHostname) { } else if (candidate.is_stun()) { // For a srflx candidate, the related address should be set to 0.0.0.0 or // ::0 - EXPECT_TRUE(webrtc::IPIsAny(raddr.ipaddr())); + EXPECT_TRUE(IPIsAny(raddr.ipaddr())); EXPECT_EQ(raddr.port(), 0); ++num_srflx_candidates; } else if (candidate.is_relay()) { @@ -2770,7 +2742,7 @@ TEST_F(BasicPortAllocatorTest, TestDoNotUseTurnServerAsStunSever) { // Test that candidates from different servers get assigned a unique local // preference (the middle 16 bits of the priority) TEST_F(BasicPortAllocatorTest, AssignsUniqueLocalPreferencetoRelayCandidates) { - allocator_->SetCandidateFilter(webrtc::CF_RELAY); + allocator_->SetCandidateFilter(CF_RELAY); allocator_->AddTurnServerForTesting( CreateTurnServers(kTurnUdpIntAddr, SocketAddress())); allocator_->AddTurnServerForTesting( @@ -2781,11 +2753,11 @@ TEST_F(BasicPortAllocatorTest, AssignsUniqueLocalPreferencetoRelayCandidates) { AddInterface(kClientAddr); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - ASSERT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + ASSERT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(3u, candidates_.size()); EXPECT_GT((candidates_[0].priority() >> 8) & 0xFFFF, (candidates_[1].priority() >> 8) & 0xFFFF); @@ -2925,18 +2897,18 @@ TEST_F(BasicPortAllocatorTest, Select2DifferentIntefaces) { AddInterface(kClientIPv6Addr5, "cell1", ADAPTER_TYPE_CELLULAR_3G); // To simplify the test, only gather UDP host candidates. - allocator().set_flags( - webrtc::PORTALLOCATOR_ENABLE_IPV6 | webrtc::PORTALLOCATOR_DISABLE_TCP | - webrtc::PORTALLOCATOR_DISABLE_STUN | webrtc::PORTALLOCATOR_DISABLE_RELAY | - webrtc::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI); + allocator().set_flags(PORTALLOCATOR_ENABLE_IPV6 | PORTALLOCATOR_DISABLE_TCP | + PORTALLOCATOR_DISABLE_STUN | + PORTALLOCATOR_DISABLE_RELAY | + PORTALLOCATOR_ENABLE_IPV6_ON_WIFI); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(2U, candidates_.size()); // ethe1 and wifi1 were selected. @@ -2955,18 +2927,18 @@ TEST_F(BasicPortAllocatorTest, Select3DifferentIntefaces) { AddInterface(kClientIPv6Addr5, "cell1", ADAPTER_TYPE_CELLULAR_3G); // To simplify the test, only gather UDP host candidates. - allocator().set_flags( - webrtc::PORTALLOCATOR_ENABLE_IPV6 | webrtc::PORTALLOCATOR_DISABLE_TCP | - webrtc::PORTALLOCATOR_DISABLE_STUN | webrtc::PORTALLOCATOR_DISABLE_RELAY | - webrtc::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI); + allocator().set_flags(PORTALLOCATOR_ENABLE_IPV6 | PORTALLOCATOR_DISABLE_TCP | + PORTALLOCATOR_DISABLE_STUN | + PORTALLOCATOR_DISABLE_RELAY | + PORTALLOCATOR_ENABLE_IPV6_ON_WIFI); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(3U, candidates_.size()); // ethe1, wifi1, and cell1 were selected. @@ -2987,18 +2959,18 @@ TEST_F(BasicPortAllocatorTest, Select4DifferentIntefaces) { AddInterface(kClientIPv6Addr5, "cell1", ADAPTER_TYPE_CELLULAR_3G); // To simplify the test, only gather UDP host candidates. - allocator().set_flags( - webrtc::PORTALLOCATOR_ENABLE_IPV6 | webrtc::PORTALLOCATOR_DISABLE_TCP | - webrtc::PORTALLOCATOR_DISABLE_STUN | webrtc::PORTALLOCATOR_DISABLE_RELAY | - webrtc::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI); + allocator().set_flags(PORTALLOCATOR_ENABLE_IPV6 | PORTALLOCATOR_DISABLE_TCP | + PORTALLOCATOR_DISABLE_STUN | + PORTALLOCATOR_DISABLE_RELAY | + PORTALLOCATOR_ENABLE_IPV6_ON_WIFI); ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP)); session_->StartGettingPorts(); - EXPECT_THAT(webrtc::WaitUntil( - [&] { return candidate_allocation_done_; }, IsTrue(), - {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), - .clock = &fake_clock}), - webrtc::IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidate_allocation_done_; }, IsTrue(), + {.timeout = TimeDelta::Millis(kDefaultAllocationTimeout), + .clock = &fake_clock}), + IsRtcOk()); EXPECT_EQ(4U, candidates_.size()); // ethe1, ethe2, wifi1, and cell1 were selected. diff --git a/p2p/dtls/dtls_stun_piggyback_controller.cc b/p2p/dtls/dtls_stun_piggyback_controller.cc index 447257a52e..5da36d48ad 100644 --- a/p2p/dtls/dtls_stun_piggyback_controller.cc +++ b/p2p/dtls/dtls_stun_piggyback_controller.cc @@ -31,8 +31,7 @@ namespace webrtc { DtlsStunPiggybackController::DtlsStunPiggybackController( - absl::AnyInvocable)> - dtls_data_callback) + absl::AnyInvocable)> dtls_data_callback) : dtls_data_callback_(std::move(dtls_data_callback)) {} DtlsStunPiggybackController::~DtlsStunPiggybackController() {} diff --git a/p2p/dtls/dtls_transport.cc b/p2p/dtls/dtls_transport.cc index 4eff3b1953..ab4f5de3ae 100644 --- a/p2p/dtls/dtls_transport.cc +++ b/p2p/dtls/dtls_transport.cc @@ -59,24 +59,24 @@ namespace webrtc { template -void AbslStringify(Sink& sink, webrtc::DtlsTransportState state) { +void AbslStringify(Sink& sink, DtlsTransportState state) { switch (state) { - case webrtc::DtlsTransportState::kNew: + case DtlsTransportState::kNew: sink.Append("kNew"); break; - case webrtc::DtlsTransportState::kConnecting: + case DtlsTransportState::kConnecting: sink.Append("kConnecting"); break; - case webrtc::DtlsTransportState::kConnected: + case DtlsTransportState::kConnected: sink.Append("kConnected"); break; - case webrtc::DtlsTransportState::kClosed: + case DtlsTransportState::kClosed: sink.Append("kClosed"); break; - case webrtc::DtlsTransportState::kFailed: + case DtlsTransportState::kFailed: sink.Append("kFailed"); break; - case webrtc::DtlsTransportState::kNumValues: + case DtlsTransportState::kNumValues: sink.Append("kNumValues"); break; } @@ -107,13 +107,13 @@ static bool IsRtpPacket(ArrayView payload) { } StreamInterfaceChannel::StreamInterfaceChannel( - webrtc::IceTransportInternal* ice_transport) + IceTransportInternal* ice_transport) : ice_transport_(ice_transport), - state_(webrtc::SS_OPEN), - packets_(kMaxPendingPackets, webrtc::kMaxDtlsPacketLen) {} + state_(SS_OPEN), + packets_(kMaxPendingPackets, kMaxDtlsPacketLen) {} void StreamInterfaceChannel::SetDtlsStunPiggybackController( - webrtc::DtlsStunPiggybackController* dtls_stun_piggyback_controller) { + DtlsStunPiggybackController* dtls_stun_piggyback_controller) { dtls_stun_piggyback_controller_ = dtls_stun_piggyback_controller; } @@ -122,16 +122,16 @@ StreamResult StreamInterfaceChannel::Read(ArrayView buffer, int& /* error */) { RTC_DCHECK_RUN_ON(&callback_sequence_); - if (state_ == webrtc::SS_CLOSED) - return webrtc::SR_EOS; - if (state_ == webrtc::SS_OPENING) - return webrtc::SR_BLOCK; + if (state_ == SS_CLOSED) + return SR_EOS; + if (state_ == SS_OPENING) + return SR_BLOCK; if (!packets_.ReadFront(buffer.data(), buffer.size(), &read)) { - return webrtc::SR_BLOCK; + return SR_BLOCK; } - return webrtc::SR_SUCCESS; + return SR_SUCCESS; } StreamResult StreamInterfaceChannel::Write(ArrayView data, @@ -149,7 +149,7 @@ StreamResult StreamInterfaceChannel::Write(ArrayView data, ice_transport_->SendPacket(reinterpret_cast(data.data()), data.size(), packet_options); written = data.size(); - return webrtc::SR_SUCCESS; + return SR_SUCCESS; } bool StreamInterfaceChannel::Flush() { @@ -174,7 +174,7 @@ bool StreamInterfaceChannel::OnPacketReceived(const char* data, size_t size) { // packet currently in packets_. RTC_LOG(LS_ERROR) << "Failed to write packet to queue."; } - FireEvent(webrtc::SE_READ, 0); + FireEvent(SE_READ, 0); return ret; } @@ -186,14 +186,14 @@ StreamState StreamInterfaceChannel::GetState() const { void StreamInterfaceChannel::Close() { RTC_DCHECK_RUN_ON(&callback_sequence_); packets_.Clear(); - state_ = webrtc::SS_CLOSED; + state_ = SS_CLOSED; } DtlsTransportInternalImpl::DtlsTransportInternalImpl( - webrtc::IceTransportInternal* ice_transport, - const webrtc::CryptoOptions& crypto_options, - webrtc::RtcEventLog* event_log, - webrtc::SSLProtocolVersion max_version) + IceTransportInternal* ice_transport, + const CryptoOptions& crypto_options, + RtcEventLog* event_log, + SSLProtocolVersion max_version) : component_(ice_transport->component()), ice_transport_(ice_transport), downward_(nullptr), @@ -208,8 +208,8 @@ DtlsTransportInternalImpl::DtlsTransportInternalImpl( return; } piggybacked_dtls_callback_( - this, ReceivedIpPacket(piggybacked_dtls_packet, - webrtc::SocketAddress())); + this, + ReceivedIpPacket(piggybacked_dtls_packet, SocketAddress())); }) { RTC_DCHECK(ice_transport_); ConnectToIceTransport(); @@ -229,7 +229,7 @@ DtlsTransportInternalImpl::~DtlsTransportInternalImpl() { } } -webrtc::DtlsTransportState DtlsTransportInternalImpl::dtls_state() const { +DtlsTransportState DtlsTransportInternalImpl::dtls_state() const { return dtls_state_; } @@ -246,7 +246,7 @@ bool DtlsTransportInternalImpl::IsDtlsActive() const { } bool DtlsTransportInternalImpl::SetLocalCertificate( - const scoped_refptr& certificate) { + const scoped_refptr& certificate) { if (dtls_active_) { if (certificate == local_certificate_) { // This may happen during renegotiation. @@ -270,12 +270,12 @@ bool DtlsTransportInternalImpl::SetLocalCertificate( return true; } -scoped_refptr -DtlsTransportInternalImpl::GetLocalCertificate() const { +scoped_refptr DtlsTransportInternalImpl::GetLocalCertificate() + const { return local_certificate_; } -bool DtlsTransportInternalImpl::SetDtlsRole(webrtc::SSLRole role) { +bool DtlsTransportInternalImpl::SetDtlsRole(SSLRole role) { if (dtls_) { RTC_DCHECK(dtls_role_); if (*dtls_role_ != role) { @@ -290,7 +290,7 @@ bool DtlsTransportInternalImpl::SetDtlsRole(webrtc::SSLRole role) { return true; } -bool DtlsTransportInternalImpl::GetDtlsRole(webrtc::SSLRole* role) const { +bool DtlsTransportInternalImpl::GetDtlsRole(SSLRole* role) const { if (!dtls_role_) { return false; } @@ -299,7 +299,7 @@ bool DtlsTransportInternalImpl::GetDtlsRole(webrtc::SSLRole* role) const { } bool DtlsTransportInternalImpl::GetSslCipherSuite(int* cipher) const { - if (dtls_state() != webrtc::DtlsTransportState::kConnected) { + if (dtls_state() != DtlsTransportState::kConnected) { return false; } @@ -308,17 +308,17 @@ bool DtlsTransportInternalImpl::GetSslCipherSuite(int* cipher) const { std::optional DtlsTransportInternalImpl::GetTlsCipherSuiteName() const { - if (dtls_state() != webrtc::DtlsTransportState::kConnected) { + if (dtls_state() != DtlsTransportState::kConnected) { return std::nullopt; } return dtls_->GetTlsCipherSuiteName(); } -webrtc::RTCError DtlsTransportInternalImpl::SetRemoteParameters( +RTCError DtlsTransportInternalImpl::SetRemoteParameters( absl::string_view digest_alg, const uint8_t* digest, size_t digest_len, - std::optional role) { + std::optional role) { Buffer remote_fingerprint_value(digest, digest_len); bool is_dtls_restart = dtls_active_ && remote_fingerprint_value_ != remote_fingerprint_value; @@ -329,17 +329,17 @@ webrtc::RTCError DtlsTransportInternalImpl::SetRemoteParameters( dtls_role_ = *role; } else { if (!SetDtlsRole(*role)) { - return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER, - "Failed to set SSL role for the transport."); + return RTCError(RTCErrorType::INVALID_PARAMETER, + "Failed to set SSL role for the transport."); } } } // Apply remote fingerprint. if (!SetRemoteFingerprint(digest_alg, digest, digest_len)) { - return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER, - "Failed to apply remote fingerprint."); + return RTCError(RTCErrorType::INVALID_PARAMETER, + "Failed to apply remote fingerprint."); } - return webrtc::RTCError::OK(); + return RTCError::OK(); } bool DtlsTransportInternalImpl::SetRemoteFingerprint( @@ -387,17 +387,17 @@ bool DtlsTransportInternalImpl::SetRemoteFingerprint( // This can occur if DTLS is set up before a remote fingerprint is // received. For instance, if we set up DTLS due to receiving an early // ClientHello. - webrtc::SSLPeerCertificateDigestError err = dtls_->SetPeerCertificateDigest( + SSLPeerCertificateDigestError err = dtls_->SetPeerCertificateDigest( remote_fingerprint_algorithm_, remote_fingerprint_value_); - if (err != webrtc::SSLPeerCertificateDigestError::NONE) { + if (err != SSLPeerCertificateDigestError::NONE) { RTC_LOG(LS_ERROR) << ToString() << ": Couldn't set DTLS certificate digest."; - set_dtls_state(webrtc::DtlsTransportState::kFailed); + set_dtls_state(DtlsTransportState::kFailed); // If the error is "verification failed", don't return false, because // this means the fingerprint was formatted correctly but didn't match // the certificate from the DTLS handshake. Thus the DTLS state should go // to "failed", but SetRemoteDescription shouldn't fail. - return err == webrtc::SSLPeerCertificateDigestError::VERIFICATION_FAILED; + return err == SSLPeerCertificateDigestError::VERIFICATION_FAILED; } return true; } @@ -406,20 +406,20 @@ bool DtlsTransportInternalImpl::SetRemoteFingerprint( // create a new one, resetting our state. if (dtls_ && fingerprint_changing) { dtls_.reset(nullptr); - set_dtls_state(webrtc::DtlsTransportState::kNew); + set_dtls_state(DtlsTransportState::kNew); set_writable(false); } if (!SetupDtls()) { - set_dtls_state(webrtc::DtlsTransportState::kFailed); + set_dtls_state(DtlsTransportState::kFailed); return false; } return true; } -std::unique_ptr -DtlsTransportInternalImpl::GetRemoteSSLCertChain() const { +std::unique_ptr DtlsTransportInternalImpl::GetRemoteSSLCertChain() + const { if (!dtls_) { return nullptr; } @@ -444,7 +444,7 @@ bool DtlsTransportInternalImpl::SetupDtls() { downward_ptr->SetDtlsStunPiggybackController( &dtls_stun_piggyback_controller_); } - dtls_ = webrtc::SSLStreamAdapter::Create( + dtls_ = SSLStreamAdapter::Create( std::move(downward), [this](SSLHandshakeError error) { OnDtlsHandshakeError(error); }, ice_transport_->field_trials()); @@ -473,7 +473,7 @@ bool DtlsTransportInternalImpl::SetupDtls() { if (remote_fingerprint_value_.size() && dtls_->SetPeerCertificateDigest(remote_fingerprint_algorithm_, remote_fingerprint_value_) != - webrtc::SSLPeerCertificateDigestError::NONE) { + SSLPeerCertificateDigestError::NONE) { RTC_LOG(LS_ERROR) << ToString() << ": Couldn't set DTLS certificate digest."; return false; @@ -504,7 +504,7 @@ bool DtlsTransportInternalImpl::SetupDtls() { } bool DtlsTransportInternalImpl::GetSrtpCryptoSuite(int* cipher) const { - if (dtls_state() != webrtc::DtlsTransportState::kConnected) { + if (dtls_state() != DtlsTransportState::kConnected) { return false; } @@ -512,7 +512,7 @@ bool DtlsTransportInternalImpl::GetSrtpCryptoSuite(int* cipher) const { } bool DtlsTransportInternalImpl::GetSslVersionBytes(int* version) const { - if (dtls_state() != webrtc::DtlsTransportState::kConnected) { + if (dtls_state() != DtlsTransportState::kConnected) { return false; } @@ -520,8 +520,8 @@ bool DtlsTransportInternalImpl::GetSslVersionBytes(int* version) const { } uint16_t DtlsTransportInternalImpl::GetSslPeerSignatureAlgorithm() const { - if (dtls_state() != webrtc::DtlsTransportState::kConnected) { - return webrtc::kSslSignatureAlgorithmUnknown; // "not applicable" + if (dtls_state() != DtlsTransportState::kConnected) { + return kSslSignatureAlgorithmUnknown; // "not applicable" } return dtls_->GetPeerSignatureAlgorithm(); } @@ -538,15 +538,15 @@ int DtlsTransportInternalImpl::SendPacket( } switch (dtls_state()) { - case webrtc::DtlsTransportState::kNew: + case DtlsTransportState::kNew: // Can't send data until the connection is active. // TODO(ekr@rtfm.com): assert here if dtls_ is NULL? return -1; - case webrtc::DtlsTransportState::kConnecting: + case DtlsTransportState::kConnecting: // Can't send data until the connection is active. return -1; - case webrtc::DtlsTransportState::kConnected: - if (flags & webrtc::PF_SRTP_BYPASS) { + case DtlsTransportState::kConnected: + if (flags & PF_SRTP_BYPASS) { RTC_DCHECK(!srtp_ciphers_.empty()); if (!IsRtpPacket( MakeArrayView(reinterpret_cast(data), size))) { @@ -559,17 +559,17 @@ int DtlsTransportInternalImpl::SendPacket( int error; return (dtls_->WriteAll( MakeArrayView(reinterpret_cast(data), size), - written, error) == webrtc::SR_SUCCESS) + written, error) == SR_SUCCESS) ? static_cast(size) : -1; } - case webrtc::DtlsTransportState::kFailed: + case DtlsTransportState::kFailed: // Can't send anything when we're failed. RTC_LOG(LS_ERROR) << ToString() << ": Couldn't send packet due to " "webrtc::DtlsTransportState::kFailed."; return -1; - case webrtc::DtlsTransportState::kClosed: + case DtlsTransportState::kClosed: // Can't send anything when we're closed. RTC_LOG(LS_ERROR) << ToString() << ": Couldn't send packet due to " @@ -581,7 +581,7 @@ int DtlsTransportInternalImpl::SendPacket( } } -webrtc::IceTransportInternal* DtlsTransportInternalImpl::ice_transport() { +IceTransportInternal* DtlsTransportInternalImpl::ice_transport() { return ice_transport_; } @@ -601,18 +601,15 @@ int DtlsTransportInternalImpl::GetError() { return ice_transport_->GetError(); } -std::optional DtlsTransportInternalImpl::network_route() - const { +std::optional DtlsTransportInternalImpl::network_route() const { return ice_transport_->network_route(); } -bool DtlsTransportInternalImpl::GetOption(webrtc::Socket::Option opt, - int* value) { +bool DtlsTransportInternalImpl::GetOption(Socket::Option opt, int* value) { return ice_transport_->GetOption(opt, value); } -int DtlsTransportInternalImpl::SetOption(webrtc::Socket::Option opt, - int value) { +int DtlsTransportInternalImpl::SetOption(Socket::Option opt, int value) { return ice_transport_->SetOption(opt, value); } @@ -635,7 +632,7 @@ void DtlsTransportInternalImpl::ConnectToIceTransport() { ice_transport_->SignalNetworkRouteChanged.connect( this, &DtlsTransportInternalImpl::OnNetworkRouteChanged); ice_transport_->SetDtlsStunPiggybackCallbacks( - webrtc::DtlsStunPiggybackCallbacks( + DtlsStunPiggybackCallbacks( [&](auto stun_message_type) { std::optional data; std::optional ack; @@ -656,12 +653,12 @@ void DtlsTransportInternalImpl::ConnectToIceTransport() { SetPiggybackDtlsDataCallback([this](PacketTransportInternal* transport, const ReceivedIpPacket& packet) { RTC_DCHECK(dtls_active_); - RTC_DCHECK(webrtc::IsDtlsPacket(packet.payload())); + RTC_DCHECK(IsDtlsPacket(packet.payload())); if (!dtls_active_) { // Not doing DTLS. return; } - if (!webrtc::IsDtlsPacket(packet.payload())) { + if (!IsDtlsPacket(packet.payload())) { return; } OnReadPacket(ice_transport_, packet, /* piggybacked= */ true); @@ -679,7 +676,7 @@ void DtlsTransportInternalImpl::ConnectToIceTransport() { // - Once the DTLS handshake completes, the state is that of the // impl again void DtlsTransportInternalImpl::OnWritableState( - webrtc::PacketTransportInternal* transport) { + PacketTransportInternal* transport) { RTC_DCHECK_RUN_ON(&thread_checker_); RTC_DCHECK(transport == ice_transport_); RTC_LOG(LS_INFO) << ToString() << ": ice_transport writable state changed to " @@ -702,10 +699,10 @@ void DtlsTransportInternalImpl::OnWritableState( } switch (dtls_state()) { - case webrtc::DtlsTransportState::kNew: + case DtlsTransportState::kNew: MaybeStartDtls(); break; - case webrtc::DtlsTransportState::kConnected: + case DtlsTransportState::kConnected: // Note: SignalWritableState fired by set_writable. if (dtls_in_stun_ && dtls_ && first_ice_writable) { // Dtls1.3 has one remaining packet after it has become kConnected (?), @@ -715,7 +712,7 @@ void DtlsTransportInternalImpl::OnWritableState( } set_writable(ice_transport_->writable()); break; - case webrtc::DtlsTransportState::kConnecting: + case DtlsTransportState::kConnecting: if (dtls_in_stun_ && dtls_) { // If DTLS piggybacking is enabled, we set the timeout // on the DTLS object (which is then different from the @@ -724,42 +721,41 @@ void DtlsTransportInternalImpl::OnWritableState( PeriodicRetransmitDtlsPacketUntilDtlsConnected(); } break; - case webrtc::DtlsTransportState::kFailed: + case DtlsTransportState::kFailed: // Should not happen. Do nothing. RTC_LOG(LS_ERROR) << ToString() << ": OnWritableState() called in state " "webrtc::DtlsTransportState::kFailed."; break; - case webrtc::DtlsTransportState::kClosed: + case DtlsTransportState::kClosed: // Should not happen. Do nothing. RTC_LOG(LS_ERROR) << ToString() << ": OnWritableState() called in state " "webrtc::DtlsTransportState::kClosed."; break; - case webrtc::DtlsTransportState::kNumValues: + case DtlsTransportState::kNumValues: RTC_DCHECK_NOTREACHED(); break; } } void DtlsTransportInternalImpl::OnReceivingState( - webrtc::PacketTransportInternal* transport) { + PacketTransportInternal* transport) { RTC_DCHECK_RUN_ON(&thread_checker_); RTC_DCHECK(transport == ice_transport_); RTC_LOG(LS_VERBOSE) << ToString() << ": ice_transport " "receiving state changed to " << ice_transport_->receiving(); - if (!dtls_active_ || dtls_state() == webrtc::DtlsTransportState::kConnected) { + if (!dtls_active_ || dtls_state() == DtlsTransportState::kConnected) { // Note: SignalReceivingState fired by set_receiving. set_receiving(ice_transport_->receiving()); } } -void DtlsTransportInternalImpl::OnReadPacket( - webrtc::PacketTransportInternal* transport, - const ReceivedIpPacket& packet, - bool piggybacked) { +void DtlsTransportInternalImpl::OnReadPacket(PacketTransportInternal* transport, + const ReceivedIpPacket& packet, + bool piggybacked) { RTC_DCHECK_RUN_ON(&thread_checker_); RTC_DCHECK(transport == ice_transport_); @@ -770,7 +766,7 @@ void DtlsTransportInternalImpl::OnReadPacket( } switch (dtls_state()) { - case webrtc::DtlsTransportState::kNew: + case DtlsTransportState::kNew: if (dtls_) { RTC_LOG(LS_INFO) << ToString() << ": Packet received before DTLS started."; @@ -780,7 +776,7 @@ void DtlsTransportInternalImpl::OnReadPacket( "doing DTLS or not."; } // Cache a client hello packet received before DTLS has actually started. - if (webrtc::IsDtlsClientHelloPacket(packet.payload())) { + if (IsDtlsClientHelloPacket(packet.payload())) { RTC_LOG(LS_INFO) << ToString() << ": Caching DTLS ClientHello packet until DTLS is " "started."; @@ -791,7 +787,7 @@ void DtlsTransportInternalImpl::OnReadPacket( // the peer has chosen the client role, and proceed with the handshake. // The fingerprint will be verified when it's set. if (!dtls_ && local_certificate_) { - SetDtlsRole(webrtc::SSL_SERVER); + SetDtlsRole(SSL_SERVER); SetupDtls(); } } else { @@ -800,18 +796,18 @@ void DtlsTransportInternalImpl::OnReadPacket( } break; - case webrtc::DtlsTransportState::kConnecting: - case webrtc::DtlsTransportState::kConnected: + case DtlsTransportState::kConnecting: + case DtlsTransportState::kConnected: // We should only get DTLS or SRTP packets; STUN's already been demuxed. // Is this potentially a DTLS packet? - if (webrtc::IsDtlsPacket(packet.payload())) { + if (IsDtlsPacket(packet.payload())) { if (!HandleDtlsPacket(packet.payload())) { RTC_LOG(LS_ERROR) << ToString() << ": Failed to handle DTLS packet."; return; } } else { // Not a DTLS packet; our handshake should be complete by now. - if (dtls_state() != webrtc::DtlsTransportState::kConnected) { + if (dtls_state() != DtlsTransportState::kConnected) { RTC_LOG(LS_ERROR) << ToString() << ": Received non-DTLS packet before DTLS " "complete."; @@ -833,23 +829,23 @@ void DtlsTransportInternalImpl::OnReadPacket( packet.CopyAndSet(ReceivedIpPacket::kSrtpEncrypted)); } break; - case webrtc::DtlsTransportState::kFailed: - case webrtc::DtlsTransportState::kClosed: - case webrtc::DtlsTransportState::kNumValues: + case DtlsTransportState::kFailed: + case DtlsTransportState::kClosed: + case DtlsTransportState::kNumValues: // This shouldn't be happening. Drop the packet. break; } } void DtlsTransportInternalImpl::OnSentPacket( - webrtc::PacketTransportInternal* /* transport */, + PacketTransportInternal* /* transport */, const SentPacketInfo& sent_packet) { RTC_DCHECK_RUN_ON(&thread_checker_); SignalSentPacket(this, sent_packet); } void DtlsTransportInternalImpl::OnReadyToSend( - webrtc::PacketTransportInternal* /* transport */) { + PacketTransportInternal* /* transport */) { RTC_DCHECK_RUN_ON(&thread_checker_); if (writable()) { SignalReadyToSend(this); @@ -860,25 +856,24 @@ void DtlsTransportInternalImpl::OnDtlsEvent(int sig, int err) { RTC_DCHECK_RUN_ON(&thread_checker_); RTC_DCHECK(dtls_); - if (sig & webrtc::SE_OPEN) { + if (sig & SE_OPEN) { // This is the first time. RTC_LOG(LS_INFO) << ToString() << ": DTLS handshake complete."; // The check for OPEN shouldn't be necessary but let's make // sure we don't accidentally frob the state if it's closed. - if (dtls_->GetState() == webrtc::SS_OPEN) { + if (dtls_->GetState() == SS_OPEN) { int ssl_version_bytes; bool ret = dtls_->GetSslVersionBytes(&ssl_version_bytes); RTC_DCHECK(ret); dtls_stun_piggyback_controller_.SetDtlsHandshakeComplete( - dtls_role_ == webrtc::SSL_CLIENT, - ssl_version_bytes == webrtc::kDtls13VersionBytes); + dtls_role_ == SSL_CLIENT, ssl_version_bytes == kDtls13VersionBytes); downward_->SetDtlsStunPiggybackController(nullptr); - set_dtls_state(webrtc::DtlsTransportState::kConnected); + set_dtls_state(DtlsTransportState::kConnected); set_writable(true); } } - if (sig & webrtc::SE_READ) { - uint8_t buf[webrtc::kMaxDtlsPacketLen]; + if (sig & SE_READ) { + uint8_t buf[kMaxDtlsPacketLen]; size_t read; int read_error; StreamResult ret; @@ -886,47 +881,47 @@ void DtlsTransportInternalImpl::OnDtlsEvent(int sig, int err) { // one packet, so read all of them. do { ret = dtls_->Read(buf, read, read_error); - if (ret == webrtc::SR_SUCCESS) { + if (ret == SR_SUCCESS) { // TODO(bugs.webrtc.org/15368): It should be possible to use information // from the original packet here to populate socket address and // timestamp. NotifyPacketReceived(ReceivedIpPacket( - MakeArrayView(buf, read), webrtc::SocketAddress(), - webrtc::Timestamp::Micros(webrtc::TimeMicros()), - EcnMarking::kNotEct, ReceivedIpPacket::kDtlsDecrypted)); - } else if (ret == webrtc::SR_EOS) { + MakeArrayView(buf, read), SocketAddress(), + Timestamp::Micros(TimeMicros()), EcnMarking::kNotEct, + ReceivedIpPacket::kDtlsDecrypted)); + } else if (ret == SR_EOS) { // Remote peer shut down the association with no error. RTC_LOG(LS_INFO) << ToString() << ": DTLS transport closed by remote"; set_writable(false); - set_dtls_state(webrtc::DtlsTransportState::kClosed); + set_dtls_state(DtlsTransportState::kClosed); NotifyOnClose(); - } else if (ret == webrtc::SR_ERROR) { + } else if (ret == SR_ERROR) { // Remote peer shut down the association with an error. RTC_LOG(LS_INFO) << ToString() << ": Closed by remote with DTLS transport error, code=" << read_error; set_writable(false); - set_dtls_state(webrtc::DtlsTransportState::kFailed); + set_dtls_state(DtlsTransportState::kFailed); NotifyOnClose(); } - } while (ret == webrtc::SR_SUCCESS); + } while (ret == SR_SUCCESS); } - if (sig & webrtc::SE_CLOSE) { - RTC_DCHECK(sig == webrtc::SE_CLOSE); // SE_CLOSE should be by itself. + if (sig & SE_CLOSE) { + RTC_DCHECK(sig == SE_CLOSE); // SE_CLOSE should be by itself. set_writable(false); if (!err) { RTC_LOG(LS_INFO) << ToString() << ": DTLS transport closed"; - set_dtls_state(webrtc::DtlsTransportState::kClosed); + set_dtls_state(DtlsTransportState::kClosed); } else { RTC_LOG(LS_INFO) << ToString() << ": DTLS transport error, code=" << err; - set_dtls_state(webrtc::DtlsTransportState::kFailed); + set_dtls_state(DtlsTransportState::kFailed); } } } void DtlsTransportInternalImpl::OnNetworkRouteChanged( - std::optional network_route) { + std::optional network_route) { RTC_DCHECK_RUN_ON(&thread_checker_); SignalNetworkRouteChanged(network_route); } @@ -947,7 +942,7 @@ void DtlsTransportInternalImpl::MaybeStartDtls() { // configuration and therefore are our fault. RTC_DCHECK_NOTREACHED() << "StartSSL failed."; RTC_LOG(LS_ERROR) << ToString() << ": Couldn't start DTLS handshake"; - set_dtls_state(webrtc::DtlsTransportState::kFailed); + set_dtls_state(DtlsTransportState::kFailed); return; } RTC_LOG(LS_INFO) @@ -955,11 +950,11 @@ void DtlsTransportInternalImpl::MaybeStartDtls() { << ": DtlsTransportInternalImpl: Started DTLS handshake active=" << IsDtlsActive() << " role=" << (*dtls_role_ == SSL_SERVER ? "server" : "client"); - set_dtls_state(webrtc::DtlsTransportState::kConnecting); + set_dtls_state(DtlsTransportState::kConnecting); // Now that the handshake has started, we can process a cached ClientHello // (if one exists). if (!cached_client_hello_.empty()) { - if (*dtls_role_ == webrtc::SSL_SERVER) { + if (*dtls_role_ == SSL_SERVER) { int size = cached_client_hello_.size(); RTC_LOG(LS_INFO) << ToString() << ": Handling #" << size << " cached DTLS ClientHello packet(s)."; @@ -1012,8 +1007,7 @@ void DtlsTransportInternalImpl::set_writable(bool writable) { } if (event_log_) { - event_log_->Log( - std::make_unique(writable)); + event_log_->Log(std::make_unique(writable)); } RTC_LOG(LS_VERBOSE) << ToString() << ": set_writable to: " << writable; writable_ = writable; @@ -1023,14 +1017,12 @@ void DtlsTransportInternalImpl::set_writable(bool writable) { SignalWritableState(this); } -void DtlsTransportInternalImpl::set_dtls_state( - webrtc::DtlsTransportState state) { +void DtlsTransportInternalImpl::set_dtls_state(DtlsTransportState state) { if (dtls_state_ == state) { return; } if (event_log_) { - event_log_->Log( - std::make_unique(state)); + event_log_->Log(std::make_unique(state)); } RTC_LOG(LS_VERBOSE) << ToString() << ": set_dtls_state from:" << static_cast(dtls_state_) << " to " @@ -1039,8 +1031,7 @@ void DtlsTransportInternalImpl::set_dtls_state( SendDtlsState(this, state); } -void DtlsTransportInternalImpl::OnDtlsHandshakeError( - webrtc::SSLHandshakeError error) { +void DtlsTransportInternalImpl::OnDtlsHandshakeError(SSLHandshakeError error) { SendDtlsHandshakeError(error); } @@ -1074,8 +1065,8 @@ void DtlsTransportInternalImpl::ConfigureHandshakeTimeout() { } void DtlsTransportInternalImpl::SetPiggybackDtlsDataCallback( - absl::AnyInvocable callback) { + absl::AnyInvocable callback) { RTC_DCHECK(callback == nullptr || !piggybacked_dtls_callback_); piggybacked_dtls_callback_ = std::move(callback); } @@ -1084,17 +1075,16 @@ bool DtlsTransportInternalImpl::IsDtlsPiggybackSupportedByPeer() { RTC_DCHECK_RUN_ON(&thread_checker_); RTC_DCHECK(ice_transport_); return dtls_in_stun_ && (dtls_stun_piggyback_controller_.state() != - webrtc::DtlsStunPiggybackController::State::OFF); + DtlsStunPiggybackController::State::OFF); } bool DtlsTransportInternalImpl::WasDtlsCompletedByPiggybacking() { RTC_DCHECK_RUN_ON(&thread_checker_); RTC_DCHECK(ice_transport_); - return dtls_in_stun_ && - (dtls_stun_piggyback_controller_.state() == - webrtc::DtlsStunPiggybackController::State::COMPLETE || - dtls_stun_piggyback_controller_.state() == - webrtc::DtlsStunPiggybackController::State::PENDING); + return dtls_in_stun_ && (dtls_stun_piggyback_controller_.state() == + DtlsStunPiggybackController::State::COMPLETE || + dtls_stun_piggyback_controller_.state() == + DtlsStunPiggybackController::State::PENDING); } // TODO (jonaso, webrtc:367395350): Switch to upcoming @@ -1113,7 +1103,7 @@ void DtlsTransportInternalImpl:: } if (ice_transport_->writable() && dtls_in_stun_) { auto data_to_send = dtls_stun_piggyback_controller_.GetDataToPiggyback( - webrtc::STUN_BINDING_INDICATION); + STUN_BINDING_INDICATION); if (!data_to_send) { // No data to send, we're done. return; @@ -1128,15 +1118,15 @@ void DtlsTransportInternalImpl:: // Set pending before we post task. pending_periodic_retransmit_dtls_packet_ = true; - webrtc::Thread::Current()->PostDelayedHighPrecisionTask( - webrtc::SafeTask(safety_flag_.flag(), - [this] { - RTC_DCHECK_RUN_ON(&thread_checker_); - // Clear pending then the PostTask runs. - pending_periodic_retransmit_dtls_packet_ = false; - PeriodicRetransmitDtlsPacketUntilDtlsConnected(); - }), - webrtc::TimeDelta::Millis(delay_ms)); + Thread::Current()->PostDelayedHighPrecisionTask( + SafeTask(safety_flag_.flag(), + [this] { + RTC_DCHECK_RUN_ON(&thread_checker_); + // Clear pending then the PostTask runs. + pending_periodic_retransmit_dtls_packet_ = false; + PeriodicRetransmitDtlsPacketUntilDtlsConnected(); + }), + TimeDelta::Millis(delay_ms)); RTC_LOG(LS_INFO) << ToString() << ": Scheduled retransmit of DTLS packet, delay_ms: " << delay_ms; diff --git a/p2p/dtls/dtls_utils.cc b/p2p/dtls/dtls_utils.cc index 226369a968..9522d1d302 100644 --- a/p2p/dtls/dtls_utils.cc +++ b/p2p/dtls/dtls_utils.cc @@ -56,7 +56,7 @@ bool IsDtlsHandshakePacket(ArrayView payload) { } uint32_t ComputeDtlsPacketHash(ArrayView dtls_packet) { - return webrtc::ComputeCrc32(dtls_packet.data(), dtls_packet.size()); + return ComputeCrc32(dtls_packet.data(), dtls_packet.size()); } bool PacketStash::AddIfUnique(ArrayView packet) { @@ -66,16 +66,16 @@ bool PacketStash::AddIfUnique(ArrayView packet) { return false; } } - packets_.push_back({.hash = h, - .buffer = std::make_unique( - packet.data(), packet.size())}); + packets_.push_back( + {.hash = h, + .buffer = std::make_unique(packet.data(), packet.size())}); return true; } void PacketStash::Add(ArrayView packet) { - packets_.push_back({.hash = ComputeDtlsPacketHash(packet), - .buffer = std::make_unique( - packet.data(), packet.size())}); + packets_.push_back( + {.hash = ComputeDtlsPacketHash(packet), + .buffer = std::make_unique(packet.data(), packet.size())}); } void PacketStash::Prune(const absl::flat_hash_set& hashes) { diff --git a/p2p/test/nat_server.cc b/p2p/test/nat_server.cc index 655357240a..d1ab308ab8 100644 --- a/p2p/test/nat_server.cc +++ b/p2p/test/nat_server.cc @@ -65,7 +65,7 @@ AddrCmp::AddrCmp(NAT* nat) size_t AddrCmp::operator()(const SocketAddress& a) const { size_t h = 0; if (use_ip) - h ^= webrtc::HashIP(a.ipaddr()); + h ^= HashIP(a.ipaddr()); if (use_port) h ^= a.port() | (a.port() << 16); return h; @@ -89,7 +89,7 @@ bool AddrCmp::operator()(const SocketAddress& a1, class NATProxyServerSocket : public AsyncProxyServerSocket { public: explicit NATProxyServerSocket(Socket* socket) - : AsyncProxyServerSocket(socket, webrtc::kNATEncodedIPv6AddressSize) { + : AsyncProxyServerSocket(socket, kNATEncodedIPv6AddressSize) { BufferInput(true); } @@ -106,13 +106,13 @@ class NATProxyServerSocket : public AsyncProxyServerSocket { int family = data[1]; RTC_DCHECK(family == AF_INET || family == AF_INET6); - if ((family == AF_INET && *len < webrtc::kNATEncodedIPv4AddressSize) || - (family == AF_INET6 && *len < webrtc::kNATEncodedIPv6AddressSize)) { + if ((family == AF_INET && *len < kNATEncodedIPv4AddressSize) || + (family == AF_INET6 && *len < kNATEncodedIPv6AddressSize)) { return; } SocketAddress dest_addr; - size_t address_length = webrtc::UnpackAddressFromNAT( + size_t address_length = UnpackAddressFromNAT( MakeArrayView(reinterpret_cast(data), *len), &dest_addr); *len -= address_length; @@ -189,7 +189,7 @@ void NATServer::OnInternalUDPPacket(AsyncPacketSocket* socket, RTC_DCHECK(internal_socket_thread_.IsCurrent()); // Read the intended destination from the wire. SocketAddress dest_addr; - size_t length = webrtc::UnpackAddressFromNAT(packet.payload(), &dest_addr); + size_t length = UnpackAddressFromNAT(packet.payload(), &dest_addr); // Find the translation for these addresses (allocating one if necessary). SocketAddressPair route(packet.source_address(), dest_addr); @@ -230,10 +230,9 @@ void NATServer::OnExternalUDPPacket(AsyncPacketSocket* socket, // Forward this packet to the internal address. // First prepend the address in a quasi-STUN format. std::unique_ptr real_buf( - new char[packet.payload().size() + webrtc::kNATEncodedIPv6AddressSize]); - size_t addrlength = webrtc::PackAddressForNAT( - real_buf.get(), - packet.payload().size() + webrtc::kNATEncodedIPv6AddressSize, + new char[packet.payload().size() + kNATEncodedIPv6AddressSize]); + size_t addrlength = PackAddressForNAT( + real_buf.get(), packet.payload().size() + kNATEncodedIPv6AddressSize, packet.source_address()); // Copy the data part after the address. AsyncSocketPacketOptions options; diff --git a/p2p/test/nat_socket_factory.cc b/p2p/test/nat_socket_factory.cc index f7fbf21784..ba580c6020 100644 --- a/p2p/test/nat_socket_factory.cc +++ b/p2p/test/nat_socket_factory.cc @@ -49,8 +49,7 @@ size_t PackAddressForNAT(char* buf, buf[0] = 0; buf[1] = family; // Writes the port. - *(reinterpret_cast(&buf[2])) = - webrtc::HostToNetwork16(remote_addr.port()); + *(reinterpret_cast(&buf[2])) = HostToNetwork16(remote_addr.port()); if (family == AF_INET) { RTC_DCHECK(buf_size >= kNATEncodedIPv4AddressSize); in_addr v4addr = ip.ipv4_address(); @@ -73,8 +72,8 @@ size_t UnpackAddressFromNAT(ArrayView buf, RTC_CHECK(buf.size() >= 8); RTC_DCHECK(buf.data()[0] == 0); int family = buf[1]; - uint16_t port = webrtc::NetworkToHost16( - *(reinterpret_cast(&buf.data()[2]))); + uint16_t port = + NetworkToHost16(*(reinterpret_cast(&buf.data()[2]))); if (family == AF_INET) { const in_addr* v4addr = reinterpret_cast(&buf.data()[4]); *remote_addr = SocketAddress(IPAddress(*v4addr), port); @@ -121,7 +120,7 @@ class NATSocket : public Socket, public sigslot::has_slots<> { // If we're not already bound (meaning `socket_` is null), bind to ANY // address. if (!socket_) { - result = BindInternal(SocketAddress(webrtc::GetAnyIP(family_), 0)); + result = BindInternal(SocketAddress(GetAnyIP(family_), 0)); if (result < 0) { return result; } diff --git a/p2p/test/test_stun_server.cc b/p2p/test/test_stun_server.cc index f440d1e186..404dcf055b 100644 --- a/p2p/test/test_stun_server.cc +++ b/p2p/test/test_stun_server.cc @@ -24,7 +24,7 @@ namespace webrtc { -std::unique_ptr> +std::unique_ptr> TestStunServer::Create(SocketServer* ss, const SocketAddress& addr, Thread& network_thread) { @@ -35,8 +35,8 @@ TestStunServer::Create(SocketServer* ss, TestStunServer* server = nullptr; network_thread.BlockingCall( [&]() { server = new TestStunServer(udp_socket, network_thread); }); - std::unique_ptr> - result(server, [&](TestStunServer* server) { + std::unique_ptr> result( + server, [&](TestStunServer* server) { network_thread.BlockingCall([server]() { delete server; }); }); return result; diff --git a/p2p/test/turn_server.cc b/p2p/test/turn_server.cc index cfc38acc67..0c49e80a7d 100644 --- a/p2p/test/turn_server.cc +++ b/p2p/test/turn_server.cc @@ -48,7 +48,6 @@ namespace webrtc { namespace { -using ::webrtc::TimeDelta; // TODO(juberti): Move this all to a future turnmessage.h // static const int IPPROTO_UDP = 17; @@ -187,7 +186,7 @@ void TurnServer::OnInternalPacket(AsyncPacketSocket* socket, InternalSocketMap::iterator iter = server_sockets_.find(socket); RTC_DCHECK(iter != server_sockets_.end()); TurnServerConnection conn(packet.source_address(), iter->second, socket); - uint16_t msg_type = webrtc::GetBE16(packet.payload().data()); + uint16_t msg_type = GetBE16(packet.payload().data()); if (!IsTurnChannelData(msg_type)) { // This is a STUN message. HandleStunMessage(&conn, packet.payload()); @@ -512,7 +511,7 @@ void TurnServer::DestroyAllocation(TurnServerAllocation* allocation) { // by all allocations. // Note: We may not find a socket if it's a TCP socket that was closed, and // the allocation is only now timing out. - if (iter != server_sockets_.end() && iter->second != webrtc::PROTO_UDP) { + if (iter != server_sockets_.end() && iter->second != PROTO_UDP) { DestroyInternalSocket(socket); } @@ -703,7 +702,7 @@ void TurnServerAllocation::HandleCreatePermissionRequest( } if (server_->reject_private_addresses_ && - webrtc::IPIsPrivate(peer_attr->GetAddress().ipaddr())) { + IPIsPrivate(peer_attr->GetAddress().ipaddr())) { SendErrorResponse(msg, STUN_ERROR_FORBIDDEN, STUN_ERROR_REASON_FORBIDDEN); return; } @@ -781,7 +780,7 @@ void TurnServerAllocation::HandleChannelBindRequest(const TurnMessage* msg) { void TurnServerAllocation::HandleChannelData(ArrayView payload) { // Extract the channel number from the data. - uint16_t channel_id = webrtc::GetBE16(payload.data()); + uint16_t channel_id = GetBE16(payload.data()); auto channel = FindChannel(channel_id); if (channel != channels_.end()) { // Send the data to the peer address. diff --git a/pc/audio_rtp_receiver_unittest.cc b/pc/audio_rtp_receiver_unittest.cc index 91614d7d69..b056bdf07e 100644 --- a/pc/audio_rtp_receiver_unittest.cc +++ b/pc/audio_rtp_receiver_unittest.cc @@ -83,7 +83,7 @@ TEST_F(AudioRtpReceiverTest, SetOutputVolumeIsCalled) { receiver_->OnSetVolume(kVolume); EXPECT_THAT(WaitUntil([&] { return set_volume_calls.load(); }, Eq(2), - {.timeout = webrtc::TimeDelta::Millis(kTimeOut)}), + {.timeout = TimeDelta::Millis(kTimeOut)}), IsRtcOk()); } diff --git a/pc/channel.cc b/pc/channel.cc index deeb8b7103..1ab970f245 100644 --- a/pc/channel.cc +++ b/pc/channel.cc @@ -55,8 +55,6 @@ namespace webrtc { namespace { -using ::webrtc::UniqueRandomIdGenerator; - // Finds a stream based on target's Primary SSRC or RIDs. // This struct is used in BaseChannel::UpdateLocalStreams_w. struct StreamFinder { @@ -117,7 +115,7 @@ void RtpSendParametersFromMediaDescription( RtpHeaderExtensions extensions = RtpExtension::DeduplicateHeaderExtensions( desc->rtp_header_extensions(), extensions_filter); const bool is_stream_active = - webrtc::RtpTransceiverDirectionHasRecv(desc->direction()); + RtpTransceiverDirectionHasRecv(desc->direction()); MediaChannelParametersFromMediaDescription(desc, extensions, is_stream_active, send_params); send_params->max_bandwidth_bps = desc->bandwidth(); @@ -166,7 +164,7 @@ BaseChannel::~BaseChannel() { } std::string BaseChannel::ToString() const { - return webrtc::StringFormat( + return StringFormat( "{mid: %s, media_type: %s}", mid().c_str(), MediaTypeToString(media_send_channel_->media_type()).c_str()); } @@ -303,8 +301,8 @@ bool BaseChannel::IsReadyToSendMedia_w() const { // Send outgoing data if we are enabled, have local and remote content, // and we have had some form of connectivity. return enabled_ && - webrtc::RtpTransceiverDirectionHasRecv(remote_content_direction_) && - webrtc::RtpTransceiverDirectionHasSend(local_content_direction_) && + RtpTransceiverDirectionHasRecv(remote_content_direction_) && + RtpTransceiverDirectionHasSend(local_content_direction_) && was_ever_writable_; } @@ -501,9 +499,9 @@ bool BaseChannel::MaybeUpdateDemuxerAndRtpExtensions_w( return true; if (!rtp_transport_->RegisterRtpDemuxerSink(demuxer_criteria_, this)) { - error_desc = webrtc::StringFormat( - "Failed to apply demuxer criteria for '%s': '%s'.", mid().c_str(), - demuxer_criteria_.ToString().c_str()); + error_desc = + StringFormat("Failed to apply demuxer criteria for '%s': '%s'.", + mid().c_str(), demuxer_criteria_.ToString().c_str()); return false; } return true; @@ -659,7 +657,7 @@ bool BaseChannel::UpdateLocalStreams_w(const std::vector& streams, continue; } if (!media_send_channel()->RemoveSendStream(old_stream.first_ssrc())) { - error_desc = webrtc::StringFormat( + error_desc = StringFormat( "Failed to remove send stream with ssrc %u from m-section with " "mid='%s'.", old_stream.first_ssrc(), mid().c_str()); @@ -685,7 +683,7 @@ bool BaseChannel::UpdateLocalStreams_w(const std::vector& streams, RTC_DCHECK(new_stream.has_ssrcs() || new_stream.has_rids()); if (new_stream.has_ssrcs() && new_stream.has_rids()) { - error_desc = webrtc::StringFormat( + error_desc = StringFormat( "Failed to add send stream: %u into m-section with mid='%s'. Stream " "has both SSRCs and RIDs.", new_stream.first_ssrc(), mid().c_str()); @@ -705,7 +703,7 @@ bool BaseChannel::UpdateLocalStreams_w(const std::vector& streams, RTC_LOG(LS_INFO) << "Add send stream ssrc: " << new_stream.ssrcs[0] << " into " << ToString(); } else { - error_desc = webrtc::StringFormat( + error_desc = StringFormat( "Failed to add send stream ssrc: %u into m-section with mid='%s'", new_stream.first_ssrc(), mid().c_str()); ret = false; @@ -720,7 +718,7 @@ bool BaseChannel::UpdateRemoteStreams_w(const MediaContentDescription* content, std::string& error_desc) { RTC_LOG_THREAD_BLOCK_COUNT(); bool needs_re_registration = false; - if (!webrtc::RtpTransceiverDirectionHasSend(content->direction())) { + if (!RtpTransceiverDirectionHasSend(content->direction())) { RTC_DLOG(LS_VERBOSE) << "UpdateRemoteStreams_w: remote side will not send " "- disable payload type demuxing for " << ToString(); @@ -747,7 +745,7 @@ bool BaseChannel::UpdateRemoteStreams_w(const MediaContentDescription* content, RTC_LOG(LS_INFO) << "Remove remote ssrc: " << old_stream.first_ssrc() << " from " << ToString() << "."; } else { - error_desc = webrtc::StringFormat( + error_desc = StringFormat( "Failed to remove remote stream with ssrc %u from m-section with " "mid='%s'.", old_stream.first_ssrc(), mid().c_str()); @@ -771,12 +769,12 @@ bool BaseChannel::UpdateRemoteStreams_w(const MediaContentDescription* content, : "unsignaled") << " to " << ToString(); } else { - error_desc = webrtc::StringFormat( - "Failed to add remote stream ssrc: %s to %s", - new_stream.has_ssrcs() - ? std::to_string(new_stream.first_ssrc()).c_str() - : "unsignaled", - ToString().c_str()); + error_desc = + StringFormat("Failed to add remote stream ssrc: %s to %s", + new_stream.has_ssrcs() + ? std::to_string(new_stream.first_ssrc()).c_str() + : "unsignaled", + ToString().c_str()); return false; } } @@ -793,8 +791,8 @@ bool BaseChannel::UpdateRemoteStreams_w(const MediaContentDescription* content, // Re-register the sink to update after changing the demuxer criteria. if (needs_re_registration && !RegisterRtpDemuxerSink_w()) { - error_desc = webrtc::StringFormat( - "Failed to set up audio demuxing for mid='%s'.", mid().c_str()); + error_desc = StringFormat("Failed to set up audio demuxing for mid='%s'.", + mid().c_str()); return false; } @@ -869,8 +867,8 @@ VoiceChannel::~VoiceChannel() { void VoiceChannel::UpdateMediaSendRecvState_w() { // Render incoming data if we're the active call, and we have the local // content. We receive data on the default channel and multiplexed streams. - bool receive = enabled() && webrtc::RtpTransceiverDirectionHasRecv( - local_content_direction()); + bool receive = + enabled() && RtpTransceiverDirectionHasRecv(local_content_direction()); media_receive_channel()->SetPlayout(receive); // Send outgoing data if we're the active call, we have the remote content, @@ -898,12 +896,11 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content, AudioReceiverParameters recv_params = last_recv_params_; MediaChannelParametersFromMediaDescription( content, header_extensions, - webrtc::RtpTransceiverDirectionHasRecv(content->direction()), - &recv_params); + RtpTransceiverDirectionHasRecv(content->direction()), &recv_params); recv_params.mid = mid(); if (!media_receive_channel()->SetReceiverParameters(recv_params)) { - error_desc = webrtc::StringFormat( + error_desc = StringFormat( "Failed to set local audio description recv parameters for m-section " "with mid='%s'.", mid().c_str()); @@ -911,7 +908,7 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content, } bool criteria_modified = false; - if (webrtc::RtpTransceiverDirectionHasRecv(content->direction())) { + if (RtpTransceiverDirectionHasRecv(content->direction())) { for (const Codec& codec : content->codecs()) { if (MaybeAddHandledPayloadType(codec.id)) { criteria_modified = true; @@ -959,7 +956,7 @@ bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content, bool parameters_applied = media_send_channel()->SetSenderParameters(send_params); if (!parameters_applied) { - error_desc = webrtc::StringFormat( + error_desc = StringFormat( "Failed to set remote audio description send parameters for m-section " "with mid='%s'.", mid().c_str()); @@ -1021,8 +1018,8 @@ VideoChannel::~VideoChannel() { void VideoChannel::UpdateMediaSendRecvState_w() { // Send outgoing data if we're the active call, we have the remote content, // and we have had some form of connectivity. - bool receive = enabled() && webrtc::RtpTransceiverDirectionHasRecv( - local_content_direction()); + bool receive = + enabled() && RtpTransceiverDirectionHasRecv(local_content_direction()); media_receive_channel()->SetReceive(receive); bool send = IsReadyToSendMedia_w(); @@ -1048,8 +1045,7 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content, MediaChannelParametersFromMediaDescription( content, header_extensions, - webrtc::RtpTransceiverDirectionHasRecv(content->direction()), - &recv_params); + RtpTransceiverDirectionHasRecv(content->direction()), &recv_params); VideoSenderParameters send_params = last_send_params_; @@ -1090,7 +1086,7 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content, send_codec.packetization = std::nullopt; needs_send_params_update = true; } else if (!has_matching_packetization) { - error_desc = webrtc::StringFormat( + error_desc = StringFormat( "Failed to set local answer due to incompatible codec " "packetization for pt='%d' specified in m-section with mid='%s'.", send_codec.id, mid().c_str()); @@ -1104,7 +1100,7 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content, } if (!media_receive_channel()->SetReceiverParameters(recv_params)) { - error_desc = webrtc::StringFormat( + error_desc = StringFormat( "Failed to set local video description recv parameters for m-section " "with mid='%s'.", mid().c_str()); @@ -1112,7 +1108,7 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content, } bool criteria_modified = false; - if (webrtc::RtpTransceiverDirectionHasRecv(content->direction())) { + if (RtpTransceiverDirectionHasRecv(content->direction())) { for (const Codec& codec : content->codecs()) { if (MaybeAddHandledPayloadType(codec.id)) criteria_modified = true; @@ -1123,7 +1119,7 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content, if (needs_send_params_update) { if (!media_send_channel()->SetSenderParameters(send_params)) { - error_desc = webrtc::StringFormat( + error_desc = StringFormat( "Failed to set send parameters for m-section with mid='%s'.", mid().c_str()); return false; @@ -1204,7 +1200,7 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content, recv_codec.packetization = std::nullopt; needs_recv_params_update = true; } else if (!has_matching_packetization) { - error_desc = webrtc::StringFormat( + error_desc = StringFormat( "Failed to set remote answer due to incompatible codec " "packetization for pt='%d' specified in m-section with mid='%s'.", recv_codec.id, mid().c_str()); @@ -1218,7 +1214,7 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content, } if (!media_send_channel()->SetSenderParameters(send_params)) { - error_desc = webrtc::StringFormat( + error_desc = StringFormat( "Failed to set remote video description send parameters for m-section " "with mid='%s'.", mid().c_str()); @@ -1234,7 +1230,7 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content, if (needs_recv_params_update) { if (!media_receive_channel()->SetReceiverParameters(recv_params)) { - error_desc = webrtc::StringFormat( + error_desc = StringFormat( "Failed to set recv parameters for m-section with mid='%s'.", mid().c_str()); return false; diff --git a/pc/data_channel_controller_unittest.cc b/pc/data_channel_controller_unittest.cc index 88f8a9e6c7..71ab9b4b18 100644 --- a/pc/data_channel_controller_unittest.cc +++ b/pc/data_channel_controller_unittest.cc @@ -64,7 +64,7 @@ class MockDataChannelTransport : public DataChannelTransportInterface { SendData, (int channel_id, const SendDataParams& params, - const webrtc::CopyOnWriteBuffer& buffer), + const CopyOnWriteBuffer& buffer), (override)); MOCK_METHOD(RTCError, CloseChannel, (int channel_id), (override)); MOCK_METHOD(void, SetDataSink, (DataChannelSink * sink), (override)); diff --git a/pc/data_channel_integrationtest.cc b/pc/data_channel_integrationtest.cc index de87405b30..e9476573b1 100644 --- a/pc/data_channel_integrationtest.cc +++ b/pc/data_channel_integrationtest.cc @@ -325,10 +325,10 @@ TEST_P(DataChannelIntegrationTest, caller()->data_channel()->Close(); EXPECT_THAT(WaitUntil([&] { return caller()->data_observer()->state(); }, - Eq(webrtc::DataChannelInterface::kClosed)), + Eq(DataChannelInterface::kClosed)), IsRtcOk()); EXPECT_THAT(WaitUntil([&] { return callee()->data_observer()->state(); }, - Eq(webrtc::DataChannelInterface::kClosed)), + Eq(DataChannelInterface::kClosed)), IsRtcOk()); } @@ -359,7 +359,7 @@ TEST_P(DataChannelIntegrationTest, EndToEndCallWithSctpDataChannelFullBuffer) { std::string data(256 * 1024, 'a'); for (size_t queued_size = 0; - queued_size < webrtc::DataChannelInterface::MaxSendQueueSize(); + queued_size < DataChannelInterface::MaxSendQueueSize(); queued_size += data.size()) { caller()->data_channel()->SendAsync(DataBuffer(data), nullptr); } diff --git a/pc/data_channel_unittest.cc b/pc/data_channel_unittest.cc index 293d33f4ba..9b4802afa3 100644 --- a/pc/data_channel_unittest.cc +++ b/pc/data_channel_unittest.cc @@ -752,12 +752,12 @@ TEST(DataChannelInterfaceTest, Coverage) { #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) TEST(DataChannelInterfaceDeathTest, SendDefaultImplDchecks) { - auto channel = webrtc::make_ref_counted(); + auto channel = make_ref_counted(); RTC_EXPECT_DEATH(channel->Send(DataBuffer("Foo")), "Check failed: false"); } TEST(DataChannelInterfaceDeathTest, SendAsyncDefaultImplDchecks) { - auto channel = webrtc::make_ref_counted(); + auto channel = make_ref_counted(); RTC_EXPECT_DEATH(channel->SendAsync(DataBuffer("Foo"), nullptr), "Check failed: false"); } diff --git a/pc/jsep_transport.cc b/pc/jsep_transport.cc index 8f0a3979c4..3a9234faea 100644 --- a/pc/jsep_transport.cc +++ b/pc/jsep_transport.cc @@ -625,9 +625,9 @@ RTCError JsepTransport::NegotiateDtlsRole( break; } } else { - if ((*current_dtls_role == webrtc::SSL_CLIENT && + if ((*current_dtls_role == SSL_CLIENT && remote_connection_role == CONNECTIONROLE_ACTIVE) || - (*current_dtls_role == webrtc::SSL_SERVER && + (*current_dtls_role == SSL_SERVER && remote_connection_role == CONNECTIONROLE_PASSIVE)) { return RTCError(RTCErrorType::INVALID_PARAMETER, "Offerer must use current negotiated role for " @@ -648,8 +648,7 @@ RTCError JsepTransport::NegotiateDtlsRole( // If local is passive, local will act as server. } - *negotiated_dtls_role = - (is_remote_server ? webrtc::SSL_CLIENT : webrtc::SSL_SERVER); + *negotiated_dtls_role = (is_remote_server ? SSL_CLIENT : SSL_SERVER); return RTCError::OK(); } diff --git a/pc/jsep_transport_controller_unittest.cc b/pc/jsep_transport_controller_unittest.cc index e8d811769f..fa42946900 100644 --- a/pc/jsep_transport_controller_unittest.cc +++ b/pc/jsep_transport_controller_unittest.cc @@ -406,7 +406,7 @@ class JsepTransportControllerTest : public JsepTransportController::Observer, // signaled correctly. std::map changed_rtp_transport_by_mid_; std::map changed_dtls_transport_by_mid_; - webrtc::PayloadTypePicker payload_type_picker_; + PayloadTypePicker payload_type_picker_; // Transport controller needs to be destroyed first, because it may issue // callbacks that modify the changed_*_by_mid in the destructor. std::unique_ptr transport_controller_; @@ -563,7 +563,7 @@ TEST_F(JsepTransportControllerTest, MaybeStartGathering) { transport_controller_->MaybeStartGathering(); EXPECT_THAT(WaitUntil([&] { return kIceGatheringGathering; }, ::testing::Eq(gathering_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(1, gathering_state_signal_count_); } @@ -711,20 +711,20 @@ TEST_F(JsepTransportControllerTest, SignalConnectionStateFailed) { fake_ice->SetConnectionCount(0); EXPECT_THAT(WaitUntil([&] { return kIceConnectionFailed; }, ::testing::Eq(connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(1, connection_state_signal_count_); EXPECT_THAT( WaitUntil([&] { return PeerConnectionInterface::kIceConnectionFailed; }, ::testing::Eq(ice_connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(1, ice_connection_state_signal_count_); EXPECT_THAT( WaitUntil( [&] { return PeerConnectionInterface::PeerConnectionState::kFailed; }, ::testing::Eq(combined_connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(1, combined_connection_state_signal_count_); } @@ -756,20 +756,20 @@ TEST_F(JsepTransportControllerTest, EXPECT_THAT(WaitUntil([&] { return kIceConnectionFailed; }, ::testing::Eq(connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(1, connection_state_signal_count_); EXPECT_THAT( WaitUntil([&] { return PeerConnectionInterface::kIceConnectionFailed; }, ::testing::Eq(ice_connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(2, ice_connection_state_signal_count_); EXPECT_THAT( WaitUntil( [&] { return PeerConnectionInterface::PeerConnectionState::kFailed; }, ::testing::Eq(combined_connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(2, combined_connection_state_signal_count_); @@ -781,14 +781,14 @@ TEST_F(JsepTransportControllerTest, fake_video_dtls->SetWritable(true); EXPECT_THAT(WaitUntil([&] { return kIceConnectionConnected; }, ::testing::Eq(connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(2, connection_state_signal_count_); EXPECT_THAT( WaitUntil( [&] { return PeerConnectionInterface::kIceConnectionConnected; }, ::testing::Eq(ice_connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(3, ice_connection_state_signal_count_); EXPECT_THAT( @@ -797,7 +797,7 @@ TEST_F(JsepTransportControllerTest, return PeerConnectionInterface::PeerConnectionState::kConnected; }, ::testing::Eq(combined_connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(3, combined_connection_state_signal_count_); } @@ -827,7 +827,7 @@ TEST_F(JsepTransportControllerTest, SignalConnectionStateComplete) { EXPECT_THAT( WaitUntil([&] { return PeerConnectionInterface::kIceConnectionChecking; }, ::testing::Eq(ice_connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(1, ice_connection_state_signal_count_); EXPECT_THAT( @@ -836,7 +836,7 @@ TEST_F(JsepTransportControllerTest, SignalConnectionStateComplete) { return PeerConnectionInterface::PeerConnectionState::kConnecting; }, ::testing::Eq(combined_connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(1, combined_connection_state_signal_count_); @@ -846,20 +846,20 @@ TEST_F(JsepTransportControllerTest, SignalConnectionStateComplete) { EXPECT_THAT(WaitUntil([&] { return kIceConnectionFailed; }, ::testing::Eq(connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(1, connection_state_signal_count_); EXPECT_THAT( WaitUntil([&] { return PeerConnectionInterface::kIceConnectionFailed; }, ::testing::Eq(ice_connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(2, ice_connection_state_signal_count_); EXPECT_THAT( WaitUntil( [&] { return PeerConnectionInterface::PeerConnectionState::kFailed; }, ::testing::Eq(combined_connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(2, combined_connection_state_signal_count_); @@ -873,14 +873,14 @@ TEST_F(JsepTransportControllerTest, SignalConnectionStateComplete) { fake_video_dtls->SetWritable(true); EXPECT_THAT(WaitUntil([&] { return kIceConnectionCompleted; }, ::testing::Eq(connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(3, connection_state_signal_count_); EXPECT_THAT( WaitUntil( [&] { return PeerConnectionInterface::kIceConnectionCompleted; }, ::testing::Eq(ice_connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(3, ice_connection_state_signal_count_); EXPECT_THAT( @@ -889,7 +889,7 @@ TEST_F(JsepTransportControllerTest, SignalConnectionStateComplete) { return PeerConnectionInterface::PeerConnectionState::kConnected; }, ::testing::Eq(combined_connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(3, combined_connection_state_signal_count_); } @@ -908,7 +908,7 @@ TEST_F(JsepTransportControllerTest, SignalIceGatheringStateGathering) { // Should be in the gathering state as soon as any transport starts gathering. EXPECT_THAT(WaitUntil([&] { return kIceGatheringGathering; }, ::testing::Eq(gathering_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(1, gathering_state_signal_count_); } @@ -929,7 +929,7 @@ TEST_F(JsepTransportControllerTest, SignalIceGatheringStateComplete) { fake_audio_dtls->fake_ice_transport()->MaybeStartGathering(); EXPECT_THAT(WaitUntil([&] { return kIceGatheringGathering; }, ::testing::Eq(gathering_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(1, gathering_state_signal_count_); @@ -941,14 +941,14 @@ TEST_F(JsepTransportControllerTest, SignalIceGatheringStateComplete) { fake_video_dtls->fake_ice_transport()->MaybeStartGathering(); EXPECT_THAT(WaitUntil([&] { return kIceGatheringGathering; }, ::testing::Eq(gathering_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(1, gathering_state_signal_count_); fake_video_dtls->fake_ice_transport()->SetCandidatesGatheringComplete(); EXPECT_THAT(WaitUntil([&] { return kIceGatheringComplete; }, ::testing::Eq(gathering_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(2, gathering_state_signal_count_); } @@ -973,9 +973,9 @@ TEST_F(JsepTransportControllerTest, EXPECT_NE(fake_audio_dtls, fake_video_dtls); fake_audio_dtls->fake_ice_transport()->MaybeStartGathering(); - EXPECT_THAT(WaitUntil([&] { return webrtc::kIceGatheringGathering; }, + EXPECT_THAT(WaitUntil([&] { return kIceGatheringGathering; }, ::testing::Eq(gathering_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(1, gathering_state_signal_count_); @@ -998,7 +998,7 @@ TEST_F(JsepTransportControllerTest, EXPECT_EQ(fake_audio_dtls, fake_video_dtls); EXPECT_THAT(WaitUntil([&] { return kIceConnectionCompleted; }, ::testing::Eq(connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(PeerConnectionInterface::kIceConnectionCompleted, ice_connection_state_); @@ -1006,7 +1006,7 @@ TEST_F(JsepTransportControllerTest, combined_connection_state_); EXPECT_THAT(WaitUntil([&] { return kIceGatheringComplete; }, ::testing::Eq(gathering_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(2, gathering_state_signal_count_); } @@ -1039,7 +1039,7 @@ TEST_F(JsepTransportControllerTest, EXPECT_THAT( WaitUntil([&] { return PeerConnectionInterface::kIceConnectionChecking; }, ::testing::Eq(ice_connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(1, ice_connection_state_signal_count_); EXPECT_THAT( @@ -1048,12 +1048,12 @@ TEST_F(JsepTransportControllerTest, return PeerConnectionInterface::PeerConnectionState::kConnecting; }, ::testing::Eq(combined_connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(1, combined_connection_state_signal_count_); EXPECT_THAT(WaitUntil([&] { return kIceGatheringGathering; }, ::testing::Eq(gathering_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(1, gathering_state_signal_count_); @@ -1067,19 +1067,19 @@ TEST_F(JsepTransportControllerTest, EXPECT_THAT( WaitUntil([&] { return PeerConnectionInterface::kIceConnectionNew; }, ::testing::Eq(ice_connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(2, ice_connection_state_signal_count_); EXPECT_THAT( WaitUntil( [&] { return PeerConnectionInterface::PeerConnectionState::kNew; }, ::testing::Eq(combined_connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(2, combined_connection_state_signal_count_); EXPECT_THAT(WaitUntil([&] { return kIceGatheringNew; }, ::testing::Eq(gathering_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(2, gathering_state_signal_count_); @@ -1089,7 +1089,7 @@ TEST_F(JsepTransportControllerTest, EXPECT_THAT( WaitUntil([&] { return PeerConnectionInterface::kIceConnectionChecking; }, ::testing::Eq(ice_connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(3, ice_connection_state_signal_count_); EXPECT_THAT( @@ -1098,12 +1098,12 @@ TEST_F(JsepTransportControllerTest, return PeerConnectionInterface::PeerConnectionState::kConnecting; }, ::testing::Eq(combined_connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(3, combined_connection_state_signal_count_); EXPECT_THAT(WaitUntil([&] { return kIceGatheringGathering; }, ::testing::Eq(gathering_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(3, gathering_state_signal_count_); } @@ -1123,7 +1123,7 @@ TEST_F(JsepTransportControllerTest, SignalCandidatesGathered) { fake_audio_dtls->fake_ice_transport(), CreateCandidate(kAudioMid1, 1)); EXPECT_THAT( WaitUntil([&] { return 1; }, ::testing::Eq(candidates_signal_count_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(1u, candidates_[kAudioMid1].size()); } @@ -1140,25 +1140,25 @@ TEST_F(JsepTransportControllerTest, IceSignalingOccursOnNetworkThread) { // connecting --> connected --> completed EXPECT_THAT(WaitUntil([&] { return kIceConnectionCompleted; }, ::testing::Eq(connection_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(2, connection_state_signal_count_); // new --> gathering --> complete EXPECT_THAT(WaitUntil([&] { return kIceGatheringComplete; }, ::testing::Eq(gathering_state_), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_EQ(2, gathering_state_signal_count_); - EXPECT_THAT(WaitUntil([&] { return candidates_[kAudioMid1].size(); }, - ::testing::Eq(1u), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), - IsRtcOk()); - EXPECT_THAT(WaitUntil([&] { return candidates_[kVideoMid1].size(); }, - ::testing::Eq(1u), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), - IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidates_[kAudioMid1].size(); }, + ::testing::Eq(1u), {.timeout = TimeDelta::Millis(kTimeout)}), + IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return candidates_[kVideoMid1].size(); }, + ::testing::Eq(1u), {.timeout = TimeDelta::Millis(kTimeout)}), + IsRtcOk()); EXPECT_EQ(2, candidates_signal_count_); EXPECT_EQ(ice_signaled_on_thread_, network_thread_.get()); diff --git a/pc/jsep_transport_unittest.cc b/pc/jsep_transport_unittest.cc index f11d41625d..1b46142e29 100644 --- a/pc/jsep_transport_unittest.cc +++ b/pc/jsep_transport_unittest.cc @@ -60,8 +60,7 @@ namespace webrtc { namespace { -using webrtc::SdpType; -using ::webrtc::test::ScopedKeyValueConfig; +using test::ScopedKeyValueConfig; static const char kIceUfrag1[] = "U001"; static const char kIcePwd1[] = "TESTICEPWD00000000000001"; @@ -185,7 +184,7 @@ class JsepTransport2Test : public ::testing::Test, public sigslot::has_slots<> { Candidate c; c.set_address(SocketAddress("192.168.1.1", 8000)); c.set_component(component); - c.set_protocol(webrtc::UDP_PROTOCOL_NAME); + c.set_protocol(UDP_PROTOCOL_NAME); c.set_priority(1); return c; } @@ -261,9 +260,9 @@ TEST_P(JsepTransport2WithRtcpMux, SetDtlsParameters) { // Create certificates. scoped_refptr local_cert = - RTCCertificate::Create(SSLIdentity::Create("local", webrtc::KT_DEFAULT)); + RTCCertificate::Create(SSLIdentity::Create("local", KT_DEFAULT)); scoped_refptr remote_cert = - RTCCertificate::Create(SSLIdentity::Create("remote", webrtc::KT_DEFAULT)); + RTCCertificate::Create(SSLIdentity::Create("remote", KT_DEFAULT)); jsep_transport_->SetLocalCertificate(local_cert); // Apply offer. @@ -287,7 +286,7 @@ TEST_P(JsepTransport2WithRtcpMux, SetDtlsParameters) { // transport descriptions. auto role = jsep_transport_->GetDtlsRole(); ASSERT_TRUE(role); - EXPECT_EQ(webrtc::SSL_SERVER, + EXPECT_EQ(SSL_SERVER, role); // Because remote description was "active". auto fake_dtls = static_cast(jsep_transport_->rtp_dtls_transport()); @@ -311,9 +310,9 @@ TEST_P(JsepTransport2WithRtcpMux, SetDtlsParametersWithPassiveAnswer) { // Create certificates. scoped_refptr local_cert = - RTCCertificate::Create(SSLIdentity::Create("local", webrtc::KT_DEFAULT)); + RTCCertificate::Create(SSLIdentity::Create("local", KT_DEFAULT)); scoped_refptr remote_cert = - RTCCertificate::Create(SSLIdentity::Create("remote", webrtc::KT_DEFAULT)); + RTCCertificate::Create(SSLIdentity::Create("remote", KT_DEFAULT)); jsep_transport_->SetLocalCertificate(local_cert); // Apply offer. @@ -337,7 +336,7 @@ TEST_P(JsepTransport2WithRtcpMux, SetDtlsParametersWithPassiveAnswer) { // transport descriptions. auto role = jsep_transport_->GetDtlsRole(); ASSERT_TRUE(role); - EXPECT_EQ(webrtc::SSL_CLIENT, + EXPECT_EQ(SSL_CLIENT, role); // Because remote description was "passive". auto fake_dtls = static_cast(jsep_transport_->rtp_dtls_transport()); @@ -422,7 +421,7 @@ TEST_P(JsepTransport2WithRtcpMux, VerifyCertificateFingerprint) { EXPECT_FALSE( jsep_transport_->VerifyCertificateFingerprint(nullptr, nullptr).ok()); - KeyType key_types[] = {webrtc::KT_RSA, webrtc::KT_ECDSA}; + KeyType key_types[] = {KT_RSA, KT_ECDSA}; for (auto& key_type : key_types) { scoped_refptr certificate = @@ -465,7 +464,7 @@ TEST_P(JsepTransport2WithRtcpMux, ValidDtlsRoleNegotiation) { // Just use the same certificate for both sides; doesn't really matter in a // non end-to-end test. scoped_refptr certificate = - RTCCertificate::Create(SSLIdentity::Create("testing", webrtc::KT_ECDSA)); + RTCCertificate::Create(SSLIdentity::Create("testing", KT_ECDSA)); JsepTransportDescription local_description = MakeJsepTransportDescription( rtcp_mux_enabled, kIceUfrag1, kIcePwd1, certificate); @@ -516,7 +515,7 @@ TEST_P(JsepTransport2WithRtcpMux, ValidDtlsRoleNegotiation) { param.local_type) .ok()); } - EXPECT_EQ(webrtc::SSL_CLIENT, *jsep_transport_->GetDtlsRole()); + EXPECT_EQ(SSL_CLIENT, *jsep_transport_->GetDtlsRole()); } // Parameters which set the SSL role to SSL_SERVER. @@ -561,7 +560,7 @@ TEST_P(JsepTransport2WithRtcpMux, ValidDtlsRoleNegotiation) { param.local_type) .ok()); } - EXPECT_EQ(webrtc::SSL_SERVER, *jsep_transport_->GetDtlsRole()); + EXPECT_EQ(SSL_SERVER, *jsep_transport_->GetDtlsRole()); } } @@ -571,7 +570,7 @@ TEST_P(JsepTransport2WithRtcpMux, InvalidDtlsRoleNegotiation) { // Just use the same certificate for both sides; doesn't really matter in a // non end-to-end test. scoped_refptr certificate = - RTCCertificate::Create(SSLIdentity::Create("testing", webrtc::KT_ECDSA)); + RTCCertificate::Create(SSLIdentity::Create("testing", KT_ECDSA)); JsepTransportDescription local_description = MakeJsepTransportDescription( rtcp_mux_enabled, kIceUfrag1, kIcePwd1, certificate); @@ -698,7 +697,7 @@ TEST_F(JsepTransport2Test, ValidDtlsReofferFromAnswerer) { // Just use the same certificate for both sides; doesn't really matter in a // non end-to-end test. scoped_refptr certificate = - RTCCertificate::Create(SSLIdentity::Create("testing", webrtc::KT_ECDSA)); + RTCCertificate::Create(SSLIdentity::Create("testing", KT_ECDSA)); bool rtcp_mux_enabled = true; jsep_transport_ = CreateJsepTransport2(rtcp_mux_enabled); jsep_transport_->SetLocalCertificate(certificate); @@ -744,7 +743,7 @@ TEST_F(JsepTransport2Test, InvalidDtlsReofferFromAnswerer) { // Just use the same certificate for both sides; doesn't really matter in a // non end-to-end test. scoped_refptr certificate = - RTCCertificate::Create(SSLIdentity::Create("testing", webrtc::KT_ECDSA)); + RTCCertificate::Create(SSLIdentity::Create("testing", KT_ECDSA)); bool rtcp_mux_enabled = true; jsep_transport_ = CreateJsepTransport2(rtcp_mux_enabled); jsep_transport_->SetLocalCertificate(certificate); @@ -789,7 +788,7 @@ TEST_F(JsepTransport2Test, InvalidDtlsReofferFromAnswerer) { // since JSEP requires generating "actpass". TEST_F(JsepTransport2Test, RemoteOfferWithCurrentNegotiatedDtlsRole) { scoped_refptr certificate = - RTCCertificate::Create(SSLIdentity::Create("testing", webrtc::KT_ECDSA)); + RTCCertificate::Create(SSLIdentity::Create("testing", KT_ECDSA)); bool rtcp_mux_enabled = true; jsep_transport_ = CreateJsepTransport2(rtcp_mux_enabled); jsep_transport_->SetLocalCertificate(certificate); @@ -815,7 +814,7 @@ TEST_F(JsepTransport2Test, RemoteOfferWithCurrentNegotiatedDtlsRole) { // Sanity check that role was actually negotiated. std::optional role = jsep_transport_->GetDtlsRole(); ASSERT_TRUE(role); - EXPECT_EQ(webrtc::SSL_CLIENT, *role); + EXPECT_EQ(SSL_CLIENT, *role); // Subsequent offer with current negotiated role of "passive". remote_desc.transport_desc.connection_role = CONNECTIONROLE_PASSIVE; @@ -833,7 +832,7 @@ TEST_F(JsepTransport2Test, RemoteOfferWithCurrentNegotiatedDtlsRole) { // role is rejected. TEST_F(JsepTransport2Test, RemoteOfferThatChangesNegotiatedDtlsRole) { scoped_refptr certificate = - RTCCertificate::Create(SSLIdentity::Create("testing", webrtc::KT_ECDSA)); + RTCCertificate::Create(SSLIdentity::Create("testing", KT_ECDSA)); bool rtcp_mux_enabled = true; jsep_transport_ = CreateJsepTransport2(rtcp_mux_enabled); jsep_transport_->SetLocalCertificate(certificate); @@ -859,7 +858,7 @@ TEST_F(JsepTransport2Test, RemoteOfferThatChangesNegotiatedDtlsRole) { // Sanity check that role was actually negotiated. std::optional role = jsep_transport_->GetDtlsRole(); ASSERT_TRUE(role); - EXPECT_EQ(webrtc::SSL_CLIENT, *role); + EXPECT_EQ(SSL_CLIENT, *role); // Subsequent offer with current negotiated role of "passive". remote_desc.transport_desc.connection_role = CONNECTIONROLE_ACTIVE; @@ -876,9 +875,9 @@ TEST_F(JsepTransport2Test, RemoteOfferThatChangesNegotiatedDtlsRole) { // Test that a remote offer which changes both fingerprint and role is accepted. TEST_F(JsepTransport2Test, RemoteOfferThatChangesFingerprintAndDtlsRole) { scoped_refptr certificate = - RTCCertificate::Create(SSLIdentity::Create("testing1", webrtc::KT_ECDSA)); + RTCCertificate::Create(SSLIdentity::Create("testing1", KT_ECDSA)); scoped_refptr certificate2 = - RTCCertificate::Create(SSLIdentity::Create("testing2", webrtc::KT_ECDSA)); + RTCCertificate::Create(SSLIdentity::Create("testing2", KT_ECDSA)); bool rtcp_mux_enabled = true; jsep_transport_ = CreateJsepTransport2(rtcp_mux_enabled); jsep_transport_->SetLocalCertificate(certificate); @@ -908,7 +907,7 @@ TEST_F(JsepTransport2Test, RemoteOfferThatChangesFingerprintAndDtlsRole) { // Sanity check that role was actually negotiated. std::optional role = jsep_transport_->GetDtlsRole(); ASSERT_TRUE(role); - EXPECT_EQ(webrtc::SSL_CLIENT, *role); + EXPECT_EQ(SSL_CLIENT, *role); // Subsequent exchange with new remote fingerprint and different role. local_desc.transport_desc.connection_role = CONNECTIONROLE_PASSIVE; @@ -923,14 +922,14 @@ TEST_F(JsepTransport2Test, RemoteOfferThatChangesFingerprintAndDtlsRole) { role = jsep_transport_->GetDtlsRole(); ASSERT_TRUE(role); - EXPECT_EQ(webrtc::SSL_SERVER, *role); + EXPECT_EQ(SSL_SERVER, *role); } // Testing that a legacy client that doesn't use the setup attribute will be // interpreted as having an active role. TEST_F(JsepTransport2Test, DtlsSetupWithLegacyAsAnswerer) { scoped_refptr certificate = - RTCCertificate::Create(SSLIdentity::Create("testing", webrtc::KT_ECDSA)); + RTCCertificate::Create(SSLIdentity::Create("testing", KT_ECDSA)); bool rtcp_mux_enabled = true; jsep_transport_ = CreateJsepTransport2(rtcp_mux_enabled); jsep_transport_->SetLocalCertificate(certificate); @@ -958,7 +957,7 @@ TEST_F(JsepTransport2Test, DtlsSetupWithLegacyAsAnswerer) { ASSERT_TRUE(role); // Since legacy answer omitted setup atribute, and we offered actpass, we // should act as passive (server). - EXPECT_EQ(webrtc::SSL_SERVER, *role); + EXPECT_EQ(SSL_SERVER, *role); } // Tests that when the RTCP mux is successfully negotiated, the RTCP transport @@ -1060,11 +1059,11 @@ class JsepTransport2HeaderExtensionTest OnReadPacket2(transport, packet); }); - auto cert1 = RTCCertificate::Create( - SSLIdentity::Create("session1", webrtc::KT_DEFAULT)); + auto cert1 = + RTCCertificate::Create(SSLIdentity::Create("session1", KT_DEFAULT)); jsep_transport1_->rtp_dtls_transport()->SetLocalCertificate(cert1); - auto cert2 = RTCCertificate::Create( - SSLIdentity::Create("session1", webrtc::KT_DEFAULT)); + auto cert2 = + RTCCertificate::Create(SSLIdentity::Create("session1", KT_DEFAULT)); jsep_transport2_->rtp_dtls_transport()->SetLocalCertificate(cert2); } @@ -1122,8 +1121,8 @@ class JsepTransport2HeaderExtensionTest memcpy(rtp_packet_data, kPcmuFrameWithExtensions, rtp_len); // In order to be able to run this test function multiple times we can not // use the same sequence number twice. Increase the sequence number by one. - webrtc::SetBE16(reinterpret_cast(rtp_packet_data) + 2, - ++sequence_number_); + SetBE16(reinterpret_cast(rtp_packet_data) + 2, + ++sequence_number_); CopyOnWriteBuffer rtp_packet(rtp_packet_data, rtp_len, packet_size); int packet_count_before = received_packet_count_; @@ -1158,8 +1157,8 @@ TEST_P(JsepTransport2HeaderExtensionTest, EncryptedHeaderExtensionNegotiation) { auto fake_dtls2 = static_cast(jsep_transport2_->rtp_dtls_transport()); - fake_dtls1->SetSrtpCryptoSuite(webrtc::kSrtpAeadAes256Gcm); - fake_dtls2->SetSrtpCryptoSuite(webrtc::kSrtpAeadAes256Gcm); + fake_dtls1->SetSrtpCryptoSuite(kSrtpAeadAes256Gcm); + fake_dtls2->SetSrtpCryptoSuite(kSrtpAeadAes256Gcm); } if (scenario == Scenario::kDtlsBeforeCallerSendOffer) { diff --git a/pc/legacy_stats_collector.cc b/pc/legacy_stats_collector.cc index 2529b88c89..e77db34c9b 100644 --- a/pc/legacy_stats_collector.cc +++ b/pc/legacy_stats_collector.cc @@ -934,7 +934,7 @@ LegacyStatsCollector::SessionStats LegacyStatsCollector::ExtractSessionInfo_n( transport_names.insert(entry.second); } - std::map transport_stats_by_name = + std::map transport_stats_by_name = pc_->GetTransportStatsByNames(transport_names); for (auto& entry : transport_stats_by_name) { @@ -1067,7 +1067,7 @@ void LegacyStatsCollector::ExtractBweInfo() { auto transceivers = pc_->GetTransceiversInternal(); std::vector video_media_channels; for (const auto& transceiver : transceivers) { - if (transceiver->media_type() != webrtc::MediaType::VIDEO) { + if (transceiver->media_type() != MediaType::VIDEO) { continue; } auto* video_channel = transceiver->internal()->channel(); @@ -1200,11 +1200,11 @@ class VideoChannelStatsGatherer final : public ChannelStatsGatherer { std::unique_ptr CreateChannelStatsGatherer( ChannelInterface* channel) { RTC_DCHECK(channel); - if (channel->media_type() == webrtc::MediaType::AUDIO) { + if (channel->media_type() == MediaType::AUDIO) { return std::make_unique( channel->AsVoiceChannel()); } else { - RTC_DCHECK_EQ(channel->media_type(), webrtc::MediaType::VIDEO); + RTC_DCHECK_EQ(channel->media_type(), MediaType::VIDEO); return std::make_unique( channel->AsVideoChannel()); } diff --git a/pc/legacy_stats_collector_unittest.cc b/pc/legacy_stats_collector_unittest.cc index 1f02180733..efe90c2395 100644 --- a/pc/legacy_stats_collector_unittest.cc +++ b/pc/legacy_stats_collector_unittest.cc @@ -745,8 +745,8 @@ static scoped_refptr CreateMockSender( EXPECT_CALL(*sender, media_type()) .WillRepeatedly( Return(track->kind() == MediaStreamTrackInterface::kAudioKind - ? webrtc::MediaType::AUDIO - : webrtc::MediaType::VIDEO)); + ? MediaType::AUDIO + : MediaType::VIDEO)); EXPECT_CALL(*sender, SetMediaChannel(_)).Times(AtMost(2)); EXPECT_CALL(*sender, SetTransceiverAsStopped()).Times(AtMost(1)); EXPECT_CALL(*sender, Stop()); @@ -762,8 +762,8 @@ static scoped_refptr CreateMockReceiver( EXPECT_CALL(*receiver, media_type()) .WillRepeatedly( Return(track->kind() == MediaStreamTrackInterface::kAudioKind - ? webrtc::MediaType::AUDIO - : webrtc::MediaType::VIDEO)); + ? MediaType::AUDIO + : MediaType::VIDEO)); EXPECT_CALL(*receiver, SetMediaChannel(_)).WillRepeatedly(Return()); EXPECT_CALL(*receiver, Stop()).WillRepeatedly(Return()); return receiver; diff --git a/pc/media_session.cc b/pc/media_session.cc index cd03330175..85e475c0ad 100644 --- a/pc/media_session.cc +++ b/pc/media_session.cc @@ -134,16 +134,15 @@ bool IsComfortNoiseCodec(const Codec& codec) { RtpTransceiverDirection NegotiateRtpTransceiverDirection( RtpTransceiverDirection offer, RtpTransceiverDirection wants) { - bool offer_send = webrtc::RtpTransceiverDirectionHasSend(offer); - bool offer_recv = webrtc::RtpTransceiverDirectionHasRecv(offer); - bool wants_send = webrtc::RtpTransceiverDirectionHasSend(wants); - bool wants_recv = webrtc::RtpTransceiverDirectionHasRecv(wants); - return webrtc::RtpTransceiverDirectionFromSendRecv(offer_recv && wants_send, - offer_send && wants_recv); + bool offer_send = RtpTransceiverDirectionHasSend(offer); + bool offer_recv = RtpTransceiverDirectionHasRecv(offer); + bool wants_send = RtpTransceiverDirectionHasSend(wants); + bool wants_recv = RtpTransceiverDirectionHasRecv(wants); + return RtpTransceiverDirectionFromSendRecv(offer_recv && wants_send, + offer_send && wants_recv); } -bool IsMediaContentOfType(const ContentInfo* content, - webrtc::MediaType media_type) { +bool IsMediaContentOfType(const ContentInfo* content, MediaType media_type) { if (!content || !content->media_description()) { return false; } @@ -242,8 +241,8 @@ void AddSimulcastToMediaDescription( RTC_DCHECK_EQ(1, description->streams().size()) << "RIDs are only supported in Unified Plan semantics."; RTC_DCHECK_EQ(1, media_description_options.sender_options.size()); - RTC_DCHECK(description->type() == webrtc::MediaType::AUDIO || - description->type() == webrtc::MediaType::VIDEO); + RTC_DCHECK(description->type() == MediaType::AUDIO || + description->type() == MediaType::VIDEO); // One RID or less indicates that simulcast is not needed. if (description->streams()[0].rids().size() <= 1) { @@ -494,7 +493,7 @@ void NegotiateRtpHeaderExtensions(const RtpHeaderExtensions& local_extensions, bool dependency_descriptor_in_local = false; bool abs_capture_time_in_local = false; - for (const webrtc::RtpExtension& ours : local_extensions) { + for (const RtpExtension& ours : local_extensions) { if (ours.uri == RtpExtension::kGenericFrameDescriptorUri00) frame_descriptor_in_local = true; else if (ours.uri == RtpExtension::kDependencyDescriptorUri) @@ -546,8 +545,8 @@ bool SetCodecsInAnswer(const MediaContentDescription* offer, StreamParamsVec* current_streams, MediaContentDescription* answer, const FieldTrialsView& field_trials) { - RTC_DCHECK(offer->type() == webrtc::MediaType::AUDIO || - offer->type() == webrtc::MediaType::VIDEO); + RTC_DCHECK(offer->type() == MediaType::AUDIO || + offer->type() == MediaType::VIDEO); answer->AddCodecs(local_codecs); answer->set_protocol(offer->protocol()); if (!AddStreamParams(media_description_options.sender_options, @@ -615,7 +614,7 @@ bool CreateMediaContentAnswer( return true; } -bool IsMediaProtocolSupported(webrtc::MediaType type, +bool IsMediaProtocolSupported(MediaType type, const std::string& protocol, bool secure_transport) { // Since not all applications serialize and deserialize the media protocol, @@ -624,7 +623,7 @@ bool IsMediaProtocolSupported(webrtc::MediaType type, return true; } - if (type == webrtc::MediaType::DATA) { + if (type == MediaType::DATA) { // Check for SCTP if (secure_transport) { // Most likely scenarios first. @@ -744,23 +743,23 @@ MediaSessionDescriptionFactory::CreateOfferOrError( } RTCError error; switch (media_description_options.type) { - case webrtc::MediaType::AUDIO: - case webrtc::MediaType::VIDEO: + case MediaType::AUDIO: + case MediaType::VIDEO: error = AddRtpContentForOffer( media_description_options, session_options, current_content, current_description, - media_description_options.type == webrtc::MediaType::AUDIO + media_description_options.type == MediaType::AUDIO ? extensions_with_ids.audio : extensions_with_ids.video, ¤t_streams, offer.get(), &ice_credentials); break; - case webrtc::MediaType::DATA: + case MediaType::DATA: error = AddDataContentForOffer(media_description_options, session_options, current_content, current_description, ¤t_streams, offer.get(), &ice_credentials); break; - case webrtc::MediaType::UNSUPPORTED: + case MediaType::UNSUPPORTED: error = AddUnsupportedContentForOffer( media_description_options, session_options, current_content, current_description, offer.get(), &ice_credentials); @@ -921,21 +920,21 @@ MediaSessionDescriptionFactory::CreateAnswerOrError( UnstoppedRtpHeaderExtensionCapabilities(header_extensions_in)); RTCError error; switch (media_description_options.type) { - case webrtc::MediaType::AUDIO: - case webrtc::MediaType::VIDEO: + case MediaType::AUDIO: + case MediaType::VIDEO: error = AddRtpContentForAnswer( media_description_options, session_options, offer_content, offer, current_content, current_description, bundle_transport, header_extensions, ¤t_streams, answer.get(), &ice_credentials); break; - case webrtc::MediaType::DATA: + case MediaType::DATA: error = AddDataContentForAnswer( media_description_options, session_options, offer_content, offer, current_content, current_description, bundle_transport, ¤t_streams, answer.get(), &ice_credentials); break; - case webrtc::MediaType::UNSUPPORTED: + case MediaType::UNSUPPORTED: error = AddUnsupportedContentForAnswer( media_description_options, session_options, offer_content, offer, current_content, current_description, bundle_transport, @@ -1052,12 +1051,12 @@ MediaSessionDescriptionFactory::GetOfferedRtpHeaderExtensionsWithIds( // Add them to `used_ids` so the local ids are not reused if a new media // type is added. for (const ContentInfo* content : current_active_contents) { - if (IsMediaContentOfType(content, webrtc::MediaType::AUDIO)) { + if (IsMediaContentOfType(content, MediaType::AUDIO)) { MergeRtpHdrExts(content->media_description()->rtp_header_extensions(), enable_encrypted_rtp_header_extensions_, &offered_extensions.audio, &all_encountered_extensions, &used_ids); - } else if (IsMediaContentOfType(content, webrtc::MediaType::VIDEO)) { + } else if (IsMediaContentOfType(content, MediaType::VIDEO)) { MergeRtpHdrExts(content->media_description()->rtp_header_extensions(), enable_encrypted_rtp_header_extensions_, &offered_extensions.video, &all_encountered_extensions, @@ -1072,11 +1071,11 @@ MediaSessionDescriptionFactory::GetOfferedRtpHeaderExtensionsWithIds( RtpHeaderExtensions filtered_extensions = filtered_rtp_header_extensions(UnstoppedOrPresentRtpHeaderExtensions( entry.header_extensions, all_encountered_extensions)); - if (entry.type == webrtc::MediaType::AUDIO) + if (entry.type == MediaType::AUDIO) MergeRtpHdrExts( filtered_extensions, enable_encrypted_rtp_header_extensions_, &offered_extensions.audio, &all_encountered_extensions, &used_ids); - else if (entry.type == webrtc::MediaType::VIDEO) + else if (entry.type == MediaType::VIDEO) MergeRtpHdrExts( filtered_extensions, enable_encrypted_rtp_header_extensions_, &offered_extensions.video, &all_encountered_extensions, &used_ids); @@ -1154,8 +1153,8 @@ RTCError MediaSessionDescriptionFactory::AddRtpContentForOffer( StreamParamsVec* current_streams, SessionDescription* session_description, IceCredentialsIterator* ice_credentials) const { - RTC_DCHECK(media_description_options.type == webrtc::MediaType::AUDIO || - media_description_options.type == webrtc::MediaType::VIDEO); + RTC_DCHECK(media_description_options.type == MediaType::AUDIO || + media_description_options.type == MediaType::VIDEO); std::vector codecs_to_include; std::string mid = media_description_options.mid; @@ -1168,7 +1167,7 @@ RTCError MediaSessionDescriptionFactory::AddRtpContentForOffer( } codecs_to_include = error_or_filtered_codecs.MoveValue(); std::unique_ptr content_description; - if (media_description_options.type == webrtc::MediaType::AUDIO) { + if (media_description_options.type == MediaType::AUDIO) { content_description = std::make_unique(); } else { content_description = std::make_unique(); @@ -1222,7 +1221,7 @@ RTCError MediaSessionDescriptionFactory::AddDataContentForOffer( data->set_protocol(secure_transport ? kMediaProtocolUdpDtlsSctp : kMediaProtocolSctp); data->set_use_sctpmap(session_options.use_obsolete_sctp_sdp); - data->set_max_message_size(webrtc::kSctpSendBufferSize); + data->set_max_message_size(kSctpSendBufferSize); auto error = CreateContentOffer(media_description_options, session_options, RtpHeaderExtensions(), ssrc_generator(), @@ -1245,8 +1244,7 @@ RTCError MediaSessionDescriptionFactory::AddUnsupportedContentForOffer( const SessionDescription* current_description, SessionDescription* desc, IceCredentialsIterator* ice_credentials) const { - RTC_CHECK( - IsMediaContentOfType(current_content, webrtc::MediaType::UNSUPPORTED)); + RTC_CHECK(IsMediaContentOfType(current_content, MediaType::UNSUPPORTED)); const UnsupportedContentDescription* current_unsupported_description = current_content->media_description()->as_unsupported(); @@ -1285,12 +1283,12 @@ RTCError MediaSessionDescriptionFactory::AddRtpContentForAnswer( StreamParamsVec* current_streams, SessionDescription* answer, IceCredentialsIterator* ice_credentials) const { - RTC_DCHECK(media_description_options.type == webrtc::MediaType::AUDIO || - media_description_options.type == webrtc::MediaType::VIDEO); + RTC_DCHECK(media_description_options.type == MediaType::AUDIO || + media_description_options.type == MediaType::VIDEO); RTC_CHECK( IsMediaContentOfType(offer_content, media_description_options.type)); const RtpMediaContentDescription* offer_content_description; - if (media_description_options.type == webrtc::MediaType::AUDIO) { + if (media_description_options.type == MediaType::AUDIO) { offer_content_description = offer_content->media_description()->as_audio(); } else { offer_content_description = offer_content->media_description()->as_video(); @@ -1336,7 +1334,7 @@ RTCError MediaSessionDescriptionFactory::AddRtpContentForAnswer( bool bundle_enabled = offer_description->HasGroup(GROUP_TYPE_BUNDLE) && session_options.bundle_enabled; std::unique_ptr answer_content; - if (media_description_options.type == webrtc::MediaType::AUDIO) { + if (media_description_options.type == MediaType::AUDIO) { answer_content = std::make_unique(); } else { answer_content = std::make_unique(); @@ -1372,7 +1370,7 @@ RTCError MediaSessionDescriptionFactory::AddRtpContentForAnswer( : transport->secure(); bool rejected = media_description_options.stopped || offer_content->rejected || !has_usable_media_codecs || - !IsMediaProtocolSupported(webrtc::MediaType::AUDIO, + !IsMediaProtocolSupported(MediaType::AUDIO, answer_content->protocol(), secure); if (rejected) { RTC_LOG(LS_INFO) << "m= section '" << media_description_options.mid @@ -1413,7 +1411,7 @@ RTCError MediaSessionDescriptionFactory::AddDataContentForAnswer( bool bundle_enabled = offer_description->HasGroup(GROUP_TYPE_BUNDLE) && session_options.bundle_enabled; - RTC_CHECK(IsMediaContentOfType(offer_content, webrtc::MediaType::DATA)); + RTC_CHECK(IsMediaContentOfType(offer_content, MediaType::DATA)); std::unique_ptr data_answer; if (offer_content->media_description()->as_sctp()) { // SCTP data content @@ -1428,11 +1426,10 @@ RTCError MediaSessionDescriptionFactory::AddDataContentForAnswer( // we do not implement infinite size messages, reply with // kSctpSendBufferSize. if (offer_data_description->max_message_size() <= 0) { - data_answer->as_sctp()->set_max_message_size(webrtc::kSctpSendBufferSize); + data_answer->as_sctp()->set_max_message_size(kSctpSendBufferSize); } else { - data_answer->as_sctp()->set_max_message_size( - std::min(offer_data_description->max_message_size(), - webrtc::kSctpSendBufferSize)); + data_answer->as_sctp()->set_max_message_size(std::min( + offer_data_description->max_message_size(), kSctpSendBufferSize)); } if (!CreateMediaContentAnswer( offer_data_description, media_description_options, session_options, @@ -1454,7 +1451,7 @@ RTCError MediaSessionDescriptionFactory::AddDataContentForAnswer( bool rejected = media_description_options.stopped || offer_content->rejected || - !IsMediaProtocolSupported(webrtc::MediaType::DATA, + !IsMediaProtocolSupported(MediaType::DATA, data_answer->protocol(), secure); auto error = AddTransportAnswer(media_description_options.mid, *data_transport, answer); @@ -1487,8 +1484,7 @@ RTCError MediaSessionDescriptionFactory::AddUnsupportedContentForAnswer( RTCErrorType::INTERNAL_ERROR, "Failed to create transport answer, unsupported transport is missing"); } - RTC_CHECK( - IsMediaContentOfType(offer_content, webrtc::MediaType::UNSUPPORTED)); + RTC_CHECK(IsMediaContentOfType(offer_content, MediaType::UNSUPPORTED)); const UnsupportedContentDescription* offer_unsupported_description = offer_content->media_description()->as_unsupported(); @@ -1514,23 +1510,23 @@ bool IsMediaContent(const ContentInfo* content) { } bool IsAudioContent(const ContentInfo* content) { - return IsMediaContentOfType(content, webrtc::MediaType::AUDIO); + return IsMediaContentOfType(content, MediaType::AUDIO); } bool IsVideoContent(const ContentInfo* content) { - return IsMediaContentOfType(content, webrtc::MediaType::VIDEO); + return IsMediaContentOfType(content, MediaType::VIDEO); } bool IsDataContent(const ContentInfo* content) { - return IsMediaContentOfType(content, webrtc::MediaType::DATA); + return IsMediaContentOfType(content, MediaType::DATA); } bool IsUnsupportedContent(const ContentInfo* content) { - return IsMediaContentOfType(content, webrtc::MediaType::UNSUPPORTED); + return IsMediaContentOfType(content, MediaType::UNSUPPORTED); } const ContentInfo* GetFirstMediaContent(const ContentInfos& contents, - webrtc::MediaType media_type) { + MediaType media_type) { for (const ContentInfo& content : contents) { if (IsMediaContentOfType(&content, media_type)) { return &content; @@ -1540,19 +1536,19 @@ const ContentInfo* GetFirstMediaContent(const ContentInfos& contents, } const ContentInfo* GetFirstAudioContent(const ContentInfos& contents) { - return GetFirstMediaContent(contents, webrtc::MediaType::AUDIO); + return GetFirstMediaContent(contents, MediaType::AUDIO); } const ContentInfo* GetFirstVideoContent(const ContentInfos& contents) { - return GetFirstMediaContent(contents, webrtc::MediaType::VIDEO); + return GetFirstMediaContent(contents, MediaType::VIDEO); } const ContentInfo* GetFirstDataContent(const ContentInfos& contents) { - return GetFirstMediaContent(contents, webrtc::MediaType::DATA); + return GetFirstMediaContent(contents, MediaType::DATA); } const ContentInfo* GetFirstMediaContent(const SessionDescription* sdesc, - webrtc::MediaType media_type) { + MediaType media_type) { if (sdesc == nullptr) { return nullptr; } @@ -1561,39 +1557,39 @@ const ContentInfo* GetFirstMediaContent(const SessionDescription* sdesc, } const ContentInfo* GetFirstAudioContent(const SessionDescription* sdesc) { - return GetFirstMediaContent(sdesc, webrtc::MediaType::AUDIO); + return GetFirstMediaContent(sdesc, MediaType::AUDIO); } const ContentInfo* GetFirstVideoContent(const SessionDescription* sdesc) { - return GetFirstMediaContent(sdesc, webrtc::MediaType::VIDEO); + return GetFirstMediaContent(sdesc, MediaType::VIDEO); } const ContentInfo* GetFirstDataContent(const SessionDescription* sdesc) { - return GetFirstMediaContent(sdesc, webrtc::MediaType::DATA); + return GetFirstMediaContent(sdesc, MediaType::DATA); } const MediaContentDescription* GetFirstMediaContentDescription( const SessionDescription* sdesc, - webrtc::MediaType media_type) { + MediaType media_type) { const ContentInfo* content = GetFirstMediaContent(sdesc, media_type); return (content ? content->media_description() : nullptr); } const AudioContentDescription* GetFirstAudioContentDescription( const SessionDescription* sdesc) { - auto desc = GetFirstMediaContentDescription(sdesc, webrtc::MediaType::AUDIO); + auto desc = GetFirstMediaContentDescription(sdesc, MediaType::AUDIO); return desc ? desc->as_audio() : nullptr; } const VideoContentDescription* GetFirstVideoContentDescription( const SessionDescription* sdesc) { - auto desc = GetFirstMediaContentDescription(sdesc, webrtc::MediaType::VIDEO); + auto desc = GetFirstMediaContentDescription(sdesc, MediaType::VIDEO); return desc ? desc->as_video() : nullptr; } const SctpDataContentDescription* GetFirstSctpDataContentDescription( const SessionDescription* sdesc) { - auto desc = GetFirstMediaContentDescription(sdesc, webrtc::MediaType::DATA); + auto desc = GetFirstMediaContentDescription(sdesc, MediaType::DATA); return desc ? desc->as_sctp() : nullptr; } @@ -1602,7 +1598,7 @@ const SctpDataContentDescription* GetFirstSctpDataContentDescription( // ContentInfo* GetFirstMediaContent(ContentInfos* contents, - webrtc::MediaType media_type) { + MediaType media_type) { for (ContentInfo& content : *contents) { if (IsMediaContentOfType(&content, media_type)) { return &content; @@ -1612,19 +1608,19 @@ ContentInfo* GetFirstMediaContent(ContentInfos* contents, } ContentInfo* GetFirstAudioContent(ContentInfos* contents) { - return GetFirstMediaContent(contents, webrtc::MediaType::AUDIO); + return GetFirstMediaContent(contents, MediaType::AUDIO); } ContentInfo* GetFirstVideoContent(ContentInfos* contents) { - return GetFirstMediaContent(contents, webrtc::MediaType::VIDEO); + return GetFirstMediaContent(contents, MediaType::VIDEO); } ContentInfo* GetFirstDataContent(ContentInfos* contents) { - return GetFirstMediaContent(contents, webrtc::MediaType::DATA); + return GetFirstMediaContent(contents, MediaType::DATA); } ContentInfo* GetFirstMediaContent(SessionDescription* sdesc, - webrtc::MediaType media_type) { + MediaType media_type) { if (sdesc == nullptr) { return nullptr; } @@ -1633,39 +1629,39 @@ ContentInfo* GetFirstMediaContent(SessionDescription* sdesc, } ContentInfo* GetFirstAudioContent(SessionDescription* sdesc) { - return GetFirstMediaContent(sdesc, webrtc::MediaType::AUDIO); + return GetFirstMediaContent(sdesc, MediaType::AUDIO); } ContentInfo* GetFirstVideoContent(SessionDescription* sdesc) { - return GetFirstMediaContent(sdesc, webrtc::MediaType::VIDEO); + return GetFirstMediaContent(sdesc, MediaType::VIDEO); } ContentInfo* GetFirstDataContent(SessionDescription* sdesc) { - return GetFirstMediaContent(sdesc, webrtc::MediaType::DATA); + return GetFirstMediaContent(sdesc, MediaType::DATA); } MediaContentDescription* GetFirstMediaContentDescription( SessionDescription* sdesc, - webrtc::MediaType media_type) { + MediaType media_type) { ContentInfo* content = GetFirstMediaContent(sdesc, media_type); return (content ? content->media_description() : nullptr); } AudioContentDescription* GetFirstAudioContentDescription( SessionDescription* sdesc) { - auto desc = GetFirstMediaContentDescription(sdesc, webrtc::MediaType::AUDIO); + auto desc = GetFirstMediaContentDescription(sdesc, MediaType::AUDIO); return desc ? desc->as_audio() : nullptr; } VideoContentDescription* GetFirstVideoContentDescription( SessionDescription* sdesc) { - auto desc = GetFirstMediaContentDescription(sdesc, webrtc::MediaType::VIDEO); + auto desc = GetFirstMediaContentDescription(sdesc, MediaType::VIDEO); return desc ? desc->as_video() : nullptr; } SctpDataContentDescription* GetFirstSctpDataContentDescription( SessionDescription* sdesc) { - auto desc = GetFirstMediaContentDescription(sdesc, webrtc::MediaType::DATA); + auto desc = GetFirstMediaContentDescription(sdesc, MediaType::DATA); return desc ? desc->as_sctp() : nullptr; } diff --git a/pc/peer_connection_bundle_unittest.cc b/pc/peer_connection_bundle_unittest.cc index e0f5643e17..56df0f92c6 100644 --- a/pc/peer_connection_bundle_unittest.cc +++ b/pc/peer_connection_bundle_unittest.cc @@ -103,8 +103,7 @@ class PeerConnectionWrapperForBundleTest : public PeerConnectionWrapper { public: using PeerConnectionWrapper::PeerConnectionWrapper; - bool AddIceCandidateToMedia(Candidate* candidate, - webrtc::MediaType media_type) { + bool AddIceCandidateToMedia(Candidate* candidate, MediaType media_type) { auto* desc = pc()->remote_description()->description(); for (size_t i = 0; i < desc->contents().size(); i++) { const auto& content = desc->contents()[i]; @@ -126,7 +125,7 @@ class PeerConnectionWrapperForBundleTest : public PeerConnectionWrapper { VoiceChannel* voice_channel() { auto transceivers = GetInternalPeerConnection()->GetTransceiversInternal(); for (const auto& transceiver : transceivers) { - if (transceiver->media_type() == webrtc::MediaType::AUDIO) { + if (transceiver->media_type() == MediaType::AUDIO) { return static_cast(transceiver->internal()->channel()); } } @@ -140,7 +139,7 @@ class PeerConnectionWrapperForBundleTest : public PeerConnectionWrapper { VideoChannel* video_channel() { auto transceivers = GetInternalPeerConnection()->GetTransceiversInternal(); for (const auto& transceiver : transceivers) { - if (transceiver->media_type() == webrtc::MediaType::VIDEO) { + if (transceiver->media_type() == MediaType::VIDEO) { return static_cast(transceiver->internal()->channel()); } } @@ -658,16 +657,16 @@ TEST_P(PeerConnectionBundleTest, // that messages are executed in the order they were posted. Candidate audio_candidate1 = CreateLocalUdpCandidate(kAudioAddress1); - ASSERT_TRUE(caller->AddIceCandidateToMedia(&audio_candidate1, - webrtc::MediaType::AUDIO)); + ASSERT_TRUE( + caller->AddIceCandidateToMedia(&audio_candidate1, MediaType::AUDIO)); Candidate video_candidate = CreateLocalUdpCandidate(kVideoAddress); - ASSERT_TRUE(caller->AddIceCandidateToMedia(&video_candidate, - webrtc::MediaType::VIDEO)); + ASSERT_TRUE( + caller->AddIceCandidateToMedia(&video_candidate, MediaType::VIDEO)); Candidate audio_candidate2 = CreateLocalUdpCandidate(kAudioAddress2); - ASSERT_TRUE(caller->AddIceCandidateToMedia(&audio_candidate2, - webrtc::MediaType::AUDIO)); + ASSERT_TRUE( + caller->AddIceCandidateToMedia(&audio_candidate2, MediaType::AUDIO)); EXPECT_THAT( WaitUntil( diff --git a/pc/peer_connection_crypto_unittest.cc b/pc/peer_connection_crypto_unittest.cc index fd0f3f6526..1a4eab3ee2 100644 --- a/pc/peer_connection_crypto_unittest.cc +++ b/pc/peer_connection_crypto_unittest.cc @@ -379,7 +379,7 @@ TEST_P(PeerConnectionCryptoDtlsCertGenTest, TestCertificateGeneration) { fake_certificate_generator->generated_failures(); }, ::testing::Gt(0), - {.timeout = webrtc::TimeDelta::Millis(kGenerateCertTimeout)}), + {.timeout = TimeDelta::Millis(kGenerateCertTimeout)}), IsRtcOk()); } else { ASSERT_EQ(fake_certificate_generator->generated_certificates(), 0); diff --git a/pc/peer_connection_data_channel_unittest.cc b/pc/peer_connection_data_channel_unittest.cc index 5cf750e456..707c34b3ef 100644 --- a/pc/peer_connection_data_channel_unittest.cc +++ b/pc/peer_connection_data_channel_unittest.cc @@ -208,11 +208,9 @@ TEST_P(PeerConnectionDataChannelTest, SctpContentAndTransportNameSetCorrectly) { auto offer = caller->CreateOffer(); const auto& offer_contents = offer->description()->contents(); - ASSERT_EQ(webrtc::MediaType::AUDIO, - offer_contents[0].media_description()->type()); + ASSERT_EQ(MediaType::AUDIO, offer_contents[0].media_description()->type()); auto audio_mid = offer_contents[0].mid(); - ASSERT_EQ(webrtc::MediaType::DATA, - offer_contents[2].media_description()->type()); + ASSERT_EQ(MediaType::DATA, offer_contents[2].media_description()->type()); auto data_mid = offer_contents[2].mid(); ASSERT_TRUE( diff --git a/pc/peer_connection_encodings_integrationtest.cc b/pc/peer_connection_encodings_integrationtest.cc index 2834f4c598..86dbb44936 100644 --- a/pc/peer_connection_encodings_integrationtest.cc +++ b/pc/peer_connection_encodings_integrationtest.cc @@ -260,7 +260,7 @@ class PeerConnectionEncodingsIntegrationTest : public ::testing::Test { absl::string_view codec_name) { std::vector codecs = pc_wrapper->pc_factory() - ->GetRtpReceiverCapabilities(webrtc::MediaType::VIDEO) + ->GetRtpReceiverCapabilities(MediaType::VIDEO) .codecs; return std::find_if(codecs.begin(), codecs.end(), [&codec_name](const RtpCodecCapability& codec) { @@ -273,7 +273,7 @@ class PeerConnectionEncodingsIntegrationTest : public ::testing::Test { absl::string_view codec_name) { std::vector codecs = pc_wrapper->pc_factory() - ->GetRtpReceiverCapabilities(webrtc::MediaType::VIDEO) + ->GetRtpReceiverCapabilities(MediaType::VIDEO) .codecs; codecs.erase(std::remove_if(codecs.begin(), codecs.end(), [&codec_name](const RtpCodecCapability& codec) { @@ -581,7 +581,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, // codec preferences because we want the sender to think SVC is a possibility. std::vector codecs = local_pc_wrapper->pc_factory() - ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO) + ->GetRtpSenderCapabilities(MediaType::VIDEO) .codecs; EXPECT_THAT(codecs[0].name, StrCaseEq("VP8")); // Attempt SVC (L3T3_KEY), which is not possible with VP8, but the sender does @@ -1355,7 +1355,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, scoped_refptr local_pc_wrapper = CreatePc(); auto transceiver_or_error = - local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::AUDIO); + local_pc_wrapper->pc()->AddTransceiver(MediaType::AUDIO); scoped_refptr audio_transceiver = transceiver_or_error.MoveValue(); RtpParameters parameters = audio_transceiver->sender()->GetParameters(); @@ -1367,7 +1367,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, scoped_refptr local_pc_wrapper = CreatePc(); auto transceiver_or_error = - local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO); + local_pc_wrapper->pc()->AddTransceiver(MediaType::VIDEO); scoped_refptr video_transceiver = transceiver_or_error.MoveValue(); RtpParameters parameters = video_transceiver->sender()->GetParameters(); @@ -1385,8 +1385,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, scoped_refptr track = stream->GetAudioTracks()[0]; std::optional pcmu = - local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::AUDIO, - "pcmu"); + local_pc_wrapper->FindFirstSendCodecWithName(MediaType::AUDIO, "pcmu"); ASSERT_TRUE(pcmu); RtpTransceiverInit init; @@ -1425,8 +1424,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, scoped_refptr track = stream->GetVideoTracks()[0]; std::optional vp9 = - local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::VIDEO, - "vp9"); + local_pc_wrapper->FindFirstSendCodecWithName(MediaType::VIDEO, "vp9"); ASSERT_TRUE(vp9); RtpTransceiverInit init; @@ -1471,8 +1469,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, scoped_refptr track = stream->GetAudioTracks()[0]; std::optional pcmu = - local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::AUDIO, - "pcmu"); + local_pc_wrapper->FindFirstSendCodecWithName(MediaType::AUDIO, "pcmu"); auto transceiver_or_error = local_pc_wrapper->pc()->AddTransceiver(track); scoped_refptr audio_transceiver = @@ -1507,8 +1504,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, scoped_refptr track = stream->GetAudioTracks()[0]; std::optional pcmu = - local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::AUDIO, - "pcmu"); + local_pc_wrapper->FindFirstSendCodecWithName(MediaType::AUDIO, "pcmu"); auto transceiver_or_error = local_pc_wrapper->pc()->AddTransceiver(track); scoped_refptr audio_transceiver = @@ -1555,8 +1551,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, scoped_refptr track = stream->GetVideoTracks()[0]; std::optional vp9 = - local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::VIDEO, - "vp9"); + local_pc_wrapper->FindFirstSendCodecWithName(MediaType::VIDEO, "vp9"); auto transceiver_or_error = local_pc_wrapper->pc()->AddTransceiver(track); scoped_refptr video_transceiver = @@ -1598,8 +1593,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, scoped_refptr track = stream->GetVideoTracks()[0]; std::optional vp9 = - local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::VIDEO, - "vp9"); + local_pc_wrapper->FindFirstSendCodecWithName(MediaType::VIDEO, "vp9"); auto transceiver_or_error = local_pc_wrapper->pc()->AddTransceiver(track); scoped_refptr video_transceiver = @@ -1644,7 +1638,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, scoped_refptr local_pc_wrapper = CreatePc(); RtpCodec dummy_codec; - dummy_codec.kind = webrtc::MediaType::AUDIO; + dummy_codec.kind = MediaType::AUDIO; dummy_codec.name = "FOOBAR"; dummy_codec.clock_rate = 90000; dummy_codec.num_channels = 2; @@ -1656,7 +1650,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, init.send_encodings.push_back(encoding_parameters); auto transceiver_or_error = - local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::AUDIO, init); + local_pc_wrapper->pc()->AddTransceiver(MediaType::AUDIO, init); EXPECT_FALSE(transceiver_or_error.ok()); EXPECT_EQ(transceiver_or_error.error().type(), RTCErrorType::UNSUPPORTED_OPERATION); @@ -1667,7 +1661,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, scoped_refptr local_pc_wrapper = CreatePc(); RtpCodec dummy_codec; - dummy_codec.kind = webrtc::MediaType::VIDEO; + dummy_codec.kind = MediaType::VIDEO; dummy_codec.name = "FOOBAR"; dummy_codec.clock_rate = 90000; @@ -1678,7 +1672,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, init.send_encodings.push_back(encoding_parameters); auto transceiver_or_error = - local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO, init); + local_pc_wrapper->pc()->AddTransceiver(MediaType::VIDEO, init); EXPECT_FALSE(transceiver_or_error.ok()); EXPECT_EQ(transceiver_or_error.error().type(), RTCErrorType::UNSUPPORTED_OPERATION); @@ -1689,13 +1683,13 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, scoped_refptr local_pc_wrapper = CreatePc(); RtpCodec dummy_codec; - dummy_codec.kind = webrtc::MediaType::AUDIO; + dummy_codec.kind = MediaType::AUDIO; dummy_codec.name = "FOOBAR"; dummy_codec.clock_rate = 90000; dummy_codec.num_channels = 2; auto transceiver_or_error = - local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::AUDIO); + local_pc_wrapper->pc()->AddTransceiver(MediaType::AUDIO); ASSERT_TRUE(transceiver_or_error.ok()); scoped_refptr audio_transceiver = transceiver_or_error.MoveValue(); @@ -1711,12 +1705,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, scoped_refptr local_pc_wrapper = CreatePc(); RtpCodec dummy_codec; - dummy_codec.kind = webrtc::MediaType::VIDEO; + dummy_codec.kind = MediaType::VIDEO; dummy_codec.name = "FOOBAR"; dummy_codec.clock_rate = 90000; auto transceiver_or_error = - local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO); + local_pc_wrapper->pc()->AddTransceiver(MediaType::VIDEO); ASSERT_TRUE(transceiver_or_error.ok()); scoped_refptr video_transceiver = transceiver_or_error.MoveValue(); @@ -1734,13 +1728,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); std::optional opus = - local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::AUDIO, - "opus"); + local_pc_wrapper->FindFirstSendCodecWithName(MediaType::AUDIO, "opus"); ASSERT_TRUE(opus); std::vector not_opus_codecs = local_pc_wrapper->pc_factory() - ->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO) + ->GetRtpSenderCapabilities(MediaType::AUDIO) .codecs; not_opus_codecs.erase( std::remove_if(not_opus_codecs.begin(), not_opus_codecs.end(), @@ -1750,7 +1743,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, not_opus_codecs.end()); auto transceiver_or_error = - local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::AUDIO); + local_pc_wrapper->pc()->AddTransceiver(MediaType::AUDIO); ASSERT_TRUE(transceiver_or_error.ok()); scoped_refptr audio_transceiver = transceiver_or_error.MoveValue(); @@ -1773,13 +1766,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); std::optional opus = - local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::AUDIO, - "opus"); + local_pc_wrapper->FindFirstSendCodecWithName(MediaType::AUDIO, "opus"); ASSERT_TRUE(opus); std::vector not_opus_codecs = local_pc_wrapper->pc_factory() - ->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO) + ->GetRtpSenderCapabilities(MediaType::AUDIO) .codecs; not_opus_codecs.erase( std::remove_if(not_opus_codecs.begin(), not_opus_codecs.end(), @@ -1789,7 +1781,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, not_opus_codecs.end()); auto transceiver_or_error = - local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::AUDIO); + local_pc_wrapper->pc()->AddTransceiver(MediaType::AUDIO); ASSERT_TRUE(transceiver_or_error.ok()); scoped_refptr audio_transceiver = transceiver_or_error.MoveValue(); @@ -1840,7 +1832,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); auto transceiver_or_error = - local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO); + local_pc_wrapper->pc()->AddTransceiver(MediaType::VIDEO); ASSERT_TRUE(transceiver_or_error.ok()); scoped_refptr video_transceiver = transceiver_or_error.MoveValue(); @@ -1901,13 +1893,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); std::optional vp8 = - local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::VIDEO, - "vp8"); + local_pc_wrapper->FindFirstSendCodecWithName(MediaType::VIDEO, "vp8"); ASSERT_TRUE(vp8); std::vector not_vp8_codecs = local_pc_wrapper->pc_factory() - ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO) + ->GetRtpSenderCapabilities(MediaType::VIDEO) .codecs; not_vp8_codecs.erase( std::remove_if(not_vp8_codecs.begin(), not_vp8_codecs.end(), @@ -1917,7 +1908,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, not_vp8_codecs.end()); auto transceiver_or_error = - local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO); + local_pc_wrapper->pc()->AddTransceiver(MediaType::VIDEO); ASSERT_TRUE(transceiver_or_error.ok()); scoped_refptr video_transceiver = transceiver_or_error.MoveValue(); @@ -1940,13 +1931,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); std::optional vp8 = - local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::VIDEO, - "vp8"); + local_pc_wrapper->FindFirstSendCodecWithName(MediaType::VIDEO, "vp8"); ASSERT_TRUE(vp8); std::vector not_vp8_codecs = local_pc_wrapper->pc_factory() - ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO) + ->GetRtpSenderCapabilities(MediaType::VIDEO) .codecs; not_vp8_codecs.erase( std::remove_if(not_vp8_codecs.begin(), not_vp8_codecs.end(), @@ -1956,7 +1946,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, not_vp8_codecs.end()); auto transceiver_or_error = - local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO); + local_pc_wrapper->pc()->AddTransceiver(MediaType::VIDEO); ASSERT_TRUE(transceiver_or_error.ok()); scoped_refptr video_transceiver = transceiver_or_error.MoveValue(); @@ -2002,13 +1992,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); std::optional opus = - local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::AUDIO, - "opus"); + local_pc_wrapper->FindFirstSendCodecWithName(MediaType::AUDIO, "opus"); ASSERT_TRUE(opus); std::vector not_opus_codecs = local_pc_wrapper->pc_factory() - ->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO) + ->GetRtpSenderCapabilities(MediaType::AUDIO) .codecs; not_opus_codecs.erase( std::remove_if(not_opus_codecs.begin(), not_opus_codecs.end(), @@ -2024,7 +2013,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, init.send_encodings.push_back(encoding_parameters); auto transceiver_or_error = - local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::AUDIO, init); + local_pc_wrapper->pc()->AddTransceiver(MediaType::AUDIO, init); ASSERT_TRUE(transceiver_or_error.ok()); scoped_refptr audio_transceiver = transceiver_or_error.MoveValue(); @@ -2051,17 +2040,15 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, std::vector send_codecs = local_pc_wrapper->pc_factory() - ->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO) + ->GetRtpSenderCapabilities(MediaType::AUDIO) .codecs; std::optional opus = - local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::AUDIO, - "opus"); + local_pc_wrapper->FindFirstSendCodecWithName(MediaType::AUDIO, "opus"); ASSERT_TRUE(opus); std::optional red = - local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::AUDIO, - "red"); + local_pc_wrapper->FindFirstSendCodecWithName(MediaType::AUDIO, "red"); ASSERT_TRUE(red); RtpTransceiverInit init; @@ -2071,7 +2058,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, init.send_encodings.push_back(encoding_parameters); auto transceiver_or_error = - local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::AUDIO, init); + local_pc_wrapper->pc()->AddTransceiver(MediaType::AUDIO, init); ASSERT_TRUE(transceiver_or_error.ok()); scoped_refptr audio_transceiver = transceiver_or_error.MoveValue(); @@ -2106,8 +2093,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, scoped_refptr local_pc_wrapper = CreatePc(); std::optional vp8 = - local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::VIDEO, - "vp8"); + local_pc_wrapper->FindFirstSendCodecWithName(MediaType::VIDEO, "vp8"); ASSERT_TRUE(vp8); RtpTransceiverInit init; @@ -2118,7 +2104,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, init.send_encodings.push_back(encoding_parameters); auto transceiver_or_error = - local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO, init); + local_pc_wrapper->pc()->AddTransceiver(MediaType::VIDEO, init); ASSERT_TRUE(transceiver_or_error.ok()); scoped_refptr video_transceiver = transceiver_or_error.MoveValue(); @@ -2136,13 +2122,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); std::optional vp8 = - local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::VIDEO, - "vp8"); + local_pc_wrapper->FindFirstSendCodecWithName(MediaType::VIDEO, "vp8"); ASSERT_TRUE(vp8); std::vector not_vp8_codecs = local_pc_wrapper->pc_factory() - ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO) + ->GetRtpSenderCapabilities(MediaType::VIDEO) .codecs; not_vp8_codecs.erase( std::remove_if(not_vp8_codecs.begin(), not_vp8_codecs.end(), @@ -2163,7 +2148,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, init.send_encodings.push_back(encoding_parameters); auto transceiver_or_error = - local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO, init); + local_pc_wrapper->pc()->AddTransceiver(MediaType::VIDEO, init); ASSERT_TRUE(transceiver_or_error.ok()); scoped_refptr video_transceiver = transceiver_or_error.MoveValue(); @@ -2194,12 +2179,10 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); std::optional vp8 = - local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::VIDEO, - "vp8"); + local_pc_wrapper->FindFirstSendCodecWithName(MediaType::VIDEO, "vp8"); ASSERT_TRUE(vp8); std::optional vp9 = - local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::VIDEO, - "vp9"); + local_pc_wrapper->FindFirstSendCodecWithName(MediaType::VIDEO, "vp9"); ASSERT_TRUE(vp9); RtpTransceiverInit init; @@ -2215,7 +2198,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, init.send_encodings.push_back(encoding_parameters); auto transceiver_or_error = - local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO, init); + local_pc_wrapper->pc()->AddTransceiver(MediaType::VIDEO, init); ASSERT_FALSE(transceiver_or_error.ok()); EXPECT_EQ(transceiver_or_error.error().type(), RTCErrorType::UNSUPPORTED_OPERATION); @@ -2232,12 +2215,10 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); std::optional vp8 = - local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::VIDEO, - "vp8"); + local_pc_wrapper->FindFirstSendCodecWithName(MediaType::VIDEO, "vp8"); ASSERT_TRUE(vp8); std::optional vp9 = - local_pc_wrapper->FindFirstSendCodecWithName(webrtc::MediaType::VIDEO, - "vp9"); + local_pc_wrapper->FindFirstSendCodecWithName(MediaType::VIDEO, "vp9"); ASSERT_TRUE(vp9); RtpTransceiverInit init; @@ -2253,7 +2234,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, init.send_encodings.push_back(encoding_parameters); auto transceiver_or_error = - local_pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO, init); + local_pc_wrapper->pc()->AddTransceiver(MediaType::VIDEO, init); ASSERT_TRUE(transceiver_or_error.ok()); } @@ -2269,7 +2250,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, ScaleToParameterChecking) { encoding.scale_resolution_down_to = {.width = 1280, .height = 720}; init.send_encodings.push_back(encoding); auto transceiver_or_error = - pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO, init); + pc_wrapper->pc()->AddTransceiver(MediaType::VIDEO, init); EXPECT_FALSE(transceiver_or_error.ok()); EXPECT_EQ(transceiver_or_error.error().type(), RTCErrorType::UNSUPPORTED_OPERATION); @@ -2279,7 +2260,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, ScaleToParameterChecking) { .height = 0}; init.send_encodings[1].scale_resolution_down_to = {.width = 0, .height = 720}; transceiver_or_error = - pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO, init); + pc_wrapper->pc()->AddTransceiver(MediaType::VIDEO, init); EXPECT_FALSE(transceiver_or_error.ok()); EXPECT_EQ(transceiver_or_error.error().type(), RTCErrorType::UNSUPPORTED_OPERATION); @@ -2293,7 +2274,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, ScaleToParameterChecking) { .height = 720}; init.send_encodings[1].scale_resolution_down_by = 2.0; transceiver_or_error = - pc_wrapper->pc()->AddTransceiver(webrtc::MediaType::VIDEO, init); + pc_wrapper->pc()->AddTransceiver(MediaType::VIDEO, init); ASSERT_TRUE(transceiver_or_error.ok()); // SetParameters: If `scale_resolution_down_to` is specified on any active @@ -3139,9 +3120,7 @@ TEST_F(PeerConnectionEncodingsFakeCodecsIntegrationTest, H265Singlecast) { ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); scoped_refptr transceiver = - local_pc_wrapper->pc() - ->AddTransceiver(webrtc::MediaType::VIDEO) - .MoveValue(); + local_pc_wrapper->pc()->AddTransceiver(MediaType::VIDEO).MoveValue(); std::vector preferred_codecs = GetCapabilitiesAndRestrictToCodec(local_pc_wrapper, "H265"); transceiver->SetCodecPreferences(preferred_codecs); @@ -3226,7 +3205,7 @@ TEST_F(PeerConnectionEncodingsFakeCodecsIntegrationTest, // H265. std::vector sender_codecs = local_pc_wrapper->pc_factory() - ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO) + ->GetRtpSenderCapabilities(MediaType::VIDEO) .codecs; auto it = std::find_if(sender_codecs.begin(), sender_codecs.end(), [](const RtpCodecCapability codec_capability) { @@ -3271,16 +3250,14 @@ TEST_F(PeerConnectionEncodingsFakeCodecsIntegrationTest, ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); scoped_refptr transceiver = - local_pc_wrapper->pc() - ->AddTransceiver(webrtc::MediaType::VIDEO) - .MoveValue(); + local_pc_wrapper->pc()->AddTransceiver(MediaType::VIDEO).MoveValue(); // Filter on codec name and assert that sender capabilities have codecs for // {sendrecv, sendonly} and the receiver capabilities have codecs for // {sendrecv, recvonly}. std::vector send_codecs = local_pc_wrapper->pc_factory() - ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO) + ->GetRtpSenderCapabilities(MediaType::VIDEO) .codecs; send_codecs.erase(std::remove_if(send_codecs.begin(), send_codecs.end(), [](const RtpCodecCapability& codec) { @@ -3289,7 +3266,7 @@ TEST_F(PeerConnectionEncodingsFakeCodecsIntegrationTest, send_codecs.end()); std::vector recv_codecs = local_pc_wrapper->pc_factory() - ->GetRtpReceiverCapabilities(webrtc::MediaType::VIDEO) + ->GetRtpReceiverCapabilities(MediaType::VIDEO) .codecs; recv_codecs.erase(std::remove_if(recv_codecs.begin(), recv_codecs.end(), [](const RtpCodecCapability& codec) { diff --git a/pc/peer_connection_factory.cc b/pc/peer_connection_factory.cc index 60820a16d7..e42457667f 100644 --- a/pc/peer_connection_factory.cc +++ b/pc/peer_connection_factory.cc @@ -168,17 +168,17 @@ void PeerConnectionFactory::SetOptions(const Options& options) { } RtpCapabilities PeerConnectionFactory::GetRtpSenderCapabilities( - webrtc::MediaType kind) const { + MediaType kind) const { RTC_DCHECK_RUN_ON(signaling_thread()); switch (kind) { - case webrtc::MediaType::AUDIO: { + case MediaType::AUDIO: { Codecs cricket_codecs; cricket_codecs = codec_vendor_.audio_send_codecs().codecs(); auto extensions = GetDefaultEnabledRtpHeaderExtensions(media_engine()->voice()); return ToRtpCapabilities(cricket_codecs, extensions); } - case webrtc::MediaType::VIDEO: { + case MediaType::VIDEO: { Codecs cricket_codecs; cricket_codecs = codec_vendor_.video_send_codecs().codecs(); auto extensions = @@ -193,17 +193,17 @@ RtpCapabilities PeerConnectionFactory::GetRtpSenderCapabilities( } RtpCapabilities PeerConnectionFactory::GetRtpReceiverCapabilities( - webrtc::MediaType kind) const { + MediaType kind) const { RTC_DCHECK_RUN_ON(signaling_thread()); switch (kind) { - case webrtc::MediaType::AUDIO: { + case MediaType::AUDIO: { Codecs cricket_codecs; cricket_codecs = codec_vendor_.audio_recv_codecs().codecs(); auto extensions = GetDefaultEnabledRtpHeaderExtensions(media_engine()->voice()); return ToRtpCapabilities(cricket_codecs, extensions); } - case webrtc::MediaType::VIDEO: { + case MediaType::VIDEO: { Codecs cricket_codecs = codec_vendor_.video_recv_codecs().codecs(); auto extensions = GetDefaultEnabledRtpHeaderExtensions(media_engine()->video()); diff --git a/pc/peer_connection_factory_unittest.cc b/pc/peer_connection_factory_unittest.cc index 7511471f3a..bde4d89d27 100644 --- a/pc/peer_connection_factory_unittest.cc +++ b/pc/peer_connection_factory_unittest.cc @@ -76,6 +76,8 @@ namespace webrtc { namespace { +using test::MockAudioProcessing; +using test::MockAudioProcessingBuilder; using ::testing::_; using ::testing::A; using ::testing::AtLeast; @@ -84,8 +86,6 @@ using ::testing::NiceMock; using ::testing::NotNull; using ::testing::Return; using ::testing::UnorderedElementsAre; -using ::webrtc::test::MockAudioProcessing; -using ::webrtc::test::MockAudioProcessingBuilder; static const char kStunIceServer[] = "stun:stun.l.google.com:19302"; static const char kTurnIceServer[] = "turn:test.com:1234"; @@ -191,7 +191,7 @@ class PeerConnectionFactoryTest : public ::testing::Test { } void VerifyAudioCodecCapability(const RtpCodecCapability& codec) { - EXPECT_EQ(codec.kind, webrtc::MediaType::AUDIO); + EXPECT_EQ(codec.kind, MediaType::AUDIO); EXPECT_FALSE(codec.name.empty()); EXPECT_GT(codec.clock_rate, 0); EXPECT_GT(codec.num_channels, 0); @@ -199,7 +199,7 @@ class PeerConnectionFactoryTest : public ::testing::Test { void VerifyVideoCodecCapability(const RtpCodecCapability& codec, bool sender) { - EXPECT_EQ(codec.kind, webrtc::MediaType::VIDEO); + EXPECT_EQ(codec.kind, MediaType::VIDEO); EXPECT_FALSE(codec.name.empty()); EXPECT_GT(codec.clock_rate, 0); if (sender) { @@ -334,7 +334,7 @@ TEST(PeerConnectionFactoryTestInternal, DISABLED_CreatePCUsingInternalModules) { TEST_F(PeerConnectionFactoryTest, CheckRtpSenderAudioCapabilities) { RtpCapabilities audio_capabilities = - factory_->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO); + factory_->GetRtpSenderCapabilities(MediaType::AUDIO); EXPECT_FALSE(audio_capabilities.codecs.empty()); for (const auto& codec : audio_capabilities.codecs) { VerifyAudioCodecCapability(codec); @@ -347,7 +347,7 @@ TEST_F(PeerConnectionFactoryTest, CheckRtpSenderAudioCapabilities) { TEST_F(PeerConnectionFactoryTest, CheckRtpSenderVideoCapabilities) { RtpCapabilities video_capabilities = - factory_->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO); + factory_->GetRtpSenderCapabilities(MediaType::VIDEO); EXPECT_FALSE(video_capabilities.codecs.empty()); for (const auto& codec : video_capabilities.codecs) { VerifyVideoCodecCapability(codec, true); @@ -360,7 +360,7 @@ TEST_F(PeerConnectionFactoryTest, CheckRtpSenderVideoCapabilities) { TEST_F(PeerConnectionFactoryTest, CheckRtpSenderRtxEnabledCapabilities) { RtpCapabilities video_capabilities = - factory_->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO); + factory_->GetRtpSenderCapabilities(MediaType::VIDEO); const auto it = std::find_if( video_capabilities.codecs.begin(), video_capabilities.codecs.end(), [](const auto& c) { return c.name == kRtxCodecName; }); @@ -370,7 +370,7 @@ TEST_F(PeerConnectionFactoryTest, CheckRtpSenderRtxEnabledCapabilities) { TEST(PeerConnectionFactoryTestInternal, CheckRtpSenderRtxDisabledCapabilities) { auto factory = CreatePeerConnectionFactoryWithRtxDisabled(); RtpCapabilities video_capabilities = - factory->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO); + factory->GetRtpSenderCapabilities(MediaType::VIDEO); const auto it = std::find_if( video_capabilities.codecs.begin(), video_capabilities.codecs.end(), [](const auto& c) { return c.name == kRtxCodecName; }); @@ -379,14 +379,14 @@ TEST(PeerConnectionFactoryTestInternal, CheckRtpSenderRtxDisabledCapabilities) { TEST_F(PeerConnectionFactoryTest, CheckRtpSenderDataCapabilities) { RtpCapabilities data_capabilities = - factory_->GetRtpSenderCapabilities(webrtc::MediaType::DATA); + factory_->GetRtpSenderCapabilities(MediaType::DATA); EXPECT_TRUE(data_capabilities.codecs.empty()); EXPECT_TRUE(data_capabilities.header_extensions.empty()); } TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverAudioCapabilities) { RtpCapabilities audio_capabilities = - factory_->GetRtpReceiverCapabilities(webrtc::MediaType::AUDIO); + factory_->GetRtpReceiverCapabilities(MediaType::AUDIO); EXPECT_FALSE(audio_capabilities.codecs.empty()); for (const auto& codec : audio_capabilities.codecs) { VerifyAudioCodecCapability(codec); @@ -399,7 +399,7 @@ TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverAudioCapabilities) { TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverVideoCapabilities) { RtpCapabilities video_capabilities = - factory_->GetRtpReceiverCapabilities(webrtc::MediaType::VIDEO); + factory_->GetRtpReceiverCapabilities(MediaType::VIDEO); EXPECT_FALSE(video_capabilities.codecs.empty()); for (const auto& codec : video_capabilities.codecs) { VerifyVideoCodecCapability(codec, false); @@ -412,7 +412,7 @@ TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverVideoCapabilities) { TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverRtxEnabledCapabilities) { RtpCapabilities video_capabilities = - factory_->GetRtpReceiverCapabilities(webrtc::MediaType::VIDEO); + factory_->GetRtpReceiverCapabilities(MediaType::VIDEO); const auto it = std::find_if( video_capabilities.codecs.begin(), video_capabilities.codecs.end(), [](const auto& c) { return c.name == kRtxCodecName; }); @@ -423,7 +423,7 @@ TEST(PeerConnectionFactoryTestInternal, CheckRtpReceiverRtxDisabledCapabilities) { auto factory = CreatePeerConnectionFactoryWithRtxDisabled(); RtpCapabilities video_capabilities = - factory->GetRtpReceiverCapabilities(webrtc::MediaType::VIDEO); + factory->GetRtpReceiverCapabilities(MediaType::VIDEO); const auto it = std::find_if( video_capabilities.codecs.begin(), video_capabilities.codecs.end(), [](const auto& c) { return c.name == kRtxCodecName; }); @@ -432,7 +432,7 @@ TEST(PeerConnectionFactoryTestInternal, TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverDataCapabilities) { RtpCapabilities data_capabilities = - factory_->GetRtpReceiverCapabilities(webrtc::MediaType::DATA); + factory_->GetRtpReceiverCapabilities(MediaType::DATA); EXPECT_TRUE(data_capabilities.codecs.empty()); EXPECT_TRUE(data_capabilities.header_extensions.empty()); } diff --git a/pc/peer_connection_field_trial_tests.cc b/pc/peer_connection_field_trial_tests.cc index a0b30b0e94..3d99c3020c 100644 --- a/pc/peer_connection_field_trial_tests.cc +++ b/pc/peer_connection_field_trial_tests.cc @@ -103,7 +103,7 @@ TEST_F(PeerConnectionFieldTrialTest, EnableDependencyDescriptorAdvertised) { "WebRTC-DependencyDescriptorAdvertised/Enabled/")); WrapperPtr caller = CreatePeerConnection(); - caller->AddTransceiver(webrtc::MediaType::VIDEO); + caller->AddTransceiver(MediaType::VIDEO); auto offer = caller->CreateOffer(); auto contents1 = offer->description()->contents(); @@ -111,7 +111,7 @@ TEST_F(PeerConnectionFieldTrialTest, EnableDependencyDescriptorAdvertised) { const MediaContentDescription* media_description1 = contents1[0].media_description(); - EXPECT_EQ(webrtc::MediaType::VIDEO, media_description1->type()); + EXPECT_EQ(MediaType::VIDEO, media_description1->type()); const RtpHeaderExtensions& rtp_header_extensions1 = media_description1->rtp_header_extensions(); @@ -137,7 +137,7 @@ TEST_F(PeerConnectionFieldTrialTest, MAYBE_InjectDependencyDescriptor) { WrapperPtr caller = CreatePeerConnection(); WrapperPtr callee = CreatePeerConnection(); - caller->AddTransceiver(webrtc::MediaType::VIDEO); + caller->AddTransceiver(MediaType::VIDEO); auto offer = caller->CreateOffer(); ContentInfos& contents1 = offer->description()->contents(); @@ -145,7 +145,7 @@ TEST_F(PeerConnectionFieldTrialTest, MAYBE_InjectDependencyDescriptor) { MediaContentDescription* media_description1 = contents1[0].media_description(); - EXPECT_EQ(webrtc::MediaType::VIDEO, media_description1->type()); + EXPECT_EQ(MediaType::VIDEO, media_description1->type()); RtpHeaderExtensions rtp_header_extensions1 = media_description1->rtp_header_extensions(); @@ -189,7 +189,7 @@ TEST_F(PeerConnectionFieldTrialTest, MAYBE_InjectDependencyDescriptor) { MediaContentDescription* media_description2 = contents2[0].media_description(); - EXPECT_EQ(webrtc::MediaType::VIDEO, media_description2->type()); + EXPECT_EQ(MediaType::VIDEO, media_description2->type()); RtpHeaderExtensions rtp_header_extensions2 = media_description2->rtp_header_extensions(); diff --git a/pc/peer_connection_histogram_unittest.cc b/pc/peer_connection_histogram_unittest.cc index ea428e4193..20d8ff823e 100644 --- a/pc/peer_connection_histogram_unittest.cc +++ b/pc/peer_connection_histogram_unittest.cc @@ -51,8 +51,7 @@ using RTCOfferAnswerOptions = PeerConnectionInterface::RTCOfferAnswerOptions; using ::testing::NiceMock; static const char kUsagePatternMetric[] = "WebRTC.PeerConnection.UsagePattern"; -static constexpr webrtc::TimeDelta kDefaultTimeout = - webrtc::TimeDelta::Millis(10000); +static constexpr TimeDelta kDefaultTimeout = TimeDelta::Millis(10000); static const SocketAddress kLocalAddrs[2] = {SocketAddress("1.1.1.1", 0), SocketAddress("2.2.2.2", 0)}; static const SocketAddress kPrivateLocalAddress("10.1.1.1", 0); diff --git a/pc/peer_connection_ice_unittest.cc b/pc/peer_connection_ice_unittest.cc index 7d763db112..9872a9a79d 100644 --- a/pc/peer_connection_ice_unittest.cc +++ b/pc/peer_connection_ice_unittest.cc @@ -269,7 +269,7 @@ class PeerConnectionIceBaseTest : public ::testing::Test { pc_wrapper_ptr->pc()); PeerConnection* pc = static_cast(pc_proxy->internal()); for (const auto& transceiver : pc->GetTransceiversInternal()) { - if (transceiver->media_type() == webrtc::MediaType::AUDIO) { + if (transceiver->media_type() == MediaType::AUDIO) { auto dtls_transport = pc->LookupDtlsTransportByMidInternal( transceiver->internal()->channel()->mid()); return dtls_transport->ice_transport()->internal()->GetIceRole(); @@ -361,11 +361,10 @@ TEST_P(PeerConnectionIceTest, OfferContainsGatheredCandidates) { // Start ICE candidate gathering by setting the local offer. ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer())); - EXPECT_THAT( - WaitUntil([&] { return caller->IsIceGatheringDone(); }, - ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kIceCandidatesTimeout)}), - IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return caller->IsIceGatheringDone(); }, + ::testing::IsTrue(), + {.timeout = TimeDelta::Millis(kIceCandidatesTimeout)}), + IsRtcOk()); auto offer = caller->CreateOffer(); EXPECT_LT(0u, caller->observer()->GetCandidatesByMline(0).size()); @@ -386,11 +385,10 @@ TEST_P(PeerConnectionIceTest, AnswerContainsGatheredCandidates) { ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); ASSERT_TRUE(callee->SetLocalDescription(callee->CreateAnswer())); - EXPECT_THAT( - WaitUntil([&] { return callee->IsIceGatheringDone(); }, - ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kIceCandidatesTimeout)}), - IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return callee->IsIceGatheringDone(); }, + ::testing::IsTrue(), + {.timeout = TimeDelta::Millis(kIceCandidatesTimeout)}), + IsRtcOk()); auto* answer = callee->pc()->local_description(); EXPECT_LT(0u, caller->observer()->GetCandidatesByMline(0).size()); @@ -717,11 +715,10 @@ TEST_P(PeerConnectionIceTest, CandidatesGeneratedForEachLocalInterface) { caller->network()->AddInterface(kLocalAddress2); caller->CreateOfferAndSetAsLocal(); - EXPECT_THAT( - WaitUntil([&] { return caller->IsIceGatheringDone(); }, - ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kIceCandidatesTimeout)}), - IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return caller->IsIceGatheringDone(); }, + ::testing::IsTrue(), + {.timeout = TimeDelta::Millis(kIceCandidatesTimeout)}), + IsRtcOk()); auto candidates = caller->observer()->GetCandidatesByMline(0); EXPECT_PRED_FORMAT2(AssertIpInCandidates, kLocalAddress1, candidates); @@ -787,7 +784,7 @@ TEST_P(PeerConnectionIceTest, AsyncAddIceCandidateIsAddedToRemoteDescription) { }); EXPECT_THAT( WaitUntil([&] { return operation_completed; }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kWaitTimeout)}), + {.timeout = TimeDelta::Millis(kWaitTimeout)}), IsRtcOk()); auto candidates = callee->GetIceCandidatesFromRemoteDescription(); @@ -839,7 +836,7 @@ TEST_P(PeerConnectionIceTest, EXPECT_FALSE(operation_completed); EXPECT_THAT( WaitUntil([&] { return answer_observer->called(); }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kWaitTimeout)}), + {.timeout = TimeDelta::Millis(kWaitTimeout)}), IsRtcOk()); // As soon as it does, AddIceCandidate() will execute without delay, so it // must also have completed. @@ -864,7 +861,7 @@ TEST_P(PeerConnectionIceTest, }); EXPECT_THAT( WaitUntil([&] { return operation_completed; }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kWaitTimeout)}), + {.timeout = TimeDelta::Millis(kWaitTimeout)}), IsRtcOk()); } @@ -901,7 +898,7 @@ TEST_P(PeerConnectionIceTest, callee = nullptr; EXPECT_THAT( WaitUntil([&] { return operation_completed; }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kWaitTimeout)}), + {.timeout = TimeDelta::Millis(kWaitTimeout)}), IsRtcOk()); } @@ -925,7 +922,7 @@ TEST_P(PeerConnectionIceTest, LocalDescriptionUpdatedWhenContinualGathering) { return caller->pc()->local_description()->candidates(0)->count(); }, ::testing::Gt(0), - {.timeout = webrtc::TimeDelta::Millis(kIceCandidatesTimeout)}), + {.timeout = TimeDelta::Millis(kIceCandidatesTimeout)}), IsRtcOk()); } @@ -952,7 +949,7 @@ TEST_P(PeerConnectionIceTest, return caller->pc()->local_description()->candidates(0)->count(); }, ::testing::Gt(0), - {.timeout = webrtc::TimeDelta::Millis(kIceCandidatesTimeout)}), + {.timeout = TimeDelta::Millis(kIceCandidatesTimeout)}), IsRtcOk()); // Remove the only network interface, causing the PeerConnection to signal @@ -965,7 +962,7 @@ TEST_P(PeerConnectionIceTest, return caller->pc()->local_description()->candidates(0)->count(); }, ::testing::Eq(0u), - {.timeout = webrtc::TimeDelta::Millis(kIceCandidatesTimeout)}), + {.timeout = TimeDelta::Millis(kIceCandidatesTimeout)}), IsRtcOk()); EXPECT_LT(0, caller->observer()->num_candidates_removed_); } @@ -983,11 +980,10 @@ TEST_P(PeerConnectionIceTest, // Start ICE candidate gathering by setting the local offer. ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer())); - EXPECT_THAT( - WaitUntil([&] { return caller->IsIceGatheringDone(); }, - ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kIceCandidatesTimeout)}), - IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return caller->IsIceGatheringDone(); }, + ::testing::IsTrue(), + {.timeout = TimeDelta::Millis(kIceCandidatesTimeout)}), + IsRtcOk()); caller->network()->RemoveInterface(kLocalAddress); diff --git a/pc/peer_connection_integrationtest.cc b/pc/peer_connection_integrationtest.cc index 95bd835301..67d9652227 100644 --- a/pc/peer_connection_integrationtest.cc +++ b/pc/peer_connection_integrationtest.cc @@ -620,9 +620,7 @@ TEST_P(PeerConnectionIntegrationTest, AudioToVideoUpgrade) { callee()->SetOfferAnswerOptions(options); } else { callee()->SetRemoteOfferHandler([this] { - callee() - ->GetFirstTransceiverOfType(webrtc::MediaType::VIDEO) - ->StopInternal(); + callee()->GetFirstTransceiverOfType(MediaType::VIDEO)->StopInternal(); }); } // Do offer/answer and make sure audio is still received end-to-end. @@ -657,8 +655,7 @@ TEST_P(PeerConnectionIntegrationTest, AudioToVideoUpgrade) { // the offer, but by default it is send only. auto transceivers = caller()->pc()->GetTransceivers(); ASSERT_EQ(2U, transceivers.size()); - ASSERT_EQ(webrtc::MediaType::VIDEO, - transceivers[1]->receiver()->media_type()); + ASSERT_EQ(MediaType::VIDEO, transceivers[1]->receiver()->media_type()); transceivers[1]->sender()->SetTrack( caller()->CreateLocalVideoTrack().get()); transceivers[1]->SetDirectionWithError( @@ -841,9 +838,7 @@ TEST_P(PeerConnectionIntegrationTest, AnswererRejectsAudioSection) { // Stopping the audio RtpTransceiver will cause the media section to be // rejected in the answer. callee()->SetRemoteOfferHandler([this] { - callee() - ->GetFirstTransceiverOfType(webrtc::MediaType::AUDIO) - ->StopInternal(); + callee()->GetFirstTransceiverOfType(MediaType::AUDIO)->StopInternal(); }); } callee()->AddTrack(callee()->CreateLocalVideoTrack()); @@ -866,8 +861,7 @@ TEST_P(PeerConnectionIntegrationTest, AnswererRejectsAudioSection) { if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) { // The caller's transceiver should have stopped after receiving the answer, // and thus no longer listed in transceivers. - EXPECT_EQ(nullptr, - caller()->GetFirstTransceiverOfType(webrtc::MediaType::AUDIO)); + EXPECT_EQ(nullptr, caller()->GetFirstTransceiverOfType(MediaType::AUDIO)); } } @@ -887,9 +881,7 @@ TEST_P(PeerConnectionIntegrationTest, AnswererRejectsVideoSection) { // Stopping the video RtpTransceiver will cause the media section to be // rejected in the answer. callee()->SetRemoteOfferHandler([this] { - callee() - ->GetFirstTransceiverOfType(webrtc::MediaType::VIDEO) - ->StopInternal(); + callee()->GetFirstTransceiverOfType(MediaType::VIDEO)->StopInternal(); }); } callee()->AddTrack(callee()->CreateLocalAudioTrack()); @@ -912,8 +904,7 @@ TEST_P(PeerConnectionIntegrationTest, AnswererRejectsVideoSection) { if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) { // The caller's transceiver should have stopped after receiving the answer, // and thus is no longer present. - EXPECT_EQ(nullptr, - caller()->GetFirstTransceiverOfType(webrtc::MediaType::VIDEO)); + EXPECT_EQ(nullptr, caller()->GetFirstTransceiverOfType(MediaType::VIDEO)); } } @@ -988,9 +979,7 @@ TEST_P(PeerConnectionIntegrationTest, VideoRejectedInSubsequentOffer) { } }); } else { - caller() - ->GetFirstTransceiverOfType(webrtc::MediaType::VIDEO) - ->StopInternal(); + caller()->GetFirstTransceiverOfType(MediaType::VIDEO)->StopInternal(); } caller()->CreateAndSetAndSignalOffer(); ASSERT_THAT( @@ -2326,9 +2315,7 @@ TEST_P(PeerConnectionIntegrationTest, callee()->SetOfferAnswerOptions(options); } else { callee()->SetRemoteOfferHandler([this] { - callee() - ->GetFirstTransceiverOfType(webrtc::MediaType::VIDEO) - ->StopInternal(); + callee()->GetFirstTransceiverOfType(MediaType::VIDEO)->StopInternal(); }); } caller()->CreateAndSetAndSignalOffer(); @@ -2350,7 +2337,7 @@ TEST_P(PeerConnectionIntegrationTest, } else { // The caller's transceiver is stopped, so we need to add another track. auto caller_transceiver = - caller()->GetFirstTransceiverOfType(webrtc::MediaType::VIDEO); + caller()->GetFirstTransceiverOfType(MediaType::VIDEO); EXPECT_EQ(nullptr, caller_transceiver.get()); caller()->AddVideoTrack(); } @@ -2421,10 +2408,10 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan, MediaFlowsAfterEarlyWarmupWithAddTransceiver) { ASSERT_TRUE(CreatePeerConnectionWrappers()); ConnectFakeSignaling(); - auto audio_result = caller()->pc()->AddTransceiver(webrtc::MediaType::AUDIO); + auto audio_result = caller()->pc()->AddTransceiver(MediaType::AUDIO); ASSERT_EQ(RTCErrorType::NONE, audio_result.error().type()); auto caller_audio_sender = audio_result.MoveValue()->sender(); - auto video_result = caller()->pc()->AddTransceiver(webrtc::MediaType::VIDEO); + auto video_result = caller()->pc()->AddTransceiver(MediaType::VIDEO); ASSERT_EQ(RTCErrorType::NONE, video_result.error().type()); auto caller_video_sender = video_result.MoveValue()->sender(); callee()->SetRemoteOfferHandler([this] { @@ -2577,11 +2564,10 @@ TEST_P(PeerConnectionIntegrationTestWithFakeClock, options.offer_to_receive_video = 1; caller()->SetOfferAnswerOptions(options); caller()->CreateAndSetAndSignalOffer(); - EXPECT_THAT( - WaitUntil([&] { return DtlsConnected(); }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(total_connection_time_ms), - .clock = &FakeClock()}), - IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return DtlsConnected(); }, ::testing::IsTrue(), + {.timeout = TimeDelta::Millis(total_connection_time_ms), + .clock = &FakeClock()}), + IsRtcOk()); // Closing the PeerConnections destroys the ports before the ScopedFakeClock. // If this is not done a DCHECK can be hit in ports.cc, because a large // negative number is calculated for the rtt due to the global clock changing. @@ -2913,7 +2899,7 @@ TEST_P(PeerConnectionIntegrationTest, GetSourcesAudio) { ASSERT_TRUE(ExpectNewFrames(media_expectations)); ASSERT_EQ(callee()->pc()->GetReceivers().size(), 1u); auto receiver = callee()->pc()->GetReceivers()[0]; - ASSERT_EQ(receiver->media_type(), webrtc::MediaType::AUDIO); + ASSERT_EQ(receiver->media_type(), MediaType::AUDIO); auto sources = receiver->GetSources(); ASSERT_GT(receiver->GetParameters().encodings.size(), 0u); EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc, @@ -2936,7 +2922,7 @@ TEST_P(PeerConnectionIntegrationTest, GetSourcesVideo) { ASSERT_TRUE(ExpectNewFrames(media_expectations)); ASSERT_EQ(callee()->pc()->GetReceivers().size(), 1u); auto receiver = callee()->pc()->GetReceivers()[0]; - ASSERT_EQ(receiver->media_type(), webrtc::MediaType::VIDEO); + ASSERT_EQ(receiver->media_type(), MediaType::VIDEO); auto sources = receiver->GetSources(); ASSERT_GT(receiver->GetParameters().encodings.size(), 0u); ASSERT_GT(sources.size(), 0u); @@ -3649,7 +3635,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan, config.sdp_semantics = SdpSemantics::kUnifiedPlan; ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config)); ConnectFakeSignaling(); - caller()->pc()->AddTransceiver(webrtc::MediaType::AUDIO); + caller()->pc()->AddTransceiver(MediaType::AUDIO); caller()->CreateAndSetAndSignalOffer(); ASSERT_THAT( @@ -3663,7 +3649,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan, while (current_size < 8) { // Double the number of tracks for (int i = 0; i < current_size; i++) { - caller()->pc()->AddTransceiver(webrtc::MediaType::AUDIO); + caller()->pc()->AddTransceiver(MediaType::AUDIO); } current_size = caller()->pc()->GetTransceivers().size(); RTC_LOG(LS_INFO) << "Renegotiating with " << current_size << " tracks"; @@ -3687,7 +3673,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan, config.sdp_semantics = SdpSemantics::kUnifiedPlan; ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config)); ConnectFakeSignaling(); - caller()->pc()->AddTransceiver(webrtc::MediaType::VIDEO); + caller()->pc()->AddTransceiver(MediaType::VIDEO); caller()->CreateAndSetAndSignalOffer(); ASSERT_THAT( @@ -3703,7 +3689,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan, while (current_size < 8) { // Double the number of tracks for (int i = 0; i < current_size; i++) { - caller()->pc()->AddTransceiver(webrtc::MediaType::VIDEO); + caller()->pc()->AddTransceiver(MediaType::VIDEO); } current_size = caller()->pc()->GetTransceivers().size(); RTC_LOG(LS_INFO) << "Renegotiating with " << current_size << " tracks"; @@ -3749,7 +3735,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan, while (current_size < 16) { // Double the number of tracks for (int i = 0; i < current_size; i++) { - caller()->pc()->AddTransceiver(webrtc::MediaType::VIDEO); + caller()->pc()->AddTransceiver(MediaType::VIDEO); } current_size = caller()->pc()->GetTransceivers().size(); RTC_LOG(LS_INFO) << "Renegotiating with " << current_size << " tracks"; @@ -3773,7 +3759,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan, TEST_F(PeerConnectionIntegrationTestUnifiedPlan, GetParametersHasEncodingsBeforeNegotiation) { ASSERT_TRUE(CreatePeerConnectionWrappers()); - auto result = caller()->pc()->AddTransceiver(webrtc::MediaType::VIDEO); + auto result = caller()->pc()->AddTransceiver(MediaType::VIDEO); auto transceiver = result.MoveValue(); auto parameters = transceiver->sender()->GetParameters(); EXPECT_EQ(parameters.encodings.size(), 1u); @@ -3785,7 +3771,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan, RtpTransceiverInit init; init.send_encodings.push_back({}); init.send_encodings[0].max_bitrate_bps = 12345; - auto result = caller()->pc()->AddTransceiver(webrtc::MediaType::VIDEO, init); + auto result = caller()->pc()->AddTransceiver(MediaType::VIDEO, init); auto transceiver = result.MoveValue(); auto parameters = transceiver->sender()->GetParameters(); ASSERT_EQ(parameters.encodings.size(), 1u); @@ -3850,7 +3836,7 @@ TEST_P(PeerConnectionIntegrationInteropTest, OneAudioLocalToNoMediaRemote) { // has the same track ID as the sending track. auto receivers = callee()->pc()->GetReceivers(); ASSERT_EQ(1u, receivers.size()); - EXPECT_EQ(webrtc::MediaType::AUDIO, receivers[0]->media_type()); + EXPECT_EQ(MediaType::AUDIO, receivers[0]->media_type()); EXPECT_EQ(receivers[0]->track()->id(), audio_sender->track()->id()); MediaExpectations media_expectations; @@ -3871,10 +3857,10 @@ TEST_P(PeerConnectionIntegrationInteropTest, OneAudioOneVideoToNoMediaRemote) { // Verify that one audio and one video receiver have been created on the // remote and that they have the same track IDs as the sending tracks. - auto audio_receivers = callee()->GetReceiversOfType(webrtc::MediaType::AUDIO); + auto audio_receivers = callee()->GetReceiversOfType(MediaType::AUDIO); ASSERT_EQ(1u, audio_receivers.size()); EXPECT_EQ(audio_receivers[0]->track()->id(), audio_sender->track()->id()); - auto video_receivers = callee()->GetReceiversOfType(webrtc::MediaType::VIDEO); + auto video_receivers = callee()->GetReceiversOfType(MediaType::VIDEO); ASSERT_EQ(1u, video_receivers.size()); EXPECT_EQ(video_receivers[0]->track()->id(), video_sender->track()->id()); @@ -3913,7 +3899,7 @@ TEST_P(PeerConnectionIntegrationInteropTest, IsRtcOk()); // Verify that only the audio track has been negotiated. - EXPECT_EQ(0u, caller()->GetReceiversOfType(webrtc::MediaType::VIDEO).size()); + EXPECT_EQ(0u, caller()->GetReceiversOfType(MediaType::VIDEO).size()); // Might also check that the callee's NegotiationNeeded flag is set. // Reverse roles. @@ -4496,7 +4482,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan, RTCConfiguration config; ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config)); ConnectFakeSignaling(); - webrtc::RtpEncodingParameters init_send_encodings; + RtpEncodingParameters init_send_encodings; init_send_encodings.active = false; caller()->pc()->AddTrack(caller()->CreateLocalAudioTrack(), {"name"}, {init_send_encodings}); diff --git a/pc/peer_connection_interface_unittest.cc b/pc/peer_connection_interface_unittest.cc index b917484e89..d95377cb93 100644 --- a/pc/peer_connection_interface_unittest.cc +++ b/pc/peer_connection_interface_unittest.cc @@ -830,7 +830,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { } scoped_refptr GetFirstReceiverOfType( - webrtc::MediaType media_type) { + MediaType media_type) { for (auto receiver : pc_->GetReceivers()) { if (receiver->media_type() == media_type) { return receiver; @@ -852,7 +852,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { } EXPECT_THAT( WaitUntil([&] { return observer->called(); }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); *desc = observer->MoveDescription(); return observer->result(); @@ -880,7 +880,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { if (pc_->signaling_state() != PeerConnectionInterface::kClosed) { EXPECT_THAT( WaitUntil([&] { return observer->called(); }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); } return observer->result(); @@ -906,7 +906,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { return false; EXPECT_THAT( WaitUntil([&] { return observer->called(); }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); return observer->called(); } @@ -917,7 +917,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { pc_->GetStats(callback.get()); EXPECT_THAT( WaitUntil([&] { return callback->called(); }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); return callback->called(); } @@ -1025,7 +1025,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { // Wait for the ice_complete message, so that SDP will have candidates. EXPECT_THAT(WaitUntil([&] { return observer_.ice_gathering_complete_; }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); } @@ -1058,13 +1058,13 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { // Verify that both OnAddStream and OnAddTrack are called. EXPECT_THAT(WaitUntil([&] { return observer_.GetLastAddedStreamId(); }, ::testing::Eq(stream_id), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_THAT( WaitUntil( [&] { return observer_.CountAddTrackEventsForStream(stream_id); }, ::testing::Eq(expected_num_tracks), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); } @@ -1182,7 +1182,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { pc_->CreateOffer(observer.get(), offer_answer_options); EXPECT_THAT( WaitUntil([&] { return observer->called(); }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); return observer->MoveDescription(); } @@ -1763,12 +1763,12 @@ TEST_P(PeerConnectionInterfaceTest, IceCandidates) { EXPECT_THAT(WaitUntil([&] { return observer_.last_candidate(); }, ::testing::Ne(nullptr), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), - IsRtcOk()); - EXPECT_THAT(WaitUntil([&] { return observer_.ice_gathering_complete_; }, - ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return observer_.ice_gathering_complete_; }, + ::testing::IsTrue(), {.timeout = TimeDelta::Millis(kTimeout)}), + IsRtcOk()); EXPECT_TRUE(pc_->AddIceCandidate(observer_.last_candidate())); } @@ -1911,7 +1911,7 @@ TEST_P(PeerConnectionInterfaceTest, GetStatsForSpecificTrack) { // Test that we can get stats on a video track. TEST_P(PeerConnectionInterfaceTest, GetStatsForVideoTrack) { InitiateCall(); - auto video_receiver = GetFirstReceiverOfType(webrtc::MediaType::VIDEO); + auto video_receiver = GetFirstReceiverOfType(MediaType::VIDEO); ASSERT_TRUE(video_receiver); EXPECT_TRUE(DoGetStats(video_receiver->track().get())); } @@ -2437,19 +2437,19 @@ TEST_P(PeerConnectionInterfaceTest, CloseAndTestStreamsAndStates) { EXPECT_EQ(2u, pc_->GetTransceivers().size()); } - auto audio_receiver = GetFirstReceiverOfType(webrtc::MediaType::AUDIO); - auto video_receiver = GetFirstReceiverOfType(webrtc::MediaType::VIDEO); + auto audio_receiver = GetFirstReceiverOfType(MediaType::AUDIO); + auto video_receiver = GetFirstReceiverOfType(MediaType::VIDEO); if (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED) { ASSERT_TRUE(audio_receiver); ASSERT_TRUE(video_receiver); // Track state may be updated asynchronously. EXPECT_THAT(WaitUntil([&] { return audio_receiver->track()->state(); }, ::testing::Eq(MediaStreamTrackInterface::kEnded), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_THAT(WaitUntil([&] { return video_receiver->track()->state(); }, ::testing::Eq(MediaStreamTrackInterface::kEnded), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); } else { ASSERT_FALSE(audio_receiver); @@ -2571,11 +2571,11 @@ TEST_F(PeerConnectionInterfaceTestPlanB, // Track state may be updated asynchronously. EXPECT_THAT(WaitUntil([&] { return audio_track2->state(); }, ::testing::Eq(MediaStreamTrackInterface::kEnded), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_THAT(WaitUntil([&] { return video_track2->state(); }, ::testing::Eq(MediaStreamTrackInterface::kEnded), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); } @@ -2587,9 +2587,9 @@ TEST_P(PeerConnectionInterfaceTest, RejectMediaContent) { // First create and set a remote offer, then reject its video content in our // answer. CreateAndSetRemoteOffer(kSdpStringWithStream1PlanB); - auto audio_receiver = GetFirstReceiverOfType(webrtc::MediaType::AUDIO); + auto audio_receiver = GetFirstReceiverOfType(MediaType::AUDIO); ASSERT_TRUE(audio_receiver); - auto video_receiver = GetFirstReceiverOfType(webrtc::MediaType::VIDEO); + auto video_receiver = GetFirstReceiverOfType(MediaType::VIDEO); ASSERT_TRUE(video_receiver); scoped_refptr remote_audio = @@ -2622,11 +2622,11 @@ TEST_P(PeerConnectionInterfaceTest, RejectMediaContent) { // Track state may be updated asynchronously. EXPECT_THAT(WaitUntil([&] { return remote_audio->state(); }, ::testing::Eq(MediaStreamTrackInterface::kEnded), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); EXPECT_THAT(WaitUntil([&] { return remote_video->state(); }, ::testing::Eq(MediaStreamTrackInterface::kEnded), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), + {.timeout = TimeDelta::Millis(kTimeout)}), IsRtcOk()); } @@ -3613,34 +3613,34 @@ TEST_F(PeerConnectionInterfaceTestPlanB, scoped_refptr video_track( CreateVideoTrack("video_track")); stream->AddTrack(audio_track); - EXPECT_THAT(WaitUntil([&] { return observer_.renegotiation_needed_; }, - ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), - IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return observer_.renegotiation_needed_; }, + ::testing::IsTrue(), {.timeout = TimeDelta::Millis(kTimeout)}), + IsRtcOk()); observer_.renegotiation_needed_ = false; CreateOfferReceiveAnswer(); stream->AddTrack(video_track); - EXPECT_THAT(WaitUntil([&] { return observer_.renegotiation_needed_; }, - ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), - IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return observer_.renegotiation_needed_; }, + ::testing::IsTrue(), {.timeout = TimeDelta::Millis(kTimeout)}), + IsRtcOk()); observer_.renegotiation_needed_ = false; CreateOfferReceiveAnswer(); stream->RemoveTrack(audio_track); - EXPECT_THAT(WaitUntil([&] { return observer_.renegotiation_needed_; }, - ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), - IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return observer_.renegotiation_needed_; }, + ::testing::IsTrue(), {.timeout = TimeDelta::Millis(kTimeout)}), + IsRtcOk()); observer_.renegotiation_needed_ = false; CreateOfferReceiveAnswer(); stream->RemoveTrack(video_track); - EXPECT_THAT(WaitUntil([&] { return observer_.renegotiation_needed_; }, - ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kTimeout)}), - IsRtcOk()); + EXPECT_THAT( + WaitUntil([&] { return observer_.renegotiation_needed_; }, + ::testing::IsTrue(), {.timeout = TimeDelta::Millis(kTimeout)}), + IsRtcOk()); observer_.renegotiation_needed_ = false; } @@ -3698,7 +3698,7 @@ TEST_P(PeerConnectionInterfaceTest, std::vector> rtp_senders = pc_->GetSenders(); ASSERT_EQ(rtp_senders.size(), 1u); - ASSERT_EQ(rtp_senders[0]->media_type(), webrtc::MediaType::VIDEO); + ASSERT_EQ(rtp_senders[0]->media_type(), MediaType::VIDEO); scoped_refptr video_rtp_sender = rtp_senders[0]; RtpParameters parameters = video_rtp_sender->GetParameters(); ASSERT_NE(parameters.degradation_preference, diff --git a/pc/peer_connection_jsep_unittest.cc b/pc/peer_connection_jsep_unittest.cc index b9a3911ca4..90016662d6 100644 --- a/pc/peer_connection_jsep_unittest.cc +++ b/pc/peer_connection_jsep_unittest.cc @@ -59,7 +59,6 @@ namespace webrtc { -using webrtc::MediaContentDescription; using RTCConfiguration = PeerConnectionInterface::RTCConfiguration; using ::testing::Combine; using ::testing::ElementsAre; @@ -129,24 +128,24 @@ TEST_F(PeerConnectionJsepTest, EmptyInitialOffer) { // section. TEST_F(PeerConnectionJsepTest, AudioOnlyInitialOffer) { auto caller = CreatePeerConnection(); - caller->AddTransceiver(webrtc::MediaType::AUDIO); + caller->AddTransceiver(MediaType::AUDIO); auto offer = caller->CreateOffer(); auto contents = offer->description()->contents(); ASSERT_EQ(1u, contents.size()); - EXPECT_EQ(webrtc::MediaType::AUDIO, contents[0].media_description()->type()); + EXPECT_EQ(MediaType::AUDIO, contents[0].media_description()->type()); } // Test than an initial offer with one video track generates one video media // section TEST_F(PeerConnectionJsepTest, VideoOnlyInitialOffer) { auto caller = CreatePeerConnection(); - caller->AddTransceiver(webrtc::MediaType::VIDEO); + caller->AddTransceiver(MediaType::VIDEO); auto offer = caller->CreateOffer(); auto contents = offer->description()->contents(); ASSERT_EQ(1u, contents.size()); - EXPECT_EQ(webrtc::MediaType::VIDEO, contents[0].media_description()->type()); + EXPECT_EQ(MediaType::VIDEO, contents[0].media_description()->type()); } // Test that an initial offer with one data channel generates one data media @@ -158,7 +157,7 @@ TEST_F(PeerConnectionJsepTest, DataOnlyInitialOffer) { auto offer = caller->CreateOffer(); auto contents = offer->description()->contents(); ASSERT_EQ(1u, contents.size()); - EXPECT_EQ(webrtc::MediaType::DATA, contents[0].media_description()->type()); + EXPECT_EQ(MediaType::DATA, contents[0].media_description()->type()); } // Test that creating multiple data channels only results in one data section @@ -178,11 +177,11 @@ TEST_F(PeerConnectionJsepTest, MultipleDataChannelsCreateOnlyOneDataSection) { // JSEP section 5.2.1. TEST_F(PeerConnectionJsepTest, MediaSectionsInInitialOfferOrderedCorrectly) { auto caller = CreatePeerConnection(); - caller->AddTransceiver(webrtc::MediaType::VIDEO); - caller->AddTransceiver(webrtc::MediaType::AUDIO); + caller->AddTransceiver(MediaType::VIDEO); + caller->AddTransceiver(MediaType::AUDIO); RtpTransceiverInit init; init.direction = RtpTransceiverDirection::kSendOnly; - caller->AddTransceiver(webrtc::MediaType::VIDEO, init); + caller->AddTransceiver(MediaType::VIDEO, init); auto offer = caller->CreateOffer(); auto contents = offer->description()->contents(); @@ -190,19 +189,19 @@ TEST_F(PeerConnectionJsepTest, MediaSectionsInInitialOfferOrderedCorrectly) { const MediaContentDescription* media_description1 = contents[0].media_description(); - EXPECT_EQ(webrtc::MediaType::VIDEO, media_description1->type()); + EXPECT_EQ(MediaType::VIDEO, media_description1->type()); EXPECT_EQ(RtpTransceiverDirection::kSendRecv, media_description1->direction()); const MediaContentDescription* media_description2 = contents[1].media_description(); - EXPECT_EQ(webrtc::MediaType::AUDIO, media_description2->type()); + EXPECT_EQ(MediaType::AUDIO, media_description2->type()); EXPECT_EQ(RtpTransceiverDirection::kSendRecv, media_description2->direction()); const MediaContentDescription* media_description3 = contents[2].media_description(); - EXPECT_EQ(webrtc::MediaType::VIDEO, media_description3->type()); + EXPECT_EQ(MediaType::VIDEO, media_description3->type()); EXPECT_EQ(RtpTransceiverDirection::kSendOnly, media_description3->direction()); } @@ -210,8 +209,8 @@ TEST_F(PeerConnectionJsepTest, MediaSectionsInInitialOfferOrderedCorrectly) { // Test that media sections in the initial offer have different mids. TEST_F(PeerConnectionJsepTest, MediaSectionsInInitialOfferHaveDifferentMids) { auto caller = CreatePeerConnection(); - caller->AddTransceiver(webrtc::MediaType::AUDIO); - caller->AddTransceiver(webrtc::MediaType::AUDIO); + caller->AddTransceiver(MediaType::AUDIO); + caller->AddTransceiver(MediaType::AUDIO); auto offer = caller->CreateOffer(); auto contents = offer->description()->contents(); @@ -222,7 +221,7 @@ TEST_F(PeerConnectionJsepTest, MediaSectionsInInitialOfferHaveDifferentMids) { TEST_F(PeerConnectionJsepTest, StoppedTransceiverHasNoMediaSectionInInitialOffer) { auto caller = CreatePeerConnection(); - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO); transceiver->StopInternal(); auto offer = caller->CreateOffer(); @@ -243,8 +242,8 @@ TEST_F(PeerConnectionJsepTest, SetLocalEmptyOfferCreatesNoTransceivers) { TEST_F(PeerConnectionJsepTest, SetLocalOfferSetsTransceiverMid) { auto caller = CreatePeerConnection(); - auto audio_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); - auto video_transceiver = caller->AddTransceiver(webrtc::MediaType::VIDEO); + auto audio_transceiver = caller->AddTransceiver(MediaType::AUDIO); + auto video_transceiver = caller->AddTransceiver(MediaType::VIDEO); auto offer = caller->CreateOffer(); auto audio_mid = offer->description()->contents()[0].mid(); @@ -262,8 +261,8 @@ TEST_F(PeerConnectionJsepTest, SetLocalOfferSetsTransceiverMid) { // transceivers, one for receiving audio and one for receiving video. TEST_F(PeerConnectionJsepTest, SetRemoteOfferCreatesTransceivers) { auto caller = CreatePeerConnection(); - auto caller_audio = caller->AddTransceiver(webrtc::MediaType::AUDIO); - auto caller_video = caller->AddTransceiver(webrtc::MediaType::VIDEO); + auto caller_audio = caller->AddTransceiver(MediaType::AUDIO); + auto caller_video = caller->AddTransceiver(MediaType::VIDEO); auto callee = CreatePeerConnection(); ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); @@ -271,12 +270,12 @@ TEST_F(PeerConnectionJsepTest, SetRemoteOfferCreatesTransceivers) { auto transceivers = callee->pc()->GetTransceivers(); ASSERT_EQ(2u, transceivers.size()); - EXPECT_EQ(webrtc::MediaType::AUDIO, transceivers[0]->media_type()); + EXPECT_EQ(MediaType::AUDIO, transceivers[0]->media_type()); EXPECT_EQ(caller_audio->mid(), transceivers[0]->mid()); EXPECT_EQ(RtpTransceiverDirection::kRecvOnly, transceivers[0]->direction()); EXPECT_EQ(0u, transceivers[0]->sender()->stream_ids().size()); - EXPECT_EQ(webrtc::MediaType::VIDEO, transceivers[1]->media_type()); + EXPECT_EQ(MediaType::VIDEO, transceivers[1]->media_type()); EXPECT_EQ(caller_video->mid(), transceivers[1]->mid()); EXPECT_EQ(RtpTransceiverDirection::kRecvOnly, transceivers[1]->direction()); EXPECT_EQ(0u, transceivers[1]->sender()->stream_ids().size()); @@ -331,7 +330,7 @@ TEST_F(PeerConnectionJsepTest, auto caller = CreatePeerConnection(); caller->AddAudioTrack("a"); auto callee = CreatePeerConnection(); - auto transceiver = callee->AddTransceiver(webrtc::MediaType::AUDIO); + auto transceiver = callee->AddTransceiver(MediaType::AUDIO); ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); @@ -411,9 +410,9 @@ TEST_F(PeerConnectionJsepTest, SetRemoteOfferReusesTransceiversOfBothTypes) { // offered media in the same order and with the same mids. TEST_F(PeerConnectionJsepTest, CreateAnswerHasSameMidsAsOffer) { auto caller = CreatePeerConnection(); - auto first_transceiver = caller->AddTransceiver(webrtc::MediaType::VIDEO); - auto second_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); - auto third_transceiver = caller->AddTransceiver(webrtc::MediaType::VIDEO); + auto first_transceiver = caller->AddTransceiver(MediaType::VIDEO); + auto second_transceiver = caller->AddTransceiver(MediaType::AUDIO); + auto third_transceiver = caller->AddTransceiver(MediaType::VIDEO); caller->CreateDataChannel("dc"); auto callee = CreatePeerConnection(); @@ -426,13 +425,13 @@ TEST_F(PeerConnectionJsepTest, CreateAnswerHasSameMidsAsOffer) { auto answer = callee->CreateAnswer(); auto contents = answer->description()->contents(); ASSERT_EQ(4u, contents.size()); - EXPECT_EQ(webrtc::MediaType::VIDEO, contents[0].media_description()->type()); + EXPECT_EQ(MediaType::VIDEO, contents[0].media_description()->type()); EXPECT_EQ(first_transceiver->mid(), contents[0].mid()); - EXPECT_EQ(webrtc::MediaType::AUDIO, contents[1].media_description()->type()); + EXPECT_EQ(MediaType::AUDIO, contents[1].media_description()->type()); EXPECT_EQ(second_transceiver->mid(), contents[1].mid()); - EXPECT_EQ(webrtc::MediaType::VIDEO, contents[2].media_description()->type()); + EXPECT_EQ(MediaType::VIDEO, contents[2].media_description()->type()); EXPECT_EQ(third_transceiver->mid(), contents[2].mid()); - EXPECT_EQ(webrtc::MediaType::DATA, contents[3].media_description()->type()); + EXPECT_EQ(MediaType::DATA, contents[3].media_description()->type()); EXPECT_EQ(offer_data->mid(), contents[3].mid()); } @@ -460,7 +459,7 @@ TEST_F(PeerConnectionJsepTest, CreateAnswerNegotiatesDirection) { auto caller = CreatePeerConnection(); RtpTransceiverInit init; init.direction = RtpTransceiverDirection::kSendOnly; - caller->AddTransceiver(webrtc::MediaType::AUDIO, init); + caller->AddTransceiver(MediaType::AUDIO, init); auto callee = CreatePeerConnection(); callee->AddAudioTrack("a"); @@ -481,7 +480,7 @@ TEST_F(PeerConnectionJsepTest, CreateAnswerNegotiatesDirection) { // property of the transceivers mentioned in the session description. TEST_F(PeerConnectionJsepTest, SetLocalAnswerUpdatesCurrentDirection) { auto caller = CreatePeerConnection(); - auto caller_audio = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto caller_audio = caller->AddTransceiver(MediaType::AUDIO); caller_audio->SetDirectionWithError(RtpTransceiverDirection::kRecvOnly); auto callee = CreatePeerConnection(); callee->AddAudioTrack("a"); @@ -525,7 +524,7 @@ TEST_F(PeerConnectionJsepTest, SetRemoteAnswerUpdatesCurrentDirection) { TEST_F(PeerConnectionJsepTest, ChangeDirectionFromRecvOnlyToSendRecvDoesNotBreakVideoNegotiation) { auto caller = CreatePeerConnection(); - auto caller_transceiver = caller->AddTransceiver(webrtc::MediaType::VIDEO); + auto caller_transceiver = caller->AddTransceiver(MediaType::VIDEO); auto callee = CreatePeerConnection(); caller_transceiver->SetDirectionWithError(RtpTransceiverDirection::kRecvOnly); @@ -543,7 +542,7 @@ TEST_F(PeerConnectionJsepTest, TEST_F(PeerConnectionJsepTest, ChangeDirectionFromRecvOnlyToSendRecvDoesNotBreakAudioNegotiation) { auto caller = CreatePeerConnection(); - auto caller_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto caller_transceiver = caller->AddTransceiver(MediaType::AUDIO); auto callee = CreatePeerConnection(); caller_transceiver->SetDirectionWithError(RtpTransceiverDirection::kRecvOnly); @@ -584,7 +583,7 @@ TEST_F(PeerConnectionJsepTest, SettingTransceiverInactiveDoesNotStopIt) { TEST_F(PeerConnectionJsepTest, ReOfferMediaSectionForAssociatedStoppedTransceiverIsRejected) { auto caller = CreatePeerConnection(); - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO); auto callee = CreatePeerConnection(); ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); @@ -606,7 +605,7 @@ TEST_F(PeerConnectionJsepTest, TEST_F(PeerConnectionJsepTest, StoppingTransceiverInOfferStopsTransceiverOnRemoteSide) { auto caller = CreatePeerConnection(); - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO); auto callee = CreatePeerConnection(); ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); @@ -629,14 +628,14 @@ TEST_F(PeerConnectionJsepTest, TEST_F(PeerConnectionJsepTest, CreateOfferDoesNotRecycleMediaSectionIfFirstStopped) { auto caller = CreatePeerConnection(); - auto first_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto first_transceiver = caller->AddTransceiver(MediaType::AUDIO); auto callee = CreatePeerConnection(); ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); ASSERT_TRUE( caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal())); - auto second_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto second_transceiver = caller->AddTransceiver(MediaType::AUDIO); first_transceiver->StopInternal(); auto reoffer = caller->CreateOffer(); @@ -652,7 +651,7 @@ TEST_F(PeerConnectionJsepTest, TEST_F(PeerConnectionJsepTest, RecycleMediaSectionWhenStoppingTransceiverOnAnswerer) { auto caller = CreatePeerConnection(); - auto first_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto first_transceiver = caller->AddTransceiver(MediaType::AUDIO); auto callee = CreatePeerConnection(); ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); @@ -714,7 +713,7 @@ TEST_F(PeerConnectionJsepTest, TEST_F(PeerConnectionJsepTest, CreateOfferRecyclesWhenOfferingTwice) { // Do a negotiation with a port 0 for the media section. auto caller = CreatePeerConnection(); - auto first_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto first_transceiver = caller->AddTransceiver(MediaType::AUDIO); auto callee = CreatePeerConnection(); ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get())); first_transceiver->StopInternal(); @@ -760,16 +759,15 @@ TEST_F(PeerConnectionJsepTest, CreateOfferRecyclesWhenOfferingTwice) { // - The new transceiver is associated with the new MID value. class RecycleMediaSectionTest : public PeerConnectionJsepTest, - public ::testing::WithParamInterface< - std::tuple> { + public ::testing::WithParamInterface> { protected: RecycleMediaSectionTest() { first_type_ = std::get<0>(GetParam()); second_type_ = std::get<1>(GetParam()); } - webrtc::MediaType first_type_; - webrtc::MediaType second_type_; + MediaType first_type_; + MediaType second_type_; }; // Test that recycling works properly when a new transceiver recycles an m= @@ -1111,18 +1109,17 @@ TEST_P(RecycleMediaSectionTest, PendingRemoteRejectedAndNotRejectedLocal) { // for the media section. This is needed for full test coverage because // MediaSession has separate functions for processing audio and video media // sections. -INSTANTIATE_TEST_SUITE_P( - PeerConnectionJsepTest, - RecycleMediaSectionTest, - Combine(Values(webrtc::MediaType::AUDIO, webrtc::MediaType::VIDEO), - Values(webrtc::MediaType::AUDIO, webrtc::MediaType::VIDEO))); +INSTANTIATE_TEST_SUITE_P(PeerConnectionJsepTest, + RecycleMediaSectionTest, + Combine(Values(MediaType::AUDIO, MediaType::VIDEO), + Values(MediaType::AUDIO, MediaType::VIDEO))); // Test that a new data channel section will not reuse a recycleable audio or // video media section. Additionally, tests that the new section is added to the // end of the session description. TEST_F(PeerConnectionJsepTest, DataChannelDoesNotRecycleMediaSection) { auto caller = CreatePeerConnection(); - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO); auto callee = CreatePeerConnection(); ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get())); @@ -1136,10 +1133,8 @@ TEST_F(PeerConnectionJsepTest, DataChannelDoesNotRecycleMediaSection) { auto offer = caller->CreateOffer(); auto offer_contents = offer->description()->contents(); ASSERT_EQ(2u, offer_contents.size()); - EXPECT_EQ(webrtc::MediaType::AUDIO, - offer_contents[0].media_description()->type()); - EXPECT_EQ(webrtc::MediaType::DATA, - offer_contents[1].media_description()->type()); + EXPECT_EQ(MediaType::AUDIO, offer_contents[0].media_description()->type()); + EXPECT_EQ(MediaType::DATA, offer_contents[1].media_description()->type()); ASSERT_TRUE( caller->SetLocalDescription(CloneSessionDescription(offer.get()))); @@ -1148,10 +1143,8 @@ TEST_F(PeerConnectionJsepTest, DataChannelDoesNotRecycleMediaSection) { auto answer = callee->CreateAnswer(); auto answer_contents = answer->description()->contents(); ASSERT_EQ(2u, answer_contents.size()); - EXPECT_EQ(webrtc::MediaType::AUDIO, - answer_contents[0].media_description()->type()); - EXPECT_EQ(webrtc::MediaType::DATA, - answer_contents[1].media_description()->type()); + EXPECT_EQ(MediaType::AUDIO, answer_contents[0].media_description()->type()); + EXPECT_EQ(MediaType::DATA, answer_contents[1].media_description()->type()); } // Test that if a new track is added to an existing session that has a data, @@ -1169,8 +1162,8 @@ TEST_F(PeerConnectionJsepTest, AudioTrackAddedAfterDataSectionInReoffer) { auto offer = caller->CreateOffer(); auto contents = offer->description()->contents(); ASSERT_EQ(2u, contents.size()); - EXPECT_EQ(webrtc::MediaType::DATA, contents[0].media_description()->type()); - EXPECT_EQ(webrtc::MediaType::AUDIO, contents[1].media_description()->type()); + EXPECT_EQ(MediaType::DATA, contents[0].media_description()->type()); + EXPECT_EQ(MediaType::AUDIO, contents[1].media_description()->type()); } // Tests for MID properties. @@ -1341,7 +1334,7 @@ TEST_F(PeerConnectionJsepTest, const std::string kTrackId = "audio_track"; auto caller = CreatePeerConnection(); - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO); transceiver->sender()->SetTrack(caller->CreateAudioTrack(kTrackId).get()); auto offer = caller->CreateOffer(); @@ -1359,11 +1352,11 @@ TEST_F(PeerConnectionJsepTest, NoMsidInOfferIfTransceiverDirectionHasNoSend) { RtpTransceiverInit init_recvonly; init_recvonly.direction = RtpTransceiverDirection::kRecvOnly; - ASSERT_TRUE(caller->AddTransceiver(webrtc::MediaType::AUDIO, init_recvonly)); + ASSERT_TRUE(caller->AddTransceiver(MediaType::AUDIO, init_recvonly)); RtpTransceiverInit init_inactive; init_inactive.direction = RtpTransceiverDirection::kInactive; - ASSERT_TRUE(caller->AddTransceiver(webrtc::MediaType::VIDEO, init_inactive)); + ASSERT_TRUE(caller->AddTransceiver(MediaType::VIDEO, init_inactive)); auto offer = caller->CreateOffer(); auto contents = offer->description()->contents(); @@ -1383,13 +1376,13 @@ TEST_F(PeerConnectionJsepTest, NoMsidInAnswerIfNoRespondingTracks) { // no tracks to send in response. RtpTransceiverInit init_recvonly; init_recvonly.direction = RtpTransceiverDirection::kRecvOnly; - caller->AddTransceiver(webrtc::MediaType::AUDIO, init_recvonly); + caller->AddTransceiver(MediaType::AUDIO, init_recvonly); // sendrecv transceiver will get negotiated to recvonly since the callee has // no tracks to send in response. RtpTransceiverInit init_sendrecv; init_sendrecv.direction = RtpTransceiverDirection::kSendRecv; - caller->AddTransceiver(webrtc::MediaType::VIDEO, init_sendrecv); + caller->AddTransceiver(MediaType::VIDEO, init_sendrecv); ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); @@ -1434,7 +1427,7 @@ TEST_F(PeerConnectionJsepTest, IncludeMsidEvenIfDirectionHasChanged) { // any MSID information for that section. TEST_F(PeerConnectionJsepTest, RemoveMsidIfTransceiverStopped) { auto caller = CreatePeerConnection(); - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO); auto callee = CreatePeerConnection(); ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get())); @@ -1591,7 +1584,7 @@ TEST_F(PeerConnectionJsepTest, CurrentDirectionResetWhenRtpTransceiverStopped) { auto caller = CreatePeerConnection(); auto callee = CreatePeerConnection(); - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO); ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get())); @@ -1903,7 +1896,7 @@ TEST_F(PeerConnectionJsepTest, AttemptToRollbackImplicitly) { TEST_F(PeerConnectionJsepTest, RollbackRemovesTransceiver) { auto caller = CreatePeerConnection(); - caller->AddTransceiver(webrtc::MediaType::AUDIO); + caller->AddTransceiver(MediaType::AUDIO); auto callee = CreatePeerConnection(); EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOffer())); ASSERT_EQ(callee->pc()->GetTransceivers().size(), 1u); @@ -1921,7 +1914,7 @@ TEST_F(PeerConnectionJsepTest, RollbackRemovesTransceiver) { TEST_F(PeerConnectionJsepTest, RollbackKeepsTransceiverAndClearsMid) { auto caller = CreatePeerConnection(); - caller->AddTransceiver(webrtc::MediaType::AUDIO); + caller->AddTransceiver(MediaType::AUDIO); auto callee = CreatePeerConnection(); EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOffer())); callee->AddAudioTrack("a"); @@ -1947,7 +1940,7 @@ TEST_F(PeerConnectionJsepTest, RollbackKeepsTransceiverAndClearsMid) { TEST_F(PeerConnectionJsepTest, RollbackKeepsTransceiverAfterAddTrackEvenWhenTrackIsNulled) { auto caller = CreatePeerConnection(); - caller->AddTransceiver(webrtc::MediaType::AUDIO); + caller->AddTransceiver(MediaType::AUDIO); auto callee = CreatePeerConnection(); EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOffer())); callee->AddAudioTrack("a"); @@ -1967,7 +1960,7 @@ TEST_F(PeerConnectionJsepTest, TEST_F(PeerConnectionJsepTest, RollbackRestoresMid) { auto caller = CreatePeerConnection(); - caller->AddTransceiver(webrtc::MediaType::AUDIO); + caller->AddTransceiver(MediaType::AUDIO); auto callee = CreatePeerConnection(); callee->AddAudioTrack("a"); auto offer = callee->CreateOffer(); @@ -1990,7 +1983,7 @@ TEST_F(PeerConnectionJsepTest, RollbackRestoresInitSendEncodings) { init.send_encodings.push_back(encoding); encoding.rid = "lo"; init.send_encodings.push_back(encoding); - caller->AddTransceiver(webrtc::MediaType::VIDEO, init); + caller->AddTransceiver(MediaType::VIDEO, init); auto encodings = caller->pc()->GetTransceivers()[0]->sender()->init_send_encodings(); EXPECT_TRUE(caller->SetLocalDescription(caller->CreateOffer())); @@ -2013,8 +2006,8 @@ TEST_F(PeerConnectionJsepTest, RollbackDoesNotAffectSendEncodings) { init.send_encodings.push_back(encoding); encoding.rid = "lo"; init.send_encodings.push_back(encoding); - caller->AddTransceiver(webrtc::MediaType::VIDEO, init); - callee->AddTransceiver(webrtc::MediaType::VIDEO); + caller->AddTransceiver(MediaType::VIDEO, init); + callee->AddTransceiver(MediaType::VIDEO); callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()); caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()); auto params = caller->pc()->GetTransceivers()[0]->sender()->GetParameters(); @@ -2054,8 +2047,7 @@ TEST_F(PeerConnectionJsepTest, RollbackRestoresMidAndRemovesTransceiver) { EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateRollback())); EXPECT_EQ(callee->pc()->GetTransceivers().size(), 1u); EXPECT_EQ(callee->pc()->GetTransceivers()[0]->mid(), mid); - EXPECT_EQ(callee->pc()->GetTransceivers()[0]->media_type(), - webrtc::MediaType::VIDEO); + EXPECT_EQ(callee->pc()->GetTransceivers()[0]->media_type(), MediaType::VIDEO); EXPECT_TRUE(callee->SetLocalDescription(std::move(offer))); EXPECT_EQ(callee->observer()->remove_track_events_.size(), callee->observer()->add_track_events_.size()); @@ -2091,9 +2083,9 @@ TEST_F(PeerConnectionJsepTest, ImplicitlyRollbackTransceiversWithSameMids) { config.sdp_semantics = SdpSemantics::kUnifiedPlan; config.enable_implicit_rollback = true; auto caller = CreatePeerConnection(config); - caller->AddTransceiver(webrtc::MediaType::VIDEO); + caller->AddTransceiver(MediaType::VIDEO); auto callee = CreatePeerConnection(config); - callee->AddTransceiver(webrtc::MediaType::VIDEO); + callee->AddTransceiver(MediaType::VIDEO); EXPECT_TRUE(callee->CreateOfferAndSetAsLocal()); auto initial_mid = callee->pc()->GetTransceivers()[0]->mid(); EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); @@ -2111,7 +2103,7 @@ TEST_F(PeerConnectionJsepTest, RollbackToNegotiatedStableState) { config.sdp_semantics = SdpSemantics::kUnifiedPlan; config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle; auto caller = CreatePeerConnection(config); - caller->AddTransceiver(webrtc::MediaType::AUDIO); + caller->AddTransceiver(MediaType::AUDIO); auto callee = CreatePeerConnection(config); EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); EXPECT_TRUE(callee->CreateAnswerAndSetAsLocal()); @@ -2162,7 +2154,7 @@ TEST_F(PeerConnectionJsepTest, RollbackHasToDestroyTransport) { TEST_F(PeerConnectionJsepTest, RollbackLocalDirectionChange) { auto caller = CreatePeerConnection(); - caller->AddTransceiver(webrtc::MediaType::AUDIO); + caller->AddTransceiver(MediaType::AUDIO); auto callee = CreatePeerConnection(); EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); EXPECT_TRUE( @@ -2185,7 +2177,7 @@ TEST_F(PeerConnectionJsepTest, RollbackLocalDirectionChange) { TEST_F(PeerConnectionJsepTest, RollbackRemoteDirectionChange) { auto caller = CreatePeerConnection(); - auto caller_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto caller_transceiver = caller->AddTransceiver(MediaType::AUDIO); auto callee = CreatePeerConnection(); callee->AddAudioTrack("a"); EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); @@ -2215,7 +2207,7 @@ TEST_F(PeerConnectionJsepTest, RollbackRemoteDirectionChange) { TEST_F(PeerConnectionJsepTest, RollbackRestoresFiredDirectionAndOnTrackCanFireAgain) { auto caller = CreatePeerConnection(); - auto caller_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto caller_transceiver = caller->AddTransceiver(MediaType::AUDIO); auto callee = CreatePeerConnection(); callee->AddAudioTrack("a"); ASSERT_EQ(callee->pc()->GetTransceivers().size(), 1u); @@ -2245,7 +2237,7 @@ TEST_F(PeerConnectionJsepTest, TEST_F(PeerConnectionJsepTest, RollbackFromInactiveToReceivingMakesOnTrackFire) { auto caller = CreatePeerConnection(); - auto caller_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto caller_transceiver = caller->AddTransceiver(MediaType::AUDIO); auto callee = CreatePeerConnection(); // Perform full O/A so that transceiver is associated. Ontrack fires. EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); @@ -2268,9 +2260,9 @@ TEST_F(PeerConnectionJsepTest, TEST_F(PeerConnectionJsepTest, RollbackAfterMultipleSLD) { auto callee = CreatePeerConnection(); - callee->AddTransceiver(webrtc::MediaType::AUDIO); + callee->AddTransceiver(MediaType::AUDIO); EXPECT_TRUE(callee->CreateOfferAndSetAsLocal()); - callee->AddTransceiver(webrtc::MediaType::VIDEO); + callee->AddTransceiver(MediaType::VIDEO); EXPECT_TRUE(callee->CreateOfferAndSetAsLocal()); callee->observer()->clear_legacy_renegotiation_needed(); callee->observer()->clear_latest_negotiation_needed_event(); @@ -2284,9 +2276,9 @@ TEST_F(PeerConnectionJsepTest, RollbackAfterMultipleSLD) { TEST_F(PeerConnectionJsepTest, NoRollbackNeeded) { auto caller = CreatePeerConnection(); - caller->AddTransceiver(webrtc::MediaType::AUDIO); + caller->AddTransceiver(MediaType::AUDIO); auto callee = CreatePeerConnection(); - callee->AddTransceiver(webrtc::MediaType::AUDIO); + callee->AddTransceiver(MediaType::AUDIO); EXPECT_TRUE(caller->CreateOfferAndSetAsLocal()); EXPECT_TRUE(caller->CreateOfferAndSetAsLocal()); EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOffer())); @@ -2319,7 +2311,7 @@ TEST_F(PeerConnectionJsepTest, DataChannelImplicitRollback) { config.sdp_semantics = SdpSemantics::kUnifiedPlan; config.enable_implicit_rollback = true; auto caller = CreatePeerConnection(config); - caller->AddTransceiver(webrtc::MediaType::VIDEO); + caller->AddTransceiver(MediaType::VIDEO); auto callee = CreatePeerConnection(config); callee->CreateDataChannel("dummy"); EXPECT_TRUE(callee->CreateOfferAndSetAsLocal()); @@ -2336,7 +2328,7 @@ TEST_F(PeerConnectionJsepTest, RollbackRemoteDataChannelThenAddTransceiver) { caller->CreateDataChannel("dummy"); EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOffer())); EXPECT_TRUE(callee->SetRemoteDescription(callee->CreateRollback())); - callee->AddTransceiver(webrtc::MediaType::VIDEO); + callee->AddTransceiver(MediaType::VIDEO); EXPECT_TRUE(callee->CreateOfferAndSetAsLocal()); } @@ -2347,7 +2339,7 @@ TEST_F(PeerConnectionJsepTest, caller->CreateDataChannel("dummy"); EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOffer())); EXPECT_TRUE(callee->SetRemoteDescription(callee->CreateRollback())); - callee->AddTransceiver(webrtc::MediaType::VIDEO); + callee->AddTransceiver(MediaType::VIDEO); callee->CreateDataChannel("dummy"); EXPECT_TRUE(callee->CreateOfferAndSetAsLocal()); } @@ -2365,7 +2357,7 @@ TEST_F(PeerConnectionJsepTest, RollbackRemoteDataChannelThenAddDataChannel) { TEST_F(PeerConnectionJsepTest, RollbackRemoteTransceiverThenAddDataChannel) { auto caller = CreatePeerConnection(); auto callee = CreatePeerConnection(); - caller->AddTransceiver(webrtc::MediaType::VIDEO); + caller->AddTransceiver(MediaType::VIDEO); EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOffer())); EXPECT_TRUE(callee->SetRemoteDescription(callee->CreateRollback())); callee->CreateDataChannel("dummy"); @@ -2376,19 +2368,19 @@ TEST_F(PeerConnectionJsepTest, RollbackRemoteTransceiverThenAddDataChannelAndTransceiver) { auto caller = CreatePeerConnection(); auto callee = CreatePeerConnection(); - caller->AddTransceiver(webrtc::MediaType::VIDEO); + caller->AddTransceiver(MediaType::VIDEO); EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOffer())); EXPECT_TRUE(callee->SetRemoteDescription(callee->CreateRollback())); callee->CreateDataChannel("dummy"); - callee->AddTransceiver(webrtc::MediaType::VIDEO); + callee->AddTransceiver(MediaType::VIDEO); EXPECT_TRUE(callee->CreateOfferAndSetAsLocal()); } TEST_F(PeerConnectionJsepTest, BundleOnlySectionDoesNotNeedRtcpMux) { auto caller = CreatePeerConnection(); auto callee = CreatePeerConnection(); - caller->AddTransceiver(webrtc::MediaType::AUDIO); - caller->AddTransceiver(webrtc::MediaType::VIDEO); + caller->AddTransceiver(MediaType::AUDIO); + caller->AddTransceiver(MediaType::VIDEO); auto offer = caller->CreateOffer(); // Remove rtcp-mux and set bundle-only on the second content. offer->description()->contents()[1].media_description()->set_rtcp_mux(false); diff --git a/pc/peer_connection_media_unittest.cc b/pc/peer_connection_media_unittest.cc index 7eab5c5851..fec693074d 100644 --- a/pc/peer_connection_media_unittest.cc +++ b/pc/peer_connection_media_unittest.cc @@ -235,7 +235,7 @@ class PeerConnectionMediaBaseTest : public ::testing::Test { RtpTransceiverDirection GetMediaContentDirection( const SessionDescriptionInterface* sdesc, - webrtc::MediaType media_type) { + MediaType media_type) { auto* content = GetFirstMediaContent(sdesc->description(), media_type); RTC_DCHECK(content); return content->media_description()->direction(); @@ -389,7 +389,7 @@ TEST_F(PeerConnectionMediaTestPlanB, SimulcastOffer) { options.num_simulcast_layers = 3; auto offer = caller->CreateOffer(options); auto* description = - GetFirstMediaContent(offer->description(), webrtc::MediaType::VIDEO) + GetFirstMediaContent(offer->description(), MediaType::VIDEO) ->media_description(); ASSERT_EQ(1u, description->streams().size()); ASSERT_TRUE(description->streams()[0].get_ssrc_group("SIM")); @@ -399,7 +399,7 @@ TEST_F(PeerConnectionMediaTestPlanB, SimulcastOffer) { caller->SetLocalDescription(std::move(offer)); auto senders = caller->pc()->GetSenders(); ASSERT_EQ(1u, senders.size()); - EXPECT_EQ(webrtc::MediaType::VIDEO, senders[0]->media_type()); + EXPECT_EQ(MediaType::VIDEO, senders[0]->media_type()); EXPECT_EQ(3u, senders[0]->GetParameters().encodings.size()); } @@ -416,7 +416,7 @@ TEST_F(PeerConnectionMediaTestPlanB, SimulcastAnswer) { options.num_simulcast_layers = 3; auto answer = callee->CreateAnswer(options); auto* description = - GetFirstMediaContent(answer->description(), webrtc::MediaType::VIDEO) + GetFirstMediaContent(answer->description(), MediaType::VIDEO) ->media_description(); ASSERT_EQ(1u, description->streams().size()); ASSERT_TRUE(description->streams()[0].get_ssrc_group("SIM")); @@ -426,7 +426,7 @@ TEST_F(PeerConnectionMediaTestPlanB, SimulcastAnswer) { callee->SetLocalDescription(std::move(answer)); auto senders = callee->pc()->GetSenders(); ASSERT_EQ(1u, senders.size()); - EXPECT_EQ(webrtc::MediaType::VIDEO, senders[0]->media_type()); + EXPECT_EQ(MediaType::VIDEO, senders[0]->media_type()); EXPECT_EQ(3u, senders[0]->GetParameters().encodings.size()); } @@ -690,8 +690,7 @@ TEST_P(PeerConnectionMediaOfferDirectionTest, VerifyDirection) { options.offer_to_receive_audio = offer_to_receive_; auto offer = caller->CreateOffer(options); - auto* content = - GetFirstMediaContent(offer->description(), webrtc::MediaType::AUDIO); + auto* content = GetFirstMediaContent(offer->description(), MediaType::AUDIO); if (expected_direction_ == RtpTransceiverDirection::kInactive) { EXPECT_FALSE(content); } else { @@ -777,7 +776,7 @@ TEST_P(PeerConnectionMediaAnswerDirectionTest, VerifyDirection) { auto expected_direction = RtpTransceiverDirectionFromSendRecv(negotiate_send, negotiate_recv); EXPECT_EQ(expected_direction, - GetMediaContentDirection(answer.get(), webrtc::MediaType::AUDIO)); + GetMediaContentDirection(answer.get(), MediaType::AUDIO)); } // Tests that the media section is rejected if and only if the callee has no @@ -838,9 +837,9 @@ TEST_P(PeerConnectionMediaTest, OfferHasDifferentDirectionForAudioVideo) { auto offer = caller->CreateOffer(options); EXPECT_EQ(RtpTransceiverDirection::kRecvOnly, - GetMediaContentDirection(offer.get(), webrtc::MediaType::AUDIO)); + GetMediaContentDirection(offer.get(), MediaType::AUDIO)); EXPECT_EQ(RtpTransceiverDirection::kSendOnly, - GetMediaContentDirection(offer.get(), webrtc::MediaType::VIDEO)); + GetMediaContentDirection(offer.get(), MediaType::VIDEO)); } TEST_P(PeerConnectionMediaTest, AnswerHasDifferentDirectionsForAudioVideo) { @@ -862,9 +861,9 @@ TEST_P(PeerConnectionMediaTest, AnswerHasDifferentDirectionsForAudioVideo) { auto answer = callee->CreateAnswer(options); EXPECT_EQ(RtpTransceiverDirection::kRecvOnly, - GetMediaContentDirection(answer.get(), webrtc::MediaType::AUDIO)); + GetMediaContentDirection(answer.get(), MediaType::AUDIO)); EXPECT_EQ(RtpTransceiverDirection::kSendOnly, - GetMediaContentDirection(answer.get(), webrtc::MediaType::VIDEO)); + GetMediaContentDirection(answer.get(), MediaType::VIDEO)); } void AddComfortNoiseCodecsToSend(FakeMediaEngine* media_engine) { @@ -982,11 +981,11 @@ TEST_P(PeerConnectionMediaTest, class PeerConnectionMediaInvalidMediaTest : public PeerConnectionMediaBaseTest, - public ::testing::WithParamInterface, - std::string>>> { + public ::testing::WithParamInterface< + std::tuple, + std::string>>> { protected: PeerConnectionMediaInvalidMediaTest() : PeerConnectionMediaBaseTest(std::get<0>(GetParam())) { @@ -995,7 +994,7 @@ class PeerConnectionMediaInvalidMediaTest expected_error_ = std::get<2>(param); } - std::function mutator_; + std::function mutator_; std::string expected_error_; }; @@ -1126,7 +1125,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, } void RenameContent(SessionDescription* desc, - webrtc::MediaType media_type, + MediaType media_type, const std::string& new_name) { auto* content = GetFirstMediaContent(desc, media_type); RTC_DCHECK(content); @@ -1153,8 +1152,8 @@ TEST_P(PeerConnectionMediaTest, AnswerHasSameMidsAsOffer) { auto callee = CreatePeerConnectionWithAudioVideo(); auto offer = caller->CreateOffer(); - RenameContent(offer->description(), webrtc::MediaType::AUDIO, kAudioMid); - RenameContent(offer->description(), webrtc::MediaType::VIDEO, kVideoMid); + RenameContent(offer->description(), MediaType::AUDIO, kAudioMid); + RenameContent(offer->description(), MediaType::VIDEO, kVideoMid); ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer))); auto answer = callee->CreateAnswer(); @@ -1172,8 +1171,8 @@ TEST_P(PeerConnectionMediaTest, ReOfferHasSameMidsAsFirstOffer) { auto callee = CreatePeerConnectionWithAudioVideo(); auto offer = caller->CreateOffer(); - RenameContent(offer->description(), webrtc::MediaType::AUDIO, kAudioMid); - RenameContent(offer->description(), webrtc::MediaType::VIDEO, kVideoMid); + RenameContent(offer->description(), MediaType::AUDIO, kAudioMid); + RenameContent(offer->description(), MediaType::VIDEO, kVideoMid); ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer))); ASSERT_TRUE(callee->SetLocalDescription(callee->CreateAnswer())); @@ -1189,8 +1188,8 @@ TEST_P(PeerConnectionMediaTest, SetRemoteDescriptionFailsWithDuplicateMids) { auto callee = CreatePeerConnectionWithAudioVideo(); auto offer = caller->CreateOffer(); - RenameContent(offer->description(), webrtc::MediaType::AUDIO, "same"); - RenameContent(offer->description(), webrtc::MediaType::VIDEO, "same"); + RenameContent(offer->description(), MediaType::AUDIO, "same"); + RenameContent(offer->description(), MediaType::VIDEO, "same"); std::string error; EXPECT_FALSE(callee->SetRemoteDescription(std::move(offer), &error)); @@ -1477,7 +1476,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto transceiver = caller->pc()->GetTransceivers().front(); auto capabilities = - caller->pc_factory()->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO); + caller->pc_factory()->GetRtpSenderCapabilities(MediaType::AUDIO); std::vector codecs; absl::c_copy_if(capabilities.codecs, std::back_inserter(codecs), @@ -1495,12 +1494,10 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto caller = CreatePeerConnectionWithAudio(); auto transceiver = caller->pc()->GetTransceivers().front(); - auto video_codecs = caller->pc_factory() - ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO) - .codecs; - auto codecs = caller->pc_factory() - ->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO) - .codecs; + auto video_codecs = + caller->pc_factory()->GetRtpSenderCapabilities(MediaType::VIDEO).codecs; + auto codecs = + caller->pc_factory()->GetRtpSenderCapabilities(MediaType::AUDIO).codecs; codecs.insert(codecs.end(), video_codecs.begin(), video_codecs.end()); auto result = transceiver->SetCodecPreferences(codecs); EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, result.type()); @@ -1522,9 +1519,8 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto transceiver = RtpTransceiverInternal(caller->pc()->GetTransceivers().front()); - auto codecs = caller->pc_factory() - ->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO) - .codecs; + auto codecs = + caller->pc_factory()->GetRtpSenderCapabilities(MediaType::AUDIO).codecs; auto codecs_only_rtx_red_fec = codecs; auto it = std::remove_if( codecs_only_rtx_red_fec.begin(), codecs_only_rtx_red_fec.end(), @@ -1542,9 +1538,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, SetCodecPreferencesAllAudioCodecs) { auto caller = CreatePeerConnectionWithAudio(); auto sender_audio_codecs = - caller->pc_factory() - ->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO) - .codecs; + caller->pc_factory()->GetRtpSenderCapabilities(MediaType::AUDIO).codecs; auto audio_transceiver = caller->pc()->GetTransceivers().front(); @@ -1561,9 +1555,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto caller = CreatePeerConnectionWithAudio(); auto sender_audio_codecs = - caller->pc_factory() - ->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO) - .codecs; + caller->pc_factory()->GetRtpSenderCapabilities(MediaType::AUDIO).codecs; std::vector empty_codecs = {}; auto audio_transceiver = caller->pc()->GetTransceivers().front(); @@ -1581,12 +1573,10 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto caller = CreatePeerConnectionWithVideo(); auto transceiver = caller->pc()->GetTransceivers().front(); - auto audio_codecs = caller->pc_factory() - ->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO) - .codecs; - auto codecs = caller->pc_factory() - ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO) - .codecs; + auto audio_codecs = + caller->pc_factory()->GetRtpSenderCapabilities(MediaType::AUDIO).codecs; + auto codecs = + caller->pc_factory()->GetRtpSenderCapabilities(MediaType::VIDEO).codecs; codecs.insert(codecs.end(), audio_codecs.begin(), audio_codecs.end()); auto result = transceiver->SetCodecPreferences(codecs); EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, result.type()); @@ -1607,9 +1597,8 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto caller = CreatePeerConnectionWithVideo(std::move(fake_engine)); auto transceiver = caller->pc()->GetTransceivers().front(); - auto codecs = caller->pc_factory() - ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO) - .codecs; + auto codecs = + caller->pc_factory()->GetRtpSenderCapabilities(MediaType::VIDEO).codecs; auto codecs_only_rtx_red_fec = codecs; auto it = std::remove_if( codecs_only_rtx_red_fec.begin(), codecs_only_rtx_red_fec.end(), @@ -1627,9 +1616,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, SetCodecPreferencesAllVideoCodecs) { auto caller = CreatePeerConnectionWithVideo(); auto sender_video_codecs = - caller->pc_factory() - ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO) - .codecs; + caller->pc_factory()->GetRtpSenderCapabilities(MediaType::VIDEO).codecs; auto video_transceiver = caller->pc()->GetTransceivers().front(); @@ -1646,9 +1633,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto caller = CreatePeerConnectionWithVideo(); auto sender_video_codecs = - caller->pc_factory() - ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO) - .codecs; + caller->pc_factory()->GetRtpSenderCapabilities(MediaType::VIDEO).codecs; std::vector empty_codecs = {}; @@ -1667,9 +1652,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto caller = CreatePeerConnectionWithVideo(); auto sender_video_codecs = - caller->pc_factory() - ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO) - .codecs; + caller->pc_factory()->GetRtpSenderCapabilities(MediaType::VIDEO).codecs; auto video_transceiver = caller->pc()->GetTransceivers().front(); @@ -1704,9 +1687,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, SetCodecPreferencesVideoWithRtx) { auto caller = CreatePeerConnectionWithVideo(std::move(caller_fake_engine)); auto sender_video_codecs = - caller->pc_factory() - ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO) - .codecs; + caller->pc_factory()->GetRtpSenderCapabilities(MediaType::VIDEO).codecs; auto video_transceiver = caller->pc()->GetTransceivers().front(); @@ -1751,9 +1732,8 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto caller = CreatePeerConnectionWithVideo(std::move(caller_fake_engine)); auto callee = CreatePeerConnection(std::move(callee_fake_engine)); - auto video_codecs = caller->pc_factory() - ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO) - .codecs; + auto video_codecs = + caller->pc_factory()->GetRtpSenderCapabilities(MediaType::VIDEO).codecs; auto send_transceiver = caller->pc()->GetTransceivers().front(); @@ -1815,9 +1795,8 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto caller = CreatePeerConnectionWithVideo(std::move(caller_fake_engine)); auto callee = CreatePeerConnection(std::move(callee_fake_engine)); - auto video_codecs = caller->pc_factory() - ->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO) - .codecs; + auto video_codecs = + caller->pc_factory()->GetRtpSenderCapabilities(MediaType::VIDEO).codecs; auto send_transceiver = caller->pc()->GetTransceivers().front(); @@ -1865,7 +1844,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto transceiver = caller->pc()->GetTransceivers().front(); auto capabilities = - caller->pc_factory()->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO); + caller->pc_factory()->GetRtpSenderCapabilities(MediaType::AUDIO); EXPECT_TRUE(transceiver->SetCodecPreferences(capabilities.codecs).ok()); options.voice_activity_detection = false; @@ -1897,12 +1876,12 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto audio_transceiver = caller->pc()->GetTransceivers()[0]; auto capabilities = - caller->pc_factory()->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO); + caller->pc_factory()->GetRtpSenderCapabilities(MediaType::AUDIO); EXPECT_TRUE(audio_transceiver->SetCodecPreferences(capabilities.codecs).ok()); auto video_transceiver = caller->pc()->GetTransceivers()[1]; capabilities = - caller->pc_factory()->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO); + caller->pc_factory()->GetRtpSenderCapabilities(MediaType::VIDEO); EXPECT_TRUE(video_transceiver->SetCodecPreferences(capabilities.codecs).ok()); RTCOfferAnswerOptions options; @@ -1942,12 +1921,12 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto audio_transceiver = caller->pc()->GetTransceivers()[0]; auto capabilities = - caller->pc_factory()->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO); + caller->pc_factory()->GetRtpSenderCapabilities(MediaType::AUDIO); EXPECT_TRUE(audio_transceiver->SetCodecPreferences(capabilities.codecs).ok()); auto video_transceiver = caller->pc()->GetTransceivers()[1]; capabilities = - caller->pc_factory()->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO); + caller->pc_factory()->GetRtpSenderCapabilities(MediaType::VIDEO); EXPECT_TRUE(video_transceiver->SetCodecPreferences(capabilities.codecs).ok()); auto answer = caller->CreateAnswer(options); @@ -1988,12 +1967,12 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto audio_transceiver = caller->pc()->GetTransceivers()[0]; auto capabilities = - caller->pc_factory()->GetRtpSenderCapabilities(webrtc::MediaType::AUDIO); + caller->pc_factory()->GetRtpSenderCapabilities(MediaType::AUDIO); EXPECT_TRUE(audio_transceiver->SetCodecPreferences(capabilities.codecs).ok()); auto video_transceiver = caller->pc()->GetTransceivers()[1]; capabilities = - caller->pc_factory()->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO); + caller->pc_factory()->GetRtpSenderCapabilities(MediaType::VIDEO); EXPECT_TRUE(video_transceiver->SetCodecPreferences(capabilities.codecs).ok()); auto reoffer = caller->CreateOffer(options); @@ -2026,8 +2005,8 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto error = audio_transceiver->SetDirectionWithError( RtpTransceiverDirection::kSendOnly); ASSERT_TRUE(error.ok()); - auto capabilities = caller->pc_factory()->GetRtpReceiverCapabilities( - webrtc::MediaType::AUDIO); + auto capabilities = + caller->pc_factory()->GetRtpReceiverCapabilities(MediaType::AUDIO); EXPECT_TRUE(audio_transceiver->SetCodecPreferences(capabilities.codecs).ok()); RTCOfferAnswerOptions options; EXPECT_TRUE(caller->SetLocalDescription(caller->CreateOffer(options))); @@ -2056,7 +2035,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, SetCodecPreferencesVideoNoRtx) { ->SetDirectionWithError(RtpTransceiverDirection::kRecvOnly) .ok()); auto capabilities = - caller->pc_factory()->GetRtpSenderCapabilities(webrtc::MediaType::VIDEO); + caller->pc_factory()->GetRtpSenderCapabilities(MediaType::VIDEO); auto it = std::remove_if(capabilities.codecs.begin(), capabilities.codecs.end(), [](const RtpCodecCapability& codec) { diff --git a/pc/peer_connection_rampup_tests.cc b/pc/peer_connection_rampup_tests.cc index 17cc3a2e0d..4cf6bacaae 100644 --- a/pc/peer_connection_rampup_tests.cc +++ b/pc/peer_connection_rampup_tests.cc @@ -63,9 +63,9 @@ namespace webrtc { namespace { -using ::webrtc::test::GetGlobalMetricsLogger; -using ::webrtc::test::ImprovementDirection; -using ::webrtc::test::Unit; +using test::GetGlobalMetricsLogger; +using test::ImprovementDirection; +using test::Unit; static const int kDefaultTestTimeMs = 15000; static const int kRampUpTimeMs = 5000; diff --git a/pc/peer_connection_rtp_unittest.cc b/pc/peer_connection_rtp_unittest.cc index ad39b76e31..a07bfc4c32 100644 --- a/pc/peer_connection_rtp_unittest.cc +++ b/pc/peer_connection_rtp_unittest.cc @@ -346,11 +346,11 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTransceiverCallsOnTrack) { auto caller = CreatePeerConnection(); auto callee = CreatePeerConnection(); - auto audio_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto audio_transceiver = caller->AddTransceiver(MediaType::AUDIO); RtpTransceiverInit video_transceiver_init; video_transceiver_init.stream_ids = {kStreamId1, kStreamId2}; auto video_transceiver = - caller->AddTransceiver(webrtc::MediaType::VIDEO, video_transceiver_init); + caller->AddTransceiver(MediaType::VIDEO, video_transceiver_init); ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get())); @@ -400,7 +400,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, SetDirectionCallsOnTrack) { auto caller = CreatePeerConnection(); auto callee = CreatePeerConnection(); - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO); EXPECT_TRUE( transceiver->SetDirectionWithError(RtpTransceiverDirection::kInactive) .ok()); @@ -431,7 +431,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, SetDirectionHoldCallsOnTrackTwice) { auto caller = CreatePeerConnection(); auto callee = CreatePeerConnection(); - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO); ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get())); EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size()); @@ -502,7 +502,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, TEST_F(PeerConnectionRtpTestUnifiedPlan, ChangeDirectionInAnswerResultsInRemoveTrackEvent) { auto caller = CreatePeerConnection(); - caller->AddTransceiver(webrtc::MediaType::AUDIO); + caller->AddTransceiver(MediaType::AUDIO); auto callee = CreatePeerConnection(); callee->AddAudioTrack("audio_track", {}); @@ -803,8 +803,8 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, TracksDoNotEndWhenSsrcChanges) { // Caller offers to receive audio and video. RtpTransceiverInit init; init.direction = RtpTransceiverDirection::kRecvOnly; - caller->AddTransceiver(webrtc::MediaType::AUDIO, init); - caller->AddTransceiver(webrtc::MediaType::VIDEO, init); + caller->AddTransceiver(MediaType::AUDIO, init); + caller->AddTransceiver(MediaType::VIDEO, init); // Callee wants to send audio and video tracks. callee->AddTrack(callee->CreateAudioTrack("audio_track"), {}); @@ -957,7 +957,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTransceiverHasCorrectInitProperties) { auto caller = CreatePeerConnection(); - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO); EXPECT_EQ(std::nullopt, transceiver->mid()); EXPECT_FALSE(transceiver->stopped()); EXPECT_EQ(RtpTransceiverDirection::kSendRecv, transceiver->direction()); @@ -970,14 +970,14 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, AddAudioTransceiverCreatesAudioSenderAndReceiver) { auto caller = CreatePeerConnection(); - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); - EXPECT_EQ(webrtc::MediaType::AUDIO, transceiver->media_type()); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO); + EXPECT_EQ(MediaType::AUDIO, transceiver->media_type()); ASSERT_TRUE(transceiver->sender()); - EXPECT_EQ(webrtc::MediaType::AUDIO, transceiver->sender()->media_type()); + EXPECT_EQ(MediaType::AUDIO, transceiver->sender()->media_type()); ASSERT_TRUE(transceiver->receiver()); - EXPECT_EQ(webrtc::MediaType::AUDIO, transceiver->receiver()->media_type()); + EXPECT_EQ(MediaType::AUDIO, transceiver->receiver()->media_type()); auto track = transceiver->receiver()->track(); ASSERT_TRUE(track); @@ -991,14 +991,14 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, AddAudioTransceiverCreatesVideoSenderAndReceiver) { auto caller = CreatePeerConnection(); - auto transceiver = caller->AddTransceiver(webrtc::MediaType::VIDEO); - EXPECT_EQ(webrtc::MediaType::VIDEO, transceiver->media_type()); + auto transceiver = caller->AddTransceiver(MediaType::VIDEO); + EXPECT_EQ(MediaType::VIDEO, transceiver->media_type()); ASSERT_TRUE(transceiver->sender()); - EXPECT_EQ(webrtc::MediaType::VIDEO, transceiver->sender()->media_type()); + EXPECT_EQ(MediaType::VIDEO, transceiver->sender()->media_type()); ASSERT_TRUE(transceiver->receiver()); - EXPECT_EQ(webrtc::MediaType::VIDEO, transceiver->receiver()->media_type()); + EXPECT_EQ(MediaType::VIDEO, transceiver->receiver()->media_type()); auto track = transceiver->receiver()->track(); ASSERT_TRUE(track); @@ -1012,7 +1012,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTransceiverShowsInLists) { auto caller = CreatePeerConnection(); - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO); EXPECT_EQ(std::vector>{transceiver}, caller->pc()->GetTransceivers()); EXPECT_EQ( @@ -1031,7 +1031,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, RtpTransceiverInit init; init.direction = RtpTransceiverDirection::kSendOnly; - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO, init); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO, init); EXPECT_EQ(RtpTransceiverDirection::kSendOnly, transceiver->direction()); } @@ -1086,7 +1086,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTransceiverWithInvalidKindReturnsError) { auto caller = CreatePeerConnection(); - auto result = caller->pc()->AddTransceiver(webrtc::MediaType::DATA); + auto result = caller->pc()->AddTransceiver(MediaType::DATA); EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, result.error().type()); } @@ -1108,7 +1108,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, AddAudioTrackCreatesAudioSender) { auto sender = caller->AddTrack(audio_track); ASSERT_TRUE(sender); - EXPECT_EQ(webrtc::MediaType::AUDIO, sender->media_type()); + EXPECT_EQ(MediaType::AUDIO, sender->media_type()); EXPECT_EQ(audio_track, sender->track()); } @@ -1121,7 +1121,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, AddVideoTrackCreatesVideoSender) { auto sender = caller->AddTrack(video_track); ASSERT_TRUE(sender); - EXPECT_EQ(webrtc::MediaType::VIDEO, sender->media_type()); + EXPECT_EQ(MediaType::VIDEO, sender->media_type()); EXPECT_EQ(video_track, sender->track()); } @@ -1145,7 +1145,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, AddFirstTrackCreatesTransceiver) { TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTrackReusesTransceiver) { auto caller = CreatePeerConnection(); - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO); auto audio_track = caller->CreateAudioTrack("a"); auto sender = caller->AddTrack(audio_track); ASSERT_TRUE(sender); @@ -1161,7 +1161,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTrackWithSendEncodingDoesNotReuseTransceiver) { auto caller = CreatePeerConnection(); - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO); auto audio_track = caller->CreateAudioTrack("a"); RtpEncodingParameters encoding; auto sender = caller->AddTrack(audio_track, {}, {encoding}); @@ -1195,8 +1195,8 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, TwoAddTrackCreatesTwoTransceivers) { TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTrackReusesTransceiverOfType) { auto caller = CreatePeerConnection(); - auto audio_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); - auto video_transceiver = caller->AddTransceiver(webrtc::MediaType::VIDEO); + auto audio_transceiver = caller->AddTransceiver(MediaType::AUDIO); + auto video_transceiver = caller->AddTransceiver(MediaType::VIDEO); auto sender = caller->AddVideoTrack("v"); ASSERT_EQ(2u, caller->pc()->GetTransceivers().size()); @@ -1211,7 +1211,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTrackDoesNotReuseTransceiverOfWrongType) { auto caller = CreatePeerConnection(); - caller->AddTransceiver(webrtc::MediaType::AUDIO); + caller->AddTransceiver(MediaType::AUDIO); auto sender = caller->AddVideoTrack("v"); auto transceivers = caller->pc()->GetTransceivers(); @@ -1226,8 +1226,8 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTrackReusesFirstMatchingTransceiver) { auto caller = CreatePeerConnection(); - caller->AddTransceiver(webrtc::MediaType::AUDIO); - caller->AddTransceiver(webrtc::MediaType::AUDIO); + caller->AddTransceiver(MediaType::AUDIO); + caller->AddTransceiver(MediaType::AUDIO); auto sender = caller->AddAudioTrack("a"); auto transceivers = caller->pc()->GetTransceivers(); @@ -1245,7 +1245,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, RtpTransceiverInit init; init.direction = RtpTransceiverDirection::kInactive; - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO, init); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO, init); EXPECT_TRUE(caller->observer()->legacy_renegotiation_needed()); EXPECT_TRUE(caller->observer()->has_negotiation_needed_event()); @@ -1268,7 +1268,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, RtpTransceiverInit init; init.direction = RtpTransceiverDirection::kRecvOnly; - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO, init); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO, init); EXPECT_TRUE(caller->observer()->legacy_renegotiation_needed()); EXPECT_TRUE(caller->observer()->has_negotiation_needed_event()); @@ -1489,8 +1489,8 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, CreateAnswerSameTrackLabel) { RtpTransceiverInit recvonly; recvonly.direction = RtpTransceiverDirection::kRecvOnly; - caller->AddTransceiver(webrtc::MediaType::AUDIO, recvonly); - caller->AddTransceiver(webrtc::MediaType::VIDEO, recvonly); + caller->AddTransceiver(MediaType::AUDIO, recvonly); + caller->AddTransceiver(MediaType::VIDEO, recvonly); ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer())); @@ -1539,7 +1539,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, EXPECT_FALSE(caller->observer()->legacy_renegotiation_needed()); EXPECT_FALSE(caller->observer()->has_negotiation_needed_event()); - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO); EXPECT_TRUE(caller->observer()->legacy_renegotiation_needed()); EXPECT_TRUE(caller->observer()->has_negotiation_needed_event()); @@ -1558,7 +1558,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, NoRenegotiationNeededAfterTransceiverSetSameDirection) { auto caller = CreatePeerConnection(); - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO); caller->observer()->clear_legacy_renegotiation_needed(); caller->observer()->clear_latest_negotiation_needed_event(); @@ -1573,7 +1573,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, NoRenegotiationNeededAfterSetDirectionOnStoppedTransceiver) { auto caller = CreatePeerConnection(); - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO); transceiver->StopInternal(); caller->observer()->clear_legacy_renegotiation_needed(); @@ -1588,7 +1588,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, CheckStoppedCurrentDirectionOnStoppedTransceiver) { auto caller = CreatePeerConnection(); - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO); transceiver->StopInternal(); EXPECT_TRUE(transceiver->stopping()); @@ -1602,7 +1602,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, CheckForInvalidStateOnStoppingTransceiver) { auto caller = CreatePeerConnection(); - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO); transceiver->StopStandard(); EXPECT_TRUE(transceiver->stopping()); @@ -1618,7 +1618,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, CheckForInvalidStateOnStoppedTransceiver) { auto caller = CreatePeerConnection(); - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO); transceiver->StopInternal(); EXPECT_TRUE(transceiver->stopping()); @@ -1634,7 +1634,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, CheckForTypeErrorForStoppedOnTransceiver) { auto caller = CreatePeerConnection(); - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO); EXPECT_EQ( RTCErrorType::INVALID_PARAMETER, transceiver->SetDirectionWithError(RtpTransceiverDirection::kStopped) @@ -1647,7 +1647,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, SetLocalDescriptionWithStoppedMediaSection) { auto caller = CreatePeerConnection(); auto callee = CreatePeerConnection(); - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO); ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get())); callee->pc()->GetTransceivers()[0]->StopStandard(); ASSERT_TRUE(callee->ExchangeOfferAnswerWith(caller.get())); @@ -1660,7 +1660,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, StopAndNegotiateCausesTransceiverToDisappear) { auto caller = CreatePeerConnection(); auto callee = CreatePeerConnection(); - auto transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto transceiver = caller->AddTransceiver(MediaType::AUDIO); ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get())); callee->pc()->GetTransceivers()[0]->StopStandard(); ASSERT_TRUE(callee->ExchangeOfferAnswerWith(caller.get())); @@ -1724,11 +1724,9 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, // Unimplemented RtpParameters: ssrc, codec_payload_type, fec, rtx, dtx, // ptime, scale_framerate_down_by, dependency_rids. init.send_encodings[0].ssrc = 1; - EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER, - caller->pc() - ->AddTransceiver(webrtc::MediaType::AUDIO, init) - .error() - .type()); + EXPECT_EQ( + RTCErrorType::UNSUPPORTED_PARAMETER, + caller->pc()->AddTransceiver(MediaType::AUDIO, init).error().type()); init.send_encodings = default_send_encodings; } @@ -1743,74 +1741,58 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, CheckForInvalidEncodingParameters) { auto default_send_encodings = init.send_encodings; init.send_encodings[0].scale_resolution_down_by = 0.5; - EXPECT_EQ(RTCErrorType::INVALID_RANGE, - caller->pc() - ->AddTransceiver(webrtc::MediaType::VIDEO, init) - .error() - .type()); + EXPECT_EQ( + RTCErrorType::INVALID_RANGE, + caller->pc()->AddTransceiver(MediaType::VIDEO, init).error().type()); init.send_encodings = default_send_encodings; init.send_encodings[0].bitrate_priority = 0; - EXPECT_EQ(RTCErrorType::INVALID_RANGE, - caller->pc() - ->AddTransceiver(webrtc::MediaType::VIDEO, init) - .error() - .type()); + EXPECT_EQ( + RTCErrorType::INVALID_RANGE, + caller->pc()->AddTransceiver(MediaType::VIDEO, init).error().type()); init.send_encodings = default_send_encodings; init.send_encodings[0].min_bitrate_bps = 200000; init.send_encodings[0].max_bitrate_bps = 100000; - EXPECT_EQ(RTCErrorType::INVALID_RANGE, - caller->pc() - ->AddTransceiver(webrtc::MediaType::VIDEO, init) - .error() - .type()); + EXPECT_EQ( + RTCErrorType::INVALID_RANGE, + caller->pc()->AddTransceiver(MediaType::VIDEO, init).error().type()); init.send_encodings = default_send_encodings; init.send_encodings[0].num_temporal_layers = 0; - EXPECT_EQ(RTCErrorType::INVALID_RANGE, - caller->pc() - ->AddTransceiver(webrtc::MediaType::VIDEO, init) - .error() - .type()); + EXPECT_EQ( + RTCErrorType::INVALID_RANGE, + caller->pc()->AddTransceiver(MediaType::VIDEO, init).error().type()); init.send_encodings = default_send_encodings; init.send_encodings[0].num_temporal_layers = 5; - EXPECT_EQ(RTCErrorType::INVALID_RANGE, - caller->pc() - ->AddTransceiver(webrtc::MediaType::VIDEO, init) - .error() - .type()); + EXPECT_EQ( + RTCErrorType::INVALID_RANGE, + caller->pc()->AddTransceiver(MediaType::VIDEO, init).error().type()); init.send_encodings = default_send_encodings; init.send_encodings[0].scalability_mode = std::nullopt; init.send_encodings[0].codec = CreateVideoCodec(SdpVideoFormat("VP8", {})).ToCodecParameters(); - EXPECT_EQ(RTCErrorType::NONE, - caller->pc() - ->AddTransceiver(webrtc::MediaType::VIDEO, init) - .error() - .type()); + EXPECT_EQ( + RTCErrorType::NONE, + caller->pc()->AddTransceiver(MediaType::VIDEO, init).error().type()); init.send_encodings = default_send_encodings; init.send_encodings[0].scalability_mode = "L1T2"; init.send_encodings[0].codec = CreateVideoCodec(SdpVideoFormat("VP8", {})).ToCodecParameters(); - EXPECT_EQ(RTCErrorType::NONE, - caller->pc() - ->AddTransceiver(webrtc::MediaType::VIDEO, init) - .error() - .type()); + EXPECT_EQ( + RTCErrorType::NONE, + caller->pc()->AddTransceiver(MediaType::VIDEO, init).error().type()); init.send_encodings = default_send_encodings; init.send_encodings[0].scalability_mode = "L2T2"; init.send_encodings[0].codec = CreateVideoCodec(SdpVideoFormat("VP8", {})).ToCodecParameters(); - EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION, - caller->pc() - ->AddTransceiver(webrtc::MediaType::VIDEO, init) - .error() - .type()); + EXPECT_EQ( + RTCErrorType::UNSUPPORTED_OPERATION, + caller->pc()->AddTransceiver(MediaType::VIDEO, init).error().type()); init.send_encodings = default_send_encodings; } @@ -1824,7 +1806,7 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, SendEncodingsPassedToSender) { init.send_encodings[0].active = false; init.send_encodings[0].max_bitrate_bps = 180000; - auto result = caller->pc()->AddTransceiver(webrtc::MediaType::AUDIO, init); + auto result = caller->pc()->AddTransceiver(MediaType::AUDIO, init); ASSERT_TRUE(result.ok()); auto init_send_encodings = result.value()->sender()->init_send_encodings(); diff --git a/pc/peer_connection_signaling_unittest.cc b/pc/peer_connection_signaling_unittest.cc index 645d62dc35..2412660fb3 100644 --- a/pc/peer_connection_signaling_unittest.cc +++ b/pc/peer_connection_signaling_unittest.cc @@ -587,7 +587,7 @@ TEST_P(PeerConnectionSignalingTest, CreateOffersAndShutdown) { // must have received a notification. EXPECT_THAT( WaitUntil([&] { return observer->called(); }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kWaitTimeout)}), + {.timeout = TimeDelta::Millis(kWaitTimeout)}), IsRtcOk()); } } @@ -603,7 +603,7 @@ TEST_P(PeerConnectionSignalingTest, CloseCreateOfferAndShutdown) { caller->pc()->CreateOffer(observer.get(), RTCOfferAnswerOptions()); caller.reset(nullptr); EXPECT_THAT(WaitUntil([&] { return observer->called(); }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kWaitTimeout)}), + {.timeout = TimeDelta::Millis(kWaitTimeout)}), IsRtcOk()); } @@ -697,7 +697,7 @@ TEST_P(PeerConnectionSignalingTest, Thread::Current()->PostTask( [&checkpoint_reached] { checkpoint_reached = true; }); EXPECT_THAT(WaitUntil([&] { return checkpoint_reached; }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kWaitTimeout)}), + {.timeout = TimeDelta::Millis(kWaitTimeout)}), IsRtcOk()); // If resolving the observer was pending, it must now have been called. EXPECT_TRUE(observer->called()); @@ -742,7 +742,7 @@ TEST_P(PeerConnectionSignalingTest, CreateOfferBlocksSetRemoteDescription) { // EXPECT_TRUE_WAIT causes messages to be processed... EXPECT_THAT( WaitUntil([&] { return offer_observer->called(); }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kWaitTimeout)}), + {.timeout = TimeDelta::Millis(kWaitTimeout)}), IsRtcOk()); // Now that the offer has been completed, SetRemoteDescription() will have // been executed next in the chain. @@ -764,7 +764,7 @@ TEST_P(PeerConnectionSignalingTest, // Wait for messages to be processed. EXPECT_THAT(WaitUntil([&] { return observer->called(); }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kWaitTimeout)}), + {.timeout = TimeDelta::Millis(kWaitTimeout)}), IsRtcOk()); EXPECT_TRUE(observer->result()); EXPECT_TRUE(caller->pc()->pending_local_description()); @@ -791,7 +791,7 @@ TEST_P(PeerConnectionSignalingTest, // Wait for messages to be processed. EXPECT_THAT(WaitUntil([&] { return observer->called(); }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kWaitTimeout)}), + {.timeout = TimeDelta::Millis(kWaitTimeout)}), IsRtcOk()); EXPECT_TRUE(observer->result()); EXPECT_TRUE(callee->pc()->current_local_description()); @@ -813,7 +813,7 @@ TEST_P(PeerConnectionSignalingTest, EXPECT_THAT( WaitUntil([&] { return caller_set_local_description_observer->called(); }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kWaitTimeout)}), + {.timeout = TimeDelta::Millis(kWaitTimeout)}), IsRtcOk()); ASSERT_TRUE(caller->pc()->pending_local_description()); @@ -833,7 +833,7 @@ TEST_P(PeerConnectionSignalingTest, EXPECT_THAT( WaitUntil([&] { return callee_set_local_description_observer->called(); }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kWaitTimeout)}), + {.timeout = TimeDelta::Millis(kWaitTimeout)}), IsRtcOk()); // Chaining guarantees SetRemoteDescription() happened before // SetLocalDescription(). @@ -850,8 +850,7 @@ TEST_P(PeerConnectionSignalingTest, EXPECT_THAT( WaitUntil( [&] { return caller_set_remote_description_observer->called(); }, - ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kWaitTimeout)}), + ::testing::IsTrue(), {.timeout = TimeDelta::Millis(kWaitTimeout)}), IsRtcOk()); EXPECT_EQ(PeerConnection::kStable, caller->signaling_state()); @@ -869,7 +868,7 @@ TEST_P(PeerConnectionSignalingTest, // The operation should fail asynchronously. EXPECT_FALSE(observer->called()); EXPECT_THAT(WaitUntil([&] { return observer->called(); }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kWaitTimeout)}), + {.timeout = TimeDelta::Millis(kWaitTimeout)}), IsRtcOk()); EXPECT_FALSE(observer->result()); // This did not affect the signaling state. @@ -891,7 +890,7 @@ TEST_P(PeerConnectionSignalingTest, // The operation should fail asynchronously. EXPECT_FALSE(observer->called()); EXPECT_THAT(WaitUntil([&] { return observer->called(); }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kWaitTimeout)}), + {.timeout = TimeDelta::Millis(kWaitTimeout)}), IsRtcOk()); EXPECT_FALSE(observer->result()); // This did not affect the signaling state. @@ -1185,7 +1184,7 @@ TEST_F(PeerConnectionSignalingUnifiedPlanTest, caller->pc()->CreateOffer(offer_observer.get(), RTCOfferAnswerOptions()); EXPECT_THAT(WaitUntil([&] { return offer_observer->was_called(); }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kWaitTimeout)}), + {.timeout = TimeDelta::Millis(kWaitTimeout)}), IsRtcOk()); } @@ -1257,7 +1256,7 @@ TEST_F(PeerConnectionSignalingUnifiedPlanTest, auto caller = CreatePeerConnection(); EXPECT_FALSE(caller->observer()->has_negotiation_needed_event()); auto transceiver = - caller->AddTransceiver(webrtc::MediaType::AUDIO, RtpTransceiverInit()); + caller->AddTransceiver(MediaType::AUDIO, RtpTransceiverInit()); EXPECT_TRUE(caller->observer()->has_negotiation_needed_event()); EXPECT_TRUE(caller->pc()->ShouldFireNegotiationNeededEvent( caller->observer()->latest_negotiation_needed_event())); @@ -1268,7 +1267,7 @@ TEST_F(PeerConnectionSignalingUnifiedPlanTest, auto caller = CreatePeerConnection(); EXPECT_FALSE(caller->observer()->has_negotiation_needed_event()); auto transceiver = - caller->AddTransceiver(webrtc::MediaType::AUDIO, RtpTransceiverInit()); + caller->AddTransceiver(MediaType::AUDIO, RtpTransceiverInit()); EXPECT_TRUE(caller->observer()->has_negotiation_needed_event()); auto observer = make_ref_counted(); @@ -1284,7 +1283,7 @@ TEST_F(PeerConnectionSignalingUnifiedPlanTest, // When the Operations Chain becomes empty again, a new negotiation needed // event will be generated that is not suppressed. EXPECT_THAT(WaitUntil([&] { return observer->called(); }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kWaitTimeout)}), + {.timeout = TimeDelta::Millis(kWaitTimeout)}), IsRtcOk()); EXPECT_TRUE(caller->observer()->has_negotiation_needed_event()); EXPECT_TRUE(caller->pc()->ShouldFireNegotiationNeededEvent( @@ -1299,7 +1298,7 @@ TEST_F(PeerConnectionSignalingUnifiedPlanTest, EXPECT_FALSE(caller->observer()->has_negotiation_needed_event()); auto transceiver = - callee->AddTransceiver(webrtc::MediaType::AUDIO, RtpTransceiverInit()); + callee->AddTransceiver(MediaType::AUDIO, RtpTransceiverInit()); EXPECT_TRUE(callee->observer()->has_negotiation_needed_event()); // Change signaling state (to "have-remote-offer") by setting a remote offer. @@ -1376,7 +1375,7 @@ TEST_F(PeerConnectionSignalingUnifiedPlanTest, LoopbackSdpIsPossible) { // and triggered surprising behavior. auto caller = CreatePeerConnection(); auto transceiver = - caller->AddTransceiver(webrtc::MediaType::AUDIO, RtpTransceiverInit()); + caller->AddTransceiver(MediaType::AUDIO, RtpTransceiverInit()); auto offer = caller->CreateOffer(RTCOfferAnswerOptions()); std::string offer_sdp; diff --git a/pc/peer_connection_simulcast_unittest.cc b/pc/peer_connection_simulcast_unittest.cc index 7ff413d497..802cc41740 100644 --- a/pc/peer_connection_simulcast_unittest.cc +++ b/pc/peer_connection_simulcast_unittest.cc @@ -138,7 +138,7 @@ class PeerConnectionSimulcastTests : public ::testing::Test { scoped_refptr AddTransceiver( PeerConnectionWrapper* pc, const std::vector& layers, - webrtc::MediaType media_type = webrtc::MediaType::VIDEO) { + MediaType media_type = MediaType::VIDEO) { auto init = CreateTransceiverInit(layers); return pc->AddTransceiver(media_type, init); } @@ -196,7 +196,7 @@ TEST_F(PeerConnectionSimulcastTests, RidsAreAutogeneratedIfNotProvided) { for (RtpEncodingParameters& parameters : init.send_encodings) { parameters.rid = ""; } - auto transceiver = pc->AddTransceiver(webrtc::MediaType::VIDEO, init); + auto transceiver = pc->AddTransceiver(MediaType::VIDEO, init); auto parameters = transceiver->sender()->GetParameters(); ASSERT_EQ(3u, parameters.encodings.size()); EXPECT_THAT(parameters.encodings, @@ -212,7 +212,7 @@ TEST_F(PeerConnectionSimulcastTests, MustSupplyAllOrNoRidsInSimulcast) { auto layers = CreateLayers({"f", "h", "remove"}, true); auto init = CreateTransceiverInit(layers); init.send_encodings[2].rid = ""; - auto error = pc->AddTransceiver(webrtc::MediaType::VIDEO, init); + auto error = pc->AddTransceiver(MediaType::VIDEO, init); EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, error.error().type()); } @@ -222,7 +222,7 @@ TEST_F(PeerConnectionSimulcastTests, ChecksForIllegalRidValues) { auto pc = pc_wrapper->pc(); auto layers = CreateLayers({"f", "h", "~q"}, true); auto init = CreateTransceiverInit(layers); - auto error = pc->AddTransceiver(webrtc::MediaType::VIDEO, init); + auto error = pc->AddTransceiver(MediaType::VIDEO, init); EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, error.error().type()); } @@ -514,8 +514,7 @@ TEST_F(PeerConnectionSimulcastTests, SimulcastAudioRejected) { auto local = CreatePeerConnectionWrapper(); auto remote = CreatePeerConnectionWrapper(); auto layers = CreateLayers({"1", "2", "3", "4"}, true); - auto transceiver = - AddTransceiver(local.get(), layers, webrtc::MediaType::AUDIO); + auto transceiver = AddTransceiver(local.get(), layers, MediaType::AUDIO); // Should only have the first layer. auto parameters = transceiver->sender()->GetParameters(); EXPECT_EQ(1u, parameters.encodings.size()); diff --git a/pc/peer_connection_svc_integrationtest.cc b/pc/peer_connection_svc_integrationtest.cc index de3b035431..9e61012710 100644 --- a/pc/peer_connection_svc_integrationtest.cc +++ b/pc/peer_connection_svc_integrationtest.cc @@ -45,8 +45,7 @@ class PeerConnectionSVCIntegrationTest scoped_refptr transceiver, absl::string_view codec_name) { RtpCapabilities capabilities = - caller()->pc_factory()->GetRtpReceiverCapabilities( - webrtc::MediaType::VIDEO); + caller()->pc_factory()->GetRtpReceiverCapabilities(MediaType::VIDEO); std::vector codecs; for (const RtpCodecCapability& codec_capability : capabilities.codecs) { if (codec_capability.name == codec_name) @@ -100,8 +99,7 @@ TEST_F(PeerConnectionSVCIntegrationTest, SetParametersAcceptsL1T3WithVP8) { ConnectFakeSignaling(); RtpCapabilities capabilities = - caller()->pc_factory()->GetRtpReceiverCapabilities( - webrtc::MediaType::VIDEO); + caller()->pc_factory()->GetRtpReceiverCapabilities(MediaType::VIDEO); std::vector vp8_codec; for (const RtpCodecCapability& codec_capability : capabilities.codecs) { if (codec_capability.name == kVp8CodecName) @@ -243,8 +241,7 @@ TEST_F(PeerConnectionSVCIntegrationTest, FallbackToL1Tx) { auto caller_transceiver = transceiver_or_error.MoveValue(); RtpCapabilities capabilities = - caller()->pc_factory()->GetRtpReceiverCapabilities( - webrtc::MediaType::VIDEO); + caller()->pc_factory()->GetRtpReceiverCapabilities(MediaType::VIDEO); std::vector send_codecs = capabilities.codecs; // Only keep VP9 in the caller send_codecs.erase(std::partition(send_codecs.begin(), send_codecs.end(), diff --git a/pc/peer_connection_wrapper.cc b/pc/peer_connection_wrapper.cc index 4f547c77e7..d95538b733 100644 --- a/pc/peer_connection_wrapper.cc +++ b/pc/peer_connection_wrapper.cc @@ -277,7 +277,7 @@ bool PeerConnectionWrapper::ExchangeOfferAnswerWith( } scoped_refptr PeerConnectionWrapper::AddTransceiver( - webrtc::MediaType media_type) { + MediaType media_type) { RTCErrorOr> result = pc()->AddTransceiver(media_type); EXPECT_EQ(RTCErrorType::NONE, result.error().type()); @@ -285,7 +285,7 @@ scoped_refptr PeerConnectionWrapper::AddTransceiver( } scoped_refptr PeerConnectionWrapper::AddTransceiver( - webrtc::MediaType media_type, + MediaType media_type, const RtpTransceiverInit& init) { RTCErrorOr> result = pc()->AddTransceiver(media_type, init); diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc index a2df878987..780fe49c47 100644 --- a/pc/rtc_stats_collector.cc +++ b/pc/rtc_stats_collector.cc @@ -132,52 +132,47 @@ std::string RTCTransportStatsIDFromTransportChannel( } std::string RTCInboundRtpStreamStatsIDFromSSRC(const std::string& transport_id, - webrtc::MediaType media_type, + MediaType media_type, uint32_t ssrc) { char buf[1024]; SimpleStringBuilder sb(buf); - sb << 'I' << transport_id - << (media_type == webrtc::MediaType::AUDIO ? 'A' : 'V') << ssrc; + sb << 'I' << transport_id << (media_type == MediaType::AUDIO ? 'A' : 'V') + << ssrc; return sb.str(); } std::string RTCOutboundRtpStreamStatsIDFromSSRC(const std::string& transport_id, - webrtc::MediaType media_type, + MediaType media_type, uint32_t ssrc) { char buf[1024]; SimpleStringBuilder sb(buf); - sb << 'O' << transport_id - << (media_type == webrtc::MediaType::AUDIO ? 'A' : 'V') << ssrc; + sb << 'O' << transport_id << (media_type == MediaType::AUDIO ? 'A' : 'V') + << ssrc; return sb.str(); } std::string RTCRemoteInboundRtpStreamStatsIdFromSourceSsrc( - webrtc::MediaType media_type, + MediaType media_type, uint32_t source_ssrc) { char buf[1024]; SimpleStringBuilder sb(buf); - sb << "RI" << (media_type == webrtc::MediaType::AUDIO ? 'A' : 'V') - << source_ssrc; + sb << "RI" << (media_type == MediaType::AUDIO ? 'A' : 'V') << source_ssrc; return sb.str(); } -std::string RTCRemoteOutboundRTPStreamStatsIDFromSSRC( - webrtc::MediaType media_type, - uint32_t source_ssrc) { +std::string RTCRemoteOutboundRTPStreamStatsIDFromSSRC(MediaType media_type, + uint32_t source_ssrc) { char buf[1024]; SimpleStringBuilder sb(buf); - sb << "RO" << (media_type == webrtc::MediaType::AUDIO ? 'A' : 'V') - << source_ssrc; + sb << "RO" << (media_type == MediaType::AUDIO ? 'A' : 'V') << source_ssrc; return sb.str(); } -std::string RTCMediaSourceStatsIDFromKindAndAttachment( - webrtc::MediaType media_type, - int attachment_id) { +std::string RTCMediaSourceStatsIDFromKindAndAttachment(MediaType media_type, + int attachment_id) { char buf[1024]; SimpleStringBuilder sb(buf); - sb << 'S' << (media_type == webrtc::MediaType::AUDIO ? 'A' : 'V') - << attachment_id; + sb << 'S' << (media_type == MediaType::AUDIO ? 'A' : 'V') << attachment_id; return sb.str(); } @@ -452,8 +447,8 @@ std::unique_ptr CreateInboundAudioStreamStats( Timestamp timestamp, RTCStatsReport* report) { auto inbound_audio = std::make_unique( - /*id=*/RTCInboundRtpStreamStatsIDFromSSRC( - transport_id, webrtc::MediaType::AUDIO, voice_receiver_info.ssrc()), + /*id=*/RTCInboundRtpStreamStatsIDFromSSRC(transport_id, MediaType::AUDIO, + voice_receiver_info.ssrc()), timestamp); SetInboundRTPStreamStatsFromMediaReceiverInfo(voice_receiver_info, inbound_audio.get()); @@ -536,7 +531,7 @@ std::unique_ptr CreateRemoteOutboundMediaStreamStats( const MediaReceiverInfo& media_receiver_info, const std::string& mid, - webrtc::MediaType media_type, + MediaType media_type, const RTCInboundRtpStreamStats& inbound_audio_stats, const std::string& transport_id, const bool stats_timestamp_with_environment_clock) { @@ -561,7 +556,7 @@ CreateRemoteOutboundMediaStreamStats( // Populate. // - RTCRtpStreamStats. stats->ssrc = media_receiver_info.ssrc(); - stats->kind = webrtc::MediaTypeToString(media_type); + stats->kind = MediaTypeToString(media_type); stats->transport_id = transport_id; if (inbound_audio_stats.codec_id.has_value()) { stats->codec_id = *inbound_audio_stats.codec_id; @@ -599,7 +594,7 @@ CreateInboundRTPStreamStatsFromVideoReceiverInfo( Timestamp timestamp, RTCStatsReport* report) { auto inbound_video = std::make_unique( - RTCInboundRtpStreamStatsIDFromSSRC(transport_id, webrtc::MediaType::VIDEO, + RTCInboundRtpStreamStatsIDFromSSRC(transport_id, MediaType::VIDEO, video_receiver_info.ssrc()), timestamp); SetInboundRTPStreamStatsFromMediaReceiverInfo(video_receiver_info, @@ -748,8 +743,8 @@ CreateOutboundRTPStreamStatsFromVoiceSenderInfo( Timestamp timestamp, RTCStatsReport* report) { auto outbound_audio = std::make_unique( - RTCOutboundRtpStreamStatsIDFromSSRC( - transport_id, webrtc::MediaType::AUDIO, voice_sender_info.ssrc()), + RTCOutboundRtpStreamStatsIDFromSSRC(transport_id, MediaType::AUDIO, + voice_sender_info.ssrc()), timestamp); SetOutboundRTPStreamStatsFromMediaSenderInfo(voice_sender_info, outbound_audio.get()); @@ -783,8 +778,8 @@ CreateOutboundRTPStreamStatsFromVideoSenderInfo( Timestamp timestamp, RTCStatsReport* report) { auto outbound_video = std::make_unique( - RTCOutboundRtpStreamStatsIDFromSSRC( - transport_id, webrtc::MediaType::VIDEO, video_sender_info.ssrc()), + RTCOutboundRtpStreamStatsIDFromSSRC(transport_id, MediaType::VIDEO, + video_sender_info.ssrc()), timestamp); SetOutboundRTPStreamStatsFromMediaSenderInfo(video_sender_info, outbound_video.get()); @@ -874,7 +869,7 @@ std::unique_ptr ProduceRemoteInboundRtpStreamStatsFromReportBlockData( const std::string& transport_id, const ReportBlockData& report_block, - webrtc::MediaType media_type, + MediaType media_type, const std::map& outbound_rtps, const RTCStatsReport& report, const bool stats_timestamp_with_environment_clock) { @@ -889,8 +884,7 @@ ProduceRemoteInboundRtpStreamStatsFromReportBlockData( media_type, report_block.source_ssrc()), arrival_timestamp); remote_inbound->ssrc = report_block.source_ssrc(); - remote_inbound->kind = - media_type == webrtc::MediaType::AUDIO ? "audio" : "video"; + remote_inbound->kind = media_type == MediaType::AUDIO ? "audio" : "video"; remote_inbound->packets_lost = report_block.cumulative_lost(); remote_inbound->fraction_lost = report_block.fraction_lost(); if (report_block.num_rtts() > 0) { @@ -1605,7 +1599,7 @@ void RTCStatsCollector::ProduceMediaSourceStats_s( static_cast(track.get()); auto audio_source_stats = std::make_unique( RTCMediaSourceStatsIDFromKindAndAttachment( - webrtc::MediaType::AUDIO, sender_internal->AttachmentId()), + MediaType::AUDIO, sender_internal->AttachmentId()), timestamp); // TODO(https://crbug.com/webrtc/10771): We shouldn't need to have an // SSRC assigned (there shouldn't need to exist a send-stream, created @@ -1643,7 +1637,7 @@ void RTCStatsCollector::ProduceMediaSourceStats_s( RTC_DCHECK_EQ(MediaStreamTrackInterface::kVideoKind, track->kind()); auto video_source_stats = std::make_unique( RTCMediaSourceStatsIDFromKindAndAttachment( - webrtc::MediaType::VIDEO, sender_internal->AttachmentId()), + MediaType::VIDEO, sender_internal->AttachmentId()), timestamp); auto* video_track = static_cast(track.get()); auto* video_source = video_track->GetSource(); @@ -1707,9 +1701,9 @@ void RTCStatsCollector::ProduceRTPStreamStats_n( Thread::ScopedDisallowBlockingCalls no_blocking_calls; for (const RtpTransceiverStatsInfo& stats : transceiver_stats_infos) { - if (stats.media_type == webrtc::MediaType::AUDIO) { + if (stats.media_type == MediaType::AUDIO) { ProduceAudioRTPStreamStats_n(timestamp, stats, report); - } else if (stats.media_type == webrtc::MediaType::VIDEO) { + } else if (stats.media_type == MediaType::VIDEO) { ProduceVideoRTPStreamStats_n(timestamp, stats, report); } else { RTC_DCHECK_NOTREACHED(); @@ -1748,7 +1742,7 @@ void RTCStatsCollector::ProduceAudioRTPStreamStats_n( if (audio_track) { inbound_audio->track_identifier = audio_track->id(); } - if (audio_device_stats_ && stats.media_type == webrtc::MediaType::AUDIO && + if (audio_device_stats_ && stats.media_type == MediaType::AUDIO && stats.current_direction && (*stats.current_direction == RtpTransceiverDirection::kSendRecv || *stats.current_direction == RtpTransceiverDirection::kRecvOnly)) { @@ -1762,7 +1756,7 @@ void RTCStatsCollector::ProduceAudioRTPStreamStats_n( } // Remote-outbound. auto remote_outbound_audio = CreateRemoteOutboundMediaStreamStats( - voice_receiver_info, mid, webrtc::MediaType::AUDIO, *inbound_audio_ptr, + voice_receiver_info, mid, MediaType::AUDIO, *inbound_audio_ptr, transport_id, stats_timestamp_with_environment_clock_); // Add stats. if (remote_outbound_audio) { @@ -1794,7 +1788,7 @@ void RTCStatsCollector::ProduceAudioRTPStreamStats_n( stats.track_media_info_map.GetAttachmentIdByTrack(audio_track.get()) .value(); outbound_audio->media_source_id = - RTCMediaSourceStatsIDFromKindAndAttachment(webrtc::MediaType::AUDIO, + RTCMediaSourceStatsIDFromKindAndAttachment(MediaType::AUDIO, attachment_id); } auto audio_outbound_pair = @@ -1815,7 +1809,7 @@ void RTCStatsCollector::ProduceAudioRTPStreamStats_n( stats.track_media_info_map.voice_media_info()->senders) { for (const auto& report_block_data : voice_sender_info.report_block_datas) { report->AddStats(ProduceRemoteInboundRtpStreamStatsFromReportBlockData( - transport_id, report_block_data, webrtc::MediaType::AUDIO, + transport_id, report_block_data, MediaType::AUDIO, audio_outbound_rtps, *report, stats_timestamp_with_environment_clock_)); } @@ -1857,7 +1851,7 @@ void RTCStatsCollector::ProduceVideoRTPStreamStats_n( } // Remote-outbound. auto remote_outbound_video = CreateRemoteOutboundMediaStreamStats( - video_receiver_info, mid, webrtc::MediaType::VIDEO, *inbound_video_ptr, + video_receiver_info, mid, MediaType::VIDEO, *inbound_video_ptr, transport_id, stats_timestamp_with_environment_clock_); // Add stats. if (remote_outbound_video) { @@ -1889,7 +1883,7 @@ void RTCStatsCollector::ProduceVideoRTPStreamStats_n( stats.track_media_info_map.GetAttachmentIdByTrack(video_track.get()) .value(); outbound_video->media_source_id = - RTCMediaSourceStatsIDFromKindAndAttachment(webrtc::MediaType::VIDEO, + RTCMediaSourceStatsIDFromKindAndAttachment(MediaType::VIDEO, attachment_id); } auto video_outbound_pair = @@ -1910,7 +1904,7 @@ void RTCStatsCollector::ProduceVideoRTPStreamStats_n( stats.track_media_info_map.video_media_info()->senders) { for (const auto& report_block_data : video_sender_info.report_block_datas) { report->AddStats(ProduceRemoteInboundRtpStreamStatsFromReportBlockData( - transport_id, report_block_data, webrtc::MediaType::VIDEO, + transport_id, report_block_data, MediaType::VIDEO, video_outbound_rtps, *report, stats_timestamp_with_environment_clock_)); } @@ -2095,7 +2089,7 @@ void RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w_n() { for (const auto& transceiver_proxy : transceivers) { RtpTransceiver* transceiver = transceiver_proxy->internal(); - webrtc::MediaType media_type = transceiver->media_type(); + MediaType media_type = transceiver->media_type(); // Prepare stats entry. The TrackMediaInfoMap will be filled in after the // stats have been fetched on the worker thread. @@ -2113,7 +2107,7 @@ void RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w_n() { stats.mid = channel->mid(); stats.transport_name = std::string(channel->transport_name()); - if (media_type == webrtc::MediaType::AUDIO) { + if (media_type == MediaType::AUDIO) { auto voice_send_channel = channel->voice_media_send_channel(); RTC_DCHECK(voice_send_stats.find(voice_send_channel) == voice_send_stats.end()); @@ -2125,7 +2119,7 @@ void RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w_n() { voice_receive_stats.end()); voice_receive_stats.insert( std::make_pair(voice_receive_channel, VoiceMediaReceiveInfo())); - } else if (media_type == webrtc::MediaType::VIDEO) { + } else if (media_type == MediaType::VIDEO) { auto video_send_channel = channel->video_media_send_channel(); RTC_DCHECK(video_send_stats.find(video_send_channel) == video_send_stats.end()); @@ -2180,14 +2174,14 @@ void RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w_n() { std::optional video_media_info; auto channel = transceiver->channel(); if (channel) { - webrtc::MediaType media_type = transceiver->media_type(); - if (media_type == webrtc::MediaType::AUDIO) { + MediaType media_type = transceiver->media_type(); + if (media_type == MediaType::AUDIO) { auto voice_send_channel = channel->voice_media_send_channel(); auto voice_receive_channel = channel->voice_media_receive_channel(); voice_media_info = VoiceMediaInfo( std::move(voice_send_stats[voice_send_channel]), std::move(voice_receive_stats[voice_receive_channel])); - } else if (media_type == webrtc::MediaType::VIDEO) { + } else if (media_type == MediaType::VIDEO) { auto video_send_channel = channel->video_media_send_channel(); auto video_receive_channel = channel->video_media_receive_channel(); video_media_info = VideoMediaInfo( @@ -2207,7 +2201,7 @@ void RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w_n() { stats.track_media_info_map.Initialize(std::move(voice_media_info), std::move(video_media_info), senders, receivers); - if (transceiver->media_type() == webrtc::MediaType::AUDIO) { + if (transceiver->media_type() == MediaType::AUDIO) { has_audio_receiver |= !receivers.empty(); } } diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc index 7ec09911ef..29216e1d7f 100644 --- a/pc/rtc_stats_collector_unittest.cc +++ b/pc/rtc_stats_collector_unittest.cc @@ -314,30 +314,30 @@ class FakeVideoTrackForStats : public MediaStreamTrack { }; scoped_refptr CreateFakeTrack( - webrtc::MediaType media_type, + MediaType media_type, const std::string& track_id, MediaStreamTrackInterface::TrackState track_state, bool create_fake_audio_processor = false) { - if (media_type == webrtc::MediaType::AUDIO) { + if (media_type == MediaType::AUDIO) { return FakeAudioTrackForStats::Create(track_id, track_state, create_fake_audio_processor); } else { - RTC_DCHECK_EQ(media_type, webrtc::MediaType::VIDEO); + RTC_DCHECK_EQ(media_type, MediaType::VIDEO); return FakeVideoTrackForStats::Create(track_id, track_state, nullptr); } } scoped_refptr CreateMockSender( - webrtc::MediaType media_type, + MediaType media_type, scoped_refptr track, uint32_t ssrc, int attachment_id, std::vector local_stream_ids) { RTC_DCHECK(!track || (track->kind() == MediaStreamTrackInterface::kAudioKind && - media_type == webrtc::MediaType::AUDIO) || + media_type == MediaType::AUDIO) || (track->kind() == MediaStreamTrackInterface::kVideoKind && - media_type == webrtc::MediaType::VIDEO)); + media_type == MediaType::VIDEO)); auto sender = make_ref_counted(); EXPECT_CALL(*sender, track()).WillRepeatedly(Return(track)); EXPECT_CALL(*sender, ssrc()).WillRepeatedly(Return(ssrc)); @@ -373,8 +373,8 @@ scoped_refptr CreateMockReceiver( EXPECT_CALL(*receiver, media_type()) .WillRepeatedly( Return(track->kind() == MediaStreamTrackInterface::kAudioKind - ? webrtc::MediaType::AUDIO - : webrtc::MediaType::VIDEO)); + ? MediaType::AUDIO + : MediaType::VIDEO)); EXPECT_CALL(*receiver, GetParameters()).WillRepeatedly(Invoke([ssrc]() { RtpParameters params; params.encodings.push_back(RtpEncodingParameters()); @@ -427,7 +427,7 @@ class RTCStatsCollectorWrapper { } scoped_refptr SetupLocalTrackAndSender( - webrtc::MediaType media_type, + MediaType media_type, const std::string& track_id, uint32_t ssrc, bool add_stream, @@ -439,7 +439,7 @@ class RTCStatsCollectorWrapper { } scoped_refptr track; - if (media_type == webrtc::MediaType::AUDIO) { + if (media_type == MediaType::AUDIO) { track = CreateFakeTrack(media_type, track_id, MediaStreamTrackInterface::kLive); if (add_stream) { @@ -465,7 +465,7 @@ class RTCStatsCollectorWrapper { } scoped_refptr SetupRemoteTrackAndReceiver( - webrtc::MediaType media_type, + MediaType media_type, const std::string& track_id, const std::string& stream_id, uint32_t ssrc) { @@ -473,7 +473,7 @@ class RTCStatsCollectorWrapper { pc_->mutable_remote_streams()->AddStream(remote_stream); scoped_refptr track; - if (media_type == webrtc::MediaType::AUDIO) { + if (media_type == MediaType::AUDIO) { track = CreateFakeTrack(media_type, track_id, MediaStreamTrackInterface::kLive); remote_stream->AddTrack(scoped_refptr( @@ -528,7 +528,7 @@ class RTCStatsCollectorWrapper { voice_media_info.senders.push_back(voice_sender_info); scoped_refptr rtp_sender = CreateMockSender( - webrtc::MediaType::AUDIO, + MediaType::AUDIO, scoped_refptr(local_audio_track), voice_sender_info.local_stats[0].ssrc, voice_sender_info.local_stats[0].ssrc + 10, local_stream_ids); @@ -566,7 +566,7 @@ class RTCStatsCollectorWrapper { video_media_info.senders.push_back(video_sender_info); video_media_info.aggregated_senders.push_back(video_sender_info); scoped_refptr rtp_sender = CreateMockSender( - webrtc::MediaType::VIDEO, + MediaType::VIDEO, scoped_refptr(local_video_track), video_sender_info.local_stats[0].ssrc, video_sender_info.local_stats[0].ssrc + 10, local_stream_ids); @@ -602,9 +602,9 @@ class RTCStatsCollectorWrapper { scoped_refptr WaitForReport( scoped_refptr callback) { EXPECT_THAT( - WaitUntil( - [&] { return callback->report() != nullptr; }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kGetStatsReportTimeoutMs)}), + WaitUntil([&] { return callback->report() != nullptr; }, + ::testing::IsTrue(), + {.timeout = TimeDelta::Millis(kGetStatsReportTimeoutMs)}), IsRtcOk()); int64_t after = TimeUTCMicros(); for (const RTCStats& stats : *callback->report()) { @@ -719,10 +719,10 @@ class RTCStatsCollectorTest : public ::testing::Test { pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info); // outbound-rtp's sender graph.sender = stats_->SetupLocalTrackAndSender( - webrtc::MediaType::VIDEO, "LocalVideoTrackID", 3, false, 50); + MediaType::VIDEO, "LocalVideoTrackID", 3, false, 50); // inbound-rtp's receiver graph.receiver = stats_->SetupRemoteTrackAndReceiver( - webrtc::MediaType::VIDEO, "RemoteVideoTrackID", "RemoteStreamId", 4); + MediaType::VIDEO, "RemoteVideoTrackID", "RemoteStreamId", 4); // peer-connection graph.peer_connection_id = "P"; // media-source (kind: video) @@ -820,11 +820,10 @@ class RTCStatsCollectorTest : public ::testing::Test { pc_->AddVoiceChannel("VoiceMid", "TransportName", media_info); // outbound-rtp's sender graph.sender = stats_->SetupLocalTrackAndSender( - webrtc::MediaType::AUDIO, "LocalAudioTrackID", kLocalSsrc, false, 50); + MediaType::AUDIO, "LocalAudioTrackID", kLocalSsrc, false, 50); // inbound-rtp's receiver graph.receiver = stats_->SetupRemoteTrackAndReceiver( - webrtc::MediaType::AUDIO, "RemoteAudioTrackID", "RemoteStreamId", - kRemoteSsrc); + MediaType::AUDIO, "RemoteAudioTrackID", "RemoteStreamId", kRemoteSsrc); // peer-connection graph.peer_connection_id = "P"; // media-source (kind: video) @@ -883,9 +882,8 @@ TEST_F(RTCStatsCollectorTest, SingleCallback) { scoped_refptr result; stats_->stats_collector()->GetStatsReport(RTCStatsObtainer::Create(&result)); EXPECT_THAT( - WaitUntil( - [&] { return result != nullptr; }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kGetStatsReportTimeoutMs)}), + WaitUntil([&] { return result != nullptr; }, ::testing::IsTrue(), + {.timeout = TimeDelta::Millis(kGetStatsReportTimeoutMs)}), IsRtcOk()); } @@ -895,19 +893,16 @@ TEST_F(RTCStatsCollectorTest, MultipleCallbacks) { stats_->stats_collector()->GetStatsReport(RTCStatsObtainer::Create(&b)); stats_->stats_collector()->GetStatsReport(RTCStatsObtainer::Create(&c)); EXPECT_THAT( - WaitUntil( - [&] { return a != nullptr; }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kGetStatsReportTimeoutMs)}), + WaitUntil([&] { return a != nullptr; }, ::testing::IsTrue(), + {.timeout = TimeDelta::Millis(kGetStatsReportTimeoutMs)}), IsRtcOk()); EXPECT_THAT( - WaitUntil( - [&] { return b != nullptr; }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kGetStatsReportTimeoutMs)}), + WaitUntil([&] { return b != nullptr; }, ::testing::IsTrue(), + {.timeout = TimeDelta::Millis(kGetStatsReportTimeoutMs)}), IsRtcOk()); EXPECT_THAT( - WaitUntil( - [&] { return c != nullptr; }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kGetStatsReportTimeoutMs)}), + WaitUntil([&] { return c != nullptr; }, ::testing::IsTrue(), + {.timeout = TimeDelta::Millis(kGetStatsReportTimeoutMs)}), IsRtcOk()); EXPECT_EQ(a.get(), b.get()); @@ -938,19 +933,16 @@ TEST_F(RTCStatsCollectorTest, MultipleCallbacksWithInvalidatedCacheInBetween) { fake_clock_.AdvanceTime(TimeDelta::Millis(51)); stats_->stats_collector()->GetStatsReport(RTCStatsObtainer::Create(&c)); EXPECT_THAT( - WaitUntil( - [&] { return a != nullptr; }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kGetStatsReportTimeoutMs)}), + WaitUntil([&] { return a != nullptr; }, ::testing::IsTrue(), + {.timeout = TimeDelta::Millis(kGetStatsReportTimeoutMs)}), IsRtcOk()); EXPECT_THAT( - WaitUntil( - [&] { return b != nullptr; }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kGetStatsReportTimeoutMs)}), + WaitUntil([&] { return b != nullptr; }, ::testing::IsTrue(), + {.timeout = TimeDelta::Millis(kGetStatsReportTimeoutMs)}), IsRtcOk()); EXPECT_THAT( - WaitUntil( - [&] { return c != nullptr; }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kGetStatsReportTimeoutMs)}), + WaitUntil([&] { return c != nullptr; }, ::testing::IsTrue(), + {.timeout = TimeDelta::Millis(kGetStatsReportTimeoutMs)}), IsRtcOk()); EXPECT_EQ(a.get(), b.get()); // The act of doing `AdvanceTime` processes all messages. If this was not the @@ -1077,7 +1069,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCCodecStatsOnlyIfReferenced) { RtpCodecParameters inbound_audio_codec; inbound_audio_codec.payload_type = 1; - inbound_audio_codec.kind = webrtc::MediaType::AUDIO; + inbound_audio_codec.kind = MediaType::AUDIO; inbound_audio_codec.name = "opus"; inbound_audio_codec.clock_rate = 1337; inbound_audio_codec.num_channels = 1; @@ -1087,7 +1079,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCCodecStatsOnlyIfReferenced) { RtpCodecParameters outbound_audio_codec; outbound_audio_codec.payload_type = 2; - outbound_audio_codec.kind = webrtc::MediaType::AUDIO; + outbound_audio_codec.kind = MediaType::AUDIO; outbound_audio_codec.name = "isac"; outbound_audio_codec.clock_rate = 1338; outbound_audio_codec.num_channels = 2; @@ -1099,7 +1091,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCCodecStatsOnlyIfReferenced) { RtpCodecParameters inbound_video_codec; inbound_video_codec.payload_type = 3; - inbound_video_codec.kind = webrtc::MediaType::VIDEO; + inbound_video_codec.kind = MediaType::VIDEO; inbound_video_codec.name = "H264"; inbound_video_codec.clock_rate = 1339; inbound_video_codec.parameters = {{"level-asymmetry-allowed", "1"}, @@ -1110,7 +1102,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCCodecStatsOnlyIfReferenced) { RtpCodecParameters outbound_video_codec; outbound_video_codec.payload_type = 4; - outbound_video_codec.kind = webrtc::MediaType::VIDEO; + outbound_video_codec.kind = MediaType::VIDEO; outbound_video_codec.name = "VP8"; outbound_video_codec.clock_rate = 1340; video_media_info.send_codecs.insert( @@ -1222,14 +1214,14 @@ TEST_F(RTCStatsCollectorTest, CodecStatsAreCollectedPerTransport) { // PT=10 RtpCodecParameters outbound_codec_pt10; outbound_codec_pt10.payload_type = 10; - outbound_codec_pt10.kind = webrtc::MediaType::VIDEO; + outbound_codec_pt10.kind = MediaType::VIDEO; outbound_codec_pt10.name = "VP8"; outbound_codec_pt10.clock_rate = 9000; // PT=11 RtpCodecParameters outbound_codec_pt11; outbound_codec_pt11.payload_type = 11; - outbound_codec_pt11.kind = webrtc::MediaType::VIDEO; + outbound_codec_pt11.kind = MediaType::VIDEO; outbound_codec_pt11.name = "VP8"; outbound_codec_pt11.clock_rate = 9000; @@ -1287,7 +1279,7 @@ TEST_F(RTCStatsCollectorTest, SamePayloadTypeButDifferentFmtpLines) { // PT=111, useinbandfec=0 RtpCodecParameters inbound_codec_pt111_nofec; inbound_codec_pt111_nofec.payload_type = 111; - inbound_codec_pt111_nofec.kind = webrtc::MediaType::AUDIO; + inbound_codec_pt111_nofec.kind = MediaType::AUDIO; inbound_codec_pt111_nofec.name = "opus"; inbound_codec_pt111_nofec.clock_rate = 48000; inbound_codec_pt111_nofec.parameters.insert( @@ -1296,7 +1288,7 @@ TEST_F(RTCStatsCollectorTest, SamePayloadTypeButDifferentFmtpLines) { // PT=111, useinbandfec=1 RtpCodecParameters inbound_codec_pt111_fec; inbound_codec_pt111_fec.payload_type = 111; - inbound_codec_pt111_fec.kind = webrtc::MediaType::AUDIO; + inbound_codec_pt111_fec.kind = MediaType::AUDIO; inbound_codec_pt111_fec.name = "opus"; inbound_codec_pt111_fec.clock_rate = 48000; inbound_codec_pt111_fec.parameters.insert( @@ -1344,7 +1336,7 @@ TEST_F(RTCStatsCollectorTest, SamePayloadTypeButDifferentFmtpLines) { // PT=112, useinbandfec=1 RtpCodecParameters inbound_codec_pt112_fec; inbound_codec_pt112_fec.payload_type = 112; - inbound_codec_pt112_fec.kind = webrtc::MediaType::AUDIO; + inbound_codec_pt112_fec.kind = MediaType::AUDIO; inbound_codec_pt112_fec.name = "opus"; inbound_codec_pt112_fec.clock_rate = 48000; inbound_codec_pt112_fec.parameters.insert( @@ -2186,7 +2178,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Audio) { RtpCodecParameters codec_parameters; codec_parameters.payload_type = 42; - codec_parameters.kind = webrtc::MediaType::AUDIO; + codec_parameters.kind = MediaType::AUDIO; codec_parameters.name = "dummy"; codec_parameters.clock_rate = 0; voice_media_info.receive_codecs.insert( @@ -2194,8 +2186,8 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Audio) { auto voice_media_channels = pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info); - stats_->SetupRemoteTrackAndReceiver( - webrtc::MediaType::AUDIO, "RemoteAudioTrackID", "RemoteStreamId", 1); + stats_->SetupRemoteTrackAndReceiver(MediaType::AUDIO, "RemoteAudioTrackID", + "RemoteStreamId", 1); // Needed for playoutId to be populated. pc_->SetAudioDeviceStats(AudioDeviceModule::Stats()); @@ -2274,8 +2266,8 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Audio_PlayoutId) { voice_media_info.receivers[0].local_stats[0].ssrc = 1; pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info); - stats_->SetupRemoteTrackAndReceiver( - webrtc::MediaType::AUDIO, "RemoteAudioTrackID", "RemoteStreamId", 1); + stats_->SetupRemoteTrackAndReceiver(MediaType::AUDIO, "RemoteAudioTrackID", + "RemoteStreamId", 1); // Needed for playoutId to be populated. pc_->SetAudioDeviceStats(AudioDeviceModule::Stats()); @@ -2361,7 +2353,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Video) { RtpCodecParameters codec_parameters; codec_parameters.payload_type = 42; - codec_parameters.kind = webrtc::MediaType::VIDEO; + codec_parameters.kind = MediaType::VIDEO; codec_parameters.name = "dummy"; codec_parameters.clock_rate = 0; video_media_info.receive_codecs.insert( @@ -2369,8 +2361,8 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Video) { auto video_media_channels = pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info); - stats_->SetupRemoteTrackAndReceiver( - webrtc::MediaType::VIDEO, "RemoteVideoTrackID", "RemoteStreamId", 1); + stats_->SetupRemoteTrackAndReceiver(MediaType::VIDEO, "RemoteVideoTrackID", + "RemoteStreamId", 1); scoped_refptr report = stats_->GetStatsReport(); @@ -2470,8 +2462,8 @@ TEST_F(RTCStatsCollectorTest, CollectRTCAudioPlayoutStats) { pc_->SetAudioDeviceStats(audio_device_stats); pc_->AddVoiceChannel("AudioMid", "TransportName", {}); - stats_->SetupRemoteTrackAndReceiver( - webrtc::MediaType::AUDIO, "RemoteAudioTrackID", "RemoteStreamId", 1); + stats_->SetupRemoteTrackAndReceiver(MediaType::AUDIO, "RemoteAudioTrackID", + "RemoteStreamId", 1); scoped_refptr report = stats_->GetStatsReport(); auto stats_of_track_type = report->GetStatsOfType(); @@ -2514,8 +2506,8 @@ TEST_F(RTCStatsCollectorTest, CollectGoogTimingFrameInfo) { video_media_info.receivers[0].timing_frame_info = timing_frame_info; pc_->AddVideoChannel("Mid0", "Transport0", video_media_info); - stats_->SetupRemoteTrackAndReceiver( - webrtc::MediaType::VIDEO, "RemoteVideoTrackID", "RemoteStreamId", 1); + stats_->SetupRemoteTrackAndReceiver(MediaType::VIDEO, "RemoteVideoTrackID", + "RemoteStreamId", 1); scoped_refptr report = stats_->GetStatsReport(); auto inbound_rtps = report->GetStatsOfType(); @@ -2544,15 +2536,15 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRtpStreamStats_Audio) { RtpCodecParameters codec_parameters; codec_parameters.payload_type = 42; - codec_parameters.kind = webrtc::MediaType::AUDIO; + codec_parameters.kind = MediaType::AUDIO; codec_parameters.name = "dummy"; codec_parameters.clock_rate = 0; voice_media_info.send_codecs.insert( std::make_pair(codec_parameters.payload_type, codec_parameters)); pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info); - stats_->SetupLocalTrackAndSender(webrtc::MediaType::AUDIO, - "LocalAudioTrackID", 1, true, + stats_->SetupLocalTrackAndSender(MediaType::AUDIO, "LocalAudioTrackID", 1, + true, /*attachment_id=*/50); scoped_refptr report = stats_->GetStatsReport(); @@ -2633,7 +2625,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRtpStreamStats_Video) { video_media_info.aggregated_senders.push_back(video_media_info.senders[0]); RtpCodecParameters codec_parameters; codec_parameters.payload_type = 42; - codec_parameters.kind = webrtc::MediaType::AUDIO; + codec_parameters.kind = MediaType::AUDIO; codec_parameters.name = "dummy"; codec_parameters.clock_rate = 0; video_media_info.send_codecs.insert( @@ -2647,8 +2639,8 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRtpStreamStats_Video) { auto video_media_channels = pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info); - stats_->SetupLocalTrackAndSender(webrtc::MediaType::VIDEO, - "LocalVideoTrackID", 1, true, + stats_->SetupLocalTrackAndSender(MediaType::VIDEO, "LocalVideoTrackID", 1, + true, /*attachment_id=*/50); scoped_refptr report = stats_->GetStatsReport(); @@ -2991,7 +2983,7 @@ TEST_F(RTCStatsCollectorTest, CollectNoStreamRTCOutboundRtpStreamStats_Audio) { RtpCodecParameters codec_parameters; codec_parameters.payload_type = 42; - codec_parameters.kind = webrtc::MediaType::AUDIO; + codec_parameters.kind = MediaType::AUDIO; codec_parameters.name = "dummy"; codec_parameters.clock_rate = 0; voice_media_info.send_codecs.insert( @@ -2999,8 +2991,8 @@ TEST_F(RTCStatsCollectorTest, CollectNoStreamRTCOutboundRtpStreamStats_Audio) { // Emulates the case where AddTrack is used without an associated MediaStream pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info); - stats_->SetupLocalTrackAndSender(webrtc::MediaType::AUDIO, - "LocalAudioTrackID", 1, false, + stats_->SetupLocalTrackAndSender(MediaType::AUDIO, "LocalAudioTrackID", 1, + false, /*attachment_id=*/50); scoped_refptr report = stats_->GetStatsReport(); @@ -3045,9 +3037,8 @@ TEST_F(RTCStatsCollectorTest, RTCAudioSourceStatsCollectedForSenderWithTrack) { voice_media_info.senders[0].apm_statistics.echo_return_loss_enhancement = 52.0; pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info); - stats_->SetupLocalTrackAndSender(webrtc::MediaType::AUDIO, - "LocalAudioTrackID", kSsrc, false, - kAttachmentId); + stats_->SetupLocalTrackAndSender(MediaType::AUDIO, "LocalAudioTrackID", kSsrc, + false, kAttachmentId); scoped_refptr report = stats_->GetStatsReport(); @@ -3088,8 +3079,8 @@ TEST_F(RTCStatsCollectorTest, RTCVideoSourceStatsCollectedForSenderWithTrack) { kVideoSourceHeight); auto video_track = FakeVideoTrackForStats::Create( "LocalVideoTrackID", MediaStreamTrackInterface::kLive, video_source); - scoped_refptr sender = CreateMockSender( - webrtc::MediaType::VIDEO, video_track, kSsrc, kAttachmentId, {}); + scoped_refptr sender = + CreateMockSender(MediaType::VIDEO, video_track, kSsrc, kAttachmentId, {}); EXPECT_CALL(*sender, Stop()); EXPECT_CALL(*sender, SetMediaChannel(_)); EXPECT_CALL(*sender, SetSendCodecs(_)); @@ -3134,7 +3125,7 @@ TEST_F(RTCStatsCollectorTest, auto video_track = FakeVideoTrackForStats::Create( "LocalVideoTrackID", MediaStreamTrackInterface::kLive, video_source); scoped_refptr sender = CreateMockSender( - webrtc::MediaType::VIDEO, video_track, kNoSsrc, kAttachmentId, {}); + MediaType::VIDEO, video_track, kNoSsrc, kAttachmentId, {}); EXPECT_CALL(*sender, Stop()); EXPECT_CALL(*sender, SetMediaChannel(_)); EXPECT_CALL(*sender, SetSendCodecs(_)); @@ -3164,8 +3155,8 @@ TEST_F(RTCStatsCollectorTest, auto video_track = FakeVideoTrackForStats::Create( "LocalVideoTrackID", MediaStreamTrackInterface::kLive, /*source=*/nullptr); - scoped_refptr sender = CreateMockSender( - webrtc::MediaType::VIDEO, video_track, kSsrc, kAttachmentId, {}); + scoped_refptr sender = + CreateMockSender(MediaType::VIDEO, video_track, kSsrc, kAttachmentId, {}); EXPECT_CALL(*sender, Stop()); EXPECT_CALL(*sender, SetMediaChannel(_)); EXPECT_CALL(*sender, SetSendCodecs(_)); @@ -3189,7 +3180,7 @@ TEST_F(RTCStatsCollectorTest, voice_media_info.senders[0].local_stats[0].ssrc = kSsrc; pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info); scoped_refptr sender = CreateMockSender( - webrtc::MediaType::AUDIO, /*track=*/nullptr, kSsrc, kAttachmentId, {}); + MediaType::AUDIO, /*track=*/nullptr, kSsrc, kAttachmentId, {}); EXPECT_CALL(*sender, Stop()); EXPECT_CALL(*sender, SetMediaChannel(_)); EXPECT_CALL(*sender, SetSendCodecs(_)); @@ -3202,18 +3193,18 @@ TEST_F(RTCStatsCollectorTest, // Parameterized tests on webrtc::MediaType (audio or video). class RTCStatsCollectorTestWithParamKind : public RTCStatsCollectorTest, - public ::testing::WithParamInterface { + public ::testing::WithParamInterface { public: RTCStatsCollectorTestWithParamKind() : media_type_(GetParam()) { - RTC_DCHECK(media_type_ == webrtc::MediaType::AUDIO || - media_type_ == webrtc::MediaType::VIDEO); + RTC_DCHECK(media_type_ == MediaType::AUDIO || + media_type_ == MediaType::VIDEO); } std::string MediaTypeCharStr() const { switch (media_type_) { - case webrtc::MediaType::AUDIO: + case MediaType::AUDIO: return "A"; - case webrtc::MediaType::VIDEO: + case MediaType::VIDEO: return "V"; default: RTC_DCHECK_NOTREACHED(); @@ -3223,9 +3214,9 @@ class RTCStatsCollectorTestWithParamKind std::string MediaTypeKind() const { switch (media_type_) { - case webrtc::MediaType::AUDIO: + case MediaType::AUDIO: return "audio"; - case webrtc::MediaType::VIDEO: + case MediaType::VIDEO: return "video"; default: RTC_DCHECK_NOTREACHED(); @@ -3240,7 +3231,7 @@ class RTCStatsCollectorTestWithParamKind const std::vector& report_block_datas, std::optional codec) { switch (media_type_) { - case webrtc::MediaType::AUDIO: { + case MediaType::AUDIO: { VoiceMediaInfo voice_media_info; for (const auto& report_block_data : report_block_datas) { VoiceSenderInfo sender; @@ -3257,7 +3248,7 @@ class RTCStatsCollectorTestWithParamKind pc_->AddVoiceChannel("mid", transport_name, voice_media_info); return; } - case webrtc::MediaType::VIDEO: { + case MediaType::VIDEO: { VideoMediaInfo video_media_info; for (const auto& report_block_data : report_block_datas) { VideoSenderInfo sender; @@ -3275,14 +3266,14 @@ class RTCStatsCollectorTestWithParamKind pc_->AddVideoChannel("mid", transport_name, video_media_info); return; } - case webrtc::MediaType::DATA: + case MediaType::DATA: default: RTC_DCHECK_NOTREACHED(); } } protected: - webrtc::MediaType media_type_; + MediaType media_type_; }; // Verifies RTCRemoteInboundRtpStreamStats members that don't require @@ -3496,8 +3487,8 @@ TEST_P(RTCStatsCollectorTestWithParamKind, INSTANTIATE_TEST_SUITE_P(All, RTCStatsCollectorTestWithParamKind, - ::testing::Values(webrtc::MediaType::AUDIO, // "/0" - webrtc::MediaType::VIDEO)); // "/1" + ::testing::Values(MediaType::AUDIO, // "/0" + MediaType::VIDEO)); // "/1" // Checks that no remote outbound stats are collected if not available in // `VoiceMediaInfo`. @@ -3548,7 +3539,7 @@ TEST_F(RTCStatsCollectorTest, pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info); scoped_refptr sender = CreateMockSender( - webrtc::MediaType::VIDEO, /*track=*/nullptr, kSsrc, kAttachmentId, {}); + MediaType::VIDEO, /*track=*/nullptr, kSsrc, kAttachmentId, {}); EXPECT_CALL(*sender, Stop()); EXPECT_CALL(*sender, SetMediaChannel(_)); EXPECT_CALL(*sender, SetSendCodecs(_)); @@ -3566,10 +3557,9 @@ TEST_F(RTCStatsCollectorTest, CollectEchoReturnLossFromTrackAudioProcessor) { pc_->mutable_local_streams()->AddStream(local_stream); // Local audio track - scoped_refptr local_audio_track = - CreateFakeTrack(webrtc::MediaType::AUDIO, "LocalAudioTrackID", - MediaStreamTrackInterface::kEnded, - /*create_fake_audio_processor=*/true); + scoped_refptr local_audio_track = CreateFakeTrack( + MediaType::AUDIO, "LocalAudioTrackID", MediaStreamTrackInterface::kEnded, + /*create_fake_audio_processor=*/true); local_stream->AddTrack(scoped_refptr( static_cast(local_audio_track.get()))); @@ -3668,9 +3658,9 @@ TEST_F(RTCStatsCollectorTest, GetStatsWithNullReceiverSelector) { // To simulate this case we create a mock sender with SSRC=0. TEST_F(RTCStatsCollectorTest, RtpIsMissingWhileSsrcIsZero) { scoped_refptr track = CreateFakeTrack( - webrtc::MediaType::AUDIO, "audioTrack", MediaStreamTrackInterface::kLive); + MediaType::AUDIO, "audioTrack", MediaStreamTrackInterface::kLive); scoped_refptr sender = - CreateMockSender(webrtc::MediaType::AUDIO, track, 0, 49, {}); + CreateMockSender(MediaType::AUDIO, track, 0, 49, {}); EXPECT_CALL(*sender, Stop()); EXPECT_CALL(*sender, SetSendCodecs(_)); pc_->AddSender(sender); @@ -3685,9 +3675,9 @@ TEST_F(RTCStatsCollectorTest, RtpIsMissingWhileSsrcIsZero) { // `voice_sender_info` stats exist yet. TEST_F(RTCStatsCollectorTest, DoNotCrashIfSsrcIsKnownButInfosAreStillMissing) { scoped_refptr track = CreateFakeTrack( - webrtc::MediaType::AUDIO, "audioTrack", MediaStreamTrackInterface::kLive); + MediaType::AUDIO, "audioTrack", MediaStreamTrackInterface::kLive); scoped_refptr sender = - CreateMockSender(webrtc::MediaType::AUDIO, track, 4711, 49, {}); + CreateMockSender(MediaType::AUDIO, track, 4711, 49, {}); EXPECT_CALL(*sender, Stop()); EXPECT_CALL(*sender, SetSendCodecs(_)); pc_->AddSender(sender); @@ -3724,14 +3714,12 @@ TEST_F(RTCStatsCollectorTest, DoNotCrashWhenGetStatsCalledDuringCallback) { stats_->stats_collector()->GetStatsReport(callback1); stats_->stats_collector()->GetStatsReport(callback2); EXPECT_THAT( - WaitUntil( - [&] { return callback1->called(); }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kGetStatsReportTimeoutMs)}), + WaitUntil([&] { return callback1->called(); }, ::testing::IsTrue(), + {.timeout = TimeDelta::Millis(kGetStatsReportTimeoutMs)}), IsRtcOk()); EXPECT_THAT( - WaitUntil( - [&] { return callback2->called(); }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kGetStatsReportTimeoutMs)}), + WaitUntil([&] { return callback2->called(); }, ::testing::IsTrue(), + {.timeout = TimeDelta::Millis(kGetStatsReportTimeoutMs)}), IsRtcOk()); } @@ -3785,9 +3773,8 @@ class FakeRTCStatsCollector : public RTCStatsCollector, void VerifyThreadUsageAndResultsMerging() { GetStatsReport(scoped_refptr(this)); EXPECT_THAT( - WaitUntil( - [&] { return HasVerifiedResults(); }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kGetStatsReportTimeoutMs)}), + WaitUntil([&] { return HasVerifiedResults(); }, ::testing::IsTrue(), + {.timeout = TimeDelta::Millis(kGetStatsReportTimeoutMs)}), IsRtcOk()); } diff --git a/pc/rtc_stats_integrationtest.cc b/pc/rtc_stats_integrationtest.cc index 7562f803a1..2e3b69dc5f 100644 --- a/pc/rtc_stats_integrationtest.cc +++ b/pc/rtc_stats_integrationtest.cc @@ -125,11 +125,10 @@ class RTCStatsIntegrationTest : public ::testing::Test { PeerConnectionInterface* pc) { scoped_refptr stats_obtainer = RTCStatsObtainer::Create(); pc->GetStats(stats_obtainer.get()); - EXPECT_THAT( - WaitUntil([&] { return stats_obtainer->report() != nullptr; }, - ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kGetStatsTimeoutMs)}), - IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return stats_obtainer->report() != nullptr; }, + ::testing::IsTrue(), + {.timeout = TimeDelta::Millis(kGetStatsTimeoutMs)}), + IsRtcOk()); return stats_obtainer->report(); } @@ -139,11 +138,10 @@ class RTCStatsIntegrationTest : public ::testing::Test { scoped_refptr selector) { scoped_refptr stats_obtainer = RTCStatsObtainer::Create(); pc->GetStats(selector, stats_obtainer); - EXPECT_THAT( - WaitUntil([&] { return stats_obtainer->report() != nullptr; }, - ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kGetStatsTimeoutMs)}), - IsRtcOk()); + EXPECT_THAT(WaitUntil([&] { return stats_obtainer->report() != nullptr; }, + ::testing::IsTrue(), + {.timeout = TimeDelta::Millis(kGetStatsTimeoutMs)}), + IsRtcOk()); return stats_obtainer->report(); } @@ -1040,7 +1038,7 @@ TEST_F(RTCStatsIntegrationTest, GetStatsFromCallee) { EXPECT_THAT( WaitUntil([&] { return GetStatsReportAndReturnTrueIfRttIsDefined(); }, ::testing::IsTrue(), - {.timeout = webrtc::TimeDelta::Millis(kMaxWaitMs)}), + {.timeout = TimeDelta::Millis(kMaxWaitMs)}), IsRtcOk()); RTCStatsReportVerifier(report.get()).VerifyReport({}); } diff --git a/pc/rtcp_mux_filter.cc b/pc/rtcp_mux_filter.cc index b444658a9b..ba48a44c03 100644 --- a/pc/rtcp_mux_filter.cc +++ b/pc/rtcp_mux_filter.cc @@ -45,7 +45,7 @@ bool RtcpMuxFilter::SetOffer(bool offer_enable, ContentSource src) { } offer_enable_ = offer_enable; - state_ = (src == webrtc::CS_LOCAL) ? ST_SENTOFFER : ST_RECEIVEDOFFER; + state_ = (src == CS_LOCAL) ? ST_SENTOFFER : ST_RECEIVEDOFFER; return true; } @@ -63,7 +63,7 @@ bool RtcpMuxFilter::SetProvisionalAnswer(bool answer_enable, if (offer_enable_) { if (answer_enable) { - if (src == webrtc::CS_REMOTE) + if (src == CS_REMOTE) state_ = ST_RECEIVEDPRANSWER; else // CS_LOCAL state_ = ST_SENTPRANSWER; @@ -71,7 +71,7 @@ bool RtcpMuxFilter::SetProvisionalAnswer(bool answer_enable, // The provisional answer doesn't want to use RTCP mux. // Go back to the original state after the offer was set and wait for next // provisional or final answer. - if (src == webrtc::CS_REMOTE) + if (src == CS_REMOTE) state_ = ST_SENTOFFER; else // CS_LOCAL state_ = ST_RECEIVEDOFFER; @@ -112,15 +112,15 @@ bool RtcpMuxFilter::SetAnswer(bool answer_enable, ContentSource src) { bool RtcpMuxFilter::ExpectOffer(bool offer_enable, ContentSource source) { return ((state_ == ST_INIT) || (state_ == ST_ACTIVE && offer_enable == offer_enable_) || - (state_ == ST_SENTOFFER && source == webrtc::CS_LOCAL) || - (state_ == ST_RECEIVEDOFFER && source == webrtc::CS_REMOTE)); + (state_ == ST_SENTOFFER && source == CS_LOCAL) || + (state_ == ST_RECEIVEDOFFER && source == CS_REMOTE)); } bool RtcpMuxFilter::ExpectAnswer(ContentSource source) { - return ((state_ == ST_SENTOFFER && source == webrtc::CS_REMOTE) || - (state_ == ST_RECEIVEDOFFER && source == webrtc::CS_LOCAL) || - (state_ == ST_SENTPRANSWER && source == webrtc::CS_LOCAL) || - (state_ == ST_RECEIVEDPRANSWER && source == webrtc::CS_REMOTE)); + return ((state_ == ST_SENTOFFER && source == CS_REMOTE) || + (state_ == ST_RECEIVEDOFFER && source == CS_LOCAL) || + (state_ == ST_SENTPRANSWER && source == CS_LOCAL) || + (state_ == ST_RECEIVEDPRANSWER && source == CS_REMOTE)); } } // namespace webrtc diff --git a/pc/rtp_parameters_conversion.cc b/pc/rtp_parameters_conversion.cc index 75a69bcaa9..63d50d23cc 100644 --- a/pc/rtp_parameters_conversion.cc +++ b/pc/rtp_parameters_conversion.cc @@ -78,9 +78,8 @@ std::optional ToRtcpFeedback( RtpCodecCapability ToRtpCodecCapability(const Codec& cricket_codec) { RtpCodecCapability codec; codec.name = cricket_codec.name; - codec.kind = cricket_codec.type == Codec::Type::kAudio - ? webrtc::MediaType::AUDIO - : webrtc::MediaType::VIDEO; + codec.kind = cricket_codec.type == Codec::Type::kAudio ? MediaType::AUDIO + : MediaType::VIDEO; codec.clock_rate.emplace(cricket_codec.clockrate); codec.preferred_payload_type.emplace(cricket_codec.id); for (const FeedbackParam& cricket_feedback : diff --git a/pc/rtp_parameters_conversion_unittest.cc b/pc/rtp_parameters_conversion_unittest.cc index 407c535173..74f370e30b 100644 --- a/pc/rtp_parameters_conversion_unittest.cc +++ b/pc/rtp_parameters_conversion_unittest.cc @@ -86,7 +86,7 @@ TEST(RtpParametersConversionTest, ToAudioRtpCodecCapability) { RtpCodecCapability codec = ToRtpCodecCapability(cricket_codec); EXPECT_EQ("foo", codec.name); - EXPECT_EQ(webrtc::MediaType::AUDIO, codec.kind); + EXPECT_EQ(MediaType::AUDIO, codec.kind); EXPECT_EQ(50, codec.preferred_payload_type); EXPECT_EQ(22222, codec.clock_rate); EXPECT_EQ(4, codec.num_channels); @@ -108,7 +108,7 @@ TEST(RtpParametersConversionTest, ToVideoRtpCodecCapability) { RtpCodecCapability codec = ToRtpCodecCapability(cricket_codec); EXPECT_EQ("VID", codec.name); - EXPECT_EQ(webrtc::MediaType::VIDEO, codec.kind); + EXPECT_EQ(MediaType::VIDEO, codec.kind); EXPECT_EQ(101, codec.preferred_payload_type); EXPECT_EQ(80000, codec.clock_rate); ASSERT_EQ(2u, codec.parameters.size()); diff --git a/pc/rtp_sender.cc b/pc/rtp_sender.cc index ef956548b5..0645133275 100644 --- a/pc/rtp_sender.cc +++ b/pc/rtp_sender.cc @@ -264,8 +264,8 @@ void RtpSenderBase::SetParametersInternal(const RtpParameters& parameters, RTCError error( RTCErrorType::UNSUPPORTED_PARAMETER, "Attempted to set an unimplemented parameter of RtpParameters."); - RTC_LOG(LS_ERROR) << error.message() << " (" - << ::webrtc::ToString(error.type()) << ")"; + RTC_LOG(LS_ERROR) << error.message() << " (" << ToString(error.type()) + << ")"; InvokeSetParametersCallback(callback, error); return; } @@ -678,7 +678,7 @@ void LocalAudioSinkAdapter::SetSink(AudioSource::Sink* sink) { } scoped_refptr AudioRtpSender::Create( - const webrtc::Environment& env, + const Environment& env, Thread* worker_thread, const std::string& id, LegacyStatsCollectorInterface* stats, @@ -687,7 +687,7 @@ scoped_refptr AudioRtpSender::Create( set_streams_observer); } -AudioRtpSender::AudioRtpSender(const webrtc::Environment& env, +AudioRtpSender::AudioRtpSender(const Environment& env, Thread* worker_thread, const std::string& id, LegacyStatsCollectorInterface* legacy_stats, diff --git a/pc/rtp_transceiver.cc b/pc/rtp_transceiver.cc index 6dae6e6146..dadc9de064 100644 --- a/pc/rtp_transceiver.cc +++ b/pc/rtp_transceiver.cc @@ -114,7 +114,7 @@ TaskQueueBase* GetCurrentTaskQueueOrThread() { } // namespace -RtpTransceiver::RtpTransceiver(webrtc::MediaType media_type, +RtpTransceiver::RtpTransceiver(MediaType media_type, ConnectionContext* context, CodecLookupHelper* codec_lookup_helper) : thread_(GetCurrentTaskQueueOrThread()), @@ -122,8 +122,7 @@ RtpTransceiver::RtpTransceiver(webrtc::MediaType media_type, media_type_(media_type), context_(context), codec_lookup_helper_(codec_lookup_helper) { - RTC_DCHECK(media_type == webrtc::MediaType::AUDIO || - media_type == webrtc::MediaType::VIDEO); + RTC_DCHECK(media_type == MediaType::AUDIO || media_type == MediaType::VIDEO); RTC_DCHECK(context_); RTC_DCHECK(codec_lookup_helper_); } @@ -144,11 +143,11 @@ RtpTransceiver::RtpTransceiver( std::move(header_extensions_to_negotiate)), on_negotiation_needed_(std::move(on_negotiation_needed)) { RTC_DCHECK(context_); - RTC_DCHECK(media_type_ == webrtc::MediaType::AUDIO || - media_type_ == webrtc::MediaType::VIDEO); + RTC_DCHECK(media_type_ == MediaType::AUDIO || + media_type_ == MediaType::VIDEO); RTC_DCHECK_EQ(sender->media_type(), receiver->media_type()); sender->internal()->SetSendCodecs( - sender->media_type() == webrtc::MediaType::VIDEO + sender->media_type() == MediaType::VIDEO ? codec_vendor().video_send_codecs().codecs() : codec_vendor().audio_send_codecs().codecs()); senders_.push_back(sender); @@ -170,7 +169,7 @@ RtpTransceiver::RtpTransceiver( header_extensions_to_negotiate_, [](const RtpHeaderExtensionCapability& ext) { return ext.uri == RtpExtension::kGenericFrameDescriptorUri00 && - ext.direction != webrtc::RtpTransceiverDirection::kStopped; + ext.direction != RtpTransceiverDirection::kStopped; }) != header_extensions_to_negotiate_.end(); if (!uses_gfd) { for (RtpHeaderExtensionCapability& ext : @@ -216,7 +215,7 @@ RTCError RtpTransceiver::CreateChannel( } std::unique_ptr new_channel; - if (media_type() == webrtc::MediaType::AUDIO) { + if (media_type() == MediaType::AUDIO) { // TODO(bugs.webrtc.org/11992): CreateVideoChannel internally switches to // the worker thread. We shouldn't be using the `call_ptr_` hack here but // simply be on the worker thread and use `call_` (update upstream code). @@ -252,7 +251,7 @@ RTCError RtpTransceiver::CreateChannel( context()->ssrc_generator()); }); } else { - RTC_DCHECK_EQ(webrtc::MediaType::VIDEO, media_type()); + RTC_DCHECK_EQ(MediaType::VIDEO, media_type()); // TODO(bugs.webrtc.org/11992): CreateVideoChannel internally switches to // the worker thread. We shouldn't be using the `call_ptr_` hack here but @@ -403,7 +402,7 @@ void RtpTransceiver::AddSender( RTC_DCHECK(!absl::c_linear_search(senders_, sender)); std::vector send_codecs = - media_type() == webrtc::MediaType::VIDEO + media_type() == MediaType::VIDEO ? codec_vendor().video_send_codecs().codecs() : codec_vendor().audio_send_codecs().codecs(); sender->internal()->SetSendCodecs(send_codecs); @@ -468,7 +467,7 @@ scoped_refptr RtpTransceiver::receiver_internal() const { return scoped_refptr(receivers_[0]->internal()); } -webrtc::MediaType RtpTransceiver::media_type() const { +MediaType RtpTransceiver::media_type() const { return media_type_; } @@ -678,10 +677,10 @@ RTCError RtpTransceiver::UpdateCodecPreferencesCaches( const std::vector& codecs) { // Get codec capabilities from media engine. std::vector send_codecs, recv_codecs; - if (media_type_ == webrtc::MediaType::AUDIO) { + if (media_type_ == MediaType::AUDIO) { send_codecs = codec_vendor().audio_send_codecs().codecs(); recv_codecs = codec_vendor().audio_recv_codecs().codecs(); - } else if (media_type_ == webrtc::MediaType::VIDEO) { + } else if (media_type_ == MediaType::VIDEO) { send_codecs = codec_vendor().video_send_codecs().codecs(); recv_codecs = codec_vendor().video_recv_codecs().codecs(); } diff --git a/pc/rtp_transceiver_unittest.cc b/pc/rtp_transceiver_unittest.cc index c9b42be8e0..0bb9c5824b 100644 --- a/pc/rtp_transceiver_unittest.cc +++ b/pc/rtp_transceiver_unittest.cc @@ -106,10 +106,9 @@ class RtpTransceiverTest : public testing::Test { TEST_F(RtpTransceiverTest, CannotSetChannelOnStoppedTransceiver) { const std::string content_name("my_mid"); auto transceiver = make_ref_counted( - webrtc::MediaType::AUDIO, context(), codec_lookup_helper()); + MediaType::AUDIO, context(), codec_lookup_helper()); auto channel1 = std::make_unique>(); - EXPECT_CALL(*channel1, media_type()) - .WillRepeatedly(Return(webrtc::MediaType::AUDIO)); + EXPECT_CALL(*channel1, media_type()).WillRepeatedly(Return(MediaType::AUDIO)); EXPECT_CALL(*channel1, mid()).WillRepeatedly(ReturnRef(content_name)); EXPECT_CALL(*channel1, SetFirstPacketReceivedCallback(_)); EXPECT_CALL(*channel1, SetRtpTransport(_)).WillRepeatedly(Return(true)); @@ -125,8 +124,7 @@ TEST_F(RtpTransceiverTest, CannotSetChannelOnStoppedTransceiver) { EXPECT_EQ(channel1_ptr, transceiver->channel()); auto channel2 = std::make_unique>(); - EXPECT_CALL(*channel2, media_type()) - .WillRepeatedly(Return(webrtc::MediaType::AUDIO)); + EXPECT_CALL(*channel2, media_type()).WillRepeatedly(Return(MediaType::AUDIO)); // Clear the current channel - required to allow SetChannel() EXPECT_CALL(*channel1_ptr, SetFirstPacketReceivedCallback(_)); @@ -142,10 +140,9 @@ TEST_F(RtpTransceiverTest, CannotSetChannelOnStoppedTransceiver) { TEST_F(RtpTransceiverTest, CanUnsetChannelOnStoppedTransceiver) { const std::string content_name("my_mid"); auto transceiver = make_ref_counted( - webrtc::MediaType::VIDEO, context(), codec_lookup_helper()); + MediaType::VIDEO, context(), codec_lookup_helper()); auto channel = std::make_unique>(); - EXPECT_CALL(*channel, media_type()) - .WillRepeatedly(Return(webrtc::MediaType::VIDEO)); + EXPECT_CALL(*channel, media_type()).WillRepeatedly(Return(MediaType::VIDEO)); EXPECT_CALL(*channel, mid()).WillRepeatedly(ReturnRef(content_name)); EXPECT_CALL(*channel, SetFirstPacketReceivedCallback(_)) .WillRepeatedly(testing::Return()); @@ -170,15 +167,14 @@ TEST_F(RtpTransceiverTest, CanUnsetChannelOnStoppedTransceiver) { class RtpTransceiverUnifiedPlanTest : public RtpTransceiverTest { public: static scoped_refptr MockReceiver( - webrtc::MediaType media_type) { + MediaType media_type) { auto receiver = make_ref_counted>(); EXPECT_CALL(*receiver.get(), media_type()) .WillRepeatedly(Return(media_type)); return receiver; } - static scoped_refptr MockSender( - webrtc::MediaType media_type) { + static scoped_refptr MockSender(MediaType media_type) { auto sender = make_ref_counted>(); EXPECT_CALL(*sender.get(), media_type()).WillRepeatedly(Return(media_type)); return sender; @@ -204,9 +200,8 @@ class RtpTransceiverUnifiedPlanTest : public RtpTransceiverTest { // Basic tests for Stop() TEST_F(RtpTransceiverUnifiedPlanTest, StopSetsDirection) { scoped_refptr receiver = - MockReceiver(webrtc::MediaType::AUDIO); - scoped_refptr sender = - MockSender(webrtc::MediaType::AUDIO); + MockReceiver(MediaType::AUDIO); + scoped_refptr sender = MockSender(MediaType::AUDIO); scoped_refptr transceiver = CreateTransceiver(sender, receiver); @@ -231,9 +226,8 @@ class RtpTransceiverFilteredCodecPreferencesTest : public RtpTransceiverUnifiedPlanTest { public: RtpTransceiverFilteredCodecPreferencesTest() - : transceiver_( - CreateTransceiver(MockSender(webrtc::MediaType::VIDEO), - MockReceiver(webrtc::MediaType::VIDEO))) {} + : transceiver_(CreateTransceiver(MockSender(MediaType::VIDEO), + MockReceiver(MediaType::VIDEO))) {} struct H264CodecCapabilities { Codec cricket_sendrecv_codec; @@ -251,8 +245,8 @@ class RtpTransceiverFilteredCodecPreferencesTest // at transceiver create time. void RecreateTransceiver() { fake_codec_lookup_helper()->Reset(); - transceiver_ = CreateTransceiver(MockSender(webrtc::MediaType::VIDEO), - MockReceiver(webrtc::MediaType::VIDEO)); + transceiver_ = CreateTransceiver(MockSender(MediaType::VIDEO), + MockReceiver(MediaType::VIDEO)); } // For H264, the profile and level IDs are entangled. This function uses @@ -596,9 +590,8 @@ class RtpTransceiverTestForHeaderExtensions } scoped_refptr receiver_ = - MockReceiver(webrtc::MediaType::AUDIO); - scoped_refptr sender_ = - MockSender(webrtc::MediaType::AUDIO); + MockReceiver(MediaType::AUDIO); + scoped_refptr sender_ = MockSender(MediaType::AUDIO); std::vector extensions_; scoped_refptr transceiver_; @@ -744,7 +737,7 @@ TEST_F(RtpTransceiverTestForHeaderExtensions, auto mock_channel_ptr = mock_channel.get(); EXPECT_CALL(*mock_channel, SetFirstPacketReceivedCallback(_)); EXPECT_CALL(*mock_channel, media_type()) - .WillRepeatedly(Return(webrtc::MediaType::AUDIO)); + .WillRepeatedly(Return(MediaType::AUDIO)); EXPECT_CALL(*mock_channel, voice_media_send_channel()) .WillRepeatedly(Return(nullptr)); EXPECT_CALL(*mock_channel, mid()).WillRepeatedly(ReturnRef(content_name)); @@ -777,7 +770,7 @@ TEST_F(RtpTransceiverTestForHeaderExtensions, ReturnsNegotiatedHdrExts) { auto mock_channel_ptr = mock_channel.get(); EXPECT_CALL(*mock_channel, SetFirstPacketReceivedCallback(_)); EXPECT_CALL(*mock_channel, media_type()) - .WillRepeatedly(Return(webrtc::MediaType::AUDIO)); + .WillRepeatedly(Return(MediaType::AUDIO)); EXPECT_CALL(*mock_channel, voice_media_send_channel()) .WillRepeatedly(Return(nullptr)); EXPECT_CALL(*mock_channel, mid()).WillRepeatedly(ReturnRef(content_name)); @@ -860,7 +853,7 @@ TEST_F(RtpTransceiverTestForHeaderExtensions, Thread::Current(), Thread::Current(), receiver_), context(), codec_lookup_helper(), extensions, /* on_negotiation_needed= */ [] {}); - std::vector header_extensions = + std::vector header_extensions = transceiver->GetHeaderExtensionsToNegotiate(); ASSERT_EQ(header_extensions.size(), 2u); EXPECT_EQ(header_extensions[0].uri, RtpExtension::kDependencyDescriptorUri); @@ -894,7 +887,7 @@ TEST_F(RtpTransceiverTestForHeaderExtensions, RtpTransceiverDirection::kSendRecv); // SVC, a single encoding with a scalabilityMode other than L1T1. - webrtc::RtpParameters svc_parameters; + RtpParameters svc_parameters; svc_parameters.encodings.resize(1); svc_parameters.encodings[0].scalability_mode = "L3T3"; @@ -908,7 +901,7 @@ TEST_F(RtpTransceiverTestForHeaderExtensions, Thread::Current(), Thread::Current(), receiver_), context(), codec_lookup_helper(), extensions, /* on_negotiation_needed= */ [] {}); - std::vector svc_extensions = + std::vector svc_extensions = svc_transceiver->GetHeaderExtensionsToNegotiate(); ASSERT_EQ(svc_extensions.size(), 2u); EXPECT_EQ(svc_extensions[0].uri, RtpExtension::kDependencyDescriptorUri); diff --git a/pc/rtp_transmission_manager.cc b/pc/rtp_transmission_manager.cc index 8c6d2a8fa8..1779ad422c 100644 --- a/pc/rtp_transmission_manager.cc +++ b/pc/rtp_transmission_manager.cc @@ -172,10 +172,9 @@ RtpTransmissionManager::AddTrackPlanB( if (adjusted_stream_ids.empty()) { adjusted_stream_ids.push_back(CreateRandomUuid()); } - webrtc::MediaType media_type = - (track->kind() == MediaStreamTrackInterface::kAudioKind - ? webrtc::MediaType::AUDIO - : webrtc::MediaType::VIDEO); + MediaType media_type = (track->kind() == MediaStreamTrackInterface::kAudioKind + ? MediaType::AUDIO + : MediaType::VIDEO); auto new_sender = CreateSender( media_type, track->id(), track, adjusted_stream_ids, init_send_encodings @@ -213,7 +212,7 @@ RtpTransmissionManager::AddTrackUnifiedPlan( FindFirstTransceiverForAddedTrack(track, init_send_encodings); if (transceiver) { RTC_LOG(LS_INFO) << "Reusing an existing " - << webrtc::MediaTypeToString(transceiver->media_type()) + << MediaTypeToString(transceiver->media_type()) << " transceiver for AddTrack."; if (transceiver->stopping()) { LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, @@ -231,11 +230,11 @@ RtpTransmissionManager::AddTrackUnifiedPlan( transceiver->internal()->sender_internal()->set_stream_ids(stream_ids); transceiver->internal()->set_reused_for_addtrack(true); } else { - webrtc::MediaType media_type = + MediaType media_type = (track->kind() == MediaStreamTrackInterface::kAudioKind - ? webrtc::MediaType::AUDIO - : webrtc::MediaType::VIDEO); - RTC_LOG(LS_INFO) << "Adding " << webrtc::MediaTypeToString(media_type) + ? MediaType::AUDIO + : MediaType::VIDEO); + RTC_LOG(LS_INFO) << "Adding " << MediaTypeToString(media_type) << " transceiver in response to a call to AddTrack."; std::string sender_id = track->id(); // Avoid creating a sender with an existing ID by generating a random ID. @@ -259,14 +258,14 @@ RtpTransmissionManager::AddTrackUnifiedPlan( scoped_refptr> RtpTransmissionManager::CreateSender( - webrtc::MediaType media_type, + MediaType media_type, const std::string& id, scoped_refptr track, const std::vector& stream_ids, const std::vector& send_encodings) { RTC_DCHECK_RUN_ON(signaling_thread()); scoped_refptr> sender; - if (media_type == webrtc::MediaType::AUDIO) { + if (media_type == MediaType::AUDIO) { RTC_DCHECK(!track || (track->kind() == MediaStreamTrackInterface::kAudioKind)); sender = RtpSenderProxyWithInternal::Create( @@ -274,7 +273,7 @@ RtpTransmissionManager::CreateSender( AudioRtpSender::Create(env_, worker_thread(), id, legacy_stats_, this)); NoteUsageEvent(UsageEvent::AUDIO_ADDED); } else { - RTC_DCHECK_EQ(media_type, webrtc::MediaType::VIDEO); + RTC_DCHECK_EQ(media_type, MediaType::VIDEO); RTC_DCHECK(!track || (track->kind() == MediaStreamTrackInterface::kVideoKind)); sender = RtpSenderProxyWithInternal::Create( @@ -290,11 +289,11 @@ RtpTransmissionManager::CreateSender( } scoped_refptr> -RtpTransmissionManager::CreateReceiver(webrtc::MediaType media_type, +RtpTransmissionManager::CreateReceiver(MediaType media_type, const std::string& receiver_id) { RTC_DCHECK_RUN_ON(signaling_thread()); scoped_refptr> receiver; - if (media_type == webrtc::MediaType::AUDIO) { + if (media_type == MediaType::AUDIO) { receiver = RtpReceiverProxyWithInternal::Create( signaling_thread(), worker_thread(), make_ref_counted(worker_thread(), receiver_id, @@ -302,7 +301,7 @@ RtpTransmissionManager::CreateReceiver(webrtc::MediaType media_type, IsUnifiedPlan())); NoteUsageEvent(UsageEvent::AUDIO_ADDED); } else { - RTC_DCHECK_EQ(media_type, webrtc::MediaType::VIDEO); + RTC_DCHECK_EQ(media_type, MediaType::VIDEO); receiver = RtpReceiverProxyWithInternal::Create( signaling_thread(), worker_thread(), make_ref_counted(worker_thread(), receiver_id, @@ -326,7 +325,7 @@ RtpTransmissionManager::CreateAndAddTransceiver( signaling_thread(), make_ref_counted( sender, receiver, context_, codec_lookup_helper_, - sender->media_type() == webrtc::MediaType::AUDIO + sender->media_type() == MediaType::AUDIO ? media_engine()->voice().GetRtpHeaderExtensions() : media_engine()->video().GetRtpHeaderExtensions(), [this_weak_ptr = weak_ptr_factory_.GetWeakPtr()]() { @@ -349,7 +348,7 @@ RtpTransmissionManager::FindFirstTransceiverForAddedTrack( } for (auto transceiver : transceivers()->List()) { if (!transceiver->sender()->track() && - webrtc::MediaTypeToString(transceiver->media_type()) == track->kind() && + MediaTypeToString(transceiver->media_type()) == track->kind() && !transceiver->internal()->has_ever_been_used_to_send() && !transceiver->stopped()) { return transceiver; @@ -396,7 +395,7 @@ RtpTransmissionManager::GetAudioTransceiver() const { // audio/video transceiver. RTC_DCHECK(!IsUnifiedPlan()); for (auto transceiver : transceivers_.List()) { - if (transceiver->media_type() == webrtc::MediaType::AUDIO) { + if (transceiver->media_type() == MediaType::AUDIO) { return transceiver; } } @@ -411,7 +410,7 @@ RtpTransmissionManager::GetVideoTransceiver() const { // audio/video transceiver. RTC_DCHECK(!IsUnifiedPlan()); for (auto transceiver : transceivers_.List()) { - if (transceiver->media_type() == webrtc::MediaType::VIDEO) { + if (transceiver->media_type() == MediaType::VIDEO) { return transceiver; } } @@ -433,7 +432,7 @@ void RtpTransmissionManager::AddAudioTrack(AudioTrackInterface* track, } // Normal case; we've never seen this track before. - auto new_sender = CreateSender(webrtc::MediaType::AUDIO, track->id(), + auto new_sender = CreateSender(MediaType::AUDIO, track->id(), scoped_refptr(track), {stream->id()}, {{}}); new_sender->internal()->SetMediaChannel(voice_media_send_channel()); @@ -480,7 +479,7 @@ void RtpTransmissionManager::AddVideoTrack(VideoTrackInterface* track, } // Normal case; we've never seen this track before. - auto new_sender = CreateSender(webrtc::MediaType::VIDEO, track->id(), + auto new_sender = CreateSender(MediaType::VIDEO, track->id(), scoped_refptr(track), {stream->id()}, {{}}); new_sender->internal()->SetMediaChannel(video_media_send_channel()); @@ -564,7 +563,7 @@ RtpTransmissionManager::RemoveAndStopReceiver( << remote_sender_info.sender_id << " doesn't exist."; return nullptr; } - if (receiver->media_type() == webrtc::MediaType::AUDIO) { + if (receiver->media_type() == MediaType::AUDIO) { GetAudioTransceiver()->internal()->RemoveReceiver(receiver.get()); } else { GetVideoTransceiver()->internal()->RemoveReceiver(receiver.get()); @@ -575,15 +574,15 @@ RtpTransmissionManager::RemoveAndStopReceiver( void RtpTransmissionManager::OnRemoteSenderAdded( const RtpSenderInfo& sender_info, MediaStreamInterface* stream, - webrtc::MediaType media_type) { + MediaType media_type) { RTC_DCHECK_RUN_ON(signaling_thread()); - RTC_LOG(LS_INFO) << "Creating " << webrtc::MediaTypeToString(media_type) + RTC_LOG(LS_INFO) << "Creating " << MediaTypeToString(media_type) << " receiver for track_id=" << sender_info.sender_id << " and stream_id=" << sender_info.stream_id; - if (media_type == webrtc::MediaType::AUDIO) { + if (media_type == MediaType::AUDIO) { CreateAudioReceiver(stream, sender_info); - } else if (media_type == webrtc::MediaType::VIDEO) { + } else if (media_type == MediaType::VIDEO) { CreateVideoReceiver(stream, sender_info); } else { RTC_DCHECK_NOTREACHED() << "Invalid media type"; @@ -593,14 +592,14 @@ void RtpTransmissionManager::OnRemoteSenderAdded( void RtpTransmissionManager::OnRemoteSenderRemoved( const RtpSenderInfo& sender_info, MediaStreamInterface* stream, - webrtc::MediaType media_type) { + MediaType media_type) { RTC_DCHECK_RUN_ON(signaling_thread()); - RTC_LOG(LS_INFO) << "Removing " << webrtc::MediaTypeToString(media_type) + RTC_LOG(LS_INFO) << "Removing " << MediaTypeToString(media_type) << " receiver for track_id=" << sender_info.sender_id << " and stream_id=" << sender_info.stream_id; scoped_refptr receiver; - if (media_type == webrtc::MediaType::AUDIO) { + if (media_type == MediaType::AUDIO) { // When the MediaEngine audio channel is destroyed, the RemoteAudioSource // will be notified which will end the AudioRtpReceiver::track(). receiver = RemoveAndStopReceiver(sender_info); @@ -609,7 +608,7 @@ void RtpTransmissionManager::OnRemoteSenderRemoved( if (audio_track) { stream->RemoveTrack(audio_track); } - } else if (media_type == webrtc::MediaType::VIDEO) { + } else if (media_type == MediaType::VIDEO) { // Stopping or destroying a VideoRtpReceiver will end the // VideoRtpReceiver::track(). receiver = RemoveAndStopReceiver(sender_info); @@ -631,7 +630,7 @@ void RtpTransmissionManager::OnRemoteSenderRemoved( void RtpTransmissionManager::OnLocalSenderAdded( const RtpSenderInfo& sender_info, - webrtc::MediaType media_type) { + MediaType media_type) { RTC_DCHECK_RUN_ON(signaling_thread()); RTC_DCHECK(!IsUnifiedPlan()); auto sender = FindSenderById(sender_info.sender_id); @@ -654,7 +653,7 @@ void RtpTransmissionManager::OnLocalSenderAdded( void RtpTransmissionManager::OnLocalSenderRemoved( const RtpSenderInfo& sender_info, - webrtc::MediaType media_type) { + MediaType media_type) { RTC_DCHECK_RUN_ON(signaling_thread()); auto sender = FindSenderById(sender_info.sender_id); if (!sender) { @@ -676,19 +675,17 @@ void RtpTransmissionManager::OnLocalSenderRemoved( } std::vector* RtpTransmissionManager::GetRemoteSenderInfos( - webrtc::MediaType media_type) { - RTC_DCHECK(media_type == webrtc::MediaType::AUDIO || - media_type == webrtc::MediaType::VIDEO); - return (media_type == webrtc::MediaType::AUDIO) ? &remote_audio_sender_infos_ - : &remote_video_sender_infos_; + MediaType media_type) { + RTC_DCHECK(media_type == MediaType::AUDIO || media_type == MediaType::VIDEO); + return (media_type == MediaType::AUDIO) ? &remote_audio_sender_infos_ + : &remote_video_sender_infos_; } std::vector* RtpTransmissionManager::GetLocalSenderInfos( - webrtc::MediaType media_type) { - RTC_DCHECK(media_type == webrtc::MediaType::AUDIO || - media_type == webrtc::MediaType::VIDEO); - return (media_type == webrtc::MediaType::AUDIO) ? &local_audio_sender_infos_ - : &local_video_sender_infos_; + MediaType media_type) { + RTC_DCHECK(media_type == MediaType::AUDIO || media_type == MediaType::VIDEO); + return (media_type == MediaType::AUDIO) ? &local_audio_sender_infos_ + : &local_video_sender_infos_; } const RtpSenderInfo* RtpTransmissionManager::FindSenderInfo( diff --git a/pc/rtp_transport.cc b/pc/rtp_transport.cc index 2ab4ef55a7..b0cef00f74 100644 --- a/pc/rtp_transport.cc +++ b/pc/rtp_transport.cc @@ -205,7 +205,7 @@ flat_set RtpTransport::GetSsrcsForSink(RtpPacketSinkInterface* sink) { } void RtpTransport::DemuxPacket(CopyOnWriteBuffer packet, - webrtc::Timestamp arrival_time, + Timestamp arrival_time, EcnMarking ecn) { RtpPacketReceived parsed_packet(&header_extension_map_); parsed_packet.set_arrival_time(arrival_time); diff --git a/pc/rtp_transport_unittest.cc b/pc/rtp_transport_unittest.cc index c4d207d6c4..b5fe2296bc 100644 --- a/pc/rtp_transport_unittest.cc +++ b/pc/rtp_transport_unittest.cc @@ -411,7 +411,7 @@ TEST(RtpTransportTest, RecursiveSetSendDoesNotCrash) { // After the wait, the ready-to-send false is observed. EXPECT_THAT(WaitUntil([&] { return observer.ready_to_send_signal_count(); }, ::testing::Eq(2), - {.timeout = webrtc::TimeDelta::Millis(kShortTimeout)}), + {.timeout = TimeDelta::Millis(kShortTimeout)}), IsRtcOk()); EXPECT_FALSE(observer.ready_to_send()); } @@ -439,7 +439,7 @@ TEST(RtpTransportTest, RecursiveOnSentPacketDoesNotCrash) { EXPECT_EQ(observer.sent_packet_count(), 1); EXPECT_THAT( WaitUntil([&] { return observer.sent_packet_count(); }, ::testing::Eq(2), - {.timeout = webrtc::TimeDelta::Millis(kShortTimeout)}), + {.timeout = TimeDelta::Millis(kShortTimeout)}), IsRtcOk()); } diff --git a/pc/scenario_tests/goog_cc_test.cc b/pc/scenario_tests/goog_cc_test.cc index 2e8ff64e8f..057267c7f9 100644 --- a/pc/scenario_tests/goog_cc_test.cc +++ b/pc/scenario_tests/goog_cc_test.cc @@ -74,7 +74,7 @@ TEST(GoogCcPeerScenarioTest, MAYBE_NoBweChangeFromVideoUnmute) { ASSERT_EQ(num_video_streams, 1); // Exactly 1 video stream. auto get_bwe = [&] { - auto callback = make_ref_counted(); + auto callback = make_ref_counted(); caller->pc()->GetStats(callback.get()); s.net()->time_controller()->Wait([&] { return callback->called(); }); auto stats = diff --git a/pc/sdp_munging_detector.cc b/pc/sdp_munging_detector.cc index af1ebd3be0..95e29d575d 100644 --- a/pc/sdp_munging_detector.cc +++ b/pc/sdp_munging_detector.cc @@ -407,14 +407,14 @@ SdpMungingType DetermineSdpMungingType( continue; } // Validate video and audio contents. - webrtc::MediaType media_type = last_created_media_description->type(); - if (media_type == webrtc::MediaType::VIDEO) { + MediaType media_type = last_created_media_description->type(); + if (media_type == MediaType::VIDEO) { type = DetermineVideoSdpMungingType(last_created_media_description, media_description_to_set); if (type != SdpMungingType::kNoModification) { return type; } - } else if (media_type == webrtc::MediaType::AUDIO) { + } else if (media_type == MediaType::AUDIO) { type = DetermineAudioSdpMungingType(last_created_media_description, media_description_to_set); if (type != SdpMungingType::kNoModification) { @@ -434,7 +434,7 @@ SdpMungingType DetermineSdpMungingType( // Codec position swapped. for (size_t j = i + 1; j < last_created_codecs.size(); j++) { if (last_created_codecs[i] == codecs_to_set[j]) { - return media_type == webrtc::MediaType::AUDIO + return media_type == MediaType::AUDIO ? SdpMungingType::kAudioCodecsReordered : SdpMungingType::kVideoCodecsReordered; } @@ -445,13 +445,13 @@ SdpMungingType DetermineSdpMungingType( return SdpMungingType::kPayloadTypes; } if (last_created_codecs[i].params != codecs_to_set[i].params) { - return media_type == webrtc::MediaType::AUDIO + return media_type == MediaType::AUDIO ? SdpMungingType::kAudioCodecsFmtp : SdpMungingType::kVideoCodecsFmtp; } if (last_created_codecs[i].feedback_params != codecs_to_set[i].feedback_params) { - return media_type == webrtc::MediaType::AUDIO + return media_type == MediaType::AUDIO ? SdpMungingType::kAudioCodecsRtcpFb : SdpMungingType::kVideoCodecsRtcpFb; } diff --git a/pc/sdp_munging_detector_unittest.cc b/pc/sdp_munging_detector_unittest.cc index 340021ebfb..caf28dd5f2 100644 --- a/pc/sdp_munging_detector_unittest.cc +++ b/pc/sdp_munging_detector_unittest.cc @@ -148,8 +148,8 @@ TEST_F(SdpMungingTest, DISABLED_ReportUMAMetricsWithNoMunging) { auto caller = CreatePeerConnection(); auto callee = CreatePeerConnection(); - caller->AddTransceiver(webrtc::MediaType::AUDIO); - caller->AddTransceiver(webrtc::MediaType::VIDEO); + caller->AddTransceiver(MediaType::AUDIO); + caller->AddTransceiver(MediaType::VIDEO); // Negotiate, gather candidates, then exchange ICE candidates. ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get())); diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc index 5d4ce93856..d155d1d5a1 100644 --- a/pc/sdp_offer_answer.cc +++ b/pc/sdp_offer_answer.cc @@ -460,8 +460,7 @@ RTCError ValidateBundledPayloadTypes(const SessionDescription& description) { continue; } const auto type = media_description->type(); - if (type == webrtc::MediaType::AUDIO || - type == webrtc::MediaType::VIDEO) { + if (type == MediaType::AUDIO || type == MediaType::VIDEO) { for (const auto& c : media_description->codecs()) { auto error = FindDuplicateCodecParameters( c.ToCodecParameters(), payload_to_codec_parameters); @@ -593,7 +592,7 @@ RTCError ValidatePayloadTypes(const SessionDescription& description) { continue; } const auto type = media_description->type(); - if (type == webrtc::MediaType::AUDIO || type == webrtc::MediaType::VIDEO) { + if (type == MediaType::AUDIO || type == MediaType::VIDEO) { for (const auto& codec : media_description->codecs()) { if (!PayloadType::IsValid(codec.id, media_description->rtcp_mux())) { LOG_AND_RETURN_ERROR( @@ -719,15 +718,15 @@ RTCError DisableSimulcastInSender(scoped_refptr sender) { // The SDP parser used to populate these values by default for the 'content // name' if an a=mid line was absent. -absl::string_view GetDefaultMidForPlanB(webrtc::MediaType media_type) { +absl::string_view GetDefaultMidForPlanB(MediaType media_type) { switch (media_type) { - case webrtc::MediaType::AUDIO: + case MediaType::AUDIO: return CN_AUDIO; - case webrtc::MediaType::VIDEO: + case MediaType::VIDEO: return CN_VIDEO; - case webrtc::MediaType::DATA: + case MediaType::DATA: return CN_DATA; - case webrtc::MediaType::UNSUPPORTED: + case MediaType::UNSUPPORTED: return "not supported"; default: // Fall through to RTC_CHECK_NOTREACHED @@ -745,13 +744,13 @@ void AddPlanBRtpSenderOptions( MediaDescriptionOptions* video_media_description_options, int num_sim_layers) { for (const auto& sender : senders) { - if (sender->media_type() == webrtc::MediaType::AUDIO) { + if (sender->media_type() == MediaType::AUDIO) { if (audio_media_description_options) { audio_media_description_options->AddAudioSender( sender->id(), sender->internal()->stream_ids()); } } else { - RTC_DCHECK(sender->media_type() == webrtc::MediaType::VIDEO); + RTC_DCHECK(sender->media_type() == MediaType::VIDEO); if (video_media_description_options) { video_media_description_options->AddVideoSender( sender->id(), sender->internal()->stream_ids(), {}, @@ -1440,7 +1439,7 @@ std::unique_ptr SdpOfferAnswerHandler::Create( PeerConnectionSdpMethods* pc, const PeerConnectionInterface::RTCConfiguration& configuration, std::unique_ptr cert_generator, - std::unique_ptr + std::unique_ptr video_bitrate_allocator_factory, ConnectionContext* context, CodecLookupHelper* codec_lookup_helper) { @@ -1454,7 +1453,7 @@ std::unique_ptr SdpOfferAnswerHandler::Create( void SdpOfferAnswerHandler::Initialize( const PeerConnectionInterface::RTCConfiguration& configuration, std::unique_ptr cert_generator, - std::unique_ptr + std::unique_ptr video_bitrate_allocator_factory, ConnectionContext* context, CodecLookupHelper* codec_lookup_helper) { @@ -1962,7 +1961,7 @@ RTCError SdpOfferAnswerHandler::ApplyLocalDescription( GetFirstAudioContent(local_description()->description()); if (audio_content) { if (audio_content->rejected) { - RemoveSenders(webrtc::MediaType::AUDIO); + RemoveSenders(MediaType::AUDIO); } else { const MediaContentDescription* audio_desc = audio_content->media_description(); @@ -1974,7 +1973,7 @@ RTCError SdpOfferAnswerHandler::ApplyLocalDescription( GetFirstVideoContent(local_description()->description()); if (video_content) { if (video_content->rejected) { - RemoveSenders(webrtc::MediaType::VIDEO); + RemoveSenders(MediaType::VIDEO); } else { const MediaContentDescription* video_desc = video_content->media_description(); @@ -2348,7 +2347,7 @@ void SdpOfferAnswerHandler::PlanBUpdateSendersAndReceivers( // and MediaStreams. if (audio_content) { if (audio_content->rejected) { - RemoveSenders(webrtc::MediaType::AUDIO); + RemoveSenders(MediaType::AUDIO); } else { bool default_audio_track_needed = !remote_peer_supports_msid_ && @@ -2363,7 +2362,7 @@ void SdpOfferAnswerHandler::PlanBUpdateSendersAndReceivers( // and MediaStreams. if (video_content) { if (video_content->rejected) { - RemoveSenders(webrtc::MediaType::VIDEO); + RemoveSenders(MediaType::VIDEO); } else { bool default_video_track_needed = !remote_peer_supports_msid_ && @@ -3881,8 +3880,8 @@ RTCError SdpOfferAnswerHandler::ValidateSessionDescription( // media section. for (const ContentInfo& content : sdesc->description()->contents()) { const MediaContentDescription& desc = *content.media_description(); - if ((desc.type() == webrtc::MediaType::AUDIO || - desc.type() == webrtc::MediaType::VIDEO) && + if ((desc.type() == MediaType::AUDIO || + desc.type() == MediaType::VIDEO) && desc.streams().size() > 1u) { LOG_AND_RETURN_ERROR( RTCErrorType::INVALID_PARAMETER, @@ -3929,13 +3928,12 @@ RTCError SdpOfferAnswerHandler::UpdateTransceiversAndDataChannels( const ContentInfos& new_contents = new_session.description()->contents(); for (size_t i = 0; i < new_contents.size(); ++i) { const ContentInfo& new_content = new_contents[i]; - webrtc::MediaType media_type = new_content.media_description()->type(); + MediaType media_type = new_content.media_description()->type(); mid_generator_.AddKnownId(new_content.mid()); auto it = bundle_groups_by_mid.find(new_content.mid()); const ContentGroup* bundle_group = it != bundle_groups_by_mid.end() ? it->second : nullptr; - if (media_type == webrtc::MediaType::AUDIO || - media_type == webrtc::MediaType::VIDEO) { + if (media_type == MediaType::AUDIO || media_type == MediaType::VIDEO) { const ContentInfo* old_local_content = nullptr; if (old_local_description && i < old_local_description->description()->contents().size()) { @@ -3996,7 +3994,7 @@ RTCError SdpOfferAnswerHandler::UpdateTransceiversAndDataChannels( if (!error.ok()) { return error; } - } else if (media_type == webrtc::MediaType::DATA) { + } else if (media_type == MediaType::DATA) { const auto data_mid = pc_->sctp_mid(); if (data_mid && new_content.mid() != data_mid.value()) { // Ignore all but the first data section. @@ -4009,7 +4007,7 @@ RTCError SdpOfferAnswerHandler::UpdateTransceiversAndDataChannels( if (!error.ok()) { return error; } - } else if (media_type == webrtc::MediaType::UNSUPPORTED) { + } else if (media_type == MediaType::UNSUPPORTED) { RTC_LOG(LS_INFO) << "Ignoring unsupported media type"; } else { LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR, @@ -4074,8 +4072,7 @@ SdpOfferAnswerHandler::AssociateTransceiver( // If no RtpTransceiver was found in the previous step, create one with a // recvonly direction. if (!transceiver) { - RTC_LOG(LS_INFO) << "Adding " - << webrtc::MediaTypeToString(media_desc->type()) + RTC_LOG(LS_INFO) << "Adding " << MediaTypeToString(media_desc->type()) << " transceiver for MID=" << content.mid() << " at i=" << mline_index << " in response to the remote description."; @@ -4274,7 +4271,7 @@ void SdpOfferAnswerHandler::FillInMissingRemoteMids( scoped_refptr> SdpOfferAnswerHandler::FindAvailableTransceiverToReceive( - webrtc::MediaType media_type) const { + MediaType media_type) const { RTC_DCHECK_RUN_ON(signaling_thread()); RTC_DCHECK(IsUnifiedPlan()); // From JSEP section 5.10 (Applying a Remote Description): @@ -4405,7 +4402,7 @@ void SdpOfferAnswerHandler::GetOptionsForPlanBOffer( // Add audio/video/data m= sections to the end if needed. if (!audio_index && offer_new_audio_description) { MediaDescriptionOptions options( - webrtc::MediaType::AUDIO, CN_AUDIO, + MediaType::AUDIO, CN_AUDIO, RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio), false); options.header_extensions = media_engine()->voice().GetRtpHeaderExtensions(); @@ -4414,7 +4411,7 @@ void SdpOfferAnswerHandler::GetOptionsForPlanBOffer( } if (!video_index && offer_new_video_description) { MediaDescriptionOptions options( - webrtc::MediaType::VIDEO, CN_VIDEO, + MediaType::VIDEO, CN_VIDEO, RtpTransceiverDirectionFromSendRecv(send_video, recv_video), false); options.header_extensions = media_engine()->video().GetRtpHeaderExtensions(); @@ -4477,11 +4474,10 @@ void SdpOfferAnswerHandler::GetOptionsForUnifiedPlanOffer( (current_remote_content && current_remote_content->rejected); const std::string& mid = (local_content ? local_content->mid() : remote_content->mid()); - webrtc::MediaType media_type = + MediaType media_type = (local_content ? local_content->media_description()->type() : remote_content->media_description()->type()); - if (media_type == webrtc::MediaType::AUDIO || - media_type == webrtc::MediaType::VIDEO) { + if (media_type == MediaType::AUDIO || media_type == MediaType::VIDEO) { // A media section is considered eligible for recycling if it is marked as // rejected in either the current local or current remote description. auto transceiver = transceivers()->FindByMid(mid); @@ -4515,14 +4511,14 @@ void SdpOfferAnswerHandler::GetOptionsForUnifiedPlanOffer( transceiver->internal()->set_mline_index(i); } } - } else if (media_type == webrtc::MediaType::UNSUPPORTED) { + } else if (media_type == MediaType::UNSUPPORTED) { RTC_DCHECK(local_content->rejected); session_options->media_description_options.push_back( MediaDescriptionOptions(media_type, mid, RtpTransceiverDirection::kInactive, /*stopped=*/true)); } else { - RTC_CHECK_EQ(webrtc::MediaType::DATA, media_type); + RTC_CHECK_EQ(MediaType::DATA, media_type); if (had_been_rejected) { session_options->media_description_options.push_back( GetMediaDescriptionOptionsForRejectedData(mid)); @@ -4580,7 +4576,7 @@ void SdpOfferAnswerHandler::GetOptionsForUnifiedPlanOffer( for (size_t i = 0; i < session_options->media_description_options.size(); i++) { auto media_description = session_options->media_description_options[i]; - if (media_description.type == webrtc::MediaType::DATA && + if (media_description.type == MediaType::DATA && media_description.stopped) { session_options->media_description_options[i] = GetMediaDescriptionOptionsForActiveData(media_description.mid); @@ -4689,9 +4685,8 @@ void SdpOfferAnswerHandler::GetOptionsForUnifiedPlanAnswer( RTC_DCHECK(remote_description()->GetType() == SdpType::kOffer); for (const ContentInfo& content : remote_description()->description()->contents()) { - webrtc::MediaType media_type = content.media_description()->type(); - if (media_type == webrtc::MediaType::AUDIO || - media_type == webrtc::MediaType::VIDEO) { + MediaType media_type = content.media_description()->type(); + if (media_type == MediaType::AUDIO || media_type == MediaType::VIDEO) { auto transceiver = transceivers()->FindByMid(content.mid()); if (transceiver) { session_options->media_description_options.push_back( @@ -4706,14 +4701,14 @@ void SdpOfferAnswerHandler::GetOptionsForUnifiedPlanAnswer( RtpTransceiverDirection::kInactive, /*stopped=*/true)); } - } else if (media_type == webrtc::MediaType::UNSUPPORTED) { + } else if (media_type == MediaType::UNSUPPORTED) { RTC_DCHECK(content.rejected); session_options->media_description_options.push_back( MediaDescriptionOptions(media_type, content.mid(), RtpTransceiverDirection::kInactive, /*stopped=*/true)); } else { - RTC_CHECK_EQ(webrtc::MediaType::DATA, media_type); + RTC_CHECK_EQ(MediaType::DATA, media_type); // Reject all data sections if data channels are disabled. // Reject a data section if it has already been rejected. // Reject all data sections except for the first one. @@ -4767,20 +4762,18 @@ RTCError SdpOfferAnswerHandler::HandleLegacyOfferOptions( RTC_DCHECK(IsUnifiedPlan()); if (options.offer_to_receive_audio == 0) { - RemoveRecvDirectionFromReceivingTransceiversOfType( - webrtc::MediaType::AUDIO); + RemoveRecvDirectionFromReceivingTransceiversOfType(MediaType::AUDIO); } else if (options.offer_to_receive_audio == 1) { - AddUpToOneReceivingTransceiverOfType(webrtc::MediaType::AUDIO); + AddUpToOneReceivingTransceiverOfType(MediaType::AUDIO); } else if (options.offer_to_receive_audio > 1) { LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER, "offer_to_receive_audio > 1 is not supported."); } if (options.offer_to_receive_video == 0) { - RemoveRecvDirectionFromReceivingTransceiversOfType( - webrtc::MediaType::VIDEO); + RemoveRecvDirectionFromReceivingTransceiversOfType(MediaType::VIDEO); } else if (options.offer_to_receive_video == 1) { - AddUpToOneReceivingTransceiverOfType(webrtc::MediaType::VIDEO); + AddUpToOneReceivingTransceiverOfType(MediaType::VIDEO); } else if (options.offer_to_receive_video > 1) { LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER, "offer_to_receive_video > 1 is not supported."); @@ -4790,12 +4783,12 @@ RTCError SdpOfferAnswerHandler::HandleLegacyOfferOptions( } void SdpOfferAnswerHandler::RemoveRecvDirectionFromReceivingTransceiversOfType( - webrtc::MediaType media_type) { + MediaType media_type) { for (const auto& transceiver : GetReceivingTransceiversOfType(media_type)) { RtpTransceiverDirection new_direction = RtpTransceiverDirectionWithRecvSet(transceiver->direction(), false); if (new_direction != transceiver->direction()) { - RTC_LOG(LS_INFO) << "Changing " << webrtc::MediaTypeToString(media_type) + RTC_LOG(LS_INFO) << "Changing " << MediaTypeToString(media_type) << " transceiver (MID=" << transceiver->mid().value_or("") << ") from " << RtpTransceiverDirectionToString( @@ -4809,11 +4802,11 @@ void SdpOfferAnswerHandler::RemoveRecvDirectionFromReceivingTransceiversOfType( } void SdpOfferAnswerHandler::AddUpToOneReceivingTransceiverOfType( - webrtc::MediaType media_type) { + MediaType media_type) { RTC_DCHECK_RUN_ON(signaling_thread()); if (GetReceivingTransceiversOfType(media_type).empty()) { RTC_LOG(LS_INFO) - << "Adding one recvonly " << webrtc::MediaTypeToString(media_type) + << "Adding one recvonly " << MediaTypeToString(media_type) << " transceiver since CreateOffer specified offer_to_receive=1"; RtpTransceiverInit init; init.direction = RtpTransceiverDirection::kRecvOnly; @@ -4823,8 +4816,7 @@ void SdpOfferAnswerHandler::AddUpToOneReceivingTransceiverOfType( } std::vector>> -SdpOfferAnswerHandler::GetReceivingTransceiversOfType( - webrtc::MediaType media_type) { +SdpOfferAnswerHandler::GetReceivingTransceiversOfType(MediaType media_type) { std::vector>> receiving_transceivers; for (const auto& transceiver : transceivers()->List()) { @@ -4867,7 +4859,7 @@ void SdpOfferAnswerHandler::RemoveRemoteStreamsIfEmpty( } } -void SdpOfferAnswerHandler::RemoveSenders(webrtc::MediaType media_type) { +void SdpOfferAnswerHandler::RemoveSenders(MediaType media_type) { RTC_DCHECK_RUN_ON(signaling_thread()); UpdateLocalSenders(std::vector(), media_type); UpdateRemoteSendersList(std::vector(), false, media_type, @@ -4876,7 +4868,7 @@ void SdpOfferAnswerHandler::RemoveSenders(webrtc::MediaType media_type) { void SdpOfferAnswerHandler::UpdateLocalSenders( const std::vector& streams, - webrtc::MediaType media_type) { + MediaType media_type) { TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::UpdateLocalSenders"); RTC_DCHECK_RUN_ON(signaling_thread()); std::vector* current_senders = @@ -4917,7 +4909,7 @@ void SdpOfferAnswerHandler::UpdateLocalSenders( void SdpOfferAnswerHandler::UpdateRemoteSendersList( const StreamParamsVec& streams, bool default_sender_needed, - webrtc::MediaType media_type, + MediaType media_type, StreamCollection* new_streams) { TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::UpdateRemoteSendersList"); RTC_DCHECK_RUN_ON(signaling_thread()); @@ -5003,7 +4995,7 @@ void SdpOfferAnswerHandler::UpdateRemoteSendersList( remote_streams_->AddStream(default_stream); new_streams->AddStream(default_stream); } - std::string default_sender_id = (media_type == webrtc::MediaType::AUDIO) + std::string default_sender_id = (media_type == MediaType::AUDIO) ? kDefaultAudioSenderId : kDefaultVideoSenderId; const RtpSenderInfo* default_sender_info = rtp_manager()->FindSenderInfo( @@ -5477,12 +5469,12 @@ void SdpOfferAnswerHandler::DestroyMediaChannels() { RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(0); for (const auto& transceiver : list) { - if (transceiver->media_type() == webrtc::MediaType::VIDEO) { + if (transceiver->media_type() == MediaType::VIDEO) { transceiver->internal()->ClearChannel(); } } for (const auto& transceiver : list) { - if (transceiver->media_type() == webrtc::MediaType::AUDIO) { + if (transceiver->media_type() == MediaType::AUDIO) { transceiver->internal()->ClearChannel(); } } @@ -5502,13 +5494,13 @@ void SdpOfferAnswerHandler::GenerateMediaDescriptionOptions( // If we already have an audio m= section, reject this extra one. if (*audio_index) { session_options->media_description_options.push_back( - MediaDescriptionOptions(webrtc::MediaType::AUDIO, content.mid(), + MediaDescriptionOptions(MediaType::AUDIO, content.mid(), RtpTransceiverDirection::kInactive, /*stopped=*/true)); } else { bool stopped = (audio_direction == RtpTransceiverDirection::kInactive); session_options->media_description_options.push_back( - MediaDescriptionOptions(webrtc::MediaType::AUDIO, content.mid(), + MediaDescriptionOptions(MediaType::AUDIO, content.mid(), audio_direction, stopped)); *audio_index = session_options->media_description_options.size() - 1; } @@ -5518,13 +5510,13 @@ void SdpOfferAnswerHandler::GenerateMediaDescriptionOptions( // If we already have an video m= section, reject this extra one. if (*video_index) { session_options->media_description_options.push_back( - MediaDescriptionOptions(webrtc::MediaType::VIDEO, content.mid(), + MediaDescriptionOptions(MediaType::VIDEO, content.mid(), RtpTransceiverDirection::kInactive, /*stopped=*/true)); } else { bool stopped = (video_direction == RtpTransceiverDirection::kInactive); session_options->media_description_options.push_back( - MediaDescriptionOptions(webrtc::MediaType::VIDEO, content.mid(), + MediaDescriptionOptions(MediaType::VIDEO, content.mid(), video_direction, stopped)); *video_index = session_options->media_description_options.size() - 1; } @@ -5532,7 +5524,7 @@ void SdpOfferAnswerHandler::GenerateMediaDescriptionOptions( media_engine()->video().GetRtpHeaderExtensions(); } else if (IsUnsupportedContent(&content)) { session_options->media_description_options.push_back( - MediaDescriptionOptions(webrtc::MediaType::UNSUPPORTED, content.mid(), + MediaDescriptionOptions(MediaType::UNSUPPORTED, content.mid(), RtpTransceiverDirection::kInactive, /*stopped=*/true)); } else { @@ -5556,7 +5548,7 @@ SdpOfferAnswerHandler::GetMediaDescriptionOptionsForActiveData( RTC_DCHECK_RUN_ON(signaling_thread()); // Direction for data sections is meaningless, but legacy endpoints might // expect sendrecv. - MediaDescriptionOptions options(webrtc::MediaType::DATA, mid, + MediaDescriptionOptions options(MediaType::DATA, mid, RtpTransceiverDirection::kSendRecv, /*stopped=*/false); return options; @@ -5566,7 +5558,7 @@ MediaDescriptionOptions SdpOfferAnswerHandler::GetMediaDescriptionOptionsForRejectedData( const std::string& mid) const { RTC_DCHECK_RUN_ON(signaling_thread()); - MediaDescriptionOptions options(webrtc::MediaType::DATA, mid, + MediaDescriptionOptions options(MediaType::DATA, mid, RtpTransceiverDirection::kInactive, /*stopped=*/true); return options; @@ -5619,15 +5611,13 @@ bool SdpOfferAnswerHandler::UpdatePayloadTypeDemuxingState( // Ignore transceivers that are not receiving. continue; } - const webrtc::MediaType media_type = - content_info.media_description()->type(); - if (media_type == webrtc::MediaType::AUDIO || - media_type == webrtc::MediaType::VIDEO) { - if (media_type == webrtc::MediaType::AUDIO && + const MediaType media_type = content_info.media_description()->type(); + if (media_type == MediaType::AUDIO || media_type == MediaType::VIDEO) { + if (media_type == MediaType::AUDIO && !mid_header_extension_missing_audio) { mid_header_extension_missing_audio = !ContentHasHeaderExtension(content_info, RtpExtension::kMidUri); - } else if (media_type == webrtc::MediaType::VIDEO && + } else if (media_type == MediaType::VIDEO && !mid_header_extension_missing_video) { mid_header_extension_missing_video = !ContentHasHeaderExtension(content_info, RtpExtension::kMidUri); @@ -5635,16 +5625,16 @@ bool SdpOfferAnswerHandler::UpdatePayloadTypeDemuxingState( const MediaContentDescription* media_desc = content_info.media_description(); for (const Codec& codec : media_desc->codecs()) { - if (media_type == webrtc::MediaType::AUDIO) { + if (media_type == MediaType::AUDIO) { if (payload_types->audio_payload_types.count(codec.id)) { // Two m= sections are using the same payload type, thus demuxing // by payload type is not possible. - if (media_type == webrtc::MediaType::AUDIO) { + if (media_type == MediaType::AUDIO) { payload_types->pt_demuxing_possible_audio = false; } } payload_types->audio_payload_types.insert(codec.id); - } else if (media_type == webrtc::MediaType::VIDEO) { + } else if (media_type == MediaType::VIDEO) { if (payload_types->video_payload_types.count(codec.id)) { // Two m= sections are using the same payload type, thus demuxing // by payload type is not possible. @@ -5686,9 +5676,8 @@ bool SdpOfferAnswerHandler::UpdatePayloadTypeDemuxingState( continue; } - const webrtc::MediaType media_type = channel->media_type(); - if (media_type != webrtc::MediaType::AUDIO && - media_type != webrtc::MediaType::VIDEO) { + const MediaType media_type = channel->media_type(); + if (media_type != MediaType::AUDIO && media_type != MediaType::VIDEO) { continue; } @@ -5702,7 +5691,7 @@ bool SdpOfferAnswerHandler::UpdatePayloadTypeDemuxingState( const ContentGroup* bundle_group = bundle_it != bundle_groups_by_mid.end() ? bundle_it->second : nullptr; bool pt_demux_enabled = RtpTransceiverDirectionHasRecv(local_direction); - if (media_type == webrtc::MediaType::AUDIO) { + if (media_type == MediaType::AUDIO) { pt_demux_enabled &= !bundle_group || (bundled_pt_demux_allowed_audio && @@ -5711,7 +5700,7 @@ bool SdpOfferAnswerHandler::UpdatePayloadTypeDemuxingState( pt_demuxing_has_been_used_audio_ = true; } } else { - RTC_DCHECK_EQ(media_type, webrtc::MediaType::VIDEO); + RTC_DCHECK_EQ(media_type, MediaType::VIDEO); pt_demux_enabled &= !bundle_group || (bundled_pt_demux_allowed_video && diff --git a/pc/sdp_offer_answer_unittest.cc b/pc/sdp_offer_answer_unittest.cc index 3019c3cced..3830fc2f28 100644 --- a/pc/sdp_offer_answer_unittest.cc +++ b/pc/sdp_offer_answer_unittest.cc @@ -132,7 +132,7 @@ class SdpOfferAnswerTest : public ::testing::Test { } std::optional FindFirstSendCodecWithName( - webrtc::MediaType media_type, + MediaType media_type, const std::string& name) const { std::vector codecs = pc_factory_->GetRtpSenderCapabilities(media_type).codecs; @@ -156,7 +156,7 @@ TEST_F(SdpOfferAnswerTest, OnTrackReturnsProxiedObject) { auto caller = CreatePeerConnection(); auto callee = CreatePeerConnection(); - auto audio_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); + auto audio_transceiver = caller->AddTransceiver(MediaType::AUDIO); ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get())); // Verify that caller->observer->OnTrack() has been called with a @@ -600,7 +600,7 @@ TEST_F(SdpOfferAnswerTest, SimulcastAnswerWithNoRidsIsRejected) { rid2.rid = "2"; init.send_encodings.push_back(rid2); - auto transceiver = pc->AddTransceiver(webrtc::MediaType::VIDEO, init); + auto transceiver = pc->AddTransceiver(MediaType::VIDEO, init); EXPECT_TRUE(pc->CreateOfferAndSetAsLocal()); auto mid = pc->pc()->local_description()->description()->contents()[0].mid(); @@ -652,10 +652,10 @@ TEST_F(SdpOfferAnswerTest, SimulcastOfferWithMixedCodec) { FieldTrials::CreateNoGlobal("WebRTC-MixedCodecSimulcast/Enabled/")); std::optional vp8_codec_capability = - FindFirstSendCodecWithName(webrtc::MediaType::VIDEO, kVp8CodecName); + FindFirstSendCodecWithName(MediaType::VIDEO, kVp8CodecName); ASSERT_TRUE(vp8_codec_capability); std::optional vp9_codec_capability = - FindFirstSendCodecWithName(webrtc::MediaType::VIDEO, kVp9CodecName); + FindFirstSendCodecWithName(MediaType::VIDEO, kVp9CodecName); ASSERT_TRUE(vp9_codec_capability); RtpTransceiverInit init; @@ -668,7 +668,7 @@ TEST_F(SdpOfferAnswerTest, SimulcastOfferWithMixedCodec) { rid2.codec = *vp9_codec_capability; init.send_encodings.push_back(rid2); - auto transceiver = pc->AddTransceiver(webrtc::MediaType::VIDEO, init); + auto transceiver = pc->AddTransceiver(MediaType::VIDEO, init); auto offer = pc->CreateOffer(); auto& offer_contents = offer->description()->contents(); auto send_codecs = offer_contents[0].media_description()->codecs(); @@ -1611,8 +1611,8 @@ TEST_F(SdpOfferAnswerTest, ReducedSizeNegotiated) { auto caller = CreatePeerConnection(); auto callee = CreatePeerConnection(); - auto audio_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); - auto video_transceiver = caller->AddTransceiver(webrtc::MediaType::VIDEO); + auto audio_transceiver = caller->AddTransceiver(MediaType::AUDIO); + auto video_transceiver = caller->AddTransceiver(MediaType::VIDEO); ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get())); auto receivers = callee->pc()->GetReceivers(); @@ -1634,8 +1634,8 @@ TEST_F(SdpOfferAnswerTest, ReducedSizeNotNegotiated) { auto caller = CreatePeerConnection(); auto callee = CreatePeerConnection(); - auto audio_transceiver = caller->AddTransceiver(webrtc::MediaType::AUDIO); - auto video_transceiver = caller->AddTransceiver(webrtc::MediaType::VIDEO); + auto audio_transceiver = caller->AddTransceiver(MediaType::AUDIO); + auto video_transceiver = caller->AddTransceiver(MediaType::VIDEO); auto offer = caller->CreateOfferAndSetAsLocal(); ASSERT_NE(offer, nullptr); @@ -1669,9 +1669,9 @@ TEST_F(SdpOfferAnswerTest, PayloadTypeMatchingWithSubsequentOfferAnswer) { // 1. Restrict codecs and set a local description and remote description. // with a different payload type. - auto video_transceiver = caller->AddTransceiver(webrtc::MediaType::VIDEO); + auto video_transceiver = caller->AddTransceiver(MediaType::VIDEO); std::vector codec_caps = - pc_factory_->GetRtpReceiverCapabilities(webrtc::MediaType::VIDEO).codecs; + pc_factory_->GetRtpReceiverCapabilities(MediaType::VIDEO).codecs; codec_caps.erase(std::remove_if(codec_caps.begin(), codec_caps.end(), [](const RtpCodecCapability& codec) { return !absl::EqualsIgnoreCase(codec.name, @@ -1700,8 +1700,7 @@ TEST_F(SdpOfferAnswerTest, PayloadTypeMatchingWithSubsequentOfferAnswer) { EXPECT_TRUE(caller->SetRemoteDescription(std::move(answer1))); // 3. sCP to reenable that codec. Payload type is not matched at this point. - codec_caps = - pc_factory_->GetRtpReceiverCapabilities(webrtc::MediaType::VIDEO).codecs; + codec_caps = pc_factory_->GetRtpReceiverCapabilities(MediaType::VIDEO).codecs; codec_caps.erase( std::remove_if(codec_caps.begin(), codec_caps.end(), [](const RtpCodecCapability& codec) { diff --git a/pc/simulcast_sdp_serializer.cc b/pc/simulcast_sdp_serializer.cc index ba90556d6c..9f3539923d 100644 --- a/pc/simulcast_sdp_serializer.cc +++ b/pc/simulcast_sdp_serializer.cc @@ -145,9 +145,9 @@ RTCErrorOr ParseSimulcastLayerList(const std::string& str) { return std::move(result); } -webrtc::RTCError ParseRidPayloadList(const std::string& payload_list, - RidDescription* rid_description, - std::vector* rid_payload_types) { +RTCError ParseRidPayloadList(const std::string& payload_list, + RidDescription* rid_description, + std::vector* rid_payload_types) { RTC_DCHECK(rid_description); RTC_DCHECK(rid_payload_types); // Check that the description doesn't have any payload types or restrictions. diff --git a/pc/srtp_session.cc b/pc/srtp_session.cc index 7fe7e04f56..6d94c4921a 100644 --- a/pc/srtp_session.cc +++ b/pc/srtp_session.cc @@ -221,7 +221,7 @@ bool SrtpSession::ProtectRtp(CopyOnWriteBuffer& buffer) { int out_len = buffer.size(); int err = srtp_protect(session_, buffer.MutableData(), &out_len); - int seq_num = webrtc::ParseRtpSequenceNumber(buffer); + int seq_num = ParseRtpSequenceNumber(buffer); if (err != srtp_err_status_ok) { RTC_LOG(LS_WARNING) << "Failed to protect SRTP packet, seqnum=" << seq_num << ", err=" << err @@ -256,7 +256,7 @@ bool SrtpSession::ProtectRtp(void* p, int in_len, int max_len, int* out_len) { *out_len = in_len; int err = srtp_protect(session_, p, out_len); - int seq_num = webrtc::ParseRtpSequenceNumber( + int seq_num = ParseRtpSequenceNumber( MakeArrayView(reinterpret_cast(p), in_len)); if (err != srtp_err_status_ok) { RTC_LOG(LS_WARNING) << "Failed to protect SRTP packet, seqnum=" << seq_num @@ -508,18 +508,17 @@ bool SrtpSession::GetSendStreamPacketIndex(CopyOnWriteBuffer& buffer, int64_t* index) { RTC_DCHECK(thread_checker_.IsCurrent()); - uint32_t ssrc = webrtc::ParseRtpSsrc(buffer); + uint32_t ssrc = ParseRtpSsrc(buffer); uint32_t roc; if (srtp_get_stream_roc(session_, ssrc, &roc) != srtp_err_status_ok) { return false; } // Calculate the extended sequence number. - uint16_t seq_num = webrtc::ParseRtpSequenceNumber(buffer); + uint16_t seq_num = ParseRtpSequenceNumber(buffer); int64_t extended_seq_num = (roc << 16) + seq_num; // Shift extended sequence number, put into network byte order - *index = - static_cast(webrtc::NetworkToHost64(extended_seq_num << 16)); + *index = static_cast(NetworkToHost64(extended_seq_num << 16)); return true; } @@ -561,7 +560,7 @@ bool SrtpSession::DoSetKey(int type, // Enable external HMAC authentication only for outgoing streams and only // for cipher suites that support it (i.e. only non-GCM cipher suites). if (type == ssrc_any_outbound && IsExternalAuthEnabled() && - !webrtc::IsGcmCryptoSuite(crypto_suite)) { + !IsGcmCryptoSuite(crypto_suite)) { policy.rtp.auth_type = EXTERNAL_HMAC_SHA1; } if (!extension_ids.empty()) { @@ -672,7 +671,7 @@ void SrtpSession::HandleEventThunk(srtp_event_data_t* ev) { // The resulting file can be replayed using the WebRTC video_replay tool and // be inspected in Wireshark using the RTP, VP8 and H264 dissectors. void SrtpSession::DumpPacket(const CopyOnWriteBuffer& buffer, bool outbound) { - int64_t time_of_day = webrtc::TimeUTCMillis() % (24 * 3600 * 1000); + int64_t time_of_day = TimeUTCMillis() % (24 * 3600 * 1000); int64_t hours = time_of_day / (3600 * 1000); int64_t minutes = (time_of_day / (60 * 1000)) % 60; int64_t seconds = (time_of_day / 1000) % 60; @@ -683,7 +682,7 @@ void SrtpSession::DumpPacket(const CopyOnWriteBuffer& buffer, bool outbound) { << hours << ":" << std::setfill('0') << std::setw(2) << minutes << ":" << std::setfill('0') << std::setw(2) << seconds << "." << std::setfill('0') << std::setw(3) << millis << " " << "000000 " - << webrtc::hex_encode_with_delimiter( + << hex_encode_with_delimiter( absl::string_view(buffer.data(), buffer.size()), ' ') << " # RTP_DUMP"; } diff --git a/pc/srtp_session_unittest.cc b/pc/srtp_session_unittest.cc index fd58ce79b5..56a38dc1f0 100644 --- a/pc/srtp_session_unittest.cc +++ b/pc/srtp_session_unittest.cc @@ -39,7 +39,7 @@ std::vector kEncryptedHeaderExtensionIds; class SrtpSessionTest : public ::testing::Test { public: SrtpSessionTest() : s1_(field_trials_), s2_(field_trials_) { - webrtc::metrics::Reset(); + metrics::Reset(); } protected: @@ -53,8 +53,7 @@ class SrtpSessionTest : public ::testing::Test { } void TestProtectRtp(int crypto_suite) { EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_)); - EXPECT_EQ(rtp_packet_.size(), - rtp_len_ + webrtc::rtp_auth_tag_len(crypto_suite)); + EXPECT_EQ(rtp_packet_.size(), rtp_len_ + rtp_auth_tag_len(crypto_suite)); // Check that Protect changed the content (up to the original length). EXPECT_NE(0, std::memcmp(kPcmuFrame, rtp_packet_.data(), rtp_len_)); rtp_len_ = rtp_packet_.size(); @@ -62,7 +61,7 @@ class SrtpSessionTest : public ::testing::Test { void TestProtectRtcp(int crypto_suite) { EXPECT_TRUE(s1_.ProtectRtcp(rtcp_packet_)); EXPECT_EQ(rtcp_packet_.size(), - rtcp_len_ + 4 + webrtc::rtcp_auth_tag_len(crypto_suite)); + rtcp_len_ + 4 + rtcp_auth_tag_len(crypto_suite)); // Check that Protect changed the content (up to the original length). EXPECT_NE(0, std::memcmp(kRtcpReport, rtcp_packet_.data(), rtcp_len_)); rtcp_len_ = rtcp_packet_.size(); @@ -90,110 +89,108 @@ class SrtpSessionTest : public ::testing::Test { // Test that we can set up the session and keys properly. TEST_F(SrtpSessionTest, TestGoodSetup) { - EXPECT_TRUE(s1_.SetSend(webrtc::kSrtpAes128CmSha1_80, webrtc::kTestKey1, + EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kEncryptedHeaderExtensionIds)); - EXPECT_TRUE(s2_.SetReceive(webrtc::kSrtpAes128CmSha1_80, webrtc::kTestKey1, + EXPECT_TRUE(s2_.SetReceive(kSrtpAes128CmSha1_80, kTestKey1, kEncryptedHeaderExtensionIds)); } // Test that we can't change the keys once set. TEST_F(SrtpSessionTest, TestBadSetup) { - EXPECT_TRUE(s1_.SetSend(webrtc::kSrtpAes128CmSha1_80, webrtc::kTestKey1, + EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kEncryptedHeaderExtensionIds)); - EXPECT_TRUE(s2_.SetReceive(webrtc::kSrtpAes128CmSha1_80, webrtc::kTestKey1, + EXPECT_TRUE(s2_.SetReceive(kSrtpAes128CmSha1_80, kTestKey1, kEncryptedHeaderExtensionIds)); - EXPECT_FALSE(s1_.SetSend(webrtc::kSrtpAes128CmSha1_80, webrtc::kTestKey2, + EXPECT_FALSE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey2, kEncryptedHeaderExtensionIds)); - EXPECT_FALSE(s2_.SetReceive(webrtc::kSrtpAes128CmSha1_80, webrtc::kTestKey2, + EXPECT_FALSE(s2_.SetReceive(kSrtpAes128CmSha1_80, kTestKey2, kEncryptedHeaderExtensionIds)); } // Test that we fail keys of the wrong length. TEST_F(SrtpSessionTest, TestKeysTooShort) { - EXPECT_FALSE( - s1_.SetSend(webrtc::kSrtpAes128CmSha1_80, - ZeroOnFreeBuffer(webrtc::kTestKey1.data(), 1), - kEncryptedHeaderExtensionIds)); - EXPECT_FALSE( - s2_.SetReceive(webrtc::kSrtpAes128CmSha1_80, - ZeroOnFreeBuffer(webrtc::kTestKey1.data(), 1), - kEncryptedHeaderExtensionIds)); + EXPECT_FALSE(s1_.SetSend(kSrtpAes128CmSha1_80, + ZeroOnFreeBuffer(kTestKey1.data(), 1), + kEncryptedHeaderExtensionIds)); + EXPECT_FALSE(s2_.SetReceive(kSrtpAes128CmSha1_80, + ZeroOnFreeBuffer(kTestKey1.data(), 1), + kEncryptedHeaderExtensionIds)); } // Test that we can encrypt and decrypt RTP/RTCP using AES_CM_128_HMAC_SHA1_80. TEST_F(SrtpSessionTest, TestProtect_AES_CM_128_HMAC_SHA1_80) { - EXPECT_TRUE(s1_.SetSend(webrtc::kSrtpAes128CmSha1_80, webrtc::kTestKey1, + EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kEncryptedHeaderExtensionIds)); - EXPECT_TRUE(s2_.SetReceive(webrtc::kSrtpAes128CmSha1_80, webrtc::kTestKey1, + EXPECT_TRUE(s2_.SetReceive(kSrtpAes128CmSha1_80, kTestKey1, kEncryptedHeaderExtensionIds)); - TestProtectRtp(webrtc::kSrtpAes128CmSha1_80); - TestProtectRtcp(webrtc::kSrtpAes128CmSha1_80); - TestUnprotectRtp(webrtc::kSrtpAes128CmSha1_80); - TestUnprotectRtcp(webrtc::kSrtpAes128CmSha1_80); + TestProtectRtp(kSrtpAes128CmSha1_80); + TestProtectRtcp(kSrtpAes128CmSha1_80); + TestUnprotectRtp(kSrtpAes128CmSha1_80); + TestUnprotectRtcp(kSrtpAes128CmSha1_80); } // Test that we can encrypt and decrypt RTP/RTCP using AES_CM_128_HMAC_SHA1_32. TEST_F(SrtpSessionTest, TestProtect_AES_CM_128_HMAC_SHA1_32) { - EXPECT_TRUE(s1_.SetSend(webrtc::kSrtpAes128CmSha1_32, webrtc::kTestKey1, + EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_32, kTestKey1, kEncryptedHeaderExtensionIds)); - EXPECT_TRUE(s2_.SetReceive(webrtc::kSrtpAes128CmSha1_32, webrtc::kTestKey1, + EXPECT_TRUE(s2_.SetReceive(kSrtpAes128CmSha1_32, kTestKey1, kEncryptedHeaderExtensionIds)); - TestProtectRtp(webrtc::kSrtpAes128CmSha1_32); - TestProtectRtcp(webrtc::kSrtpAes128CmSha1_32); - TestUnprotectRtp(webrtc::kSrtpAes128CmSha1_32); - TestUnprotectRtcp(webrtc::kSrtpAes128CmSha1_32); + TestProtectRtp(kSrtpAes128CmSha1_32); + TestProtectRtcp(kSrtpAes128CmSha1_32); + TestUnprotectRtp(kSrtpAes128CmSha1_32); + TestUnprotectRtcp(kSrtpAes128CmSha1_32); } TEST_F(SrtpSessionTest, TestGetSendStreamPacketIndex) { - EXPECT_TRUE(s1_.SetSend(webrtc::kSrtpAes128CmSha1_32, webrtc::kTestKey1, + EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_32, kTestKey1, kEncryptedHeaderExtensionIds)); int64_t index; EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_, &index)); // `index` will be shifted by 16. - int64_t be64_index = static_cast(webrtc::NetworkToHost64(1 << 16)); + int64_t be64_index = static_cast(NetworkToHost64(1 << 16)); EXPECT_EQ(be64_index, index); } // Test that we fail to unprotect if someone tampers with the RTP/RTCP paylaods. TEST_F(SrtpSessionTest, TestTamperReject) { - EXPECT_TRUE(s1_.SetSend(webrtc::kSrtpAes128CmSha1_80, webrtc::kTestKey1, + EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kEncryptedHeaderExtensionIds)); - EXPECT_TRUE(s2_.SetReceive(webrtc::kSrtpAes128CmSha1_80, webrtc::kTestKey1, + EXPECT_TRUE(s2_.SetReceive(kSrtpAes128CmSha1_80, kTestKey1, kEncryptedHeaderExtensionIds)); - TestProtectRtp(webrtc::kSrtpAes128CmSha1_80); + TestProtectRtp(kSrtpAes128CmSha1_80); rtp_packet_.MutableData()[0] = 0x12; EXPECT_FALSE(s2_.UnprotectRtp(rtp_packet_)); EXPECT_METRIC_THAT( - webrtc::metrics::Samples("WebRTC.PeerConnection.SrtpUnprotectError"), + metrics::Samples("WebRTC.PeerConnection.SrtpUnprotectError"), ElementsAre(Pair(srtp_err_status_bad_param, 1))); - TestProtectRtcp(webrtc::kSrtpAes128CmSha1_80); + TestProtectRtcp(kSrtpAes128CmSha1_80); rtcp_packet_.MutableData()[1] = 0x34; EXPECT_FALSE(s2_.UnprotectRtcp(rtcp_packet_)); EXPECT_METRIC_THAT( - webrtc::metrics::Samples("WebRTC.PeerConnection.SrtcpUnprotectError"), + metrics::Samples("WebRTC.PeerConnection.SrtcpUnprotectError"), ElementsAre(Pair(srtp_err_status_auth_fail, 1))); } // Test that we fail to unprotect if the payloads are not authenticated. TEST_F(SrtpSessionTest, TestUnencryptReject) { - EXPECT_TRUE(s1_.SetSend(webrtc::kSrtpAes128CmSha1_80, webrtc::kTestKey1, + EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kEncryptedHeaderExtensionIds)); - EXPECT_TRUE(s2_.SetReceive(webrtc::kSrtpAes128CmSha1_80, webrtc::kTestKey1, + EXPECT_TRUE(s2_.SetReceive(kSrtpAes128CmSha1_80, kTestKey1, kEncryptedHeaderExtensionIds)); EXPECT_FALSE(s2_.UnprotectRtp(rtp_packet_)); EXPECT_METRIC_THAT( - webrtc::metrics::Samples("WebRTC.PeerConnection.SrtpUnprotectError"), + metrics::Samples("WebRTC.PeerConnection.SrtpUnprotectError"), ElementsAre(Pair(srtp_err_status_auth_fail, 1))); EXPECT_FALSE(s2_.UnprotectRtcp(rtcp_packet_)); EXPECT_METRIC_THAT( - webrtc::metrics::Samples("WebRTC.PeerConnection.SrtcpUnprotectError"), + metrics::Samples("WebRTC.PeerConnection.SrtcpUnprotectError"), ElementsAre(Pair(srtp_err_status_cant_check, 1))); } // Test that we fail when using buffers that are too small. TEST_F(SrtpSessionTest, TestBuffersTooSmall) { - EXPECT_TRUE(s1_.SetSend(webrtc::kSrtpAes128CmSha1_80, webrtc::kTestKey1, + EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kEncryptedHeaderExtensionIds)); // This buffer does not have extra capacity which we treat as an error. CopyOnWriteBuffer rtp_packet(rtp_packet_.data(), rtp_packet_.size(), @@ -211,41 +208,41 @@ TEST_F(SrtpSessionTest, TestReplay) { static const uint16_t seqnum_small = 10; static const uint16_t replay_window = 1024; - EXPECT_TRUE(s1_.SetSend(webrtc::kSrtpAes128CmSha1_80, webrtc::kTestKey1, + EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kEncryptedHeaderExtensionIds)); - EXPECT_TRUE(s2_.SetReceive(webrtc::kSrtpAes128CmSha1_80, webrtc::kTestKey1, + EXPECT_TRUE(s2_.SetReceive(kSrtpAes128CmSha1_80, kTestKey1, kEncryptedHeaderExtensionIds)); // Initial sequence number. - webrtc::SetBE16(rtp_packet_.MutableData() + 2, seqnum_big); + SetBE16(rtp_packet_.MutableData() + 2, seqnum_big); EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_)); rtp_packet_.SetData(kPcmuFrame, sizeof(kPcmuFrame)); // Replay within the 1024 window should succeed. - webrtc::SetBE16(rtp_packet_.MutableData() + 2, - seqnum_big - replay_window + 1); + SetBE16(rtp_packet_.MutableData() + 2, + seqnum_big - replay_window + 1); EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_)); rtp_packet_.SetData(kPcmuFrame, sizeof(kPcmuFrame)); // Replay out side of the 1024 window should fail. - webrtc::SetBE16(rtp_packet_.MutableData() + 2, - seqnum_big - replay_window - 1); + SetBE16(rtp_packet_.MutableData() + 2, + seqnum_big - replay_window - 1); EXPECT_FALSE(s1_.ProtectRtp(rtp_packet_)); rtp_packet_.SetData(kPcmuFrame, sizeof(kPcmuFrame)); // Increment sequence number to a small number. - webrtc::SetBE16(rtp_packet_.MutableData() + 2, seqnum_small); + SetBE16(rtp_packet_.MutableData() + 2, seqnum_small); EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_)); // Replay around 0 but out side of the 1024 window should fail. - webrtc::SetBE16(rtp_packet_.MutableData() + 2, - kMaxSeqnum + seqnum_small - replay_window - 1); + SetBE16(rtp_packet_.MutableData() + 2, + kMaxSeqnum + seqnum_small - replay_window - 1); EXPECT_FALSE(s1_.ProtectRtp(rtp_packet_)); rtp_packet_.SetData(kPcmuFrame, sizeof(kPcmuFrame)); // Replay around 0 but within the 1024 window should succeed. for (uint16_t seqnum = 65000; seqnum < 65003; ++seqnum) { - webrtc::SetBE16(rtp_packet_.MutableData() + 2, seqnum); + SetBE16(rtp_packet_.MutableData() + 2, seqnum); EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_)); rtp_packet_.SetData(kPcmuFrame, sizeof(kPcmuFrame)); } @@ -255,14 +252,14 @@ TEST_F(SrtpSessionTest, TestReplay) { // without the fix, the loop above would keep incrementing local sequence // number in libsrtp, eventually the new sequence number would go out side // of the window. - webrtc::SetBE16(rtp_packet_.MutableData() + 2, seqnum_small + 1); + SetBE16(rtp_packet_.MutableData() + 2, seqnum_small + 1); EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_)); } TEST_F(SrtpSessionTest, RemoveSsrc) { - EXPECT_TRUE(s1_.SetSend(webrtc::kSrtpAes128CmSha1_80, webrtc::kTestKey1, + EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kEncryptedHeaderExtensionIds)); - EXPECT_TRUE(s2_.SetReceive(webrtc::kSrtpAes128CmSha1_80, webrtc::kTestKey1, + EXPECT_TRUE(s2_.SetReceive(kSrtpAes128CmSha1_80, kTestKey1, kEncryptedHeaderExtensionIds)); // Encrypt and decrypt the packet once. EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_)); @@ -294,9 +291,9 @@ TEST_F(SrtpSessionTest, ProtectUnprotectWrapAroundRocMismatch) { // failures when it wraps around with packet loss. Pick your starting // sequence number in the lower half of the range for robustness reasons, // see packet_sequencer.cc for the code doing so. - EXPECT_TRUE(s1_.SetSend(webrtc::kSrtpAes128CmSha1_80, webrtc::kTestKey1, + EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kEncryptedHeaderExtensionIds)); - EXPECT_TRUE(s2_.SetReceive(webrtc::kSrtpAes128CmSha1_80, webrtc::kTestKey1, + EXPECT_TRUE(s2_.SetReceive(kSrtpAes128CmSha1_80, kTestKey1, kEncryptedHeaderExtensionIds)); // Buffers include enough room for the 10 byte SRTP auth tag so we can // encrypt in place. @@ -349,9 +346,9 @@ TEST_F(SrtpSessionTest, ProtectUnprotectWrapAroundRocMismatch) { } TEST_F(SrtpSessionTest, ProtectGetPacketIndex) { - EXPECT_TRUE(s1_.SetSend(webrtc::kSrtpAes128CmSha1_80, webrtc::kTestKey1, + EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kEncryptedHeaderExtensionIds)); - EXPECT_TRUE(s2_.SetReceive(webrtc::kSrtpAes128CmSha1_80, webrtc::kTestKey1, + EXPECT_TRUE(s2_.SetReceive(kSrtpAes128CmSha1_80, kTestKey1, kEncryptedHeaderExtensionIds)); // Buffers include enough room for the 10 byte SRTP auth tag so we can // encrypt in place. diff --git a/pc/test/svc_e2e_tests.cc b/pc/test/svc_e2e_tests.cc index 60469c76b5..a5496822fb 100644 --- a/pc/test/svc_e2e_tests.cc +++ b/pc/test/svc_e2e_tests.cc @@ -60,12 +60,12 @@ using ::testing::Optional; using ::testing::UnitTest; using ::testing::Values; using ::testing::ValuesIn; -using ::webrtc::webrtc_pc_e2e::EmulatedSFUConfig; -using ::webrtc::webrtc_pc_e2e::PeerConfigurer; -using ::webrtc::webrtc_pc_e2e::RunParams; -using ::webrtc::webrtc_pc_e2e::ScreenShareConfig; -using ::webrtc::webrtc_pc_e2e::VideoCodecConfig; -using ::webrtc::webrtc_pc_e2e::VideoConfig; +using webrtc_pc_e2e::EmulatedSFUConfig; +using webrtc_pc_e2e::PeerConfigurer; +using webrtc_pc_e2e::RunParams; +using webrtc_pc_e2e::ScreenShareConfig; +using webrtc_pc_e2e::VideoCodecConfig; +using webrtc_pc_e2e::VideoConfig; std::unique_ptr CreateTestFixture(absl::string_view test_case_name, diff --git a/pc/track_media_info_map.cc b/pc/track_media_info_map.cc index a1eef98fe7..6cd24448bb 100644 --- a/pc/track_media_info_map.cc +++ b/pc/track_media_info_map.cc @@ -58,7 +58,7 @@ void GetAudioAndVideoTrackBySsrc( RTC_DCHECK(remote_audio_track_by_ssrc->empty()); RTC_DCHECK(remote_video_track_by_ssrc->empty()); for (const auto& rtp_sender : rtp_senders) { - webrtc::MediaType media_type = rtp_sender->media_type(); + MediaType media_type = rtp_sender->media_type(); MediaStreamTrackInterface* track = rtp_sender->track().get(); if (!track) { continue; @@ -66,7 +66,7 @@ void GetAudioAndVideoTrackBySsrc( // TODO(deadbeef): `ssrc` should be removed in favor of `GetParameters`. uint32_t ssrc = rtp_sender->ssrc(); if (ssrc != 0) { - if (media_type == webrtc::MediaType::AUDIO) { + if (media_type == MediaType::AUDIO) { RTC_DCHECK(local_audio_track_by_ssrc->find(ssrc) == local_audio_track_by_ssrc->end()); (*local_audio_track_by_ssrc)[ssrc] = @@ -80,21 +80,21 @@ void GetAudioAndVideoTrackBySsrc( } } for (const auto& rtp_receiver : rtp_receivers) { - webrtc::MediaType media_type = rtp_receiver->media_type(); + MediaType media_type = rtp_receiver->media_type(); MediaStreamTrackInterface* track = rtp_receiver->track().get(); RTC_DCHECK(track); RtpParameters params = rtp_receiver->GetParameters(); for (const RtpEncodingParameters& encoding : params.encodings) { if (!encoding.ssrc) { - if (media_type == webrtc::MediaType::AUDIO) { + if (media_type == MediaType::AUDIO) { *unsignaled_audio_track = static_cast(track); } else { - RTC_DCHECK(media_type == webrtc::MediaType::VIDEO); + RTC_DCHECK(media_type == MediaType::VIDEO); *unsignaled_video_track = static_cast(track); } continue; } - if (media_type == webrtc::MediaType::AUDIO) { + if (media_type == MediaType::AUDIO) { RTC_DCHECK(remote_audio_track_by_ssrc->find(*encoding.ssrc) == remote_audio_track_by_ssrc->end()); (*remote_audio_track_by_ssrc)[*encoding.ssrc] = diff --git a/pc/track_media_info_map_unittest.cc b/pc/track_media_info_map_unittest.cc index a9487cc507..cd6bf36a2e 100644 --- a/pc/track_media_info_map_unittest.cc +++ b/pc/track_media_info_map_unittest.cc @@ -55,7 +55,7 @@ RtpParameters CreateRtpParametersWithSsrcs( } scoped_refptr CreateMockRtpSender( - webrtc::MediaType media_type, + MediaType media_type, std::initializer_list ssrcs, scoped_refptr track) { uint32_t first_ssrc; @@ -77,7 +77,7 @@ scoped_refptr CreateMockRtpSender( } scoped_refptr CreateMockRtpReceiver( - webrtc::MediaType media_type, + MediaType media_type, std::initializer_list ssrcs, scoped_refptr track) { auto receiver = make_ref_counted(); @@ -121,8 +121,8 @@ class TrackMediaInfoMapTest : public ::testing::Test { MediaStreamTrackInterface* local_track) { scoped_refptr rtp_sender = CreateMockRtpSender( local_track->kind() == MediaStreamTrackInterface::kAudioKind - ? webrtc::MediaType::AUDIO - : webrtc::MediaType::VIDEO, + ? MediaType::AUDIO + : MediaType::VIDEO, ssrcs, scoped_refptr(local_track)); rtp_senders_.push_back(rtp_sender); @@ -150,8 +150,8 @@ class TrackMediaInfoMapTest : public ::testing::Test { MediaStreamTrackInterface* remote_track) { auto rtp_receiver = CreateMockRtpReceiver( remote_track->kind() == MediaStreamTrackInterface::kAudioKind - ? webrtc::MediaType::AUDIO - : webrtc::MediaType::VIDEO, + ? MediaType::AUDIO + : MediaType::VIDEO, ssrcs, scoped_refptr(remote_track)); rtp_receivers_.push_back(rtp_receiver); diff --git a/test/peer_scenario/peer_scenario_client.cc b/test/peer_scenario/peer_scenario_client.cc index 2c688e0129..f5a207c4e9 100644 --- a/test/peer_scenario/peer_scenario_client.cc +++ b/test/peer_scenario/peer_scenario_client.cc @@ -210,11 +210,10 @@ class LambdaSetRemoteDescriptionObserver class FakeVideoEncoderFactory : public VideoEncoderFactory { public: std::vector GetSupportedFormats() const override { - const absl::InlinedVector - kSupportedScalabilityModes = {webrtc::ScalabilityMode::kL1T1, - webrtc::ScalabilityMode::kL1T2, - webrtc::ScalabilityMode::kL1T3}; + const absl::InlinedVector + kSupportedScalabilityModes = {ScalabilityMode::kL1T1, + ScalabilityMode::kL1T2, + ScalabilityMode::kL1T3}; return {SdpVideoFormat(kVp8CodecName, {}, kSupportedScalabilityModes)}; } std::unique_ptr Create(const Environment& env, diff --git a/test/peer_scenario/scenario_connection.cc b/test/peer_scenario/scenario_connection.cc index 9e35696ad1..bdbba3b9f0 100644 --- a/test/peer_scenario/scenario_connection.cc +++ b/test/peer_scenario/scenario_connection.cc @@ -113,7 +113,7 @@ class ScenarioIceConnectionImpl : public ScenarioIceConnection, std::unique_ptr ScenarioIceConnection::Create( const Environment& env, - webrtc::test::NetworkEmulationManagerImpl* net, + test::NetworkEmulationManagerImpl* net, IceConnectionObserver* observer) { return std::make_unique(env, net, observer); } @@ -155,7 +155,7 @@ ScenarioIceConnectionImpl::ScenarioIceConnectionImpl( port_allocator_->Initialize(); RTC_CHECK(port_allocator_->SetConfiguration(/*stun_servers*/ {}, /*turn_servers*/ {}, 0, - webrtc::NO_PRUNE)); + NO_PRUNE)); jsep_controller_->SetLocalCertificate(certificate_); }); } @@ -208,7 +208,7 @@ void ScenarioIceConnectionImpl::SendRtcpPacket( void ScenarioIceConnectionImpl::SetRemoteSdp(SdpType type, const std::string& remote_sdp) { RTC_DCHECK_RUN_ON(signaling_thread_); - remote_description_ = webrtc::CreateSessionDescription(type, remote_sdp); + remote_description_ = CreateSessionDescription(type, remote_sdp); jsep_controller_->SubscribeIceCandidateGathered( [this](const std::string& transport, const std::vector& candidate) { @@ -237,7 +237,7 @@ void ScenarioIceConnectionImpl::SetRemoteSdp(SdpType type, void ScenarioIceConnectionImpl::SetLocalSdp(SdpType type, const std::string& local_sdp) { RTC_DCHECK_RUN_ON(signaling_thread_); - local_description_ = webrtc::CreateSessionDescription(type, local_sdp); + local_description_ = CreateSessionDescription(type, local_sdp); auto res = jsep_controller_->SetLocalDescription( local_description_->GetType(), local_description_->description(), remote_description_ ? remote_description_->description() : nullptr); diff --git a/test/peer_scenario/tests/bwe_ramp_up_test.cc b/test/peer_scenario/tests/bwe_ramp_up_test.cc index ddd9d15bad..b8cee3e051 100644 --- a/test/peer_scenario/tests/bwe_ramp_up_test.cc +++ b/test/peer_scenario/tests/bwe_ramp_up_test.cc @@ -52,8 +52,7 @@ using ::testing::WithParamInterface; scoped_refptr GetStatsAndProcess( PeerScenario& s, PeerScenarioClient* client) { - auto stats_collector = - make_ref_counted(); + auto stats_collector = make_ref_counted(); client->pc()->GetStats(stats_collector.get()); s.ProcessMessages(TimeDelta::Millis(0)); RTC_CHECK(stats_collector->called()); @@ -214,13 +213,13 @@ INSTANTIATE_TEST_SUITE_P( .expected_bwe_min = DataRate::KilobitsPerSec(2500), }, { - .network_capacity = webrtc::DataRate::KilobitsPerSec(500), - .expected_bwe_min = webrtc::DataRate::KilobitsPerSec(400), + .network_capacity = DataRate::KilobitsPerSec(500), + .expected_bwe_min = DataRate::KilobitsPerSec(400), }})); class MockRtpSenderObserver : public RtpSenderObserverInterface { public: - MOCK_METHOD(void, OnFirstPacketSent, (webrtc::MediaType)); + MOCK_METHOD(void, OnFirstPacketSent, (MediaType)); }; // Test that caller and callee BWE rampup even if no media packets are sent. @@ -233,7 +232,7 @@ TEST_P(BweRampupWithInitialProbeTest, BweRampUpBothDirectionsWithoutMedia) { PeerScenarioClient* caller = s.CreateClient({}); PeerScenarioClient* callee = s.CreateClient({}); - auto transceiver = caller->pc()->AddTransceiver(webrtc::MediaType::VIDEO); + auto transceiver = caller->pc()->AddTransceiver(MediaType::VIDEO); ASSERT_TRUE(transceiver.error().ok()); MockRtpSenderObserver observer; @@ -326,7 +325,7 @@ TEST(BweRampupTest, CanReconfigureBweAfterStopingVideo) { // Send a TCP messages to the receiver using the same downlink node. // This is done just to force a lower BWE than the link capacity. - webrtc::TcpMessageRoute* tcp_route = s.net()->CreateTcpRoute( + TcpMessageRoute* tcp_route = s.net()->CreateTcpRoute( s.net()->CreateRoute({caller_node}), s.net()->CreateRoute({callee_node})); DataRate bwe_before_restart = DataRate::Zero(); diff --git a/test/peer_scenario/tests/l4s_test.cc b/test/peer_scenario/tests/l4s_test.cc index 3ac13e0005..b68fba2658 100644 --- a/test/peer_scenario/tests/l4s_test.cc +++ b/test/peer_scenario/tests/l4s_test.cc @@ -94,8 +94,7 @@ class RtcpFeedbackCounter { scoped_refptr GetStatsAndProcess( PeerScenario& s, PeerScenarioClient* client) { - auto stats_collector = - make_ref_counted(); + auto stats_collector = make_ref_counted(); client->pc()->GetStats(stats_collector.get()); s.ProcessMessages(TimeDelta::Millis(0)); RTC_CHECK(stats_collector->called()); From 15e3558219ec0cff2de1cf6d23fd9da89b954004 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Fri, 9 May 2025 05:02:06 -0700 Subject: [PATCH 105/292] Roll chromium_revision efebdb6499..4852d4ffee (1457935:1458067) Change log: https://chromium.googlesource.com/chromium/src/+log/efebdb6499..4852d4ffee Full diff: https://chromium.googlesource.com/chromium/src/+/efebdb6499..4852d4ffee Changed dependencies * src/build: https://chromium.googlesource.com/chromium/src/build/+log/009c5fbcdd..f94ef0c502 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/62c918c015..39709f3763 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/4694d84c9b..7461945a13 * src/third_party/androidx/cipd: 3f6bb_ds1Ab_INPp6lc8i778IRbWZSwVm2QBzfQFTMoC..6oMBB-fWfNxprryb8ZyOcOKsrHEOYQzxe7DaKSjHx7cC * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/f371c85a8d..45cab558a8 * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/74f545d009..168822b896 * src/third_party/libunwind/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git/+log/81e2cb40a7..4acde988c5 * src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/e3e030ec6e..61d7ed3011 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/40cdeeca2e..e2cf09bef7 DEPS diff: https://chromium.googlesource.com/chromium/src/+/efebdb6499..4852d4ffee/DEPS No update to Clang. BUG=None Change-Id: I533b4417c933fe09c3eb692f71594235ae84dcb2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390543 Commit-Queue: Autoroller Bot-Commit: Autoroller Cr-Commit-Position: refs/heads/main@{#44577} --- DEPS | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/DEPS b/DEPS index 6588ed5c3b..b06ee5156e 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': 'efebdb649977730f38af9c17a2a4e53e22f02ee9', + 'chromium_revision': '4852d4ffeefa31265675175f00edcf84c5e30ecd', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -75,7 +75,7 @@ deps = { 'src/base': 'https://chromium.googlesource.com/chromium/src/base@07cde36d11f84e1ca905028b7b96ab273fe76520', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@009c5fbcdd84d02ac6099abfa9c59efd4aa6391b', + 'https://chromium.googlesource.com/chromium/src/build@f94ef0c5025b01d7e3fc5b17d5458e89791c89a9', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@077a66f30fcf281b066fafb6dfc60818c238efb6', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -88,9 +88,9 @@ deps = { 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@62c918c0155b79bb0edc172b6e41396e143e22a8', + 'https://chromium.googlesource.com/chromium/src/testing@39709f37638c9e666908dcb05a40fa8a34cda350', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@4694d84c9bd8f526078f36d441053bfe6373214f', + 'https://chromium.googlesource.com/chromium/src/third_party@7461945a13bd879828bbd4800e4f4fecd9ccf8ea', 'src/buildtools/linux64': { 'packages': [ @@ -329,9 +329,9 @@ deps = { 'src/third_party/libc++abi/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@7ff13dd8c66d5047a3ee1e1ba26ddeb70d3d8740', 'src/third_party/llvm-libc/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@e3e030ec6ee1674bf2195d0cfd0a4bf5fee16537', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@61d7ed30110f97d6842304bc9035ee9cdae6b5e5', 'src/third_party/libunwind/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@81e2cb40a70de2b6978e6d8658891ded9a77f7e3', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@4acde988c52e6a325e3ce2ac761284f295f2ab39', 'src/third_party/test_fonts/test_fonts': { 'dep_type': 'gcs', @@ -480,11 +480,11 @@ deps = { }, 'src/third_party/boringssl/src': - 'https://boringssl.googlesource.com/boringssl.git@f371c85a8d58bf911cba6073bcca6d497af81dc9', + 'https://boringssl.googlesource.com/boringssl.git@45cab558a838e1a546391406e7cbe1eec9b6e643', 'src/third_party/breakpad/breakpad': 'https://chromium.googlesource.com/breakpad/breakpad.git@b319c12317824bfd0e92828db20bbfe0ec6e31f8', 'src/third_party/catapult': - 'https://chromium.googlesource.com/catapult.git@74f545d009ceef464f840938ae079ffbc112a102', + 'https://chromium.googlesource.com/catapult.git@168822b896c38f7d0079a2483c81f3d5cbc1d933', 'src/third_party/ced/src': { 'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5', }, @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@40cdeeca2ea585f4a1057b559081baa457e10c99', + 'https://chromium.googlesource.com/chromium/src/tools@e2cf09bef7cbd4f461768d61f2758b4eabd374e7', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': '3f6bb_ds1Ab_INPp6lc8i778IRbWZSwVm2QBzfQFTMoC', + 'version': '6oMBB-fWfNxprryb8ZyOcOKsrHEOYQzxe7DaKSjHx7cC', }, ], 'condition': 'checkout_android and non_git_source', From 564364f174acfdc83916262c818b4007735bea0b Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Fri, 9 May 2025 11:02:38 -0700 Subject: [PATCH 106/292] Roll chromium_revision 4852d4ffee..5eae54daf4 (1458067:1458214) Change log: https://chromium.googlesource.com/chromium/src/+log/4852d4ffee..5eae54daf4 Full diff: https://chromium.googlesource.com/chromium/src/+/4852d4ffee..5eae54daf4 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/07cde36d11..41a37104fc * src/build: https://chromium.googlesource.com/chromium/src/build/+log/f94ef0c502..b121fee0f3 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/8d03237122..0102327db9 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/39709f3763..0ac0603693 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/7461945a13..682251224b * src/third_party/androidx/cipd: 6oMBB-fWfNxprryb8ZyOcOKsrHEOYQzxe7DaKSjHx7cC..HYTISQQTMh5XcYAUlGBAU6HIXb_FD0kMzljZh0v03j0C * src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/2fdf452fd1..948a46574c * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/a7571b1596..d1f9fa6c92 * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/b58c61966d..1fde64ca4b * src/third_party/llvm-build/Release+Asserts: Linux_x64/clang-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Linux_x64/clang-tidy-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Linux_x64/clangd-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Mac/clang-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Mac/clang-tidy-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Mac/clangd-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Mac/llvm-code-coverage-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Mac_arm64/clang-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Mac_arm64/clangd-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Win/clang-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Win/clang-tidy-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Win/clang-win-runtime-library-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Win/clangd-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Win/llvm-code-coverage-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz,Win/llvmobjdump-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz..Linux_x64/clang-llvmorg-21-init-9266-g09006611-2.tar.xz,Linux_x64/clang-tidy-llvmorg-21-init-9266-g09006611-2.tar.xz,Linux_x64/clangd-llvmorg-21-init-9266-g09006611-2.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-2.tar.xz,Linux_x64/llvmobjdump-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac/clang-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac/clang-mac-runtime-library-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac/clang-tidy-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac/clangd-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac/llvm-code-coverage-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac_arm64/clang-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac_arm64/clang-tidy-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac_arm64/clangd-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-2.tar.xz,Win/clang-llvmorg-21-init-9266-g09006611-2.tar.xz,Win/clang-tidy-llvmorg-21-init-9266-g09006611-2.tar.xz,Win/clang-win-runtime-library-llvmorg-21-init-9266-g09006611-2.tar.xz,Win/clangd-llvmorg-21-init-9266-g09006611-2.tar.xz,Win/llvm-code-coverage-llvmorg-21-init-9266-g09006611-2.tar.xz,Win/llvmobjdump-llvmorg-21-init-9266-g09006611-2.tar.xz * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/993d41d0c3..f2e70843f7 * src/third_party/rust-toolchain: Linux_x64/rust-toolchain-4a0969e06dbeaaa43914d2d00b2e843d49aa3886-1-llvmorg-21-init-10851-gcd6c4b61.tar.xz,Mac/rust-toolchain-4a0969e06dbeaaa43914d2d00b2e843d49aa3886-1-llvmorg-21-init-10851-gcd6c4b61.tar.xz,Mac_arm64/rust-toolchain-4a0969e06dbeaaa43914d2d00b2e843d49aa3886-1-llvmorg-21-init-10851-gcd6c4b61.tar.xz,Win/rust-toolchain-4a0969e06dbeaaa43914d2d00b2e843d49aa3886-1-llvmorg-21-init-10851-gcd6c4b61.tar.xz..Linux_x64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Mac/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Mac_arm64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Win/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/e2cf09bef7..b130fc850c * src/tools/luci-go: git_revision:a9a10995e2889126ace91faf4052949e38c69d33..git_revision:4486e4bb411a38fa21ba426d9a89052402fe610f * src/tools/luci-go: git_revision:a9a10995e2889126ace91faf4052949e38c69d33..git_revision:4486e4bb411a38fa21ba426d9a89052402fe610f * src/tools/luci-go: git_revision:a9a10995e2889126ace91faf4052949e38c69d33..git_revision:4486e4bb411a38fa21ba426d9a89052402fe610f DEPS diff: https://chromium.googlesource.com/chromium/src/+/4852d4ffee..5eae54daf4/DEPS Clang version changed llvmorg-21-init-10851-gcd6c4b61:llvmorg-21-init-9266-g09006611 Details: https://chromium.googlesource.com/chromium/src/+/4852d4ffee..5eae54daf4/tools/clang/scripts/update.py BUG=None Change-Id: Icd16a6ebf880561b81fc60f98d276022e0dc8867 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390507 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44578} --- DEPS | 222 +++++++++++++++++++++++++++++------------------------------ 1 file changed, 111 insertions(+), 111 deletions(-) diff --git a/DEPS b/DEPS index b06ee5156e..a8357e008b 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '4852d4ffeefa31265675175f00edcf84c5e30ecd', + 'chromium_revision': '5eae54daf4571ee507e01d0b0708ee55c85c0d51', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@07cde36d11f84e1ca905028b7b96ab273fe76520', + 'https://chromium.googlesource.com/chromium/src/base@41a37104fc3634509a7289f21a5de7bd59499fe6', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@f94ef0c5025b01d7e3fc5b17d5458e89791c89a9', + 'https://chromium.googlesource.com/chromium/src/build@b121fee0f3208b9e5d23ed536e2f9b7474682e86', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@077a66f30fcf281b066fafb6dfc60818c238efb6', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@8d03237122a4d1490aee8079b839feb54ec8cda8', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@0102327db9c59f933348504fe5a1259100c56b10', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@39709f37638c9e666908dcb05a40fa8a34cda350', + 'https://chromium.googlesource.com/chromium/src/testing@0ac0603693a53de2c76f4210f84105d4db1f8037', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@7461945a13bd879828bbd4800e4f4fecd9ccf8ea', + 'https://chromium.googlesource.com/chromium/src/third_party@682251224b464c210978d21c077de02d7524217c', 'src/buildtools/linux64': { 'packages': [ @@ -142,143 +142,143 @@ deps = { 'objects': [ { # The Android libclang_rt.builtins libraries are currently only included in the Linux clang package. - 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'sha256sum': 'e5e5a685fde6a4ddd211913ccf6f8e62ac7e6567fec26d68d5bff3aeac42c330', - 'size_bytes': 2376076, - 'generation': 1746525671928280, + 'object_name': 'Win/llvm-code-coverage-llvmorg-21-init-9266-g09006611-2.tar.xz', + 'sha256sum': '1a5e31c5e69ea7b84fdfbff27024670cf2eec538c68fc4213a5bcf576263b0ca', + 'size_bytes': 2370540, + 'generation': 1746105761636379, 'condition': 'host_os == "win" and checkout_clang_coverage_tools', }, { - 'object_name': 'Win/llvmobjdump-llvmorg-21-init-10851-gcd6c4b61-1.tar.xz', - 'sha256sum': 'd02d624d3ecfe5dbaa2e93b2da5a6443e4f25b9fe2fab43b8b540fccc91e4770', - 'size_bytes': 5687508, - 'generation': 1746525671637063, + 'object_name': 'Win/llvmobjdump-llvmorg-21-init-9266-g09006611-2.tar.xz', + 'sha256sum': '3ac42a8e816d1fa89594bfe882d033b6c351551b055b9d78867a2b3d24ac2451', + 'size_bytes': 5698696, + 'generation': 1746105761558030, 'condition': '(checkout_linux or checkout_mac or checkout_android) and host_os == "win"', }, ] @@ -290,31 +290,31 @@ deps = { 'bucket': 'chromium-browser-clang', 'objects': [ { - 'object_name': 'Linux_x64/rust-toolchain-4a0969e06dbeaaa43914d2d00b2e843d49aa3886-1-llvmorg-21-init-10851-gcd6c4b61.tar.xz', - 'sha256sum': '2d8498b5dcab2954c8212564798252c3edc9c0181ea8b05d3d6ae42b7b6adf7d', - 'size_bytes': 118439104, - 'generation': 1746525613682798, + 'object_name': 'Linux_x64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz', + 'sha256sum': '378c432f7739bb5da11aad7b3a2687f8252565eae5f0dcfc55c39a15382c519c', + 'size_bytes': 118598336, + 'generation': 1745271335898717, 'condition': 'host_os == "linux" and non_git_source', }, { - 'object_name': 'Mac/rust-toolchain-4a0969e06dbeaaa43914d2d00b2e843d49aa3886-1-llvmorg-21-init-10851-gcd6c4b61.tar.xz', - 'sha256sum': '93e57858344a978ce2df2d9926c541d8e25cd9def0054328dba4ac13ff633d34', - 'size_bytes': 111691112, - 'generation': 1746525615270255, + 'object_name': 'Mac/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz', + 'sha256sum': 'bf05c8b5e90d6904de02dca9b3e4cb5e45a1a56207e7af1fbb3a10707704a26a', + 'size_bytes': 111932536, + 'generation': 1745271337336068, 'condition': 'host_os == "mac" and host_cpu == "x64"', }, { - 'object_name': 'Mac_arm64/rust-toolchain-4a0969e06dbeaaa43914d2d00b2e843d49aa3886-1-llvmorg-21-init-10851-gcd6c4b61.tar.xz', - 'sha256sum': '4c1f9d7ff8c1febdb755ad971b0ae8d9b3c5bed5b30b96ea00caa784dfb86427', - 'size_bytes': 101778572, - 'generation': 1746525616855514, + 'object_name': 'Mac_arm64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz', + 'sha256sum': '1aec99f479ff28cefe44ed739844833e016a1da255cf3c17d79e59a273246615', + 'size_bytes': 101605468, + 'generation': 1745271339727037, 'condition': 'host_os == "mac" and host_cpu == "arm64"', }, { - 'object_name': 'Win/rust-toolchain-4a0969e06dbeaaa43914d2d00b2e843d49aa3886-1-llvmorg-21-init-10851-gcd6c4b61.tar.xz', - 'sha256sum': '7d2484e229f4015d3711f1294d11048764ec22ebaf8266701abf350cd51c897b', - 'size_bytes': 193318196, - 'generation': 1746525618539172, + 'object_name': 'Win/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz', + 'sha256sum': 'b291520613a3ebc415e4576a7fa31d840a5ebf4ab9be6e9dc5d90062dc001c1e', + 'size_bytes': 193280372, + 'generation': 1745271341223097, 'condition': 'host_os == "win"', }, ], @@ -323,7 +323,7 @@ deps = { 'src/third_party/clang-format/script': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@37f6e68a107df43b7d7e044fd36a13cbae3413f2', 'src/third_party/compiler-rt/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@2fdf452fd1b09f2e0bb9fdfe8a162c86cc70dbc2', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@948a46574ce380c306ae4f01d2a630d93847b88f', 'src/third_party/libc++/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@a01c02c9d4acbdae3b7e8a2f3ee58579a9c29f96', 'src/third_party/libc++abi/src': @@ -497,13 +497,13 @@ deps = { 'src/third_party/crc32c/src': 'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558', 'src/third_party/depot_tools': - 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@a7571b1596305d6a6fe26c788f24e40c579fcda8', + 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@d1f9fa6c922b20d8034fe4c7f3a62f8a824c561b', 'src/third_party/ffmpeg': 'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@dcdd0fa51b65a0b1688ff6b8f0cc81908f09ded2', 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@b58c61966d84dfefbf75f8fb0d2fadae948ac4db', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@1fde64ca4bf841e94ca2498c504a48620ea7784c', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '993d41d0c39335e35fe577808e8f704e58d9dbfb', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'f2e70843f73b1e17fea3de511344f65b4bc8ad9c', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@e2cf09bef7cbd4f461768d61f2758b4eabd374e7', + 'https://chromium.googlesource.com/chromium/src/tools@b130fc850c78da6b362d869774d22bf118899f5b', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': '6oMBB-fWfNxprryb8ZyOcOKsrHEOYQzxe7DaKSjHx7cC', + 'version': 'HYTISQQTMh5XcYAUlGBAU6HIXb_FD0kMzljZh0v03j0C', }, ], 'condition': 'checkout_android and non_git_source', @@ -793,15 +793,15 @@ deps = { 'packages': [ { 'package': 'infra/tools/luci/cas/${{platform}}', - 'version': 'git_revision:a9a10995e2889126ace91faf4052949e38c69d33', + 'version': 'git_revision:4486e4bb411a38fa21ba426d9a89052402fe610f', }, { 'package': 'infra/tools/luci/isolate/${{platform}}', - 'version': 'git_revision:a9a10995e2889126ace91faf4052949e38c69d33', + 'version': 'git_revision:4486e4bb411a38fa21ba426d9a89052402fe610f', }, { 'package': 'infra/tools/luci/swarming/${{platform}}', - 'version': 'git_revision:a9a10995e2889126ace91faf4052949e38c69d33', + 'version': 'git_revision:4486e4bb411a38fa21ba426d9a89052402fe610f', } ], 'dep_type': 'cipd', From c42f165cf4e7917405c017a5b937f9b4bae74733 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Fri, 9 May 2025 15:02:10 -0700 Subject: [PATCH 107/292] Roll chromium_revision 5eae54daf4..a48aecb5d3 (1458214:1458369) Change log: https://chromium.googlesource.com/chromium/src/+log/5eae54daf4..a48aecb5d3 Full diff: https://chromium.googlesource.com/chromium/src/+/5eae54daf4..a48aecb5d3 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/41a37104fc..994477767d * src/build: https://chromium.googlesource.com/chromium/src/build/+log/b121fee0f3..92d42d9ad2 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/0102327db9..745c72082f * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/0ac0603693..dd165a40b2 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/682251224b..0bc55e4159 * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/168822b896..5255e1a11a * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/1fde64ca4b..61c101f683 * src/third_party/libc++abi/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git/+log/7ff13dd8c6..1efb5e6d7c * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/f2e70843f7..829564494c * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/b130fc850c..b5cffc6be3 DEPS diff: https://chromium.googlesource.com/chromium/src/+/5eae54daf4..a48aecb5d3/DEPS No update to Clang. BUG=None Change-Id: I7e915f3337bcd89a27eb6cee6244e17edd9d7dc7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390546 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44579} --- DEPS | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/DEPS b/DEPS index a8357e008b..bbbb9fb024 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '5eae54daf4571ee507e01d0b0708ee55c85c0d51', + 'chromium_revision': 'a48aecb5d32aca77ba5e0e1881ca2e3c2100e430', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@41a37104fc3634509a7289f21a5de7bd59499fe6', + 'https://chromium.googlesource.com/chromium/src/base@994477767d99fcf1e3b34a25ae523c7eb7fd608a', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@b121fee0f3208b9e5d23ed536e2f9b7474682e86', + 'https://chromium.googlesource.com/chromium/src/build@92d42d9ad23dbee90f88c54bd7c53f347cb7040a', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@077a66f30fcf281b066fafb6dfc60818c238efb6', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@0102327db9c59f933348504fe5a1259100c56b10', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@745c72082f1477a061d5e13f17dd346763d28998', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@0ac0603693a53de2c76f4210f84105d4db1f8037', + 'https://chromium.googlesource.com/chromium/src/testing@dd165a40b256cfe9418d2bad7ef850f160300980', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@682251224b464c210978d21c077de02d7524217c', + 'https://chromium.googlesource.com/chromium/src/third_party@0bc55e4159bac499a909f9f27b4b0db2d8a946bb', 'src/buildtools/linux64': { 'packages': [ @@ -327,7 +327,7 @@ deps = { 'src/third_party/libc++/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@a01c02c9d4acbdae3b7e8a2f3ee58579a9c29f96', 'src/third_party/libc++abi/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@7ff13dd8c66d5047a3ee1e1ba26ddeb70d3d8740', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@1efb5e6d7c5eee01624f3730a935285405c9cd22', 'src/third_party/llvm-libc/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@61d7ed30110f97d6842304bc9035ee9cdae6b5e5', 'src/third_party/libunwind/src': @@ -484,7 +484,7 @@ deps = { 'src/third_party/breakpad/breakpad': 'https://chromium.googlesource.com/breakpad/breakpad.git@b319c12317824bfd0e92828db20bbfe0ec6e31f8', 'src/third_party/catapult': - 'https://chromium.googlesource.com/catapult.git@168822b896c38f7d0079a2483c81f3d5cbc1d933', + 'https://chromium.googlesource.com/catapult.git@5255e1a11a4d0bf747f304816a2523f952be9228', 'src/third_party/ced/src': { 'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5', }, @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@1fde64ca4bf841e94ca2498c504a48620ea7784c', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@61c101f683f673a7917430bfee10f9e667b3a546', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'f2e70843f73b1e17fea3de511344f65b4bc8ad9c', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '829564494ca9d7bc1c8b15998c5f1f3536cdacf8', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@b130fc850c78da6b362d869774d22bf118899f5b', + 'https://chromium.googlesource.com/chromium/src/tools@b5cffc6be310bf444947c20cd714e299e4923568', 'src/third_party/espresso': { 'packages': [ From 28f489e5a2b0c551ba1464697921666ae60bcb2c Mon Sep 17 00:00:00 2001 From: webrtc-version-updater Date: Fri, 9 May 2025 21:08:51 -0700 Subject: [PATCH 108/292] Update WebRTC code version (2025-05-10T04:08:17). Bug: None Change-Id: I6bf17e56b03c30157fe43b166a1ba58267f9fd12 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390548 Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Cr-Commit-Position: refs/heads/main@{#44580} --- call/version.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/call/version.cc b/call/version.cc index 210d105d53..156e478d32 100644 --- a/call/version.cc +++ b/call/version.cc @@ -13,7 +13,7 @@ namespace webrtc { // The timestamp is always in UTC. -const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-09T04:02:12"; +const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-10T04:08:17"; void LoadWebRTCVersionInRegister() { // Using volatile to instruct the compiler to not optimize `p` away even From 4d0b18c5c428aa610bd121442f3c741a3cd563b3 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Sat, 10 May 2025 01:11:25 -0700 Subject: [PATCH 109/292] Roll chromium_revision a48aecb5d3..4effae884f (1458369:1458473) Change log: https://chromium.googlesource.com/chromium/src/+log/a48aecb5d3..4effae884f Full diff: https://chromium.googlesource.com/chromium/src/+/a48aecb5d3..4effae884f Changed dependencies * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/745c72082f..b128eca85f * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/dd165a40b2..ab12b4b4c9 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/0bc55e4159..30c00d0f4d * src/third_party/androidx/cipd: HYTISQQTMh5XcYAUlGBAU6HIXb_FD0kMzljZh0v03j0C..1jSxEkUFGSfsQTosCWjXFmS0h3Lk_BRPBBLwBNosOVYC * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/45cab558a8..136284f854 * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/5255e1a11a..0c5acc073d * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/d1f9fa6c92..977c37458f * src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/61d7ed3011..4a2940b40b * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/829564494c..6ffe623d28 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/b5cffc6be3..b9d3fbbda7 DEPS diff: https://chromium.googlesource.com/chromium/src/+/a48aecb5d3..4effae884f/DEPS No update to Clang. BUG=None Change-Id: I061cf66c2b7e72ddfc5a4e783d9db6ed380e1b42 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390549 Commit-Queue: Autoroller Bot-Commit: Autoroller Cr-Commit-Position: refs/heads/main@{#44581} --- DEPS | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/DEPS b/DEPS index bbbb9fb024..a067a94538 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': 'a48aecb5d32aca77ba5e0e1881ca2e3c2100e430', + 'chromium_revision': '4effae884f53c80ac1b2a41c2b3aa6b27b0a31a3', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@745c72082f1477a061d5e13f17dd346763d28998', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@b128eca85f82e669877c6df05d834456d225f3f4', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@dd165a40b256cfe9418d2bad7ef850f160300980', + 'https://chromium.googlesource.com/chromium/src/testing@ab12b4b4c9bf4ea505ef5d0f4ed63dbf10cead00', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@0bc55e4159bac499a909f9f27b4b0db2d8a946bb', + 'https://chromium.googlesource.com/chromium/src/third_party@30c00d0f4dae87cb4709312008d098a98de9ede5', 'src/buildtools/linux64': { 'packages': [ @@ -329,7 +329,7 @@ deps = { 'src/third_party/libc++abi/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@1efb5e6d7c5eee01624f3730a935285405c9cd22', 'src/third_party/llvm-libc/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@61d7ed30110f97d6842304bc9035ee9cdae6b5e5', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@4a2940b40b394ca57312aa9bbc8af430fe9a5340', 'src/third_party/libunwind/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@4acde988c52e6a325e3ce2ac761284f295f2ab39', @@ -480,11 +480,11 @@ deps = { }, 'src/third_party/boringssl/src': - 'https://boringssl.googlesource.com/boringssl.git@45cab558a838e1a546391406e7cbe1eec9b6e643', + 'https://boringssl.googlesource.com/boringssl.git@136284f8548bc7fb43e99e7f69e03fab57168e8b', 'src/third_party/breakpad/breakpad': 'https://chromium.googlesource.com/breakpad/breakpad.git@b319c12317824bfd0e92828db20bbfe0ec6e31f8', 'src/third_party/catapult': - 'https://chromium.googlesource.com/catapult.git@5255e1a11a4d0bf747f304816a2523f952be9228', + 'https://chromium.googlesource.com/catapult.git@0c5acc073dd14893f2a75a51819ad81627098551', 'src/third_party/ced/src': { 'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5', }, @@ -497,7 +497,7 @@ deps = { 'src/third_party/crc32c/src': 'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558', 'src/third_party/depot_tools': - 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@d1f9fa6c922b20d8034fe4c7f3a62f8a824c561b', + 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@977c37458fda472d8822a8b57e4a83a7bc747471', 'src/third_party/ffmpeg': 'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@dcdd0fa51b65a0b1688ff6b8f0cc81908f09ded2', 'src/third_party/flatbuffers/src': @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '829564494ca9d7bc1c8b15998c5f1f3536cdacf8', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '6ffe623d283575c00047cf9abf90c3659505a592', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@b5cffc6be310bf444947c20cd714e299e4923568', + 'https://chromium.googlesource.com/chromium/src/tools@b9d3fbbda7b3606e4954991b008fa1e839f1ad90', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'HYTISQQTMh5XcYAUlGBAU6HIXb_FD0kMzljZh0v03j0C', + 'version': '1jSxEkUFGSfsQTosCWjXFmS0h3Lk_BRPBBLwBNosOVYC', }, ], 'condition': 'checkout_android and non_git_source', From c6e125e640f54ba73d5cebfca28b3c2cdf609e70 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Fri, 9 May 2025 10:36:09 +0000 Subject: [PATCH 110/292] [cleanup] Fix redundant webrtc name specifier This CL was uploaded by git cl split. R=hta@webrtc.org No-Iwyu: LSC Bug: webrtc:42232595 Change-Id: I1dbd586b71231c70733b666155705181021eb292 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390602 Auto-Submit: Evan Shrubsole Commit-Queue: Evan Shrubsole Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44582} --- media/sctp/dcsctp_transport.cc | 2 +- media/sctp/dcsctp_transport_unittest.cc | 11 +++++------ 2 files changed, 6 insertions(+), 7 deletions(-) diff --git a/media/sctp/dcsctp_transport.cc b/media/sctp/dcsctp_transport.cc index af2e6eaed3..529de89a41 100644 --- a/media/sctp/dcsctp_transport.cc +++ b/media/sctp/dcsctp_transport.cc @@ -706,7 +706,7 @@ void DcSctpTransport::OnTransportWritableState( } void DcSctpTransport::OnDtlsTransportState(DtlsTransportInternal* transport, - webrtc::DtlsTransportState state) { + DtlsTransportState state) { if (state == DtlsTransportState::kNew && socket_) { // IF DTLS restart (DtlsTransportState::kNew) // THEN diff --git a/media/sctp/dcsctp_transport_unittest.cc b/media/sctp/dcsctp_transport_unittest.cc index d629722078..fe16899f26 100644 --- a/media/sctp/dcsctp_transport_unittest.cc +++ b/media/sctp/dcsctp_transport_unittest.cc @@ -57,7 +57,7 @@ class MockDataChannelSink : public DataChannelSink { // DataChannelSink MOCK_METHOD(void, OnDataReceived, - (int, DataMessageType, const webrtc::CopyOnWriteBuffer&)); + (int, DataMessageType, const CopyOnWriteBuffer&)); MOCK_METHOD(void, OnChannelClosing, (int)); MOCK_METHOD(void, OnChannelClosed, (int)); MOCK_METHOD(void, OnReadyToSend, ()); @@ -82,7 +82,7 @@ class Peer { .Times(1) .WillOnce(Return(ByMove(std::move(socket_ptr)))); - sctp_transport_ = std::make_unique( + sctp_transport_ = std::make_unique( env_, Thread::Current(), &fake_dtls_transport_, std::move(mock_dcsctp_socket_factory)); sctp_transport_->SetDataChannelSink(&sink_); @@ -90,10 +90,10 @@ class Peer { } FakeDtlsTransport fake_dtls_transport_; - webrtc::SimulatedClock simulated_clock_; + SimulatedClock simulated_clock_; Environment env_; dcsctp::MockDcSctpSocket* socket_; - std::unique_ptr sctp_transport_; + std::unique_ptr sctp_transport_; NiceMock sink_; }; } // namespace @@ -290,8 +290,7 @@ TEST(DcSctpTransportTest, DeliversMessage) { AutoThread main_thread; Peer peer_a; - EXPECT_CALL(peer_a.sink_, - OnDataReceived(1, webrtc::DataMessageType::kBinary, _)) + EXPECT_CALL(peer_a.sink_, OnDataReceived(1, DataMessageType::kBinary, _)) .Times(1); peer_a.sctp_transport_->OpenStream(1, kDefaultPriority); From f31651c42704d3c40e2a266cf87486b8e22aafc6 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Fri, 9 May 2025 10:30:51 +0000 Subject: [PATCH 111/292] [cleanup] Fix redundant webrtc name specifier This CL was uploaded by git cl split. R=hta@webrtc.org No-Iwyu: LSC Bug: webrtc:42232595 Change-Id: I49ffb81b392def7fb1eef3749e2dcf6e5f57bd06 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390600 Reviewed-by: Harald Alvestrand Commit-Queue: Evan Shrubsole Auto-Submit: Evan Shrubsole Cr-Commit-Position: refs/heads/main@{#44583} --- api/transport/field_trial_based_config.cc | 2 +- api/transport/stun.cc | 49 ++++---- api/transport/stun_unittest.cc | 20 ++-- .../data_channel_benchmark/grpc_signaling.cc | 26 ++--- .../peer_connection_client.cc | 61 +++++----- .../frame_analyzer/video_quality_analysis.cc | 4 +- .../video_quality_analysis_unittest.cc | 3 +- rtc_tools/network_tester/packet_sender.cc | 8 +- .../rtc_event_log_visualizer/analyzer.cc | 105 +++++++++--------- .../rtc_event_log_visualizer/plot_base.cc | 21 ++-- rtc_tools/rtp_generator/rtp_generator.cc | 35 +++--- rtc_tools/video_encoder/video_encoder.cc | 23 ++-- rtc_tools/video_file_reader_unittest.cc | 12 +- rtc_tools/video_file_writer_unittest.cc | 22 ++-- test/network/emulated_turn_server.cc | 10 +- test/network/fake_network_socket_server.cc | 2 +- test/network/feedback_generator.cc | 10 +- test/network/feedback_generator_unittest.cc | 4 +- test/network/network_emulation_pc_unittest.cc | 6 +- test/network/network_emulation_unittest.cc | 6 +- test/network/schedulable_network_behavior.cc | 28 ++--- 21 files changed, 214 insertions(+), 243 deletions(-) diff --git a/api/transport/field_trial_based_config.cc b/api/transport/field_trial_based_config.cc index ea3ce21512..81abd07411 100644 --- a/api/transport/field_trial_based_config.cc +++ b/api/transport/field_trial_based_config.cc @@ -16,6 +16,6 @@ namespace webrtc { std::string FieldTrialBasedConfig::GetValue(absl::string_view key) const { - return webrtc::field_trial::FindFullName(std::string(key)); + return field_trial::FindFullName(std::string(key)); } } // namespace webrtc diff --git a/api/transport/stun.cc b/api/transport/stun.cc index dd7c142829..b5e74e49df 100644 --- a/api/transport/stun.cc +++ b/api/transport/stun.cc @@ -381,7 +381,7 @@ bool StunMessage::ValidateMessageIntegrityOfType(int mi_attr_type, } // Getting the message length from the STUN header. - uint16_t msg_length = webrtc::GetBE16(&data[2]); + uint16_t msg_length = GetBE16(&data[2]); if (size != (msg_length + kStunHeaderSize)) { return false; } @@ -392,8 +392,8 @@ bool StunMessage::ValidateMessageIntegrityOfType(int mi_attr_type, while (current_pos + 4 <= size) { uint16_t attr_type, attr_length; // Getting attribute type and length. - attr_type = webrtc::GetBE16(&data[current_pos]); - attr_length = webrtc::GetBE16(&data[current_pos + sizeof(attr_type)]); + attr_type = GetBE16(&data[current_pos]); + attr_length = GetBE16(&data[current_pos + sizeof(attr_type)]); // If M-I, sanity check it, and break out. if (attr_type == mi_attr_type) { @@ -434,14 +434,12 @@ bool StunMessage::ValidateMessageIntegrityOfType(int mi_attr_type, // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ // |0 0| STUN Message Type | Message Length | // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - webrtc::SetBE16(temp_data.get() + 2, - static_cast(new_adjusted_len)); + SetBE16(temp_data.get() + 2, static_cast(new_adjusted_len)); } char hmac[kStunMessageIntegritySize]; - size_t ret = webrtc::ComputeHmac(webrtc::DIGEST_SHA_1, password.c_str(), - password.size(), temp_data.get(), mi_pos, - hmac, sizeof(hmac)); + size_t ret = ComputeHmac(DIGEST_SHA_1, password.c_str(), password.size(), + temp_data.get(), mi_pos, hmac, sizeof(hmac)); RTC_DCHECK(ret == sizeof(hmac)); if (ret != sizeof(hmac)) { return false; @@ -481,9 +479,8 @@ bool StunMessage::AddMessageIntegrityOfType(int attr_type, int msg_len_for_hmac = static_cast( buf.Length() - kStunAttributeHeaderSize - msg_integrity_attr->length()); char hmac[kStunMessageIntegritySize]; - size_t ret = - webrtc::ComputeHmac(webrtc::DIGEST_SHA_1, key.data(), key.size(), - buf.Data(), msg_len_for_hmac, hmac, sizeof(hmac)); + size_t ret = ComputeHmac(DIGEST_SHA_1, key.data(), key.size(), buf.Data(), + msg_len_for_hmac, hmac, sizeof(hmac)); RTC_DCHECK(ret == sizeof(hmac)); if (ret != sizeof(hmac)) { RTC_LOG(LS_ERROR) << "HMAC computation failed. Message-Integrity " @@ -511,26 +508,26 @@ bool StunMessage::ValidateFingerprint(const char* data, size_t size) { // Skip the rest if the magic cookie isn't present. const char* magic_cookie = data + kStunTransactionIdOffset - kStunMagicCookieLength; - if (webrtc::GetBE32(magic_cookie) != kStunMagicCookie) + if (GetBE32(magic_cookie) != kStunMagicCookie) return false; // Check the fingerprint type and length. const char* fingerprint_attr_data = data + size - fingerprint_attr_size; - if (webrtc::GetBE16(fingerprint_attr_data) != STUN_ATTR_FINGERPRINT || - webrtc::GetBE16(fingerprint_attr_data + sizeof(uint16_t)) != + if (GetBE16(fingerprint_attr_data) != STUN_ATTR_FINGERPRINT || + GetBE16(fingerprint_attr_data + sizeof(uint16_t)) != StunUInt32Attribute::SIZE) return false; // Check the fingerprint value. uint32_t fingerprint = - webrtc::GetBE32(fingerprint_attr_data + kStunAttributeHeaderSize); + GetBE32(fingerprint_attr_data + kStunAttributeHeaderSize); return ((fingerprint ^ STUN_FINGERPRINT_XOR_VALUE) == - webrtc::ComputeCrc32(data, size - fingerprint_attr_size)); + ComputeCrc32(data, size - fingerprint_attr_size)); } // static std::string StunMessage::GenerateTransactionId() { - return webrtc::CreateRandomString(kStunTransactionIdLength); + return CreateRandomString(kStunTransactionIdLength); } bool StunMessage::IsStunMethod(ArrayView methods, @@ -543,10 +540,10 @@ bool StunMessage::IsStunMethod(ArrayView methods, // Skip the rest if the magic cookie isn't present. const char* magic_cookie = data + kStunTransactionIdOffset - kStunMagicCookieLength; - if (webrtc::GetBE32(magic_cookie) != kStunMagicCookie) + if (GetBE32(magic_cookie) != kStunMagicCookie) return false; - int method = webrtc::GetBE16(data); + int method = GetBE16(data); for (int m : methods) { if (m == method) { return true; @@ -570,7 +567,7 @@ bool StunMessage::AddFingerprint() { int msg_len_for_crc32 = static_cast( buf.Length() - kStunAttributeHeaderSize - fingerprint_attr->length()); - uint32_t c = webrtc::ComputeCrc32(buf.Data(), msg_len_for_crc32); + uint32_t c = ComputeCrc32(buf.Data(), msg_len_for_crc32); // Insert the correct CRC-32, XORed with a constant, into the attribute. fingerprint_attr->SetValue(c ^ STUN_FINGERPRINT_XOR_VALUE); @@ -609,7 +606,7 @@ bool StunMessage::Read(ByteBufferReader* buf) { static_assert(sizeof(magic_cookie_int) == kStunMagicCookieLength, "Integer size mismatch: magic_cookie_int and kStunMagicCookie"); std::memcpy(&magic_cookie_int, magic_cookie.data(), sizeof(magic_cookie_int)); - if (webrtc::NetworkToHost32(magic_cookie_int) != kStunMagicCookie) { + if (NetworkToHost32(magic_cookie_int) != kStunMagicCookie) { // If magic cookie is invalid it means that the peer implements // RFC3489 instead of RFC5389. transaction_id.insert(0, magic_cookie); @@ -978,8 +975,7 @@ IPAddress StunXorAddressAttribute::GetXoredIP() const { switch (ip.family()) { case AF_INET: { in_addr v4addr = ip.ipv4_address(); - v4addr.s_addr = - (v4addr.s_addr ^ webrtc::HostToNetwork32(kStunMagicCookie)); + v4addr.s_addr = (v4addr.s_addr ^ HostToNetwork32(kStunMagicCookie)); return IPAddress(v4addr); } case AF_INET6: { @@ -992,8 +988,7 @@ IPAddress StunXorAddressAttribute::GetXoredIP() const { uint32_t* ip_as_ints = reinterpret_cast(&v6addr.s6_addr); // Transaction ID is in network byte order, but magic cookie // is stored in host byte order. - ip_as_ints[0] = - (ip_as_ints[0] ^ webrtc::HostToNetwork32(kStunMagicCookie)); + ip_as_ints[0] = (ip_as_ints[0] ^ HostToNetwork32(kStunMagicCookie)); ip_as_ints[1] = (ip_as_ints[1] ^ transactionid_as_ints[0]); ip_as_ints[2] = (ip_as_ints[2] ^ transactionid_as_ints[1]); ip_as_ints[3] = (ip_as_ints[3] ^ transactionid_as_ints[2]); @@ -1388,8 +1383,8 @@ bool ComputeStunCredentialHash(const std::string& username, input += password; char digest[MessageDigest::kMaxSize]; - size_t size = webrtc::ComputeDigest(webrtc::DIGEST_MD5, input.c_str(), - input.size(), digest, sizeof(digest)); + size_t size = ComputeDigest(DIGEST_MD5, input.c_str(), input.size(), digest, + sizeof(digest)); if (size == 0) { return false; } diff --git a/api/transport/stun_unittest.cc b/api/transport/stun_unittest.cc index 793973e037..bcbb2badc6 100644 --- a/api/transport/stun_unittest.cc +++ b/api/transport/stun_unittest.cc @@ -1194,15 +1194,15 @@ TEST_F(StunTest, ValidateMessageIntegrity) { // Again, but with the lengths matching what is claimed in the headers. EXPECT_FALSE(StunMessage::ValidateMessageIntegrityForTesting( reinterpret_cast(kStunMessageWithZeroLength), - kStunHeaderSize + webrtc::GetBE16(&kStunMessageWithZeroLength[2]), + kStunHeaderSize + GetBE16(&kStunMessageWithZeroLength[2]), kRfc5769SampleMsgPassword)); EXPECT_FALSE(StunMessage::ValidateMessageIntegrityForTesting( reinterpret_cast(kStunMessageWithExcessLength), - kStunHeaderSize + webrtc::GetBE16(&kStunMessageWithExcessLength[2]), + kStunHeaderSize + GetBE16(&kStunMessageWithExcessLength[2]), kRfc5769SampleMsgPassword)); EXPECT_FALSE(StunMessage::ValidateMessageIntegrityForTesting( reinterpret_cast(kStunMessageWithSmallLength), - kStunHeaderSize + webrtc::GetBE16(&kStunMessageWithSmallLength[2]), + kStunHeaderSize + GetBE16(&kStunMessageWithSmallLength[2]), kRfc5769SampleMsgPassword)); // Check that a too-short HMAC doesn't cause buffer overflow. @@ -1285,15 +1285,15 @@ TEST_F(StunTest, ValidateMessageIntegrity32) { // Again, but with the lengths matching what is claimed in the headers. EXPECT_FALSE(StunMessage::ValidateMessageIntegrity32ForTesting( reinterpret_cast(kStunMessageWithZeroLength), - kStunHeaderSize + webrtc::GetBE16(&kStunMessageWithZeroLength[2]), + kStunHeaderSize + GetBE16(&kStunMessageWithZeroLength[2]), kRfc5769SampleMsgPassword)); EXPECT_FALSE(StunMessage::ValidateMessageIntegrity32ForTesting( reinterpret_cast(kStunMessageWithExcessLength), - kStunHeaderSize + webrtc::GetBE16(&kStunMessageWithExcessLength[2]), + kStunHeaderSize + GetBE16(&kStunMessageWithExcessLength[2]), kRfc5769SampleMsgPassword)); EXPECT_FALSE(StunMessage::ValidateMessageIntegrity32ForTesting( reinterpret_cast(kStunMessageWithSmallLength), - kStunHeaderSize + webrtc::GetBE16(&kStunMessageWithSmallLength[2]), + kStunHeaderSize + GetBE16(&kStunMessageWithSmallLength[2]), kRfc5769SampleMsgPassword)); // Check that a too-short HMAC doesn't cause buffer overflow. @@ -1706,24 +1706,24 @@ TEST_F(StunTest, SizeRestrictionOnAttributes) { } TEST_F(StunTest, ValidateMessageIntegrityWithParser) { - webrtc::metrics::Reset(); // Ensure counters start from zero. + metrics::Reset(); // Ensure counters start from zero. // Try the messages from RFC 5769. StunMessage message; ByteBufferReader reader(kRfc5769SampleRequest); EXPECT_TRUE(message.Read(&reader)); EXPECT_EQ(message.ValidateMessageIntegrity(kRfc5769SampleMsgPassword), StunMessage::IntegrityStatus::kIntegrityOk); - EXPECT_EQ(webrtc::metrics::NumEvents( + EXPECT_EQ(metrics::NumEvents( "WebRTC.Stun.Integrity.Request", static_cast(StunMessage::IntegrityStatus::kIntegrityOk)), 1); EXPECT_EQ(message.RevalidateMessageIntegrity("Invalid password"), StunMessage::IntegrityStatus::kIntegrityBad); - EXPECT_EQ(webrtc::metrics::NumEvents( + EXPECT_EQ(metrics::NumEvents( "WebRTC.Stun.Integrity.Request", static_cast(StunMessage::IntegrityStatus::kIntegrityBad)), 1); - EXPECT_EQ(webrtc::metrics::NumSamples("WebRTC.Stun.Integrity.Request"), 2); + EXPECT_EQ(metrics::NumSamples("WebRTC.Stun.Integrity.Request"), 2); } } // namespace webrtc diff --git a/rtc_tools/data_channel_benchmark/grpc_signaling.cc b/rtc_tools/data_channel_benchmark/grpc_signaling.cc index 33fb60028c..9073b0078b 100644 --- a/rtc_tools/data_channel_benchmark/grpc_signaling.cc +++ b/rtc_tools/data_channel_benchmark/grpc_signaling.cc @@ -29,7 +29,7 @@ using GrpcSignaling::SessionDescription; using GrpcSignaling::SignalingMessage; template -class SessionData : public webrtc::SignalingInterface { +class SessionData : public SignalingInterface { public: SessionData() {} explicit SessionData(T* stream) : stream_(stream) {} @@ -102,8 +102,8 @@ void ProcessMessages(StreamReader* stream, SessionData* session) { while (stream->Read(&message)) { switch (message.Content_case()) { case SignalingMessage::ContentCase::kCandidate: { - webrtc::SdpParseError error; - auto jsep_candidate = std::make_unique( + SdpParseError error; + auto jsep_candidate = std::make_unique( message.candidate().mid(), message.candidate().mline_index()); if (!jsep_candidate->Initialize(message.candidate().description(), &error)) { @@ -121,10 +121,9 @@ void ProcessMessages(StreamReader* stream, SessionData* session) { auto& description = message.description(); auto content = description.content(); - auto sdp = webrtc::CreateSessionDescription( - description.type() == SessionDescription::OFFER - ? webrtc::SdpType::kOffer - : webrtc::SdpType::kAnswer, + auto sdp = CreateSessionDescription( + description.type() == SessionDescription::OFFER ? SdpType::kOffer + : SdpType::kAnswer, description.content()); session->remote_description_callback_(std::move(sdp)); break; @@ -138,10 +137,9 @@ void ProcessMessages(StreamReader* stream, SessionData* session) { class GrpcNegotiationServer : public GrpcSignalingServerInterface, public PeerConnectionSignaling::Service { public: - GrpcNegotiationServer( - std::function callback, - int port, - bool oneshot) + GrpcNegotiationServer(std::function callback, + int port, + bool oneshot) : connect_callback_(std::move(callback)), requested_port_(port), oneshot_(oneshot) {} @@ -196,7 +194,7 @@ class GrpcNegotiationServer : public GrpcSignalingServerInterface, } private: - std::function connect_callback_; + std::function connect_callback_; int requested_port_; int selected_port_; bool oneshot_; @@ -236,7 +234,7 @@ class GrpcNegotiationClient : public GrpcSignalingClientInterface { return true; } - webrtc::SignalingInterface* signaling_client() override { return &session_; } + SignalingInterface* signaling_client() override { return &session_; } private: std::shared_ptr channel_; @@ -252,7 +250,7 @@ class GrpcNegotiationClient : public GrpcSignalingClientInterface { std::unique_ptr GrpcSignalingServerInterface::Create( - std::function callback, + std::function callback, int port, bool oneshot) { return std::make_unique(std::move(callback), port, diff --git a/rtc_tools/data_channel_benchmark/peer_connection_client.cc b/rtc_tools/data_channel_benchmark/peer_connection_client.cc index 74175bd7da..74f1216e2b 100644 --- a/rtc_tools/data_channel_benchmark/peer_connection_client.cc +++ b/rtc_tools/data_channel_benchmark/peer_connection_client.cc @@ -128,15 +128,15 @@ class CreateSessionDescriptionObserverAdapter namespace webrtc { PeerConnectionClient::PeerConnectionClient( - webrtc::PeerConnectionFactoryInterface* factory, - webrtc::SignalingInterface* signaling) + PeerConnectionFactoryInterface* factory, + SignalingInterface* signaling) : signaling_(signaling) { signaling_->OnIceCandidate( - [&](std::unique_ptr candidate) { + [&](std::unique_ptr candidate) { AddIceCandidate(std::move(candidate)); }); signaling_->OnRemoteDescription( - [&](std::unique_ptr sdp) { + [&](std::unique_ptr sdp) { SetRemoteDescription(std::move(sdp)); }); InitializePeerConnection(factory); @@ -148,11 +148,11 @@ PeerConnectionClient::~PeerConnectionClient() { scoped_refptr PeerConnectionClient::CreateDefaultFactory(Thread* signaling_thread) { - auto factory = webrtc::CreatePeerConnectionFactory( + auto factory = CreatePeerConnectionFactory( /*network_thread=*/nullptr, /*worker_thread=*/nullptr, /*signaling_thread*/ signaling_thread, - /*default_adm=*/nullptr, webrtc::CreateBuiltinAudioEncoderFactory(), - webrtc::CreateBuiltinAudioDecoderFactory(), + /*default_adm=*/nullptr, CreateBuiltinAudioEncoderFactory(), + CreateBuiltinAudioDecoderFactory(), std::make_unique>(), @@ -170,17 +170,17 @@ PeerConnectionClient::CreateDefaultFactory(Thread* signaling_thread) { } bool PeerConnectionClient::InitializePeerConnection( - webrtc::PeerConnectionFactoryInterface* factory) { + PeerConnectionFactoryInterface* factory) { RTC_CHECK(factory) << "Must call InitializeFactory before InitializePeerConnection"; - webrtc::PeerConnectionInterface::RTCConfiguration config; - webrtc::PeerConnectionInterface::IceServer server; + PeerConnectionInterface::RTCConfiguration config; + PeerConnectionInterface::IceServer server; server.urls.push_back(kStunServer); config.servers.push_back(server); - config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; + config.sdp_semantics = SdpSemantics::kUnifiedPlan; - webrtc::PeerConnectionDependencies dependencies(this); + PeerConnectionDependencies dependencies(this); auto result = factory->CreatePeerConnectionOrError(config, std::move(dependencies)); @@ -199,8 +199,7 @@ bool PeerConnectionClient::StartPeerConnection() { RTC_LOG(LS_INFO) << "Creating offer"; peer_connection_->SetLocalDescription( - SetLocalDescriptionObserverAdapter::Create([this]( - webrtc::RTCError error) { + SetLocalDescriptionObserverAdapter::Create([this](RTCError error) { if (error.ok()) signaling_->SendDescription(peer_connection_->local_description()); })); @@ -210,7 +209,7 @@ bool PeerConnectionClient::StartPeerConnection() { bool PeerConnectionClient::IsConnected() { return peer_connection_->peer_connection_state() == - webrtc::PeerConnectionInterface::PeerConnectionState::kConnected; + PeerConnectionInterface::PeerConnectionState::kConnected; } // Disconnect from the call. @@ -234,11 +233,11 @@ void PeerConnectionClient::DeletePeerConnection() { } void PeerConnectionClient::OnIceConnectionChange( - webrtc::PeerConnectionInterface::IceConnectionState new_state) { - if (new_state == webrtc::PeerConnectionInterface::IceConnectionState:: - kIceConnectionCompleted) { + PeerConnectionInterface::IceConnectionState new_state) { + if (new_state == + PeerConnectionInterface::IceConnectionState::kIceConnectionCompleted) { RTC_LOG(LS_INFO) << "State is updating to connected"; - } else if (new_state == webrtc::PeerConnectionInterface::IceConnectionState:: + } else if (new_state == PeerConnectionInterface::IceConnectionState:: kIceConnectionDisconnected) { RTC_LOG(LS_INFO) << "Disconnecting from peer"; Disconnect(); @@ -246,19 +245,19 @@ void PeerConnectionClient::OnIceConnectionChange( } void PeerConnectionClient::OnIceGatheringChange( - webrtc::PeerConnectionInterface::IceGatheringState new_state) { - if (new_state == webrtc::PeerConnectionInterface::kIceGatheringComplete) { + PeerConnectionInterface::IceGatheringState new_state) { + if (new_state == PeerConnectionInterface::kIceGatheringComplete) { RTC_LOG(LS_INFO) << "Client is ready to receive remote SDP"; } } void PeerConnectionClient::OnIceCandidate( - const webrtc::IceCandidateInterface* candidate) { + const IceCandidateInterface* candidate) { signaling_->SendIceCandidate(candidate); } void PeerConnectionClient::OnDataChannel( - scoped_refptr channel) { + scoped_refptr channel) { RTC_LOG(LS_INFO) << __FUNCTION__ << " remote datachannel created"; if (on_data_channel_callback_) on_data_channel_callback_(channel); @@ -266,8 +265,7 @@ void PeerConnectionClient::OnDataChannel( } void PeerConnectionClient::SetOnDataChannel( - std::function)> - callback) { + std::function)> callback) { on_data_channel_callback_ = callback; } @@ -275,28 +273,27 @@ void PeerConnectionClient::OnNegotiationNeededEvent(uint32_t event_id) { RTC_LOG(LS_INFO) << "OnNegotiationNeededEvent"; peer_connection_->SetLocalDescription( - SetLocalDescriptionObserverAdapter::Create([this]( - webrtc::RTCError error) { + SetLocalDescriptionObserverAdapter::Create([this](RTCError error) { if (error.ok()) signaling_->SendDescription(peer_connection_->local_description()); })); } bool PeerConnectionClient::SetRemoteDescription( - std::unique_ptr desc) { + std::unique_ptr desc) { RTC_LOG(LS_INFO) << "SetRemoteDescription"; auto type = desc->GetType(); peer_connection_->SetRemoteDescription( std::move(desc), - SetRemoteDescriptionObserverAdapter::Create([&](webrtc::RTCError) { + SetRemoteDescriptionObserverAdapter::Create([&](RTCError) { RTC_LOG(LS_INFO) << "SetRemoteDescription done"; - if (type == webrtc::SdpType::kOffer) { + if (type == SdpType::kOffer) { // Got an offer from the remote, need to set an answer and send it. peer_connection_->SetLocalDescription( SetLocalDescriptionObserverAdapter::Create( - [this](webrtc::RTCError error) { + [this](RTCError error) { if (error.ok()) signaling_->SendDescription( peer_connection_->local_description()); @@ -308,7 +305,7 @@ bool PeerConnectionClient::SetRemoteDescription( } void PeerConnectionClient::AddIceCandidate( - std::unique_ptr candidate) { + std::unique_ptr candidate) { RTC_LOG(LS_INFO) << "AddIceCandidate"; peer_connection_->AddIceCandidate( diff --git a/rtc_tools/frame_analyzer/video_quality_analysis.cc b/rtc_tools/frame_analyzer/video_quality_analysis.cc index 1229d0a85f..36dc933506 100644 --- a/rtc_tools/frame_analyzer/video_quality_analysis.cc +++ b/rtc_tools/frame_analyzer/video_quality_analysis.cc @@ -55,8 +55,8 @@ double Ssim(const scoped_refptr& ref_buffer, } std::vector RunAnalysis( - const scoped_refptr& reference_video, - const scoped_refptr& test_video, + const scoped_refptr& reference_video, + const scoped_refptr& test_video, const std::vector& test_frame_indices) { std::vector results; for (size_t i = 0; i < test_video->number_of_frames(); ++i) { diff --git a/rtc_tools/frame_analyzer/video_quality_analysis_unittest.cc b/rtc_tools/frame_analyzer/video_quality_analysis_unittest.cc index d0227fb4b3..2b18388e81 100644 --- a/rtc_tools/frame_analyzer/video_quality_analysis_unittest.cc +++ b/rtc_tools/frame_analyzer/video_quality_analysis_unittest.cc @@ -136,8 +136,7 @@ TEST(VideoQualityAnalysisTest, PrintMaxRepeatedAndSkippedFramesDecodeErrorInTest) { ResultsContainer result; - std::string log_filename = - TempFilename(webrtc::test::OutputPath(), "log.log"); + std::string log_filename = TempFilename(test::OutputPath(), "log.log"); FILE* logfile = fopen(log_filename.c_str(), "w"); ASSERT_TRUE(logfile != NULL); diff --git a/rtc_tools/network_tester/packet_sender.cc b/rtc_tools/network_tester/packet_sender.cc index 0ab5b77ffc..abc2240408 100644 --- a/rtc_tools/network_tester/packet_sender.cc +++ b/rtc_tools/network_tester/packet_sender.cc @@ -28,7 +28,7 @@ namespace { absl::AnyInvocable SendPacketTask( PacketSender* packet_sender, - scoped_refptr task_safety_flag, + scoped_refptr task_safety_flag, int64_t target_time_ms = TimeMillis()) { return [target_time_ms, packet_sender, task_safety_flag = std::move(task_safety_flag)]() mutable { @@ -48,7 +48,7 @@ absl::AnyInvocable SendPacketTask( absl::AnyInvocable UpdateTestSettingTask( PacketSender* packet_sender, std::unique_ptr config_reader, - scoped_refptr task_safety_flag) { + scoped_refptr task_safety_flag) { return [packet_sender, config_reader = std::move(config_reader), task_safety_flag = std::move(task_safety_flag)]() mutable { if (!task_safety_flag->alive()) { @@ -72,8 +72,8 @@ absl::AnyInvocable UpdateTestSettingTask( PacketSender::PacketSender( TestController* test_controller, - webrtc::TaskQueueBase* worker_queue, - scoped_refptr task_safety_flag, + TaskQueueBase* worker_queue, + scoped_refptr task_safety_flag, const std::string& config_file_path) : packet_size_(0), send_interval_ms_(0), diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer.cc b/rtc_tools/rtc_event_log_visualizer/analyzer.cc index 4aea669675..544f144cc1 100644 --- a/rtc_tools/rtc_event_log_visualizer/analyzer.cc +++ b/rtc_tools/rtc_event_log_visualizer/analyzer.cc @@ -282,44 +282,43 @@ absl::string_view GetIceCandidateTypeAsString(IceCandidateType type) { } } -std::string GetProtocolAsString(webrtc::IceCandidatePairProtocol protocol) { +std::string GetProtocolAsString(IceCandidatePairProtocol protocol) { switch (protocol) { - case webrtc::IceCandidatePairProtocol::kUdp: + case IceCandidatePairProtocol::kUdp: return kProtocolUdp; - case webrtc::IceCandidatePairProtocol::kTcp: + case IceCandidatePairProtocol::kTcp: return kProtocolTcp; - case webrtc::IceCandidatePairProtocol::kSsltcp: + case IceCandidatePairProtocol::kSsltcp: return kProtocolSsltcp; - case webrtc::IceCandidatePairProtocol::kTls: + case IceCandidatePairProtocol::kTls: return kProtocolTls; default: return kUnknownEnumValue; } } -std::string GetAddressFamilyAsString( - webrtc::IceCandidatePairAddressFamily family) { +std::string GetAddressFamilyAsString(IceCandidatePairAddressFamily family) { switch (family) { - case webrtc::IceCandidatePairAddressFamily::kIpv4: + case IceCandidatePairAddressFamily::kIpv4: return kAddressFamilyIpv4; - case webrtc::IceCandidatePairAddressFamily::kIpv6: + case IceCandidatePairAddressFamily::kIpv6: return kAddressFamilyIpv6; default: return kUnknownEnumValue; } } -std::string GetNetworkTypeAsString(webrtc::IceCandidateNetworkType type) { +std::string GetNetworkTypeAsString(IceCandidateNetworkType type) { switch (type) { - case webrtc::IceCandidateNetworkType::kEthernet: + case IceCandidateNetworkType::kEthernet: return kNetworkTypeEthernet; - case webrtc::IceCandidateNetworkType::kLoopback: + case IceCandidateNetworkType::kLoopback: return kNetworkTypeLoopback; - case webrtc::IceCandidateNetworkType::kWifi: + case IceCandidateNetworkType::kWifi: return kNetworkTypeWifi; - case webrtc::IceCandidateNetworkType::kVpn: + case IceCandidateNetworkType::kVpn: return kNetworkTypeVpn; - case webrtc::IceCandidateNetworkType::kCellular: + case IceCandidateNetworkType::kCellular: return kNetworkTypeCellular; default: return kUnknownEnumValue; @@ -435,19 +434,19 @@ struct PacketLossSummary { Timestamp base_time = Timestamp::MinusInfinity(); }; -float GetHighestSeqNumber(const webrtc::rtcp::ReportBlock& block) { +float GetHighestSeqNumber(const rtcp::ReportBlock& block) { return block.extended_high_seq_num(); } -float GetFractionLost(const webrtc::rtcp::ReportBlock& block) { +float GetFractionLost(const rtcp::ReportBlock& block) { return static_cast(block.fraction_lost()) / 256 * 100; } -float GetCumulativeLost(const webrtc::rtcp::ReportBlock& block) { +float GetCumulativeLost(const rtcp::ReportBlock& block) { return block.cumulative_lost(); } -float DelaySinceLastSr(const webrtc::rtcp::ReportBlock& block) { +float DelaySinceLastSr(const rtcp::ReportBlock& block) { return static_cast(block.delay_since_last_sr()) / 65536; } @@ -534,35 +533,35 @@ void EventLogAnalyzer::InitializeMapOfNamedGraphs(bool show_detector_state, bool show_alr_state, bool show_link_capacity) { plots_.RegisterPlot("incoming_packet_sizes", [this](Plot* plot) { - this->CreatePacketGraph(webrtc::kIncomingPacket, plot); + this->CreatePacketGraph(kIncomingPacket, plot); }); plots_.RegisterPlot("outgoing_packet_sizes", [this](Plot* plot) { - this->CreatePacketGraph(webrtc::kOutgoingPacket, plot); + this->CreatePacketGraph(kOutgoingPacket, plot); }); plots_.RegisterPlot("incoming_rtcp_types", [this](Plot* plot) { - this->CreateRtcpTypeGraph(webrtc::kIncomingPacket, plot); + this->CreateRtcpTypeGraph(kIncomingPacket, plot); }); plots_.RegisterPlot("outgoing_rtcp_types", [this](Plot* plot) { - this->CreateRtcpTypeGraph(webrtc::kOutgoingPacket, plot); + this->CreateRtcpTypeGraph(kOutgoingPacket, plot); }); plots_.RegisterPlot("incoming_packet_count", [this](Plot* plot) { - this->CreateAccumulatedPacketsGraph(webrtc::kIncomingPacket, plot); + this->CreateAccumulatedPacketsGraph(kIncomingPacket, plot); }); plots_.RegisterPlot("outgoing_packet_count", [this](Plot* plot) { - this->CreateAccumulatedPacketsGraph(webrtc::kOutgoingPacket, plot); + this->CreateAccumulatedPacketsGraph(kOutgoingPacket, plot); }); plots_.RegisterPlot("incoming_packet_rate", [this](Plot* plot) { - this->CreatePacketRateGraph(webrtc::kIncomingPacket, plot); + this->CreatePacketRateGraph(kIncomingPacket, plot); }); plots_.RegisterPlot("outgoing_packet_rate", [this](Plot* plot) { - this->CreatePacketRateGraph(webrtc::kOutgoingPacket, plot); + this->CreatePacketRateGraph(kOutgoingPacket, plot); }); plots_.RegisterPlot("total_incoming_packet_rate", [this](Plot* plot) { - this->CreateTotalPacketRateGraph(webrtc::kIncomingPacket, plot); + this->CreateTotalPacketRateGraph(kIncomingPacket, plot); }); plots_.RegisterPlot("total_outgoing_packet_rate", [this](Plot* plot) { - this->CreateTotalPacketRateGraph(webrtc::kOutgoingPacket, plot); + this->CreateTotalPacketRateGraph(kOutgoingPacket, plot); }); plots_.RegisterPlot("audio_playout", [this](Plot* plot) { this->CreatePlayoutGraph(plot); }); @@ -572,10 +571,10 @@ void EventLogAnalyzer::InitializeMapOfNamedGraphs(bool show_detector_state, }); plots_.RegisterPlot("incoming_audio_level", [this](Plot* plot) { - this->CreateAudioLevelGraph(webrtc::kIncomingPacket, plot); + this->CreateAudioLevelGraph(kIncomingPacket, plot); }); plots_.RegisterPlot("outgoing_audio_level", [this](Plot* plot) { - this->CreateAudioLevelGraph(webrtc::kOutgoingPacket, plot); + this->CreateAudioLevelGraph(kOutgoingPacket, plot); }); plots_.RegisterPlot("incoming_sequence_number_delta", [this](Plot* plot) { this->CreateSequenceNumberGraph(plot); @@ -596,16 +595,16 @@ void EventLogAnalyzer::InitializeMapOfNamedGraphs(bool show_detector_state, plot, show_detector_state, show_alr_state, show_link_capacity); }); plots_.RegisterPlot("incoming_stream_bitrate", [this](Plot* plot) { - this->CreateStreamBitrateGraph(webrtc::kIncomingPacket, plot); + this->CreateStreamBitrateGraph(kIncomingPacket, plot); }); plots_.RegisterPlot("outgoing_stream_bitrate", [this](Plot* plot) { - this->CreateStreamBitrateGraph(webrtc::kOutgoingPacket, plot); + this->CreateStreamBitrateGraph(kOutgoingPacket, plot); }); plots_.RegisterPlot("incoming_layer_bitrate_allocation", [this](Plot* plot) { - this->CreateBitrateAllocationGraph(webrtc::kIncomingPacket, plot); + this->CreateBitrateAllocationGraph(kIncomingPacket, plot); }); plots_.RegisterPlot("outgoing_layer_bitrate_allocation", [this](Plot* plot) { - this->CreateBitrateAllocationGraph(webrtc::kOutgoingPacket, plot); + this->CreateBitrateAllocationGraph(kOutgoingPacket, plot); }); plots_.RegisterPlot("simulated_receiveside_bwe", [this](Plot* plot) { this->CreateReceiveSideBweSimulationGraph(plot); @@ -632,54 +631,54 @@ void EventLogAnalyzer::InitializeMapOfNamedGraphs(bool show_detector_state, this->CreateFractionLossGraph(plot); }); plots_.RegisterPlot("incoming_timestamps", [this](Plot* plot) { - this->CreateTimestampGraph(webrtc::kIncomingPacket, plot); + this->CreateTimestampGraph(kIncomingPacket, plot); }); plots_.RegisterPlot("outgoing_timestamps", [this](Plot* plot) { - this->CreateTimestampGraph(webrtc::kOutgoingPacket, plot); + this->CreateTimestampGraph(kOutgoingPacket, plot); }); plots_.RegisterPlot("incoming_rtcp_fraction_lost", [this](Plot* plot) { - this->CreateSenderAndReceiverReportPlot( - webrtc::kIncomingPacket, GetFractionLost, - "Fraction lost (incoming RTCP)", "Loss rate (percent)", plot); + this->CreateSenderAndReceiverReportPlot(kIncomingPacket, GetFractionLost, + "Fraction lost (incoming RTCP)", + "Loss rate (percent)", plot); }); plots_.RegisterPlot("outgoing_rtcp_fraction_lost", [this](Plot* plot) { - this->CreateSenderAndReceiverReportPlot( - webrtc::kOutgoingPacket, GetFractionLost, - "Fraction lost (outgoing RTCP)", "Loss rate (percent)", plot); + this->CreateSenderAndReceiverReportPlot(kOutgoingPacket, GetFractionLost, + "Fraction lost (outgoing RTCP)", + "Loss rate (percent)", plot); }); plots_.RegisterPlot("incoming_rtcp_cumulative_lost", [this](Plot* plot) { this->CreateSenderAndReceiverReportPlot( - webrtc::kIncomingPacket, GetCumulativeLost, + kIncomingPacket, GetCumulativeLost, "Cumulative lost packets (incoming RTCP)", "Packets", plot); }); plots_.RegisterPlot("outgoing_rtcp_cumulative_lost", [this](Plot* plot) { this->CreateSenderAndReceiverReportPlot( - webrtc::kOutgoingPacket, GetCumulativeLost, + kOutgoingPacket, GetCumulativeLost, "Cumulative lost packets (outgoing RTCP)", "Packets", plot); }); plots_.RegisterPlot("incoming_rtcp_highest_seq_number", [this](Plot* plot) { this->CreateSenderAndReceiverReportPlot( - webrtc::kIncomingPacket, GetHighestSeqNumber, + kIncomingPacket, GetHighestSeqNumber, "Highest sequence number (incoming RTCP)", "Sequence number", plot); }); plots_.RegisterPlot("outgoing_rtcp_highest_seq_number", [this](Plot* plot) { this->CreateSenderAndReceiverReportPlot( - webrtc::kOutgoingPacket, GetHighestSeqNumber, + kOutgoingPacket, GetHighestSeqNumber, "Highest sequence number (outgoing RTCP)", "Sequence number", plot); }); plots_.RegisterPlot("incoming_rtcp_delay_since_last_sr", [this](Plot* plot) { this->CreateSenderAndReceiverReportPlot( - webrtc::kIncomingPacket, DelaySinceLastSr, + kIncomingPacket, DelaySinceLastSr, "Delay since last received sender report (incoming RTCP)", "Time (s)", plot); }); plots_.RegisterPlot("outgoing_rtcp_delay_since_last_sr", [this](Plot* plot) { this->CreateSenderAndReceiverReportPlot( - webrtc::kOutgoingPacket, DelaySinceLastSr, + kOutgoingPacket, DelaySinceLastSr, "Delay since last received sender report (outgoing RTCP)", "Time (s)", plot); }); @@ -1592,16 +1591,16 @@ void EventLogAnalyzer::CreateEcnFeedbackGraph(Plot* plot, for (const rtcp::CongestionControlFeedback::PacketInfo& info : feedback.congestion_feedback.packets()) { switch (info.ecn) { - case webrtc::EcnMarking::kNotEct: + case EcnMarking::kNotEct: ++not_ect_count; break; - case webrtc::EcnMarking::kEct1: + case EcnMarking::kEct1: ++ect_1_count; break; - case webrtc::EcnMarking::kEct0: + case EcnMarking::kEct0: RTC_LOG(LS_ERROR) << "unexpected ect(0)"; break; - case webrtc::EcnMarking::kCe: + case EcnMarking::kCe: ++ce_count; break; } diff --git a/rtc_tools/rtc_event_log_visualizer/plot_base.cc b/rtc_tools/rtc_event_log_visualizer/plot_base.cc index ddfa5e056e..481c81cd40 100644 --- a/rtc_tools/rtc_event_log_visualizer/plot_base.cc +++ b/rtc_tools/rtc_event_log_visualizer/plot_base.cc @@ -262,9 +262,9 @@ void Plot::PrintPythonCode(absl::string_view figure_output_path) const { } } -void Plot::ExportProtobuf(webrtc::analytics::Chart* chart) const { +void Plot::ExportProtobuf(analytics::Chart* chart) const { for (size_t i = 0; i < series_list_.size(); i++) { - webrtc::analytics::DataSet* data_set = chart->add_data_sets(); + analytics::DataSet* data_set = chart->add_data_sets(); for (const auto& point : series_list_[i].points) { data_set->add_x_values(point.x); } @@ -273,15 +273,15 @@ void Plot::ExportProtobuf(webrtc::analytics::Chart* chart) const { } if (series_list_[i].line_style == LineStyle::kBar) { - data_set->set_style(webrtc::analytics::ChartStyle::BAR_CHART); + data_set->set_style(analytics::ChartStyle::BAR_CHART); } else if (series_list_[i].line_style == LineStyle::kLine) { - data_set->set_style(webrtc::analytics::ChartStyle::LINE_CHART); + data_set->set_style(analytics::ChartStyle::LINE_CHART); } else if (series_list_[i].line_style == LineStyle::kStep) { - data_set->set_style(webrtc::analytics::ChartStyle::LINE_STEP_CHART); + data_set->set_style(analytics::ChartStyle::LINE_STEP_CHART); } else if (series_list_[i].line_style == LineStyle::kNone) { - data_set->set_style(webrtc::analytics::ChartStyle::SCATTER_CHART); + data_set->set_style(analytics::ChartStyle::SCATTER_CHART); } else { - data_set->set_style(webrtc::analytics::ChartStyle::UNDEFINED); + data_set->set_style(analytics::ChartStyle::UNDEFINED); } if (series_list_[i].point_style == PointStyle::kHighlight) @@ -300,7 +300,7 @@ void Plot::ExportProtobuf(webrtc::analytics::Chart* chart) const { chart->set_id(id_); for (const auto& kv : yaxis_tick_labels_) { - webrtc::analytics::TickLabel* tick = chart->add_yaxis_tick_labels(); + analytics::TickLabel* tick = chart->add_yaxis_tick_labels(); tick->set_value(kv.first); tick->set_label(kv.second); } @@ -334,10 +334,9 @@ void PlotCollection::PrintPythonCode( } void PlotCollection::ExportProtobuf( - webrtc::analytics::ChartCollection* collection) const { + analytics::ChartCollection* collection) const { for (const auto& plot : plots_) { - webrtc::analytics::Chart* protobuf_representation = - collection->add_charts(); + analytics::Chart* protobuf_representation = collection->add_charts(); plot->ExportProtobuf(protobuf_representation); } if (calltime_to_utc_ms_) { diff --git a/rtc_tools/rtp_generator/rtp_generator.cc b/rtc_tools/rtp_generator/rtp_generator.cc index 51228a4b65..3152bc1834 100644 --- a/rtc_tools/rtp_generator/rtp_generator.cc +++ b/rtc_tools/rtp_generator/rtp_generator.cc @@ -192,15 +192,15 @@ RtpGenerator::RtpGenerator(const RtpGeneratorOptions& options) : options_(options), env_(CreateEnvironment()), video_encoder_factory_( - std::make_unique>()), + std::make_unique< + VideoEncoderFactoryTemplate>()), video_decoder_factory_( - std::make_unique>()), + std::make_unique< + VideoDecoderFactoryTemplate>()), video_bitrate_allocator_factory_( CreateBuiltinVideoBitrateAllocatorFactory()), call_(Call::Create(CallConfig(env_))) { @@ -208,9 +208,9 @@ RtpGenerator::RtpGenerator(const RtpGeneratorOptions& options) constexpr int kMaxBitrateBps = 2500000; // 2.5 Mbps int stream_count = 0; - webrtc::VideoEncoder::EncoderInfo encoder_info; + VideoEncoder::EncoderInfo encoder_info; for (const auto& send_config : options.video_streams) { - webrtc::VideoSendStream::Config video_config(this); + VideoSendStream::Config video_config(this); video_config.encoder_settings.encoder_factory = video_encoder_factory_.get(); video_config.encoder_settings.bitrate_allocator_factory = @@ -268,9 +268,8 @@ RtpGenerator::RtpGenerator(const RtpGeneratorOptions& options) VideoSendStream* video_send_stream = call_->CreateVideoSendStream( std::move(video_config), std::move(encoder_config)); - video_send_stream->SetSource( - frame_generator.get(), - webrtc::DegradationPreference::MAINTAIN_FRAMERATE); + video_send_stream->SetSource(frame_generator.get(), + DegradationPreference::MAINTAIN_FRAMERATE); // Store these objects so we can destropy them at the end. frame_generators_.push_back(std::move(frame_generator)); video_send_streams_.push_back(video_send_stream); @@ -287,8 +286,7 @@ void RtpGenerator::GenerateRtpDump(const std::string& rtp_dump_path) { rtp_dump_writer_.reset(test::RtpFileWriter::Create( test::RtpFileWriter::kRtpDump, rtp_dump_path)); - call_->SignalChannelNetworkState(webrtc::MediaType::VIDEO, - webrtc::kNetworkUp); + call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp); for (VideoSendStream* send_stream : video_send_streams_) { send_stream->Start(); } @@ -296,8 +294,7 @@ void RtpGenerator::GenerateRtpDump(const std::string& rtp_dump_path) { // Spinlock until all the durations end. WaitUntilAllVideoStreamsFinish(); - call_->SignalChannelNetworkState(webrtc::MediaType::VIDEO, - webrtc::kNetworkDown); + call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkDown); } bool RtpGenerator::SendRtp(ArrayView packet, @@ -346,12 +343,12 @@ void RtpGenerator::WaitUntilAllVideoStreamsFinish() { test::RtpPacket RtpGenerator::DataToRtpPacket(const uint8_t* packet, size_t packet_len) { - webrtc::test::RtpPacket rtp_packet; + test::RtpPacket rtp_packet; memcpy(rtp_packet.data, packet, packet_len); rtp_packet.length = packet_len; rtp_packet.original_length = packet_len; rtp_packet.time_ms = - webrtc::Clock::GetRealTimeClock()->TimeInMilliseconds() - start_ms_; + Clock::GetRealTimeClock()->TimeInMilliseconds() - start_ms_; return rtp_packet; } diff --git a/rtc_tools/video_encoder/video_encoder.cc b/rtc_tools/video_encoder/video_encoder.cc index 130308415d..b252162bcf 100644 --- a/rtc_tools/video_encoder/video_encoder.cc +++ b/rtc_tools/video_encoder/video_encoder.cc @@ -189,14 +189,14 @@ uint32_t GetDefaultTargetBitrate(const VideoCodecType codec, }; size_t codec_index = 0; switch (codec) { - case webrtc::kVideoCodecVP8: - case webrtc::kVideoCodecH264: + case kVideoCodecVP8: + case kVideoCodecH264: codec_index = 0; break; - case webrtc::kVideoCodecVP9: + case kVideoCodecVP9: codec_index = 1; break; - case webrtc::kVideoCodecAV1: + case kVideoCodecAV1: codec_index = 2; break; default: @@ -264,7 +264,7 @@ class BitstreamProcessor final : public EncodedImageCallback, } } - void ValidatePSNR(webrtc::VideoFrame& frame) { + void ValidatePSNR(VideoFrame& frame) { RTC_CHECK(validate_psnr_); video_decoder_->Decode(*encoded_image_, /*dont_care=*/0); double psnr = I420PSNR(*frame.video_frame_buffer()->ToI420(), @@ -494,10 +494,9 @@ class TestVideoEncoderFactoryWrapper final { RTC_CHECK(video_encoder); // Initialize video encoder. - const webrtc::VideoEncoder::Settings kSettings( - webrtc::VideoEncoder::Capabilities(false), - /*number_of_cores=*/1, - /*max_payload_size=*/0); + const VideoEncoder::Settings kSettings(VideoEncoder::Capabilities(false), + /*number_of_cores=*/1, + /*max_payload_size=*/0); int ret = video_encoder->InitEncode(&video_codec_setting, kSettings); RTC_CHECK_EQ(ret, WEBRTC_VIDEO_CODEC_OK); @@ -508,12 +507,12 @@ class TestVideoEncoderFactoryWrapper final { env, video_codec_setting); RTC_CHECK(bitrate_allocator); - webrtc::VideoBitrateAllocation allocation = + VideoBitrateAllocation allocation = bitrate_allocator->GetAllocation(bitrate_kbps * 1000, frame_rate_fps); RTC_LOG(LS_INFO) << allocation.ToString(); - video_encoder->SetRates(webrtc::VideoEncoder::RateControlParameters( - allocation, frame_rate_fps)); + video_encoder->SetRates( + VideoEncoder::RateControlParameters(allocation, frame_rate_fps)); return video_encoder; } diff --git a/rtc_tools/video_file_reader_unittest.cc b/rtc_tools/video_file_reader_unittest.cc index a357699d37..cd81dd1785 100644 --- a/rtc_tools/video_file_reader_unittest.cc +++ b/rtc_tools/video_file_reader_unittest.cc @@ -24,7 +24,7 @@ class Y4mFileReaderTest : public ::testing::Test { public: void SetUp() override { const std::string filename = - TempFilename(webrtc::test::OutputPath(), "test_video_file.y4m"); + TempFilename(test::OutputPath(), "test_video_file.y4m"); // Create simple test video of size 6x4. FILE* file = fopen(filename.c_str(), "wb"); @@ -45,11 +45,11 @@ class Y4mFileReaderTest : public ::testing::Test { fclose(file); // Open the newly created file. - video = webrtc::test::OpenY4mFile(filename); + video = test::OpenY4mFile(filename); ASSERT_TRUE(video); } - scoped_refptr video; + scoped_refptr video; }; TEST_F(Y4mFileReaderTest, TestParsingFileHeader) { @@ -77,7 +77,7 @@ class YuvFileReaderTest : public ::testing::Test { public: void SetUp() override { const std::string filename = - TempFilename(webrtc::test::OutputPath(), "test_video_file.yuv"); + TempFilename(test::OutputPath(), "test_video_file.yuv"); // Create simple test video of size 6x4. FILE* file = fopen(filename.c_str(), "wb"); @@ -95,11 +95,11 @@ class YuvFileReaderTest : public ::testing::Test { fclose(file); // Open the newly created file. - video = webrtc::test::OpenYuvFile(filename, 6, 4); + video = test::OpenYuvFile(filename, 6, 4); ASSERT_TRUE(video); } - scoped_refptr video; + scoped_refptr video; }; TEST_F(YuvFileReaderTest, TestParsingFileHeader) { diff --git a/rtc_tools/video_file_writer_unittest.cc b/rtc_tools/video_file_writer_unittest.cc index bcf10c0d54..7c8b65df8d 100644 --- a/rtc_tools/video_file_writer_unittest.cc +++ b/rtc_tools/video_file_writer_unittest.cc @@ -26,8 +26,7 @@ namespace test { class VideoFileWriterTest : public ::testing::Test { public: void SetUp() override { - video_filename_ = - TempFilename(webrtc::test::OutputPath(), "test_video_file.y4m"); + video_filename_ = TempFilename(test::OutputPath(), "test_video_file.y4m"); // Create simple test video of size 6x4. FILE* file = fopen(video_filename_.c_str(), "wb"); @@ -46,7 +45,7 @@ class VideoFileWriterTest : public ::testing::Test { fclose(file); // Open the newly created file. - video_ = webrtc::test::OpenY4mFile(video_filename_); + video_ = test::OpenY4mFile(video_filename_); ASSERT_TRUE(video_); ASSERT_EQ(video_->number_of_frames(), 2u); } @@ -68,9 +67,9 @@ class VideoFileWriterTest : public ::testing::Test { } // Create an unique filename, e.g. test_video_file2.y4mZapata. written_video_filename_ = - TempFilename(webrtc::test::OutputPath(), "test_video_file2.y4m"); - webrtc::test::WriteY4mVideoToFile(video_, written_video_filename_, fps); - written_video_ = webrtc::test::OpenY4mFile(written_video_filename_); + TempFilename(test::OutputPath(), "test_video_file2.y4m"); + test::WriteY4mVideoToFile(video_, written_video_filename_, fps); + written_video_ = test::OpenY4mFile(written_video_filename_); ASSERT_TRUE(written_video_); } @@ -82,18 +81,17 @@ class VideoFileWriterTest : public ::testing::Test { } // Create an unique filename, e.g. test_video_file2.yuvZapata. written_video_filename_ = - TempFilename(webrtc::test::OutputPath(), "test_video_file2.yuv"); - webrtc::test::WriteYuvVideoToFile(video_, written_video_filename_, fps); - written_video_ = - webrtc::test::OpenYuvFile(written_video_filename_, width, height); + TempFilename(test::OutputPath(), "test_video_file2.yuv"); + test::WriteYuvVideoToFile(video_, written_video_filename_, fps); + written_video_ = test::OpenYuvFile(written_video_filename_, width, height); ASSERT_TRUE(written_video_); } const int width = 6; const int height = 4; const int fps = 60; - scoped_refptr video_; - scoped_refptr written_video_; + scoped_refptr video_; + scoped_refptr written_video_; // Each video object must be backed by file! std::string video_filename_; std::string written_video_filename_; diff --git a/test/network/emulated_turn_server.cc b/test/network/emulated_turn_server.cc index 6e5cc089ae..fe001d6f8d 100644 --- a/test/network/emulated_turn_server.cc +++ b/test/network/emulated_turn_server.cc @@ -87,8 +87,8 @@ namespace test { // sending data back into the emulated network. class EmulatedTURNServer::AsyncPacketSocketWrapper : public AsyncPacketSocket { public: - AsyncPacketSocketWrapper(webrtc::test::EmulatedTURNServer* turn_server, - webrtc::EmulatedEndpoint* endpoint, + AsyncPacketSocketWrapper(test::EmulatedTURNServer* turn_server, + EmulatedEndpoint* endpoint, uint16_t port) : turn_server_(turn_server), endpoint_(endpoint), @@ -126,8 +126,8 @@ class EmulatedTURNServer::AsyncPacketSocketWrapper : public AsyncPacketSocket { void SetError(int error) override {} private: - webrtc::test::EmulatedTURNServer* const turn_server_; - webrtc::EmulatedEndpoint* const endpoint_; + test::EmulatedTURNServer* const turn_server_; + EmulatedEndpoint* const endpoint_; const SocketAddress local_address_; }; @@ -182,7 +182,7 @@ AsyncPacketSocket* EmulatedTURNServer::Wrap(EmulatedEndpoint* endpoint) { return socket; } -void EmulatedTURNServer::OnPacketReceived(webrtc::EmulatedIpPacket packet) { +void EmulatedTURNServer::OnPacketReceived(EmulatedIpPacket packet) { // Copy from EmulatedEndpoint to webrtc::AsyncPacketSocket. thread_->PostTask([this, packet(std::move(packet))]() { RTC_DCHECK_RUN_ON(thread_.get()); diff --git a/test/network/fake_network_socket_server.cc b/test/network/fake_network_socket_server.cc index c9a1fb50a5..26e8911d98 100644 --- a/test/network/fake_network_socket_server.cc +++ b/test/network/fake_network_socket_server.cc @@ -296,7 +296,7 @@ void FakeNetworkSocketServer::SetMessageQueue(Thread* thread) { } // Always returns true (if return false, it won't be invoked again...) -bool FakeNetworkSocketServer::Wait(webrtc::TimeDelta max_wait_duration, +bool FakeNetworkSocketServer::Wait(TimeDelta max_wait_duration, bool process_io) { RTC_DCHECK(thread_ == Thread::Current()); if (!max_wait_duration.IsZero()) diff --git a/test/network/feedback_generator.cc b/test/network/feedback_generator.cc index 01887d0479..eedbb19a06 100644 --- a/test/network/feedback_generator.cc +++ b/test/network/feedback_generator.cc @@ -40,11 +40,6 @@ #include "test/network/simulated_network.h" namespace webrtc { -using ::webrtc::RtcEventRtcpPacketIncoming; -using ::webrtc::RtcEventRtpPacketOutgoing; -using ::webrtc::RtpPacketToSend; -using ::webrtc::TimeDelta; -using ::webrtc::TransportPacketsFeedback; namespace { Environment GetEnvironment(NetworkEmulationManager& net) { @@ -54,9 +49,8 @@ Environment GetEnvironment(NetworkEmulationManager& net) { return factory.Create(); } -webrtc::EmulatedRoute* CreateRoute( - webrtc::NetworkEmulationManager& net, - std::vector link) { +EmulatedRoute* CreateRoute(NetworkEmulationManager& net, + std::vector link) { if (!link.empty()) { return net.CreateRoute(link); } else { diff --git a/test/network/feedback_generator_unittest.cc b/test/network/feedback_generator_unittest.cc index b2d572567b..ff2073da05 100644 --- a/test/network/feedback_generator_unittest.cc +++ b/test/network/feedback_generator_unittest.cc @@ -98,8 +98,8 @@ TEST(FeedbackGeneratorTest, WritesToEventLog) { } TEST(FeedbackGeneratorWithoutNetworkTest, ReportsFeedbackForSentPackets) { - auto network_emulation_manager = CreateNetworkEmulationManager( - {.time_mode = webrtc::TimeMode::kSimulated}); + auto network_emulation_manager = + CreateNetworkEmulationManager({.time_mode = TimeMode::kSimulated}); std::unique_ptr gen = CreateFeedbackGeneratorWithoutNetwork( FeedbackGeneratorWithoutNetwork::Config(), diff --git a/test/network/network_emulation_pc_unittest.cc b/test/network/network_emulation_pc_unittest.cc index 5a248450ad..376471bbfc 100644 --- a/test/network/network_emulation_pc_unittest.cc +++ b/test/network/network_emulation_pc_unittest.cc @@ -92,7 +92,7 @@ scoped_refptr CreatePeerConnection( // This test does not support TCP rtc_configuration.port_allocator_config.flags = PORTALLOCATOR_DISABLE_TCP; if (turn_server != nullptr) { - webrtc::PeerConnectionInterface::IceServer server; + PeerConnectionInterface::IceServer server; server.username = turn_server->GetIceServerConfig().username; server.password = turn_server->GetIceServerConfig().username; server.urls.push_back(turn_server->GetIceServerConfig().url); @@ -161,7 +161,7 @@ TEST(NetworkEmulationManagerPCTest, Run) { std::move(bob_observer)); SendTask(signaling_thread.get(), [&]() { - scoped_refptr source = + scoped_refptr source = alice_pcf->CreateAudioSource(AudioOptions()); scoped_refptr track = alice_pcf->CreateAudioTrack("audio", source.get()); @@ -272,7 +272,7 @@ TEST(NetworkEmulationManagerPCTest, RunTURN) { std::move(bob_observer)); SendTask(signaling_thread.get(), [&]() { - scoped_refptr source = + scoped_refptr source = alice_pcf->CreateAudioSource(AudioOptions()); scoped_refptr track = alice_pcf->CreateAudioTrack("audio", source.get()); diff --git a/test/network/network_emulation_unittest.cc b/test/network/network_emulation_unittest.cc index eb158c0753..7ebd0d9075 100644 --- a/test/network/network_emulation_unittest.cc +++ b/test/network/network_emulation_unittest.cc @@ -86,7 +86,7 @@ class SocketReader : public sigslot::has_slots<> { return received_count_; } - webrtc::EcnMarking LastEcnMarking() const { + EcnMarking LastEcnMarking() const { MutexLock lock(&lock_); return last_ecn_mark_; } @@ -95,7 +95,7 @@ class SocketReader : public sigslot::has_slots<> { Socket* const socket_; Thread* const network_thread_; Buffer payload_; - webrtc::EcnMarking last_ecn_mark_; + EcnMarking last_ecn_mark_; mutable Mutex lock_; int received_count_ RTC_GUARDED_BY(lock_) = 0; @@ -440,7 +440,7 @@ TEST(NetworkEmulationManagerTest, EcnMarkingIsPropagated) { network_manager.time_controller()->AdvanceTime(TimeDelta::Seconds(1)); EXPECT_EQ(r2.ReceivedCount(), 1); - EXPECT_EQ(r2.LastEcnMarking(), webrtc::EcnMarking::kEct1); + EXPECT_EQ(r2.LastEcnMarking(), EcnMarking::kEct1); std::atomic received_stats_count{0}; nt1->GetStats([&](EmulatedNetworkStats st) { diff --git a/test/network/schedulable_network_behavior.cc b/test/network/schedulable_network_behavior.cc index 114754dc64..f602eb39f6 100644 --- a/test/network/schedulable_network_behavior.cc +++ b/test/network/schedulable_network_behavior.cc @@ -30,7 +30,6 @@ namespace webrtc { namespace { -using ::webrtc::BuiltInNetworkBehaviorConfig; void UpdateConfigFromSchedule( const network_behaviour::NetworkConfigScheduleItem& schedule_item, @@ -77,8 +76,8 @@ BuiltInNetworkBehaviorConfig GetInitialConfig( SchedulableNetworkBehavior::SchedulableNetworkBehavior( network_behaviour::NetworkConfigSchedule schedule, uint64_t random_seed, - webrtc::Clock& clock, - absl::AnyInvocable start_callback) + Clock& clock, + absl::AnyInvocable start_callback) : SimulatedNetwork(GetInitialConfig(schedule), random_seed), schedule_(std::move(schedule)), start_condition_(std::move(start_callback)), @@ -90,19 +89,18 @@ SchedulableNetworkBehavior::SchedulableNetworkBehavior( sequence_checker_.Detach(); } -bool SchedulableNetworkBehavior::EnqueuePacket( - webrtc::PacketInFlightInfo packet_info) { +bool SchedulableNetworkBehavior::EnqueuePacket(PacketInFlightInfo packet_info) { RTC_DCHECK_RUN_ON(&sequence_checker_); if (first_send_time_.IsInfinite() && - start_condition_(webrtc::Timestamp::Micros(packet_info.send_time_us))) { - first_send_time_ = webrtc::Timestamp::Micros(packet_info.send_time_us); + start_condition_(Timestamp::Micros(packet_info.send_time_us))) { + first_send_time_ = Timestamp::Micros(packet_info.send_time_us); if (schedule_.item().size() > 1) { RTC_CHECK_LT(next_schedule_index_, schedule_.item().size()); - webrtc::TimeDelta delay = - webrtc::TimeDelta::Millis(schedule_.item()[next_schedule_index_] - .time_since_first_sent_packet_ms()); + TimeDelta delay = + TimeDelta::Millis(schedule_.item()[next_schedule_index_] + .time_since_first_sent_packet_ms()); schedule_task_ = RepeatingTaskHandle::DelayedStart( - webrtc::TaskQueueBase::Current(), delay, + TaskQueueBase::Current(), delay, [this] { return UpdateConfigAndReschedule(); }); } } @@ -118,9 +116,8 @@ TimeDelta SchedulableNetworkBehavior::UpdateConfigAndReschedule() { UpdateConfigFromSchedule(next_config, config_); SimulatedNetwork::SetConfig(config_, reschedule_time); next_schedule_index_ = ++next_schedule_index_ % schedule_.item().size(); - webrtc::TimeDelta delay = webrtc::TimeDelta::Zero(); - webrtc::TimeDelta time_since_first_sent_packet = - reschedule_time - first_send_time_; + TimeDelta delay = TimeDelta::Zero(); + TimeDelta time_since_first_sent_packet = reschedule_time - first_send_time_; if (next_schedule_index_ != 0) { delay = std::max(TimeDelta::Millis(schedule_.item()[next_schedule_index_] .time_since_first_sent_packet_ms()) - @@ -136,8 +133,7 @@ TimeDelta SchedulableNetworkBehavior::UpdateConfigAndReschedule() { TimeDelta::Millis(schedule_.repeat_schedule_after_last_ms()) + TimeDelta::Millis(schedule_.item()[schedule_.item().size() - 1] .time_since_first_sent_packet_ms()); - delay = - webrtc::TimeDelta::Millis(schedule_.repeat_schedule_after_last_ms()); + delay = TimeDelta::Millis(schedule_.repeat_schedule_after_last_ms()); RTC_DCHECK_GE(delay, TimeDelta::Zero()); } From 2b87a42aa568cfb815b9863730d8fb3337ed9a1f Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Fri, 9 May 2025 10:35:38 +0000 Subject: [PATCH 112/292] [cleanup] Fix redundant webrtc name specifier This CL was uploaded by git cl split. R=hta@webrtc.org No-Iwyu: LSC Bug: webrtc:42232595 Change-Id: Ibcffecf0d1e04b73850501f0f2cb88f39384a79e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390620 Reviewed-by: Harald Alvestrand Auto-Submit: Evan Shrubsole Commit-Queue: Evan Shrubsole Cr-Commit-Position: refs/heads/main@{#44584} --- common_video/bitrate_adjuster_unittest.cc | 2 +- common_video/libyuv/libyuv_unittest.cc | 18 ++- common_video/video_frame_unittest.cc | 16 +-- .../rtc_event_log_encoder_new_format.cc | 14 +- logging/rtc_event_log/rtc_event_log_parser.cc | 5 +- video/buffered_frame_decryptor.cc | 7 +- video/config/encoder_stream_factory.cc | 82 ++++++------ .../config/encoder_stream_factory_unittest.cc | 3 +- video/config/simulcast.cc | 120 ++++++++---------- video/config/simulcast_unittest.cc | 74 ++++++----- ...rame_instrumentation_generator_unittest.cc | 2 +- video/end_to_end_tests/bandwidth_tests.cc | 2 +- .../resolution_bitrate_limits_tests.cc | 36 +++--- video/end_to_end_tests/rtp_rtcp_tests.cc | 2 +- video/frame_cadence_adapter_unittest.cc | 2 +- video/frame_decode_timing.cc | 3 +- video/frame_decode_timing_unittest.cc | 4 +- .../frame_encode_metadata_writer_unittest.cc | 3 +- video/full_stack_tests.cc | 12 +- video/pc_full_stack_tests.cc | 24 ++-- video/receive_statistics_proxy.cc | 2 +- video/receive_statistics_proxy_unittest.cc | 96 +++++++------- video/rtp_video_stream_receiver2.cc | 2 +- video/rtp_video_stream_receiver2_unittest.cc | 16 +-- video/send_statistics_proxy.cc | 4 +- video/video_analyzer.cc | 8 +- video/video_loopback.cc | 4 +- video/video_quality_test.cc | 18 +-- video/video_receive_stream2.cc | 24 ++-- video/video_receive_stream2_unittest.cc | 14 +- video/video_send_stream_impl.cc | 16 +-- video/video_send_stream_impl_unittest.cc | 5 +- video/video_send_stream_tests.cc | 30 ++--- .../video_source_sink_controller_unittest.cc | 8 +- video/video_stream_encoder.cc | 20 ++- 35 files changed, 329 insertions(+), 369 deletions(-) diff --git a/common_video/bitrate_adjuster_unittest.cc b/common_video/bitrate_adjuster_unittest.cc index 1ff911c4be..8660533281 100644 --- a/common_video/bitrate_adjuster_unittest.cc +++ b/common_video/bitrate_adjuster_unittest.cc @@ -34,7 +34,7 @@ class BitrateAdjusterTest : public ::testing::Test { const size_t frame_size_bytes = (bitrate_bps * frame_interval_ms) / (8 * 1000); for (size_t i = 0; i < update_frame_interval; ++i) { - clock_.AdvanceTime(webrtc::TimeDelta::Millis(frame_interval_ms)); + clock_.AdvanceTime(TimeDelta::Millis(frame_interval_ms)); adjuster_.Update(frame_size_bytes); } } diff --git a/common_video/libyuv/libyuv_unittest.cc b/common_video/libyuv/libyuv_unittest.cc index fbaf67cad4..b0e97943ea 100644 --- a/common_video/libyuv/libyuv_unittest.cc +++ b/common_video/libyuv/libyuv_unittest.cc @@ -97,8 +97,7 @@ TestLibYuv::TestLibYuv() frame_length_(CalcBufferSize(VideoType::kI420, 352, 288)) {} void TestLibYuv::SetUp() { - const std::string input_file_name = - webrtc::test::ResourcePath("foreman_cif", "yuv"); + const std::string input_file_name = test::ResourcePath("foreman_cif", "yuv"); source_file_ = fopen(input_file_name.c_str(), "rb"); ASSERT_TRUE(source_file_ != NULL) << "Cannot read file: " << input_file_name << "\n"; @@ -106,12 +105,11 @@ void TestLibYuv::SetUp() { scoped_refptr buffer( test::ReadI420Buffer(width_, height_, source_file_)); - orig_frame_ = - std::make_unique(VideoFrame::Builder() - .set_video_frame_buffer(buffer) - .set_rotation(webrtc::kVideoRotation_0) - .set_timestamp_us(0) - .build()); + orig_frame_ = std::make_unique(VideoFrame::Builder() + .set_video_frame_buffer(buffer) + .set_rotation(kVideoRotation_0) + .set_timestamp_us(0) + .build()); } void TestLibYuv::TearDown() { @@ -125,7 +123,7 @@ TEST_F(TestLibYuv, ConvertTest) { // Reading YUV frame - testing on the first frame of the foreman sequence int j = 0; std::string output_file_name = - webrtc::test::OutputPath() + "LibYuvTest_conversion.yuv"; + test::OutputPath() + "LibYuvTest_conversion.yuv"; FILE* output_file = fopen(output_file_name.c_str(), "wb"); ASSERT_TRUE(output_file != NULL); @@ -291,7 +289,7 @@ TEST_F(TestLibYuv, ConvertTest) { TEST_F(TestLibYuv, ConvertAlignedFrame) { // Reading YUV frame - testing on the first frame of the foreman sequence std::string output_file_name = - webrtc::test::OutputPath() + "LibYuvTest_conversion.yuv"; + test::OutputPath() + "LibYuvTest_conversion.yuv"; FILE* output_file = fopen(output_file_name.c_str(), "wb"); ASSERT_TRUE(output_file != NULL); diff --git a/common_video/video_frame_unittest.cc b/common_video/video_frame_unittest.cc index c2611aa3ff..21ddf8d85e 100644 --- a/common_video/video_frame_unittest.cc +++ b/common_video/video_frame_unittest.cc @@ -146,7 +146,7 @@ void CheckCrop(const T& frame, template void CheckRotate(int width, int height, - webrtc::VideoRotation rotation, + VideoRotation rotation, const T& rotated) { int rotated_width = width; int rotated_height = height; @@ -204,7 +204,7 @@ TEST(TestVideoFrame, WidthHeightValues) { VideoFrame frame = VideoFrame::Builder() .set_video_frame_buffer(I420Buffer::Create(10, 10, 10, 14, 90)) - .set_rotation(webrtc::kVideoRotation_0) + .set_rotation(kVideoRotation_0) .set_timestamp_ms(789) .build(); const int valid_value = 10; @@ -251,10 +251,8 @@ TEST(TestVideoFrame, ShallowCopy) { VideoFrame frame2(frame1); EXPECT_EQ(frame1.video_frame_buffer(), frame2.video_frame_buffer()); - const webrtc::I420BufferInterface* yuv1 = - frame1.video_frame_buffer()->GetI420(); - const webrtc::I420BufferInterface* yuv2 = - frame2.video_frame_buffer()->GetI420(); + const I420BufferInterface* yuv1 = frame1.video_frame_buffer()->GetI420(); + const I420BufferInterface* yuv2 = frame2.video_frame_buffer()->GetI420(); EXPECT_EQ(yuv1->DataY(), yuv2->DataY()); EXPECT_EQ(yuv1->DataU(), yuv2->DataU()); EXPECT_EQ(yuv1->DataV(), yuv2->DataV()); @@ -276,8 +274,8 @@ TEST(TestVideoFrame, ShallowCopy) { } TEST(TestVideoFrame, TextureInitialValues) { - VideoFrame frame = test::FakeNativeBuffer::CreateFrame( - 640, 480, 100, 10, webrtc::kVideoRotation_0); + VideoFrame frame = + test::FakeNativeBuffer::CreateFrame(640, 480, 100, 10, kVideoRotation_0); EXPECT_EQ(640, frame.width()); EXPECT_EQ(480, frame.height()); EXPECT_EQ(100u, frame.rtp_timestamp()); @@ -425,7 +423,7 @@ class TestPlanarYuvBufferRotate : public ::testing::Test { TYPED_TEST_SUITE_P(TestPlanarYuvBufferRotate); TYPED_TEST_P(TestPlanarYuvBufferRotate, Rotates) { - for (const webrtc::VideoRotation& rotation : this->RotationParams) { + for (const VideoRotation& rotation : this->RotationParams) { scoped_refptr buffer = CreateGradient(640, 480); scoped_refptr rotated_buffer = TypeParam::Rotate(*buffer, rotation); diff --git a/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc b/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc index d6fce461d0..38bf7cf165 100644 --- a/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc +++ b/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc @@ -170,19 +170,19 @@ bool ConvertToProtoFormat(const std::vector& extensions, } rtclog2::DtlsTransportStateEvent::DtlsTransportState ConvertToProtoFormat( - webrtc::DtlsTransportState state) { + DtlsTransportState state) { switch (state) { - case webrtc::DtlsTransportState::kNew: + case DtlsTransportState::kNew: return rtclog2::DtlsTransportStateEvent::DTLS_TRANSPORT_NEW; - case webrtc::DtlsTransportState::kConnecting: + case DtlsTransportState::kConnecting: return rtclog2::DtlsTransportStateEvent::DTLS_TRANSPORT_CONNECTING; - case webrtc::DtlsTransportState::kConnected: + case DtlsTransportState::kConnected: return rtclog2::DtlsTransportStateEvent::DTLS_TRANSPORT_CONNECTED; - case webrtc::DtlsTransportState::kClosed: + case DtlsTransportState::kClosed: return rtclog2::DtlsTransportStateEvent::DTLS_TRANSPORT_CLOSED; - case webrtc::DtlsTransportState::kFailed: + case DtlsTransportState::kFailed: return rtclog2::DtlsTransportStateEvent::DTLS_TRANSPORT_FAILED; - case webrtc::DtlsTransportState::kNumValues: + case DtlsTransportState::kNumValues: RTC_DCHECK_NOTREACHED(); } RTC_DCHECK_NOTREACHED(); diff --git a/logging/rtc_event_log/rtc_event_log_parser.cc b/logging/rtc_event_log/rtc_event_log_parser.cc index aff018197a..61ab4f5ec9 100644 --- a/logging/rtc_event_log/rtc_event_log_parser.cc +++ b/logging/rtc_event_log/rtc_event_log_parser.cc @@ -1123,8 +1123,7 @@ ParsedRtcEventLog::LoggedRtpStreamView::LoggedRtpStreamView( // is not stored in the event log. // TODO(ivoc): Remove this once this mapping is stored in the event log for // audio streams. Tracking bug: webrtc:6399 -webrtc::RtpHeaderExtensionMap -ParsedRtcEventLog::GetDefaultHeaderExtensionMap() { +RtpHeaderExtensionMap ParsedRtcEventLog::GetDefaultHeaderExtensionMap() { // Values from before the default RTP header extension IDs were removed. constexpr int kAudioLevelDefaultId = 1; constexpr int kTimestampOffsetDefaultId = 2; @@ -1136,7 +1135,7 @@ ParsedRtcEventLog::GetDefaultHeaderExtensionMap() { constexpr int kVideoTimingDefaultId = 8; constexpr int kDependencyDescriptorDefaultId = 9; - webrtc::RtpHeaderExtensionMap default_map(/*extmap_allow_mixed=*/true); + RtpHeaderExtensionMap default_map(/*extmap_allow_mixed=*/true); default_map.Register(kAudioLevelDefaultId); default_map.Register(kTimestampOffsetDefaultId); default_map.Register(kAbsSendTimeDefaultId); diff --git a/video/buffered_frame_decryptor.cc b/video/buffered_frame_decryptor.cc index 29ec8aec22..03fc2eb567 100644 --- a/video/buffered_frame_decryptor.cc +++ b/video/buffered_frame_decryptor.cc @@ -73,7 +73,7 @@ BufferedFrameDecryptor::FrameDecision BufferedFrameDecryptor::DecryptFrame( } // Retrieve the maximum possible size of the decrypted payload. const size_t max_plaintext_byte_size = - frame_decryptor_->GetMaxPlaintextByteSize(webrtc::MediaType::VIDEO, + frame_decryptor_->GetMaxPlaintextByteSize(MediaType::VIDEO, frame->size()); RTC_CHECK_LE(max_plaintext_byte_size, frame->size()); // Place the decrypted frame inline into the existing frame. @@ -88,9 +88,8 @@ BufferedFrameDecryptor::FrameDecision BufferedFrameDecryptor::DecryptFrame( // Attempt to decrypt the video frame. const FrameDecryptorInterface::Result decrypt_result = - frame_decryptor_->Decrypt(webrtc::MediaType::VIDEO, /*csrcs=*/{}, - additional_data, *frame, - inline_decrypted_bitstream); + frame_decryptor_->Decrypt(MediaType::VIDEO, /*csrcs=*/{}, additional_data, + *frame, inline_decrypted_bitstream); // Optionally call the callback if there was a change in status if (decrypt_result.status != last_status_) { last_status_ = decrypt_result.status; diff --git a/video/config/encoder_stream_factory.cc b/video/config/encoder_stream_factory.cc index 601c750d36..8820c98944 100644 --- a/video/config/encoder_stream_factory.cc +++ b/video/config/encoder_stream_factory.cc @@ -42,7 +42,6 @@ namespace webrtc { namespace { -using ::webrtc::FieldTrialsView; const int kMinLayerSize = 16; @@ -73,10 +72,10 @@ bool IsScaleFactorsPowerOfTwo(const VideoEncoderConfig& config) { } bool IsTemporalLayersSupported(VideoCodecType codec_type) { - return codec_type == webrtc::VideoCodecType::kVideoCodecVP8 || - codec_type == webrtc::VideoCodecType::kVideoCodecVP9 || - codec_type == webrtc::VideoCodecType::kVideoCodecAV1 || - codec_type == webrtc::VideoCodecType::kVideoCodecH265; + return codec_type == VideoCodecType::kVideoCodecVP8 || + codec_type == VideoCodecType::kVideoCodecVP9 || + codec_type == VideoCodecType::kVideoCodecAV1 || + codec_type == VideoCodecType::kVideoCodecH265; } size_t FindRequiredActiveLayers(const VideoEncoderConfig& encoder_config) { @@ -111,14 +110,14 @@ static int GetMaxDefaultVideoBitrateKbps(int width, int GetDefaultMaxQp(VideoCodecType codec_type) { switch (codec_type) { - case webrtc::kVideoCodecH264: - case webrtc::kVideoCodecH265: + case kVideoCodecH264: + case kVideoCodecH265: return kDefaultVideoMaxQpH26x; - case webrtc::kVideoCodecVP8: - case webrtc::kVideoCodecVP9: - case webrtc::kVideoCodecGeneric: + case kVideoCodecVP8: + case kVideoCodecVP9: + case kVideoCodecGeneric: return kDefaultVideoMaxQpVpx; - case webrtc::kVideoCodecAV1: + case kVideoCodecAV1: return kDefaultVideoMaxQpAv1; } } @@ -131,7 +130,7 @@ int NormalizeSimulcastSize(const FieldTrialsView& field_trials, size_t simulcast_layers) { int base2_exponent = static_cast(simulcast_layers) - 1; const std::optional experimental_base2_exponent = - webrtc::NormalizeSimulcastSizeExperiment::GetBase2Exponent(field_trials); + NormalizeSimulcastSizeExperiment::GetBase2Exponent(field_trials); if (experimental_base2_exponent && (size > (1 << *experimental_base2_exponent))) { base2_exponent = *experimental_base2_exponent; @@ -149,10 +148,10 @@ void OverrideStreamSettings( // Allow an experiment to override the minimum bitrate for the lowest // spatial layer. The experiment's configuration has the lowest priority. - layers[0].min_bitrate_bps = experimental_min_bitrate - .value_or(webrtc::DataRate::BitsPerSec( - webrtc::kDefaultMinVideoBitrateBps)) - .bps(); + layers[0].min_bitrate_bps = + experimental_min_bitrate + .value_or(DataRate::BitsPerSec(kDefaultMinVideoBitrateBps)) + .bps(); const bool temporal_layers_supported = IsTemporalLayersSupported(encoder_config.codec_type); @@ -220,14 +219,14 @@ void OverrideStreamSettings( bool is_highest_layer_max_bitrate_configured = encoder_config.simulcast_layers[layers.size() - 1].max_bitrate_bps > 0; - bool is_screencast = encoder_config.content_type == - webrtc::VideoEncoderConfig::ContentType::kScreen; + bool is_screencast = + encoder_config.content_type == VideoEncoderConfig::ContentType::kScreen; if (!is_screencast && !is_highest_layer_max_bitrate_configured && encoder_config.max_bitrate_bps > 0) { // No application-configured maximum for the largest layer. // If there is bitrate leftover, give it to the largest layer. - webrtc::BoostMaxSimulcastLayer( - webrtc::DataRate::BitsPerSec(encoder_config.max_bitrate_bps), &layers); + BoostMaxSimulcastLayer(DataRate::BitsPerSec(encoder_config.max_bitrate_bps), + &layers); } // Sort the layers by max_bitrate_bps, they might not always be from @@ -259,7 +258,7 @@ void OverrideStreamSettings( } // namespace EncoderStreamFactory::EncoderStreamFactory( - const webrtc::VideoEncoder::EncoderInfo& encoder_info, + const VideoEncoder::EncoderInfo& encoder_info, std::optional restrictions) : encoder_info_requested_resolution_alignment_( encoder_info.requested_resolution_alignment), @@ -296,7 +295,7 @@ std::vector EncoderStreamFactory::CreateEncoderStreams( std::vector streams; if (is_simulcast || - webrtc::SimulcastUtility::IsConferenceModeScreenshare(encoder_config)) { + SimulcastUtility::IsConferenceModeScreenshare(encoder_config)) { streams = CreateSimulcastOrConferenceModeScreenshareStreams( trials, frame_width, frame_height, encoder_config, experimental_min_bitrate); @@ -318,8 +317,8 @@ std::vector EncoderStreamFactory::CreateDefaultVideoStreams( int height, const VideoEncoderConfig& encoder_config, const std::optional& experimental_min_bitrate) const { - bool is_screencast = encoder_config.content_type == - webrtc::VideoEncoderConfig::ContentType::kScreen; + bool is_screencast = + encoder_config.content_type == VideoEncoderConfig::ContentType::kScreen; // The max bitrate specified by the API. // - `encoder_config.simulcast_layers[0].max_bitrate_bps` comes from the first @@ -346,8 +345,8 @@ std::vector EncoderStreamFactory::CreateDefaultVideoStreams( int min_bitrate_bps = experimental_min_bitrate - ? webrtc::saturated_cast(experimental_min_bitrate->bps()) - : webrtc::kDefaultMinVideoBitrateBps; + ? saturated_cast(experimental_min_bitrate->bps()) + : kDefaultMinVideoBitrateBps; if (encoder_config.simulcast_layers[0].min_bitrate_bps > 0) { // Use set min bitrate. min_bitrate_bps = encoder_config.simulcast_layers[0].min_bitrate_bps; @@ -388,7 +387,7 @@ std::vector EncoderStreamFactory::CreateDefaultVideoStreams( kMinLayerSize); } - if (encoder_config.codec_type == webrtc::VideoCodecType::kVideoCodecVP9) { + if (encoder_config.codec_type == VideoCodecType::kVideoCodecVP9) { RTC_DCHECK(encoder_config.encoder_specific_settings); // Use VP9 SVC layering from codec settings which might be initialized // though field trial in ConfigureVideoEncoderSettings. @@ -407,12 +406,12 @@ std::vector EncoderStreamFactory::CreateDefaultVideoStreams( (layer.num_temporal_layers > 1u || num_spatial_layers > 1)) { // In SVC mode, the VP9 max bitrate is determined by SvcConfig, instead of // GetMaxDefaultVideoBitrateKbps(). - std::vector svc_layers = - webrtc::GetSvcConfig(width, height, max_framerate, - /*first_active_layer=*/0, num_spatial_layers, - *layer.num_temporal_layers, is_screencast); + std::vector svc_layers = + GetSvcConfig(width, height, max_framerate, + /*first_active_layer=*/0, num_spatial_layers, + *layer.num_temporal_layers, is_screencast); int sum_max_bitrates_kbps = 0; - for (const webrtc::SpatialLayer& spatial_layer : svc_layers) { + for (const SpatialLayer& spatial_layer : svc_layers) { sum_max_bitrates_kbps += spatial_layer.maxBitrate; } RTC_DCHECK_GE(sum_max_bitrates_kbps, 0); @@ -468,9 +467,9 @@ EncoderStreamFactory::CreateSimulcastOrConferenceModeScreenshareStreams( // Use legacy simulcast screenshare if conference mode is explicitly enabled // or use the regular simulcast configuration path which is generic. - std::vector layers = webrtc::GetSimulcastConfig( + std::vector layers = GetSimulcastConfig( resolutions, - webrtc::SimulcastUtility::IsConferenceModeScreenshare(encoder_config), + SimulcastUtility::IsConferenceModeScreenshare(encoder_config), IsTemporalLayersSupported(encoder_config.codec_type), trials, encoder_config.codec_type); @@ -509,8 +508,8 @@ Resolution EncoderStreamFactory::GetLayerResolutionFromScaleResolutionDownTo( VideoSinkWants wants; wants.is_active = true; wants.target_pixel_count = restrictions_->target_pixels_per_frame(); - wants.max_pixel_count = webrtc::dchecked_cast( - restrictions_->max_pixels_per_frame().value_or( + wants.max_pixel_count = + dchecked_cast(restrictions_->max_pixels_per_frame().value_or( std::numeric_limits::max())); wants.aggregates.emplace(VideoSinkWants::Aggregates()); wants.resolution_alignment = encoder_info_requested_resolution_alignment_; @@ -532,7 +531,7 @@ std::vector EncoderStreamFactory::GetStreamResolutions( int height, const VideoEncoderConfig& encoder_config) const { std::vector resolutions; - if (webrtc::SimulcastUtility::IsConferenceModeScreenshare(encoder_config)) { + if (SimulcastUtility::IsConferenceModeScreenshare(encoder_config)) { for (size_t i = 0; i < encoder_config.number_of_streams; ++i) { resolutions.push_back({.width = width, .height = height}); } @@ -540,7 +539,7 @@ std::vector EncoderStreamFactory::GetStreamResolutions( size_t min_num_layers = FindRequiredActiveLayers(encoder_config); size_t max_num_layers = !encoder_config.HasScaleResolutionDownTo() - ? webrtc::LimitSimulcastLayerCount( + ? LimitSimulcastLayerCount( min_num_layers, encoder_config.number_of_streams, width, height, trials, encoder_config.codec_type) : encoder_config.number_of_streams; @@ -556,8 +555,8 @@ std::vector EncoderStreamFactory::GetStreamResolutions( if (encoder_config.HasScaleResolutionDownTo() && restrictions_.has_value() && restrictions_->max_pixels_per_frame().has_value()) { - int max_pixels = webrtc::dchecked_cast( - restrictions_->max_pixels_per_frame().value()); + int max_pixels = + dchecked_cast(restrictions_->max_pixels_per_frame().value()); int prev_pixel_count = encoder_config.simulcast_layers[0] .scale_resolution_down_to.value_or(Resolution()) @@ -592,8 +591,7 @@ std::vector EncoderStreamFactory::GetStreamResolutions( } const bool norm_size_configured = - webrtc::NormalizeSimulcastSizeExperiment::GetBase2Exponent(trials) - .has_value(); + NormalizeSimulcastSizeExperiment::GetBase2Exponent(trials).has_value(); const int normalized_width = (default_scale_factors_used || norm_size_configured) && (width >= kMinLayerSize) diff --git a/video/config/encoder_stream_factory_unittest.cc b/video/config/encoder_stream_factory_unittest.cc index a36efaab01..5c93527282 100644 --- a/video/config/encoder_stream_factory_unittest.cc +++ b/video/config/encoder_stream_factory_unittest.cc @@ -42,7 +42,6 @@ using ::testing::Not; using ::testing::SizeIs; using ::testing::TestWithParam; using ::testing::Values; -using ::webrtc::EncoderStreamFactory; struct CreateVideoStreamParams { int width = 0; @@ -482,7 +481,7 @@ INSTANTIATE_TEST_SUITE_P( .scalability_mode = ScalabilityMode::kL1T2})}}), Values( #ifdef RTC_ENABLE_H265 - webrtc::kVideoCodecH265, + kVideoCodecH265, #endif VideoCodecType::kVideoCodecAV1, VideoCodecType::kVideoCodecVP9))); diff --git a/video/config/simulcast.cc b/video/config/simulcast.cc index 66e9005282..acc2e15d51 100644 --- a/video/config/simulcast.cc +++ b/video/config/simulcast.cc @@ -37,7 +37,6 @@ namespace webrtc { namespace { -using ::webrtc::FieldTrialsView; constexpr char kUseLegacySimulcastLayerLimitFieldTrial[] = "WebRTC-LegacySimulcastLayerLimit"; @@ -47,16 +46,16 @@ constexpr double kDefaultMaxRoundupRate = 0.1; // Limits for legacy conference screensharing mode. Currently used for the // lower of the two simulcast streams. constexpr DataRate kScreenshareDefaultTl0Bitrate = - webrtc::DataRate::KilobitsPerSec(200); + DataRate::KilobitsPerSec(200); constexpr DataRate kScreenshareDefaultTl1Bitrate = - webrtc::DataRate::KilobitsPerSec(1000); + DataRate::KilobitsPerSec(1000); // Min/max bitrate for the higher one of the two simulcast stream used for // screen content. constexpr DataRate kScreenshareHighStreamMinBitrate = - webrtc::DataRate::KilobitsPerSec(600); + DataRate::KilobitsPerSec(600); constexpr DataRate kScreenshareHighStreamMaxBitrate = - webrtc::DataRate::KilobitsPerSec(1250); + DataRate::KilobitsPerSec(1250); constexpr int kDefaultNumTemporalLayers = 3; constexpr int kScreenshareMaxSimulcastLayers = 2; @@ -83,62 +82,47 @@ struct SimulcastFormat { // simulcast layers at what bitrates (maximum, target, and minimum). // Important!! Keep this table from high resolution to low resolution. constexpr const SimulcastFormat kSimulcastFormatsVP8[] = { - {1920, 1080, 3, webrtc::DataRate::KilobitsPerSec(5000), - webrtc::DataRate::KilobitsPerSec(4000), - webrtc::DataRate::KilobitsPerSec(800)}, - {1280, 720, 3, webrtc::DataRate::KilobitsPerSec(2500), - webrtc::DataRate::KilobitsPerSec(2500), - webrtc::DataRate::KilobitsPerSec(600)}, - {960, 540, 3, webrtc::DataRate::KilobitsPerSec(1200), - webrtc::DataRate::KilobitsPerSec(1200), - webrtc::DataRate::KilobitsPerSec(350)}, - {640, 360, 2, webrtc::DataRate::KilobitsPerSec(700), - webrtc::DataRate::KilobitsPerSec(500), - webrtc::DataRate::KilobitsPerSec(150)}, - {480, 270, 2, webrtc::DataRate::KilobitsPerSec(450), - webrtc::DataRate::KilobitsPerSec(350), - webrtc::DataRate::KilobitsPerSec(150)}, - {320, 180, 1, webrtc::DataRate::KilobitsPerSec(200), - webrtc::DataRate::KilobitsPerSec(150), - webrtc::DataRate::KilobitsPerSec(30)}, + {1920, 1080, 3, DataRate::KilobitsPerSec(5000), + DataRate::KilobitsPerSec(4000), DataRate::KilobitsPerSec(800)}, + {1280, 720, 3, DataRate::KilobitsPerSec(2500), + DataRate::KilobitsPerSec(2500), DataRate::KilobitsPerSec(600)}, + {960, 540, 3, DataRate::KilobitsPerSec(1200), + DataRate::KilobitsPerSec(1200), DataRate::KilobitsPerSec(350)}, + {640, 360, 2, DataRate::KilobitsPerSec(700), DataRate::KilobitsPerSec(500), + DataRate::KilobitsPerSec(150)}, + {480, 270, 2, DataRate::KilobitsPerSec(450), DataRate::KilobitsPerSec(350), + DataRate::KilobitsPerSec(150)}, + {320, 180, 1, DataRate::KilobitsPerSec(200), DataRate::KilobitsPerSec(150), + DataRate::KilobitsPerSec(30)}, // As the resolution goes down, interpolate the target and max bitrates down // towards zero. The min bitrate is still limited at 30 kbps and the target // and the max will be capped from below accordingly. - {0, 0, 1, webrtc::DataRate::KilobitsPerSec(0), - webrtc::DataRate::KilobitsPerSec(0), - webrtc::DataRate::KilobitsPerSec(30)}}; + {0, 0, 1, DataRate::KilobitsPerSec(0), DataRate::KilobitsPerSec(0), + DataRate::KilobitsPerSec(30)}}; // These tables describe from which resolution we can use how many // simulcast layers at what bitrates (maximum, target, and minimum). // Important!! Keep this table from high resolution to low resolution. constexpr const SimulcastFormat kSimulcastFormatsVP9[] = { - {1920, 1080, 3, webrtc::DataRate::KilobitsPerSec(3367), - webrtc::DataRate::KilobitsPerSec(3367), - webrtc::DataRate::KilobitsPerSec(769)}, - {1280, 720, 3, webrtc::DataRate::KilobitsPerSec(1524), - webrtc::DataRate::KilobitsPerSec(1524), - webrtc::DataRate::KilobitsPerSec(481)}, - {960, 540, 3, webrtc::DataRate::KilobitsPerSec(879), - webrtc::DataRate::KilobitsPerSec(879), - webrtc::DataRate::KilobitsPerSec(337)}, - {640, 360, 2, webrtc::DataRate::KilobitsPerSec(420), - webrtc::DataRate::KilobitsPerSec(420), - webrtc::DataRate::KilobitsPerSec(193)}, - {480, 270, 2, webrtc::DataRate::KilobitsPerSec(257), - webrtc::DataRate::KilobitsPerSec(257), - webrtc::DataRate::KilobitsPerSec(121)}, - {320, 180, 1, webrtc::DataRate::KilobitsPerSec(142), - webrtc::DataRate::KilobitsPerSec(142), - webrtc::DataRate::KilobitsPerSec(30)}, - {240, 135, 1, webrtc::DataRate::KilobitsPerSec(101), - webrtc::DataRate::KilobitsPerSec(101), - webrtc::DataRate::KilobitsPerSec(30)}, + {1920, 1080, 3, DataRate::KilobitsPerSec(3367), + DataRate::KilobitsPerSec(3367), DataRate::KilobitsPerSec(769)}, + {1280, 720, 3, DataRate::KilobitsPerSec(1524), + DataRate::KilobitsPerSec(1524), DataRate::KilobitsPerSec(481)}, + {960, 540, 3, DataRate::KilobitsPerSec(879), DataRate::KilobitsPerSec(879), + DataRate::KilobitsPerSec(337)}, + {640, 360, 2, DataRate::KilobitsPerSec(420), DataRate::KilobitsPerSec(420), + DataRate::KilobitsPerSec(193)}, + {480, 270, 2, DataRate::KilobitsPerSec(257), DataRate::KilobitsPerSec(257), + DataRate::KilobitsPerSec(121)}, + {320, 180, 1, DataRate::KilobitsPerSec(142), DataRate::KilobitsPerSec(142), + DataRate::KilobitsPerSec(30)}, + {240, 135, 1, DataRate::KilobitsPerSec(101), DataRate::KilobitsPerSec(101), + DataRate::KilobitsPerSec(30)}, // As the resolution goes down, interpolate the target and max bitrates down // towards zero. The min bitrate is still limited at 30 kbps and the target // and the max will be capped from below accordingly. - {0, 0, 1, webrtc::DataRate::KilobitsPerSec(0), - webrtc::DataRate::KilobitsPerSec(0), - webrtc::DataRate::KilobitsPerSec(30)}}; + {0, 0, 1, DataRate::KilobitsPerSec(0), DataRate::KilobitsPerSec(0), + DataRate::KilobitsPerSec(30)}}; constexpr DataRate Interpolate(const DataRate& a, const DataRate& b, @@ -154,16 +138,16 @@ bool EnableLowresBitrateInterpolation(const FieldTrialsView& trials) { int GetDefaultSimulcastTemporalLayers(VideoCodecType codec) { switch (codec) { - case webrtc::kVideoCodecVP8: - case webrtc::kVideoCodecVP9: - case webrtc::kVideoCodecAV1: - case webrtc::kVideoCodecH264: - case webrtc::kVideoCodecGeneric: + case kVideoCodecVP8: + case kVideoCodecVP9: + case kVideoCodecAV1: + case kVideoCodecH264: + case kVideoCodecGeneric: return kDefaultNumTemporalLayers; // For codec type that has no software fallback, defaults to L1T1 for // initial simulcast setup, as this is the only scalability mode secure to // be supported. - case webrtc::kVideoCodecH265: + case kVideoCodecH265: return 1; } RTC_DCHECK_NOTREACHED() << "Unsupported codec."; @@ -175,15 +159,15 @@ std::vector GetSimulcastFormats( VideoCodecType codec) { std::vector formats; switch (codec) { - case webrtc::kVideoCodecGeneric: - case webrtc::kVideoCodecVP8: - case webrtc::kVideoCodecH264: + case kVideoCodecGeneric: + case kVideoCodecVP8: + case kVideoCodecH264: formats.insert(formats.begin(), std::begin(kSimulcastFormatsVP8), std::end(kSimulcastFormatsVP8)); break; - case webrtc::kVideoCodecVP9: - case webrtc::kVideoCodecAV1: - case webrtc::kVideoCodecH265: + case kVideoCodecVP9: + case kVideoCodecAV1: + case kVideoCodecH265: formats.insert(formats.begin(), std::begin(kSimulcastFormatsVP9), std::end(kSimulcastFormatsVP9)); break; @@ -323,7 +307,7 @@ std::vector GetScreenshareLayers(size_t max_layers, layers[0].width = width; layers[0].height = height; layers[0].max_framerate = 5; - layers[0].min_bitrate_bps = webrtc::kDefaultMinVideoBitrateBps; + layers[0].min_bitrate_bps = kDefaultMinVideoBitrateBps; layers[0].target_bitrate_bps = kScreenshareDefaultTl0Bitrate.bps(); layers[0].max_bitrate_bps = kScreenshareDefaultTl1Bitrate.bps(); layers[0].num_temporal_layers = temporal_layers_supported ? 2 : 1; @@ -340,7 +324,7 @@ std::vector GetScreenshareLayers(size_t max_layers, // layers were enabled. max_bitrate_bps = static_cast( kScreenshareHighStreamMaxBitrate.bps() * - webrtc::SimulcastRateAllocator::GetTemporalRateAllocation( + SimulcastRateAllocator::GetTemporalRateAllocation( kScreenshareTemporalLayers, 0, base_heavy_tl3_rate_alloc)); } else { // Experimental temporal layer mode used, use increased max bitrate. @@ -378,8 +362,8 @@ size_t LimitSimulcastLayerCount(size_t min_num_layers, // than configured `max_ratio`. pixels_down is the selected index in // kSimulcastFormats based on pixels. FieldTrialOptional max_ratio("max_ratio"); - webrtc::ParseFieldTrial({&max_ratio}, - trials.Lookup("WebRTC-SimulcastLayerLimitRoundUp")); + ParseFieldTrial({&max_ratio}, + trials.Lookup("WebRTC-SimulcastLayerLimitRoundUp")); size_t reduced_num_layers = std::max(min_num_layers, @@ -413,14 +397,14 @@ void BoostMaxSimulcastLayer(DataRate max_bitrate, DataRate GetTotalMaxBitrate(const std::vector& layers) { if (layers.empty()) - return webrtc::DataRate::Zero(); + return DataRate::Zero(); int total_max_bitrate_bps = 0; for (size_t s = 0; s < layers.size() - 1; ++s) { total_max_bitrate_bps += layers[s].target_bitrate_bps; } total_max_bitrate_bps += layers.back().max_bitrate_bps; - return webrtc::DataRate::BitsPerSec(total_max_bitrate_bps); + return DataRate::BitsPerSec(total_max_bitrate_bps); } std::vector GetSimulcastConfig( diff --git a/video/config/simulcast_unittest.cc b/video/config/simulcast_unittest.cc index 94b48fea56..628e4ea131 100644 --- a/video/config/simulcast_unittest.cc +++ b/video/config/simulcast_unittest.cc @@ -73,7 +73,7 @@ const std::vector GetH265SimulcastBitrates720p() { std::vector CreateResolutions(int max_width, int max_height, int num_streams) { - std::vector resolutions(num_streams); + std::vector resolutions(num_streams); for (int i = 0; i < num_streams; ++i) { resolutions[i].width = max_width >> (num_streams - i - 1); resolutions[i].height = max_height >> (num_streams - i - 1); @@ -107,10 +107,10 @@ TEST(SimulcastTest, BandwidthAboveTotalMaxBitrateGivenToHighestStream) { streams[1].target_bitrate_bps = 200000; streams[2].max_bitrate_bps = 400000; - const webrtc::DataRate one_bps = webrtc::DataRate::BitsPerSec(1); + const DataRate one_bps = DataRate::BitsPerSec(1); // No bitrate above the total max to give to the highest stream. - const webrtc::DataRate max_total_bitrate = GetTotalMaxBitrate(streams); + const DataRate max_total_bitrate = GetTotalMaxBitrate(streams); BoostMaxSimulcastLayer(max_total_bitrate, &streams); EXPECT_EQ(400000, streams[2].max_bitrate_bps); EXPECT_EQ(max_total_bitrate, GetTotalMaxBitrate(streams)); @@ -129,7 +129,7 @@ TEST(SimulcastTest, GetConfig) { const size_t kMaxLayers = 3; std::vector streams = GetSimulcastConfig(CreateResolutions(1280, 720, kMaxLayers), - !kScreenshare, true, trials, webrtc::kVideoCodecVP8); + !kScreenshare, true, trials, kVideoCodecVP8); ASSERT_THAT(streams, SizeIs(kMaxLayers)); EXPECT_EQ(320u, streams[0].width); @@ -157,7 +157,7 @@ TEST(SimulcastTest, GetConfigH265) { const size_t kMaxLayers = 3; std::vector streams = GetSimulcastConfig(CreateResolutions(1280, 720, kMaxLayers), - !kScreenshare, true, trials, webrtc::kVideoCodecH265); + !kScreenshare, true, trials, kVideoCodecH265); ASSERT_THAT(streams, SizeIs(kMaxLayers)); EXPECT_EQ(320u, streams[0].width); @@ -188,7 +188,7 @@ TEST(SimulcastTest, GetConfigWithBaseHeavyVP8TL3RateAllocation) { const size_t kMaxLayers = 3; std::vector streams = GetSimulcastConfig(CreateResolutions(1280, 720, kMaxLayers), - !kScreenshare, true, trials, webrtc::kVideoCodecVP8); + !kScreenshare, true, trials, kVideoCodecVP8); ASSERT_THAT(streams, SizeIs(kMaxLayers)); EXPECT_EQ(kExpected[0].min_bitrate_bps, streams[0].min_bitrate_bps); @@ -209,7 +209,7 @@ TEST(SimulcastTest, GetConfigWithLimitedMaxLayers) { const size_t kMaxLayers = 2; std::vector streams = GetSimulcastConfig(CreateResolutions(1280, 720, kMaxLayers), - !kScreenshare, true, trials, webrtc::kVideoCodecVP8); + !kScreenshare, true, trials, kVideoCodecVP8); ASSERT_THAT(streams, SizeIs(kMaxLayers)); EXPECT_EQ(640u, streams[0].width); @@ -224,7 +224,7 @@ TEST(SimulcastTest, GetConfigForScreenshareSimulcast) { std::vector{{.width = 1400, .height = 800}, {.width = 1400, .height = 800}, {.width = 1400, .height = 800}}, - kScreenshare, true, trials, webrtc::kVideoCodecVP8); + kScreenshare, true, trials, kVideoCodecVP8); EXPECT_THAT(streams, SizeIs(2)); for (size_t i = 0; i < streams.size(); ++i) { @@ -244,7 +244,7 @@ TEST(SimulcastTest, GetConfigForScreenshareSimulcastWithLimitedMaxLayers) { ExplicitKeyValueConfig trials(""); std::vector streams = GetSimulcastConfig( std::vector{{.width = 1400, .height = 800}}, kScreenshare, - true, trials, webrtc::kVideoCodecVP8); + true, trials, kVideoCodecVP8); EXPECT_THAT(streams, SizeIs(1)); } @@ -252,7 +252,7 @@ TEST(SimulcastTest, AveragesBitratesForNonStandardResolution) { ExplicitKeyValueConfig trials(""); std::vector streams = GetSimulcastConfig(std::vector{{.width = 900, .height = 800}}, - !kScreenshare, true, trials, webrtc::kVideoCodecVP8); + !kScreenshare, true, trials, kVideoCodecVP8); ASSERT_THAT(streams, SizeIs(1)); EXPECT_EQ(900u, streams[0].width); @@ -273,7 +273,7 @@ TEST(SimulcastTest, BitratesForCloseToStandardResolution) { std::vector streams = GetSimulcastConfig(CreateResolutions(kWidth, kHeight, kMaxLayers), - !kScreenshare, true, trials, webrtc::kVideoCodecVP8); + !kScreenshare, true, trials, kVideoCodecVP8); ASSERT_THAT(streams, SizeIs(kMaxLayers)); EXPECT_EQ(kWidth, streams[2].width); @@ -296,18 +296,18 @@ TEST(SimulcastTest, MaxLayersWithRoundUpDisabled) { const int kMaxLayers = 3; size_t num_layers = LimitSimulcastLayerCount(kMinLayers, kMaxLayers, 960, 540, - trials, webrtc::kVideoCodecVP8); + trials, kVideoCodecVP8); EXPECT_EQ(num_layers, 3u); // <960x540: 2 layers num_layers = LimitSimulcastLayerCount(kMinLayers, kMaxLayers, 960, 539, - trials, webrtc::kVideoCodecVP8); + trials, kVideoCodecVP8); EXPECT_EQ(num_layers, 2u); num_layers = LimitSimulcastLayerCount(kMinLayers, kMaxLayers, 480, 270, - trials, webrtc::kVideoCodecVP8); + trials, kVideoCodecVP8); EXPECT_EQ(num_layers, 2u); // <480x270: 1 layer num_layers = LimitSimulcastLayerCount(kMinLayers, kMaxLayers, 480, 269, - trials, webrtc::kVideoCodecVP8); + trials, kVideoCodecVP8); EXPECT_EQ(num_layers, 1u); } @@ -318,24 +318,24 @@ TEST(SimulcastTest, MaxLayersWithDefaultRoundUpRatio) { const int kMaxLayers = 3; size_t num_layers = LimitSimulcastLayerCount(kMinLayers, kMaxLayers, 960, 540, - trials, webrtc::kVideoCodecVP8); + trials, kVideoCodecVP8); EXPECT_EQ(num_layers, 3u); // Lowest cropped height where max layers from higher resolution is used. num_layers = LimitSimulcastLayerCount(kMinLayers, kMaxLayers, 960, 512, - trials, webrtc::kVideoCodecVP8); + trials, kVideoCodecVP8); EXPECT_EQ(num_layers, 3u); num_layers = LimitSimulcastLayerCount(kMinLayers, kMaxLayers, 960, 508, - trials, webrtc::kVideoCodecVP8); + trials, kVideoCodecVP8); EXPECT_EQ(num_layers, 2u); num_layers = LimitSimulcastLayerCount(kMinLayers, kMaxLayers, 480, 270, - trials, webrtc::kVideoCodecVP8); + trials, kVideoCodecVP8); EXPECT_EQ(num_layers, 2u); // Lowest cropped height where max layers from higher resolution is used. num_layers = LimitSimulcastLayerCount(kMinLayers, kMaxLayers, 480, 256, - trials, webrtc::kVideoCodecVP8); + trials, kVideoCodecVP8); EXPECT_EQ(num_layers, 2u); num_layers = LimitSimulcastLayerCount(kMinLayers, kMaxLayers, 480, 254, - trials, webrtc::kVideoCodecVP8); + trials, kVideoCodecVP8); EXPECT_EQ(num_layers, 1u); } @@ -347,14 +347,14 @@ TEST(SimulcastTest, MaxLayersWithRoundUpRatio) { const int kMaxLayers = 3; size_t num_layers = LimitSimulcastLayerCount(kMinLayers, kMaxLayers, 480, 270, - trials, webrtc::kVideoCodecVP8); + trials, kVideoCodecVP8); EXPECT_EQ(num_layers, 2u); // Lowest cropped height where max layers from higher resolution is used. num_layers = LimitSimulcastLayerCount(kMinLayers, kMaxLayers, 480, 252, - trials, webrtc::kVideoCodecVP8); + trials, kVideoCodecVP8); EXPECT_EQ(num_layers, 2u); num_layers = LimitSimulcastLayerCount(kMinLayers, kMaxLayers, 480, 250, - trials, webrtc::kVideoCodecVP8); + trials, kVideoCodecVP8); EXPECT_EQ(num_layers, 1u); } @@ -366,7 +366,7 @@ TEST(SimulcastTest, BitratesInterpolatedForResBelow180p) { const size_t kMaxLayers = 3; std::vector streams = GetSimulcastConfig( CreateResolutions(/*max_width=*/960, /*max_height=*/540, kMaxLayers), - !kScreenshare, true, trials, webrtc::kVideoCodecVP8); + !kScreenshare, true, trials, kVideoCodecVP8); ASSERT_THAT(streams, SizeIs(kMaxLayers)); EXPECT_EQ(240u, streams[0].width); @@ -383,7 +383,7 @@ TEST(SimulcastTest, BitratesConsistentForVerySmallRes) { std::vector streams = GetSimulcastConfig(std::vector{{.width = 1, .height = 1}}, - !kScreenshare, true, trials, webrtc::kVideoCodecVP8); + !kScreenshare, true, trials, kVideoCodecVP8); ASSERT_THAT(streams, SizeIs(1)); EXPECT_EQ(1u, streams[0].width); @@ -401,7 +401,7 @@ TEST(SimulcastTest, const size_t kMaxLayers = 3; std::vector streams = GetSimulcastConfig( CreateResolutions(/*max_width=*/960, /*max_height=*/540, kMaxLayers), - !kScreenshare, true, trials, webrtc::kVideoCodecVP8); + !kScreenshare, true, trials, kVideoCodecVP8); ASSERT_THAT(streams, SizeIs(kMaxLayers)); EXPECT_EQ(240u, streams[0].width); @@ -419,10 +419,10 @@ struct BitrateLimitsTestParams { }; using BitrateLimitsTest = - TestWithParam>; + TestWithParam>; TEST_P(BitrateLimitsTest, VerifyBitrateLimits) { - const auto codec_type = std::get(GetParam()); + const auto codec_type = std::get(GetParam()); const auto test_params = std::get(GetParam()); ExplicitKeyValueConfig trials(""); std::vector streams = GetSimulcastConfig( @@ -442,7 +442,7 @@ TEST_P(BitrateLimitsTest, VerifyBitrateLimits) { INSTANTIATE_TEST_SUITE_P( Vp8H264, BitrateLimitsTest, - Combine(Values(webrtc::kVideoCodecVP8, webrtc::kVideoCodecH264), + Combine(Values(kVideoCodecVP8, kVideoCodecH264), Values(BitrateLimitsTestParams{.width = 1920, .height = 1080, .expected_min_bitrate_kbps{150, 350, @@ -460,8 +460,7 @@ INSTANTIATE_TEST_SUITE_P( .expected_min_bitrate_kbps{30, 150, 350}, .expected_max_bitrate_kbps{200, 450, 1200}})), [](const TestParamInfo& info) { - return CodecTypeToPayloadString( - std::get(info.param)) + + return CodecTypeToPayloadString(std::get(info.param)) + std::to_string( std::get(info.param).height); }); @@ -472,10 +471,10 @@ INSTANTIATE_TEST_SUITE_P( Combine( Values( #ifdef RTC_ENABLE_H265 - webrtc::kVideoCodecH265, + kVideoCodecH265, #endif - webrtc::kVideoCodecAV1, - webrtc::kVideoCodecVP9), + kVideoCodecAV1, + kVideoCodecVP9), Values( BitrateLimitsTestParams{.width = 1920, .height = 1080, @@ -491,8 +490,7 @@ INSTANTIATE_TEST_SUITE_P( .expected_min_bitrate_kbps{30, 121, 337}, .expected_max_bitrate_kbps{101, 257, 879}})), [](const TestParamInfo& info) { - return CodecTypeToPayloadString( - std::get(info.param)) + + return CodecTypeToPayloadString(std::get(info.param)) + std::to_string( std::get(info.param).height); }); @@ -506,7 +504,7 @@ TEST(SimulcastTest, GetConfigForH265) { const size_t kMaxLayers = 3; std::vector streams = GetSimulcastConfig(CreateResolutions(1280, 720, kMaxLayers), - !kScreenshare, true, trials, webrtc::kVideoCodecH265); + !kScreenshare, true, trials, kVideoCodecH265); ASSERT_THAT(streams, SizeIs(kMaxLayers)); for (size_t i = 0; i < streams.size(); ++i) { diff --git a/video/corruption_detection/frame_instrumentation_generator_unittest.cc b/video/corruption_detection/frame_instrumentation_generator_unittest.cc index d4362c012d..fff4cb497c 100644 --- a/video/corruption_detection/frame_instrumentation_generator_unittest.cc +++ b/video/corruption_detection/frame_instrumentation_generator_unittest.cc @@ -685,7 +685,7 @@ TEST(FrameInstrumentationGeneratorTest, QueuesAtMostThreeInputFrames) { VideoCodecType::kVideoCodecVP8); bool frames_destroyed[4] = {}; - class TestBuffer : public webrtc::I420Buffer { + class TestBuffer : public I420Buffer { public: TestBuffer(int width, int height, bool* frame_destroyed_indicator) : I420Buffer(width, height), diff --git a/video/end_to_end_tests/bandwidth_tests.cc b/video/end_to_end_tests/bandwidth_tests.cc index d327a7670d..2f6ec09f6d 100644 --- a/video/end_to_end_tests/bandwidth_tests.cc +++ b/video/end_to_end_tests/bandwidth_tests.cc @@ -327,7 +327,7 @@ TEST_F(BandwidthEndToEndTest, RembWithSendSideBwe) { TEST_F(BandwidthEndToEndTest, ReportsSetEncoderRates) { // If these fields trial are on, we get lower bitrates than expected by this // test, due to the packetization overhead and encoder pushback. - webrtc::test::ScopedFieldTrials field_trials( + test::ScopedFieldTrials field_trials( std::string(field_trial::GetFieldTrialString()) + "WebRTC-VideoRateControl/bitrate_adjuster:false/"); class EncoderRateStatsTest : public test::EndToEndTest, diff --git a/video/end_to_end_tests/resolution_bitrate_limits_tests.cc b/video/end_to_end_tests/resolution_bitrate_limits_tests.cc index a8503d9499..786535a4aa 100644 --- a/video/end_to_end_tests/resolution_bitrate_limits_tests.cc +++ b/video/end_to_end_tests/resolution_bitrate_limits_tests.cc @@ -228,7 +228,7 @@ class InitEncodeTest : public test::EndToEndTest, }; TEST_P(ResolutionBitrateLimitsTest, LimitsApplied) { - webrtc::test::ScopedFieldTrials field_trials( + test::ScopedFieldTrials field_trials( "WebRTC-GetEncoderInfoOverride/" "frame_size_pixels:921600," "min_start_bitrate_bps:0," @@ -272,7 +272,7 @@ TEST_F(ResolutionBitrateLimitsWithScalabilityModeTest, } TEST_F(ResolutionBitrateLimitsWithScalabilityModeTest, OneStreamLimitsAppliedForOneSpatialLayer) { - webrtc::test::ScopedFieldTrials field_trials( + test::ScopedFieldTrials field_trials( "WebRTC-GetEncoderInfoOverride/" "frame_size_pixels:921600," "min_start_bitrate_bps:0," @@ -291,7 +291,7 @@ TEST_F(ResolutionBitrateLimitsWithScalabilityModeTest, TEST_F(ResolutionBitrateLimitsWithScalabilityModeTest, OneStreamLimitsNotAppliedForMultipleSpatialLayers) { - webrtc::test::ScopedFieldTrials field_trials( + test::ScopedFieldTrials field_trials( "WebRTC-GetEncoderInfoOverride/" "frame_size_pixels:230400|921600," "min_start_bitrate_bps:0|0," @@ -324,7 +324,7 @@ TEST_P(ResolutionBitrateLimitsTest, EncodingsApplied) { } TEST_P(ResolutionBitrateLimitsTest, IntersectionApplied) { - webrtc::test::ScopedFieldTrials field_trials( + test::ScopedFieldTrials field_trials( "WebRTC-GetEncoderInfoOverride/" "frame_size_pixels:921600," "min_start_bitrate_bps:0," @@ -343,7 +343,7 @@ TEST_P(ResolutionBitrateLimitsTest, IntersectionApplied) { } TEST_P(ResolutionBitrateLimitsTest, LimitsAppliedMiddleActive) { - webrtc::test::ScopedFieldTrials field_trials( + test::ScopedFieldTrials field_trials( "WebRTC-GetEncoderInfoOverride/" "frame_size_pixels:230400|921600," "min_start_bitrate_bps:0|0," @@ -360,7 +360,7 @@ TEST_P(ResolutionBitrateLimitsTest, LimitsAppliedMiddleActive) { } TEST_P(ResolutionBitrateLimitsTest, EncodingMinMaxBitrateAppliedMiddleActive) { - webrtc::test::ScopedFieldTrials field_trials( + test::ScopedFieldTrials field_trials( "WebRTC-GetEncoderInfoOverride/" "frame_size_pixels:230400|921600," "min_start_bitrate_bps:0|0," @@ -383,7 +383,7 @@ TEST_P(ResolutionBitrateLimitsTest, EncodingMinMaxBitrateAppliedMiddleActive) { } TEST_P(ResolutionBitrateLimitsTest, MinBitrateNotAboveEncodingMax) { - webrtc::test::ScopedFieldTrials field_trials( + test::ScopedFieldTrials field_trials( "WebRTC-GetEncoderInfoOverride/" "frame_size_pixels:230400|921600," "min_start_bitrate_bps:0|0," @@ -404,7 +404,7 @@ TEST_P(ResolutionBitrateLimitsTest, MinBitrateNotAboveEncodingMax) { } TEST_P(ResolutionBitrateLimitsTest, MaxBitrateNotBelowEncodingMin) { - webrtc::test::ScopedFieldTrials field_trials( + test::ScopedFieldTrials field_trials( "WebRTC-GetEncoderInfoOverride/" "frame_size_pixels:230400|921600," "min_start_bitrate_bps:0|0," @@ -464,7 +464,7 @@ TEST_F(ResolutionBitrateLimitsWithScalabilityModeTest, } TEST_P(ResolutionBitrateLimitsTest, LimitsAppliedHighestActive) { - webrtc::test::ScopedFieldTrials field_trials( + test::ScopedFieldTrials field_trials( "WebRTC-GetEncoderInfoOverride/" "frame_size_pixels:230400|921600," "min_start_bitrate_bps:0|0," @@ -481,7 +481,7 @@ TEST_P(ResolutionBitrateLimitsTest, LimitsAppliedHighestActive) { } TEST_P(ResolutionBitrateLimitsTest, EncodingMinMaxBitrateAppliedHighestActive) { - webrtc::test::ScopedFieldTrials field_trials( + test::ScopedFieldTrials field_trials( "WebRTC-GetEncoderInfoOverride/" "frame_size_pixels:230400|921600," "min_start_bitrate_bps:0|0," @@ -506,7 +506,7 @@ TEST_P(ResolutionBitrateLimitsTest, EncodingMinMaxBitrateAppliedHighestActive) { } TEST_P(ResolutionBitrateLimitsTest, LimitsNotAppliedLowestActive) { - webrtc::test::ScopedFieldTrials field_trials( + test::ScopedFieldTrials field_trials( "WebRTC-GetEncoderInfoOverride/" "frame_size_pixels:230400|921600," "min_start_bitrate_bps:0|0," @@ -527,7 +527,7 @@ TEST_P(ResolutionBitrateLimitsTest, LimitsNotAppliedLowestActive) { TEST_F(ResolutionBitrateLimitsWithScalabilityModeTest, LimitsAppliedForVp9OneSpatialLayer) { - webrtc::test::ScopedFieldTrials field_trials( + test::ScopedFieldTrials field_trials( "WebRTC-GetEncoderInfoOverride/" "frame_size_pixels:230400|921600," "min_start_bitrate_bps:0|0," @@ -547,7 +547,7 @@ TEST_F(ResolutionBitrateLimitsWithScalabilityModeTest, TEST_F(ResolutionBitrateLimitsWithScalabilityModeTest, LimitsNotAppliedForVp9MultipleSpatialLayers) { - webrtc::test::ScopedFieldTrials field_trials( + test::ScopedFieldTrials field_trials( "WebRTC-GetEncoderInfoOverride/" "frame_size_pixels:230400|921600," "min_start_bitrate_bps:0|0," @@ -570,7 +570,7 @@ TEST_F(ResolutionBitrateLimitsWithScalabilityModeTest, TEST_F(ResolutionBitrateLimitsWithScalabilityModeTest, OneStreamLimitsAppliedForAv1OneSpatialLayer) { - webrtc::test::ScopedFieldTrials field_trials( + test::ScopedFieldTrials field_trials( "WebRTC-GetEncoderInfoOverride/" "frame_size_pixels:921600," "min_start_bitrate_bps:0," @@ -589,7 +589,7 @@ TEST_F(ResolutionBitrateLimitsWithScalabilityModeTest, TEST_F(ResolutionBitrateLimitsWithScalabilityModeTest, LimitsAppliedForAv1SingleSpatialLayer) { - webrtc::test::ScopedFieldTrials field_trials( + test::ScopedFieldTrials field_trials( "WebRTC-GetEncoderInfoOverride/" "frame_size_pixels:230400|921600," "min_start_bitrate_bps:0|0," @@ -609,7 +609,7 @@ TEST_F(ResolutionBitrateLimitsWithScalabilityModeTest, TEST_F(ResolutionBitrateLimitsWithScalabilityModeTest, EncodingMinMaxBitrateAppliedForAv1SingleSpatialLayer) { - webrtc::test::ScopedFieldTrials field_trials( + test::ScopedFieldTrials field_trials( "WebRTC-GetEncoderInfoOverride/" "frame_size_pixels:921600," "min_start_bitrate_bps:0," @@ -631,7 +631,7 @@ TEST_F(ResolutionBitrateLimitsWithScalabilityModeTest, TEST_F(ResolutionBitrateLimitsWithScalabilityModeTest, LimitsNotAppliedForAv1MultipleSpatialLayers) { - webrtc::test::ScopedFieldTrials field_trials( + test::ScopedFieldTrials field_trials( "WebRTC-GetEncoderInfoOverride/" "frame_size_pixels:230400|921600," "min_start_bitrate_bps:0|0," @@ -653,7 +653,7 @@ TEST_F(ResolutionBitrateLimitsWithScalabilityModeTest, } TEST_P(ResolutionBitrateLimitsTest, LimitsNotAppliedSimulcast) { - webrtc::test::ScopedFieldTrials field_trials( + test::ScopedFieldTrials field_trials( "WebRTC-GetEncoderInfoOverride/" "frame_size_pixels:230400|921600," "min_start_bitrate_bps:0|0," diff --git a/video/end_to_end_tests/rtp_rtcp_tests.cc b/video/end_to_end_tests/rtp_rtcp_tests.cc index 712d4a82bc..1e0f1d78bc 100644 --- a/video/end_to_end_tests/rtp_rtcp_tests.cc +++ b/video/end_to_end_tests/rtp_rtcp_tests.cc @@ -362,7 +362,7 @@ void RtpRtcpEndToEndTest::TestRtpStatePreservation( // to wait for a second for periodic Sender Report. rtcp::RapidResyncRequest force_send_sr_back_request; Buffer packet = force_send_sr_back_request.Build(); - static_cast(receive_transport_.get()) + static_cast(receive_transport_.get()) ->SendRtcp(packet, /*packet_options=*/{}); } CreateFrameGeneratorCapturer(30, 1280, 720); diff --git a/video/frame_cadence_adapter_unittest.cc b/video/frame_cadence_adapter_unittest.cc index b40171eaca..285b917eb7 100644 --- a/video/frame_cadence_adapter_unittest.cc +++ b/video/frame_cadence_adapter_unittest.cc @@ -1295,7 +1295,7 @@ TEST_F(ZeroHertzQueueOverloadTest, TEST_F(ZeroHertzQueueOverloadTest, QueueOverloadIsDisabledForZeroHerzWhenKillSwitchIsEnabled) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( field_trials_, "WebRTC-ZeroHertzQueueOverload/Disabled/"); adapter_.reset(); adapter_ = CreateAdapter(field_trials, time_controller_.GetClock()); diff --git a/video/frame_decode_timing.cc b/video/frame_decode_timing.cc index 04b8e22c5f..9871ba9784 100644 --- a/video/frame_decode_timing.cc +++ b/video/frame_decode_timing.cc @@ -22,8 +22,7 @@ namespace webrtc { -FrameDecodeTiming::FrameDecodeTiming(Clock* clock, - webrtc::VCMTiming const* timing) +FrameDecodeTiming::FrameDecodeTiming(Clock* clock, VCMTiming const* timing) : clock_(clock), timing_(timing) { RTC_DCHECK(clock_); RTC_DCHECK(timing_); diff --git a/video/frame_decode_timing_unittest.cc b/video/frame_decode_timing_unittest.cc index e4359b5c2a..da7a098ad0 100644 --- a/video/frame_decode_timing_unittest.cc +++ b/video/frame_decode_timing_unittest.cc @@ -35,10 +35,10 @@ using ::testing::Optional; namespace { -class FakeVCMTiming : public webrtc::VCMTiming { +class FakeVCMTiming : public VCMTiming { public: explicit FakeVCMTiming(Clock* clock, const FieldTrialsView& field_trials) - : webrtc::VCMTiming(clock, field_trials) {} + : VCMTiming(clock, field_trials) {} Timestamp RenderTime(uint32_t frame_timestamp, Timestamp now) const override { RTC_DCHECK(render_time_map_.contains(frame_timestamp)); diff --git a/video/frame_encode_metadata_writer_unittest.cc b/video/frame_encode_metadata_writer_unittest.cc index bb584b8dbd..4580caab93 100644 --- a/video/frame_encode_metadata_writer_unittest.cc +++ b/video/frame_encode_metadata_writer_unittest.cc @@ -386,8 +386,7 @@ TEST(FrameEncodeMetadataWriterTest, CopiesColorSpace) { bitrate_allocation.SetBitrate(0, 0, 500000); encode_timer.OnSetRates(bitrate_allocation, 30); - webrtc::ColorSpace color_space = - CreateTestColorSpace(/*with_hdr_metadata=*/true); + ColorSpace color_space = CreateTestColorSpace(/*with_hdr_metadata=*/true); image.SetRtpTimestamp(static_cast(kTimestampMs * 90)); VideoFrame frame = VideoFrame::Builder() .set_timestamp_ms(kTimestampMs) diff --git a/video/full_stack_tests.cc b/video/full_stack_tests.cc index 8245e6ff00..f8c809c970 100644 --- a/video/full_stack_tests.cc +++ b/video/full_stack_tests.cc @@ -978,7 +978,7 @@ TEST(FullStackTest, Screenshare_Slides_Vp9_3sl_High_Fps) { #if !defined(WEBRTC_MAC) TEST(FullStackTest, Vp9ksvc_3sl_High) { - webrtc::test::ScopedFieldTrials override_trials( + test::ScopedFieldTrials override_trials( AppendFieldTrials("WebRTC-Vp9IssueKeyFrameOnLayerDeactivation/Enabled/")); auto fixture = CreateVideoQualityTestFixture(); ParamsWithLogging simulcast; @@ -993,7 +993,7 @@ TEST(FullStackTest, Vp9ksvc_3sl_High) { } TEST(FullStackTest, Vp9ksvc_3sl_Low) { - webrtc::test::ScopedFieldTrials override_trials( + test::ScopedFieldTrials override_trials( AppendFieldTrials("WebRTC-Vp9IssueKeyFrameOnLayerDeactivation/Enabled/")); auto fixture = CreateVideoQualityTestFixture(); ParamsWithLogging simulcast; @@ -1008,7 +1008,7 @@ TEST(FullStackTest, Vp9ksvc_3sl_Low) { } TEST(FullStackTest, Vp9ksvc_3sl_Low_Bw_Limited) { - webrtc::test::ScopedFieldTrials override_trials( + test::ScopedFieldTrials override_trials( AppendFieldTrials("WebRTC-Vp9IssueKeyFrameOnLayerDeactivation/Enabled/")); auto fixture = CreateVideoQualityTestFixture(); ParamsWithLogging simulcast; @@ -1024,7 +1024,7 @@ TEST(FullStackTest, Vp9ksvc_3sl_Low_Bw_Limited) { } TEST(FullStackTest, Vp9ksvc_3sl_Medium_Network_Restricted) { - webrtc::test::ScopedFieldTrials override_trials( + test::ScopedFieldTrials override_trials( AppendFieldTrials("WebRTC-Vp9IssueKeyFrameOnLayerDeactivation/Enabled/")); auto fixture = CreateVideoQualityTestFixture(); ParamsWithLogging simulcast; @@ -1042,7 +1042,7 @@ TEST(FullStackTest, Vp9ksvc_3sl_Medium_Network_Restricted) { // TODO(webrtc:9722): Remove when experiment is cleaned up. TEST(FullStackTest, Vp9ksvc_3sl_Medium_Network_Restricted_Trusted_Rate) { - webrtc::test::ScopedFieldTrials override_trials( + test::ScopedFieldTrials override_trials( AppendFieldTrials("WebRTC-Vp9IssueKeyFrameOnLayerDeactivation/Enabled/")); auto fixture = CreateVideoQualityTestFixture(); ParamsWithLogging simulcast; @@ -1087,7 +1087,7 @@ TEST(FullStackTest, MAYBE_Simulcast_HD_High) { simulcast.ss[0] = { streams, 2, 1, 0, InterLayerPredMode::kOn, std::vector(), true}; - webrtc::test::ScopedFieldTrials override_trials(AppendFieldTrials( + test::ScopedFieldTrials override_trials(AppendFieldTrials( "WebRTC-ForceSimulatedOveruseIntervalMs/1000-50000-300/")); fixture->RunWithAnalyzer(simulcast); } diff --git a/video/pc_full_stack_tests.cc b/video/pc_full_stack_tests.cc index 841f961ff0..f0c4c0ebf7 100644 --- a/video/pc_full_stack_tests.cc +++ b/video/pc_full_stack_tests.cc @@ -39,14 +39,14 @@ namespace webrtc { -using ::webrtc::webrtc_pc_e2e::AudioConfig; -using ::webrtc::webrtc_pc_e2e::EmulatedSFUConfig; -using ::webrtc::webrtc_pc_e2e::PeerConfigurer; -using ::webrtc::webrtc_pc_e2e::RunParams; -using ::webrtc::webrtc_pc_e2e::ScreenShareConfig; -using ::webrtc::webrtc_pc_e2e::VideoCodecConfig; -using ::webrtc::webrtc_pc_e2e::VideoConfig; -using ::webrtc::webrtc_pc_e2e::VideoSimulcastConfig; +using webrtc_pc_e2e::AudioConfig; +using webrtc_pc_e2e::EmulatedSFUConfig; +using webrtc_pc_e2e::PeerConfigurer; +using webrtc_pc_e2e::RunParams; +using webrtc_pc_e2e::ScreenShareConfig; +using webrtc_pc_e2e::VideoCodecConfig; +using webrtc_pc_e2e::VideoConfig; +using webrtc_pc_e2e::VideoSimulcastConfig; namespace { @@ -1235,7 +1235,7 @@ ParamsWithLogging::Video SimulcastVp8VideoLow() { #if defined(RTC_ENABLE_VP9) TEST(PCFullStackTest, Pc_Screenshare_Slides_Vp9_3sl_High_Fps) { - webrtc::test::ScopedFieldTrials override_trials( + test::ScopedFieldTrials override_trials( AppendFieldTrials("WebRTC-Vp9InterLayerPred/" "Enabled,inter_layer_pred_mode:on/")); std::unique_ptr network_emulation_manager = @@ -1269,7 +1269,7 @@ TEST(PCFullStackTest, Pc_Screenshare_Slides_Vp9_3sl_High_Fps) { } TEST(PCFullStackTest, Pc_Vp9svc_3sl_High) { - webrtc::test::ScopedFieldTrials override_trials( + test::ScopedFieldTrials override_trials( AppendFieldTrials("WebRTC-Vp9InterLayerPred/" "Enabled,inter_layer_pred_mode:on/")); std::unique_ptr network_emulation_manager = @@ -1302,7 +1302,7 @@ TEST(PCFullStackTest, Pc_Vp9svc_3sl_High) { } TEST(PCFullStackTest, Pc_Vp9svc_3sl_Low) { - webrtc::test::ScopedFieldTrials override_trials( + test::ScopedFieldTrials override_trials( AppendFieldTrials("WebRTC-Vp9InterLayerPred/" "Enabled,inter_layer_pred_mode:on/")); std::unique_ptr network_emulation_manager = @@ -1434,7 +1434,7 @@ TEST(PCFullStackTest, VP9KSVC_3SL_Medium_Network_Restricted_Trusted_Rate) { #define MAYBE_Pc_Simulcast_HD_High Pc_Simulcast_HD_High #endif TEST(PCFullStackTest, MAYBE_Pc_Simulcast_HD_High) { - webrtc::test::ScopedFieldTrials override_trials(AppendFieldTrials( + test::ScopedFieldTrials override_trials(AppendFieldTrials( "WebRTC-ForceSimulatedOveruseIntervalMs/1000-50000-300/")); std::unique_ptr network_emulation_manager = CreateNetworkEmulationManager(); diff --git a/video/receive_statistics_proxy.cc b/video/receive_statistics_proxy.cc index 22d46e4cc9..b89ae42538 100644 --- a/video/receive_statistics_proxy.cc +++ b/video/receive_statistics_proxy.cc @@ -614,7 +614,7 @@ void ReceiveStatisticsProxy::OnDecodedFrame(const VideoFrame& frame, VideoContentType content_type, VideoFrameType frame_type) { TimeDelta processing_delay = TimeDelta::Zero(); - webrtc::Timestamp current_time = clock_->CurrentTime(); + Timestamp current_time = clock_->CurrentTime(); // TODO(bugs.webrtc.org/13984): some tests do not fill packet_infos(). TimeDelta assembly_time = TimeDelta::Zero(); if (frame.packet_infos().size() > 0) { diff --git a/video/receive_statistics_proxy_unittest.cc b/video/receive_statistics_proxy_unittest.cc index a198cd6196..5e06231a38 100644 --- a/video/receive_statistics_proxy_unittest.cc +++ b/video/receive_statistics_proxy_unittest.cc @@ -128,7 +128,7 @@ class ReceiveStatisticsProxyTest : public ::testing::Test { TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameIncreasesFramesDecoded) { EXPECT_EQ(0u, statistics_proxy_->GetStats().frames_decoded); - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); for (uint32_t i = 1; i <= 3; ++i) { statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(), VideoContentType::UNSPECIFIED, @@ -141,7 +141,7 @@ TEST_F(ReceiveStatisticsProxyTest, DecodedFpsIsReported) { const Frequency kFps = Frequency::Hertz(20); const int kRequiredSamples = TimeDelta::Seconds(metrics::kMinRunTimeInSeconds) * kFps; - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); for (int i = 0; i < kRequiredSamples; ++i) { statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(), VideoContentType::UNSPECIFIED, @@ -159,7 +159,7 @@ TEST_F(ReceiveStatisticsProxyTest, DecodedFpsIsNotReportedForTooFewSamples) { const Frequency kFps = Frequency::Hertz(20); const int kRequiredSamples = TimeDelta::Seconds(metrics::kMinRunTimeInSeconds) * kFps; - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); for (int i = 0; i < kRequiredSamples - 1; ++i) { statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(), VideoContentType::UNSPECIFIED, @@ -174,7 +174,7 @@ TEST_F(ReceiveStatisticsProxyTest, DecodedFpsIsNotReportedForTooFewSamples) { TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameWithQpDoesNotResetFramesDecodedOrTotalDecodeTime) { EXPECT_EQ(0u, statistics_proxy_->GetStats().frames_decoded); - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); TimeDelta expected_total_decode_time = TimeDelta::Zero(); unsigned int expected_frames_decoded = 0; for (uint32_t i = 1; i <= 3; ++i) { @@ -204,7 +204,7 @@ TEST_F(ReceiveStatisticsProxyTest, TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameIncreasesProcessingDelay) { const TimeDelta kProcessingDelay = TimeDelta::Millis(10); EXPECT_EQ(0u, statistics_proxy_->GetStats().frames_decoded); - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); TimeDelta expected_total_processing_delay = TimeDelta::Zero(); unsigned int expected_frames_decoded = 0; // We set receive time fixed and increase the clock by 10ms @@ -242,7 +242,7 @@ TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameIncreasesProcessingDelay) { TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameIncreasesAssemblyTime) { const TimeDelta kAssemblyTime = TimeDelta::Millis(7); EXPECT_EQ(0u, statistics_proxy_->GetStats().frames_decoded); - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); TimeDelta expected_total_assembly_time = TimeDelta::Zero(); unsigned int expected_frames_decoded = 0; unsigned int expected_frames_assembled_from_multiple_packets = 0; @@ -316,7 +316,7 @@ TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameIncreasesAssemblyTime) { TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameIncreasesQpSum) { EXPECT_EQ(std::nullopt, statistics_proxy_->GetStats().qp_sum); - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); statistics_proxy_->OnDecodedFrame(frame, 3u, TimeDelta::Zero(), VideoContentType::UNSPECIFIED, VideoFrameType::kVideoFrameKey); @@ -329,7 +329,7 @@ TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameIncreasesQpSum) { TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameIncreasesTotalDecodeTime) { EXPECT_EQ(std::nullopt, statistics_proxy_->GetStats().qp_sum); - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); statistics_proxy_->OnDecodedFrame(frame, 3u, TimeDelta::Millis(4), VideoContentType::UNSPECIFIED, VideoFrameType::kVideoFrameKey); @@ -343,7 +343,7 @@ TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameIncreasesTotalDecodeTime) { TEST_F(ReceiveStatisticsProxyTest, ReportsContentType) { const std::string kRealtimeString("realtime"); const std::string kScreenshareString("screen"); - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); EXPECT_EQ(kRealtimeString, videocontenttypehelpers::ToString( statistics_proxy_->GetStats().content_type)); statistics_proxy_->OnDecodedFrame(frame, 3u, TimeDelta::Zero(), @@ -359,7 +359,7 @@ TEST_F(ReceiveStatisticsProxyTest, ReportsContentType) { } TEST_F(ReceiveStatisticsProxyTest, ReportsMaxInterframeDelay) { - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); const TimeDelta kInterframeDelay1 = TimeDelta::Millis(100); const TimeDelta kInterframeDelay2 = TimeDelta::Millis(200); const TimeDelta kInterframeDelay3 = TimeDelta::Millis(100); @@ -390,7 +390,7 @@ TEST_F(ReceiveStatisticsProxyTest, ReportsMaxInterframeDelay) { } TEST_F(ReceiveStatisticsProxyTest, ReportInterframeDelayInWindow) { - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); const TimeDelta kInterframeDelay1 = TimeDelta::Millis(900); const TimeDelta kInterframeDelay2 = TimeDelta::Millis(750); const TimeDelta kInterframeDelay3 = TimeDelta::Millis(700); @@ -428,7 +428,7 @@ TEST_F(ReceiveStatisticsProxyTest, ReportsFreezeMetrics) { EXPECT_EQ(0u, stats.freeze_count); EXPECT_FALSE(stats.total_freezes_duration_ms); - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); for (size_t i = 0; i < VideoQualityObserver::kMinFrameSamplesToDetectFreeze; ++i) { time_controller_.AdvanceTime(TimeDelta::Millis(30)); @@ -449,7 +449,7 @@ TEST_F(ReceiveStatisticsProxyTest, ReportsPauseMetrics) { ASSERT_EQ(0u, stats.pause_count); ASSERT_EQ(0u, stats.total_pauses_duration_ms); - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); statistics_proxy_->OnRenderedFrame(MetaData(frame)); // Pause. @@ -467,7 +467,7 @@ TEST_F(ReceiveStatisticsProxyTest, PauseBeforeFirstAndAfterLastFrameIgnored) { ASSERT_EQ(0u, stats.pause_count); ASSERT_EQ(0u, stats.total_pauses_duration_ms); - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); // Pause -> Frame -> Pause time_controller_.AdvanceTime(TimeDelta::Seconds(5)); @@ -489,7 +489,7 @@ TEST_F(ReceiveStatisticsProxyTest, ReportsTotalInterFrameDelay) { VideoReceiveStreamInterface::Stats stats = statistics_proxy_->GetStats(); ASSERT_EQ(0.0, stats.total_inter_frame_delay); - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); // Emulate delay before first frame is rendered. This is needed to ensure // that frame duration only covers time since first frame is rendered and @@ -508,7 +508,7 @@ TEST_F(ReceiveStatisticsProxyTest, ReportsTotalSquaredInterFrameDelay) { VideoReceiveStreamInterface::Stats stats = statistics_proxy_->GetStats(); ASSERT_EQ(0.0, stats.total_squared_inter_frame_delay); - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); for (int i = 0; i <= 10; ++i) { time_controller_.AdvanceTime(TimeDelta::Millis(30)); statistics_proxy_->OnRenderedFrame(MetaData(frame)); @@ -522,7 +522,7 @@ TEST_F(ReceiveStatisticsProxyTest, ReportsTotalSquaredInterFrameDelay) { } TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameWithoutQpQpSumWontExist) { - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); EXPECT_EQ(std::nullopt, statistics_proxy_->GetStats().qp_sum); statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(), VideoContentType::UNSPECIFIED, @@ -531,7 +531,7 @@ TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameWithoutQpQpSumWontExist) { } TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameWithoutQpResetsQpSum) { - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); EXPECT_EQ(std::nullopt, statistics_proxy_->GetStats().qp_sum); statistics_proxy_->OnDecodedFrame(frame, 3u, TimeDelta::Zero(), VideoContentType::UNSPECIFIED, @@ -545,7 +545,7 @@ TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameWithoutQpResetsQpSum) { TEST_F(ReceiveStatisticsProxyTest, OnRenderedFrameIncreasesFramesRendered) { EXPECT_EQ(0u, statistics_proxy_->GetStats().frames_rendered); - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); for (uint32_t i = 1; i <= 3; ++i) { statistics_proxy_->OnRenderedFrame(MetaData(frame)); EXPECT_EQ(i, statistics_proxy_->GetStats().frames_rendered); @@ -695,7 +695,7 @@ TEST_F(ReceiveStatisticsProxyTest, TEST_F(ReceiveStatisticsProxyTest, GetStatsReportsFrameCounts) { const int kKeyFrames = 3; const int kDeltaFrames = 22; - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); for (int i = 0; i < kKeyFrames; i++) { statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(), VideoContentType::UNSPECIFIED, @@ -774,7 +774,7 @@ TEST_F(ReceiveStatisticsProxyTest, LifetimeHistogramIsUpdated) { time_controller_.AdvanceTime(kLifetime); // Need at least one decoded frame to report stream lifetime. - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); statistics_proxy_->OnCompleteFrame(true, 1000, VideoContentType::UNSPECIFIED); statistics_proxy_->OnDecodedFrame( frame, std::nullopt, TimeDelta::Millis(1000), @@ -920,7 +920,7 @@ TEST_F(ReceiveStatisticsProxyTest, KeyFrameHistogramNotUpdatedForTooFewSamples) { const bool kIsKeyFrame = false; const int kFrameSizeBytes = 1000; - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); for (int i = 0; i < kMinRequiredSamples - 1; ++i) { statistics_proxy_->OnCompleteFrame(kIsKeyFrame, kFrameSizeBytes, @@ -945,7 +945,7 @@ TEST_F(ReceiveStatisticsProxyTest, KeyFrameHistogramUpdatedForMinRequiredSamples) { const bool kIsKeyFrame = false; const int kFrameSizeBytes = 1000; - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); for (int i = 0; i < kMinRequiredSamples; ++i) { statistics_proxy_->OnCompleteFrame(kIsKeyFrame, kFrameSizeBytes, @@ -970,7 +970,7 @@ TEST_F(ReceiveStatisticsProxyTest, TEST_F(ReceiveStatisticsProxyTest, KeyFrameHistogramIsUpdated) { const int kFrameSizeBytes = 1000; - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); for (int i = 0; i < kMinRequiredSamples; ++i) { statistics_proxy_->OnCompleteFrame(true, kFrameSizeBytes, @@ -1058,7 +1058,7 @@ TEST_F(ReceiveStatisticsProxyTest, TimingHistogramsAreUpdated) { TEST_F(ReceiveStatisticsProxyTest, DoesNotReportStaleFramerates) { const Frequency kDefaultFps = Frequency::Hertz(30); - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); for (int i = 0; i < kDefaultFps.hertz(); ++i) { // Since OnRenderedFrame is never called the fps in each sample will be 0, @@ -1137,7 +1137,7 @@ TEST_F(ReceiveStatisticsProxyTest, ReceivedFrameHistogramsAreUpdated) { } TEST_F(ReceiveStatisticsProxyTest, ZeroDelayReportedIfFrameNotDelayed) { - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(), VideoContentType::UNSPECIFIED, VideoFrameType::kVideoFrameKey); @@ -1160,7 +1160,7 @@ TEST_F(ReceiveStatisticsProxyTest, ZeroDelayReportedIfFrameNotDelayed) { TEST_F(ReceiveStatisticsProxyTest, DelayedFrameHistogramsAreNotUpdatedIfMinRuntimeHasNotPassed) { - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(), VideoContentType::UNSPECIFIED, VideoFrameType::kVideoFrameKey); @@ -1182,7 +1182,7 @@ TEST_F(ReceiveStatisticsProxyTest, TEST_F(ReceiveStatisticsProxyTest, DelayedFramesHistogramsAreNotUpdatedIfNoRenderedFrames) { - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(), VideoContentType::UNSPECIFIED, VideoFrameType::kVideoFrameKey); @@ -1199,7 +1199,7 @@ TEST_F(ReceiveStatisticsProxyTest, } TEST_F(ReceiveStatisticsProxyTest, DelayReportedIfFrameIsDelayed) { - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(), VideoContentType::UNSPECIFIED, VideoFrameType::kVideoFrameKey); @@ -1224,7 +1224,7 @@ TEST_F(ReceiveStatisticsProxyTest, DelayReportedIfFrameIsDelayed) { } TEST_F(ReceiveStatisticsProxyTest, AverageDelayOfDelayedFramesIsReported) { - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(), VideoContentType::UNSPECIFIED, VideoFrameType::kVideoFrameKey); @@ -1342,7 +1342,7 @@ INSTANTIATE_TEST_SUITE_P(_, TEST_P(ReceiveStatisticsProxyTestWithFreezeDuration, FreezeDetection) { VideoReceiveStreamInterface::Stats stats = statistics_proxy_->GetStats(); EXPECT_EQ(0u, stats.freeze_count); - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); // Add a very long frame. This is need to verify that average frame // duration, which is supposed to be calculated as mean of durations of @@ -1364,9 +1364,9 @@ TEST_P(ReceiveStatisticsProxyTestWithFreezeDuration, FreezeDetection) { class ReceiveStatisticsProxyTestWithContent : public ReceiveStatisticsProxyTest, - public ::testing::WithParamInterface { + public ::testing::WithParamInterface { protected: - const webrtc::VideoContentType content_type_{GetParam()}; + const VideoContentType content_type_{GetParam()}; }; INSTANTIATE_TEST_SUITE_P(ContentTypes, @@ -1376,7 +1376,7 @@ INSTANTIATE_TEST_SUITE_P(ContentTypes, TEST_P(ReceiveStatisticsProxyTestWithContent, InterFrameDelaysAreReported) { const TimeDelta kInterFrameDelay = TimeDelta::Millis(33); - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); for (int i = 0; i < kMinRequiredSamples; ++i) { statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(), @@ -1413,7 +1413,7 @@ TEST_P(ReceiveStatisticsProxyTestWithContent, InterFrameDelaysPercentilesAreReported) { const TimeDelta kInterFrameDelay = TimeDelta::Millis(33); const int kLastFivePercentsSamples = kMinRequiredSamples * 5 / 100; - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); for (int i = 0; i <= kMinRequiredSamples - kLastFivePercentsSamples; ++i) { time_controller_.AdvanceTime(kInterFrameDelay); @@ -1451,7 +1451,7 @@ TEST_P(ReceiveStatisticsProxyTestWithContent, TEST_P(ReceiveStatisticsProxyTestWithContent, MaxInterFrameDelayOnlyWithValidAverage) { const TimeDelta kInterFrameDelay = TimeDelta::Millis(33); - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); for (int i = 0; i < kMinRequiredSamples; ++i) { statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(), @@ -1475,7 +1475,7 @@ TEST_P(ReceiveStatisticsProxyTestWithContent, TEST_P(ReceiveStatisticsProxyTestWithContent, MaxInterFrameDelayOnlyWithPause) { const TimeDelta kInterFrameDelay = TimeDelta::Millis(33); - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); for (int i = 0; i <= kMinRequiredSamples; ++i) { statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(), @@ -1563,7 +1563,7 @@ TEST_P(ReceiveStatisticsProxyTestWithContent, FreezesAreReported) { const TimeDelta kFreezeDelay = TimeDelta::Millis(200); const TimeDelta kCallDuration = kMinRequiredSamples * kInterFrameDelay + kFreezeDelay; - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); for (int i = 0; i < kMinRequiredSamples; ++i) { VideoFrameMetaData meta = MetaData(frame); @@ -1612,7 +1612,7 @@ TEST_P(ReceiveStatisticsProxyTestWithContent, HarmonicFrameRateIsReported) { const TimeDelta kPauseDuration = TimeDelta::Seconds(10); const TimeDelta kCallDuration = kMinRequiredSamples * kFrameDuration + kFreezeDuration + kPauseDuration; - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); for (int i = 0; i < kMinRequiredSamples; ++i) { time_controller_.AdvanceTime(kFrameDuration); @@ -1661,7 +1661,7 @@ TEST_P(ReceiveStatisticsProxyTestWithContent, HarmonicFrameRateIsReported) { TEST_P(ReceiveStatisticsProxyTestWithContent, PausesAreIgnored) { const TimeDelta kInterFrameDelay = TimeDelta::Millis(33); const TimeDelta kPauseDuration = TimeDelta::Seconds(10); - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); for (int i = 0; i <= kMinRequiredSamples; ++i) { VideoFrameMetaData meta = MetaData(frame); @@ -1706,7 +1706,7 @@ TEST_P(ReceiveStatisticsProxyTestWithContent, PausesAreIgnored) { TEST_P(ReceiveStatisticsProxyTestWithContent, ManyPausesAtTheBeginning) { const TimeDelta kInterFrameDelay = TimeDelta::Millis(33); const TimeDelta kPauseDuration = TimeDelta::Seconds(10); - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); for (int i = 0; i <= kMinRequiredSamples; ++i) { statistics_proxy_->OnDecodedFrame(frame, std::nullopt, TimeDelta::Zero(), @@ -1736,8 +1736,8 @@ TEST_P(ReceiveStatisticsProxyTestWithContent, ManyPausesAtTheBeginning) { TEST_P(ReceiveStatisticsProxyTestWithContent, TimeInHdReported) { const TimeDelta kInterFrameDelay = TimeDelta::Millis(20); - webrtc::VideoFrame frame_hd = CreateFrame(1280, 720); - webrtc::VideoFrame frame_sd = CreateFrame(640, 360); + VideoFrame frame_hd = CreateFrame(1280, 720); + VideoFrame frame_sd = CreateFrame(640, 360); // HD frames. for (int i = 0; i < kMinRequiredSamples; ++i) { @@ -1777,7 +1777,7 @@ TEST_P(ReceiveStatisticsProxyTestWithContent, TimeInBlockyVideoReported) { const TimeDelta kInterFrameDelay = TimeDelta::Millis(20); const int kHighQp = 80; const int kLowQp = 30; - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); // High quality frames. for (int i = 0; i < kMinRequiredSamples; ++i) { @@ -1821,9 +1821,9 @@ TEST_P(ReceiveStatisticsProxyTestWithContent, DownscalesReported) { // To ensure long enough call duration. const TimeDelta kInterFrameDelay = TimeDelta::Seconds(2); - webrtc::VideoFrame frame_hd = CreateFrame(1280, 720); - webrtc::VideoFrame frame_sd = CreateFrame(640, 360); - webrtc::VideoFrame frame_ld = CreateFrame(320, 180); + VideoFrame frame_hd = CreateFrame(1280, 720); + VideoFrame frame_sd = CreateFrame(640, 360); + VideoFrame frame_ld = CreateFrame(320, 180); // Call once to pass content type. statistics_proxy_->OnDecodedFrame(frame_hd, std::nullopt, TimeDelta::Zero(), @@ -1854,7 +1854,7 @@ TEST_P(ReceiveStatisticsProxyTestWithContent, DecodeTimeReported) { const int kLowQp = 30; const TimeDelta kDecodeTime = TimeDelta::Millis(7); - webrtc::VideoFrame frame = CreateFrame(kWidth, kHeight); + VideoFrame frame = CreateFrame(kWidth, kHeight); for (int i = 0; i < kMinRequiredSamples; ++i) { statistics_proxy_->OnDecodedFrame(frame, kLowQp, kDecodeTime, content_type_, diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc index bdd2cc22c6..3e41758685 100644 --- a/video/rtp_video_stream_receiver2.cc +++ b/video/rtp_video_stream_receiver2.cc @@ -393,7 +393,7 @@ RtpVideoStreamReceiver2::~RtpVideoStreamReceiver2() { void RtpVideoStreamReceiver2::AddReceiveCodec( uint8_t payload_type, VideoCodecType video_codec, - const webrtc::CodecParameterMap& codec_params, + const CodecParameterMap& codec_params, bool raw_payload) { RTC_DCHECK_RUN_ON(&packet_sequence_checker_); if (codec_params.count(kH264FmtpSpsPpsIdrInKeyframe) > 0 || diff --git a/video/rtp_video_stream_receiver2_unittest.cc b/video/rtp_video_stream_receiver2_unittest.cc index 6a8f96368c..344060156a 100644 --- a/video/rtp_video_stream_receiver2_unittest.cc +++ b/video/rtp_video_stream_receiver2_unittest.cc @@ -348,7 +348,7 @@ TEST_F(RtpVideoStreamReceiver2Test, CacheColorSpaceFromLastPacketOfKeyframe) { received_packet_generator.SetColorSpace(kColorSpace); // Prepare the receiver for VP9. - webrtc::CodecParameterMap codec_params; + CodecParameterMap codec_params; rtp_video_stream_receiver_->AddReceiveCodec(kVp9PayloadType, kVideoCodecVP9, codec_params, /*raw_payload=*/false); @@ -476,7 +476,7 @@ TEST_F(RtpVideoStreamReceiver2Test, const std::vector kDeltaFramePayload = {5, 6, 7, 8, 9}; // Prepare the receiver for VP9. - webrtc::CodecParameterMap codec_params; + CodecParameterMap codec_params; rtp_video_stream_receiver_->AddReceiveCodec(kVp9PayloadType, kVideoCodecVP9, codec_params, /*raw_payload=*/false); @@ -553,7 +553,7 @@ TEST_F(RtpVideoStreamReceiver2Test, const std::vector kDeltaFramePayload = {5, 6, 7, 8, 9}; // Prepare the receiver for VP9. - webrtc::CodecParameterMap codec_params; + CodecParameterMap codec_params; rtp_video_stream_receiver_->AddReceiveCodec(kVp9PayloadType, kVideoCodecVP9, codec_params, /*raw_payload=*/false); @@ -799,7 +799,7 @@ INSTANTIATE_TEST_SUITE_P(SpsPpsIdrIsKeyframeAndH26xPacketBuffer, TEST_P(RtpVideoStreamReceiver2TestH264, InBandSpsPps) { constexpr int kH264PayloadType = 98; - webrtc::CodecParameterMap codec_params; + CodecParameterMap codec_params; rtp_video_stream_receiver_->AddReceiveCodec(kH264PayloadType, kVideoCodecH264, codec_params, /*raw_payload=*/false); @@ -856,7 +856,7 @@ TEST_P(RtpVideoStreamReceiver2TestH264, InBandSpsPps) { TEST_P(RtpVideoStreamReceiver2TestH264, OutOfBandFmtpSpsPps) { constexpr int kH264PayloadType = 99; - webrtc::CodecParameterMap codec_params; + CodecParameterMap codec_params; // Example parameter sets from https://tools.ietf.org/html/rfc3984#section-8.2 // . codec_params.insert({kH264FmtpSpropParameterSets, "Z0IACpZTBYmI,aMljiA=="}); @@ -903,7 +903,7 @@ TEST_P(RtpVideoStreamReceiver2TestH264, OutOfBandFmtpSpsPps) { } TEST_P(RtpVideoStreamReceiver2TestH264, ForceSpsPpsIdrIsKeyframe) { - webrtc::CodecParameterMap codec_params; + CodecParameterMap codec_params; // Forcing can be done either with field trial or codec_params. if (!env_.field_trials().IsEnabled("WebRTC-SpsPpsIdrIsH264Keyframe")) { codec_params.insert({kH264FmtpSpsPpsIdrInKeyframe, ""}); @@ -1063,7 +1063,7 @@ TEST_P(RtpVideoStreamReceiver2TestPadding, EmptyPaddingInMediaStream) { rtp_packet_slice.Parse(raw_rtp_slice.data(), raw_rtp_slice.size()); // Prepare the receiver for H264. - webrtc::CodecParameterMap codec_params; + CodecParameterMap codec_params; rtp_video_stream_receiver_->AddReceiveCodec(kH264PayloadType, kVideoCodecH264, codec_params, false); rtp_video_stream_receiver_->StartReceive(); @@ -1770,7 +1770,7 @@ class RtpVideoStreamReceiver2TestH265 : public RtpVideoStreamReceiver2Test { TEST_F(RtpVideoStreamReceiver2TestH265, H265Bitstream) { constexpr int kH265PayloadType = 98; - webrtc::CodecParameterMap codec_params; + CodecParameterMap codec_params; rtp_video_stream_receiver_->AddReceiveCodec(kH265PayloadType, kVideoCodecH265, codec_params, /*raw_payload=*/false); diff --git a/video/send_statistics_proxy.cc b/video/send_statistics_proxy.cc index 38873b4fa4..e6858dfef9 100644 --- a/video/send_statistics_proxy.cc +++ b/video/send_statistics_proxy.cc @@ -147,7 +147,7 @@ std::optional GetFallbackMaxPixels(const std::string& group) { } std::optional GetFallbackMaxPixelsIfFieldTrialEnabled( - const webrtc::FieldTrialsView& field_trials) { + const FieldTrialsView& field_trials) { std::string group = field_trials.Lookup(kVp8ForcedFallbackEncoderFieldTrial); return (absl::StartsWith(group, "Enabled")) ? GetFallbackMaxPixels(group.substr(7)) @@ -155,7 +155,7 @@ std::optional GetFallbackMaxPixelsIfFieldTrialEnabled( } std::optional GetFallbackMaxPixelsIfFieldTrialDisabled( - const webrtc::FieldTrialsView& field_trials) { + const FieldTrialsView& field_trials) { std::string group = field_trials.Lookup(kVp8ForcedFallbackEncoderFieldTrial); return (absl::StartsWith(group, "Disabled")) ? GetFallbackMaxPixels(group.substr(8)) diff --git a/video/video_analyzer.cc b/video/video_analyzer.cc index 94f06ee465..159bd8ca2e 100644 --- a/video/video_analyzer.cc +++ b/video/video_analyzer.cc @@ -78,10 +78,10 @@ ABSL_FLAG(bool, namespace webrtc { namespace { -using ::webrtc::test::GetGlobalMetricsLogger; -using ::webrtc::test::ImprovementDirection; -using ::webrtc::test::Metric; -using ::webrtc::test::Unit; +using test::GetGlobalMetricsLogger; +using test::ImprovementDirection; +using test::Metric; +using test::Unit; constexpr TimeDelta kSendStatsPollingInterval = TimeDelta::Seconds(1); constexpr size_t kMaxComparisons = 10; diff --git a/video/video_loopback.cc b/video/video_loopback.cc index b8659d13cf..f2ab82509a 100644 --- a/video/video_loopback.cc +++ b/video/video_loopback.cc @@ -442,9 +442,9 @@ int RunLoopbackTest(int argc, char* argv[]) { // InitFieldTrialsFromString stores the char*, so the char array must outlive // the application. const std::string field_trials = absl::GetFlag(FLAGS_force_fieldtrials); - webrtc::field_trial::InitFieldTrialsFromString(field_trials.c_str()); + field_trial::InitFieldTrialsFromString(field_trials.c_str()); - webrtc::test::RunTest(webrtc::Loopback); + test::RunTest(Loopback); return 0; } } // namespace webrtc diff --git a/video/video_quality_test.cc b/video/video_quality_test.cc index 66a8a25f39..d4e1754f04 100644 --- a/video/video_quality_test.cc +++ b/video/video_quality_test.cc @@ -340,7 +340,7 @@ std::unique_ptr VideoQualityTest::CreateVideoDecoder( const SdpVideoFormat& format) { std::unique_ptr decoder; if (format.name == "FakeCodec") { - decoder = webrtc::FakeVideoDecoderFactory::CreateVideoDecoder(); + decoder = FakeVideoDecoderFactory::CreateVideoDecoder(); } else { decoder = decoder_factory_->Create(env, format); } @@ -644,14 +644,14 @@ void VideoQualityTest::FillScalabilitySettings( const std::vector& sl_descriptors) { if (params->ss[video_idx].streams.empty() && params->ss[video_idx].infer_streams) { - webrtc::VideoEncoder::EncoderInfo encoder_info; - webrtc::VideoEncoderConfig encoder_config; + VideoEncoder::EncoderInfo encoder_info; + VideoEncoderConfig encoder_config; encoder_config.codec_type = PayloadStringToCodecType(params->video[video_idx].codec); encoder_config.content_type = params->screenshare[video_idx].enabled - ? webrtc::VideoEncoderConfig::ContentType::kScreen - : webrtc::VideoEncoderConfig::ContentType::kRealtimeVideo; + ? VideoEncoderConfig::ContentType::kScreen + : VideoEncoderConfig::ContentType::kRealtimeVideo; encoder_config.max_bitrate_bps = params->video[video_idx].max_bitrate_bps; encoder_config.min_transmit_bitrate_bps = params->video[video_idx].min_transmit_bps; @@ -737,7 +737,7 @@ void VideoQualityTest::SetupVideo(Transport* send_transport, RTC_CHECK(num_video_streams_ > 0); video_encoder_configs_.resize(num_video_streams_); std::string generic_codec_name; - webrtc::VideoEncoder::EncoderInfo encoder_info; + VideoEncoder::EncoderInfo encoder_info; for (size_t video_idx = 0; video_idx < num_video_streams_; ++video_idx) { VideoSendStream::Config config(send_transport); config.rtp.extmap_allow_mixed = true; @@ -1423,7 +1423,7 @@ void VideoQualityTest::InitializeAudioDevice(CallConfig* send_call_config, // The real ADM requires extra initialization: setting default devices, // setting up number of channels etc. Helper class also calls // AudioDeviceModule::Init(). - webrtc::adm_helpers::Init(audio_device.get()); + adm_helpers::Init(audio_device.get()); } else { audio_device->Init(); } @@ -1448,8 +1448,8 @@ void VideoQualityTest::SetupAudio(Transport* transport) { if (params_.call.send_side_bwe) { audio_send_config.rtp.extensions.push_back( - webrtc::RtpExtension(webrtc::RtpExtension::kTransportSequenceNumberUri, - kTransportSequenceNumberExtensionId)); + RtpExtension(RtpExtension::kTransportSequenceNumberUri, + kTransportSequenceNumberExtensionId)); audio_send_config.min_bitrate_bps = kOpusMinBitrateBps; audio_send_config.max_bitrate_bps = kOpusBitrateFbBps; // Only allow ANA when send-side BWE is enabled. diff --git a/video/video_receive_stream2.cc b/video/video_receive_stream2.cc index 3aced70817..c7c1c30175 100644 --- a/video/video_receive_stream2.cc +++ b/video/video_receive_stream2.cc @@ -132,11 +132,11 @@ class WebRtcRecordableEncodedFrame : public RecordableEncodedFrame { return buffer_; } - std::optional color_space() const override { + std::optional color_space() const override { return color_space_; } - std::optional video_rotation() const override { + std::optional video_rotation() const override { return video_rotation_; } @@ -156,8 +156,8 @@ class WebRtcRecordableEncodedFrame : public RecordableEncodedFrame { VideoCodecType codec_; bool is_key_frame_; EncodedResolution resolution_; - std::optional color_space_; - std::optional video_rotation_; + std::optional color_space_; + std::optional video_rotation_; }; RenderResolution InitialDecoderResolution(const FieldTrialsView& field_trials) { @@ -174,21 +174,21 @@ RenderResolution InitialDecoderResolution(const FieldTrialsView& field_trials) { // Video decoder class to be used for unknown codecs. Doesn't support decoding // but logs messages to LS_ERROR. -class NullVideoDecoder : public webrtc::VideoDecoder { +class NullVideoDecoder : public VideoDecoder { public: bool Configure(const Settings& settings) override { RTC_LOG(LS_ERROR) << "Can't initialize NullVideoDecoder."; return true; } - int32_t Decode(const webrtc::EncodedImage& input_image, + int32_t Decode(const EncodedImage& input_image, int64_t render_time_ms) override { RTC_LOG(LS_ERROR) << "The NullVideoDecoder doesn't support decoding."; return WEBRTC_VIDEO_CODEC_OK; } int32_t RegisterDecodeCompleteCallback( - webrtc::DecodedImageCallback* callback) override { + DecodedImageCallback* callback) override { RTC_LOG(LS_ERROR) << "Can't register decode complete callback on NullVideoDecoder."; return WEBRTC_VIDEO_CODEC_OK; @@ -710,7 +710,7 @@ void VideoReceiveStream2::OnFrame(const VideoFrame& video_frame) { frame_meta.decode_timestamp); })); - webrtc::MutexLock lock(&pending_resolution_mutex_); + MutexLock lock(&pending_resolution_mutex_); if (pending_resolution_.has_value()) { if (!pending_resolution_->empty() && (video_frame.width() != static_cast(pending_resolution_->width) || @@ -729,7 +729,7 @@ void VideoReceiveStream2::OnFrame(const VideoFrame& video_frame) { } void VideoReceiveStream2::SetFrameDecryptor( - scoped_refptr frame_decryptor) { + scoped_refptr frame_decryptor) { rtp_video_stream_receiver_.SetFrameDecryptor(std::move(frame_decryptor)); } @@ -973,7 +973,7 @@ int VideoReceiveStream2::DecodeAndMaybeDispatchEncodedFrame( RTC_LOG(LS_ERROR) << "About to halt recordable encoded frame output due " "to too many buffered frames."; - webrtc::MutexLock lock(&pending_resolution_mutex_); + MutexLock lock(&pending_resolution_mutex_); if (IsKeyFrameAndUnspecifiedResolution(*frame_ptr) && !pending_resolution_.has_value()) pending_resolution_.emplace(); @@ -1001,7 +1001,7 @@ int VideoReceiveStream2::DecodeAndMaybeDispatchEncodedFrame( { // Fish out `pending_resolution_` to avoid taking the mutex on every lap // or dispatching under the mutex in the flush loop. - webrtc::MutexLock lock(&pending_resolution_mutex_); + MutexLock lock(&pending_resolution_mutex_); if (pending_resolution_.has_value()) pending_resolution = *pending_resolution_; } @@ -1134,7 +1134,7 @@ void VideoReceiveStream2::UpdatePlayoutDelays() const { } } -std::vector VideoReceiveStream2::GetSources() const { +std::vector VideoReceiveStream2::GetSources() const { RTC_DCHECK_RUN_ON(&worker_sequence_checker_); return source_tracker_.GetSources(); } diff --git a/video/video_receive_stream2_unittest.cc b/video/video_receive_stream2_unittest.cc index 61e18724b6..b19e0017e3 100644 --- a/video/video_receive_stream2_unittest.cc +++ b/video/video_receive_stream2_unittest.cc @@ -257,12 +257,10 @@ class VideoReceiveStream2Test : public ::testing::TestWithParam { video_receive_stream_ = nullptr; } timing_ = new VCMTiming(&env_.clock(), env_.field_trials()); - video_receive_stream_ = - std::make_unique( - env_, &fake_call_, kDefaultNumCpuCores, &packet_router_, - config_.Copy(), &call_stats_, absl::WrapUnique(timing_), - &nack_periodic_processor_, - UseMetronome() ? &decode_sync_ : nullptr); + video_receive_stream_ = std::make_unique( + env_, &fake_call_, kDefaultNumCpuCores, &packet_router_, config_.Copy(), + &call_stats_, absl::WrapUnique(timing_), &nack_periodic_processor_, + UseMetronome() ? &decode_sync_ : nullptr); video_receive_stream_->RegisterWithTransport( &rtp_stream_receiver_controller_); if (state) @@ -283,7 +281,7 @@ class VideoReceiveStream2Test : public ::testing::TestWithParam { test::RtcpPacketParser rtcp_packet_parser_; PacketRouter packet_router_; RtpStreamReceiverController rtp_stream_receiver_controller_; - std::unique_ptr video_receive_stream_; + std::unique_ptr video_receive_stream_; VCMTiming* timing_; test::FakeMetronome fake_metronome_; DecodeSynchronizer decode_sync_; @@ -595,7 +593,7 @@ TEST_P(VideoReceiveStream2Test, PassesNtpTime) { } TEST_P(VideoReceiveStream2Test, PassesRotation) { - const webrtc::VideoRotation kRotation = webrtc::kVideoRotation_180; + const VideoRotation kRotation = kVideoRotation_180; std::unique_ptr test_frame = test::FakeFrameBuilder() .Id(0) diff --git a/video/video_send_stream_impl.cc b/video/video_send_stream_impl.cc index 9f093f5f82..c0c8dff453 100644 --- a/video/video_send_stream_impl.cc +++ b/video/video_send_stream_impl.cc @@ -228,10 +228,9 @@ int GetEncoderPriorityBitrate(std::string codec_name, const FieldTrialsView& field_trials) { int priority_bitrate = 0; if (PayloadStringToCodecType(codec_name) == VideoCodecType::kVideoCodecAV1) { - webrtc::FieldTrialParameter av1_priority_bitrate("bitrate", 0); - webrtc::ParseFieldTrial( - {&av1_priority_bitrate}, - field_trials.Lookup("WebRTC-AV1-OverridePriorityBitrate")); + FieldTrialParameter av1_priority_bitrate("bitrate", 0); + ParseFieldTrial({&av1_priority_bitrate}, + field_trials.Lookup("WebRTC-AV1-OverridePriorityBitrate")); priority_bitrate = av1_priority_bitrate; } return priority_bitrate; @@ -299,9 +298,8 @@ size_t CalculateMaxHeaderSize(const RtpConfig& config) { VideoStreamEncoder::BitrateAllocationCallbackType GetBitrateAllocationCallbackType(const VideoSendStream::Config& config, const FieldTrialsView& field_trials) { - if (webrtc::RtpExtension::FindHeaderExtensionByUri( - config.rtp.extensions, - webrtc::RtpExtension::kVideoLayersAllocationUri, + if (RtpExtension::FindHeaderExtensionByUri( + config.rtp.extensions, RtpExtension::kVideoLayersAllocationUri, config.crypto_options.srtp.enable_encrypted_rtp_header_extensions ? RtpExtension::Filter::kPreferEncryptedExtension : RtpExtension::Filter::kDiscardEncryptedExtension)) { @@ -349,7 +347,7 @@ std::unique_ptr CreateVideoStreamEncoder( VideoStreamEncoder::BitrateAllocationCallbackType bitrate_allocation_callback_type, Metronome* metronome, - webrtc::VideoEncoderFactory::EncoderSelectorInterface* encoder_selector) { + VideoEncoderFactory::EncoderSelectorInterface* encoder_selector) { std::unique_ptr encoder_queue = env.task_queue_factory().CreateTaskQueue( "EncoderQueue", TaskQueueFactory::Priority::NORMAL); @@ -542,7 +540,7 @@ VideoSendStreamImpl::GetAdaptationResources() { } void VideoSendStreamImpl::SetSource( - VideoSourceInterface* source, + VideoSourceInterface* source, const DegradationPreference& degradation_preference) { RTC_DCHECK_RUN_ON(&thread_checker_); video_stream_encoder_->SetSource(source, degradation_preference); diff --git a/video/video_send_stream_impl_unittest.cc b/video/video_send_stream_impl_unittest.cc index 7a11b78eba..f0023f12b9 100644 --- a/video/video_send_stream_impl_unittest.cc +++ b/video/video_send_stream_impl_unittest.cc @@ -126,8 +126,7 @@ class MockRtpVideoSender : public RtpVideoSenderInterface { MOCK_METHOD(void, SetEncodingData, (size_t, size_t, size_t), (override)); MOCK_METHOD(std::vector, GetSentRtpPacketInfos, - (uint32_t ssrc, - webrtc::ArrayView sequence_numbers), + (uint32_t ssrc, ArrayView sequence_numbers), (const, override)); MOCK_METHOD(void, SetFecAllowed, (bool fec_allowed), (override)); @@ -217,7 +216,7 @@ class VideoSendStreamImplTest : public ::testing::Test { protected: GlobalSimulatedTimeController time_controller_; - webrtc::test::ScopedKeyValueConfig field_trials_; + test::ScopedKeyValueConfig field_trials_; NiceMock transport_; NiceMock transport_controller_; NiceMock bitrate_allocator_; diff --git a/video/video_send_stream_tests.cc b/video/video_send_stream_tests.cc index 69e0c8edf3..d73601ebc2 100644 --- a/video/video_send_stream_tests.cc +++ b/video/video_send_stream_tests.cc @@ -126,7 +126,7 @@ namespace webrtc { namespace test { class VideoSendStreamPeer { public: - explicit VideoSendStreamPeer(webrtc::VideoSendStream* base_class_stream) + explicit VideoSendStreamPeer(VideoSendStream* base_class_stream) : internal_stream_( static_cast(base_class_stream)) {} std::optional GetPacingFactorOverride() const { @@ -1754,8 +1754,8 @@ TEST_F(VideoSendStreamTest, ChangingNetworkRoute) { } private: - webrtc::SequenceChecker module_process_thread_; - webrtc::SequenceChecker task_queue_thread_; + SequenceChecker module_process_thread_; + SequenceChecker task_queue_thread_; TaskQueueBase* const task_queue_; RtpHeaderExtensionMap extensions_; Call* call_ RTC_GUARDED_BY(task_queue_thread_); @@ -1771,7 +1771,7 @@ TEST_F(VideoSendStreamTest, DISABLED_RelayToDirectRoute) { static const int kStartBitrateBps = 300000; static const int kRelayBandwidthCapBps = 800000; static const int kMinPacketsToSend = 100; - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( field_trials_, "WebRTC-Bwe-NetworkRouteConstraints/relay_cap:" + std::to_string(kRelayBandwidthCapBps) + "bps/"); @@ -1858,8 +1858,8 @@ TEST_F(VideoSendStreamTest, DISABLED_RelayToDirectRoute) { } private: - webrtc::SequenceChecker module_process_thread_; - webrtc::SequenceChecker task_queue_thread_; + SequenceChecker module_process_thread_; + SequenceChecker task_queue_thread_; TaskQueueBase* const task_queue_; Call* call_ RTC_GUARDED_BY(task_queue_thread_); int packets_sent_ RTC_GUARDED_BY(task_queue_thread_); @@ -2049,11 +2049,11 @@ class MaxPaddingSetTest : public test::SendTest { } private: - webrtc::SequenceChecker task_queue_thread_; + SequenceChecker task_queue_thread_; Call* call_ RTC_GUARDED_BY(task_queue_thread_) = nullptr; VideoSendStream::Config send_stream_config_{nullptr}; VideoEncoderConfig encoder_config_; - webrtc::SequenceChecker module_process_thread_; + SequenceChecker module_process_thread_; uint32_t packets_sent_ RTC_GUARDED_BY(task_queue_thread_) = 0; bool running_without_padding_ RTC_GUARDED_BY(task_queue_thread_); T* const stream_resetter_; @@ -2729,7 +2729,7 @@ TEST_F(VideoSendStreamTest, ReconfigureBitratesSetsEncoderBitratesCorrectly) { // TODO(bugs.webrtc.org/12058): If these fields trial are on, we get lower // bitrates than expected by this test, due to encoder pushback and subtracted // overhead. - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( field_trials_, "WebRTC-VideoRateControl/bitrate_adjuster:false/"); class EncoderBitrateThresholdObserver : public test::SendTest, @@ -2919,11 +2919,11 @@ TEST_F(VideoSendStreamTest, ReconfigureBitratesSetsEncoderBitratesCorrectly) { int num_rate_allocator_creations_; int num_encoder_initializations_; - webrtc::Call* call_; - webrtc::VideoSendStream* send_stream_; + Call* call_; + VideoSendStream* send_stream_; test::VideoEncoderProxyFactory encoder_factory_; std::unique_ptr bitrate_allocator_factory_; - webrtc::VideoEncoderConfig encoder_config_; + VideoEncoderConfig encoder_config_; } test(env(), task_queue()); RunBaseTest(&test); @@ -3380,7 +3380,7 @@ class Vp9HeaderObserver : public test::SendTest { test::FunctionVideoEncoderFactory encoder_factory_; const Vp9TestParams params_; VideoCodecVP9 vp9_settings_; - webrtc::VideoEncoderConfig encoder_config_; + VideoEncoderConfig encoder_config_; bool last_packet_marker_ = false; std::optional last_packet_sequence_number_; std::optional last_packet_timestamp_; @@ -3979,7 +3979,7 @@ class ContentSwitchTest : public test::SendTest { internal_send_peer.GetPacingFactorOverride().value_or(0.0f); float expected_pacing_factor = 1.1; // Strict pacing factor. VideoSendStream::Stats stats = send_stream_->GetStats(); - if (stats.content_type == webrtc::VideoContentType::SCREENSHARE) { + if (stats.content_type == VideoContentType::SCREENSHARE) { expected_pacing_factor = 1.0f; // Currently used pacing factor in ALR. } @@ -4123,7 +4123,7 @@ void VideoSendStreamTest::TestTemporalLayers( VideoSendStream::Config* send_config, std::vector* receive_configs, VideoEncoderConfig* encoder_config) override { - webrtc::VideoEncoder::EncoderInfo encoder_info; + VideoEncoder::EncoderInfo encoder_info; send_config->encoder_settings.encoder_factory = encoder_factory_; send_config->rtp.payload_name = payload_name_; send_config->rtp.payload_type = diff --git a/video/video_source_sink_controller_unittest.cc b/video/video_source_sink_controller_unittest.cc index bc4d263166..603c726a8c 100644 --- a/video/video_source_sink_controller_unittest.cc +++ b/video/video_source_sink_controller_unittest.cc @@ -43,13 +43,9 @@ class MockVideoSourceWithVideoFrame : public VideoSourceInterface { MOCK_METHOD(void, AddOrUpdateSink, - (webrtc::VideoSinkInterface*, - const webrtc::VideoSinkWants&), - (override)); - MOCK_METHOD(void, - RemoveSink, - (webrtc::VideoSinkInterface*), + (VideoSinkInterface*, const VideoSinkWants&), (override)); + MOCK_METHOD(void, RemoveSink, (VideoSinkInterface*), (override)); MOCK_METHOD(void, RequestRefreshFrame, (), (override)); }; diff --git a/video/video_stream_encoder.cc b/video/video_stream_encoder.cc index 86239444f7..212108ed06 100644 --- a/video/video_stream_encoder.cc +++ b/video/video_stream_encoder.cc @@ -346,9 +346,9 @@ VideoLayersAllocation CreateVideoLayersAllocation( encoder_config.VP9().interLayerPred == InterLayerPredMode::kOn; layers_allocation.resolution_and_frame_rate_is_valid = true; - std::vector aggregated_spatial_bitrate( - webrtc::kMaxTemporalStreams, DataRate::Zero()); - for (int si = 0; si < webrtc::kMaxSpatialLayers; ++si) { + std::vector aggregated_spatial_bitrate(kMaxTemporalStreams, + DataRate::Zero()); + for (int si = 0; si < kMaxSpatialLayers; ++si) { layers_allocation.resolution_and_frame_rate_is_valid = true; if (!target_bitrate.IsSpatialLayerUsed(si) || target_bitrate.GetSpatialLayerSum(si) == 0) { @@ -692,10 +692,9 @@ VideoStreamEncoder::VideoStreamEncoder( const VideoStreamEncoderSettings& settings, std::unique_ptr overuse_detector, std::unique_ptr frame_cadence_adapter, - std::unique_ptr - encoder_queue, + std::unique_ptr encoder_queue, BitrateAllocationCallbackType allocation_cb_type, - webrtc::VideoEncoderFactory::EncoderSelectorInterface* encoder_selector) + VideoEncoderFactory::EncoderSelectorInterface* encoder_selector) : env_(env), worker_queue_(TaskQueueBase::Current()), number_of_cores_(number_of_cores), @@ -1012,7 +1011,7 @@ void VideoStreamEncoder::ConfigureEncoder(VideoEncoderConfig config, ReconfigureEncoder(); } else { - webrtc::InvokeSetParametersCallback(callback, webrtc::RTCError::OK()); + InvokeSetParametersCallback(callback, RTCError::OK()); } }); } @@ -1485,7 +1484,7 @@ void VideoStreamEncoder::ReconfigureEncoder() { stream_resource_manager_.ConfigureQualityScaler(info); stream_resource_manager_.ConfigureBandwidthQualityScaler(info); - webrtc::RTCError encoder_configuration_result = webrtc::RTCError::OK(); + RTCError encoder_configuration_result = RTCError::OK(); if (!encoder_initialized_) { RTC_LOG(LS_WARNING) << "Failed to initialize " @@ -1497,14 +1496,13 @@ void VideoStreamEncoder::ReconfigureEncoder() { RequestEncoderSwitch(); } else { encoder_configuration_result = - webrtc::RTCError(RTCErrorType::UNSUPPORTED_OPERATION); + RTCError(RTCErrorType::UNSUPPORTED_OPERATION); } } if (!encoder_configuration_callbacks_.empty()) { for (auto& callback : encoder_configuration_callbacks_) { - webrtc::InvokeSetParametersCallback(callback, - encoder_configuration_result); + InvokeSetParametersCallback(callback, encoder_configuration_result); } encoder_configuration_callbacks_.clear(); } From ecb5a9f301a0daea4469f52b76e8b659b56d424d Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Fri, 9 May 2025 10:39:21 +0000 Subject: [PATCH 113/292] [cleanup] Fix redundant webrtc name specifier This CL was uploaded by git cl split. R=hta@webrtc.org No-Iwyu: LSC Bug: webrtc:42232595 Change-Id: I88cc8d0ddeafa2817e59e49f2a300de23e484dab Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390622 Reviewed-by: Harald Alvestrand Commit-Queue: Evan Shrubsole Auto-Submit: Evan Shrubsole Cr-Commit-Position: refs/heads/main@{#44585} --- test/scenario/column_printer.cc | 9 ++++----- test/scenario/probing_test.cc | 5 ++--- test/scenario/video_stream.cc | 2 +- test/scenario/video_stream_unittest.cc | 8 ++++---- 4 files changed, 11 insertions(+), 13 deletions(-) diff --git a/test/scenario/column_printer.cc b/test/scenario/column_printer.cc index 046925ceaa..76746869c0 100644 --- a/test/scenario/column_printer.cc +++ b/test/scenario/column_printer.cc @@ -15,10 +15,9 @@ namespace test { ColumnPrinter::ColumnPrinter(const ColumnPrinter&) = default; ColumnPrinter::~ColumnPrinter() = default; -ColumnPrinter::ColumnPrinter( - const char* headers, - std::function printer, - size_t max_length) +ColumnPrinter::ColumnPrinter(const char* headers, + std::function printer, + size_t max_length) : headers_(headers), printer_(printer), max_length_(max_length) {} ColumnPrinter ColumnPrinter::Fixed(const char* headers, std::string fields) { @@ -29,7 +28,7 @@ ColumnPrinter ColumnPrinter::Fixed(const char* headers, std::string fields) { ColumnPrinter ColumnPrinter::Lambda( const char* headers, - std::function printer, + std::function printer, size_t max_length) { return ColumnPrinter(headers, printer, max_length); } diff --git a/test/scenario/probing_test.cc b/test/scenario/probing_test.cc index 86653ced9b..f3cbe977c0 100644 --- a/test/scenario/probing_test.cc +++ b/test/scenario/probing_test.cc @@ -86,9 +86,8 @@ TEST(ProbingTest, ProbesRampsUpWhenVideoEncoderConfigChanges) { VideoStreamConfig video_config; video_config.encoder.codec = VideoStreamConfig::Encoder::Codec::kVideoCodecVP8; - video_config.encoder.simulcast_streams = {webrtc::ScalabilityMode::kL1T3, - webrtc::ScalabilityMode::kL1T3, - webrtc::ScalabilityMode::kL1T3}; + video_config.encoder.simulcast_streams = { + ScalabilityMode::kL1T3, ScalabilityMode::kL1T3, ScalabilityMode::kL1T3}; video_config.source.generator.width = 1280; video_config.source.generator.height = 720; diff --git a/test/scenario/video_stream.cc b/test/scenario/video_stream.cc index be211be9f0..0fe68161b2 100644 --- a/test/scenario/video_stream.cc +++ b/test/scenario/video_stream.cc @@ -195,7 +195,7 @@ CreateEncoderSpecificSettings(VideoStreamConfig config) { } VideoEncoderConfig CreateVideoEncoderConfig(VideoStreamConfig config) { - webrtc::VideoEncoder::EncoderInfo encoder_info; + VideoEncoder::EncoderInfo encoder_info; VideoEncoderConfig encoder_config; encoder_config.codec_type = config.encoder.codec; encoder_config.content_type = ConvertContentType(config.encoder.content_type); diff --git a/test/scenario/video_stream_unittest.cc b/test/scenario/video_stream_unittest.cc index b0d1f4d161..0ece8b0eff 100644 --- a/test/scenario/video_stream_unittest.cc +++ b/test/scenario/video_stream_unittest.cc @@ -104,9 +104,9 @@ TEST(VideoStreamTest, ReceivesVp8SimulcastFrames) { c->encoder.implementation = CodecImpl::kSoftware; c->encoder.codec = Codec::kVideoCodecVP8; // Enable simulcast. - c->encoder.simulcast_streams = {webrtc::ScalabilityMode::kL1T1, - webrtc::ScalabilityMode::kL1T1, - webrtc::ScalabilityMode::kL1T1}; + c->encoder.simulcast_streams = {ScalabilityMode::kL1T1, + ScalabilityMode::kL1T1, + ScalabilityMode::kL1T1}; }); s.RunFor(kRunTime); } @@ -218,7 +218,7 @@ TEST(VideoStreamTest, ResolutionAdaptsToAvailableBandwidth) { c->encoder.implementation = CodecImpl::kSoftware; c->encoder.codec = Codec::kVideoCodecVP9; // Enable SVC. - c->encoder.simulcast_streams = {webrtc::ScalabilityMode::kL2T1}; + c->encoder.simulcast_streams = {ScalabilityMode::kL2T1}; }); // Run for a few seconds, until streams have stabilized, From da1a66d203b49123eae3d3687f10dfd750ba00e9 Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Tue, 6 May 2025 13:36:17 +0200 Subject: [PATCH 114/292] Delete deprecated AudioDeviceModule::Create Bug: webrtc:413413572 Change-Id: I2d58ccc968b881acc9c5394fa188c827ca407a40 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390001 Reviewed-by: Henrik Andreassson Commit-Queue: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#44586} --- api/audio/BUILD.gn | 3 --- api/audio/audio_device.h | 8 +------- modules/audio_device/audio_device_impl.cc | 18 ++---------------- modules/audio_device/audio_device_impl.h | 6 ------ 4 files changed, 3 insertions(+), 32 deletions(-) diff --git a/api/audio/BUILD.gn b/api/audio/BUILD.gn index b762e1505b..09662a4efd 100644 --- a/api/audio/BUILD.gn +++ b/api/audio/BUILD.gn @@ -16,10 +16,7 @@ rtc_source_set("audio_device") { ] deps = [ "..:ref_count", - "..:scoped_refptr", - "../../rtc_base:checks", "../../rtc_base:stringutils", - "../task_queue", ] } diff --git a/api/audio/audio_device.h b/api/audio/audio_device.h index e6df56dc9a..5db7d7ecdd 100644 --- a/api/audio/audio_device.h +++ b/api/audio/audio_device.h @@ -13,10 +13,9 @@ #include #include + #include "api/audio/audio_device_defines.h" #include "api/ref_count.h" -#include "api/scoped_refptr.h" -#include "api/task_queue/task_queue_factory.h" namespace webrtc { @@ -60,11 +59,6 @@ class AudioDeviceModule : public webrtc::RefCountInterface { }; public: - // Creates a default ADM for usage in production code. - [[deprecated("Use CreateAudioDeviceModule instead")]] - static scoped_refptr Create( - AudioLayer audio_layer, TaskQueueFactory* task_queue_factory); - // Retrieve the currently utilized audio layer virtual int32_t ActiveAudioLayer(AudioLayer* audioLayer) const = 0; diff --git a/modules/audio_device/audio_device_impl.cc b/modules/audio_device/audio_device_impl.cc index eaeb89c420..da98848b96 100644 --- a/modules/audio_device/audio_device_impl.cc +++ b/modules/audio_device/audio_device_impl.cc @@ -73,20 +73,6 @@ namespace webrtc { absl_nullable scoped_refptr AudioDeviceModuleImpl::Create(const Environment& env, AudioLayer audio_layer) { - return Create(audio_layer, &env.task_queue_factory()); -} - -scoped_refptr AudioDeviceModule::Create( - AudioLayer audio_layer, - TaskQueueFactory* task_queue_factory) { - RTC_DLOG(LS_INFO) << __FUNCTION__; - return AudioDeviceModuleImpl::Create(audio_layer, task_queue_factory); -} - -// static -absl_nullable scoped_refptr -AudioDeviceModuleImpl::Create(AudioLayer audio_layer, - TaskQueueFactory* task_queue_factory) { RTC_DLOG(LS_INFO) << __FUNCTION__; // The "AudioDeviceModule::kWindowsCoreAudio2" audio layer has its own @@ -107,8 +93,8 @@ AudioDeviceModuleImpl::Create(AudioLayer audio_layer, } // Create the generic reference counted (platform independent) implementation. - auto audio_device = - make_ref_counted(audio_layer, task_queue_factory); + auto audio_device = make_ref_counted( + audio_layer, &env.task_queue_factory()); // Ensure that the current platform is supported. if (audio_device->CheckPlatform() == -1) { diff --git a/modules/audio_device/audio_device_impl.h b/modules/audio_device/audio_device_impl.h index 93707f21a9..609daf56cf 100644 --- a/modules/audio_device/audio_device_impl.h +++ b/modules/audio_device/audio_device_impl.h @@ -50,12 +50,6 @@ class AudioDeviceModuleImpl : public AudioDeviceModuleForTest { const Environment& env, AudioLayer audio_layer); - // TODO: bugs.webrtc.org/413413572 - Remove this variant when top level - // `AudioDeviceModule::Create` is no longer used. - static absl_nullable scoped_refptr Create( - AudioLayer audio_layer, - TaskQueueFactory* task_queue_factory); - AudioDeviceModuleImpl(AudioLayer audio_layer, TaskQueueFactory* task_queue_factory); // If `create_detached` is true, created ADM can be used on another thread From 5bd374cd52044b96dce4a1105f401fb573d2e9e2 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Fri, 9 May 2025 10:38:15 +0000 Subject: [PATCH 115/292] [cleanup] Fix redundant webrtc name specifier This CL was uploaded by git cl split. R=hta@webrtc.org No-Iwyu: LSC Bug: webrtc:42232595 Change-Id: Id49aa1f2c28952c719aa3fc91519a290582e16e0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390621 Reviewed-by: Harald Alvestrand Commit-Queue: Evan Shrubsole Auto-Submit: Evan Shrubsole Cr-Commit-Position: refs/heads/main@{#44587} --- .../codecs/test/video_codec_unittest.cc | 2 +- ...videocodec_test_fixture_config_unittest.cc | 6 +- .../test/videocodec_test_fixture_impl.cc | 42 ++++++----- .../codecs/test/videocodec_test_libvpx.cc | 2 +- .../codecs/test/videoprocessor.cc | 19 +++-- .../codecs/vp8/libvpx_vp8_decoder.cc | 9 +-- .../codecs/vp8/libvpx_vp8_encoder.cc | 2 +- .../codecs/vp8/test/vp8_impl_unittest.cc | 7 +- .../codecs/vp9/libvpx_vp9_decoder.cc | 9 +-- .../codecs/vp9/test/vp9_impl_unittest.cc | 11 ++- .../video_coding/deprecated/frame_buffer.cc | 2 +- .../video_coding/frame_helpers_unittest.cc | 8 +- .../video_coding/generic_decoder_unittest.cc | 2 +- modules/video_coding/media_opt_util.cc | 7 +- ...n_detection_settings_generator_unittest.cc | 18 ++--- .../utility/ivf_file_reader_unittest.cc | 5 +- .../utility/ivf_file_writer_unittest.cc | 5 +- .../utility/simulcast_rate_allocator.cc | 2 +- .../utility/simulcast_test_fixture_impl.cc | 48 ++++++------ .../video_coding/utility/simulcast_utility.cc | 4 +- .../video_codec_initializer_unittest.cc | 10 ++- .../video_coding/video_receiver2_unittest.cc | 4 +- .../balanced_degradation_settings_unittest.cc | 74 +++++++++---------- rtc_base/experiments/field_trial_parser.cc | 2 +- .../min_video_bitrate_experiment.cc | 21 +++--- .../quality_scaler_settings_unittest.cc | 2 +- .../quality_scaling_experiment_unittest.cc | 32 ++++---- .../experiments/struct_parameters_parser.cc | 6 +- test/call_config_utils.cc | 2 +- test/direct_transport.cc | 4 +- test/encoder_settings.cc | 4 +- test/fake_decoder.cc | 2 +- test/fake_vp8_decoder.cc | 2 +- test/field_trial.cc | 13 ++-- test/frame_generator.cc | 2 +- test/frame_generator_capturer.cc | 3 +- test/frame_utils.cc | 10 +-- test/gl/gl_renderer.cc | 4 +- test/linux/glx_renderer.cc | 2 +- .../audio/default_audio_quality_analyzer.cc | 4 +- .../analyzer/video/analyzing_video_sink.cc | 2 +- .../video/default_video_quality_analyzer.cc | 40 +++++----- ..._video_quality_analyzer_frame_in_flight.cc | 10 +-- ...ideo_quality_analyzer_frames_comparator.cc | 2 +- ...quality_analyzer_frames_comparator_test.cc | 2 +- ...ideo_quality_analyzer_metric_names_test.cc | 8 +- .../default_video_quality_analyzer_test.cc | 2 +- .../video/example_video_quality_analyzer.cc | 32 ++++---- .../video/quality_analyzing_video_decoder.cc | 13 ++-- .../video/quality_analyzing_video_encoder.cc | 2 +- .../video/simulcast_dummy_buffer_helper.cc | 8 +- test/pc/e2e/analyzer/video/video_dumping.cc | 2 +- .../e2e/analyzer/video/video_dumping_test.cc | 12 +-- ...video_quality_analyzer_injection_helper.cc | 4 +- .../video/video_quality_metrics_reporter.cc | 4 +- test/pc/e2e/cross_media_metrics_reporter.cc | 8 +- test/pc/e2e/media/media_helper.cc | 2 +- .../e2e/network_quality_metrics_reporter.cc | 4 +- test/pc/e2e/peer_connection_quality_test.cc | 36 +++++---- ...nnection_quality_test_metric_names_test.cc | 10 +-- .../e2e/peer_connection_quality_test_test.cc | 2 +- test/pc/e2e/sdp/sdp_changer.cc | 13 ++-- ..._based_network_quality_metrics_reporter.cc | 4 +- ...d_network_quality_metrics_reporter_test.cc | 10 +-- test/pc/e2e/stats_poller.cc | 10 +-- test/pc/e2e/stats_poller_test.cc | 2 +- test/pc/e2e/test_peer.cc | 8 +- test/pc/e2e/test_peer_factory.cc | 2 +- test/run_loop.cc | 2 +- test/scoped_key_value_config.cc | 2 +- test/test_main_lib.cc | 11 ++- .../copy_to_file_audio_capturer_unittest.cc | 4 +- test/testsupport/file_utils.cc | 8 +- test/testsupport/file_utils_unittest.cc | 55 +++++++------- ...xed_fps_video_frame_writer_adapter_test.cc | 2 +- .../ivf_video_frame_generator_unittest.cc | 7 +- test/testsupport/perf_test.cc | 17 ++--- test/testsupport/video_frame_writer.cc | 4 +- .../video_frame_writer_unittest.cc | 8 +- test/testsupport/y4m_frame_generator.cc | 14 ++-- test/testsupport/y4m_frame_reader_unittest.cc | 4 +- test/testsupport/y4m_frame_writer_unittest.cc | 4 +- test/testsupport/yuv_frame_reader_unittest.cc | 4 +- test/testsupport/yuv_frame_writer_unittest.cc | 4 +- .../time_controller_conformance_test.cc | 10 +-- test/vcm_capturer.cc | 2 +- test/video_codec_tester.cc | 6 +- test/video_codec_tester_unittest.cc | 4 +- 88 files changed, 405 insertions(+), 434 deletions(-) diff --git a/modules/video_coding/codecs/test/video_codec_unittest.cc b/modules/video_coding/codecs/test/video_codec_unittest.cc index 54be5b3c99..966da6b879 100644 --- a/modules/video_coding/codecs/test/video_codec_unittest.cc +++ b/modules/video_coding/codecs/test/video_codec_unittest.cc @@ -81,7 +81,7 @@ void VideoCodecUnitTest::FakeDecodeCompleteCallback::Decoded( } void VideoCodecUnitTest::SetUp() { - webrtc::test::CodecSettings(kVideoCodecVP8, &codec_settings_); + test::CodecSettings(kVideoCodecVP8, &codec_settings_); codec_settings_.startBitrate = kStartBitrate; codec_settings_.maxBitrate = kMaxBitrate; codec_settings_.maxFramerate = kMaxFramerate; diff --git a/modules/video_coding/codecs/test/videocodec_test_fixture_config_unittest.cc b/modules/video_coding/codecs/test/videocodec_test_fixture_config_unittest.cc index d9d43e067b..581a6286a5 100644 --- a/modules/video_coding/codecs/test/videocodec_test_fixture_config_unittest.cc +++ b/modules/video_coding/codecs/test/videocodec_test_fixture_config_unittest.cc @@ -42,20 +42,20 @@ TEST(Config, NumberOfCoresWithoutUseSingleCore) { TEST(Config, NumberOfTemporalLayersIsOne) { Config config; - webrtc::test::CodecSettings(kVideoCodecH264, &config.codec_settings); + test::CodecSettings(kVideoCodecH264, &config.codec_settings); EXPECT_EQ(1u, config.NumberOfTemporalLayers()); } TEST(Config, NumberOfTemporalLayers_Vp8) { Config config; - webrtc::test::CodecSettings(kVideoCodecVP8, &config.codec_settings); + test::CodecSettings(kVideoCodecVP8, &config.codec_settings); config.codec_settings.VP8()->numberOfTemporalLayers = kNumTemporalLayers; EXPECT_EQ(kNumTemporalLayers, config.NumberOfTemporalLayers()); } TEST(Config, NumberOfTemporalLayers_Vp9) { Config config; - webrtc::test::CodecSettings(kVideoCodecVP9, &config.codec_settings); + test::CodecSettings(kVideoCodecVP9, &config.codec_settings); config.codec_settings.VP9()->numberOfTemporalLayers = kNumTemporalLayers; EXPECT_EQ(kNumTemporalLayers, config.NumberOfTemporalLayers()); } diff --git a/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc b/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc index 5974266285..6d8f2eecdf 100644 --- a/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc +++ b/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc @@ -229,7 +229,7 @@ void VideoCodecTestFixtureImpl::Config::SetCodecSettings( size_t height) { codec_name = codec_name_to_set; VideoCodecType codec_type = PayloadStringToCodecType(codec_name); - webrtc::test::CodecSettings(codec_type, &codec_settings); + test::CodecSettings(codec_type, &codec_settings); // TODO(brandtr): Move the setting of `width` and `height` to the tests, and // DCHECK that they are set before initializing the codec instead. @@ -390,22 +390,22 @@ std::string VideoCodecTestFixtureImpl::Config::CodecName() const { // TODO(kthelgason): Move this out of the test fixture impl and // make available as a shared utility class. void VideoCodecTestFixtureImpl::H264KeyframeChecker::CheckEncodedFrame( - webrtc::VideoCodecType codec, + VideoCodecType codec, const EncodedImage& encoded_frame) const { EXPECT_EQ(kVideoCodecH264, codec); bool contains_sps = false; bool contains_pps = false; bool contains_idr = false; - const std::vector nalu_indices = - webrtc::H264::FindNaluIndices(encoded_frame); - for (const webrtc::H264::NaluIndex& index : nalu_indices) { - webrtc::H264::NaluType nalu_type = webrtc::H264::ParseNaluType( - encoded_frame.data()[index.payload_start_offset]); - if (nalu_type == webrtc::H264::NaluType::kSps) { + const std::vector nalu_indices = + H264::FindNaluIndices(encoded_frame); + for (const H264::NaluIndex& index : nalu_indices) { + H264::NaluType nalu_type = + H264::ParseNaluType(encoded_frame.data()[index.payload_start_offset]); + if (nalu_type == H264::NaluType::kSps) { contains_sps = true; - } else if (nalu_type == webrtc::H264::NaluType::kPps) { + } else if (nalu_type == H264::NaluType::kPps) { contains_pps = true; - } else if (nalu_type == webrtc::H264::NaluType::kIdr) { + } else if (nalu_type == H264::NaluType::kIdr) { contains_idr = true; } } @@ -457,16 +457,18 @@ class VideoCodecTestFixtureImpl::CpuProcessTime final { }; VideoCodecTestFixtureImpl::VideoCodecTestFixtureImpl(Config config) - : encoder_factory_(std::make_unique>()), - decoder_factory_(std::make_unique>()), + : encoder_factory_( + std::make_unique< + VideoEncoderFactoryTemplate>()), + decoder_factory_( + std::make_unique< + VideoDecoderFactoryTemplate>()), env_(CreateEnvironment()), config_(config) {} diff --git a/modules/video_coding/codecs/test/videocodec_test_libvpx.cc b/modules/video_coding/codecs/test/videocodec_test_libvpx.cc index 481fc775c1..c67790d959 100644 --- a/modules/video_coding/codecs/test/videocodec_test_libvpx.cc +++ b/modules/video_coding/codecs/test/videocodec_test_libvpx.cc @@ -51,7 +51,7 @@ const size_t kNumFirstFramesToSkipAtRdPerfAnalysis = 60; class QpFrameChecker : public VideoCodecTestFixture::EncodedFrameChecker { public: - void CheckEncodedFrame(webrtc::VideoCodecType codec, + void CheckEncodedFrame(VideoCodecType codec, const EncodedImage& encoded_frame) const override { int qp; if (codec == kVideoCodecVP8) { diff --git a/modules/video_coding/codecs/test/videoprocessor.cc b/modules/video_coding/codecs/test/videoprocessor.cc index 5b5c2bb80a..bec0e353c6 100644 --- a/modules/video_coding/codecs/test/videoprocessor.cc +++ b/modules/video_coding/codecs/test/videoprocessor.cc @@ -70,13 +70,13 @@ size_t GetMaxNaluSizeBytes(const EncodedImage& encoded_frame, if (config.codec_settings.codecType != kVideoCodecH264) return 0; - std::vector nalu_indices = - webrtc::H264::FindNaluIndices(encoded_frame); + std::vector nalu_indices = + H264::FindNaluIndices(encoded_frame); RTC_CHECK(!nalu_indices.empty()); size_t max_size = 0; - for (const webrtc::H264::NaluIndex& index : nalu_indices) + for (const H264::NaluIndex& index : nalu_indices) max_size = std::max(max_size, index.payload_size); return max_size; @@ -155,7 +155,7 @@ void CalculateFrameQuality(const I420BufferInterface& ref_buffer, } // namespace VideoProcessor::VideoProcessor(const Environment& env, - webrtc::VideoEncoder* encoder, + VideoEncoder* encoder, VideoDecoderList* decoders, FrameReader* input_frame_reader, const VideoCodecTestFixture::Config& config, @@ -275,7 +275,7 @@ void VideoProcessor::ProcessFrame() { .set_video_frame_buffer(buffer) .set_rtp_timestamp(static_cast(timestamp)) .set_timestamp_ms(static_cast(timestamp / kMsToRtpTimestamp)) - .set_rotation(webrtc::kVideoRotation_0) + .set_rotation(kVideoRotation_0) .build(); // Store input frame as a reference for quality calculations. if (config_.decode && !config_.measure_cpu) { @@ -382,9 +382,8 @@ int32_t VideoProcessor::VideoProcessorDecodeCompleteCallback::Decoded( return 0; } -void VideoProcessor::FrameEncoded( - const webrtc::EncodedImage& encoded_image, - const webrtc::CodecSpecificInfo& codec_specific) { +void VideoProcessor::FrameEncoded(const EncodedImage& encoded_image, + const CodecSpecificInfo& codec_specific) { RTC_DCHECK_RUN_ON(&sequence_checker_); // For the highest measurement accuracy of the encode time, the start/stop @@ -457,7 +456,7 @@ void VideoProcessor::FrameEncoded( frame_stat->non_ref_for_inter_layer_pred = true; } - const webrtc::EncodedImage* encoded_image_for_decode = &encoded_image; + const EncodedImage* encoded_image_for_decode = &encoded_image; if (config_.decode || !encoded_frame_writers_->empty()) { if (num_spatial_layers > 1) { encoded_image_for_decode = BuildAndStoreSuperframe( @@ -660,7 +659,7 @@ void VideoProcessor::DecodeFrame(const EncodedImage& encoded_image, decoders_->at(spatial_idx)->Decode(encoded_image, 0); } -const webrtc::EncodedImage* VideoProcessor::BuildAndStoreSuperframe( +const EncodedImage* VideoProcessor::BuildAndStoreSuperframe( const EncodedImage& encoded_image, const VideoCodecType /* codec */, size_t /* frame_number */, diff --git a/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc b/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc index 3eef8e5c11..8bb121ff0d 100644 --- a/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc +++ b/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc @@ -263,11 +263,10 @@ int LibvpxVp8Decoder::Decode(const EncodedImage& input_image, return WEBRTC_VIDEO_CODEC_OK; } -int LibvpxVp8Decoder::ReturnFrame( - const vpx_image_t* img, - uint32_t timestamp, - int qp, - const webrtc::ColorSpace* explicit_color_space) { +int LibvpxVp8Decoder::ReturnFrame(const vpx_image_t* img, + uint32_t timestamp, + int qp, + const ColorSpace* explicit_color_space) { if (img == NULL) { // Decoder OK and NULL image => No show frame return WEBRTC_VIDEO_CODEC_NO_OUTPUT; diff --git a/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc b/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc index 1ec4ab985b..8730a47ffc 100644 --- a/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc +++ b/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc @@ -778,7 +778,7 @@ int LibvpxVp8Encoder::InitEncode(const VideoCodec* inst, // to converge somewhat. Set a minim filter size of 1.25 since some // outlier pixels deviate a bit from truth even at very low QP, // seeminly by bleeding into neighbours. - webrtc::CorruptionDetectionSettingsGenerator::TransientParameters{ + CorruptionDetectionSettingsGenerator::TransientParameters{ .max_qp = 127, .keyframe_threshold_offset = 1, .keyframe_stddev_offset = 2.0, diff --git a/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc b/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc index c0aca08ca1..2f537f0a91 100644 --- a/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc +++ b/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc @@ -70,9 +70,8 @@ using ::testing::NiceMock; using ::testing::Return; using ::testing::Values; using ::testing::WithParamInterface; -using EncoderInfo = webrtc::VideoEncoder::EncoderInfo; -using FramerateFractions = - absl::InlinedVector; +using EncoderInfo = VideoEncoder::EncoderInfo; +using FramerateFractions = absl::InlinedVector; namespace { constexpr uint32_t kLegacyScreenshareTl0BitrateKbps = 200; @@ -103,7 +102,7 @@ class TestVp8Impl : public VideoCodecUnitTest { } void ModifyCodecSettings(VideoCodec* codec_settings) override { - webrtc::test::CodecSettings(kVideoCodecVP8, codec_settings); + test::CodecSettings(kVideoCodecVP8, codec_settings); codec_settings->width = kWidth; codec_settings->height = kHeight; codec_settings->SetVideoEncoderComplexity( diff --git a/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc b/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc index 4d4e147d50..ee9b5e54f3 100644 --- a/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc +++ b/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc @@ -265,11 +265,10 @@ int LibvpxVp9Decoder::Decode(const EncodedImage& input_image, return WEBRTC_VIDEO_CODEC_OK; } -int LibvpxVp9Decoder::ReturnFrame( - const vpx_image_t* img, - uint32_t timestamp, - int qp, - const webrtc::ColorSpace* explicit_color_space) { +int LibvpxVp9Decoder::ReturnFrame(const vpx_image_t* img, + uint32_t timestamp, + int qp, + const ColorSpace* explicit_color_space) { if (img == nullptr) { // Decoder OK and nullptr image => No show frame. return WEBRTC_VIDEO_CODEC_NO_OUTPUT; diff --git a/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc b/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc index 3152ccbb1e..e23fe8b561 100644 --- a/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc +++ b/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc @@ -93,9 +93,8 @@ using ::testing::SizeIs; using ::testing::TypedEq; using ::testing::UnorderedElementsAreArray; using ::testing::WithArg; -using EncoderInfo = webrtc::VideoEncoder::EncoderInfo; -using FramerateFractions = - absl::InlinedVector; +using EncoderInfo = VideoEncoder::EncoderInfo; +using FramerateFractions = absl::InlinedVector; constexpr size_t kWidth = 1280; constexpr size_t kHeight = 720; @@ -108,7 +107,7 @@ const VideoEncoder::Settings kSettings(kCapabilities, VideoCodec DefaultCodecSettings() { VideoCodec codec_settings; - webrtc::test::CodecSettings(kVideoCodecVP9, &codec_settings); + test::CodecSettings(kVideoCodecVP9, &codec_settings); codec_settings.width = kWidth; codec_settings.height = kHeight; codec_settings.startBitrate = kBitrateKbps; @@ -145,7 +144,7 @@ class TestVp9Impl : public VideoCodecUnitTest { } void ModifyCodecSettings(VideoCodec* codec_settings) override { - webrtc::test::CodecSettings(kVideoCodecVP9, codec_settings); + test::CodecSettings(kVideoCodecVP9, codec_settings); codec_settings->width = kWidth; codec_settings->height = kHeight; ConfigureSvc(*codec_settings); @@ -2072,7 +2071,7 @@ INSTANTIATE_TEST_SUITE_P(All, class TestVp9ImplFrameDropping : public TestVp9Impl { protected: void ModifyCodecSettings(VideoCodec* codec_settings) override { - webrtc::test::CodecSettings(kVideoCodecVP9, codec_settings); + test::CodecSettings(kVideoCodecVP9, codec_settings); // We need to encode quite a lot of frames in this test. Use low resolution // to reduce execution time. codec_settings->width = 64; diff --git a/modules/video_coding/deprecated/frame_buffer.cc b/modules/video_coding/deprecated/frame_buffer.cc index 395f596f46..5b1af328a5 100644 --- a/modules/video_coding/deprecated/frame_buffer.cc +++ b/modules/video_coding/deprecated/frame_buffer.cc @@ -35,7 +35,7 @@ VCMFrameBuffer::VCMFrameBuffer() VCMFrameBuffer::~VCMFrameBuffer() {} -webrtc::VideoFrameType VCMFrameBuffer::FrameType() const { +VideoFrameType VCMFrameBuffer::FrameType() const { return _sessionInfo.FrameType(); } diff --git a/modules/video_coding/frame_helpers_unittest.cc b/modules/video_coding/frame_helpers_unittest.cc index ed319b68a3..2e5f4b33f2 100644 --- a/modules/video_coding/frame_helpers_unittest.cc +++ b/modules/video_coding/frame_helpers_unittest.cc @@ -33,11 +33,9 @@ using ::testing::ElementsAre; constexpr uint32_t kRtpTimestamp = 123456710; -webrtc::scoped_refptr CreateEncodedImageBufferOfSizeN( - size_t n, - uint8_t x) { - webrtc::scoped_refptr buffer = - EncodedImageBuffer::Create(n); +scoped_refptr CreateEncodedImageBufferOfSizeN(size_t n, + uint8_t x) { + scoped_refptr buffer = EncodedImageBuffer::Create(n); for (size_t i = 0; i < n; ++i) { buffer->data()[i] = static_cast(x + i); } diff --git a/modules/video_coding/generic_decoder_unittest.cc b/modules/video_coding/generic_decoder_unittest.cc index 52fc057956..29f8bdd59a 100644 --- a/modules/video_coding/generic_decoder_unittest.cc +++ b/modules/video_coding/generic_decoder_unittest.cc @@ -115,7 +115,7 @@ class GenericDecoderTest : public ::testing::Test { Clock* const clock_; test::ScopedKeyValueConfig field_trials_; VCMTiming timing_; - webrtc::test::FakeDecoder decoder_; + test::FakeDecoder decoder_; VCMDecodedFrameCallback vcm_callback_; VCMGenericDecoder generic_decoder_; ReceiveCallback user_callback_; diff --git a/modules/video_coding/media_opt_util.cc b/modules/video_coding/media_opt_util.cc index b0fb8f835a..1a7ba66554 100644 --- a/modules/video_coding/media_opt_util.cc +++ b/modules/video_coding/media_opt_util.cc @@ -447,10 +447,9 @@ bool VCMFecMethod::ProtectionFactor(const VCMProtectionParameters* parameters) { int VCMFecMethod::BitsPerFrame(const VCMProtectionParameters* parameters) { // When temporal layers are available FEC will only be applied on the base // layer. - const float bitRateRatio = - webrtc::SimulcastRateAllocator::GetTemporalRateAllocation( - parameters->numLayers, 0, - rate_control_settings_.Vp8BaseHeavyTl3RateAllocation()); + const float bitRateRatio = SimulcastRateAllocator::GetTemporalRateAllocation( + parameters->numLayers, 0, + rate_control_settings_.Vp8BaseHeavyTl3RateAllocation()); float frameRateRatio = powf(1 / 2.0, parameters->numLayers - 1); float bitRate = parameters->bitRate * bitRateRatio; float frameRate = parameters->frameRate * frameRateRatio; diff --git a/modules/video_coding/utility/corruption_detection_settings_generator_unittest.cc b/modules/video_coding/utility/corruption_detection_settings_generator_unittest.cc index 7e74b32afc..59cc95f58c 100644 --- a/modules/video_coding/utility/corruption_detection_settings_generator_unittest.cc +++ b/modules/video_coding/utility/corruption_detection_settings_generator_unittest.cc @@ -29,7 +29,7 @@ TEST(CorruptionDetectionSettingsGenerator, ExponentialFunctionStdDev) { .exponent_factor = 0.01857465, .exponent_offset = -4.26470513}, CorruptionDetectionSettingsGenerator::ErrorThresholds{}, - webrtc::CorruptionDetectionSettingsGenerator::TransientParameters{}); + CorruptionDetectionSettingsGenerator::TransientParameters{}); // 0.006 * e^(0.01857465 * 20 + 4.26470513) ~= 0.612 CorruptionDetectionFilterSettings settings = @@ -49,7 +49,7 @@ TEST(CorruptionDetectionSettingsGenerator, ExponentialFunctionThresholds) { .exponent_offset = -4.26470513}, CorruptionDetectionSettingsGenerator::ErrorThresholds{.luma = 5, .chroma = 6}, - webrtc::CorruptionDetectionSettingsGenerator::TransientParameters{}); + CorruptionDetectionSettingsGenerator::TransientParameters{}); CorruptionDetectionFilterSettings settings = settings_generator.OnFrame(/*is_keyframe=*/true, /*qp=*/20); @@ -62,7 +62,7 @@ TEST(CorruptionDetectionSettingsGenerator, RationalFunctionStdDev) { CorruptionDetectionSettingsGenerator::RationalFunctionParameters{ .numerator_factor = -5.5, .denumerator_term = -97, .offset = -1}, CorruptionDetectionSettingsGenerator::ErrorThresholds{}, - webrtc::CorruptionDetectionSettingsGenerator::TransientParameters{}); + CorruptionDetectionSettingsGenerator::TransientParameters{}); // (20 * -5.5) / (20 - 97) - 1 ~= 0.429 CorruptionDetectionFilterSettings settings = @@ -80,7 +80,7 @@ TEST(CorruptionDetectionSettingsGenerator, RationalFunctionThresholds) { .numerator_factor = -5.5, .denumerator_term = -97, .offset = -1}, CorruptionDetectionSettingsGenerator::ErrorThresholds{.luma = 5, .chroma = 6}, - webrtc::CorruptionDetectionSettingsGenerator::TransientParameters{}); + CorruptionDetectionSettingsGenerator::TransientParameters{}); CorruptionDetectionFilterSettings settings = settings_generator.OnFrame(/*is_keyframe=*/true, /*qp=*/20); @@ -96,7 +96,7 @@ TEST(CorruptionDetectionSettingsGenerator, TransientStdDevOffset) { CorruptionDetectionSettingsGenerator::ErrorThresholds{}, // Two frames with adjusted settings, including the keyframe. // Adjust the keyframe std_dev by 2. - webrtc::CorruptionDetectionSettingsGenerator::TransientParameters{ + CorruptionDetectionSettingsGenerator::TransientParameters{ .keyframe_stddev_offset = 2.0, .keyframe_offset_duration_frames = 2, }); @@ -128,7 +128,7 @@ TEST(CorruptionDetectionSettingsGenerator, TransientThresholdOffsets) { .chroma = 3}, // Two frames with adjusted settings, including the keyframe. // Adjust the error thresholds by 2. - webrtc::CorruptionDetectionSettingsGenerator::TransientParameters{ + CorruptionDetectionSettingsGenerator::TransientParameters{ .keyframe_threshold_offset = 2, .keyframe_offset_duration_frames = 2, }); @@ -169,7 +169,7 @@ TEST(CorruptionDetectionSettingsGenerator, StdDevUpperBound) { CorruptionDetectionSettingsGenerator::RationalFunctionParameters{ .numerator_factor = 1, .denumerator_term = 0, .offset = 41}, CorruptionDetectionSettingsGenerator::ErrorThresholds{}, - webrtc::CorruptionDetectionSettingsGenerator::TransientParameters{}); + CorruptionDetectionSettingsGenerator::TransientParameters{}); // `std_dev` capped at max 40.0, which is the limit for the protocol. EXPECT_THAT( @@ -183,7 +183,7 @@ TEST(CorruptionDetectionSettingsGenerator, StdDevLowerBound) { CorruptionDetectionSettingsGenerator::RationalFunctionParameters{ .numerator_factor = 1, .denumerator_term = 0, .offset = 1}, CorruptionDetectionSettingsGenerator::ErrorThresholds{}, - webrtc::CorruptionDetectionSettingsGenerator::TransientParameters{ + CorruptionDetectionSettingsGenerator::TransientParameters{ .std_dev_lower_bound = 5.0}); // `std_dev` capped at lower bound of 5.0. @@ -200,7 +200,7 @@ TEST(CorruptionDetectionSettingsGenerator, TreatsLargeQpChangeAsKeyFrame) { .chroma = 3}, // Two frames with adjusted settings, including the keyframe. // Adjust the error thresholds by 2. - webrtc::CorruptionDetectionSettingsGenerator::TransientParameters{ + CorruptionDetectionSettingsGenerator::TransientParameters{ .max_qp = 100, .keyframe_threshold_offset = 2, .keyframe_offset_duration_frames = 1, diff --git a/modules/video_coding/utility/ivf_file_reader_unittest.cc b/modules/video_coding/utility/ivf_file_reader_unittest.cc index bfd2b6140f..ede43e07f4 100644 --- a/modules/video_coding/utility/ivf_file_reader_unittest.cc +++ b/modules/video_coding/utility/ivf_file_reader_unittest.cc @@ -38,10 +38,9 @@ constexpr uint8_t kDummyPayload[4] = {'0', '1', '2', '3'}; class IvfFileReaderTest : public ::testing::Test { protected: void SetUp() override { - file_name_ = - webrtc::test::TempFilename(webrtc::test::OutputPath(), "test_file.ivf"); + file_name_ = test::TempFilename(test::OutputPath(), "test_file.ivf"); } - void TearDown() override { webrtc::test::RemoveFile(file_name_); } + void TearDown() override { test::RemoveFile(file_name_); } bool WriteDummyTestFrames(IvfFileWriter* file_writer, VideoCodecType codec_type, diff --git a/modules/video_coding/utility/ivf_file_writer_unittest.cc b/modules/video_coding/utility/ivf_file_writer_unittest.cc index 6be8a6c228..007448fd14 100644 --- a/modules/video_coding/utility/ivf_file_writer_unittest.cc +++ b/modules/video_coding/utility/ivf_file_writer_unittest.cc @@ -38,10 +38,9 @@ constexpr int kDefaultHeight = 720; class IvfFileWriterTest : public ::testing::Test { protected: void SetUp() override { - file_name_ = - webrtc::test::TempFilename(webrtc::test::OutputPath(), "test_file"); + file_name_ = test::TempFilename(test::OutputPath(), "test_file"); } - void TearDown() override { webrtc::test::RemoveFile(file_name_); } + void TearDown() override { test::RemoveFile(file_name_); } bool WriteDummyTestFrames(VideoCodecType codec_type, int width, diff --git a/modules/video_coding/utility/simulcast_rate_allocator.cc b/modules/video_coding/utility/simulcast_rate_allocator.cc index 1157d1a76f..2811d52a0c 100644 --- a/modules/video_coding/utility/simulcast_rate_allocator.cc +++ b/modules/video_coding/utility/simulcast_rate_allocator.cc @@ -330,7 +330,7 @@ SimulcastRateAllocator::ScreenshareTemporalLayerAllocation( return allocation; } -const VideoCodec& webrtc::SimulcastRateAllocator::GetCodec() const { +const VideoCodec& SimulcastRateAllocator::GetCodec() const { return codec_; } diff --git a/modules/video_coding/utility/simulcast_test_fixture_impl.cc b/modules/video_coding/utility/simulcast_test_fixture_impl.cc index a8a3e80fdd..129d76368b 100644 --- a/modules/video_coding/utility/simulcast_test_fixture_impl.cc +++ b/modules/video_coding/utility/simulcast_test_fixture_impl.cc @@ -311,12 +311,12 @@ void SimulcastTestFixtureImpl::SetUpCodec(const int* temporal_layer_profile) { EXPECT_TRUE(decoder_->Configure(decoder_settings)); input_buffer_ = I420Buffer::Create(kDefaultWidth, kDefaultHeight); input_buffer_->InitializeData(); - input_frame_ = std::make_unique( - webrtc::VideoFrame::Builder() - .set_video_frame_buffer(input_buffer_) - .set_rotation(webrtc::kVideoRotation_0) - .set_timestamp_us(0) - .build()); + input_frame_ = + std::make_unique(VideoFrame::Builder() + .set_video_frame_buffer(input_buffer_) + .set_rotation(kVideoRotation_0) + .set_timestamp_us(0) + .build()); } void SimulcastTestFixtureImpl::SetUpRateAllocator() { @@ -687,12 +687,12 @@ void SimulcastTestFixtureImpl::SwitchingToOneStream(int width, int height) { input_buffer_ = I420Buffer::Create(settings_.width, settings_.height); input_buffer_->InitializeData(); - input_frame_ = std::make_unique( - webrtc::VideoFrame::Builder() - .set_video_frame_buffer(input_buffer_) - .set_rotation(webrtc::kVideoRotation_0) - .set_timestamp_us(0) - .build()); + input_frame_ = + std::make_unique(VideoFrame::Builder() + .set_video_frame_buffer(input_buffer_) + .set_rotation(kVideoRotation_0) + .set_timestamp_us(0) + .build()); // The for loop above did not set the bitrate of the highest layer. settings_.simulcastStream[settings_.numberOfSimulcastStreams - 1].maxBitrate = @@ -732,12 +732,12 @@ void SimulcastTestFixtureImpl::SwitchingToOneStream(int width, int height) { // Resize `input_frame_` to the new resolution. input_buffer_ = I420Buffer::Create(settings_.width, settings_.height); input_buffer_->InitializeData(); - input_frame_ = std::make_unique( - webrtc::VideoFrame::Builder() - .set_video_frame_buffer(input_buffer_) - .set_rotation(webrtc::kVideoRotation_0) - .set_timestamp_us(0) - .build()); + input_frame_ = + std::make_unique(VideoFrame::Builder() + .set_video_frame_buffer(input_buffer_) + .set_rotation(kVideoRotation_0) + .set_timestamp_us(0) + .build()); EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types)); } @@ -896,12 +896,12 @@ void SimulcastTestFixtureImpl::TestStrideEncodeDecode() { int stride_uv = ((kDefaultWidth + 1) / 2) + 5; input_buffer_ = I420Buffer::Create(kDefaultWidth, kDefaultHeight, stride_y, stride_uv, stride_uv); - input_frame_ = std::make_unique( - webrtc::VideoFrame::Builder() - .set_video_frame_buffer(input_buffer_) - .set_rotation(webrtc::kVideoRotation_0) - .set_timestamp_us(0) - .build()); + input_frame_ = + std::make_unique(VideoFrame::Builder() + .set_video_frame_buffer(input_buffer_) + .set_rotation(kVideoRotation_0) + .set_timestamp_us(0) + .build()); // Set color. int plane_offset[kNumOfPlanes]; diff --git a/modules/video_coding/utility/simulcast_utility.cc b/modules/video_coding/utility/simulcast_utility.cc index ec1ad067cb..b6327aeeb4 100644 --- a/modules/video_coding/utility/simulcast_utility.cc +++ b/modules/video_coding/utility/simulcast_utility.cc @@ -89,8 +89,8 @@ bool SimulcastUtility::IsConferenceModeScreenshare( return encoder_config.content_type == VideoEncoderConfig::ContentType::kScreen && encoder_config.legacy_conference_mode && - (encoder_config.codec_type == webrtc::VideoCodecType::kVideoCodecVP8 || - encoder_config.codec_type == webrtc::VideoCodecType::kVideoCodecH264); + (encoder_config.codec_type == VideoCodecType::kVideoCodecVP8 || + encoder_config.codec_type == VideoCodecType::kVideoCodecH264); } int SimulcastUtility::NumberOfTemporalLayers(const VideoCodec& codec, diff --git a/modules/video_coding/video_codec_initializer_unittest.cc b/modules/video_coding/video_codec_initializer_unittest.cc index 849c780aca..f97bf88c58 100644 --- a/modules/video_coding/video_codec_initializer_unittest.cc +++ b/modules/video_coding/video_codec_initializer_unittest.cc @@ -84,15 +84,17 @@ class VideoCodecInitializerTest : public ::testing::Test { ASSERT_FALSE(num_spatial_streams.has_value()); VideoCodecVP8 vp8_settings = VideoEncoder::GetDefaultVp8Settings(); vp8_settings.numberOfTemporalLayers = num_temporal_streams; - config_.encoder_specific_settings = make_ref_counted< - webrtc::VideoEncoderConfig::Vp8EncoderSpecificSettings>(vp8_settings); + config_.encoder_specific_settings = + make_ref_counted( + vp8_settings); } else if (type == VideoCodecType::kVideoCodecVP9) { ASSERT_TRUE(num_spatial_streams.has_value()); VideoCodecVP9 vp9_settings = VideoEncoder::GetDefaultVp9Settings(); vp9_settings.numberOfSpatialLayers = num_spatial_streams.value(); vp9_settings.numberOfTemporalLayers = num_temporal_streams; - config_.encoder_specific_settings = make_ref_counted< - webrtc::VideoEncoderConfig::Vp9EncoderSpecificSettings>(vp9_settings); + config_.encoder_specific_settings = + make_ref_counted( + vp9_settings); } } diff --git a/modules/video_coding/video_receiver2_unittest.cc b/modules/video_coding/video_receiver2_unittest.cc index f7460a5310..8bce1de57c 100644 --- a/modules/video_coding/video_receiver2_unittest.cc +++ b/modules/video_coding/video_receiver2_unittest.cc @@ -54,7 +54,7 @@ class TestEncodedFrame : public EncodedFrame { SetPacketInfos(CreatePacketInfos(3)); } - void SetReceivedTime(webrtc::Timestamp received_time) { + void SetReceivedTime(Timestamp received_time) { received_time_ = received_time; } @@ -63,7 +63,7 @@ class TestEncodedFrame : public EncodedFrame { int64_t RenderTime() const override { return _renderTimeMs; } private: - webrtc::Timestamp received_time_ = webrtc::Timestamp::Millis(0); + Timestamp received_time_ = Timestamp::Millis(0); }; class VideoReceiver2Test : public ::testing::Test { diff --git a/rtc_base/experiments/balanced_degradation_settings_unittest.cc b/rtc_base/experiments/balanced_degradation_settings_unittest.cc index a9ad43398a..4f4ef11b69 100644 --- a/rtc_base/experiments/balanced_degradation_settings_unittest.cc +++ b/rtc_base/experiments/balanced_degradation_settings_unittest.cc @@ -58,7 +58,7 @@ void VerifyIsDefault( } // namespace TEST(BalancedDegradationSettings, GetsDefaultConfigIfNoList) { - webrtc::test::ScopedKeyValueConfig field_trials(""); + test::ScopedKeyValueConfig field_trials(""); BalancedDegradationSettings settings(field_trials); VerifyIsDefault(settings.GetConfigs()); EXPECT_TRUE(settings.CanAdaptUp(kVideoCodecVP8, 1, /*bitrate_bps*/ 1)); @@ -73,7 +73,7 @@ TEST(BalancedDegradationSettings, GetsDefaultConfigIfNoList) { } TEST(BalancedDegradationSettings, GetsConfig) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:11|22|33,fps:5|15|25,other:4|5|6/"); BalancedDegradationSettings settings(field_trials); @@ -115,7 +115,7 @@ TEST(BalancedDegradationSettings, GetsConfig) { } TEST(BalancedDegradationSettings, GetsDefaultConfigForZeroFpsValue) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:0|15|25/"); BalancedDegradationSettings settings(field_trials); @@ -123,7 +123,7 @@ TEST(BalancedDegradationSettings, GetsDefaultConfigForZeroFpsValue) { } TEST(BalancedDegradationSettings, GetsDefaultConfigIfPixelsDecreases) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|999|3000,fps:5|15|25/"); BalancedDegradationSettings settings(field_trials); @@ -131,7 +131,7 @@ TEST(BalancedDegradationSettings, GetsDefaultConfigIfPixelsDecreases) { } TEST(BalancedDegradationSettings, GetsDefaultConfigIfFramerateDecreases) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:5|4|25/"); BalancedDegradationSettings settings(field_trials); @@ -139,7 +139,7 @@ TEST(BalancedDegradationSettings, GetsDefaultConfigIfFramerateDecreases) { } TEST(BalancedDegradationSettings, GetsConfigWithSpecificFps) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:5|15|25,vp8_fps:7|8|9,vp9_fps:9|10|11," "h264_fps:11|12|13,av1_fps:1|2|3,generic_fps:13|14|15/"); @@ -182,7 +182,7 @@ TEST(BalancedDegradationSettings, GetsConfigWithSpecificFps) { } TEST(BalancedDegradationSettings, GetsDefaultConfigForZeroVp8FpsValue) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:7|15|25,vp8_fps:0|15|25/"); BalancedDegradationSettings settings(field_trials); @@ -190,7 +190,7 @@ TEST(BalancedDegradationSettings, GetsDefaultConfigForZeroVp8FpsValue) { } TEST(BalancedDegradationSettings, GetsDefaultConfigForInvalidFpsValue) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:7|15|25,vp8_fps:10|15|2000/"); BalancedDegradationSettings settings(field_trials); @@ -198,7 +198,7 @@ TEST(BalancedDegradationSettings, GetsDefaultConfigForInvalidFpsValue) { } TEST(BalancedDegradationSettings, GetsDefaultConfigIfVp8FramerateDecreases) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:4|5|25,vp8_fps:5|4|25/"); BalancedDegradationSettings settings(field_trials); @@ -206,7 +206,7 @@ TEST(BalancedDegradationSettings, GetsDefaultConfigIfVp8FramerateDecreases) { } TEST(BalancedDegradationSettings, GetsMinFps) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:5|15|25/"); BalancedDegradationSettings settings(field_trials); @@ -221,7 +221,7 @@ TEST(BalancedDegradationSettings, GetsMinFps) { } TEST(BalancedDegradationSettings, GetsVp8MinFps) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:5|15|25,vp8_fps:7|10|12/"); BalancedDegradationSettings settings(field_trials); @@ -236,7 +236,7 @@ TEST(BalancedDegradationSettings, GetsVp8MinFps) { } TEST(BalancedDegradationSettings, GetsMaxFps) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:5|15|25/"); BalancedDegradationSettings settings(field_trials); @@ -249,7 +249,7 @@ TEST(BalancedDegradationSettings, GetsMaxFps) { } TEST(BalancedDegradationSettings, GetsVp8MaxFps) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:5|15|25,vp8_fps:7|10|12/"); BalancedDegradationSettings settings(field_trials); @@ -262,7 +262,7 @@ TEST(BalancedDegradationSettings, GetsVp8MaxFps) { } TEST(BalancedDegradationSettings, GetsVp9Fps) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:5|15|25,vp9_fps:7|10|12/"); BalancedDegradationSettings settings(field_trials); @@ -271,7 +271,7 @@ TEST(BalancedDegradationSettings, GetsVp9Fps) { } TEST(BalancedDegradationSettings, GetsH264Fps) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:5|15|25,h264_fps:8|11|13/"); BalancedDegradationSettings settings(field_trials); @@ -280,7 +280,7 @@ TEST(BalancedDegradationSettings, GetsH264Fps) { } TEST(BalancedDegradationSettings, GetsGenericFps) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:5|15|25,generic_fps:9|12|14/"); BalancedDegradationSettings settings(field_trials); @@ -290,7 +290,7 @@ TEST(BalancedDegradationSettings, GetsGenericFps) { } TEST(BalancedDegradationSettings, GetsUnlimitedForMaxValidFps) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:5|15|100,vp8_fps:30|100|100/"); const int kUnlimitedFps = std::numeric_limits::max(); @@ -302,7 +302,7 @@ TEST(BalancedDegradationSettings, GetsUnlimitedForMaxValidFps) { } TEST(BalancedDegradationSettings, GetsConfigWithBitrate) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:11|22|33,fps:5|15|25,kbps:44|88|99,kbps_res:55|111|222," "vp8_kbps:11|12|13,vp8_kbps_res:14|15|16," @@ -349,7 +349,7 @@ TEST(BalancedDegradationSettings, GetsConfigWithBitrate) { } TEST(BalancedDegradationSettings, GetsDefaultConfigIfBitrateDecreases) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:11|22|33,fps:5|15|25,kbps:44|43|99/"); BalancedDegradationSettings settings(field_trials); @@ -358,7 +358,7 @@ TEST(BalancedDegradationSettings, GetsDefaultConfigIfBitrateDecreases) { TEST(BalancedDegradationSettings, GetsDefaultConfigIfBitrateDecreasesWithUnsetValue) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:11|22|33,fps:5|15|25,kbps:44|0|43/"); BalancedDegradationSettings settings(field_trials); @@ -367,7 +367,7 @@ TEST(BalancedDegradationSettings, TEST(BalancedDegradationSettings, CanAdaptUp) { VideoCodecType vp8 = kVideoCodecVP8; - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000|4000,fps:5|15|25|30,kbps:0|80|0|90," "vp9_kbps:40|50|60|70/"); @@ -382,7 +382,7 @@ TEST(BalancedDegradationSettings, CanAdaptUp) { } TEST(BalancedDegradationSettings, CanAdaptUpWithCodecType) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000|4000,fps:5|15|25|30,vp8_kbps:0|30|40|50," "vp9_kbps:0|60|70|80,h264_kbps:0|55|65|75,av1_kbps:0|77|88|99," @@ -402,7 +402,7 @@ TEST(BalancedDegradationSettings, CanAdaptUpWithCodecType) { TEST(BalancedDegradationSettings, CanAdaptUpResolution) { VideoCodecType vp8 = kVideoCodecVP8; - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000|4000,fps:5|15|25|30,kbps_res:0|80|0|90," "vp9_kbps_res:40|50|60|70/"); @@ -417,7 +417,7 @@ TEST(BalancedDegradationSettings, CanAdaptUpResolution) { } TEST(BalancedDegradationSettings, CanAdaptUpResolutionWithCodecType) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000|4000,fps:5|15|25|30,vp8_kbps_res:0|30|40|50," "vp9_kbps_res:0|60|70|80,h264_kbps_res:0|55|65|75," @@ -436,7 +436,7 @@ TEST(BalancedDegradationSettings, CanAdaptUpResolutionWithCodecType) { } TEST(BalancedDegradationSettings, GetsFpsDiff) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:5|15|25,fps_diff:0|-2|3/"); BalancedDegradationSettings settings(field_trials); @@ -450,7 +450,7 @@ TEST(BalancedDegradationSettings, GetsFpsDiff) { } TEST(BalancedDegradationSettings, GetsNoFpsDiffIfValueBelowMinSetting) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:5|15|25,fps_diff:-100|-99|-101/"); // Min valid fps_diff setting: -99. @@ -461,7 +461,7 @@ TEST(BalancedDegradationSettings, GetsNoFpsDiffIfValueBelowMinSetting) { } TEST(BalancedDegradationSettings, QpThresholdsNotSetByDefault) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:5|15|25/"); BalancedDegradationSettings settings(field_trials); @@ -473,7 +473,7 @@ TEST(BalancedDegradationSettings, QpThresholdsNotSetByDefault) { } TEST(BalancedDegradationSettings, GetsConfigWithQpThresholds) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:5|15|25,vp8_qp_low:89|90|88," "vp8_qp_high:90|91|92,vp9_qp_low:27|28|29,vp9_qp_high:120|130|140," @@ -518,7 +518,7 @@ TEST(BalancedDegradationSettings, GetsConfigWithQpThresholds) { } TEST(BalancedDegradationSettings, GetsDefaultConfigIfOnlyHasLowThreshold) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:5|15|25,vp8_qp_low:89|90|88/"); BalancedDegradationSettings settings(field_trials); @@ -526,7 +526,7 @@ TEST(BalancedDegradationSettings, GetsDefaultConfigIfOnlyHasLowThreshold) { } TEST(BalancedDegradationSettings, GetsDefaultConfigIfOnlyHasHighThreshold) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:5|15|25,vp8_qp_high:90|91|92/"); BalancedDegradationSettings settings(field_trials); @@ -534,7 +534,7 @@ TEST(BalancedDegradationSettings, GetsDefaultConfigIfOnlyHasHighThreshold) { } TEST(BalancedDegradationSettings, GetsDefaultConfigIfLowEqualsHigh) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:5|15|25," "vp8_qp_low:89|90|88,vp8_qp_high:90|91|88/"); @@ -543,7 +543,7 @@ TEST(BalancedDegradationSettings, GetsDefaultConfigIfLowEqualsHigh) { } TEST(BalancedDegradationSettings, GetsDefaultConfigIfLowGreaterThanHigh) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:5|15|25," "vp8_qp_low:89|90|88,vp8_qp_high:90|91|87/"); @@ -552,7 +552,7 @@ TEST(BalancedDegradationSettings, GetsDefaultConfigIfLowGreaterThanHigh) { } TEST(BalancedDegradationSettings, GetsDefaultConfigForZeroQpValue) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:5|15|25," "vp8_qp_low:89|0|88,vp8_qp_high:90|91|92/"); @@ -561,7 +561,7 @@ TEST(BalancedDegradationSettings, GetsDefaultConfigForZeroQpValue) { } TEST(BalancedDegradationSettings, GetsVp8QpThresholds) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:5|15|25," "vp8_qp_low:89|90|88,vp8_qp_high:90|91|92/"); @@ -577,7 +577,7 @@ TEST(BalancedDegradationSettings, GetsVp8QpThresholds) { } TEST(BalancedDegradationSettings, GetsVp9QpThresholds) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:5|15|25," "vp9_qp_low:55|56|57,vp9_qp_high:155|156|157/"); @@ -589,7 +589,7 @@ TEST(BalancedDegradationSettings, GetsVp9QpThresholds) { } TEST(BalancedDegradationSettings, GetsH264QpThresholds) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:5|15|25," "h264_qp_low:21|22|23,h264_qp_high:41|43|42/"); @@ -601,7 +601,7 @@ TEST(BalancedDegradationSettings, GetsH264QpThresholds) { } TEST(BalancedDegradationSettings, GetsGenericQpThresholds) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-BalancedDegradationSettings/" "pixels:1000|2000|3000,fps:5|15|25," "generic_qp_low:2|3|4,generic_qp_high:22|23|24/"); diff --git a/rtc_base/experiments/field_trial_parser.cc b/rtc_base/experiments/field_trial_parser.cc index 1288c72c78..fa00c4e32f 100644 --- a/rtc_base/experiments/field_trial_parser.cc +++ b/rtc_base/experiments/field_trial_parser.cc @@ -193,7 +193,7 @@ bool FieldTrialFlag::Get() const { return value_; } -webrtc::FieldTrialFlag::operator bool() const { +FieldTrialFlag::operator bool() const { return value_; } diff --git a/rtc_base/experiments/min_video_bitrate_experiment.cc b/rtc_base/experiments/min_video_bitrate_experiment.cc index dd216b8b27..2bd8911f36 100644 --- a/rtc_base/experiments/min_video_bitrate_experiment.cc +++ b/rtc_base/experiments/min_video_bitrate_experiment.cc @@ -68,22 +68,21 @@ std::optional GetExperimentalMinVideoBitrate( } if (field_trials.IsEnabled(kMinVideoBitrateExperiment)) { - webrtc::FieldTrialFlag enabled("Enabled"); + FieldTrialFlag enabled("Enabled"); // Backwards-compatibility with an old experiment - a generic minimum which, // if set, applies to all codecs. - webrtc::FieldTrialOptional min_video_bitrate("br"); + FieldTrialOptional min_video_bitrate("br"); // New experiment - per-codec minimum bitrate. - webrtc::FieldTrialOptional min_bitrate_vp8("vp8_br"); - webrtc::FieldTrialOptional min_bitrate_vp9("vp9_br"); - webrtc::FieldTrialOptional min_bitrate_av1("av1_br"); - webrtc::FieldTrialOptional min_bitrate_h264("h264_br"); - - webrtc::ParseFieldTrial( - {&enabled, &min_video_bitrate, &min_bitrate_vp8, &min_bitrate_vp9, - &min_bitrate_av1, &min_bitrate_h264}, - field_trials.Lookup(kMinVideoBitrateExperiment)); + FieldTrialOptional min_bitrate_vp8("vp8_br"); + FieldTrialOptional min_bitrate_vp9("vp9_br"); + FieldTrialOptional min_bitrate_av1("av1_br"); + FieldTrialOptional min_bitrate_h264("h264_br"); + + ParseFieldTrial({&enabled, &min_video_bitrate, &min_bitrate_vp8, + &min_bitrate_vp9, &min_bitrate_av1, &min_bitrate_h264}, + field_trials.Lookup(kMinVideoBitrateExperiment)); if (min_video_bitrate) { if (min_bitrate_vp8 || min_bitrate_vp9 || min_bitrate_av1 || diff --git a/rtc_base/experiments/quality_scaler_settings_unittest.cc b/rtc_base/experiments/quality_scaler_settings_unittest.cc index 578fe97b03..06d0dab80a 100644 --- a/rtc_base/experiments/quality_scaler_settings_unittest.cc +++ b/rtc_base/experiments/quality_scaler_settings_unittest.cc @@ -17,7 +17,7 @@ namespace webrtc { namespace { TEST(QualityScalerSettingsTest, ValuesNotSetByDefault) { - webrtc::test::ScopedKeyValueConfig field_trials(""); + test::ScopedKeyValueConfig field_trials(""); const auto settings = QualityScalerSettings(field_trials); EXPECT_FALSE(settings.MinFrames()); EXPECT_FALSE(settings.InitialScaleFactor()); diff --git a/rtc_base/experiments/quality_scaling_experiment_unittest.cc b/rtc_base/experiments/quality_scaling_experiment_unittest.cc index 0c1450557a..c8dab44565 100644 --- a/rtc_base/experiments/quality_scaling_experiment_unittest.cc +++ b/rtc_base/experiments/quality_scaling_experiment_unittest.cc @@ -41,7 +41,7 @@ void ExpectEqualConfig(QualityScalingExperiment::Config a, #if !defined(WEBRTC_IOS) // TODO(bugs.webrtc.org/12401): investigate why QualityScaler kicks in on iOS. TEST(QualityScalingExperimentTest, DefaultEnabledWithoutFieldTrial) { - webrtc::test::ScopedKeyValueConfig field_trials(""); + test::ScopedKeyValueConfig field_trials(""); EXPECT_TRUE(QualityScalingExperiment::Enabled(field_trials)); } #else @@ -52,7 +52,7 @@ TEST(QualityScalingExperimentTest, DefaultDisabledWithoutFieldTrialIOS) { #endif TEST(QualityScalingExperimentTest, EnabledWithFieldTrial) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-QualityScaling/Enabled/"); EXPECT_TRUE(QualityScalingExperiment::Enabled(field_trials)); } @@ -60,7 +60,7 @@ TEST(QualityScalingExperimentTest, EnabledWithFieldTrial) { TEST(QualityScalingExperimentTest, ParseSettings) { const QualityScalingExperiment::Settings kExpected = {1, 2, 3, 4, 5, 6, 7, 8, 0.9f, 0.99f, 1}; - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-QualityScaling/Enabled-1,2,3,4,5,6,7,8,0.9,0.99,1/"); const auto settings = QualityScalingExperiment::ParseSettings(field_trials); EXPECT_TRUE(settings); @@ -70,7 +70,7 @@ TEST(QualityScalingExperimentTest, ParseSettings) { #if !defined(WEBRTC_IOS) // TODO(bugs.webrtc.org/12401): investigate why QualityScaler kicks in on iOS. TEST(QualityScalingExperimentTest, ParseSettingsUsesDefaultsWithoutFieldTrial) { - webrtc::test::ScopedKeyValueConfig field_trials(""); + test::ScopedKeyValueConfig field_trials(""); // Uses some default hard coded values. EXPECT_TRUE(QualityScalingExperiment::ParseSettings(field_trials)); } @@ -82,13 +82,13 @@ TEST(QualityScalingExperimentTest, ParseSettingsFailsWithoutFieldTrial) { #endif TEST(QualityScalingExperimentTest, ParseSettingsFailsWithInvalidFieldTrial) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-QualityScaling/Enabled-invalid/"); EXPECT_FALSE(QualityScalingExperiment::ParseSettings(field_trials)); } TEST(QualityScalingExperimentTest, GetConfig) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-QualityScaling/Enabled-1,2,3,4,5,6,7,8,0.9,0.99,0/"); const auto config = QualityScalingExperiment::GetConfig(field_trials); EXPECT_EQ(0.9f, config.alpha_high); @@ -97,7 +97,7 @@ TEST(QualityScalingExperimentTest, GetConfig) { } TEST(QualityScalingExperimentTest, GetsDefaultConfigForInvalidFieldTrial) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-QualityScaling/Enabled-invalid/"); const auto config = QualityScalingExperiment::GetConfig(field_trials); ExpectEqualConfig(config, QualityScalingExperiment::Config()); @@ -106,14 +106,14 @@ TEST(QualityScalingExperimentTest, GetsDefaultConfigForInvalidFieldTrial) { TEST(QualityScalingExperimentTest, GetsDefaultAlphaForInvalidValue) { QualityScalingExperiment::Config expected_config; expected_config.use_all_drop_reasons = true; - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-QualityScaling/Enabled-1,2,3,4,5,6,7,8,0.99,0.9,1/"); const auto config = QualityScalingExperiment::GetConfig(field_trials); ExpectEqualConfig(config, expected_config); } TEST(QualityScalingExperimentTest, GetVp8Thresholds) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-QualityScaling/Enabled-1,2,3,4,5,6,0,0,0.9,0.99,1/"); const auto thresholds = QualityScalingExperiment::GetQpThresholds(kVideoCodecVP8, field_trials); @@ -123,7 +123,7 @@ TEST(QualityScalingExperimentTest, GetVp8Thresholds) { } TEST(QualityScalingExperimentTest, GetThresholdsFailsForInvalidVp8Value) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-QualityScaling/Enabled-0,0,3,4,5,6,7,8,0.9,0.99,1/"); const auto thresholds = QualityScalingExperiment::GetQpThresholds(kVideoCodecVP8, field_trials); @@ -131,7 +131,7 @@ TEST(QualityScalingExperimentTest, GetThresholdsFailsForInvalidVp8Value) { } TEST(QualityScalingExperimentTest, GetVp9Thresholds) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-QualityScaling/Enabled-1,2,3,4,5,6,0,0,0.9,0.99,1/"); const auto thresholds = QualityScalingExperiment::GetQpThresholds(kVideoCodecVP9, field_trials); @@ -141,7 +141,7 @@ TEST(QualityScalingExperimentTest, GetVp9Thresholds) { } TEST(QualityScalingExperimentTest, GetThresholdsFailsForInvalidVp9Value) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-QualityScaling/Enabled-1,2,0,0,5,6,7,8,0.9,0.99,1/"); const auto thresholds = QualityScalingExperiment::GetQpThresholds(kVideoCodecVP9, field_trials); @@ -149,7 +149,7 @@ TEST(QualityScalingExperimentTest, GetThresholdsFailsForInvalidVp9Value) { } TEST(QualityScalingExperimentTest, GetH264Thresholds) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-QualityScaling/Enabled-1,2,3,4,5,6,0,0,0.9,0.99,1/"); const auto thresholds = QualityScalingExperiment::GetQpThresholds(kVideoCodecH264, field_trials); @@ -159,7 +159,7 @@ TEST(QualityScalingExperimentTest, GetH264Thresholds) { } TEST(QualityScalingExperimentTest, GetThresholdsFailsForInvalidH264Value) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-QualityScaling/Enabled-1,2,3,4,0,0,7,8,0.9,0.99,1/"); const auto thresholds = QualityScalingExperiment::GetQpThresholds(kVideoCodecH264, field_trials); @@ -167,7 +167,7 @@ TEST(QualityScalingExperimentTest, GetThresholdsFailsForInvalidH264Value) { } TEST(QualityScalingExperimentTest, GetGenericThresholds) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-QualityScaling/Enabled-1,2,3,4,0,0,7,8,0.9,0.99,1/"); const auto thresholds = QualityScalingExperiment::GetQpThresholds( kVideoCodecGeneric, field_trials); @@ -177,7 +177,7 @@ TEST(QualityScalingExperimentTest, GetGenericThresholds) { } TEST(QualityScalingExperimentTest, GetThresholdsFailsForInvalidGenericValue) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( "WebRTC-Video-QualityScaling/Enabled-1,2,3,4,5,6,0,0,0.9,0.99,1/"); const auto thresholds = QualityScalingExperiment::GetQpThresholds( kVideoCodecGeneric, field_trials); diff --git a/rtc_base/experiments/struct_parameters_parser.cc b/rtc_base/experiments/struct_parameters_parser.cc index bb46453a87..f999601afe 100644 --- a/rtc_base/experiments/struct_parameters_parser.cc +++ b/rtc_base/experiments/struct_parameters_parser.cc @@ -38,13 +38,13 @@ inline void StringEncode(std::string* target, unsigned val) { *target += absl::StrCat(val); } inline void StringEncode(std::string* target, DataRate val) { - *target += webrtc::ToString(val); + *target += ToString(val); } inline void StringEncode(std::string* target, DataSize val) { - *target += webrtc::ToString(val); + *target += ToString(val); } inline void StringEncode(std::string* target, TimeDelta val) { - *target += webrtc::ToString(val); + *target += ToString(val); } template diff --git a/test/call_config_utils.cc b/test/call_config_utils.cc index d93c902bbe..ff586821f7 100644 --- a/test/call_config_utils.cc +++ b/test/call_config_utils.cc @@ -20,7 +20,7 @@ namespace test { // back into a valid object. This will not initialize the decoders or the // renderer. VideoReceiveStreamInterface::Config ParseVideoReceiveStreamJsonConfig( - webrtc::Transport* transport, + Transport* transport, const Json::Value& json) { auto receive_config = VideoReceiveStreamInterface::Config(transport); for (const auto& decoder_json : json["decoders"]) { diff --git a/test/direct_transport.cc b/test/direct_transport.cc index 7d96b81e5d..5b8e53aa2a 100644 --- a/test/direct_transport.cc +++ b/test/direct_transport.cc @@ -96,10 +96,10 @@ bool DirectTransport::SendRtp(ArrayView data, const RtpHeaderExtensionMap* extensions = nullptr; MediaType media_type = demuxer_.GetMediaType(data.data(), data.size()); switch (demuxer_.GetMediaType(data.data(), data.size())) { - case webrtc::MediaType::AUDIO: + case MediaType::AUDIO: extensions = &audio_extensions_; break; - case webrtc::MediaType::VIDEO: + case MediaType::VIDEO: extensions = &video_extensions_; break; default: diff --git a/test/encoder_settings.cc b/test/encoder_settings.cc index d5009ca9ef..d8c9348c43 100644 --- a/test/encoder_settings.cc +++ b/test/encoder_settings.cc @@ -29,7 +29,7 @@ const int DefaultVideoStreamFactory::kDefaultMinBitratePerStream[] = { std::vector CreateVideoStreams( int width, int height, - const webrtc::VideoEncoderConfig& encoder_config) { + const VideoEncoderConfig& encoder_config) { RTC_DCHECK(encoder_config.number_of_streams <= DefaultVideoStreamFactory::kMaxNumberOfStreams); @@ -109,7 +109,7 @@ std::vector DefaultVideoStreamFactory::CreateEncoderStreams( const FieldTrialsView& /*field_trials*/, int frame_width, int frame_height, - const webrtc::VideoEncoderConfig& encoder_config) { + const VideoEncoderConfig& encoder_config) { return CreateVideoStreams(frame_width, frame_height, encoder_config); } diff --git a/test/fake_decoder.cc b/test/fake_decoder.cc index 36c6dd5c9a..d077ae7b5a 100644 --- a/test/fake_decoder.cc +++ b/test/fake_decoder.cc @@ -50,7 +50,7 @@ int32_t FakeDecoder::Decode(const EncodedImage& input, int64_t render_time_ms) { I420Buffer::SetBlack(buffer.get()); VideoFrame frame = VideoFrame::Builder() .set_video_frame_buffer(buffer) - .set_rotation(webrtc::kVideoRotation_0) + .set_rotation(kVideoRotation_0) .set_timestamp_ms(render_time_ms) .build(); frame.set_rtp_timestamp(input.RtpTimestamp()); diff --git a/test/fake_vp8_decoder.cc b/test/fake_vp8_decoder.cc index 3a4793986b..9471d11280 100644 --- a/test/fake_vp8_decoder.cc +++ b/test/fake_vp8_decoder.cc @@ -55,7 +55,7 @@ int32_t FakeVp8Decoder::Decode(const EncodedImage& input, VideoFrame frame = VideoFrame::Builder() .set_video_frame_buffer(I420Buffer::Create(width_, height_)) - .set_rotation(webrtc::kVideoRotation_0) + .set_rotation(kVideoRotation_0) .set_timestamp_ms(render_time_ms) .build(); frame.set_rtp_timestamp(input.RtpTimestamp()); diff --git a/test/field_trial.cc b/test/field_trial.cc index 3d6c6ac617..5267d7a457 100644 --- a/test/field_trial.cc +++ b/test/field_trial.cc @@ -21,18 +21,17 @@ namespace test { ScopedFieldTrials::ScopedFieldTrials(absl::string_view config) : current_field_trials_(config), - previous_field_trials_(webrtc::field_trial::GetFieldTrialString()) { - RTC_CHECK(webrtc::field_trial::FieldTrialsStringIsValid( - current_field_trials_.c_str())) + previous_field_trials_(field_trial::GetFieldTrialString()) { + RTC_CHECK( + field_trial::FieldTrialsStringIsValid(current_field_trials_.c_str())) << "Invalid field trials string: " << current_field_trials_; - webrtc::field_trial::InitFieldTrialsFromString(current_field_trials_.c_str()); + field_trial::InitFieldTrialsFromString(current_field_trials_.c_str()); } ScopedFieldTrials::~ScopedFieldTrials() { - RTC_CHECK( - webrtc::field_trial::FieldTrialsStringIsValid(previous_field_trials_)) + RTC_CHECK(field_trial::FieldTrialsStringIsValid(previous_field_trials_)) << "Invalid field trials string: " << previous_field_trials_; - webrtc::field_trial::InitFieldTrialsFromString(previous_field_trials_); + field_trial::InitFieldTrialsFromString(previous_field_trials_); } } // namespace test diff --git a/test/frame_generator.cc b/test/frame_generator.cc index 42f4617b26..aab3991c36 100644 --- a/test/frame_generator.cc +++ b/test/frame_generator.cc @@ -139,7 +139,7 @@ void SquareGenerator::Square::Draw( return; // Optionally draw on alpha plane if given. - const webrtc::I420ABufferInterface* yuva_buffer = frame_buffer->GetI420A(); + const I420ABufferInterface* yuva_buffer = frame_buffer->GetI420A(); for (int y = y_; y < y_ + length; ++y) { uint8_t* pos_y = (const_cast(yuva_buffer->DataA()) + x_ + y * yuva_buffer->StrideA()); diff --git a/test/frame_generator_capturer.cc b/test/frame_generator_capturer.cc index 0e2f2ee952..48051cd4eb 100644 --- a/test/frame_generator_capturer.cc +++ b/test/frame_generator_capturer.cc @@ -107,8 +107,7 @@ void FrameGeneratorCapturer::InsertFrame() { // Skip frames that are identical to the previous one but still send at // least one frame every second. if (number_of_frames_skipped_ < target_capture_fps_ - 1 && - webrtc::test::FrameBufsEqual(last_frame_captured_, - frame_data.buffer)) { + test::FrameBufsEqual(last_frame_captured_, frame_data.buffer)) { ++number_of_frames_skipped_; return; } diff --git a/test/frame_utils.cc b/test/frame_utils.cc index 1e2019bb7f..1a45c04a2a 100644 --- a/test/frame_utils.cc +++ b/test/frame_utils.cc @@ -35,7 +35,7 @@ bool EqualPlane(const uint8_t* data1, return true; } -bool FramesEqual(const webrtc::VideoFrame& f1, const webrtc::VideoFrame& f2) { +bool FramesEqual(const VideoFrame& f1, const VideoFrame& f2) { if (f1.rtp_timestamp() != f2.rtp_timestamp() || f1.ntp_time_ms() != f2.ntp_time_ms() || f1.render_time_ms() != f2.render_time_ms()) { @@ -44,8 +44,8 @@ bool FramesEqual(const webrtc::VideoFrame& f1, const webrtc::VideoFrame& f2) { return FrameBufsEqual(f1.video_frame_buffer(), f2.video_frame_buffer()); } -bool FrameBufsEqual(const scoped_refptr& f1, - const scoped_refptr& f2) { +bool FrameBufsEqual(const scoped_refptr& f1, + const scoped_refptr& f2) { if (f1 == f2) { return true; } @@ -59,8 +59,8 @@ bool FrameBufsEqual(const scoped_refptr& f1, return false; } - scoped_refptr f1_i420 = f1->ToI420(); - scoped_refptr f2_i420 = f2->ToI420(); + scoped_refptr f1_i420 = f1->ToI420(); + scoped_refptr f2_i420 = f2->ToI420(); return EqualPlane(f1_i420->DataY(), f2_i420->DataY(), f1_i420->StrideY(), f2_i420->StrideY(), f1_i420->width(), f1_i420->height()) && EqualPlane(f1_i420->DataU(), f2_i420->DataU(), f1_i420->StrideU(), diff --git a/test/gl/gl_renderer.cc b/test/gl/gl_renderer.cc index 10162ee056..21a11953f7 100644 --- a/test/gl/gl_renderer.cc +++ b/test/gl/gl_renderer.cc @@ -70,7 +70,7 @@ void GlRenderer::ResizeVideo(size_t width, size_t height) { GL_UNSIGNED_INT_8_8_8_8, static_cast(buffer_)); } -void GlRenderer::OnFrame(const webrtc::VideoFrame& frame) { +void GlRenderer::OnFrame(const VideoFrame& frame) { RTC_DCHECK(is_init_); if (static_cast(frame.width()) != width_ || @@ -78,7 +78,7 @@ void GlRenderer::OnFrame(const webrtc::VideoFrame& frame) { ResizeVideo(frame.width(), frame.height()); } - webrtc::ConvertFromI420(frame, VideoType::kBGRA, 0, buffer_); + ConvertFromI420(frame, VideoType::kBGRA, 0, buffer_); glEnable(GL_TEXTURE_2D); glBindTexture(GL_TEXTURE_2D, texture_); diff --git a/test/linux/glx_renderer.cc b/test/linux/glx_renderer.cc index 509a6c286e..43f2cdafd8 100644 --- a/test/linux/glx_renderer.cc +++ b/test/linux/glx_renderer.cc @@ -141,7 +141,7 @@ void GlxRenderer::Resize(size_t width, size_t height) { XConfigureWindow(display_, window_, CWWidth | CWHeight, &wc); } -void GlxRenderer::OnFrame(const webrtc::VideoFrame& frame) { +void GlxRenderer::OnFrame(const VideoFrame& frame) { if (static_cast(frame.width()) != width_ || static_cast(frame.height()) != height_) { Resize(static_cast(frame.width()), diff --git a/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.cc b/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.cc index 7c92cd228f..d282bee308 100644 --- a/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.cc +++ b/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.cc @@ -33,8 +33,8 @@ namespace webrtc { namespace webrtc_pc_e2e { -using ::webrtc::test::ImprovementDirection; -using ::webrtc::test::Unit; +using test::ImprovementDirection; +using test::Unit; DefaultAudioQualityAnalyzer::DefaultAudioQualityAnalyzer( test::MetricsLogger* const metrics_logger) diff --git a/test/pc/e2e/analyzer/video/analyzing_video_sink.cc b/test/pc/e2e/analyzer/video/analyzing_video_sink.cc index 9cba75cfd2..8b04799d29 100644 --- a/test/pc/e2e/analyzer/video/analyzing_video_sink.cc +++ b/test/pc/e2e/analyzer/video/analyzing_video_sink.cc @@ -120,7 +120,7 @@ void AnalyzingVideoSink::OnFrame(const VideoFrame& frame) { } } -void AnalyzingVideoSink::LogMetrics(webrtc::test::MetricsLogger& metrics_logger, +void AnalyzingVideoSink::LogMetrics(test::MetricsLogger& metrics_logger, absl::string_view test_case_name) const { if (report_infra_stats_) { MutexLock lock(&mutex_); diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc index 42f45abc7a..54984e5d0a 100644 --- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc +++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc @@ -51,9 +51,9 @@ namespace webrtc { namespace { -using ::webrtc::test::ImprovementDirection; -using ::webrtc::test::Unit; -using ::webrtc::webrtc_pc_e2e::MetricMetadataKey; +using test::ImprovementDirection; +using test::Unit; +using webrtc_pc_e2e::MetricMetadataKey; constexpr int kBitsInByte = 8; constexpr absl::string_view kSkipRenderedFrameReasonProcessed = "processed"; @@ -140,7 +140,7 @@ SamplesStatsCounter::StatsSample StatsSample(double value, } // namespace DefaultVideoQualityAnalyzer::DefaultVideoQualityAnalyzer( - webrtc::Clock* clock, + Clock* clock, test::MetricsLogger* metrics_logger, DefaultVideoQualityAnalyzerOptions options) : options_(options), @@ -175,7 +175,7 @@ void DefaultVideoQualityAnalyzer::Start(std::string test_case_name, uint16_t DefaultVideoQualityAnalyzer::OnFrameCaptured( absl::string_view peer_name, const std::string& stream_label, - const webrtc::VideoFrame& frame) { + const VideoFrame& frame) { // `next_frame_id` is atomic, so we needn't lock here. Timestamp captured_time = Now(); Timestamp start_time = Timestamp::MinusInfinity(); @@ -288,9 +288,8 @@ uint16_t DefaultVideoQualityAnalyzer::OnFrameCaptured( return frame_id; } -void DefaultVideoQualityAnalyzer::OnFramePreEncode( - absl::string_view peer_name, - const webrtc::VideoFrame& frame) { +void DefaultVideoQualityAnalyzer::OnFramePreEncode(absl::string_view peer_name, + const VideoFrame& frame) { Timestamp processing_started = Now(); MutexLock lock(&mutex_); RTC_CHECK_EQ(state_, State::kActive) @@ -323,7 +322,7 @@ void DefaultVideoQualityAnalyzer::OnFramePreEncode( void DefaultVideoQualityAnalyzer::OnFrameEncoded( absl::string_view peer_name, uint16_t frame_id, - const webrtc::EncodedImage& encoded_image, + const EncodedImage& encoded_image, const EncoderStats& stats, bool discarded) { if (discarded) @@ -392,14 +391,14 @@ void DefaultVideoQualityAnalyzer::OnFrameEncoded( void DefaultVideoQualityAnalyzer::OnFrameDropped( absl::string_view peer_name, - webrtc::EncodedImageCallback::DropReason reason) { + EncodedImageCallback::DropReason reason) { // Here we do nothing, because we will see this drop on renderer side. } void DefaultVideoQualityAnalyzer::OnFramePreDecode( absl::string_view peer_name, uint16_t frame_id, - const webrtc::EncodedImage& input_image) { + const EncodedImage& input_image) { Timestamp processing_started = Now(); MutexLock lock(&mutex_); RTC_CHECK_EQ(state_, State::kActive) @@ -449,10 +448,9 @@ void DefaultVideoQualityAnalyzer::OnFramePreDecode( } } -void DefaultVideoQualityAnalyzer::OnFrameDecoded( - absl::string_view peer_name, - const webrtc::VideoFrame& frame, - const DecoderStats& stats) { +void DefaultVideoQualityAnalyzer::OnFrameDecoded(absl::string_view peer_name, + const VideoFrame& frame, + const DecoderStats& stats) { Timestamp processing_started = Now(); MutexLock lock(&mutex_); RTC_CHECK_EQ(state_, State::kActive) @@ -496,9 +494,8 @@ void DefaultVideoQualityAnalyzer::OnFrameDecoded( } } -void DefaultVideoQualityAnalyzer::OnFrameRendered( - absl::string_view peer_name, - const webrtc::VideoFrame& frame) { +void DefaultVideoQualityAnalyzer::OnFrameRendered(absl::string_view peer_name, + const VideoFrame& frame) { Timestamp processing_started = Now(); MutexLock lock(&mutex_); RTC_CHECK_EQ(state_, State::kActive) @@ -599,10 +596,9 @@ void DefaultVideoQualityAnalyzer::OnFrameRendered( } } -void DefaultVideoQualityAnalyzer::OnEncoderError( - absl::string_view peer_name, - const webrtc::VideoFrame& frame, - int32_t error_code) { +void DefaultVideoQualityAnalyzer::OnEncoderError(absl::string_view peer_name, + const VideoFrame& frame, + int32_t error_code) { RTC_LOG(LS_ERROR) << "Encoder error for frame.id=" << frame.id() << ", code=" << error_code; } diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frame_in_flight.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frame_in_flight.cc index fb96ee6ee5..6dbdfdc58e 100644 --- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frame_in_flight.cc +++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frame_in_flight.cc @@ -78,7 +78,7 @@ bool FrameInFlight::HaveAllPeersReceived() const { } void FrameInFlight::OnFrameEncoded( - webrtc::Timestamp time, + Timestamp time, std::optional time_between_encoded_frames, VideoFrameType frame_type, DataSize encoded_image_size, @@ -113,8 +113,8 @@ void FrameInFlight::OnFrameEncoded( } void FrameInFlight::OnFramePreDecode(size_t peer, - webrtc::Timestamp received_time, - webrtc::Timestamp decode_start_time, + Timestamp received_time, + Timestamp decode_start_time, VideoFrameType frame_type, DataSize encoded_image_size) { receiver_stats_[peer].received_time = received_time; @@ -132,7 +132,7 @@ bool FrameInFlight::HasReceivedTime(size_t peer) const { } void FrameInFlight::OnFrameDecoded(size_t peer, - webrtc::Timestamp time, + Timestamp time, int width, int height, const StreamCodecInfo& used_decoder, @@ -158,7 +158,7 @@ bool FrameInFlight::HasDecodeEndTime(size_t peer) const { return it->second.decode_end_time.IsFinite(); } -void FrameInFlight::OnFrameRendered(size_t peer, webrtc::Timestamp time) { +void FrameInFlight::OnFrameRendered(size_t peer, Timestamp time) { receiver_stats_[peer].rendered_time = time; } diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.cc index 4a9906a401..de277e5983 100644 --- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.cc +++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.cc @@ -33,7 +33,7 @@ namespace webrtc { namespace { -using ::webrtc::webrtc_pc_e2e::SampleMetadataKey; +using webrtc_pc_e2e::SampleMetadataKey; constexpr TimeDelta kFreezeThreshold = TimeDelta::Millis(150); constexpr int kMaxActiveComparisons = 10; diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator_test.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator_test.cc index 571c3c602f..9540840717 100644 --- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator_test.cc +++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator_test.cc @@ -34,7 +34,7 @@ using ::testing::IsEmpty; using ::testing::Pair; using ::testing::SizeIs; -using StatsSample = ::webrtc::SamplesStatsCounter::StatsSample; +using StatsSample = SamplesStatsCounter::StatsSample; DefaultVideoQualityAnalyzerOptions AnalyzerOptionsForTest() { DefaultVideoQualityAnalyzerOptions options; diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_metric_names_test.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_metric_names_test.cc index f3f9caac0a..6bc493bf79 100644 --- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_metric_names_test.cc +++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_metric_names_test.cc @@ -43,10 +43,10 @@ using ::testing::Contains; using ::testing::SizeIs; using ::testing::UnorderedElementsAre; -using ::webrtc::test::DefaultMetricsLogger; -using ::webrtc::test::ImprovementDirection; -using ::webrtc::test::Metric; -using ::webrtc::test::Unit; +using test::DefaultMetricsLogger; +using test::ImprovementDirection; +using test::Metric; +using test::Unit; constexpr int kAnalyzerMaxThreadsCount = 1; constexpr int kFrameWidth = 320; diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc index 4aa80bf224..94dad5782e 100644 --- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc +++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc @@ -44,7 +44,7 @@ using ::testing::Test; using ::testing::TestWithParam; using ::testing::ValuesIn; -using StatsSample = ::webrtc::SamplesStatsCounter::StatsSample; +using StatsSample = SamplesStatsCounter::StatsSample; constexpr int kAnalyzerMaxThreadsCount = 1; constexpr TimeDelta kMaxFramesInFlightStorageDuration = TimeDelta::Seconds(3); diff --git a/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc b/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc index f55e886c46..3946b9c0bd 100644 --- a/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc +++ b/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc @@ -25,7 +25,7 @@ void ExampleVideoQualityAnalyzer::Start(std::string test_case_name, uint16_t ExampleVideoQualityAnalyzer::OnFrameCaptured( absl::string_view peer_name, const std::string& stream_label, - const webrtc::VideoFrame& frame) { + const VideoFrame& frame) { MutexLock lock(&lock_); uint16_t frame_id = next_frame_id_++; if (frame_id == VideoFrame::kNotSetId) { @@ -50,9 +50,8 @@ uint16_t ExampleVideoQualityAnalyzer::OnFrameCaptured( return frame_id; } -void ExampleVideoQualityAnalyzer::OnFramePreEncode( - absl::string_view peer_name, - const webrtc::VideoFrame& frame) { +void ExampleVideoQualityAnalyzer::OnFramePreEncode(absl::string_view peer_name, + const VideoFrame& frame) { MutexLock lock(&lock_); ++frames_pre_encoded_; } @@ -60,7 +59,7 @@ void ExampleVideoQualityAnalyzer::OnFramePreEncode( void ExampleVideoQualityAnalyzer::OnFrameEncoded( absl::string_view peer_name, uint16_t frame_id, - const webrtc::EncodedImage& encoded_image, + const EncodedImage& encoded_image, const EncoderStats& stats, bool discarded) { MutexLock lock(&lock_); @@ -69,7 +68,7 @@ void ExampleVideoQualityAnalyzer::OnFrameEncoded( void ExampleVideoQualityAnalyzer::OnFrameDropped( absl::string_view peer_name, - webrtc::EncodedImageCallback::DropReason reason) { + EncodedImageCallback::DropReason reason) { RTC_LOG(LS_INFO) << "Frame dropped by encoder"; MutexLock lock(&lock_); ++frames_dropped_; @@ -78,31 +77,28 @@ void ExampleVideoQualityAnalyzer::OnFrameDropped( void ExampleVideoQualityAnalyzer::OnFramePreDecode( absl::string_view peer_name, uint16_t frame_id, - const webrtc::EncodedImage& encoded_image) { + const EncodedImage& encoded_image) { MutexLock lock(&lock_); ++frames_received_; } -void ExampleVideoQualityAnalyzer::OnFrameDecoded( - absl::string_view peer_name, - const webrtc::VideoFrame& frame, - const DecoderStats& stats) { +void ExampleVideoQualityAnalyzer::OnFrameDecoded(absl::string_view peer_name, + const VideoFrame& frame, + const DecoderStats& stats) { MutexLock lock(&lock_); ++frames_decoded_; } -void ExampleVideoQualityAnalyzer::OnFrameRendered( - absl::string_view peer_name, - const webrtc::VideoFrame& frame) { +void ExampleVideoQualityAnalyzer::OnFrameRendered(absl::string_view peer_name, + const VideoFrame& frame) { MutexLock lock(&lock_); frames_in_flight_.erase(frame.id()); ++frames_rendered_; } -void ExampleVideoQualityAnalyzer::OnEncoderError( - absl::string_view peer_name, - const webrtc::VideoFrame& frame, - int32_t error_code) { +void ExampleVideoQualityAnalyzer::OnEncoderError(absl::string_view peer_name, + const VideoFrame& frame, + int32_t error_code) { RTC_LOG(LS_ERROR) << "Failed to encode frame " << frame.id() << ". Code: " << error_code; } diff --git a/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.cc b/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.cc index 1b51402c48..f990adf472 100644 --- a/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.cc +++ b/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.cc @@ -189,19 +189,18 @@ int32_t QualityAnalyzingVideoDecoder::DecoderCallback::IrrelevantSimulcastStreamDecoded( uint16_t frame_id, uint32_t timestamp_ms) { - webrtc::VideoFrame dummy_frame = - webrtc::VideoFrame::Builder() - .set_video_frame_buffer(GetDummyFrameBuffer()) - .set_rtp_timestamp(timestamp_ms) - .set_id(frame_id) - .build(); + VideoFrame dummy_frame = VideoFrame::Builder() + .set_video_frame_buffer(GetDummyFrameBuffer()) + .set_rtp_timestamp(timestamp_ms) + .set_id(frame_id) + .build(); MutexLock lock(&callback_mutex_); RTC_DCHECK(delegate_callback_); delegate_callback_->Decoded(dummy_frame, std::nullopt, std::nullopt); return WEBRTC_VIDEO_CODEC_OK; } -scoped_refptr +scoped_refptr QualityAnalyzingVideoDecoder::DecoderCallback::GetDummyFrameBuffer() { if (!dummy_frame_buffer_) { dummy_frame_buffer_ = CreateDummyFrameBuffer(); diff --git a/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.cc b/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.cc index 65e2f12594..c11a0e942c 100644 --- a/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.cc +++ b/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.cc @@ -27,7 +27,7 @@ namespace webrtc_pc_e2e { namespace { using EmulatedSFUConfigMap = - ::webrtc::webrtc_pc_e2e::QualityAnalyzingVideoEncoder::EmulatedSFUConfigMap; + webrtc_pc_e2e::QualityAnalyzingVideoEncoder::EmulatedSFUConfigMap; constexpr size_t kMaxFrameInPipelineCount = 1000; constexpr double kNoMultiplier = 1.0; diff --git a/test/pc/e2e/analyzer/video/simulcast_dummy_buffer_helper.cc b/test/pc/e2e/analyzer/video/simulcast_dummy_buffer_helper.cc index 955406ab83..e6925466e6 100644 --- a/test/pc/e2e/analyzer/video/simulcast_dummy_buffer_helper.cc +++ b/test/pc/e2e/analyzer/video/simulcast_dummy_buffer_helper.cc @@ -22,9 +22,9 @@ constexpr char kIrrelatedSimulcastStreamFrameData[] = "Dummy!"; } // namespace -scoped_refptr CreateDummyFrameBuffer() { +scoped_refptr CreateDummyFrameBuffer() { // Use i420 buffer here as default one and supported by all codecs. - scoped_refptr buffer = webrtc::I420Buffer::Create(2, 2); + scoped_refptr buffer = I420Buffer::Create(2, 2); memcpy(buffer->MutableDataY(), kIrrelatedSimulcastStreamFrameData, 2); memcpy(buffer->MutableDataY() + buffer->StrideY(), kIrrelatedSimulcastStreamFrameData + 2, 2); @@ -33,11 +33,11 @@ scoped_refptr CreateDummyFrameBuffer() { return buffer; } -bool IsDummyFrame(const webrtc::VideoFrame& video_frame) { +bool IsDummyFrame(const VideoFrame& video_frame) { if (video_frame.width() != 2 || video_frame.height() != 2) { return false; } - scoped_refptr buffer = + scoped_refptr buffer = video_frame.video_frame_buffer()->ToI420(); if (memcmp(buffer->DataY(), kIrrelatedSimulcastStreamFrameData, 2) != 0) { return false; diff --git a/test/pc/e2e/analyzer/video/video_dumping.cc b/test/pc/e2e/analyzer/video/video_dumping.cc index 4fec0a8f9e..6acacec106 100644 --- a/test/pc/e2e/analyzer/video/video_dumping.cc +++ b/test/pc/e2e/analyzer/video/video_dumping.cc @@ -70,7 +70,7 @@ class BroadcastingFrameWriter final : public test::VideoFrameWriter { : delegates_(std::move(delegates)) {} ~BroadcastingFrameWriter() override { Close(); } - bool WriteFrame(const webrtc::VideoFrame& frame) override { + bool WriteFrame(const VideoFrame& frame) override { for (auto& delegate : delegates_) { if (!delegate->WriteFrame(frame)) { return false; diff --git a/test/pc/e2e/analyzer/video/video_dumping_test.cc b/test/pc/e2e/analyzer/video/video_dumping_test.cc index 2b96fda024..3688fe5b8a 100644 --- a/test/pc/e2e/analyzer/video/video_dumping_test.cc +++ b/test/pc/e2e/analyzer/video/video_dumping_test.cc @@ -63,11 +63,11 @@ std::vector AsVector(const uint8_t* data, size_t size) { return out; } -void AssertFramesEqual(scoped_refptr actual, +void AssertFramesEqual(scoped_refptr actual, scoped_refptr expected) { ASSERT_THAT(actual->width(), Eq(expected->width())); ASSERT_THAT(actual->height(), Eq(expected->height())); - scoped_refptr expected_i420 = expected->ToI420(); + scoped_refptr expected_i420 = expected->ToI420(); int height = actual->height(); @@ -102,10 +102,10 @@ class VideoDumpingTest : public Test { ~VideoDumpingTest() override = default; void SetUp() override { - video_filename_ = webrtc::test::TempFilename(webrtc::test::OutputPath(), - "video_dumping_test"); - ids_filename_ = webrtc::test::TempFilename(webrtc::test::OutputPath(), - "video_dumping_test"); + video_filename_ = + test::TempFilename(test::OutputPath(), "video_dumping_test"); + ids_filename_ = + test::TempFilename(test::OutputPath(), "video_dumping_test"); } void TearDown() override { diff --git a/test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.cc b/test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.cc index 643dc946fd..c175859bfa 100644 --- a/test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.cc +++ b/test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.cc @@ -37,9 +37,9 @@ namespace webrtc { namespace webrtc_pc_e2e { namespace { -using webrtc::webrtc_pc_e2e::VideoConfig; +using webrtc_pc_e2e::VideoConfig; using EmulatedSFUConfigMap = - ::webrtc::webrtc_pc_e2e::QualityAnalyzingVideoEncoder::EmulatedSFUConfigMap; + webrtc_pc_e2e::QualityAnalyzingVideoEncoder::EmulatedSFUConfigMap; class AnalyzingFramePreprocessor : public test::TestVideoCapturer::FramePreprocessor { diff --git a/test/pc/e2e/analyzer/video/video_quality_metrics_reporter.cc b/test/pc/e2e/analyzer/video/video_quality_metrics_reporter.cc index d717cbb8ce..11e586af9c 100644 --- a/test/pc/e2e/analyzer/video/video_quality_metrics_reporter.cc +++ b/test/pc/e2e/analyzer/video/video_quality_metrics_reporter.cc @@ -39,8 +39,8 @@ namespace webrtc { namespace webrtc_pc_e2e { namespace { -using ::webrtc::test::ImprovementDirection; -using ::webrtc::test::Unit; +using test::ImprovementDirection; +using test::Unit; SamplesStatsCounter BytesPerSecondToKbps(const SamplesStatsCounter& counter) { return counter * 0.008; diff --git a/test/pc/e2e/cross_media_metrics_reporter.cc b/test/pc/e2e/cross_media_metrics_reporter.cc index 3315ec6b84..9edc91a0a3 100644 --- a/test/pc/e2e/cross_media_metrics_reporter.cc +++ b/test/pc/e2e/cross_media_metrics_reporter.cc @@ -31,8 +31,8 @@ namespace webrtc { namespace webrtc_pc_e2e { -using ::webrtc::test::ImprovementDirection; -using ::webrtc::test::Unit; +using test::ImprovementDirection; +using test::Unit; CrossMediaMetricsReporter::CrossMediaMetricsReporter( test::MetricsLogger* metrics_logger) @@ -124,7 +124,7 @@ void CrossMediaMetricsReporter::StopAndReportResults() { "audio_ahead_ms", GetTestCaseName(pair.second.audio_stream_info.stream_label, sync_group), pair.second.audio_ahead_ms, Unit::kMilliseconds, - webrtc::test::ImprovementDirection::kSmallerIsBetter, + test::ImprovementDirection::kSmallerIsBetter, std::move(audio_metric_metadata)); std::map video_metric_metadata{ @@ -139,7 +139,7 @@ void CrossMediaMetricsReporter::StopAndReportResults() { "video_ahead_ms", GetTestCaseName(pair.second.video_stream_info.stream_label, sync_group), pair.second.video_ahead_ms, Unit::kMilliseconds, - webrtc::test::ImprovementDirection::kSmallerIsBetter, + test::ImprovementDirection::kSmallerIsBetter, std::move(video_metric_metadata)); } } diff --git a/test/pc/e2e/media/media_helper.cc b/test/pc/e2e/media/media_helper.cc index 0b4de88e9f..ad83e8d091 100644 --- a/test/pc/e2e/media/media_helper.cc +++ b/test/pc/e2e/media/media_helper.cc @@ -39,7 +39,7 @@ void MediaHelper::MaybeAddAudio(TestPeer* peer) { return; } const AudioConfig& audio_config = peer->params().audio_config.value(); - scoped_refptr source = + scoped_refptr source = peer->pc_factory()->CreateAudioSource(audio_config.audio_options); scoped_refptr track = peer->pc_factory()->CreateAudioTrack(*audio_config.stream_label, diff --git a/test/pc/e2e/network_quality_metrics_reporter.cc b/test/pc/e2e/network_quality_metrics_reporter.cc index bccc15ee57..3cb820eb62 100644 --- a/test/pc/e2e/network_quality_metrics_reporter.cc +++ b/test/pc/e2e/network_quality_metrics_reporter.cc @@ -22,8 +22,8 @@ namespace webrtc { namespace webrtc_pc_e2e { namespace { -using ::webrtc::test::ImprovementDirection; -using ::webrtc::test::Unit; +using test::ImprovementDirection; +using test::Unit; constexpr TimeDelta kStatsWaitTimeout = TimeDelta::Seconds(1); diff --git a/test/pc/e2e/peer_connection_quality_test.cc b/test/pc/e2e/peer_connection_quality_test.cc index ce177d4e4d..764a1bce0f 100644 --- a/test/pc/e2e/peer_connection_quality_test.cc +++ b/test/pc/e2e/peer_connection_quality_test.cc @@ -85,8 +85,8 @@ namespace webrtc { namespace webrtc_pc_e2e { namespace { -using ::webrtc::test::ImprovementDirection; -using ::webrtc::test::Unit; +using test::ImprovementDirection; +using test::Unit; constexpr TimeDelta kDefaultTimeout = TimeDelta::Seconds(10); constexpr char kSignalThreadName[] = "signaling_thread"; @@ -116,7 +116,7 @@ class FixturePeerConnectionObserver : public MockPeerConnectionObserver { // of reconnect this callback can be called again, so it should be tolerant // to such behavior. FixturePeerConnectionObserver( - std::function)> + std::function)> on_track_callback, std::function on_connected_callback) : on_track_callback_(std::move(on_track_callback)), @@ -136,7 +136,7 @@ class FixturePeerConnectionObserver : public MockPeerConnectionObserver { } private: - std::function)> + std::function)> on_track_callback_; std::function on_connected_callback_; }; @@ -235,7 +235,7 @@ PeerConnectionE2EQualityTest::PeerHandle* PeerConnectionE2EQualityTest::AddPeer( } void PeerConnectionE2EQualityTest::Run(RunParams run_params) { - webrtc::webrtc_pc_e2e::PeerParamsPreprocessor params_preprocessor; + webrtc_pc_e2e::PeerParamsPreprocessor params_preprocessor; for (auto& peer_configuration : peer_configurations_) { params_preprocessor.SetDefaultValuesForMissingParams(*peer_configuration); params_preprocessor.ValidateParams(*peer_configuration); @@ -280,9 +280,9 @@ void PeerConnectionE2EQualityTest::Run(RunParams run_params) { time_controller_.GetClock()); // Create a `task_queue_`. - task_queue_ = std::make_unique( + task_queue_ = std::make_unique( time_controller_.GetTaskQueueFactory()->CreateTaskQueue( - "pc_e2e_quality_test", webrtc::TaskQueueFactory::Priority::NORMAL)); + "pc_e2e_quality_test", TaskQueueFactory::Priority::NORMAL)); // Create call participants: Alice and Bob. // Audio streams are intercepted in AudioDeviceModule, so if it is required to @@ -358,16 +358,16 @@ void PeerConnectionE2EQualityTest::Run(RunParams run_params) { // Start RTCEventLog recording if requested. if (alice_->params().rtc_event_log_path) { - auto alice_rtc_event_log = std::make_unique( + auto alice_rtc_event_log = std::make_unique( alice_->params().rtc_event_log_path.value()); alice_->pc()->StartRtcEventLog(std::move(alice_rtc_event_log), - webrtc::RtcEventLog::kImmediateOutput); + RtcEventLog::kImmediateOutput); } if (bob_->params().rtc_event_log_path) { - auto bob_rtc_event_log = std::make_unique( + auto bob_rtc_event_log = std::make_unique( bob_->params().rtc_event_log_path.value()); bob_->pc()->StartRtcEventLog(std::move(bob_rtc_event_log), - webrtc::RtcEventLog::kImmediateOutput); + RtcEventLog::kImmediateOutput); } // Setup alive logging. It is done to prevent test infra to think that test is @@ -521,8 +521,7 @@ void PeerConnectionE2EQualityTest::SetupCallOnSignalingThread( // multiple audio streams, then we need transceiver for each Bob's audio // stream. RTCErrorOr> result = - alice_->AddTransceiver(webrtc::MediaType::AUDIO, - receive_only_transceiver_init); + alice_->AddTransceiver(MediaType::AUDIO, receive_only_transceiver_init); RTC_CHECK(result.ok()); alice_transceivers_counter++; } @@ -561,7 +560,7 @@ void PeerConnectionE2EQualityTest::SetupCallOnSignalingThread( alice_video_transceivers_non_simulcast_counter++; } RTCErrorOr> result = - alice_->AddTransceiver(webrtc::MediaType::VIDEO, transceiver_params); + alice_->AddTransceiver(MediaType::VIDEO, transceiver_params); RTC_CHECK(result.ok()); alice_transceivers_counter++; @@ -572,8 +571,7 @@ void PeerConnectionE2EQualityTest::SetupCallOnSignalingThread( for (size_t i = alice_video_transceivers_non_simulcast_counter; i < bob_->configurable_params().video_configs.size(); ++i) { RTCErrorOr> result = - alice_->AddTransceiver(webrtc::MediaType::VIDEO, - receive_only_transceiver_init); + alice_->AddTransceiver(MediaType::VIDEO, receive_only_transceiver_init); RTC_CHECK(result.ok()); alice_transceivers_counter++; } @@ -598,19 +596,19 @@ void PeerConnectionE2EQualityTest::SetPeerCodecPreferences(TestPeer* peer) { peer->params().video_codecs, true, peer->params().use_ulp_fec, peer->params().use_flex_fec, peer->pc_factory() - ->GetRtpReceiverCapabilities(webrtc::MediaType::VIDEO) + ->GetRtpReceiverCapabilities(MediaType::VIDEO) .codecs); std::vector without_rtx_video_capabilities = FilterVideoCodecCapabilities( peer->params().video_codecs, false, peer->params().use_ulp_fec, peer->params().use_flex_fec, peer->pc_factory() - ->GetRtpReceiverCapabilities(webrtc::MediaType::VIDEO) + ->GetRtpReceiverCapabilities(MediaType::VIDEO) .codecs); // Set codecs for transceivers for (auto transceiver : peer->pc()->GetTransceivers()) { - if (transceiver->media_type() == webrtc::MediaType::VIDEO) { + if (transceiver->media_type() == MediaType::VIDEO) { if (transceiver->sender()->init_send_encodings().size() > 1) { // If transceiver's sender has more then 1 send encodings, it means it // has multiple simulcast streams, so we need disable RTX on it. diff --git a/test/pc/e2e/peer_connection_quality_test_metric_names_test.cc b/test/pc/e2e/peer_connection_quality_test_metric_names_test.cc index a3eea5d825..757ce74bc8 100644 --- a/test/pc/e2e/peer_connection_quality_test_metric_names_test.cc +++ b/test/pc/e2e/peer_connection_quality_test_metric_names_test.cc @@ -43,11 +43,11 @@ namespace { using ::testing::IsSupersetOf; using ::testing::UnorderedElementsAre; -using ::webrtc::test::DefaultMetricsLogger; -using ::webrtc::test::ImprovementDirection; -using ::webrtc::test::Metric; -using ::webrtc::test::Unit; -using ::webrtc::webrtc_pc_e2e::PeerConfigurer; +using test::DefaultMetricsLogger; +using test::ImprovementDirection; +using test::Metric; +using test::Unit; +using webrtc_pc_e2e::PeerConfigurer; std::string GetExpectedTestCaseName(const std::string& stream_label) { if (!absl::GetFlag(FLAGS_isolated_script_test_perf_output).empty()) { diff --git a/test/pc/e2e/peer_connection_quality_test_test.cc b/test/pc/e2e/peer_connection_quality_test_test.cc index 35e299bf37..a945b0491f 100644 --- a/test/pc/e2e/peer_connection_quality_test_test.cc +++ b/test/pc/e2e/peer_connection_quality_test_test.cc @@ -36,7 +36,7 @@ namespace { using ::testing::Eq; using ::testing::Test; -using ::webrtc::webrtc_pc_e2e::PeerConfigurer; +using webrtc_pc_e2e::PeerConfigurer; // Remove files and directories in a directory non-recursively. void CleanDir(absl::string_view dir, size_t expected_output_files_count) { diff --git a/test/pc/e2e/sdp/sdp_changer.cc b/test/pc/e2e/sdp/sdp_changer.cc index 6c74740cac..6aa774bf69 100644 --- a/test/pc/e2e/sdp/sdp_changer.cc +++ b/test/pc/e2e/sdp/sdp_changer.cc @@ -133,7 +133,7 @@ void SignalingInterceptor::FillSimulcastContext( SessionDescriptionInterface* offer) { for (auto& content : offer->description()->contents()) { MediaContentDescription* media_desc = content.media_description(); - if (media_desc->type() != webrtc::MediaType::VIDEO) { + if (media_desc->type() != MediaType::VIDEO) { continue; } if (media_desc->HasSimulcast()) { @@ -189,7 +189,7 @@ LocalAndRemoteSdp SignalingInterceptor::PatchOffer( for (auto& content : offer->description()->contents()) { context_.mids_order.push_back(content.mid()); MediaContentDescription* media_desc = content.media_description(); - if (media_desc->type() != webrtc::MediaType::VIDEO) { + if (media_desc->type() != MediaType::VIDEO) { continue; } if (content.media_description()->streams().empty()) { @@ -249,7 +249,7 @@ LocalAndRemoteSdp SignalingInterceptor::PatchVp8Offer( // Swap mid and rid extensions, so remote peer will understand rid as mid. // Also remove rid extension. - std::vector extensions = + std::vector extensions = prototype_media_desc->rtp_header_extensions(); for (auto ext_it = extensions.begin(); ext_it != extensions.end();) { if (ext_it->uri == RtpExtension::kRidUri) { @@ -333,7 +333,7 @@ LocalAndRemoteSdp SignalingInterceptor::PatchVp9Offer( } for (auto& content : offer->description()->contents()) { - if (content.media_description()->type() != webrtc::MediaType::VIDEO) { + if (content.media_description()->type() != MediaType::VIDEO) { // We are interested in only video tracks continue; } @@ -381,7 +381,7 @@ LocalAndRemoteSdp SignalingInterceptor::PatchAnswer( const VideoCodecConfig& first_codec) { for (auto& content : answer->description()->contents()) { MediaContentDescription* media_desc = content.media_description(); - if (media_desc->type() != webrtc::MediaType::VIDEO) { + if (media_desc->type() != MediaType::VIDEO) { continue; } if (content.media_description()->direction() != @@ -433,8 +433,7 @@ LocalAndRemoteSdp SignalingInterceptor::PatchVp8Answer( // Patch `media_desc` to make it simulcast answer description. // Restore mid/rid rtp header extensions - std::vector extensions = - media_desc->rtp_header_extensions(); + std::vector extensions = media_desc->rtp_header_extensions(); // First remove existing rid/mid header extensions. extensions.erase(std::remove_if(extensions.begin(), extensions.end(), [](const webrtc::RtpExtension& e) { diff --git a/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc b/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc index 333d6818bf..32f6eb4d59 100644 --- a/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc +++ b/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc @@ -49,8 +49,8 @@ namespace webrtc { namespace webrtc_pc_e2e { namespace { -using ::webrtc::test::ImprovementDirection; -using ::webrtc::test::Unit; +using test::ImprovementDirection; +using test::Unit; using NetworkLayerStats = StatsBasedNetworkQualityMetricsReporter::NetworkLayerStats; diff --git a/test/pc/e2e/stats_based_network_quality_metrics_reporter_test.cc b/test/pc/e2e/stats_based_network_quality_metrics_reporter_test.cc index fe2947aa79..b06324361e 100644 --- a/test/pc/e2e/stats_based_network_quality_metrics_reporter_test.cc +++ b/test/pc/e2e/stats_based_network_quality_metrics_reporter_test.cc @@ -39,11 +39,11 @@ namespace { using ::testing::UnorderedElementsAre; -using ::webrtc::test::DefaultMetricsLogger; -using ::webrtc::test::ImprovementDirection; -using ::webrtc::test::Metric; -using ::webrtc::test::Unit; -using ::webrtc::webrtc_pc_e2e::PeerConfigurer; +using test::DefaultMetricsLogger; +using test::ImprovementDirection; +using test::Metric; +using test::Unit; +using webrtc_pc_e2e::PeerConfigurer; // Adds a peer with some audio and video (the client should not care about // details about audio and video configs). diff --git a/test/pc/e2e/stats_poller.cc b/test/pc/e2e/stats_poller.cc index 49daf20c1b..23db7acbd3 100644 --- a/test/pc/e2e/stats_poller.cc +++ b/test/pc/e2e/stats_poller.cc @@ -32,7 +32,7 @@ void InternalStatsObserver::OnStatsDelivered( StatsPoller::StatsPoller(std::vector observers, std::map peers) : observers_(std::move(observers)) { - webrtc::MutexLock lock(&mutex_); + MutexLock lock(&mutex_); for (auto& peer : peers) { pollers_.push_back(make_ref_counted( peer.first, peer.second, observers_)); @@ -42,7 +42,7 @@ StatsPoller::StatsPoller(std::vector observers, StatsPoller::StatsPoller(std::vector observers, std::map peers) : observers_(std::move(observers)) { - webrtc::MutexLock lock(&mutex_); + MutexLock lock(&mutex_); for (auto& peer : peers) { pollers_.push_back(make_ref_counted( peer.first, peer.second, observers_)); @@ -50,7 +50,7 @@ StatsPoller::StatsPoller(std::vector observers, } void StatsPoller::PollStatsAndNotifyObservers() { - webrtc::MutexLock lock(&mutex_); + MutexLock lock(&mutex_); for (auto& poller : pollers_) { poller->PollStats(); } @@ -58,13 +58,13 @@ void StatsPoller::PollStatsAndNotifyObservers() { void StatsPoller::RegisterParticipantInCall(absl::string_view peer_name, StatsProvider* peer) { - webrtc::MutexLock lock(&mutex_); + MutexLock lock(&mutex_); pollers_.push_back( make_ref_counted(peer_name, peer, observers_)); } bool StatsPoller::UnregisterParticipantInCall(absl::string_view peer_name) { - webrtc::MutexLock lock(&mutex_); + MutexLock lock(&mutex_); for (auto it = pollers_.begin(); it != pollers_.end(); ++it) { if ((*it)->pc_label() == peer_name) { pollers_.erase(it); diff --git a/test/pc/e2e/stats_poller_test.cc b/test/pc/e2e/stats_poller_test.cc index 754145f629..ce6ea2476c 100644 --- a/test/pc/e2e/stats_poller_test.cc +++ b/test/pc/e2e/stats_poller_test.cc @@ -41,7 +41,7 @@ class MockStatsObserver : public StatsObserverInterface { MOCK_METHOD(void, OnStatsReports, (absl::string_view pc_label, - const webrtc::scoped_refptr& report)); + const scoped_refptr& report)); }; TEST(StatsPollerTest, UnregisterParticipantAddedInCtor) { diff --git a/test/pc/e2e/test_peer.cc b/test/pc/e2e/test_peer.cc index a0c5dc48f3..9028f947dc 100644 --- a/test/pc/e2e/test_peer.cc +++ b/test/pc/e2e/test_peer.cc @@ -38,20 +38,20 @@ namespace webrtc_pc_e2e { namespace { class SetRemoteDescriptionCallback - : public webrtc::SetRemoteDescriptionObserverInterface { + : public SetRemoteDescriptionObserverInterface { public: - void OnSetRemoteDescriptionComplete(webrtc::RTCError error) override { + void OnSetRemoteDescriptionComplete(RTCError error) override { is_called_ = true; error_ = error; } bool is_called() const { return is_called_; } - webrtc::RTCError error() const { return error_; } + RTCError error() const { return error_; } private: bool is_called_ = false; - webrtc::RTCError error_; + RTCError error_; }; } // namespace diff --git a/test/pc/e2e/test_peer_factory.cc b/test/pc/e2e/test_peer_factory.cc index 4ff994ada1..85d356a182 100644 --- a/test/pc/e2e/test_peer_factory.cc +++ b/test/pc/e2e/test_peer_factory.cc @@ -48,7 +48,7 @@ namespace webrtc_pc_e2e { namespace { using EmulatedSFUConfigMap = - ::webrtc::webrtc_pc_e2e::QualityAnalyzingVideoEncoder::EmulatedSFUConfigMap; + webrtc_pc_e2e::QualityAnalyzingVideoEncoder::EmulatedSFUConfigMap; constexpr int16_t kGeneratedAudioMaxAmplitude = 32000; constexpr int kDefaultSamplingFrequencyInHz = 48000; diff --git a/test/run_loop.cc b/test/run_loop.cc index d3495e00e8..7bcde87022 100644 --- a/test/run_loop.cc +++ b/test/run_loop.cc @@ -51,7 +51,7 @@ void RunLoop::FakeSocketServer::FailNextWait() { fail_next_wait_ = true; } -bool RunLoop::FakeSocketServer::Wait(webrtc::TimeDelta max_wait_duration, +bool RunLoop::FakeSocketServer::Wait(TimeDelta max_wait_duration, bool process_io) { if (fail_next_wait_) { fail_next_wait_ = false; diff --git a/test/scoped_key_value_config.cc b/test/scoped_key_value_config.cc index df84462637..6f1230da4b 100644 --- a/test/scoped_key_value_config.cc +++ b/test/scoped_key_value_config.cc @@ -115,7 +115,7 @@ std::string ScopedKeyValueConfig::LookupRecurse(absl::string_view key) const { // When at the root, check the global string so that test programs using // a mix between ScopedKeyValueConfig and the global string continue to work - return webrtc::field_trial::FindFullName(std::string(key)); + return field_trial::FindFullName(std::string(key)); } } // namespace test diff --git a/test/test_main_lib.cc b/test/test_main_lib.cc index 8931adf776..8aa63c395d 100644 --- a/test/test_main_lib.cc +++ b/test/test_main_lib.cc @@ -204,10 +204,9 @@ class TestMainImpl : public TestMain { if (absl::GetFlag(FLAGS_export_perf_results_new_api)) { exporters.push_back(std::make_unique()); if (!absl::GetFlag(FLAGS_webrtc_test_metrics_output_path).empty()) { - exporters.push_back( - std::make_unique( - webrtc::test::MetricsSetProtoFileExporter::Options( - absl::GetFlag(FLAGS_webrtc_test_metrics_output_path)))); + exporters.push_back(std::make_unique( + test::MetricsSetProtoFileExporter::Options( + absl::GetFlag(FLAGS_webrtc_test_metrics_output_path)))); } if (!absl::GetFlag(FLAGS_isolated_script_test_perf_output).empty()) { exporters.push_back( @@ -224,12 +223,12 @@ class TestMainImpl : public TestMain { std::string perf_output_file = absl::GetFlag(FLAGS_isolated_script_test_perf_output); if (!perf_output_file.empty()) { - if (!webrtc::test::WritePerfResults(perf_output_file)) { + if (!test::WritePerfResults(perf_output_file)) { return 1; } } if (metrics_to_plot) { - webrtc::test::PrintPlottableResults(*metrics_to_plot); + test::PrintPlottableResults(*metrics_to_plot); } } diff --git a/test/testsupport/copy_to_file_audio_capturer_unittest.cc b/test/testsupport/copy_to_file_audio_capturer_unittest.cc index e97b69860a..023b61da2b 100644 --- a/test/testsupport/copy_to_file_audio_capturer_unittest.cc +++ b/test/testsupport/copy_to_file_audio_capturer_unittest.cc @@ -23,8 +23,8 @@ namespace test { class CopyToFileAudioCapturerTest : public ::testing::Test { protected: void SetUp() override { - temp_filename_ = webrtc::test::TempFilename( - webrtc::test::OutputPath(), "copy_to_file_audio_capturer_unittest"); + temp_filename_ = test::TempFilename(test::OutputPath(), + "copy_to_file_audio_capturer_unittest"); std::unique_ptr delegate = TestAudioDeviceModule::CreatePulsedNoiseCapturer(32000, 48000); capturer_ = std::make_unique(std::move(delegate), diff --git a/test/testsupport/file_utils.cc b/test/testsupport/file_utils.cc index 963514c88d..e41046c7f9 100644 --- a/test/testsupport/file_utils.cc +++ b/test/testsupport/file_utils.cc @@ -94,18 +94,18 @@ bool DirExists(absl::string_view directory_name) { } std::string OutputPath() { - return webrtc::test::internal::OutputPath(); + return test::internal::OutputPath(); } std::string OutputPathWithRandomDirectory() { - std::string path = webrtc::test::internal::OutputPath(); + std::string path = test::internal::OutputPath(); std::string rand_dir = path + CreateRandomUuid(); RTC_CHECK(CreateDir(rand_dir)) << "Failed to create dir: " << rand_dir; return rand_dir + std::string(kPathDelimiter); } std::string WorkingDir() { - return webrtc::test::internal::WorkingDir(); + return test::internal::WorkingDir(); } // Generate a temporary filename in a safe way. @@ -254,7 +254,7 @@ bool RemoveFile(absl::string_view file_name) { } std::string ResourcePath(absl::string_view name, absl::string_view extension) { - return webrtc::test::internal::ResourcePath(name, extension); + return test::internal::ResourcePath(name, extension); } std::string JoinFilename(absl::string_view dir, absl::string_view name) { diff --git a/test/testsupport/file_utils_unittest.cc b/test/testsupport/file_utils_unittest.cc index adb9a2a0ca..104db59221 100644 --- a/test/testsupport/file_utils_unittest.cc +++ b/test/testsupport/file_utils_unittest.cc @@ -76,9 +76,7 @@ class FileUtilsTest : public ::testing::Test { FileUtilsTest() {} ~FileUtilsTest() override {} // Runs before the first test - static void SetUpTestSuite() { - original_working_dir_ = webrtc::test::WorkingDir(); - } + static void SetUpTestSuite() { original_working_dir_ = test::WorkingDir(); } void SetUp() override { ASSERT_EQ(chdir(original_working_dir_.c_str()), 0); } void TearDown() override { ASSERT_EQ(chdir(original_working_dir_.c_str()), 0); @@ -104,7 +102,7 @@ std::string ExpectedRootDirByPlatform() { TEST_F(FileUtilsTest, OutputPathFromUnchangedWorkingDir) { std::string expected_end = ExpectedRootDirByPlatform(); - std::string result = webrtc::test::OutputPath(); + std::string result = test::OutputPath(); ASSERT_THAT(result, EndsWith(expected_end)); } @@ -115,7 +113,7 @@ TEST_F(FileUtilsTest, OutputPathFromRootWorkingDir) { ASSERT_EQ(0, chdir(kPathDelimiter.data())); std::string expected_end = ExpectedRootDirByPlatform(); - std::string result = webrtc::test::OutputPath(); + std::string result = test::OutputPath(); ASSERT_THAT(result, EndsWith(expected_end)); } @@ -125,7 +123,7 @@ TEST_F(FileUtilsTest, RandomOutputPathFromUnchangedWorkingDir) { std::string fixed_first_uuid = "def01482-f829-429a-bfd4-841706e92cdd"; std::string expected_end = ExpectedRootDirByPlatform() + fixed_first_uuid + std::string(kPathDelimiter); - std::string result = webrtc::test::OutputPathWithRandomDirectory(); + std::string result = test::OutputPathWithRandomDirectory(); ASSERT_THAT(result, EndsWith(expected_end)); } @@ -137,23 +135,23 @@ TEST_F(FileUtilsTest, RandomOutputPathFromRootWorkingDir) { std::string fixed_first_uuid = "def01482-f829-429a-bfd4-841706e92cdd"; std::string expected_end = ExpectedRootDirByPlatform() + fixed_first_uuid + std::string(kPathDelimiter); - std::string result = webrtc::test::OutputPathWithRandomDirectory(); + std::string result = test::OutputPathWithRandomDirectory(); ASSERT_THAT(result, EndsWith(expected_end)); } TEST_F(FileUtilsTest, TempFilename) { - std::string temp_filename = webrtc::test::TempFilename( - webrtc::test::OutputPath(), "TempFilenameTest"); - ASSERT_TRUE(webrtc::test::FileExists(temp_filename)) + std::string temp_filename = + test::TempFilename(test::OutputPath(), "TempFilenameTest"); + ASSERT_TRUE(test::FileExists(temp_filename)) << "Couldn't find file: " << temp_filename; remove(temp_filename.c_str()); } TEST_F(FileUtilsTest, GenerateTempFilename) { - std::string temp_filename = webrtc::test::GenerateTempFilename( - webrtc::test::OutputPath(), "TempFilenameTest"); - ASSERT_FALSE(webrtc::test::FileExists(temp_filename)) + std::string temp_filename = + test::GenerateTempFilename(test::OutputPath(), "TempFilenameTest"); + ASSERT_FALSE(test::FileExists(temp_filename)) << "File exists: " << temp_filename; FILE* file = fopen(temp_filename.c_str(), "wb"); ASSERT_TRUE(file != NULL) << "Failed to open file: " << temp_filename; @@ -174,20 +172,20 @@ TEST_F(FileUtilsTest, MAYBE_CreateDir) { test::OutputPathWithRandomDirectory() + "fileutils-unittest-empty-dir"; // Make sure it's removed if a previous test has failed: remove(directory.c_str()); - ASSERT_TRUE(webrtc::test::CreateDir(directory)); + ASSERT_TRUE(test::CreateDir(directory)); remove(directory.c_str()); } TEST_F(FileUtilsTest, WorkingDirReturnsValue) { // This will obviously be different depending on where the webrtc checkout is, // so just check something is returned. - std::string working_dir = webrtc::test::WorkingDir(); + std::string working_dir = test::WorkingDir(); ASSERT_GT(working_dir.length(), 0u); } TEST_F(FileUtilsTest, ResourcePathReturnsCorrectPath) { - std::string result = webrtc::test::ResourcePath( - Path("video_coding/frame-ethernet-ii"), "pcap"); + std::string result = + test::ResourcePath(Path("video_coding/frame-ethernet-ii"), "pcap"); #if defined(WEBRTC_IOS) // iOS bundles resources straight into the bundle root. std::string expected_end = Path("/frame-ethernet-ii.pcap"); @@ -205,7 +203,7 @@ TEST_F(FileUtilsTest, ResourcePathReturnsCorrectPath) { TEST_F(FileUtilsTest, ResourcePathFromRootWorkingDir) { ASSERT_EQ(0, chdir(kPathDelimiter.data())); - std::string resource = webrtc::test::ResourcePath("whatever", "ext"); + std::string resource = test::ResourcePath("whatever", "ext"); #if !defined(WEBRTC_IOS) ASSERT_NE(resource.find("resources"), std::string::npos); #endif @@ -215,37 +213,36 @@ TEST_F(FileUtilsTest, ResourcePathFromRootWorkingDir) { TEST_F(FileUtilsTest, GetFileSizeExistingFile) { // Create a file with some dummy data in. - std::string temp_filename = webrtc::test::TempFilename( - webrtc::test::OutputPath(), "fileutils_unittest"); + std::string temp_filename = + test::TempFilename(test::OutputPath(), "fileutils_unittest"); FILE* file = fopen(temp_filename.c_str(), "wb"); ASSERT_TRUE(file != NULL) << "Failed to open file: " << temp_filename; ASSERT_GT(fprintf(file, "%s", "Dummy data"), 0) << "Failed to write to file: " << temp_filename; fclose(file); - ASSERT_GT(webrtc::test::GetFileSize(temp_filename), 0u); + ASSERT_GT(test::GetFileSize(temp_filename), 0u); remove(temp_filename.c_str()); } TEST_F(FileUtilsTest, GetFileSizeNonExistingFile) { - ASSERT_EQ(0u, webrtc::test::GetFileSize("non-existing-file.tmp")); + ASSERT_EQ(0u, test::GetFileSize("non-existing-file.tmp")); } TEST_F(FileUtilsTest, DirExists) { // Check that an existing directory is recognized as such. - ASSERT_TRUE(webrtc::test::DirExists(webrtc::test::OutputPath())) + ASSERT_TRUE(test::DirExists(test::OutputPath())) << "Existing directory not found"; // Check that a non-existing directory is recognized as such. std::string directory = "direxists-unittest-non_existing-dir"; - ASSERT_FALSE(webrtc::test::DirExists(directory)) - << "Non-existing directory found"; + ASSERT_FALSE(test::DirExists(directory)) << "Non-existing directory found"; // Check that an existing file is not recognized as an existing directory. - std::string temp_filename = webrtc::test::TempFilename( - webrtc::test::OutputPath(), "TempFilenameTest"); - ASSERT_TRUE(webrtc::test::FileExists(temp_filename)) + std::string temp_filename = + test::TempFilename(test::OutputPath(), "TempFilenameTest"); + ASSERT_TRUE(test::FileExists(temp_filename)) << "Couldn't find file: " << temp_filename; - ASSERT_FALSE(webrtc::test::DirExists(temp_filename)) + ASSERT_FALSE(test::DirExists(temp_filename)) << "Existing file recognized as existing directory"; remove(temp_filename.c_str()); } diff --git a/test/testsupport/fixed_fps_video_frame_writer_adapter_test.cc b/test/testsupport/fixed_fps_video_frame_writer_adapter_test.cc index 5ee4701cc9..473e63dae3 100644 --- a/test/testsupport/fixed_fps_video_frame_writer_adapter_test.cc +++ b/test/testsupport/fixed_fps_video_frame_writer_adapter_test.cc @@ -36,7 +36,7 @@ class InMemoryVideoWriter : public VideoFrameWriter { public: ~InMemoryVideoWriter() override = default; - bool WriteFrame(const webrtc::VideoFrame& frame) override { + bool WriteFrame(const VideoFrame& frame) override { MutexLock lock(&mutex_); received_frames_.push_back(frame); return true; diff --git a/test/testsupport/ivf_video_frame_generator_unittest.cc b/test/testsupport/ivf_video_frame_generator_unittest.cc index ff0dd08d87..a2c3fea3c2 100644 --- a/test/testsupport/ivf_video_frame_generator_unittest.cc +++ b/test/testsupport/ivf_video_frame_generator_unittest.cc @@ -101,10 +101,9 @@ class IvfFileWriterEncodedCallback : public EncodedImageCallback { class IvfVideoFrameGeneratorTest : public ::testing::Test { protected: void SetUp() override { - file_name_ = - webrtc::test::TempFilename(webrtc::test::OutputPath(), "test_file.ivf"); + file_name_ = test::TempFilename(test::OutputPath(), "test_file.ivf"); } - void TearDown() override { webrtc::test::RemoveFile(file_name_); } + void TearDown() override { test::RemoveFile(file_name_); } VideoFrame BuildFrame(FrameGeneratorInterface::VideoFrameData frame_data) { return VideoFrame::Builder() @@ -121,7 +120,7 @@ class IvfVideoFrameGeneratorTest : public ::testing::Test { std::nullopt); VideoCodec codec_settings; - webrtc::test::CodecSettings(video_codec_type, &codec_settings); + test::CodecSettings(video_codec_type, &codec_settings); codec_settings.width = kWidth; codec_settings.height = kHeight; codec_settings.maxFramerate = kMaxFramerate; diff --git a/test/testsupport/perf_test.cc b/test/testsupport/perf_test.cc index c274ee1a7b..cec361bd8c 100644 --- a/test/testsupport/perf_test.cc +++ b/test/testsupport/perf_test.cc @@ -31,15 +31,14 @@ namespace test { namespace { -std::string UnitWithDirection( - absl::string_view units, - webrtc::test::ImproveDirection improve_direction) { +std::string UnitWithDirection(absl::string_view units, + test::ImproveDirection improve_direction) { switch (improve_direction) { - case webrtc::test::ImproveDirection::kNone: + case test::ImproveDirection::kNone: return std::string(units); - case webrtc::test::ImproveDirection::kSmallerIsBetter: + case test::ImproveDirection::kSmallerIsBetter: return std::string(units) + "_smallerIsBetter"; - case webrtc::test::ImproveDirection::kBiggerIsBetter: + case test::ImproveDirection::kBiggerIsBetter: return std::string(units) + "_biggerIsBetter"; } } @@ -69,7 +68,7 @@ void OutputListToStream(std::ostream* ostream, const Container& values) { struct PlottableCounter { std::string graph_name; std::string trace_name; - webrtc::SamplesStatsCounter counter; + SamplesStatsCounter counter; std::string units; }; @@ -84,7 +83,7 @@ class PlottableCounterPrinter { void AddCounter(absl::string_view graph_name, absl::string_view trace_name, - const webrtc::SamplesStatsCounter& counter, + const SamplesStatsCounter& counter, absl::string_view units) { MutexLock lock(&mutex_); plottable_counters_.push_back({std::string(graph_name), @@ -181,7 +180,7 @@ class ResultsLinePrinter { const ArrayView values, absl::string_view units, const bool important, - webrtc::test::ImproveDirection improve_direction) { + test::ImproveDirection improve_direction) { std::ostringstream value_stream; value_stream.precision(8); OutputListToStream(&value_stream, values); diff --git a/test/testsupport/video_frame_writer.cc b/test/testsupport/video_frame_writer.cc index 09af7f12ea..96f1b0d810 100644 --- a/test/testsupport/video_frame_writer.cc +++ b/test/testsupport/video_frame_writer.cc @@ -71,7 +71,7 @@ Y4mVideoFrameWriterImpl::Y4mVideoFrameWriterImpl(std::string output_file_name, RTC_CHECK(frame_writer_->Init()); } -bool Y4mVideoFrameWriterImpl::WriteFrame(const webrtc::VideoFrame& frame) { +bool Y4mVideoFrameWriterImpl::WriteFrame(const VideoFrame& frame) { Buffer frame_buffer = ExtractI420BufferWithSize(frame, width_, height_); RTC_CHECK_EQ(frame_buffer.size(), frame_writer_->FrameLength()); return frame_writer_->WriteFrame(frame_buffer.data()); @@ -97,7 +97,7 @@ YuvVideoFrameWriterImpl::YuvVideoFrameWriterImpl(std::string output_file_name, RTC_CHECK(frame_writer_->Init()); } -bool YuvVideoFrameWriterImpl::WriteFrame(const webrtc::VideoFrame& frame) { +bool YuvVideoFrameWriterImpl::WriteFrame(const VideoFrame& frame) { Buffer frame_buffer = ExtractI420BufferWithSize(frame, width_, height_); RTC_CHECK_EQ(frame_buffer.size(), frame_writer_->FrameLength()); return frame_writer_->WriteFrame(frame_buffer.data()); diff --git a/test/testsupport/video_frame_writer_unittest.cc b/test/testsupport/video_frame_writer_unittest.cc index 30ebc526d7..de8c618216 100644 --- a/test/testsupport/video_frame_writer_unittest.cc +++ b/test/testsupport/video_frame_writer_unittest.cc @@ -56,8 +56,8 @@ scoped_refptr CreateI420Buffer(int width, int height) { return buffer; } -void AssertI420BuffersEq(scoped_refptr actual, - scoped_refptr expected) { +void AssertI420BuffersEq(scoped_refptr actual, + scoped_refptr expected) { ASSERT_TRUE(actual); ASSERT_EQ(actual->width(), expected->width()); @@ -93,8 +93,8 @@ class VideoFrameWriterTest : public ::testing::Test { ~VideoFrameWriterTest() override = default; void SetUp() override { - temp_filename_ = webrtc::test::TempFilename(webrtc::test::OutputPath(), - "video_frame_writer_unittest"); + temp_filename_ = + test::TempFilename(test::OutputPath(), "video_frame_writer_unittest"); frame_writer_ = CreateFrameWriter(); } diff --git a/test/testsupport/y4m_frame_generator.cc b/test/testsupport/y4m_frame_generator.cc index bd3c8f4127..1c5e0c8a58 100644 --- a/test/testsupport/y4m_frame_generator.cc +++ b/test/testsupport/y4m_frame_generator.cc @@ -50,15 +50,14 @@ Y4mFrameGenerator::Y4mFrameGenerator(absl::string_view filename, RTC_CHECK_GT(height_, 0); // Delegate the actual reads (from NextFrame) to a Y4mReader. - frame_reader_ = webrtc::test::CreateY4mFrameReader( + frame_reader_ = test::CreateY4mFrameReader( filename_, ToYuvFrameReaderRepeatMode(repeat_mode_)); } Y4mFrameGenerator::VideoFrameData Y4mFrameGenerator::NextFrame() { - webrtc::VideoFrame::UpdateRect update_rect{0, 0, static_cast(width_), - static_cast(height_)}; - scoped_refptr next_frame_buffer = - frame_reader_->PullFrame(); + VideoFrame::UpdateRect update_rect{0, 0, static_cast(width_), + static_cast(height_)}; + scoped_refptr next_frame_buffer = frame_reader_->PullFrame(); if (!next_frame_buffer || (static_cast(next_frame_buffer->width()) == width_ && @@ -67,9 +66,8 @@ Y4mFrameGenerator::VideoFrameData Y4mFrameGenerator::NextFrame() { } // Allocate a new buffer and return scaled version. - scoped_refptr scaled_buffer( - I420Buffer::Create(width_, height_)); - webrtc::I420Buffer::SetBlack(scaled_buffer.get()); + scoped_refptr scaled_buffer(I420Buffer::Create(width_, height_)); + I420Buffer::SetBlack(scaled_buffer.get()); scaled_buffer->ScaleFrom(*next_frame_buffer->ToI420()); return VideoFrameData(scaled_buffer, update_rect); } diff --git a/test/testsupport/y4m_frame_reader_unittest.cc b/test/testsupport/y4m_frame_reader_unittest.cc index 205cbdeb39..406c69eee3 100644 --- a/test/testsupport/y4m_frame_reader_unittest.cc +++ b/test/testsupport/y4m_frame_reader_unittest.cc @@ -41,8 +41,8 @@ class Y4mFrameReaderTest : public ::testing::Test { ~Y4mFrameReaderTest() override = default; void SetUp() override { - filepath_ = webrtc::test::TempFilename(webrtc::test::OutputPath(), - "y4m_frame_reader_unittest"); + filepath_ = + test::TempFilename(test::OutputPath(), "y4m_frame_reader_unittest"); FILE* file = fopen(filepath_.c_str(), "wb"); fwrite(kFileHeader, 1, sizeof(kFileHeader) - 1, file); for (int n = 0; n < kNumFrames; ++n) { diff --git a/test/testsupport/y4m_frame_writer_unittest.cc b/test/testsupport/y4m_frame_writer_unittest.cc index f12a4b8e4f..ebb34f53ce 100644 --- a/test/testsupport/y4m_frame_writer_unittest.cc +++ b/test/testsupport/y4m_frame_writer_unittest.cc @@ -38,8 +38,8 @@ class Y4mFrameWriterTest : public ::testing::Test { ~Y4mFrameWriterTest() override = default; void SetUp() override { - temp_filename_ = webrtc::test::TempFilename(webrtc::test::OutputPath(), - "y4m_frame_writer_unittest"); + temp_filename_ = + test::TempFilename(test::OutputPath(), "y4m_frame_writer_unittest"); frame_writer_.reset(new Y4mFrameWriterImpl(temp_filename_, kFrameWidth, kFrameHeight, kFrameRate)); ASSERT_TRUE(frame_writer_->Init()); diff --git a/test/testsupport/yuv_frame_reader_unittest.cc b/test/testsupport/yuv_frame_reader_unittest.cc index 82d242ff32..9d633d7006 100644 --- a/test/testsupport/yuv_frame_reader_unittest.cc +++ b/test/testsupport/yuv_frame_reader_unittest.cc @@ -38,8 +38,8 @@ class YuvFrameReaderTest : public ::testing::Test { ~YuvFrameReaderTest() override = default; void SetUp() override { - filepath_ = webrtc::test::TempFilename(webrtc::test::OutputPath(), - "yuv_frame_reader_unittest"); + filepath_ = + test::TempFilename(test::OutputPath(), "yuv_frame_reader_unittest"); CreateYuvFileAndReader(/*num_frames=*/3, RepeatMode::kSingle); } diff --git a/test/testsupport/yuv_frame_writer_unittest.cc b/test/testsupport/yuv_frame_writer_unittest.cc index 13ed715b9e..22a048b7a2 100644 --- a/test/testsupport/yuv_frame_writer_unittest.cc +++ b/test/testsupport/yuv_frame_writer_unittest.cc @@ -34,8 +34,8 @@ class YuvFrameWriterTest : public ::testing::Test { ~YuvFrameWriterTest() override = default; void SetUp() override { - temp_filename_ = webrtc::test::TempFilename(webrtc::test::OutputPath(), - "yuv_frame_writer_unittest"); + temp_filename_ = + test::TempFilename(test::OutputPath(), "yuv_frame_writer_unittest"); frame_writer_.reset( new YuvFrameWriterImpl(temp_filename_, kFrameWidth, kFrameHeight)); ASSERT_TRUE(frame_writer_->Init()); diff --git a/test/time_controller/time_controller_conformance_test.cc b/test/time_controller/time_controller_conformance_test.cc index 5e0dcf85cc..265404156e 100644 --- a/test/time_controller/time_controller_conformance_test.cc +++ b/test/time_controller/time_controller_conformance_test.cc @@ -45,11 +45,11 @@ std::unique_ptr CreateTimeController(TimeMode mode) { } } -std::string ParamsToString(const TestParamInfo& param) { +std::string ParamsToString(const TestParamInfo& param) { switch (param.param) { - case webrtc::TimeMode::kRealTime: + case TimeMode::kRealTime: return "RealTime"; - case webrtc::TimeMode::kSimulated: + case TimeMode::kSimulated: return "SimulatedTime"; default: RTC_DCHECK_NOTREACHED() << "Time mode not supported"; @@ -76,7 +76,7 @@ class ExecutionOrderKeeper { // Tests conformance between real time and simulated time time controller. class SimulatedRealTimeControllerConformanceTest - : public TestWithParam {}; + : public TestWithParam {}; TEST_P(SimulatedRealTimeControllerConformanceTest, ThreadPostOrderTest) { std::unique_ptr time_controller = @@ -153,7 +153,7 @@ TEST_P(SimulatedRealTimeControllerConformanceTest, std::unique_ptr time_controller = CreateTimeController(GetParam()); auto task_queue = time_controller->GetTaskQueueFactory()->CreateTaskQueue( - "task_queue", webrtc::TaskQueueFactory::Priority::NORMAL); + "task_queue", TaskQueueFactory::Priority::NORMAL); // Tasks on thread have to be executed in order in which they were // posted/invoked. diff --git a/test/vcm_capturer.cc b/test/vcm_capturer.cc index 0a9226ef6f..bc022c1a80 100644 --- a/test/vcm_capturer.cc +++ b/test/vcm_capturer.cc @@ -41,7 +41,7 @@ bool VcmCapturer::Init(size_t width, return false; } - vcm_ = webrtc::VideoCaptureFactory::Create(unique_name); + vcm_ = VideoCaptureFactory::Create(unique_name); if (!vcm_) { return false; } diff --git a/test/video_codec_tester.cc b/test/video_codec_tester.cc index 7c40fc2e5e..edd917f878 100644 --- a/test/video_codec_tester.cc +++ b/test/video_codec_tester.cc @@ -108,7 +108,7 @@ using PacingMode = PacingSettings::PacingMode; using VideoCodecStats = VideoCodecTester::VideoCodecStats; using DecodeCallback = absl::AnyInvocable; -using webrtc::test::ImprovementDirection; +using test::ImprovementDirection; constexpr Frequency k90kHz = Frequency::Hertz(90000); @@ -588,8 +588,8 @@ class VideoCodecAnalyzer : public VideoCodecTester::VideoCodecStats { } Frame superframe = subframes.back(); - for (const Frame& frame : webrtc::ArrayView(subframes).subview( - 0, subframes.size() - 1)) { + for (const Frame& frame : + ArrayView(subframes).subview(0, subframes.size() - 1)) { superframe.decoded |= frame.decoded; superframe.encoded |= frame.encoded; superframe.frame_size += frame.frame_size; diff --git a/test/video_codec_tester_unittest.cc b/test/video_codec_tester_unittest.cc index d78c7c6416..a24a1db072 100644 --- a/test/video_codec_tester_unittest.cc +++ b/test/video_codec_tester_unittest.cc @@ -103,8 +103,8 @@ scoped_refptr CreateYuvBuffer(uint8_t y = 0, // TODO(ssilkin): Wrap this into a class that removes file in dtor. std::string CreateYuvFile(int width, int height, int num_frames) { - std::string path = webrtc::test::TempFilename(webrtc::test::OutputPath(), - "video_codec_tester_unittest"); + std::string path = + test::TempFilename(test::OutputPath(), "video_codec_tester_unittest"); FILE* file = fopen(path.c_str(), "wb"); for (int frame_num = 0; frame_num < num_frames; ++frame_num) { // For purposes of testing quality estimation, we need Y, U, V values in From 2f5c2662392e5393d0cd9a2f00e7cf1c45381e98 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Mon, 12 May 2025 07:43:09 +0000 Subject: [PATCH 116/292] [cleanup] Fix redundant webrtc name specifier This CL was uploaded by git cl split. R=hta@webrtc.org No-Iwyu: LSC Bug: webrtc:42232595 Change-Id: Ia71cdef9e95eec403edf857dd3d01bd8348e4a64 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390560 Auto-Submit: Evan Shrubsole Commit-Queue: Evan Shrubsole Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44588} --- api/candidate.cc | 20 +- api/create_peerconnection_factory.cc | 2 +- api/enable_media.cc | 2 +- api/field_trials_unittest.cc | 4 +- api/rtp_parameters.cc | 53 +- api/rtp_parameters_unittest.cc | 1 - api/sequence_checker_unittest.cc | 2 +- api/task_queue/task_queue_test.cc | 34 +- media/base/adapted_video_track_source.cc | 6 +- media/base/codec_comparators.cc | 5 +- media/base/codec_comparators_unittest.cc | 9 +- media/base/codec_list.cc | 7 +- media/base/codec_list_unittest.cc | 6 +- media/base/codec_unittest.cc | 66 ++- media/base/fake_frame_source.cc | 12 +- media/base/fake_media_engine.cc | 12 +- media/base/media_engine.cc | 18 +- media/base/rtp_utils.cc | 24 +- media/base/turn_utils.cc | 10 +- media/base/video_adapter.cc | 2 +- media/base/video_adapter_unittest.cc | 19 +- media/base/video_broadcaster.cc | 2 +- media/base/video_common_unittest.cc | 4 +- media/engine/fake_video_codec_factory.cc | 9 +- media/engine/fake_webrtc_call.cc | 14 +- media/engine/fake_webrtc_video_engine.cc | 5 +- .../internal_decoder_factory_unittest.cc | 2 +- media/engine/internal_encoder_factory.cc | 4 +- media/engine/simulcast_encoder_adapter.cc | 18 +- .../simulcast_encoder_adapter_unittest.cc | 10 +- media/engine/webrtc_media_engine.cc | 3 +- media/engine/webrtc_media_engine_unittest.cc | 2 +- media/engine/webrtc_video_engine.cc | 150 +++-- media/engine/webrtc_video_engine_unittest.cc | 524 ++++++++---------- 34 files changed, 486 insertions(+), 575 deletions(-) diff --git a/api/candidate.cc b/api/candidate.cc index 30a6c1b87b..bb12f06a3b 100644 --- a/api/candidate.cc +++ b/api/candidate.cc @@ -48,8 +48,8 @@ Candidate::Candidate() : id_(CreateRandomString(8)), component_(ICE_CANDIDATE_COMPONENT_DEFAULT), priority_(0), - network_type_(webrtc::ADAPTER_TYPE_UNKNOWN), - underlying_type_for_vpn_(webrtc::ADAPTER_TYPE_UNKNOWN), + network_type_(ADAPTER_TYPE_UNKNOWN), + underlying_type_for_vpn_(ADAPTER_TYPE_UNKNOWN), generation_(0), network_id_(0), network_cost_(0) {} @@ -73,8 +73,8 @@ Candidate::Candidate(int component, username_(username), password_(password), type_(type), - network_type_(webrtc::ADAPTER_TYPE_UNKNOWN), - underlying_type_for_vpn_(webrtc::ADAPTER_TYPE_UNKNOWN), + network_type_(ADAPTER_TYPE_UNKNOWN), + underlying_type_for_vpn_(ADAPTER_TYPE_UNKNOWN), generation_(generation), foundation_(foundation), network_id_(network_id), @@ -102,7 +102,7 @@ bool Candidate::is_relay() const { } absl::string_view Candidate::type_name() const { - return webrtc::IceCandidateTypeToString(type_); + return IceCandidateTypeToString(type_); } bool Candidate::IsEquivalent(const Candidate& c) const { @@ -157,7 +157,7 @@ uint32_t Candidate::GetPriority(uint32_t type_preference, // local preference = (NIC Type << 8 | Addr_Pref) + relay preference. // The relay preference is based on the number of TURN servers, the // first TURN server gets the highest preference. - int addr_pref = webrtc::IPAddressPrecedence(address_.ipaddr()); + int addr_pref = IPAddressPrecedence(address_.ipaddr()); int local_preference = ((network_adapter_preference << 8) | addr_pref) + relay_preference; @@ -205,7 +205,7 @@ Candidate Candidate::ToSanitizedCopy(bool use_hostname_address, // IP needs to be redacted, but no hostname available. SocketAddress redacted_addr("redacted-ip.invalid", address().port()); copy.set_address(redacted_addr); - } else if (webrtc::IPFromString(address().hostname(), &ip)) { + } else if (IPFromString(address().hostname(), &ip)) { // The hostname is an IP literal, and needs to be redacted too. SocketAddress redacted_addr("redacted-literal.invalid", address().port()); copy.set_address(redacted_addr); @@ -216,7 +216,7 @@ Candidate Candidate::ToSanitizedCopy(bool use_hostname_address, } if (filter_related_address) { copy.set_related_address( - webrtc::EmptySocketAddressWithFamily(copy.address().family())); + EmptySocketAddressWithFamily(copy.address().family())); } if (filter_ufrag) { copy.set_username(""); @@ -253,13 +253,13 @@ void Candidate::ComputeFoundation(const SocketAddress& base_address, // - 1 (that is, a 64-bit positive integer). This number is used in // connectivity checks to detect and repair this case [...] sb << absl::StrCat(tie_breaker); - foundation_ = absl::StrCat(webrtc::ComputeCrc32(sb.Release())); + foundation_ = absl::StrCat(ComputeCrc32(sb.Release())); } void Candidate::ComputePrflxFoundation() { RTC_DCHECK(is_prflx()); RTC_DCHECK(!id_.empty()); - foundation_ = absl::StrCat(webrtc::ComputeCrc32(id_)); + foundation_ = absl::StrCat(ComputeCrc32(id_)); } void Candidate::Assign(std::string& s, absl::string_view view) { diff --git a/api/create_peerconnection_factory.cc b/api/create_peerconnection_factory.cc index e0b50fc02e..aec4f12549 100644 --- a/api/create_peerconnection_factory.cc +++ b/api/create_peerconnection_factory.cc @@ -49,7 +49,7 @@ scoped_refptr CreatePeerConnectionFactory( dependencies.worker_thread = worker_thread; dependencies.signaling_thread = signaling_thread; dependencies.event_log_factory = std::make_unique(); - dependencies.env = webrtc::CreateEnvironment(std::move(field_trials)); + dependencies.env = CreateEnvironment(std::move(field_trials)); if (network_thread) { // TODO(bugs.webrtc.org/13145): Add an webrtc::SocketFactory* argument. diff --git a/api/enable_media.cc b/api/enable_media.cc index 7d6d0cc3df..25b1bafc9c 100644 --- a/api/enable_media.cc +++ b/api/enable_media.cc @@ -35,7 +35,7 @@ class MediaFactoryImpl : public MediaFactory { ~MediaFactoryImpl() override = default; std::unique_ptr CreateCall(CallConfig config) override { - return webrtc::Call::Create(std::move(config)); + return Call::Create(std::move(config)); } std::unique_ptr CreateMediaEngine( diff --git a/api/field_trials_unittest.cc b/api/field_trials_unittest.cc index d7a02513c7..9d492529d1 100644 --- a/api/field_trials_unittest.cc +++ b/api/field_trials_unittest.cc @@ -19,8 +19,8 @@ namespace webrtc { namespace { -using ::webrtc::field_trial::FieldTrialsAllowedInScopeForTesting; -using ::webrtc::test::ScopedFieldTrials; +using field_trial::FieldTrialsAllowedInScopeForTesting; +using test::ScopedFieldTrials; TEST(FieldTrialsTest, EmptyStringHasNoEffect) { FieldTrialsAllowedInScopeForTesting k({"MyCoolTrial"}); diff --git a/api/rtp_parameters.cc b/api/rtp_parameters.cc index bb2f500fd4..c6560d4bb0 100644 --- a/api/rtp_parameters.cc +++ b/api/rtp_parameters.cc @@ -166,35 +166,32 @@ constexpr int RtpExtension::kOneByteHeaderExtensionMaxId; constexpr int RtpExtension::kOneByteHeaderExtensionMaxValueSize; bool RtpExtension::IsSupportedForAudio(absl::string_view uri) { - return uri == webrtc::RtpExtension::kAudioLevelUri || - uri == webrtc::RtpExtension::kAbsSendTimeUri || - uri == webrtc::RtpExtension::kAbsoluteCaptureTimeUri || - uri == webrtc::RtpExtension::kTransportSequenceNumberUri || - uri == webrtc::RtpExtension::kTransportSequenceNumberV2Uri || - uri == webrtc::RtpExtension::kMidUri || - uri == webrtc::RtpExtension::kRidUri || - uri == webrtc::RtpExtension::kRepairedRidUri; + return uri == RtpExtension::kAudioLevelUri || + uri == RtpExtension::kAbsSendTimeUri || + uri == RtpExtension::kAbsoluteCaptureTimeUri || + uri == RtpExtension::kTransportSequenceNumberUri || + uri == RtpExtension::kTransportSequenceNumberV2Uri || + uri == RtpExtension::kMidUri || uri == RtpExtension::kRidUri || + uri == RtpExtension::kRepairedRidUri; } bool RtpExtension::IsSupportedForVideo(absl::string_view uri) { - return uri == webrtc::RtpExtension::kTimestampOffsetUri || - uri == webrtc::RtpExtension::kAbsSendTimeUri || - uri == webrtc::RtpExtension::kAbsoluteCaptureTimeUri || - uri == webrtc::RtpExtension::kVideoRotationUri || - uri == webrtc::RtpExtension::kTransportSequenceNumberUri || - uri == webrtc::RtpExtension::kTransportSequenceNumberV2Uri || - uri == webrtc::RtpExtension::kPlayoutDelayUri || - uri == webrtc::RtpExtension::kVideoContentTypeUri || - uri == webrtc::RtpExtension::kVideoTimingUri || - uri == webrtc::RtpExtension::kMidUri || - uri == webrtc::RtpExtension::kGenericFrameDescriptorUri00 || - uri == webrtc::RtpExtension::kDependencyDescriptorUri || - uri == webrtc::RtpExtension::kColorSpaceUri || - uri == webrtc::RtpExtension::kRidUri || - uri == webrtc::RtpExtension::kRepairedRidUri || - uri == webrtc::RtpExtension::kVideoLayersAllocationUri || - uri == webrtc::RtpExtension::kVideoFrameTrackingIdUri || - uri == webrtc::RtpExtension::kCorruptionDetectionUri; + return uri == RtpExtension::kTimestampOffsetUri || + uri == RtpExtension::kAbsSendTimeUri || + uri == RtpExtension::kAbsoluteCaptureTimeUri || + uri == RtpExtension::kVideoRotationUri || + uri == RtpExtension::kTransportSequenceNumberUri || + uri == RtpExtension::kTransportSequenceNumberV2Uri || + uri == RtpExtension::kPlayoutDelayUri || + uri == RtpExtension::kVideoContentTypeUri || + uri == RtpExtension::kVideoTimingUri || uri == RtpExtension::kMidUri || + uri == RtpExtension::kGenericFrameDescriptorUri00 || + uri == RtpExtension::kDependencyDescriptorUri || + uri == RtpExtension::kColorSpaceUri || uri == RtpExtension::kRidUri || + uri == RtpExtension::kRepairedRidUri || + uri == RtpExtension::kVideoLayersAllocationUri || + uri == RtpExtension::kVideoFrameTrackingIdUri || + uri == RtpExtension::kCorruptionDetectionUri; } bool RtpExtension::IsEncryptionSupported(absl::string_view uri) { @@ -207,7 +204,7 @@ bool RtpExtension::IsEncryptionSupported(absl::string_view uri) { // encrypted (which can't be done by Chromium). uri != webrtc::RtpExtension::kAbsSendTimeUri && #endif - uri != webrtc::RtpExtension::kEncryptHeaderExtensionsUri; + uri != RtpExtension::kEncryptHeaderExtensionsUri; } // Returns whether a header extension with the given URI exists. @@ -228,7 +225,7 @@ const RtpExtension* RtpExtension::FindHeaderExtensionByUri( const std::vector& extensions, absl::string_view uri, Filter filter) { - const webrtc::RtpExtension* fallback_extension = nullptr; + const RtpExtension* fallback_extension = nullptr; for (const auto& extension : extensions) { if (extension.uri != uri) { continue; diff --git a/api/rtp_parameters_unittest.cc b/api/rtp_parameters_unittest.cc index 0dc3cb5bd6..7d26e9d2b0 100644 --- a/api/rtp_parameters_unittest.cc +++ b/api/rtp_parameters_unittest.cc @@ -16,7 +16,6 @@ namespace webrtc { -using webrtc::RtpExtension; static const char kExtensionUri1[] = "extension-uri1"; static const char kExtensionUri2[] = "extension-uri2"; diff --git a/api/sequence_checker_unittest.cc b/api/sequence_checker_unittest.cc index 41eef3a7a7..b6e432c2d3 100644 --- a/api/sequence_checker_unittest.cc +++ b/api/sequence_checker_unittest.cc @@ -44,7 +44,7 @@ class CompileTimeTestForGuardedBy { private: int guarded_ RTC_GUARDED_BY(sequence_checker_); - ::webrtc::SequenceChecker sequence_checker_; + SequenceChecker sequence_checker_; }; void RunOnDifferentThread(FunctionView run) { diff --git a/api/task_queue/task_queue_test.cc b/api/task_queue/task_queue_test.cc index 1366a39cd8..a7c62262e3 100644 --- a/api/task_queue/task_queue_test.cc +++ b/api/task_queue/task_queue_test.cc @@ -36,20 +36,20 @@ void SleepFor(TimeDelta duration) { } std::unique_ptr CreateTaskQueue( - const std::unique_ptr& factory, + const std::unique_ptr& factory, absl::string_view task_queue_name, TaskQueueFactory::Priority priority = TaskQueueFactory::Priority::NORMAL) { return factory->CreateTaskQueue(task_queue_name, priority); } TEST_P(TaskQueueTest, Construct) { - std::unique_ptr factory = GetParam()(nullptr); + std::unique_ptr factory = GetParam()(nullptr); auto queue = CreateTaskQueue(factory, "Construct"); EXPECT_FALSE(queue->IsCurrent()); } TEST_P(TaskQueueTest, PostAndCheckCurrent) { - std::unique_ptr factory = GetParam()(nullptr); + std::unique_ptr factory = GetParam()(nullptr); Event event; auto queue = CreateTaskQueue(factory, "PostAndCheckCurrent"); @@ -67,7 +67,7 @@ TEST_P(TaskQueueTest, PostAndCheckCurrent) { } TEST_P(TaskQueueTest, PostCustomTask) { - std::unique_ptr factory = GetParam()(nullptr); + std::unique_ptr factory = GetParam()(nullptr); Event ran; auto queue = CreateTaskQueue(factory, "PostCustomImplementation"); @@ -86,7 +86,7 @@ TEST_P(TaskQueueTest, PostCustomTask) { } TEST_P(TaskQueueTest, PostDelayedZero) { - std::unique_ptr factory = GetParam()(nullptr); + std::unique_ptr factory = GetParam()(nullptr); Event event; auto queue = CreateTaskQueue(factory, "PostDelayedZero"); @@ -95,7 +95,7 @@ TEST_P(TaskQueueTest, PostDelayedZero) { } TEST_P(TaskQueueTest, PostFromQueue) { - std::unique_ptr factory = GetParam()(nullptr); + std::unique_ptr factory = GetParam()(nullptr); Event event; auto queue = CreateTaskQueue(factory, "PostFromQueue"); @@ -105,7 +105,7 @@ TEST_P(TaskQueueTest, PostFromQueue) { } TEST_P(TaskQueueTest, PostDelayed) { - std::unique_ptr factory = GetParam()(nullptr); + std::unique_ptr factory = GetParam()(nullptr); Event event; auto queue = CreateTaskQueue(factory, "PostDelayed", TaskQueueFactory::Priority::HIGH); @@ -127,7 +127,7 @@ TEST_P(TaskQueueTest, PostDelayed) { } TEST_P(TaskQueueTest, PostMultipleDelayed) { - std::unique_ptr factory = GetParam()(nullptr); + std::unique_ptr factory = GetParam()(nullptr); auto queue = CreateTaskQueue(factory, "PostMultipleDelayed"); std::vector events(100); @@ -146,7 +146,7 @@ TEST_P(TaskQueueTest, PostMultipleDelayed) { } TEST_P(TaskQueueTest, PostDelayedAfterDestruct) { - std::unique_ptr factory = GetParam()(nullptr); + std::unique_ptr factory = GetParam()(nullptr); Event run; Event deleted; auto queue = CreateTaskQueue(factory, "PostDelayedAfterDestruct"); @@ -161,7 +161,7 @@ TEST_P(TaskQueueTest, PostDelayedAfterDestruct) { } TEST_P(TaskQueueTest, PostDelayedHighPrecisionAfterDestruct) { - std::unique_ptr factory = GetParam()(nullptr); + std::unique_ptr factory = GetParam()(nullptr); Event run; Event deleted; auto queue = @@ -178,7 +178,7 @@ TEST_P(TaskQueueTest, PostDelayedHighPrecisionAfterDestruct) { } TEST_P(TaskQueueTest, PostedUnexecutedClosureDestroyedOnTaskQueue) { - std::unique_ptr factory = GetParam()(nullptr); + std::unique_ptr factory = GetParam()(nullptr); auto queue = CreateTaskQueue(factory, "PostedUnexecutedClosureDestroyedOnTaskQueue"); TaskQueueBase* queue_ptr = queue.get(); @@ -198,7 +198,7 @@ TEST_P(TaskQueueTest, PostedUnexecutedClosureDestroyedOnTaskQueue) { } TEST_P(TaskQueueTest, PostedClosureDestroyedOnTaskQueue) { - std::unique_ptr factory = GetParam()(nullptr); + std::unique_ptr factory = GetParam()(nullptr); auto queue = CreateTaskQueue(factory, "PostedClosureDestroyedOnTaskQueue"); TaskQueueBase* queue_ptr = queue.get(); Event finished; @@ -215,7 +215,7 @@ TEST_P(TaskQueueTest, PostedClosureDestroyedOnTaskQueue) { } TEST_P(TaskQueueTest, PostedExecutedClosureDestroyedOnTaskQueue) { - std::unique_ptr factory = GetParam()(nullptr); + std::unique_ptr factory = GetParam()(nullptr); auto queue = CreateTaskQueue(factory, "PostedExecutedClosureDestroyedOnTaskQueue"); TaskQueueBase* queue_ptr = queue.get(); @@ -229,7 +229,7 @@ TEST_P(TaskQueueTest, PostedExecutedClosureDestroyedOnTaskQueue) { } TEST_P(TaskQueueTest, PostAndReuse) { - std::unique_ptr factory = GetParam()(nullptr); + std::unique_ptr factory = GetParam()(nullptr); Event event; auto post_queue = CreateTaskQueue(factory, "PostQueue"); auto reply_queue = CreateTaskQueue(factory, "ReplyQueue"); @@ -274,7 +274,7 @@ TEST_P(TaskQueueTest, PostALot) { explicit BlockingCounter(int initial_count) : count_(initial_count) {} void DecrementCount() { - if (count_.DecRef() == webrtc::RefCountReleaseStatus::kDroppedLastRef) { + if (count_.DecRef() == RefCountReleaseStatus::kDroppedLastRef) { event_.Set(); } } @@ -285,7 +285,7 @@ TEST_P(TaskQueueTest, PostALot) { Event event_; }; - std::unique_ptr factory = GetParam()(nullptr); + std::unique_ptr factory = GetParam()(nullptr); static constexpr int kTaskCount = 0xffff; Event posting_done; BlockingCounter all_destroyed(kTaskCount); @@ -329,7 +329,7 @@ TEST_P(TaskQueueTest, PostALot) { // unit test, run it under TSan or some other tool that is able to // directly detect data races. TEST_P(TaskQueueTest, PostTwoWithSharedUnprotectedState) { - std::unique_ptr factory = GetParam()(nullptr); + std::unique_ptr factory = GetParam()(nullptr); struct SharedState { // First task will set this value to 1 and second will assert it. int state = 0; diff --git a/media/base/adapted_video_track_source.cc b/media/base/adapted_video_track_source.cc index f015956d19..c2ec91b1fd 100644 --- a/media/base/adapted_video_track_source.cc +++ b/media/base/adapted_video_track_source.cc @@ -53,13 +53,13 @@ void AdaptedVideoTrackSource::OnFrame(const VideoFrame& frame) { true was just added. The VideoBroadcaster enforces synchronization for us in this case, by not passing the frame on to sinks which don't want it. */ - if (apply_rotation() && frame.rotation() != webrtc::kVideoRotation_0 && + if (apply_rotation() && frame.rotation() != kVideoRotation_0 && buffer->type() == VideoFrameBuffer::Type::kI420) { /* Apply pending rotation. */ VideoFrame rotated_frame(frame); rotated_frame.set_video_frame_buffer( I420Buffer::Rotate(*buffer->GetI420(), frame.rotation())); - rotated_frame.set_rotation(webrtc::kVideoRotation_0); + rotated_frame.set_rotation(kVideoRotation_0); broadcaster_.OnFrame(rotated_frame); } else { broadcaster_.OnFrame(frame); @@ -109,7 +109,7 @@ bool AdaptedVideoTrackSource::AdaptFrame(int width, } if (!video_adapter_.AdaptFrameResolution( - width, height, time_us * webrtc::kNumNanosecsPerMicrosec, crop_width, + width, height, time_us * kNumNanosecsPerMicrosec, crop_width, crop_height, out_width, out_height)) { broadcaster_.OnDiscardedFrame(); // VideoAdapter dropped the frame. diff --git a/media/base/codec_comparators.cc b/media/base/codec_comparators.cc index 3dbf316ebf..4209d8cfa7 100644 --- a/media/base/codec_comparators.cc +++ b/media/base/codec_comparators.cc @@ -398,14 +398,13 @@ bool IsSameRtpCodecIgnoringLevel(const Codec& codec, // Some video codecs are compatible with others (e.g. same profile but // different level). This comparison looks at the relevant parameters, // ignoring ones that are either irrelevant or unrecognized. - if (rtp_codec.kind == webrtc::MediaType::VIDEO && rtp_codec.IsMediaCodec()) { + if (rtp_codec.kind == MediaType::VIDEO && rtp_codec.IsMediaCodec()) { return IsSameCodecSpecific(rtp_codec.name, params1, rtp_codec2.name, params2); } // audio/RED should ignore the parameters which specify payload types so // can not be compared. - if (rtp_codec.kind == webrtc::MediaType::AUDIO && - rtp_codec.name == kRedCodecName) { + if (rtp_codec.kind == MediaType::AUDIO && rtp_codec.name == kRedCodecName) { return true; } diff --git a/media/base/codec_comparators_unittest.cc b/media/base/codec_comparators_unittest.cc index 214bff85a7..b7dde12dec 100644 --- a/media/base/codec_comparators_unittest.cc +++ b/media/base/codec_comparators_unittest.cc @@ -390,8 +390,7 @@ TEST(CodecTest, TestOpusAudioCodecWithDifferentParameters) { // Matches does not compare parameters for audio. EXPECT_TRUE(opus_with_fec.Matches(opus_without_fec)); - webrtc::RtpCodecParameters rtp_opus_with_fec = - opus_with_fec.ToCodecParameters(); + RtpCodecParameters rtp_opus_with_fec = opus_with_fec.ToCodecParameters(); // MatchesRtpCodec takes parameters into account. EXPECT_TRUE(opus_with_fec.MatchesRtpCodec(rtp_opus_with_fec)); EXPECT_FALSE(opus_without_fec.MatchesRtpCodec(rtp_opus_with_fec)); @@ -480,19 +479,19 @@ TEST(CodecTest, TestVP9CodecMatches) { Codec c_no_profile = CreateVideoCodec(95, kVp9CodecName); Codec c_profile0 = CreateVideoCodec(95, kVp9CodecName); - c_profile0.params[webrtc::kVP9FmtpProfileId] = kProfile0; + c_profile0.params[kVP9FmtpProfileId] = kProfile0; EXPECT_TRUE(c_profile0.Matches(c_no_profile)); { Codec c_profile0_eq = CreateVideoCodec(95, kVp9CodecName); - c_profile0_eq.params[webrtc::kVP9FmtpProfileId] = kProfile0; + c_profile0_eq.params[kVP9FmtpProfileId] = kProfile0; EXPECT_TRUE(c_profile0.Matches(c_profile0_eq)); } { Codec c_profile2 = CreateVideoCodec(95, kVp9CodecName); - c_profile2.params[webrtc::kVP9FmtpProfileId] = kProfile2; + c_profile2.params[kVP9FmtpProfileId] = kProfile2; EXPECT_FALSE(c_profile0.Matches(c_profile2)); EXPECT_FALSE(c_no_profile.Matches(c_profile2)); } diff --git a/media/base/codec_list.cc b/media/base/codec_list.cc index 27183a0279..30c6c5f16a 100644 --- a/media/base/codec_list.cc +++ b/media/base/codec_list.cc @@ -23,9 +23,6 @@ namespace webrtc { -using webrtc::RTCError; -using webrtc::RTCErrorOr; -using webrtc::RTCErrorType; namespace { @@ -46,7 +43,7 @@ RTCError CheckInputConsistency(const std::vector& codecs) { } } } - for (const webrtc::Codec& codec : codecs) { + for (const Codec& codec : codecs) { switch (codec.GetResiliencyType()) { case Codec::ResiliencyType::kRed: // Check that the target codec exists @@ -74,7 +71,7 @@ RTCError CheckInputConsistency(const std::vector& codecs) { break; } int associated_pt; - if (!(webrtc::FromString(apt_it->second, &associated_pt))) { + if (!(FromString(apt_it->second, &associated_pt))) { RTC_LOG(LS_ERROR) << "Non-numeric argument to rtx apt: " << codec << " apt=" << apt_it->second; LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, diff --git a/media/base/codec_list_unittest.cc b/media/base/codec_list_unittest.cc index 161b59d788..b7bc1dde47 100644 --- a/media/base/codec_list_unittest.cc +++ b/media/base/codec_list_unittest.cc @@ -22,8 +22,6 @@ namespace webrtc { namespace { -using webrtc::RTCErrorOr; -using webrtc::RTCErrorType; TEST(CodecList, StoreAndRecall) { CodecList empty_list = CodecList::CreateFromTrustedData(std::vector{}); @@ -36,7 +34,7 @@ TEST(CodecList, StoreAndRecall) { } TEST(CodecList, RejectIllegalConstructorArguments) { - std::vector apt_without_number{webrtc::CreateVideoCodec( + std::vector apt_without_number{CreateVideoCodec( {SdpVideoFormat{"rtx", CodecParameterMap{{"apt", "not-a-number"}}}})}; apt_without_number[0].id = 96; RTCErrorOr checked_codec_list = @@ -54,7 +52,7 @@ TEST(CodecList, CrashOnIllegalConstructorArguments) { // Checks that can't be enabled log things instead. // Note: DCHECK is on in some release builds, so we can't use // EXPECT_DEBUG_DEATH here. - std::vector apt_without_number{webrtc::CreateVideoCodec( + std::vector apt_without_number{CreateVideoCodec( {SdpVideoFormat{"rtx", CodecParameterMap{{"apt", "not-a-number"}}}})}; apt_without_number[0].id = 96; #if RTC_DCHECK_IS_ON diff --git a/media/base/codec_unittest.cc b/media/base/codec_unittest.cc index e00b08d8c2..3a1bfef713 100644 --- a/media/base/codec_unittest.cc +++ b/media/base/codec_unittest.cc @@ -254,9 +254,9 @@ TEST(CodecTest, TestValidateCodecFormat) { TEST(CodecTest, TestToCodecParameters) { Codec v = CreateVideoCodec(96, "V"); v.SetParam("p1", "v1"); - webrtc::RtpCodecParameters codec_params_1 = v.ToCodecParameters(); + RtpCodecParameters codec_params_1 = v.ToCodecParameters(); EXPECT_EQ(96, codec_params_1.payload_type); - EXPECT_EQ(webrtc::MediaType::VIDEO, codec_params_1.kind); + EXPECT_EQ(MediaType::VIDEO, codec_params_1.kind); EXPECT_EQ("V", codec_params_1.name); EXPECT_EQ(kVideoCodecClockrate, codec_params_1.clock_rate); EXPECT_EQ(std::nullopt, codec_params_1.num_channels); @@ -266,9 +266,9 @@ TEST(CodecTest, TestToCodecParameters) { Codec a = CreateAudioCodec(97, "A", 44100, 2); a.SetParam("p1", "a1"); - webrtc::RtpCodecParameters codec_params_2 = a.ToCodecParameters(); + RtpCodecParameters codec_params_2 = a.ToCodecParameters(); EXPECT_EQ(97, codec_params_2.payload_type); - EXPECT_EQ(webrtc::MediaType::AUDIO, codec_params_2.kind); + EXPECT_EQ(MediaType::AUDIO, codec_params_2.kind); EXPECT_EQ("A", codec_params_2.name); EXPECT_EQ(44100, codec_params_2.clock_rate); EXPECT_EQ(2, codec_params_2.num_channels); @@ -278,22 +278,21 @@ TEST(CodecTest, TestToCodecParameters) { } TEST(CodecTest, H264CostrainedBaselineIsAddedIfH264IsSupported) { - const std::vector kExplicitlySupportedFormats = { - webrtc::CreateH264Format(webrtc::H264Profile::kProfileBaseline, - webrtc::H264Level::kLevel3_1, "1"), - webrtc::CreateH264Format(webrtc::H264Profile::kProfileBaseline, - webrtc::H264Level::kLevel3_1, "0")}; - - std::vector supported_formats = - kExplicitlySupportedFormats; + const std::vector kExplicitlySupportedFormats = { + CreateH264Format(H264Profile::kProfileBaseline, H264Level::kLevel3_1, + "1"), + CreateH264Format(H264Profile::kProfileBaseline, H264Level::kLevel3_1, + "0")}; + + std::vector supported_formats = kExplicitlySupportedFormats; AddH264ConstrainedBaselineProfileToSupportedFormats(&supported_formats); - const webrtc::SdpVideoFormat kH264ConstrainedBasedlinePacketization1 = - webrtc::CreateH264Format(webrtc::H264Profile::kProfileConstrainedBaseline, - webrtc::H264Level::kLevel3_1, "1"); - const webrtc::SdpVideoFormat kH264ConstrainedBasedlinePacketization0 = - webrtc::CreateH264Format(webrtc::H264Profile::kProfileConstrainedBaseline, - webrtc::H264Level::kLevel3_1, "0"); + const SdpVideoFormat kH264ConstrainedBasedlinePacketization1 = + CreateH264Format(H264Profile::kProfileConstrainedBaseline, + H264Level::kLevel3_1, "1"); + const SdpVideoFormat kH264ConstrainedBasedlinePacketization0 = + CreateH264Format(H264Profile::kProfileConstrainedBaseline, + H264Level::kLevel3_1, "0"); EXPECT_EQ(supported_formats[0], kExplicitlySupportedFormats[0]); EXPECT_EQ(supported_formats[1], kExplicitlySupportedFormats[1]); @@ -302,13 +301,11 @@ TEST(CodecTest, H264CostrainedBaselineIsAddedIfH264IsSupported) { } TEST(CodecTest, H264CostrainedBaselineIsNotAddedIfH264IsUnsupported) { - const std::vector kExplicitlySupportedFormats = { + const std::vector kExplicitlySupportedFormats = { {kVp9CodecName, - {{webrtc::kVP9FmtpProfileId, - VP9ProfileToString(webrtc::VP9Profile::kProfile0)}}}}; + {{kVP9FmtpProfileId, VP9ProfileToString(VP9Profile::kProfile0)}}}}; - std::vector supported_formats = - kExplicitlySupportedFormats; + std::vector supported_formats = kExplicitlySupportedFormats; AddH264ConstrainedBaselineProfileToSupportedFormats(&supported_formats); EXPECT_EQ(supported_formats[0], kExplicitlySupportedFormats[0]); @@ -316,18 +313,17 @@ TEST(CodecTest, H264CostrainedBaselineIsNotAddedIfH264IsUnsupported) { } TEST(CodecTest, H264CostrainedBaselineNotAddedIfAlreadySpecified) { - const std::vector kExplicitlySupportedFormats = { - webrtc::CreateH264Format(webrtc::H264Profile::kProfileBaseline, - webrtc::H264Level::kLevel3_1, "1"), - webrtc::CreateH264Format(webrtc::H264Profile::kProfileBaseline, - webrtc::H264Level::kLevel3_1, "0"), - webrtc::CreateH264Format(webrtc::H264Profile::kProfileConstrainedBaseline, - webrtc::H264Level::kLevel3_1, "1"), - webrtc::CreateH264Format(webrtc::H264Profile::kProfileConstrainedBaseline, - webrtc::H264Level::kLevel3_1, "0")}; - - std::vector supported_formats = - kExplicitlySupportedFormats; + const std::vector kExplicitlySupportedFormats = { + CreateH264Format(H264Profile::kProfileBaseline, H264Level::kLevel3_1, + "1"), + CreateH264Format(H264Profile::kProfileBaseline, H264Level::kLevel3_1, + "0"), + CreateH264Format(H264Profile::kProfileConstrainedBaseline, + H264Level::kLevel3_1, "1"), + CreateH264Format(H264Profile::kProfileConstrainedBaseline, + H264Level::kLevel3_1, "0")}; + + std::vector supported_formats = kExplicitlySupportedFormats; AddH264ConstrainedBaselineProfileToSupportedFormats(&supported_formats); EXPECT_EQ(supported_formats[0], kExplicitlySupportedFormats[0]); diff --git a/media/base/fake_frame_source.cc b/media/base/fake_frame_source.cc index 41a014313f..7c87553985 100644 --- a/media/base/fake_frame_source.cc +++ b/media/base/fake_frame_source.cc @@ -48,12 +48,12 @@ void FakeFrameSource::SetRotation(VideoRotation rotation) { VideoFrame FakeFrameSource::GetFrameRotationApplied() { switch (rotation_) { - case webrtc::kVideoRotation_0: - case webrtc::kVideoRotation_180: - return GetFrame(width_, height_, webrtc::kVideoRotation_0, interval_us_); - case webrtc::kVideoRotation_90: - case webrtc::kVideoRotation_270: - return GetFrame(height_, width_, webrtc::kVideoRotation_0, interval_us_); + case kVideoRotation_0: + case kVideoRotation_180: + return GetFrame(width_, height_, kVideoRotation_0, interval_us_); + case kVideoRotation_90: + case kVideoRotation_270: + return GetFrame(height_, width_, kVideoRotation_0, interval_us_); } RTC_DCHECK_NOTREACHED() << "Invalid rotation value: " << static_cast(rotation_); diff --git a/media/base/fake_media_engine.cc b/media/base/fake_media_engine.cc index 5485c02441..35e8f137d9 100644 --- a/media/base/fake_media_engine.cc +++ b/media/base/fake_media_engine.cc @@ -47,7 +47,6 @@ #include "rtc_base/system/file_wrapper.h" namespace webrtc { -using webrtc::TaskQueueBase; FakeVoiceMediaReceiveChannel::DtmfInfo::DtmfInfo(uint32_t ssrc, int event_code, @@ -561,12 +560,11 @@ bool FakeVideoMediaReceiveChannel::GetStats(VideoMediaReceiveInfo* /* info */) { } FakeVoiceEngine::FakeVoiceEngine() - : encoder_factory_(webrtc::make_ref_counted(this)), - decoder_factory_( - webrtc::make_ref_counted(this)) { + : encoder_factory_(make_ref_counted(this)), + decoder_factory_(make_ref_counted(this)) { // Add a fake audio codec. Note that the name must not be "" as there are // sanity checks against that. - SetCodecs({webrtc::CreateAudioCodec(101, "fake_audio_codec", 8000, 1)}); + SetCodecs({CreateAudioCodec(101, "fake_audio_codec", 8000, 1)}); } void FakeVoiceEngine::Init() {} scoped_refptr FakeVoiceEngine::GetAudioState() const { @@ -635,8 +633,8 @@ void FakeVoiceEngine::SetRtpHeaderExtensions( FakeVideoEngine::FakeVideoEngine() : capture_(false) { // Add a fake video codec. Note that the name must not be "" as there are // sanity checks against that. - send_codecs_.push_back(webrtc::CreateVideoCodec(111, "fake_video_codec")); - recv_codecs_.push_back(webrtc::CreateVideoCodec(111, "fake_video_codec")); + send_codecs_.push_back(CreateVideoCodec(111, "fake_video_codec")); + recv_codecs_.push_back(CreateVideoCodec(111, "fake_video_codec")); } bool FakeVideoEngine::SetOptions(const VideoOptions& options) { options_ = options; diff --git a/media/base/media_engine.cc b/media/base/media_engine.cc index f1f5173027..e9d9c6c72a 100644 --- a/media/base/media_engine.cc +++ b/media/base/media_engine.cc @@ -40,10 +40,9 @@ bool SupportsMode(const Codec& codec, if (!scalability_mode.has_value()) { return true; } - return absl::c_any_of( - codec.scalability_modes, [&](webrtc::ScalabilityMode mode) { - return ScalabilityModeToString(mode) == *scalability_mode; - }); + return absl::c_any_of(codec.scalability_modes, [&](ScalabilityMode mode) { + return ScalabilityModeToString(mode) == *scalability_mode; + }); } } // namespace @@ -90,8 +89,6 @@ std::vector GetDefaultEnabledRtpHeaderExtensions( RTCError CheckScalabilityModeValues(const RtpParameters& rtp_parameters, ArrayView send_codecs, std::optional send_codec) { - using webrtc::RTCErrorType; - if (send_codecs.empty()) { // This is an audio sender or an extra check in the stack where the codec // list is not available and we can't check the scalability_mode values. @@ -101,7 +98,7 @@ RTCError CheckScalabilityModeValues(const RtpParameters& rtp_parameters, for (size_t i = 0; i < rtp_parameters.encodings.size(); ++i) { if (rtp_parameters.encodings[i].codec) { bool codecFound = false; - for (const webrtc::Codec& codec : send_codecs) { + for (const Codec& codec : send_codecs) { if (IsSameRtpCodecIgnoringLevel(codec, *rtp_parameters.encodings[i].codec) && SupportsMode(codec, rtp_parameters.encodings[i].scalability_mode)) { @@ -120,7 +117,7 @@ RTCError CheckScalabilityModeValues(const RtpParameters& rtp_parameters, if (rtp_parameters.encodings[i].scalability_mode) { if (!send_codec) { bool scalabilityModeFound = false; - for (const webrtc::Codec& codec : send_codecs) { + for (const Codec& codec : send_codecs) { for (const auto& scalability_mode : codec.scalability_modes) { if (ScalabilityModeToString(scalability_mode) == *rtp_parameters.encodings[i].scalability_mode) { @@ -164,8 +161,6 @@ RTCError CheckRtpParametersValues(const RtpParameters& rtp_parameters, ArrayView send_codecs, std::optional send_codec, const FieldTrialsView& field_trials) { - using webrtc::RTCErrorType; - bool has_scale_resolution_down_to = false; for (size_t i = 0; i < rtp_parameters.encodings.size(); ++i) { if (rtp_parameters.encodings[i].bitrate_priority <= 0) { @@ -198,7 +193,7 @@ RTCError CheckRtpParametersValues(const RtpParameters& rtp_parameters, if (rtp_parameters.encodings[i].num_temporal_layers) { if (*rtp_parameters.encodings[i].num_temporal_layers < 1 || *rtp_parameters.encodings[i].num_temporal_layers > - webrtc::kMaxTemporalStreams) { + kMaxTemporalStreams) { LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE, "Attempted to set RtpParameters " "num_temporal_layers to an invalid number."); @@ -252,7 +247,6 @@ RTCError CheckRtpParametersInvalidModificationAndValues( ArrayView send_codecs, std::optional send_codec, const FieldTrialsView& field_trials) { - using webrtc::RTCErrorType; if (rtp_parameters.encodings.size() != old_rtp_parameters.encodings.size()) { LOG_AND_RETURN_ERROR( RTCErrorType::INVALID_MODIFICATION, diff --git a/media/base/rtp_utils.cc b/media/base/rtp_utils.cc index 4a905b2fe7..7b4fe196dd 100644 --- a/media/base/rtp_utils.cc +++ b/media/base/rtp_utils.cc @@ -102,10 +102,10 @@ void UpdateRtpAuthTag(uint8_t* rtp, size_t auth_required_length = length - tag_length + kRocLength; uint8_t output[64]; - size_t result = webrtc::ComputeHmac( - webrtc::DIGEST_SHA_1, &packet_time_params.srtp_auth_key[0], - packet_time_params.srtp_auth_key.size(), rtp, auth_required_length, - output, sizeof(output)); + size_t result = + ComputeHmac(DIGEST_SHA_1, &packet_time_params.srtp_auth_key[0], + packet_time_params.srtp_auth_key.size(), rtp, + auth_required_length, output, sizeof(output)); if (result < tag_length) { RTC_DCHECK_NOTREACHED(); @@ -147,7 +147,7 @@ bool GetRtcpSsrc(const void* data, size_t len, uint32_t* value) { // SDES packet parsing is not supported. if (pl_type == kRtcpTypeSDES) return false; - *value = webrtc::GetBE32(static_cast(data) + 4); + *value = GetBE32(static_cast(data) + 4); return true; } @@ -176,10 +176,10 @@ absl::string_view RtpPacketTypeToString(RtpPacketType packet_type) { } RtpPacketType InferRtpPacketType(ArrayView packet) { - if (webrtc::IsRtcpPacket(packet)) { + if (IsRtcpPacket(packet)) { return RtpPacketType::kRtcp; } - if (webrtc::IsRtpPacket(packet)) { + if (IsRtpPacket(packet)) { return RtpPacketType::kRtp; } return RtpPacketType::kUnknown; @@ -219,7 +219,7 @@ bool ValidateRtpHeader(const uint8_t* rtp, // Getting extension profile length. // Length is in 32 bit words. - uint16_t extension_length_in_32bits = webrtc::GetBE16(rtp + 2); + uint16_t extension_length_in_32bits = GetBE16(rtp + 2); size_t extension_length = extension_length_in_32bits * 4; size_t rtp_header_length = extension_length + @@ -267,9 +267,9 @@ bool UpdateRtpAbsSendTimeExtension(uint8_t* rtp, rtp += header_length_without_extension; // Getting extension profile ID and length. - uint16_t profile_id = webrtc::GetBE16(rtp); + uint16_t profile_id = GetBE16(rtp); // Length is in 32 bit words. - uint16_t extension_length_in_32bits = webrtc::GetBE16(rtp + 2); + uint16_t extension_length_in_32bits = GetBE16(rtp + 2); size_t extension_length = extension_length_in_32bits * 4; rtp += kRtpExtensionHeaderLen; // Moving past extension header. @@ -373,14 +373,14 @@ bool ApplyPacketOptions(uint8_t* data, // indication. size_t rtp_start_pos; size_t rtp_length; - if (!webrtc::UnwrapTurnPacket(data, length, &rtp_start_pos, &rtp_length)) { + if (!UnwrapTurnPacket(data, length, &rtp_start_pos, &rtp_length)) { RTC_DCHECK_NOTREACHED(); return false; } // Making sure we have a valid RTP packet at the end. auto packet = MakeArrayView(data + rtp_start_pos, rtp_length); - if (!webrtc::IsRtpPacket(packet) || + if (!IsRtpPacket(packet) || !ValidateRtpHeader(data + rtp_start_pos, rtp_length, nullptr)) { RTC_DCHECK_NOTREACHED(); return false; diff --git a/media/base/turn_utils.cc b/media/base/turn_utils.cc index 4e4783f965..d006bb892d 100644 --- a/media/base/turn_utils.cc +++ b/media/base/turn_utils.cc @@ -31,7 +31,7 @@ bool IsTurnSendIndicationPacket(const uint8_t* data, size_t length) { return false; } - uint16_t type = webrtc::GetBE16(data); + uint16_t type = GetBE16(data); return (type == TURN_SEND_INDICATION); } @@ -52,7 +52,7 @@ bool UnwrapTurnPacket(const uint8_t* packet, // / Application Data / // / / // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - size_t length = webrtc::GetBE16(&packet[2]); + size_t length = GetBE16(&packet[2]); if (length + kTurnChannelHeaderLength > packet_size) { return false; } @@ -64,7 +64,7 @@ bool UnwrapTurnPacket(const uint8_t* packet, if (IsTurnSendIndicationPacket(packet, packet_size)) { // Validate STUN message length. - const size_t stun_message_length = webrtc::GetBE16(&packet[2]); + const size_t stun_message_length = GetBE16(&packet[2]); if (stun_message_length + kStunHeaderSize != packet_size) { return false; } @@ -94,8 +94,8 @@ bool UnwrapTurnPacket(const uint8_t* packet, } // Getting attribute type and length. - attr_type = webrtc::GetBE16(&packet[pos]); - attr_length = webrtc::GetBE16(&packet[pos + sizeof(attr_type)]); + attr_type = GetBE16(&packet[pos]); + attr_length = GetBE16(&packet[pos + sizeof(attr_type)]); pos += kAttrHeaderLength; // Skip STUN_DATA_ATTR header. diff --git a/media/base/video_adapter.cc b/media/base/video_adapter.cc index a4f88145f4..2b5dcf6513 100644 --- a/media/base/video_adapter.cc +++ b/media/base/video_adapter.cc @@ -309,7 +309,7 @@ void VideoAdapter::OnOutputFormatRequest( target_aspect_ratio = std::make_pair(format->width, format->height); max_pixel_count = format->width * format->height; if (format->interval > 0) - max_fps = webrtc::kNumNanosecsPerSec / format->interval; + max_fps = kNumNanosecsPerSec / format->interval; } OnOutputFormatRequest(target_aspect_ratio, max_pixel_count, max_fps); } diff --git a/media/base/video_adapter_unittest.cc b/media/base/video_adapter_unittest.cc index 135afa5f4a..98b699657e 100644 --- a/media/base/video_adapter_unittest.cc +++ b/media/base/video_adapter_unittest.cc @@ -35,10 +35,10 @@ constexpr int kWidth = 1280; constexpr int kHeight = 720; constexpr int kDefaultFps = 30; +using test::ScopedFieldTrials; using ::testing::_; using ::testing::Eq; using ::testing::Pair; -using ::webrtc::test::ScopedFieldTrials; VideoSinkWants BuildSinkWants(std::optional target_pixel_count, int max_pixel_count, @@ -84,8 +84,7 @@ class VideoAdapterTest : public ::testing::Test, frame_source_(std::make_unique( kWidth, kHeight, - VideoFormat::FpsToInterval(kDefaultFps) / - webrtc::kNumNanosecsPerMicrosec)), + VideoFormat::FpsToInterval(kDefaultFps) / kNumNanosecsPerMicrosec)), adapter_(source_resolution_alignment), adapter_wrapper_(std::make_unique(&adapter_)), use_new_format_request_(GetParam()) {} @@ -117,8 +116,8 @@ class VideoAdapterTest : public ::testing::Test, int out_height; if (video_adapter_->AdaptFrameResolution( in_width, in_height, - frame.timestamp_us() * webrtc::kNumNanosecsPerMicrosec, - &cropped_width, &cropped_height, &out_width, &out_height)) { + frame.timestamp_us() * kNumNanosecsPerMicrosec, &cropped_width, + &cropped_height, &out_width, &out_height)) { stats_.cropped_width = cropped_width; stats_.cropped_height = cropped_height; stats_.out_width = out_width; @@ -164,7 +163,7 @@ class VideoAdapterTest : public ::testing::Test, } adapter_.OnOutputFormatRequest( VideoFormat(width, height, fps ? VideoFormat::FpsToInterval(*fps) : 0, - webrtc::FOURCC_I420)); + FOURCC_I420)); } // Return pair of @@ -1058,7 +1057,7 @@ TEST(VideoAdapterTestMultipleOrientation, TestNormal) { EXPECT_TRUE(video_adapter.AdaptFrameResolution( /* in_width= */ 480, /* in_height= */ 640, - /* in_timestamp_ns= */ webrtc::kNumNanosecsPerSec / 30, &cropped_width, + /* in_timestamp_ns= */ kNumNanosecsPerSec / 30, &cropped_width, &cropped_height, &out_width, &out_height)); EXPECT_EQ(360, cropped_width); EXPECT_EQ(640, cropped_height); @@ -1086,7 +1085,7 @@ TEST(VideoAdapterTestMultipleOrientation, TestForcePortrait) { EXPECT_TRUE(video_adapter.AdaptFrameResolution( /* in_width= */ 480, /* in_height= */ 640, - /* in_timestamp_ns= */ webrtc::kNumNanosecsPerSec / 30, &cropped_width, + /* in_timestamp_ns= */ kNumNanosecsPerSec / 30, &cropped_width, &cropped_height, &out_width, &out_height)); EXPECT_EQ(360, cropped_width); EXPECT_EQ(640, cropped_height); @@ -1195,8 +1194,8 @@ TEST_P(VideoAdapterTest, AdaptResolutionWithSinkAlignment) { std::numeric_limits::max(), sink_alignment)); EXPECT_TRUE(adapter_.AdaptFrameResolution( kSourceWidth, kSourceHeight, - frame_num * webrtc::kNumNanosecsPerSec / kSourceFramerate, - &cropped_width_, &cropped_height_, &out_width_, &out_height_)); + frame_num * kNumNanosecsPerSec / kSourceFramerate, &cropped_width_, + &cropped_height_, &out_width_, &out_height_)); EXPECT_EQ(out_width_ % sink_alignment, 0); EXPECT_EQ(out_height_ % sink_alignment, 0); diff --git a/media/base/video_broadcaster.cc b/media/base/video_broadcaster.cc index 7f5cf85958..ac26324e31 100644 --- a/media/base/video_broadcaster.cc +++ b/media/base/video_broadcaster.cc @@ -74,7 +74,7 @@ void VideoBroadcaster::OnFrame(const VideoFrame& frame) { bool current_frame_was_discarded = false; for (auto& sink_pair : sink_pairs()) { if (sink_pair.wants.rotation_applied && - frame.rotation() != webrtc::kVideoRotation_0) { + frame.rotation() != kVideoRotation_0) { // Calls to OnFrame are not synchronized with changes to the sink wants. // When rotation_applied is set to true, one or a few frames may get here // with rotation still pending. Protect sinks that don't expect any diff --git a/media/base/video_common_unittest.cc b/media/base/video_common_unittest.cc index a813aa3fa7..37b5c5521c 100644 --- a/media/base/video_common_unittest.cc +++ b/media/base/video_common_unittest.cc @@ -40,8 +40,8 @@ TEST(VideoCommonTest, TestCanonicalFourCC) { // Test conversion between interval and fps TEST(VideoCommonTest, TestVideoFormatFps) { EXPECT_EQ(VideoFormat::kMinimumInterval, VideoFormat::FpsToInterval(0)); - EXPECT_EQ(webrtc::kNumNanosecsPerSec / 20, VideoFormat::FpsToInterval(20)); - EXPECT_EQ(20, VideoFormat::IntervalToFps(webrtc::kNumNanosecsPerSec / 20)); + EXPECT_EQ(kNumNanosecsPerSec / 20, VideoFormat::FpsToInterval(20)); + EXPECT_EQ(20, VideoFormat::IntervalToFps(kNumNanosecsPerSec / 20)); EXPECT_EQ(0, VideoFormat::IntervalToFps(0)); } diff --git a/media/engine/fake_video_codec_factory.cc b/media/engine/fake_video_codec_factory.cc index 70811f9568..d7b9b2e24f 100644 --- a/media/engine/fake_video_codec_factory.cc +++ b/media/engine/fake_video_codec_factory.cc @@ -32,11 +32,10 @@ namespace webrtc { std::vector FakeVideoEncoderFactory::GetSupportedFormats() const { - const absl::InlinedVector - kSupportedScalabilityModes = {webrtc::ScalabilityMode::kL1T1, - webrtc::ScalabilityMode::kL1T2, - webrtc::ScalabilityMode::kL1T3}; + const absl::InlinedVector + kSupportedScalabilityModes = {ScalabilityMode::kL1T1, + ScalabilityMode::kL1T2, + ScalabilityMode::kL1T3}; return std::vector( 1, SdpVideoFormat(kFakeCodecFactoryCodecName, {}, diff --git a/media/engine/fake_webrtc_call.cc b/media/engine/fake_webrtc_call.cc index fb8d985a14..523ce610c4 100644 --- a/media/engine/fake_webrtc_call.cc +++ b/media/engine/fake_webrtc_call.cc @@ -58,8 +58,6 @@ namespace webrtc { -using ::webrtc::Environment; -using ::webrtc::ParseRtpSsrc; FakeAudioSendStream::FakeAudioSendStream(int id, const AudioSendStream::Config& config) @@ -68,7 +66,7 @@ FakeAudioSendStream::FakeAudioSendStream(int id, void FakeAudioSendStream::Reconfigure(const AudioSendStream::Config& config, SetParametersCallback callback) { config_ = config; - webrtc::InvokeSetParametersCallback(callback, RTCError::OK()); + InvokeSetParametersCallback(callback, RTCError::OK()); } const AudioSendStream::Config& FakeAudioSendStream::GetConfig() const { @@ -353,7 +351,7 @@ void FakeVideoSendStream::ReconfigureVideoEncoder( codec_settings_set_ = config.encoder_specific_settings != nullptr; encoder_config_ = std::move(config); ++num_encoder_reconfigurations_; - webrtc::InvokeSetParametersCallback(callback, RTCError::OK()); + InvokeSetParametersCallback(callback, RTCError::OK()); } void FakeVideoSendStream::Start() { @@ -467,8 +465,8 @@ FakeCall::FakeCall(const Environment& env, : env_(env), network_thread_(network_thread), worker_thread_(worker_thread), - audio_network_state_(webrtc::kNetworkUp), - video_network_state_(webrtc::kNetworkUp), + audio_network_state_(kNetworkUp), + video_network_state_(kNetworkUp), num_created_send_streams_(0), num_created_receive_streams_(0) {} @@ -537,13 +535,13 @@ NetworkState FakeCall::GetNetworkState(MediaType media) const { case MediaType::ANY: case MediaType::UNSUPPORTED: ADD_FAILURE() << "GetNetworkState called with unknown parameter."; - return webrtc::kNetworkDown; + return kNetworkDown; } // Even though all the values for the enum class are listed above,the compiler // will emit a warning as the method may be called with a value outside of the // valid enum range, unless this case is also handled. ADD_FAILURE() << "GetNetworkState called with unknown parameter."; - return webrtc::kNetworkDown; + return kNetworkDown; } AudioSendStream* FakeCall::CreateAudioSendStream( diff --git a/media/engine/fake_webrtc_video_engine.cc b/media/engine/fake_webrtc_video_engine.cc index 89a536d449..b016da355d 100644 --- a/media/engine/fake_webrtc_video_engine.cc +++ b/media/engine/fake_webrtc_video_engine.cc @@ -42,7 +42,6 @@ namespace webrtc { namespace { -using ::webrtc::Environment; static constexpr TimeDelta kEventTimeout = TimeDelta::Seconds(10); @@ -103,7 +102,7 @@ std::vector FakeWebRtcVideoDecoderFactory::GetSupportedFormats() const { std::vector formats; - for (const webrtc::SdpVideoFormat& format : supported_codec_formats_) { + for (const SdpVideoFormat& format : supported_codec_formats_) { // We need to test erroneous scenarios, so just warn if there's // a duplicate. if (format.IsCodecInList(formats)) { @@ -232,7 +231,7 @@ std::vector FakeWebRtcVideoEncoderFactory::GetSupportedFormats() const { std::vector formats; - for (const webrtc::SdpVideoFormat& format : formats_) { + for (const SdpVideoFormat& format : formats_) { // Don't add same codec twice. if (!format.IsCodecInList(formats)) formats.push_back(format); diff --git a/media/engine/internal_decoder_factory_unittest.cc b/media/engine/internal_decoder_factory_unittest.cc index a7cd2762dc..17c3b1381f 100644 --- a/media/engine/internal_decoder_factory_unittest.cc +++ b/media/engine/internal_decoder_factory_unittest.cc @@ -28,7 +28,7 @@ using ::testing::Contains; using ::testing::Field; using ::testing::Not; -using ::webrtc::field_trial::InitFieldTrialsFromString; +using field_trial::InitFieldTrialsFromString; #ifdef RTC_ENABLE_VP9 constexpr bool kVp9Enabled = true; diff --git a/media/engine/internal_encoder_factory.cc b/media/engine/internal_encoder_factory.cc index a00c056f0a..9c02ac13bf 100644 --- a/media/engine/internal_encoder_factory.cc +++ b/media/engine/internal_encoder_factory.cc @@ -33,14 +33,14 @@ namespace webrtc { namespace { using Factory = - VideoEncoderFactoryTemplate; + LibvpxVp9EncoderTemplateAdapter>; } // namespace std::vector InternalEncoderFactory::GetSupportedFormats() diff --git a/media/engine/simulcast_encoder_adapter.cc b/media/engine/simulcast_encoder_adapter.cc index 0a1e108c8a..ebd0e1b473 100644 --- a/media/engine/simulcast_encoder_adapter.cc +++ b/media/engine/simulcast_encoder_adapter.cc @@ -115,7 +115,7 @@ int VerifyCodec(const VideoCodec* codec_settings) { if (codec_settings->width <= 1 || codec_settings->height <= 1) { return WEBRTC_VIDEO_CODEC_ERR_PARAMETER; } - if (codec_settings->codecType == webrtc::kVideoCodecVP8 && + if (codec_settings->codecType == kVideoCodecVP8 && codec_settings->VP8().automaticResizeOn && CountActiveStreams(*codec_settings) > 1) { return WEBRTC_VIDEO_CODEC_ERR_PARAMETER; @@ -591,7 +591,7 @@ int SimulcastEncoderAdapter::Encode( // TODO(ilnik): Consider scaling UpdateRect together with the buffer. VideoFrame frame(input_image); frame.set_video_frame_buffer(dst_buffer); - frame.set_rotation(webrtc::kVideoRotation_0); + frame.set_rotation(kVideoRotation_0); frame.set_update_rect( VideoFrame::UpdateRect{0, 0, frame.width(), frame.height()}); int ret = layer.encoder().Encode(frame, &stream_frame_types); @@ -799,13 +799,13 @@ SimulcastEncoderAdapter::FetchOrCreateEncoderContext( return encoder_context; } -webrtc::VideoCodec SimulcastEncoderAdapter::MakeStreamCodec( - const webrtc::VideoCodec& codec, +VideoCodec SimulcastEncoderAdapter::MakeStreamCodec( + const VideoCodec& codec, int stream_idx, uint32_t start_bitrate_kbps, bool is_lowest_quality_stream, bool is_highest_quality_stream) { - webrtc::VideoCodec codec_params = codec; + VideoCodec codec_params = codec; const SimulcastStream& stream_params = codec.simulcastStream[stream_idx]; codec_params.numberOfSimulcastStreams = 0; @@ -845,7 +845,7 @@ webrtc::VideoCodec SimulcastEncoderAdapter::MakeStreamCodec( codec_params.qpMax = kLowestResMaxQp; } } - if (codec.codecType == webrtc::kVideoCodecVP8) { + if (codec.codecType == kVideoCodecVP8) { codec_params.VP8()->numberOfTemporalLayers = stream_params.numberOfTemporalLayers; if (!is_highest_quality_stream) { @@ -854,15 +854,15 @@ webrtc::VideoCodec SimulcastEncoderAdapter::MakeStreamCodec( int pixels_per_frame = codec_params.width * codec_params.height; if (pixels_per_frame < 352 * 288) { codec_params.SetVideoEncoderComplexity( - webrtc::VideoCodecComplexity::kComplexityHigher); + VideoCodecComplexity::kComplexityHigher); } // Turn off denoising for all streams but the highest resolution. codec_params.VP8()->denoisingOn = false; } - } else if (codec.codecType == webrtc::kVideoCodecH264) { + } else if (codec.codecType == kVideoCodecH264) { codec_params.H264()->numberOfTemporalLayers = stream_params.numberOfTemporalLayers; - } else if (codec.codecType == webrtc::kVideoCodecVP9 && + } else if (codec.codecType == kVideoCodecVP9 && scalability_mode.has_value() && !only_active_stream) { // If VP9 simulcast then explicitly set a single spatial layer for each // simulcast stream. diff --git a/media/engine/simulcast_encoder_adapter_unittest.cc b/media/engine/simulcast_encoder_adapter_unittest.cc index c8e18057e8..86475311ff 100644 --- a/media/engine/simulcast_encoder_adapter_unittest.cc +++ b/media/engine/simulcast_encoder_adapter_unittest.cc @@ -301,7 +301,7 @@ class MockVideoEncoder : public VideoEncoder { image._encodedWidth = width; image._encodedHeight = height; CodecSpecificInfo codec_specific_info; - codec_specific_info.codecType = webrtc::kVideoCodecVP8; + codec_specific_info.codecType = kVideoCodecVP8; callback_->OnEncodedImage(image, &codec_specific_info); } @@ -617,7 +617,7 @@ class TestSimulcastEncoderAdapterFake : public ::testing::Test, InitRefCodec(0, &ref_codec); ref_codec.qpMax = 45; ref_codec.SetVideoEncoderComplexity( - webrtc::VideoCodecComplexity::kComplexityHigher); + VideoCodecComplexity::kComplexityHigher); ref_codec.VP8()->denoisingOn = false; ref_codec.startBitrate = 100; // Should equal to the target bitrate. VerifyCodec(ref_codec, 0); @@ -1366,8 +1366,7 @@ TEST_F(TestSimulcastEncoderAdapterFake, DoesNotAlterMaxQpForScreenshare) { VideoCodec ref_codec; InitRefCodec(0, &ref_codec); ref_codec.qpMax = kHighMaxQp; - ref_codec.SetVideoEncoderComplexity( - webrtc::VideoCodecComplexity::kComplexityHigher); + ref_codec.SetVideoEncoderComplexity(VideoCodecComplexity::kComplexityHigher); ref_codec.VP8()->denoisingOn = false; ref_codec.startBitrate = 100; // Should equal to the target bitrate. VerifyCodec(ref_codec, 0); @@ -1400,8 +1399,7 @@ TEST_F(TestSimulcastEncoderAdapterFake, VideoCodec ref_codec; InitRefCodec(2, &ref_codec, true /* reverse_layer_order */); ref_codec.qpMax = kHighMaxQp; - ref_codec.SetVideoEncoderComplexity( - webrtc::VideoCodecComplexity::kComplexityHigher); + ref_codec.SetVideoEncoderComplexity(VideoCodecComplexity::kComplexityHigher); ref_codec.VP8()->denoisingOn = false; ref_codec.startBitrate = 100; // Should equal to the target bitrate. VerifyCodec(ref_codec, 2); diff --git a/media/engine/webrtc_media_engine.cc b/media/engine/webrtc_media_engine.cc index 60bf57f775..7565d9a540 100644 --- a/media/engine/webrtc_media_engine.cc +++ b/media/engine/webrtc_media_engine.cc @@ -37,8 +37,7 @@ void DiscardRedundantExtensions( bool found = false; for (const char* uri : extensions_decreasing_prio) { auto it = absl::c_find_if( - *extensions, - [uri](const webrtc::RtpExtension& rhs) { return rhs.uri == uri; }); + *extensions, [uri](const RtpExtension& rhs) { return rhs.uri == uri; }); if (it != extensions->end()) { if (found) { extensions->erase(it); diff --git a/media/engine/webrtc_media_engine_unittest.cc b/media/engine/webrtc_media_engine_unittest.cc index 91893fa08a..ec41af8c29 100644 --- a/media/engine/webrtc_media_engine_unittest.cc +++ b/media/engine/webrtc_media_engine_unittest.cc @@ -22,7 +22,7 @@ namespace webrtc { namespace { -using webrtc::test::ScopedKeyValueConfig; +using test::ScopedKeyValueConfig; std::vector MakeUniqueExtensions() { std::vector result; diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc index 3645bd054f..6f1d2f51ef 100644 --- a/media/engine/webrtc_video_engine.cc +++ b/media/engine/webrtc_video_engine.cc @@ -106,10 +106,7 @@ namespace webrtc { namespace { -using ::webrtc::ParseRtpPayloadType; -using ::webrtc::ParseRtpSsrc; - -constexpr int64_t kUnsignaledSsrcCooldownMs = webrtc::kNumMillisecsPerSec / 2; +constexpr int64_t kUnsignaledSsrcCooldownMs = kNumMillisecsPerSec / 2; // This constant is really an on/off, lower-level configurable NACK history // duration hasn't been implemented. @@ -176,8 +173,7 @@ std::vector GetDefaultSupportedFormats( std::vector supported_formats = factory->GetSupportedFormats(); if (is_decoder_factory) { - webrtc::AddH264ConstrainedBaselineProfileToSupportedFormats( - &supported_formats); + AddH264ConstrainedBaselineProfileToSupportedFormats(&supported_formats); } if (supported_formats.empty()) @@ -205,7 +201,7 @@ std::vector GetDefaultSupportedFormats( RTCErrorOr AssignPayloadType(const SdpVideoFormat& format, PayloadTypePicker& pt_mapper, const FieldTrialsView& trials) { - Codec codec = webrtc::CreateVideoCodec(format); + Codec codec = CreateVideoCodec(format); RTCErrorOr result = pt_mapper.SuggestMapping(codec, /* excluder= */ nullptr); if (!result.ok()) { @@ -220,8 +216,7 @@ RTCErrorOr AssignPayloadType(const SdpVideoFormat& format, // (VP8, VP9, AV1, H264, and RED). RTCErrorOr AddRtx(const Codec& primary_codec, PayloadTypePicker& pt_mapper) { - Codec rtx_codec = - webrtc::CreateVideoRtxCodec(Codec::kIdNotSet, primary_codec.id); + Codec rtx_codec = CreateVideoRtxCodec(Codec::kIdNotSet, primary_codec.id); RTCErrorOr result = pt_mapper.SuggestMapping(rtx_codec, /* excluder= */ nullptr); if (!result.ok()) { @@ -394,8 +389,8 @@ static bool ValidateStreamParams(const StreamParams& sp) { // Returns true if the given codec is disallowed from doing simulcast. bool IsCodecDisabledForSimulcast(bool legacy_scalability_mode, VideoCodecType codec_type) { - if (legacy_scalability_mode && (codec_type == webrtc::kVideoCodecVP9 || - codec_type == webrtc::kVideoCodecAV1)) { + if (legacy_scalability_mode && + (codec_type == kVideoCodecVP9 || codec_type == kVideoCodecAV1)) { return true; } @@ -424,9 +419,8 @@ std::optional NumSpatialLayersFromEncoding( if (idx >= rtp_parameters.encodings.size()) return std::nullopt; - std::optional scalability_mode = - webrtc::ScalabilityModeFromString( - rtp_parameters.encodings[idx].scalability_mode.value_or("")); + std::optional scalability_mode = ScalabilityModeFromString( + rtp_parameters.encodings[idx].scalability_mode.value_or("")); return scalability_mode ? std::optional( ScalabilityModeToNumSpatialLayers(*scalability_mode)) @@ -492,7 +486,7 @@ bool IsActiveFromEncodings( // with this ssrc does not exist. auto encoding_it = std::find_if( encodings.begin(), encodings.end(), - [ssrc = ssrc.value()](const webrtc::RtpEncodingParameters& encoding) { + [ssrc = ssrc.value()](const RtpEncodingParameters& encoding) { return encoding.ssrc.has_value() && encoding.ssrc.value() == ssrc; }); return encoding_it != encodings.end() ? encoding_it->active : false; @@ -547,11 +541,11 @@ void FallbackToDefaultScalabilityModeIfNotSupported( config)) { encoding.scalability_mode = (encoding.scalability_mode != - std::string(webrtc::kDefaultScalabilityModeStr) && - IsScalabilityModeSupportedByCodec( - codec, webrtc::kDefaultScalabilityModeStr, config)) - ? webrtc::kDefaultScalabilityModeStr - : webrtc::kNoLayeringScalabilityModeStr; + std::string(kDefaultScalabilityModeStr) && + IsScalabilityModeSupportedByCodec(codec, kDefaultScalabilityModeStr, + config)) + ? kDefaultScalabilityModeStr + : kNoLayeringScalabilityModeStr; RTC_LOG(LS_INFO) << " -> " << *encoding.scalability_mode; } } @@ -577,12 +571,11 @@ RTCErrorOr> MapCodecs( UlpfecConfig ulpfec_config; std::optional flexfec_payload_type; - for (const webrtc::Codec& in_codec : codecs) { + for (const Codec& in_codec : codecs) { const int payload_type = in_codec.id; if (payload_codec_type.find(payload_type) != payload_codec_type.end()) { - if (webrtc::MatchesWithCodecRules(defined_codecs.at(in_codec.id), - in_codec)) { + if (MatchesWithCodecRules(defined_codecs.at(in_codec.id), in_codec)) { // Ignore second occurence of the same codec. // This can happen with multiple H.264 profiles. continue; @@ -776,8 +769,7 @@ void ExtractCodecInformation( int ParseReceiveBufferSize(const FieldTrialsView& trials) { FieldTrialParameter size_bytes("size_bytes", kVideoRtpRecvBufferSize); - webrtc::ParseFieldTrial({&size_bytes}, - trials.Lookup("WebRTC-ReceiveBufferSize")); + ParseFieldTrial({&size_bytes}, trials.Lookup("WebRTC-ReceiveBufferSize")); if (size_bytes.Get() < 10'000 || size_bytes.Get() > 10'000'000) { RTC_LOG(LS_WARNING) << "WebRTC-ReceiveBufferSize out of bounds: " << size_bytes.Get(); @@ -881,21 +873,19 @@ WebRtcVideoEngine::GetRtpHeaderExtensions() const { // resolve conflicts. int id = 1; for (const auto& uri : - {webrtc::RtpExtension::kTimestampOffsetUri, - webrtc::RtpExtension::kAbsSendTimeUri, - webrtc::RtpExtension::kVideoRotationUri, - webrtc::RtpExtension::kTransportSequenceNumberUri, - webrtc::RtpExtension::kPlayoutDelayUri, - webrtc::RtpExtension::kVideoContentTypeUri, - webrtc::RtpExtension::kVideoTimingUri, - webrtc::RtpExtension::kColorSpaceUri, webrtc::RtpExtension::kMidUri, - webrtc::RtpExtension::kRidUri, webrtc::RtpExtension::kRepairedRidUri}) { + {RtpExtension::kTimestampOffsetUri, RtpExtension::kAbsSendTimeUri, + RtpExtension::kVideoRotationUri, + RtpExtension::kTransportSequenceNumberUri, + RtpExtension::kPlayoutDelayUri, RtpExtension::kVideoContentTypeUri, + RtpExtension::kVideoTimingUri, RtpExtension::kColorSpaceUri, + RtpExtension::kMidUri, RtpExtension::kRidUri, + RtpExtension::kRepairedRidUri}) { result.emplace_back(uri, id++, RtpTransceiverDirection::kSendRecv); } result.emplace_back(RtpExtension::kCorruptionDetectionUri, id++, /*preferred_encrypt=*/true, RtpTransceiverDirection::kStopped); - for (const auto& uri : {webrtc::RtpExtension::kAbsoluteCaptureTimeUri}) { + for (const auto& uri : {RtpExtension::kAbsoluteCaptureTimeUri}) { result.emplace_back(uri, id, RtpTransceiverDirection::kStopped); } result.emplace_back(RtpExtension::kGenericFrameDescriptorUri00, id, @@ -1030,7 +1020,7 @@ WebRtcVideoSendChannel::WebRtcVideoSendStream::ConfigureVideoEncoderSettings( {{"off", InterLayerPredMode::kOff}, {"on", InterLayerPredMode::kOn}, {"onkeypic", InterLayerPredMode::kOnKeyPic}}); - webrtc::ParseFieldTrial( + ParseFieldTrial( {&interlayer_pred_experiment_enabled, &inter_layer_pred_mode}, call_->trials().Lookup("WebRTC-Vp9InterLayerPred")); if (interlayer_pred_experiment_enabled) { @@ -1407,7 +1397,7 @@ RtpParameters WebRtcVideoSendChannel::GetRtpSendParameters( RtpParameters rtp_params = it->second->GetRtpParameters(); // Need to add the common list of codecs to the send stream-specific // RTP parameters. - for (const webrtc::Codec& codec : send_params_.codecs) { + for (const Codec& codec : send_params_.codecs) { if (send_codec() && send_codec()->codec.id == codec.id) { // Put the current send codec to the front of the codecs list. RTC_DCHECK_EQ(codec.name, send_codec()->codec.name); @@ -1432,8 +1422,8 @@ RTCError WebRtcVideoSendChannel::SetRtpSendParameters( RTC_LOG(LS_ERROR) << "Attempting to set RTP send parameters for stream " "with ssrc " << ssrc << " which doesn't exist."; - return webrtc::InvokeSetParametersCallback( - callback, RTCError(RTCErrorType::INTERNAL_ERROR)); + return InvokeSetParametersCallback(callback, + RTCError(RTCErrorType::INTERNAL_ERROR)); } // TODO(deadbeef): Handle setting parameters with a list of codecs in a @@ -1442,8 +1432,8 @@ RTCError WebRtcVideoSendChannel::SetRtpSendParameters( if (current_parameters.codecs != parameters.codecs) { RTC_DLOG(LS_ERROR) << "Using SetParameters to change the set of codecs " "is not currently supported."; - return webrtc::InvokeSetParametersCallback( - callback, RTCError(RTCErrorType::INTERNAL_ERROR)); + return InvokeSetParametersCallback(callback, + RTCError(RTCErrorType::INTERNAL_ERROR)); } if (!parameters.encodings.empty()) { @@ -1451,19 +1441,19 @@ RTCError WebRtcVideoSendChannel::SetRtpSendParameters( // https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-16#section-5 // TODO(deadbeef): Change values depending on whether we are sending a // keyframe or non-keyframe. - DiffServCodePoint new_dscp = webrtc::DSCP_DEFAULT; + DiffServCodePoint new_dscp = DSCP_DEFAULT; switch (parameters.encodings[0].network_priority) { case Priority::kVeryLow: - new_dscp = webrtc::DSCP_CS1; + new_dscp = DSCP_CS1; break; case Priority::kLow: - new_dscp = webrtc::DSCP_DEFAULT; + new_dscp = DSCP_DEFAULT; break; case Priority::kMedium: - new_dscp = webrtc::DSCP_AF42; + new_dscp = DSCP_AF42; break; case Priority::kHigh: - new_dscp = webrtc::DSCP_AF41; + new_dscp = DSCP_AF41; break; } @@ -1475,7 +1465,7 @@ RTCError WebRtcVideoSendChannel::SetRtpSendParameters( ResolveSendCodecs(*send_codec_, send_codecs_, parameters.encodings, negotiated_codecs_, &send_codecs); if (!error.ok()) { - return webrtc::InvokeSetParametersCallback(callback, error); + return InvokeSetParametersCallback(callback, error); } if (send_codecs_ != send_codecs) { @@ -1642,7 +1632,7 @@ bool WebRtcVideoSendChannel::GetStats(VideoMediaSendInfo* info) { // Log stats periodically. bool log_stats = false; - int64_t now_ms = webrtc::TimeMillis(); + int64_t now_ms = TimeMillis(); if (last_send_stats_log_ms_ == -1 || now_ms - last_send_stats_log_ms_ > kStatsLogIntervalMs) { last_send_stats_log_ms_ = now_ms; @@ -1721,8 +1711,8 @@ void WebRtcVideoSendChannel::OnPacketSent(const SentPacketInfo& sent_packet) { void WebRtcVideoSendChannel::OnReadyToSend(bool ready) { RTC_DCHECK_RUN_ON(&network_thread_checker_); RTC_LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready."); - call_->SignalChannelNetworkState( - MediaType::VIDEO, ready ? webrtc::kNetworkUp : webrtc::kNetworkDown); + call_->SignalChannelNetworkState(MediaType::VIDEO, + ready ? kNetworkUp : kNetworkDown); } void WebRtcVideoSendChannel::OnNetworkRouteChanged( @@ -2011,12 +2001,12 @@ void WebRtcVideoSendChannel::WebRtcVideoSendStream::SetCodec( } } - const bool has_lntf = webrtc::HasLntf(codec_settings.codec); + const bool has_lntf = HasLntf(codec_settings.codec); parameters_.config.rtp.lntf.enabled = has_lntf; parameters_.config.encoder_settings.capabilities.loss_notification = has_lntf; parameters_.config.rtp.nack.rtp_history_ms = - webrtc::HasNack(codec_settings.codec) ? kNackHistoryMs : 0; + HasNack(codec_settings.codec) ? kNackHistoryMs : 0; parameters_.codec_settings = codec_settings; @@ -2126,7 +2116,7 @@ RTCError WebRtcVideoSendChannel::WebRtcVideoSendStream::SetRtpParameters( RTCError error = CheckRtpParametersInvalidModificationAndValues( rtp_parameters_, new_parameters, call_->trials()); if (!error.ok()) { - return webrtc::InvokeSetParametersCallback(callback, error); + return InvokeSetParametersCallback(callback, error); } bool new_param = false; @@ -2205,7 +2195,7 @@ RTCError WebRtcVideoSendChannel::WebRtcVideoSendStream::SetRtpParameters( } GenerateKeyFrame(key_frames_requested_by_rid); } - return webrtc::InvokeSetParametersCallback(callback, RTCError::OK()); + return InvokeSetParametersCallback(callback, RTCError::OK()); } RtpParameters WebRtcVideoSendChannel::WebRtcVideoSendStream::GetRtpParameters() @@ -2257,7 +2247,7 @@ WebRtcVideoSendChannel::WebRtcVideoSendStream::CreateVideoEncoderConfig( const Codec& codec) const { RTC_DCHECK_RUN_ON(&thread_checker_); VideoEncoderConfig encoder_config; - encoder_config.codec_type = webrtc::PayloadStringToCodecType(codec.name); + encoder_config.codec_type = PayloadStringToCodecType(codec.name); encoder_config.video_format = SdpVideoFormat(codec.name, codec.params); bool is_screencast = parameters_.options.is_screencast.value_or(false); @@ -2276,8 +2266,7 @@ WebRtcVideoSendChannel::WebRtcVideoSendStream::CreateVideoEncoderConfig( // `legacy_scalability_mode` and codec used. encoder_config.number_of_streams = parameters_.config.rtp.ssrcs.size(); bool legacy_scalability_mode = true; - for (const webrtc::RtpEncodingParameters& encoding : - rtp_parameters_.encodings) { + for (const RtpEncodingParameters& encoding : rtp_parameters_.encodings) { if (encoding.scalability_mode.has_value() && (encoding.scale_resolution_down_by.has_value() || encoding.scale_resolution_down_to.has_value())) { @@ -2339,7 +2328,7 @@ WebRtcVideoSendChannel::WebRtcVideoSendStream::CreateVideoEncoderConfig( encoder_config.simulcast_layers[i].active = rtp_parameters_.encodings[i].active; encoder_config.simulcast_layers[i].scalability_mode = - webrtc::ScalabilityModeFromString( + ScalabilityModeFromString( rtp_parameters_.encodings[i].scalability_mode.value_or("")); if (rtp_parameters_.encodings[i].min_bitrate_bps) { encoder_config.simulcast_layers[i].min_bitrate_bps = @@ -2389,7 +2378,7 @@ void WebRtcVideoSendChannel::WebRtcVideoSendStream::ReconfigureEncoder( if (!stream_) { // The webrtc::VideoSendStream `stream_` has not yet been created but other // parameters has changed. - webrtc::InvokeSetParametersCallback(callback, RTCError::OK()); + InvokeSetParametersCallback(callback, RTCError::OK()); return; } @@ -2420,7 +2409,7 @@ void WebRtcVideoSendChannel::WebRtcVideoSendStream::ReconfigureEncoder( // The app is switching between legacy and standard modes, recreate instead // of reconfiguring to avoid number of streams not matching in lower layers. RecreateWebRtcStream(); - webrtc::InvokeSetParametersCallback(callback, RTCError::OK()); + InvokeSetParametersCallback(callback, RTCError::OK()); return; } @@ -2454,7 +2443,7 @@ WebRtcVideoSendChannel::WebRtcVideoSendStream::GetPerLayerVideoSenderInfos( } else { stats = stream_->GetStats(); if (log_stats) - RTC_LOG(LS_INFO) << stats.ToString(webrtc::TimeMillis()); + RTC_LOG(LS_INFO) << stats.ToString(TimeMillis()); // Metrics that are in common for all substreams. common_info.adapt_changes = stats.number_of_cpu_adapt_changes; @@ -2819,7 +2808,7 @@ RtpParameters WebRtcVideoReceiveChannel::GetRtpReceiverParameters( rtp_params.header_extensions = recv_rtp_extensions_; // Add codecs, which any stream is prepared to receive. - for (const webrtc::Codec& codec : recv_params_.codecs) { + for (const Codec& codec : recv_params_.codecs) { rtp_params.codecs.push_back(codec.ToCodecParameters()); } @@ -2839,7 +2828,7 @@ RtpParameters WebRtcVideoReceiveChannel::GetDefaultRtpReceiveParameters() rtp_params.encodings.emplace_back(); // Add codecs, which any stream is prepared to receive. - for (const webrtc::Codec& codec : recv_params_.codecs) { + for (const Codec& codec : recv_params_.codecs) { rtp_params.codecs.push_back(codec.ToCodecParameters()); } @@ -2871,8 +2860,7 @@ bool WebRtcVideoReceiveChannel::GetChangedReceiverParameters( /*is_decoder_factory=*/true, /*include_rtx=*/true, call_->trials()); for (const VideoCodecSettings& mapped_codec : mapped_codecs) { - if (!webrtc::FindMatchingVideoCodec(local_supported_codecs, - mapped_codec.codec)) { + if (!FindMatchingVideoCodec(local_supported_codecs, mapped_codec.codec)) { RTC_LOG(LS_ERROR) << "GetChangedReceiverParameters called with " "unsupported video codec: " << mapped_codec.codec.ToString(); @@ -3189,7 +3177,7 @@ bool WebRtcVideoReceiveChannel::GetStats(VideoMediaReceiveInfo* info) { // Log stats periodically. bool log_stats = false; - int64_t now_ms = webrtc::TimeMillis(); + int64_t now_ms = TimeMillis(); if (last_receive_stats_log_ms_ == -1 || now_ms - last_receive_stats_log_ms_ > kStatsLogIntervalMs) { last_receive_stats_log_ms_ = now_ms; @@ -3214,8 +3202,8 @@ void WebRtcVideoReceiveChannel::FillReceiverStats( void WebRtcVideoReceiveChannel::FillReceiveCodecStats( VideoMediaReceiveInfo* video_media_info) { for (const auto& receiver : video_media_info->receivers) { - auto codec = absl::c_find_if( - recv_params_.codecs, [&receiver](const webrtc::Codec& c) { + auto codec = + absl::c_find_if(recv_params_.codecs, [&receiver](const Codec& c) { return receiver.codec_payload_type && *receiver.codec_payload_type == c.id; }); @@ -3300,7 +3288,7 @@ bool WebRtcVideoReceiveChannel::MaybeCreateDefaultReceiveStream( // of creating decoders on every packet eats up processing time (e.g. // https://crbug.com/1069603) and this cooldown prevents that. if (last_unsignalled_ssrc_creation_time_ms_.has_value()) { - int64_t now_ms = webrtc::TimeMillis(); + int64_t now_ms = TimeMillis(); if (now_ms - last_unsignalled_ssrc_creation_time_ms_.value() < kUnsignaledSsrcCooldownMs) { // We've already created an unsignalled ssrc stream within the last @@ -3314,7 +3302,7 @@ bool WebRtcVideoReceiveChannel::MaybeCreateDefaultReceiveStream( // RTX SSRC not yet known. ReCreateDefaultReceiveStream(packet.Ssrc(), std::nullopt); - last_unsignalled_ssrc_creation_time_ms_ = webrtc::TimeMillis(); + last_unsignalled_ssrc_creation_time_ms_ = TimeMillis(); return true; } @@ -3456,15 +3444,13 @@ WebRtcVideoReceiveChannel::WebRtcVideoReceiveStream::WebRtcVideoReceiveStream( const VideoCodecSettings& codec = recv_codecs.front(); config_.rtp.ulpfec_payload_type = codec.ulpfec.ulpfec_payload_type; config_.rtp.red_payload_type = codec.ulpfec.red_payload_type; - config_.rtp.lntf.enabled = webrtc::HasLntf(codec.codec); - config_.rtp.nack.rtp_history_ms = - webrtc::HasNack(codec.codec) ? kNackHistoryMs : 0; + config_.rtp.lntf.enabled = HasLntf(codec.codec); + config_.rtp.nack.rtp_history_ms = HasNack(codec.codec) ? kNackHistoryMs : 0; if (codec.rtx_time && config_.rtp.nack.rtp_history_ms != 0) { config_.rtp.nack.rtp_history_ms = *codec.rtx_time; } - config_.rtp.rtcp_xr.receiver_reference_time_report = - webrtc::HasRrtr(codec.codec); + config_.rtp.rtcp_xr.receiver_reference_time_report = HasRrtr(codec.codec); if (codec.ulpfec.red_rtx_payload_type != -1) { config_.rtp @@ -3545,14 +3531,14 @@ bool WebRtcVideoReceiveChannel::WebRtcVideoReceiveStream::ReconfigureCodecs( config_.rtp.ulpfec_payload_type); } - const bool has_lntf = webrtc::HasLntf(codec.codec); + const bool has_lntf = HasLntf(codec.codec); if (config_.rtp.lntf.enabled != has_lntf) { config_.rtp.lntf.enabled = has_lntf; stream_->SetLossNotificationEnabled(has_lntf); } int new_history_ms = config_.rtp.nack.rtp_history_ms; - const int rtp_history_ms = webrtc::HasNack(codec.codec) ? kNackHistoryMs : 0; + const int rtp_history_ms = HasNack(codec.codec) ? kNackHistoryMs : 0; if (rtp_history_ms != config_.rtp.nack.rtp_history_ms) { new_history_ms = rtp_history_ms; } @@ -3568,7 +3554,7 @@ bool WebRtcVideoReceiveChannel::WebRtcVideoReceiveStream::ReconfigureCodecs( stream_->SetNackHistory(TimeDelta::Millis(new_history_ms)); } - const bool has_rtr = webrtc::HasRrtr(codec.codec); + const bool has_rtr = HasRrtr(codec.codec); if (has_rtr != config_.rtp.rtcp_xr.receiver_reference_time_report) { config_.rtp.rtcp_xr.receiver_reference_time_report = has_rtr; stream_->SetRtcpXr(config_.rtp.rtcp_xr); @@ -3742,7 +3728,7 @@ void WebRtcVideoReceiveChannel::WebRtcVideoReceiveStream::OnFrame( const VideoFrame& frame) { MutexLock lock(&sink_lock_); - int64_t time_now_ms = webrtc::TimeMillis(); + int64_t time_now_ms = TimeMillis(); if (first_frame_timestamp_ < 0) first_frame_timestamp_ = time_now_ms; int64_t elapsed_time_ms = time_now_ms - first_frame_timestamp_; @@ -3928,7 +3914,7 @@ WebRtcVideoReceiveChannel::WebRtcVideoReceiveStream::GetVideoReceiverInfo( // present if DLRR is enabled. if (log_stats) - RTC_LOG(LS_INFO) << stats.ToString(webrtc::TimeMillis()); + RTC_LOG(LS_INFO) << stats.ToString(TimeMillis()); return info; } @@ -4018,9 +4004,9 @@ void WebRtcVideoReceiveChannel::ProcessReceivedPacket( // It would likely be good if extensions where merged per BUNDLE and // applied directly in RtpTransport::DemuxPacket; packet.IdentifyExtensions(recv_rtp_extension_map_); - packet.set_payload_type_frequency(webrtc::kVideoPayloadTypeFrequency); + packet.set_payload_type_frequency(kVideoPayloadTypeFrequency); if (!packet.arrival_time().IsFinite()) { - packet.set_arrival_time(Timestamp::Micros(webrtc::TimeMicros())); + packet.set_arrival_time(Timestamp::Micros(TimeMicros())); } call_->Receiver()->DeliverRtpPacket( diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc index 0d2803ba3c..8d7f803b44 100644 --- a/media/engine/webrtc_video_engine_unittest.cc +++ b/media/engine/webrtc_video_engine_unittest.cc @@ -127,6 +127,9 @@ #include "video/config/simulcast.h" #include "video/config/video_encoder_config.h" +namespace webrtc { +namespace { + using ::testing::_; using ::testing::Combine; using ::testing::Contains; @@ -149,8 +152,6 @@ using ::webrtc::test::FunctionVideoDecoderFactory; using ::webrtc::test::RtcpPacketParser; using ::webrtc::test::ScopedKeyValueConfig; -namespace webrtc { -namespace { static const uint8_t kRedRtxPayloadType = 125; @@ -175,31 +176,31 @@ constexpr size_t kNumSimulcastStreams = 3; static const char kUnsupportedExtensionName[] = "urn:ietf:params:rtp-hdrext:unsupported"; -webrtc::Codec RemoveFeedbackParams(webrtc::Codec&& codec) { - codec.feedback_params = webrtc::FeedbackParams(); +Codec RemoveFeedbackParams(Codec&& codec) { + codec.feedback_params = FeedbackParams(); return std::move(codec); } -void VerifyCodecHasDefaultFeedbackParams(const webrtc::Codec& codec, +void VerifyCodecHasDefaultFeedbackParams(const Codec& codec, bool lntf_expected) { - EXPECT_EQ(lntf_expected, codec.HasFeedbackParam(webrtc::FeedbackParam( + EXPECT_EQ(lntf_expected, codec.HasFeedbackParam(FeedbackParam( kRtcpFbParamLntf, kParamValueEmpty))); EXPECT_TRUE(codec.HasFeedbackParam( - webrtc::FeedbackParam(kRtcpFbParamNack, kParamValueEmpty))); + FeedbackParam(kRtcpFbParamNack, kParamValueEmpty))); EXPECT_TRUE(codec.HasFeedbackParam( - webrtc::FeedbackParam(kRtcpFbParamNack, kRtcpFbNackParamPli))); + FeedbackParam(kRtcpFbParamNack, kRtcpFbNackParamPli))); EXPECT_TRUE(codec.HasFeedbackParam( - webrtc::FeedbackParam(kRtcpFbParamRemb, kParamValueEmpty))); + FeedbackParam(kRtcpFbParamRemb, kParamValueEmpty))); EXPECT_TRUE(codec.HasFeedbackParam( - webrtc::FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty))); + FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty))); EXPECT_TRUE(codec.HasFeedbackParam( - webrtc::FeedbackParam(kRtcpFbParamCcm, kRtcpFbCcmParamFir))); + FeedbackParam(kRtcpFbParamCcm, kRtcpFbCcmParamFir))); } // Return true if any codec in `codecs` is an RTX codec with associated // payload type `payload_type`. -bool HasRtxCodec(const std::vector& codecs, int payload_type) { - for (const webrtc::Codec& codec : codecs) { +bool HasRtxCodec(const std::vector& codecs, int payload_type) { + for (const Codec& codec : codecs) { int associated_payload_type; if (absl::EqualsIgnoreCase(codec.name.c_str(), "rtx") && codec.GetParam(kCodecParamAssociatedPayloadType, @@ -213,8 +214,8 @@ bool HasRtxCodec(const std::vector& codecs, int payload_type) { // Return true if any codec in `codecs` is an RTX codec, independent of // payload type. -bool HasAnyRtxCodec(const std::vector& codecs) { - for (const webrtc::Codec& codec : codecs) { +bool HasAnyRtxCodec(const std::vector& codecs) { + for (const Codec& codec : codecs) { if (absl::EqualsIgnoreCase(codec.name.c_str(), "rtx")) { return true; } @@ -230,16 +231,15 @@ const int* FindKeyByValue(const std::map& m, int v) { return nullptr; } -bool HasRtxReceiveAssociation( - const webrtc::VideoReceiveStreamInterface::Config& config, - int payload_type) { +bool HasRtxReceiveAssociation(const VideoReceiveStreamInterface::Config& config, + int payload_type) { return FindKeyByValue(config.rtp.rtx_associated_payload_types, payload_type) != nullptr; } // Check that there's an Rtx payload type for each decoder. bool VerifyRtxReceiveAssociations( - const webrtc::VideoReceiveStreamInterface::Config& config) { + const VideoReceiveStreamInterface::Config& config) { for (const auto& decoder : config.decoders) { if (!HasRtxReceiveAssociation(config, decoder.payload_type)) return false; @@ -247,15 +247,13 @@ bool VerifyRtxReceiveAssociations( return true; } -scoped_refptr CreateBlackFrameBuffer(int width, - int height) { - scoped_refptr buffer = - webrtc::I420Buffer::Create(width, height); - webrtc::I420Buffer::SetBlack(buffer.get()); +scoped_refptr CreateBlackFrameBuffer(int width, int height) { + scoped_refptr buffer = I420Buffer::Create(width, height); + I420Buffer::SetBlack(buffer.get()); return buffer; } -void VerifySendStreamHasRtxTypes(const webrtc::VideoSendStream::Config& config, +void VerifySendStreamHasRtxTypes(const VideoSendStream::Config& config, const std::map& rtx_types) { std::map::const_iterator it; it = rtx_types.find(config.rtp.payload_type); @@ -288,17 +286,16 @@ int GetMaxDefaultBitrateBps(size_t width, size_t height) { } } -class MockVideoSource - : public webrtc::VideoSourceInterface { +class MockVideoSource : public VideoSourceInterface { public: MOCK_METHOD(void, AddOrUpdateSink, - (webrtc::VideoSinkInterface * sink, - const webrtc::VideoSinkWants& wants), + (VideoSinkInterface * sink, + const VideoSinkWants& wants), (override)); MOCK_METHOD(void, RemoveSink, - (webrtc::VideoSinkInterface * sink), + (VideoSinkInterface * sink), (override)); }; @@ -306,23 +303,23 @@ class MockNetworkInterface : public MediaChannelNetworkInterface { public: MOCK_METHOD(bool, SendPacket, - (webrtc::CopyOnWriteBuffer * packet, - const webrtc::AsyncSocketPacketOptions& options), + (CopyOnWriteBuffer * packet, + const AsyncSocketPacketOptions& options), (override)); MOCK_METHOD(bool, SendRtcp, - (webrtc::CopyOnWriteBuffer * packet, - const webrtc::AsyncSocketPacketOptions& options), + (CopyOnWriteBuffer * packet, + const AsyncSocketPacketOptions& options), (override)); MOCK_METHOD(int, SetOption, - (SocketType type, webrtc::Socket::Option opt, int option), + (SocketType type, Socket::Option opt, int option), (override)); }; -std::vector GetStreamResolutions( - const std::vector& streams) { - std::vector res; +std::vector GetStreamResolutions( + const std::vector& streams) { + std::vector res; for (const auto& s : streams) { if (s.active) { res.push_back({checked_cast(s.width), checked_cast(s.height)}); @@ -396,7 +393,7 @@ class WebRtcVideoEngineTest : public ::testing::Test { encoder_factory_(new FakeWebRtcVideoEncoderFactory), decoder_factory_(new FakeWebRtcVideoDecoderFactory), video_bitrate_allocator_factory_( - webrtc::CreateBuiltinVideoBitrateAllocatorFactory()), + CreateBuiltinVideoBitrateAllocatorFactory()), engine_( std::unique_ptr(encoder_factory_), std::unique_ptr(decoder_factory_), @@ -470,7 +467,7 @@ TEST_F(WebRtcVideoEngineTest, AddSupportedVideoCodecType("VP9"); AddSupportedVideoCodecType("AV1"); AddSupportedVideoCodecType("H264"); - for (const webrtc::Codec& codec : engine_.LegacySendCodecs()) { + for (const Codec& codec : engine_.LegacySendCodecs()) { if (codec.name != kRtxCodecName) continue; int associated_payload_type; @@ -478,7 +475,7 @@ TEST_F(WebRtcVideoEngineTest, &associated_payload_type)); EXPECT_EQ(codec.id, associated_payload_type + 1); } - for (const webrtc::Codec& codec : engine_.LegacyRecvCodecs()) { + for (const Codec& codec : engine_.LegacyRecvCodecs()) { if (codec.name != kRtxCodecName) continue; int associated_payload_type; @@ -757,7 +754,7 @@ TEST_F(WebRtcVideoEngineTest, UseFactoryForVp8WhenSupported) { EXPECT_EQ(0, encoder_factory_->GetNumCreatedEncoders()); EXPECT_TRUE(send_channel->SetSend(true)); FrameForwarder frame_forwarder; - FakeFrameSource frame_source(1280, 720, webrtc::kNumMicrosecsPerSec / 30); + FakeFrameSource frame_source(1280, 720, kNumMicrosecsPerSec / 30); EXPECT_TRUE(send_channel->SetVideoSend(kSsrc, nullptr, &frame_forwarder)); frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); time_controller_.AdvanceTime(TimeDelta::Zero()); @@ -785,10 +782,6 @@ TEST_F(WebRtcVideoEngineTest, UseFactoryForVp8WhenSupported) { // TODO(deadbeef): This test should be updated if/when we start // adding RTX codecs for unrecognized codec names. TEST_F(WebRtcVideoEngineTest, RtxCodecAddedForH264Codec) { - using webrtc::H264Level; - using webrtc::H264Profile; - using webrtc::H264ProfileLevelId; - using webrtc::H264ProfileLevelIdToString; SdpVideoFormat h264_constrained_baseline("H264"); h264_constrained_baseline.parameters[kH264FmtpProfileLevelId] = *H264ProfileLevelIdToString(H264ProfileLevelId( @@ -810,17 +803,15 @@ TEST_F(WebRtcVideoEngineTest, RtxCodecAddedForH264Codec) { // Now search for RTX codecs for them. Expect that they all have associated // RTX codecs. EXPECT_TRUE(HasRtxCodec( - codecs, webrtc::FindMatchingVideoCodec( - codecs, webrtc::CreateVideoCodec(h264_constrained_baseline)) + codecs, FindMatchingVideoCodec( + codecs, CreateVideoCodec(h264_constrained_baseline)) ->id)); EXPECT_TRUE(HasRtxCodec( - codecs, webrtc::FindMatchingVideoCodec( - codecs, webrtc::CreateVideoCodec(h264_constrained_high)) - ->id)); - EXPECT_TRUE( - HasRtxCodec(codecs, webrtc::FindMatchingVideoCodec( - codecs, webrtc::CreateVideoCodec(h264_high)) - ->id)); + codecs, + FindMatchingVideoCodec(codecs, CreateVideoCodec(h264_constrained_high)) + ->id)); + EXPECT_TRUE(HasRtxCodec( + codecs, FindMatchingVideoCodec(codecs, CreateVideoCodec(h264_high))->id)); } #if defined(RTC_ENABLE_VP9) @@ -843,7 +834,7 @@ TEST_F(WebRtcVideoEngineTest, PropagatesInputFrameTimestamp) { EXPECT_TRUE(send_channel->AddSendStream(StreamParams::CreateLegacy(kSsrc))); FrameForwarder frame_forwarder; - FakeFrameSource frame_source(1280, 720, webrtc::kNumMicrosecsPerSec / 60); + FakeFrameSource frame_source(1280, 720, kNumMicrosecsPerSec / 60); EXPECT_TRUE(send_channel->SetVideoSend(kSsrc, nullptr, &frame_forwarder)); send_channel->SetSend(true); @@ -863,14 +854,12 @@ TEST_F(WebRtcVideoEngineTest, PropagatesInputFrameTimestamp) { last_timestamp = timestamp; } - frame_forwarder.IncomingCapturedFrame( - frame_source.GetFrame(1280, 720, VideoRotation::kVideoRotation_0, - webrtc::kNumMicrosecsPerSec / 30)); + frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame( + 1280, 720, VideoRotation::kVideoRotation_0, kNumMicrosecsPerSec / 30)); last_timestamp = stream->GetLastTimestamp(); for (int i = 0; i < 10; i++) { - frame_forwarder.IncomingCapturedFrame( - frame_source.GetFrame(1280, 720, VideoRotation::kVideoRotation_0, - webrtc::kNumMicrosecsPerSec / 30)); + frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame( + 1280, 720, VideoRotation::kVideoRotation_0, kNumMicrosecsPerSec / 30)); int64_t timestamp = stream->GetLastTimestamp(); int64_t interval = timestamp - last_timestamp; @@ -888,7 +877,7 @@ TEST_F(WebRtcVideoEngineTest, PropagatesInputFrameTimestamp) { void WebRtcVideoEngineTest::AssignDefaultAptRtxTypes() { std::vector engine_codecs = engine_.LegacySendCodecs(); RTC_DCHECK(!engine_codecs.empty()); - for (const webrtc::Codec& codec : engine_codecs) { + for (const Codec& codec : engine_codecs) { if (codec.name == "rtx") { int associated_payload_type; if (codec.GetParam(kCodecParamAssociatedPayloadType, @@ -903,7 +892,7 @@ void WebRtcVideoEngineTest::AssignDefaultCodec() { std::vector engine_codecs = engine_.LegacySendCodecs(); RTC_DCHECK(!engine_codecs.empty()); bool codec_set = false; - for (const webrtc::Codec& codec : engine_codecs) { + for (const Codec& codec : engine_codecs) { if (!codec_set && codec.name != "rtx" && codec.name != "red" && codec.name != "ulpfec" && codec.name != "flexfec-03") { default_codec_ = codec; @@ -925,7 +914,7 @@ size_t WebRtcVideoEngineTest::GetEngineCodecIndex( // an internal H264 codec from the engine with a different H264 profile. if (absl::EqualsIgnoreCase(name.c_str(), kH264CodecName)) { const std::optional profile_level_id = - webrtc::ParseSdpForH264ProfileLevelId(engine_codec.params); + ParseSdpForH264ProfileLevelId(engine_codec.params); if (profile_level_id->profile != H264Profile::kProfileConstrainedBaseline) { continue; @@ -962,8 +951,7 @@ WebRtcVideoEngineTest::SetSendParamsWithAllSupportedCodecs() { video_bitrate_allocator_factory_.get()); VideoSenderParameters parameters; // We need to look up the codec in the engine to get the correct payload type. - for (const webrtc::SdpVideoFormat& format : - encoder_factory_->GetSupportedFormats()) { + for (const SdpVideoFormat& format : encoder_factory_->GetSupportedFormats()) { Codec engine_codec = GetEngineCodec(format.name); if (!absl::c_linear_search(parameters.codecs, engine_codec)) { parameters.codecs.push_back(engine_codec); @@ -991,8 +979,7 @@ WebRtcVideoEngineTest::SetRecvParamsWithSupportedCodecs( std::unique_ptr WebRtcVideoEngineTest::SetRecvParamsWithAllSupportedCodecs() { std::vector codecs; - for (const webrtc::SdpVideoFormat& format : - decoder_factory_->GetSupportedFormats()) { + for (const SdpVideoFormat& format : decoder_factory_->GetSupportedFormats()) { Codec engine_codec = GetEngineCodec(format.name); if (!absl::c_linear_search(codecs, engine_codec)) { codecs.push_back(engine_codec); @@ -1103,12 +1090,12 @@ TEST_F(WebRtcVideoEngineTest, UsesSimulcastAdapterForVp8Factories) { std::vector ssrcs = MAKE_VECTOR(kSsrcs3); - EXPECT_TRUE(send_channel->AddSendStream( - webrtc::CreateSimStreamParams("cname", ssrcs))); + EXPECT_TRUE( + send_channel->AddSendStream(CreateSimStreamParams("cname", ssrcs))); EXPECT_TRUE(send_channel->SetSend(true)); FrameForwarder frame_forwarder; - FakeFrameSource frame_source(1280, 720, webrtc::kNumMicrosecsPerSec / 60); + FakeFrameSource frame_source(1280, 720, kNumMicrosecsPerSec / 60); EXPECT_TRUE( send_channel->SetVideoSend(ssrcs.front(), nullptr, &frame_forwarder)); frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); @@ -1139,7 +1126,7 @@ TEST_F(WebRtcVideoEngineTest, ChannelWithH264CanChangeToVp8) { // Frame source. FrameForwarder frame_forwarder; - FakeFrameSource frame_source(1280, 720, webrtc::kNumMicrosecsPerSec / 30); + FakeFrameSource frame_source(1280, 720, kNumMicrosecsPerSec / 30); std::unique_ptr send_channel = engine_.CreateSendChannel(call_.get(), GetMediaConfig(), VideoOptions(), @@ -1183,15 +1170,15 @@ TEST_F(WebRtcVideoEngineTest, std::vector ssrcs = MAKE_VECTOR(kSsrcs3); - EXPECT_TRUE(send_channel->AddSendStream( - webrtc::CreateSimStreamParams("cname", ssrcs))); + EXPECT_TRUE( + send_channel->AddSendStream(CreateSimStreamParams("cname", ssrcs))); EXPECT_TRUE(send_channel->SetSend(true)); // Send a fake frame, or else the media engine will configure the simulcast // encoder adapter at a low-enough size that it'll only create a single // encoder layer. FrameForwarder frame_forwarder; - FakeFrameSource frame_source(1280, 720, webrtc::kNumMicrosecsPerSec / 30); + FakeFrameSource frame_source(1280, 720, kNumMicrosecsPerSec / 30); EXPECT_TRUE( send_channel->SetVideoSend(ssrcs.front(), nullptr, &frame_forwarder)); frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); @@ -1199,7 +1186,7 @@ TEST_F(WebRtcVideoEngineTest, ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(2)); ASSERT_TRUE(encoder_factory_->encoders()[0]->WaitForInitEncode()); - EXPECT_EQ(webrtc::kVideoCodecVP8, + EXPECT_EQ(kVideoCodecVP8, encoder_factory_->encoders()[0]->GetCodecSettings().codecType); send_channel.reset(); @@ -1224,14 +1211,14 @@ TEST_F(WebRtcVideoEngineTest, // Send a frame of 720p. This should trigger a "real" encoder initialization. FrameForwarder frame_forwarder; - FakeFrameSource frame_source(1280, 720, webrtc::kNumMicrosecsPerSec / 30); + FakeFrameSource frame_source(1280, 720, kNumMicrosecsPerSec / 30); EXPECT_TRUE(send_channel->SetVideoSend(kSsrc, nullptr, &frame_forwarder)); frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); time_controller_.AdvanceTime(TimeDelta::Zero()); ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(1)); ASSERT_EQ(1u, encoder_factory_->encoders().size()); ASSERT_TRUE(encoder_factory_->encoders()[0]->WaitForInitEncode()); - EXPECT_EQ(webrtc::kVideoCodecH264, + EXPECT_EQ(kVideoCodecH264, encoder_factory_->encoders()[0]->GetCodecSettings().codecType); send_channel.reset(); @@ -1252,12 +1239,12 @@ TEST_F(WebRtcVideoEngineTest, SimulcastEnabledForH264) { EXPECT_TRUE(send_channel->SetSenderParameters(parameters)); const std::vector ssrcs = MAKE_VECTOR(kSsrcs3); - EXPECT_TRUE(send_channel->AddSendStream( - webrtc::CreateSimStreamParams("cname", ssrcs))); + EXPECT_TRUE( + send_channel->AddSendStream(CreateSimStreamParams("cname", ssrcs))); // Send a frame of 720p. This should trigger a "real" encoder initialization. FrameForwarder frame_forwarder; - FakeFrameSource frame_source(1280, 720, webrtc::kNumMicrosecsPerSec / 30); + FakeFrameSource frame_source(1280, 720, kNumMicrosecsPerSec / 30); EXPECT_TRUE(send_channel->SetVideoSend(ssrcs[0], nullptr, &frame_forwarder)); frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); time_controller_.AdvanceTime(TimeDelta::Zero()); @@ -1266,7 +1253,7 @@ TEST_F(WebRtcVideoEngineTest, SimulcastEnabledForH264) { ASSERT_EQ(1u, encoder_factory_->encoders().size()); FakeWebRtcVideoEncoder* encoder = encoder_factory_->encoders()[0]; ASSERT_TRUE(encoder_factory_->encoders()[0]->WaitForInitEncode()); - EXPECT_EQ(webrtc::kVideoCodecH264, encoder->GetCodecSettings().codecType); + EXPECT_EQ(kVideoCodecH264, encoder->GetCodecSettings().codecType); EXPECT_LT(1u, encoder->GetCodecSettings().numberOfSimulcastStreams); EXPECT_TRUE(send_channel->SetVideoSend(ssrcs[0], nullptr, nullptr)); } @@ -1443,7 +1430,7 @@ TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, Vp8) { std::unique_ptr rate_allocator_factory = std::make_unique(); EXPECT_CALL(*rate_allocator_factory, - Create(_, Field(&VideoCodec::codecType, webrtc::kVideoCodecVP8))) + Create(_, Field(&VideoCodec::codecType, kVideoCodecVP8))) .WillOnce([] { return std::make_unique(); }); FieldTrialBasedConfig trials; WebRtcVideoEngine engine( @@ -1520,7 +1507,7 @@ TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, Vp8) { // Set capturer. FrameForwarder frame_forwarder; - FakeFrameSource frame_source(1280, 720, webrtc::kNumMicrosecsPerSec / 30); + FakeFrameSource frame_source(1280, 720, kNumMicrosecsPerSec / 30); EXPECT_TRUE(send_channel->SetVideoSend(send_ssrc, nullptr, &frame_forwarder)); // Sending one frame will allocate the encoder. frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); @@ -1560,7 +1547,7 @@ TEST_F(WebRtcVideoEngineTest, DISABLED_RecreatesEncoderOnContentTypeChange) { ASSERT_TRUE(send_channel->SetSenderParameters(parameters)); FrameForwarder frame_forwarder; - FakeFrameSource frame_source(1280, 720, webrtc::kNumMicrosecsPerSec / 30); + FakeFrameSource frame_source(1280, 720, kNumMicrosecsPerSec / 30); VideoOptions options; EXPECT_TRUE(send_channel->SetVideoSend(kSsrc, &options, &frame_forwarder)); @@ -1632,7 +1619,7 @@ class WebRtcVideoChannelEncodedFrameCallbackTest : public ::testing::Test { time_controller_.GetClock())), call_(Call::Create(CallConfig(env_))), video_bitrate_allocator_factory_( - webrtc::CreateBuiltinVideoBitrateAllocatorFactory()), + CreateBuiltinVideoBitrateAllocatorFactory()), engine_( std::make_unique< VideoEncoderFactoryTemplateAddSendStream(DefaultSendStreamParams())); frame_forwarder_ = std::make_unique(); frame_source_ = std::make_unique( - 640, 480, webrtc::kNumMicrosecsPerSec / kFramerate); + 640, 480, kNumMicrosecsPerSec / kFramerate); EXPECT_TRUE( send_channel_->SetVideoSend(kSsrc, nullptr, frame_forwarder_.get())); } @@ -1901,7 +1888,7 @@ class WebRtcVideoChannelBaseTest : public ::testing::Test { bool SetOneCodec(const Codec& codec) { frame_source_ = std::make_unique( - kVideoWidth, kVideoHeight, webrtc::kNumMicrosecsPerSec / kFramerate); + kVideoWidth, kVideoHeight, kNumMicrosecsPerSec / kFramerate); bool sending = SendImpl()->sending(); bool success = SetSend(false); @@ -1998,13 +1985,13 @@ class WebRtcVideoChannelBaseTest : public ::testing::Test { } Codec GetEngineCodec(const std::string& name) { - for (const webrtc::Codec& engine_codec : engine_.LegacySendCodecs()) { + for (const Codec& engine_codec : engine_.LegacySendCodecs()) { if (absl::EqualsIgnoreCase(name, engine_codec.name)) return engine_codec; } // This point should never be reached. ADD_FAILURE() << "Unrecognized codec name: " << name; - return webrtc::CreateVideoCodec(0, ""); + return CreateVideoCodec(0, ""); } Codec DefaultCodec() { return GetEngineCodec("VP8"); } @@ -2202,7 +2189,7 @@ TEST_F(WebRtcVideoChannelBaseTest, GetStatsMultipleSendStreams) { const int kTestWidth = 160; const int kTestHeight = 120; FakeFrameSource frame_source(kTestWidth, kTestHeight, - webrtc::kNumMicrosecsPerSec / 5); + kNumMicrosecsPerSec / 5); EXPECT_TRUE(send_channel_->AddSendStream(StreamParams::CreateLegacy(5678))); EXPECT_TRUE(send_channel_->SetVideoSend(5678, nullptr, &frame_forwarder)); EXPECT_TRUE( @@ -2381,8 +2368,6 @@ TEST_F(WebRtcVideoChannelBaseTest, SimulateConference) { // Tests that we can add and remove capturers and frames are sent out properly TEST_F(WebRtcVideoChannelBaseTest, DISABLED_AddRemoveCapturer) { - using webrtc::Codec; - Codec codec = DefaultCodec(); const int time_between_send_ms = VideoFormat::FpsToInterval(kFramerate); EXPECT_TRUE(SetOneCodec(codec)); @@ -2393,8 +2378,8 @@ TEST_F(WebRtcVideoChannelBaseTest, DISABLED_AddRemoveCapturer) { EXPECT_FRAME(1, kVideoWidth, kVideoHeight); FrameForwarder frame_forwarder; - FakeFrameSource frame_source(480, 360, webrtc::kNumMicrosecsPerSec / 30, - webrtc::kNumMicrosecsPerSec / 30); + FakeFrameSource frame_source(480, 360, kNumMicrosecsPerSec / 30, + kNumMicrosecsPerSec / 30); // TODO(nisse): This testcase fails if we don't configure // screencast. It's unclear why, I see nothing obvious in this @@ -2480,7 +2465,7 @@ TEST_F(WebRtcVideoChannelBaseTest, AddRemoveCapturerMultipleSources) { FrameForwarder frame_forwarder1; FakeFrameSource frame_source(kVideoWidth, kVideoHeight, - webrtc::kNumMicrosecsPerSec / kFramerate); + kNumMicrosecsPerSec / kFramerate); // Set up additional stream 2. FakeVideoRenderer renderer2; @@ -2503,13 +2488,13 @@ TEST_F(WebRtcVideoChannelBaseTest, AddRemoveCapturerMultipleSources) { const int kTestHeight = 120; frame_forwarder1.IncomingCapturedFrame(frame_source.GetFrame( kTestWidth, kTestHeight, VideoRotation::kVideoRotation_0, - webrtc::kNumMicrosecsPerSec / kFramerate)); + kNumMicrosecsPerSec / kFramerate)); time_controller_.AdvanceTime(kFrameDuration); EXPECT_FRAME_ON_RENDERER(renderer1, 1, kTestWidth, kTestHeight); // Capture a frame with additional capturer2, frames should be received frame_forwarder2.IncomingCapturedFrame(frame_source.GetFrame( kTestWidth, kTestHeight, VideoRotation::kVideoRotation_0, - webrtc::kNumMicrosecsPerSec / kFramerate)); + kNumMicrosecsPerSec / kFramerate)); time_controller_.AdvanceTime(kFrameDuration); EXPECT_FRAME_ON_RENDERER(renderer2, 1, kTestWidth, kTestHeight); // Successfully remove the capturer. @@ -2713,7 +2698,7 @@ class WebRtcVideoChannelTest : public WebRtcVideoEngineTest { WebRtcVideoChannelTest() : WebRtcVideoChannelTest("") {} explicit WebRtcVideoChannelTest(const char* field_trials) : WebRtcVideoEngineTest(field_trials), - frame_source_(1280, 720, webrtc::kNumMicrosecsPerSec / 30), + frame_source_(1280, 720, kNumMicrosecsPerSec / 30), last_ssrc_(0) {} void SetUp() override { AddSupportedVideoCodecType("VP8"); @@ -2775,13 +2760,13 @@ class WebRtcVideoChannelTest : public WebRtcVideoEngineTest { } Codec GetEngineCodec(const std::string& name) { - for (const webrtc::Codec& engine_codec : engine_.LegacySendCodecs()) { + for (const Codec& engine_codec : engine_.LegacySendCodecs()) { if (absl::EqualsIgnoreCase(name, engine_codec.name)) return engine_codec; } // This point should never be reached. ADD_FAILURE() << "Unrecognized codec name: " << name; - return webrtc::CreateVideoCodec(0, ""); + return CreateVideoCodec(0, ""); } Codec DefaultCodec() { return GetEngineCodec("VP8"); } @@ -3025,9 +3010,9 @@ class WebRtcVideoChannelTest : public WebRtcVideoEngineTest { } if (with_rtx) { return AddSendStream( - webrtc::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs)); + CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs)); } - return AddSendStream(webrtc::CreateSimStreamParams("cname", ssrcs)); + return AddSendStream(CreateSimStreamParams("cname", ssrcs)); } int GetMaxEncoderBitrate() { @@ -3107,16 +3092,16 @@ TEST_F(WebRtcVideoChannelTest, RecvStreamWithSimAndRtx) { // Send side. const std::vector ssrcs = MAKE_VECTOR(kSsrcs1); const std::vector rtx_ssrcs = MAKE_VECTOR(kRtxSsrcs1); - FakeVideoSendStream* send_stream = AddSendStream( - webrtc::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs)); + FakeVideoSendStream* send_stream = + AddSendStream(CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs)); ASSERT_EQ(rtx_ssrcs.size(), send_stream->GetConfig().rtp.rtx.ssrcs.size()); for (size_t i = 0; i < rtx_ssrcs.size(); ++i) EXPECT_EQ(rtx_ssrcs[i], send_stream->GetConfig().rtp.rtx.ssrcs[i]); // Receiver side. - FakeVideoReceiveStream* recv_stream = AddRecvStream( - webrtc::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs)); + FakeVideoReceiveStream* recv_stream = + AddRecvStream(CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs)); EXPECT_FALSE( recv_stream->GetConfig().rtp.rtx_associated_payload_types.empty()); EXPECT_TRUE(VerifyRtxReceiveAssociations(recv_stream->GetConfig())) @@ -3544,7 +3529,7 @@ TEST_F(WebRtcVideoChannelTest, ReconfiguresEncodersWhenNotSending) { EXPECT_EQ(0u, streams[0].height); FrameForwarder frame_forwarder; - FakeFrameSource frame_source(1280, 720, webrtc::kNumMicrosecsPerSec / 30); + FakeFrameSource frame_source(1280, 720, kNumMicrosecsPerSec / 30); EXPECT_TRUE( send_channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder)); @@ -3567,7 +3552,7 @@ TEST_F(WebRtcVideoChannelTest, UsesCorrectSettingsForScreencast) { AddSendStream(); FrameForwarder frame_forwarder; - FakeFrameSource frame_source(1280, 720, webrtc::kNumMicrosecsPerSec / 30); + FakeFrameSource frame_source(1280, 720, kNumMicrosecsPerSec / 30); VideoOptions min_bitrate_options; min_bitrate_options.screencast_min_bitrate_kbps = kScreenshareMinBitrateKbps; EXPECT_TRUE(send_channel_->SetVideoSend(last_ssrc_, &min_bitrate_options, @@ -3627,7 +3612,7 @@ TEST_F(WebRtcVideoChannelTest, VideoOptions options; options.is_screencast = true; FrameForwarder frame_forwarder; - FakeFrameSource frame_source(1280, 720, webrtc::kNumMicrosecsPerSec / 30); + FakeFrameSource frame_source(1280, 720, kNumMicrosecsPerSec / 30); EXPECT_TRUE( send_channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder)); EXPECT_TRUE(send_channel_->SetSend(true)); @@ -3993,7 +3978,7 @@ TEST_F(Vp9SettingsTest, MultipleSsrcsEnablesSvc) { std::vector ssrcs = MAKE_VECTOR(kSsrcs3); FakeVideoSendStream* stream = - AddSendStream(webrtc::CreateSimStreamParams("cname", ssrcs)); + AddSendStream(CreateSimStreamParams("cname", ssrcs)); VideoSendStream::Config config = stream->GetConfig().Copy(); @@ -4022,7 +4007,7 @@ TEST_F(Vp9SettingsTest, SvcModeCreatesSingleRtpStream) { std::vector ssrcs = MAKE_VECTOR(kSsrcs3); FakeVideoSendStream* stream = - AddSendStream(webrtc::CreateSimStreamParams("cname", ssrcs)); + AddSendStream(CreateSimStreamParams("cname", ssrcs)); VideoSendStream::Config config = stream->GetConfig().Copy(); @@ -4053,7 +4038,7 @@ TEST_F(Vp9SettingsTest, AllEncodingParametersCopied) { std::vector ssrcs = MAKE_VECTOR(kSsrcs3); FakeVideoSendStream* stream = - AddSendStream(webrtc::CreateSimStreamParams("cname", ssrcs)); + AddSendStream(CreateSimStreamParams("cname", ssrcs)); RtpParameters parameters = send_channel_->GetRtpSendParameters(ssrcs[0]); ASSERT_EQ(kNumSpatialLayers, parameters.encodings.size()); @@ -4084,7 +4069,7 @@ TEST_F(Vp9SettingsTest, MaxBitrateDeterminedBySvcResolutions) { std::vector ssrcs = MAKE_VECTOR(kSsrcs3); FakeVideoSendStream* stream = - AddSendStream(webrtc::CreateSimStreamParams("cname", ssrcs)); + AddSendStream(CreateSimStreamParams("cname", ssrcs)); VideoSendStream::Config config = stream->GetConfig().Copy(); @@ -4128,7 +4113,7 @@ TEST_F(Vp9SettingsTest, Vp9SvcTargetBitrateCappedByMax) { std::vector ssrcs = MAKE_VECTOR(kSsrcs3); FakeVideoSendStream* stream = - AddSendStream(webrtc::CreateSimStreamParams("cname", ssrcs)); + AddSendStream(CreateSimStreamParams("cname", ssrcs)); VideoSendStream::Config config = stream->GetConfig().Copy(); @@ -4237,7 +4222,7 @@ INSTANTIATE_TEST_SUITE_P( TEST_F(WebRtcVideoChannelTest, VerifyMinBitrate) { std::vector streams = AddSendStream()->GetVideoStreams(); ASSERT_EQ(1u, streams.size()); - EXPECT_EQ(webrtc::kDefaultMinVideoBitrateBps, streams[0].min_bitrate_bps); + EXPECT_EQ(kDefaultMinVideoBitrateBps, streams[0].min_bitrate_bps); } TEST_F(WebRtcVideoChannelTest, VerifyMinBitrateWithForcedFallbackFieldTrial) { @@ -4430,7 +4415,7 @@ TEST_F(WebRtcVideoChannelTest, EstimatesNtpStartTimeCorrectly) { .set_video_frame_buffer(CreateBlackFrameBuffer(4, 4)) .set_rtp_timestamp(kInitialTimestamp) .set_timestamp_us(0) - .set_rotation(webrtc::kVideoRotation_0) + .set_rotation(kVideoRotation_0) .build(); // Initial NTP time is not available on the first frame, but should still be // able to be estimated. @@ -4470,8 +4455,8 @@ TEST_F(WebRtcVideoChannelTest, SetDefaultSendCodecs) { // Using a RTX setup to verify that the default RTX payload type is good. const std::vector ssrcs = MAKE_VECTOR(kSsrcs1); const std::vector rtx_ssrcs = MAKE_VECTOR(kRtxSsrcs1); - FakeVideoSendStream* stream = AddSendStream( - webrtc::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs)); + FakeVideoSendStream* stream = + AddSendStream(CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs)); VideoSendStream::Config config = stream->GetConfig().Copy(); // Make sure NACK and FEC are enabled on the correct payload types. @@ -4520,9 +4505,8 @@ TEST_F(WebRtcVideoChannelTest, FlexfecSendCodecWithoutSsrcNotExposedByDefault) { } TEST_F(WebRtcVideoChannelTest, FlexfecSendCodecWithSsrcNotExposedByDefault) { - FakeVideoSendStream* stream = - AddSendStream(webrtc::CreatePrimaryWithFecFrStreamParams( - "cname", kSsrcs1[0], kFlexfecSsrc)); + FakeVideoSendStream* stream = AddSendStream( + CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc)); VideoSendStream::Config config = stream->GetConfig().Copy(); EXPECT_EQ(-1, config.rtp.flexfec.payload_type); @@ -4539,8 +4523,8 @@ TEST_F(WebRtcVideoChannelTest, FlexfecRecvCodecWithoutSsrcNotExposedByDefault) { } TEST_F(WebRtcVideoChannelTest, FlexfecRecvCodecWithSsrcExposedByDefault) { - AddRecvStream(webrtc::CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], - kFlexfecSsrc)); + AddRecvStream( + CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc)); const std::vector& streams = fake_call_->GetFlexfecReceiveStreams(); @@ -4558,7 +4542,7 @@ class WebRtcVideoChannelFlexfecRecvTest : public WebRtcVideoChannelTest { TEST_F(WebRtcVideoChannelFlexfecRecvTest, DefaultFlexfecCodecHasRembFeedbackParam) { - EXPECT_TRUE(webrtc::HasRemb(GetEngineCodec("flexfec-03"))); + EXPECT_TRUE(HasRemb(GetEngineCodec("flexfec-03"))); } TEST_F(WebRtcVideoChannelFlexfecRecvTest, SetDefaultRecvCodecsWithoutSsrc) { @@ -4579,8 +4563,8 @@ TEST_F(WebRtcVideoChannelFlexfecRecvTest, SetDefaultRecvCodecsWithoutSsrc) { } TEST_F(WebRtcVideoChannelFlexfecRecvTest, SetDefaultRecvCodecsWithSsrc) { - AddRecvStream(webrtc::CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], - kFlexfecSsrc)); + AddRecvStream( + CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc)); const std::vector& streams = fake_call_->GetFlexfecReceiveStreams(); @@ -4612,8 +4596,8 @@ TEST_F(WebRtcVideoChannelFlexfecRecvTest, recv_parameters.codecs.push_back(GetEngineCodec("VP8")); ASSERT_TRUE(receive_channel_->SetReceiverParameters(recv_parameters)); - AddRecvStream(webrtc::CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], - kFlexfecSsrc)); + AddRecvStream( + CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc)); EXPECT_EQ(1, fake_call_->GetNumCreatedReceiveStreams()); const std::vector& video_streams = fake_call_->GetVideoReceiveStreams(); @@ -4654,8 +4638,8 @@ TEST_F(WebRtcVideoChannelFlexfecRecvTest, recv_parameters.codecs.push_back(GetEngineCodec("flexfec-03")); ASSERT_TRUE(receive_channel_->SetReceiverParameters(recv_parameters)); - AddRecvStream(webrtc::CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], - kFlexfecSsrc)); + AddRecvStream( + CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc)); EXPECT_EQ(2, fake_call_->GetNumCreatedReceiveStreams()); EXPECT_EQ(1U, fake_call_->GetFlexfecReceiveStreams().size()); const std::vector& video_streams = @@ -4697,8 +4681,8 @@ TEST_F(WebRtcVideoChannelFlexfecRecvTest, DuplicateFlexfecCodecIsDropped) { recv_parameters.codecs.push_back(duplicate); ASSERT_TRUE(receive_channel_->SetReceiverParameters(recv_parameters)); - AddRecvStream(webrtc::CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], - kFlexfecSsrc)); + AddRecvStream( + CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc)); const std::vector& streams = fake_call_->GetFlexfecReceiveStreams(); @@ -4729,9 +4713,8 @@ TEST_F(WebRtcVideoChannelFlexfecSendRecvTest, SetDefaultSendCodecsWithoutSsrc) { } TEST_F(WebRtcVideoChannelFlexfecSendRecvTest, SetDefaultSendCodecsWithSsrc) { - FakeVideoSendStream* stream = - AddSendStream(webrtc::CreatePrimaryWithFecFrStreamParams( - "cname", kSsrcs1[0], kFlexfecSsrc)); + FakeVideoSendStream* stream = AddSendStream( + CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc)); VideoSendStream::Config config = stream->GetConfig().Copy(); EXPECT_EQ(GetEngineCodec("flexfec-03").id, config.rtp.flexfec.payload_type); @@ -4764,8 +4747,8 @@ TEST_F(WebRtcVideoChannelFlexfecSendRecvTest, SetSendCodecsWithoutFec) { } TEST_F(WebRtcVideoChannelFlexfecRecvTest, SetRecvCodecsWithFec) { - AddRecvStream(webrtc::CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], - kFlexfecSsrc)); + AddRecvStream( + CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc)); VideoReceiverParameters recv_parameters; recv_parameters.codecs.push_back(GetEngineCodec("VP8")); @@ -4821,9 +4804,8 @@ TEST_F(WebRtcVideoChannelFlexfecRecvTest, parameters.codecs.push_back(GetEngineCodec("flexfec-03")); ASSERT_TRUE(send_channel_->SetSenderParameters(parameters)); - FakeVideoSendStream* stream = - AddSendStream(webrtc::CreatePrimaryWithFecFrStreamParams( - "cname", kSsrcs1[0], kFlexfecSsrc)); + FakeVideoSendStream* stream = AddSendStream( + CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc)); VideoSendStream::Config config = stream->GetConfig().Copy(); EXPECT_EQ(-1, config.rtp.flexfec.payload_type); @@ -4834,11 +4816,10 @@ TEST_F(WebRtcVideoChannelFlexfecRecvTest, TEST_F(WebRtcVideoChannelTest, SetSendCodecRejectsRtxWithoutAssociatedPayloadType) { const int kUnusedPayloadType = 127; - EXPECT_FALSE( - webrtc::FindCodecById(engine_.LegacySendCodecs(), kUnusedPayloadType)); + EXPECT_FALSE(FindCodecById(engine_.LegacySendCodecs(), kUnusedPayloadType)); VideoSenderParameters parameters; - Codec rtx_codec = webrtc::CreateVideoCodec(kUnusedPayloadType, "rtx"); + Codec rtx_codec = CreateVideoCodec(kUnusedPayloadType, "rtx"); parameters.codecs.push_back(rtx_codec); EXPECT_FALSE(send_channel_->SetSenderParameters(parameters)) << "RTX codec without associated payload type should be rejected."; @@ -4848,13 +4829,11 @@ TEST_F(WebRtcVideoChannelTest, SetSendCodecRejectsRtxWithoutMatchingVideoCodec) { const int kUnusedPayloadType1 = 126; const int kUnusedPayloadType2 = 127; - EXPECT_FALSE( - webrtc::FindCodecById(engine_.LegacySendCodecs(), kUnusedPayloadType1)); - EXPECT_FALSE( - webrtc::FindCodecById(engine_.LegacySendCodecs(), kUnusedPayloadType2)); + EXPECT_FALSE(FindCodecById(engine_.LegacySendCodecs(), kUnusedPayloadType1)); + EXPECT_FALSE(FindCodecById(engine_.LegacySendCodecs(), kUnusedPayloadType2)); { - Codec rtx_codec = webrtc::CreateVideoRtxCodec(kUnusedPayloadType1, - GetEngineCodec("VP8").id); + Codec rtx_codec = + CreateVideoRtxCodec(kUnusedPayloadType1, GetEngineCodec("VP8").id); VideoSenderParameters parameters; parameters.codecs.push_back(GetEngineCodec("VP8")); parameters.codecs.push_back(rtx_codec); @@ -4862,7 +4841,7 @@ TEST_F(WebRtcVideoChannelTest, } { Codec rtx_codec = - webrtc::CreateVideoRtxCodec(kUnusedPayloadType1, kUnusedPayloadType2); + CreateVideoRtxCodec(kUnusedPayloadType1, kUnusedPayloadType2); VideoSenderParameters parameters; parameters.codecs.push_back(GetEngineCodec("VP8")); parameters.codecs.push_back(rtx_codec); @@ -4874,10 +4853,8 @@ TEST_F(WebRtcVideoChannelTest, TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithChangedRtxPayloadType) { const int kUnusedPayloadType1 = 126; const int kUnusedPayloadType2 = 127; - EXPECT_FALSE( - webrtc::FindCodecById(engine_.LegacySendCodecs(), kUnusedPayloadType1)); - EXPECT_FALSE( - webrtc::FindCodecById(engine_.LegacySendCodecs(), kUnusedPayloadType2)); + EXPECT_FALSE(FindCodecById(engine_.LegacySendCodecs(), kUnusedPayloadType1)); + EXPECT_FALSE(FindCodecById(engine_.LegacySendCodecs(), kUnusedPayloadType2)); // SSRCs for RTX. StreamParams params = StreamParams::CreateLegacy(kSsrcs1[0]); @@ -4887,7 +4864,7 @@ TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithChangedRtxPayloadType) { // Original payload type for RTX. VideoSenderParameters parameters; parameters.codecs.push_back(GetEngineCodec("VP8")); - Codec rtx_codec = webrtc::CreateVideoCodec(kUnusedPayloadType1, "rtx"); + Codec rtx_codec = CreateVideoCodec(kUnusedPayloadType1, "rtx"); rtx_codec.SetParam("apt", GetEngineCodec("VP8").id); parameters.codecs.push_back(rtx_codec); EXPECT_TRUE(send_channel_->SetSenderParameters(parameters)); @@ -4936,9 +4913,8 @@ TEST_F(WebRtcVideoChannelFlexfecSendRecvTest, parameters.codecs.push_back(GetEngineCodec("flexfec-03")); ASSERT_TRUE(send_channel_->SetSenderParameters(parameters)); - FakeVideoSendStream* stream = - AddSendStream(webrtc::CreatePrimaryWithFecFrStreamParams( - "cname", kSsrcs1[0], kFlexfecSsrc)); + FakeVideoSendStream* stream = AddSendStream( + CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc)); VideoSendStream::Config config = stream->GetConfig().Copy(); EXPECT_EQ(GetEngineCodec("flexfec-03").id, config.rtp.flexfec.payload_type); @@ -4957,7 +4933,7 @@ TEST_F(WebRtcVideoChannelFlexfecSendRecvTest, TEST_F(WebRtcVideoChannelTest, SetSendCodecsChangesExistingStreams) { VideoSenderParameters parameters; - Codec codec = webrtc::CreateVideoCodec(100, "VP8"); + Codec codec = CreateVideoCodec(100, "VP8"); codec.SetParam(kCodecParamMaxQuantization, kDefaultVideoMaxQpVpx); parameters.codecs.push_back(codec); @@ -5012,8 +4988,8 @@ TEST_F(WebRtcVideoChannelTest, SetSenderParametersRemovesSelectedCodecFromRtpParameters) { EXPECT_TRUE(AddSendStream()); VideoSenderParameters parameters; - parameters.codecs.push_back(webrtc::CreateVideoCodec(100, "VP8")); - parameters.codecs.push_back(webrtc::CreateVideoCodec(100, "VP9")); + parameters.codecs.push_back(CreateVideoCodec(100, "VP8")); + parameters.codecs.push_back(CreateVideoCodec(100, "VP9")); send_channel_->SetSenderParameters(parameters); RtpParameters initial_params = @@ -5031,7 +5007,7 @@ TEST_F(WebRtcVideoChannelTest, send_channel_->SetRtpSendParameters(last_ssrc_, initial_params).ok()); parameters.codecs.clear(); - parameters.codecs.push_back(webrtc::CreateVideoCodec(100, "VP8")); + parameters.codecs.push_back(CreateVideoCodec(100, "VP8")); send_channel_->SetSenderParameters(parameters); // Since VP9 is no longer negotiated, the RTP parameters should not have a @@ -5221,8 +5197,8 @@ TEST_F(WebRtcVideoChannelTest, ASSERT_TRUE(send_channel_->SetSenderParameters(parameters)); send_channel_->SetSend(true); - FakeVideoSendStream* stream = AddSendStream( - webrtc::CreateSimStreamParams("cname", MAKE_VECTOR(kSsrcs3))); + FakeVideoSendStream* stream = + AddSendStream(CreateSimStreamParams("cname", MAKE_VECTOR(kSsrcs3))); // Send a frame to make sure this scales up to >1 stream (simulcast). FrameForwarder frame_forwarder; @@ -5233,7 +5209,7 @@ TEST_F(WebRtcVideoChannelTest, std::vector streams = stream->GetVideoStreams(); ASSERT_GT(streams.size(), 1u) << "Without simulcast this test doesn't make sense."; - int initial_max_bitrate_bps = webrtc::GetTotalMaxBitrate(streams).bps(); + int initial_max_bitrate_bps = GetTotalMaxBitrate(streams).bps(); EXPECT_GT(initial_max_bitrate_bps, 0); parameters.max_bandwidth_bps = initial_max_bitrate_bps * 2; @@ -5241,7 +5217,7 @@ TEST_F(WebRtcVideoChannelTest, // Insert a frame to update the encoder config. frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame()); streams = stream->GetVideoStreams(); - int increased_max_bitrate_bps = webrtc::GetTotalMaxBitrate(streams).bps(); + int increased_max_bitrate_bps = GetTotalMaxBitrate(streams).bps(); EXPECT_EQ(initial_max_bitrate_bps * 2, increased_max_bitrate_bps); EXPECT_TRUE(send_channel_->SetVideoSend(kSsrcs3[0], nullptr, nullptr)); @@ -5313,14 +5289,12 @@ TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithOnlyVp8) { TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithRtx) { const int kUnusedPayloadType1 = 126; const int kUnusedPayloadType2 = 127; - EXPECT_FALSE( - webrtc::FindCodecById(engine_.LegacyRecvCodecs(), kUnusedPayloadType1)); - EXPECT_FALSE( - webrtc::FindCodecById(engine_.LegacyRecvCodecs(), kUnusedPayloadType2)); + EXPECT_FALSE(FindCodecById(engine_.LegacyRecvCodecs(), kUnusedPayloadType1)); + EXPECT_FALSE(FindCodecById(engine_.LegacyRecvCodecs(), kUnusedPayloadType2)); VideoReceiverParameters parameters; parameters.codecs.push_back(GetEngineCodec("VP8")); - Codec rtx_codec = webrtc::CreateVideoCodec(kUnusedPayloadType1, "rtx"); + Codec rtx_codec = CreateVideoCodec(kUnusedPayloadType1, "rtx"); parameters.codecs.push_back(rtx_codec); EXPECT_FALSE(receive_channel_->SetReceiverParameters(parameters)) << "RTX codec without associated payload should be rejected."; @@ -5332,7 +5306,7 @@ TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithRtx) { parameters.codecs[1].SetParam("apt", GetEngineCodec("VP8").id); EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); - Codec rtx_codec2 = webrtc::CreateVideoCodec(kUnusedPayloadType2, "rtx"); + Codec rtx_codec2 = CreateVideoCodec(kUnusedPayloadType2, "rtx"); rtx_codec2.SetParam("apt", rtx_codec.id); parameters.codecs.push_back(rtx_codec2); @@ -5382,9 +5356,9 @@ TEST_F(WebRtcVideoChannelTest, DuplicateUlpfecCodecIsDropped) { VideoReceiverParameters parameters; parameters.codecs.push_back(GetEngineCodec("VP8")); parameters.codecs.push_back( - webrtc::CreateVideoCodec(kFirstUlpfecPayloadType, kUlpfecCodecName)); + CreateVideoCodec(kFirstUlpfecPayloadType, kUlpfecCodecName)); parameters.codecs.push_back( - webrtc::CreateVideoCodec(kSecondUlpfecPayloadType, kUlpfecCodecName)); + CreateVideoCodec(kSecondUlpfecPayloadType, kUlpfecCodecName)); ASSERT_TRUE(receive_channel_->SetReceiverParameters(parameters)); FakeVideoReceiveStream* recv_stream = AddRecvStream(); @@ -5399,9 +5373,9 @@ TEST_F(WebRtcVideoChannelTest, DuplicateRedCodecIsDropped) { VideoReceiverParameters parameters; parameters.codecs.push_back(GetEngineCodec("VP8")); parameters.codecs.push_back( - webrtc::CreateVideoCodec(kFirstRedPayloadType, kRedCodecName)); + CreateVideoCodec(kFirstRedPayloadType, kRedCodecName)); parameters.codecs.push_back( - webrtc::CreateVideoCodec(kSecondRedPayloadType, kRedCodecName)); + CreateVideoCodec(kSecondRedPayloadType, kRedCodecName)); ASSERT_TRUE(receive_channel_->SetReceiverParameters(parameters)); FakeVideoReceiveStream* recv_stream = AddRecvStream(); @@ -5412,10 +5386,8 @@ TEST_F(WebRtcVideoChannelTest, DuplicateRedCodecIsDropped) { TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithChangedRtxPayloadType) { const int kUnusedPayloadType1 = 126; const int kUnusedPayloadType2 = 127; - EXPECT_FALSE( - webrtc::FindCodecById(engine_.LegacyRecvCodecs(), kUnusedPayloadType1)); - EXPECT_FALSE( - webrtc::FindCodecById(engine_.LegacyRecvCodecs(), kUnusedPayloadType2)); + EXPECT_FALSE(FindCodecById(engine_.LegacyRecvCodecs(), kUnusedPayloadType1)); + EXPECT_FALSE(FindCodecById(engine_.LegacyRecvCodecs(), kUnusedPayloadType2)); // SSRCs for RTX. StreamParams params = StreamParams::CreateLegacy(kSsrcs1[0]); @@ -5425,7 +5397,7 @@ TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithChangedRtxPayloadType) { // Original payload type for RTX. VideoReceiverParameters parameters; parameters.codecs.push_back(GetEngineCodec("VP8")); - Codec rtx_codec = webrtc::CreateVideoCodec(kUnusedPayloadType1, "rtx"); + Codec rtx_codec = CreateVideoCodec(kUnusedPayloadType1, "rtx"); rtx_codec.SetParam("apt", GetEngineCodec("VP8").id); parameters.codecs.push_back(rtx_codec); EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); @@ -5456,10 +5428,8 @@ TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithChangedRtxPayloadType) { TEST_F(WebRtcVideoChannelTest, SetRecvCodecsRtxWithRtxTime) { const int kUnusedPayloadType1 = 126; const int kUnusedPayloadType2 = 127; - EXPECT_FALSE( - webrtc::FindCodecById(engine_.LegacyRecvCodecs(), kUnusedPayloadType1)); - EXPECT_FALSE( - webrtc::FindCodecById(engine_.LegacyRecvCodecs(), kUnusedPayloadType2)); + EXPECT_FALSE(FindCodecById(engine_.LegacyRecvCodecs(), kUnusedPayloadType1)); + EXPECT_FALSE(FindCodecById(engine_.LegacyRecvCodecs(), kUnusedPayloadType2)); // SSRCs for RTX. StreamParams params = StreamParams::CreateLegacy(kSsrcs1[0]); @@ -5469,7 +5439,7 @@ TEST_F(WebRtcVideoChannelTest, SetRecvCodecsRtxWithRtxTime) { // Payload type for RTX. VideoReceiverParameters parameters; parameters.codecs.push_back(GetEngineCodec("VP8")); - Codec rtx_codec = webrtc::CreateVideoCodec(kUnusedPayloadType1, "rtx"); + Codec rtx_codec = CreateVideoCodec(kUnusedPayloadType1, "rtx"); rtx_codec.SetParam("apt", GetEngineCodec("VP8").id); parameters.codecs.push_back(rtx_codec); EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); @@ -5546,7 +5516,7 @@ TEST_F(WebRtcVideoChannelTest, SetRecvCodecsAcceptDefaultCodecs) { TEST_F(WebRtcVideoChannelTest, SetRecvCodecsRejectUnsupportedCodec) { VideoReceiverParameters parameters; parameters.codecs.push_back(GetEngineCodec("VP8")); - parameters.codecs.push_back(webrtc::CreateVideoCodec(101, "WTF3")); + parameters.codecs.push_back(CreateVideoCodec(101, "WTF3")); EXPECT_FALSE(receive_channel_->SetReceiverParameters(parameters)); } @@ -5579,8 +5549,8 @@ TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithoutFecDisablesFec) { } TEST_F(WebRtcVideoChannelFlexfecRecvTest, SetRecvParamsWithoutFecDisablesFec) { - AddRecvStream(webrtc::CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], - kFlexfecSsrc)); + AddRecvStream( + CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc)); const std::vector& streams = fake_call_->GetFlexfecReceiveStreams(); @@ -5627,8 +5597,8 @@ TEST_F(WebRtcVideoChannelTest, SetSendParamsWithFecEnablesFec) { TEST_F(WebRtcVideoChannelFlexfecSendRecvTest, SetSendRecvParamsWithFecEnablesFec) { - AddRecvStream(webrtc::CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], - kFlexfecSsrc)); + AddRecvStream( + CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc)); const std::vector& streams = fake_call_->GetFlexfecReceiveStreams(); @@ -5848,17 +5818,16 @@ TEST_F(WebRtcVideoChannelTest, TestSetRecvRtcpReducedSize) { } TEST_F(WebRtcVideoChannelTest, OnReadyToSendSignalsNetworkState) { - EXPECT_EQ(webrtc::kNetworkUp, fake_call_->GetNetworkState(MediaType::VIDEO)); - EXPECT_EQ(webrtc::kNetworkUp, fake_call_->GetNetworkState(MediaType::AUDIO)); + EXPECT_EQ(kNetworkUp, fake_call_->GetNetworkState(MediaType::VIDEO)); + EXPECT_EQ(kNetworkUp, fake_call_->GetNetworkState(MediaType::AUDIO)); send_channel_->OnReadyToSend(false); - EXPECT_EQ(webrtc::kNetworkDown, - fake_call_->GetNetworkState(MediaType::VIDEO)); - EXPECT_EQ(webrtc::kNetworkUp, fake_call_->GetNetworkState(MediaType::AUDIO)); + EXPECT_EQ(kNetworkDown, fake_call_->GetNetworkState(MediaType::VIDEO)); + EXPECT_EQ(kNetworkUp, fake_call_->GetNetworkState(MediaType::AUDIO)); send_channel_->OnReadyToSend(true); - EXPECT_EQ(webrtc::kNetworkUp, fake_call_->GetNetworkState(MediaType::VIDEO)); - EXPECT_EQ(webrtc::kNetworkUp, fake_call_->GetNetworkState(MediaType::AUDIO)); + EXPECT_EQ(kNetworkUp, fake_call_->GetNetworkState(MediaType::VIDEO)); + EXPECT_EQ(kNetworkUp, fake_call_->GetNetworkState(MediaType::AUDIO)); } TEST_F(WebRtcVideoChannelTest, GetStatsReportsSentCodecName) { @@ -6339,7 +6308,7 @@ TEST_F(WebRtcVideoChannelTest, // Create simulcast stream from both SSRCs. // `kSsrc1` is the "main" ssrc used for getting parameters. FakeVideoSendStream* stream = - AddSendStream(webrtc::CreateSimStreamParams("cname", {kSsrc1, kSsrc2})); + AddSendStream(CreateSimStreamParams("cname", {kSsrc1, kSsrc2})); RtpParameters parameters = send_channel_->GetRtpSendParameters(kSsrc1); ASSERT_EQ(2u, parameters.encodings.size()); @@ -6378,8 +6347,8 @@ TEST_F(WebRtcVideoChannelTest, OutboundRtpIsActiveComesFromAnyEncodingInSvc) { // Configuring SVC is done the same way that simulcast is configured, the only // difference is that the VP9 codec is used. This triggers special hacks that // we depend on because we don't have a proper SVC API yet. - FakeVideoSendStream* stream = AddSendStream( - webrtc::CreateSimStreamParams("cname", {kSsrc1, kSsrc2, kSsrc3})); + FakeVideoSendStream* stream = + AddSendStream(CreateSimStreamParams("cname", {kSsrc1, kSsrc2, kSsrc3})); // Expect that we got SVC. EXPECT_EQ(stream->GetEncoderConfig().number_of_streams, 1u); VideoCodecVP9 vp9_settings; @@ -6973,7 +6942,7 @@ TEST_F(WebRtcVideoChannelTest, DefaultReceiveStreamReconfiguresToUseRtx) { << "Default receive stream should not have configured RTX"; EXPECT_TRUE(receive_channel_->AddRecvStream( - webrtc::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs))); + CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs))); ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size()) << "AddRecvStream should have reconfigured, not added a new receiver."; recv_stream = fake_call_->GetVideoReceiveStreams()[0]; @@ -6993,8 +6962,7 @@ TEST_F(WebRtcVideoChannelTest, RejectsAddingStreamsWithMissingSsrcsForRtx) { const std::vector ssrcs = MAKE_VECTOR(kSsrcs1); const std::vector rtx_ssrcs = MAKE_VECTOR(kRtxSsrcs1); - StreamParams sp = - webrtc::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs); + StreamParams sp = CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs); sp.ssrcs = ssrcs; // Without RTXs, this is the important part. EXPECT_FALSE(send_channel_->AddSendStream(sp)); @@ -7007,8 +6975,7 @@ TEST_F(WebRtcVideoChannelTest, RejectsAddingStreamsWithOverlappingRtxSsrcs) { const std::vector ssrcs = MAKE_VECTOR(kSsrcs1); const std::vector rtx_ssrcs = MAKE_VECTOR(kRtxSsrcs1); - StreamParams sp = - webrtc::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs); + StreamParams sp = CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs); EXPECT_TRUE(send_channel_->AddSendStream(sp)); EXPECT_TRUE(receive_channel_->AddRecvStream(sp)); @@ -7033,14 +7000,13 @@ TEST_F(WebRtcVideoChannelTest, EXPECT_TRUE(send_channel_->SetSenderParameters(send_parameters_)); StreamParams sp = - webrtc::CreateSimStreamParams("cname", MAKE_VECTOR(kFirstStreamSsrcs)); + CreateSimStreamParams("cname", MAKE_VECTOR(kFirstStreamSsrcs)); EXPECT_TRUE(send_channel_->AddSendStream(sp)); EXPECT_TRUE(receive_channel_->AddRecvStream(sp)); // One of the SSRCs is already used in previous streams, using it should fail. - sp = webrtc::CreateSimStreamParams("cname", - MAKE_VECTOR(kOverlappingStreamSsrcs)); + sp = CreateSimStreamParams("cname", MAKE_VECTOR(kOverlappingStreamSsrcs)); EXPECT_FALSE(send_channel_->AddSendStream(sp)); EXPECT_FALSE(receive_channel_->AddRecvStream(sp)); @@ -7058,7 +7024,7 @@ TEST_F(WebRtcVideoChannelTest, ReportsSsrcGroupsInStats) { static const uint32_t kSenderSsrcs[] = {4, 7, 10}; static const uint32_t kSenderRtxSsrcs[] = {5, 8, 11}; - StreamParams sender_sp = webrtc::CreateSimWithRtxStreamParams( + StreamParams sender_sp = CreateSimWithRtxStreamParams( "cname", MAKE_VECTOR(kSenderSsrcs), MAKE_VECTOR(kSenderRtxSsrcs)); EXPECT_TRUE(send_channel_->AddSendStream(sender_sp)); @@ -7066,7 +7032,7 @@ TEST_F(WebRtcVideoChannelTest, ReportsSsrcGroupsInStats) { static const uint32_t kReceiverSsrcs[] = {3}; static const uint32_t kReceiverRtxSsrcs[] = {2}; - StreamParams receiver_sp = webrtc::CreateSimWithRtxStreamParams( + StreamParams receiver_sp = CreateSimWithRtxStreamParams( "cname", MAKE_VECTOR(kReceiverSsrcs), MAKE_VECTOR(kReceiverRtxSsrcs)); EXPECT_TRUE(receive_channel_->AddRecvStream(receiver_sp)); @@ -7522,12 +7488,11 @@ void WebRtcVideoChannelTest::TestReceiveUnsignaledSsrcPacket( uint8_t payload_type, bool expect_created_receive_stream) { // kRedRtxPayloadType must currently be unused. - EXPECT_FALSE( - webrtc::FindCodecById(engine_.LegacyRecvCodecs(), kRedRtxPayloadType)); + EXPECT_FALSE(FindCodecById(engine_.LegacyRecvCodecs(), kRedRtxPayloadType)); // Add a RED RTX codec. Codec red_rtx_codec = - webrtc::CreateVideoRtxCodec(kRedRtxPayloadType, GetEngineCodec("red").id); + CreateVideoRtxCodec(kRedRtxPayloadType, GetEngineCodec("red").id); recv_parameters_.codecs.push_back(red_rtx_codec); EXPECT_TRUE(receive_channel_->SetReceiverParameters(recv_parameters_)); @@ -7694,7 +7659,7 @@ TEST_F(WebRtcVideoChannelTest, ReceiveDifferentUnsignaledSsrc) { .set_video_frame_buffer(CreateBlackFrameBuffer(4, 4)) .set_rtp_timestamp(100) .set_timestamp_us(0) - .set_rotation(webrtc::kVideoRotation_0) + .set_rotation(kVideoRotation_0) .build(); recv_stream->InjectFrame(video_frame); EXPECT_EQ(1, renderer.num_rendered_frames()); @@ -7713,7 +7678,7 @@ TEST_F(WebRtcVideoChannelTest, ReceiveDifferentUnsignaledSsrc) { .set_video_frame_buffer(CreateBlackFrameBuffer(4, 4)) .set_rtp_timestamp(200) .set_timestamp_us(0) - .set_rotation(webrtc::kVideoRotation_0) + .set_rotation(kVideoRotation_0) .build(); recv_stream->InjectFrame(video_frame2); EXPECT_EQ(2, renderer.num_rendered_frames()); @@ -7835,13 +7800,13 @@ TEST_F(WebRtcVideoChannelTest, // we are just testing the behavior of // EncoderStreamFactory::CreateEncoderStreams. ASSERT_EQ(1UL, stream->GetVideoStreams().size()); - EXPECT_EQ(webrtc::kDefaultMinVideoBitrateBps, + EXPECT_EQ(kDefaultMinVideoBitrateBps, stream->GetVideoStreams()[0].min_bitrate_bps); // Set a low max bitrate & check that VideoStream.min_bitrate_bps is limited // by this amount. parameters = send_channel_->GetRtpSendParameters(last_ssrc_); - int low_max_bitrate_bps = webrtc::kDefaultMinVideoBitrateBps - 1000; + int low_max_bitrate_bps = kDefaultMinVideoBitrateBps - 1000; parameters.encodings[0].max_bitrate_bps = low_max_bitrate_bps; EXPECT_TRUE(send_channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); @@ -7879,7 +7844,7 @@ TEST_F(WebRtcVideoChannelTest, ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps); ASSERT_TRUE(send_channel_->SetSenderParameters(send_parameters_)); ASSERT_EQ(1UL, stream->GetVideoStreams().size()); - EXPECT_EQ(webrtc::kDefaultMinVideoBitrateBps, + EXPECT_EQ(kDefaultMinVideoBitrateBps, stream->GetVideoStreams()[0].min_bitrate_bps); EXPECT_EQ(send_parameters_.max_bandwidth_bps, stream->GetVideoStreams()[0].max_bitrate_bps); @@ -7964,7 +7929,7 @@ TEST_F(WebRtcVideoChannelTest, TEST_F(WebRtcVideoChannelTest, CannotSetSimulcastRtpSendParametersWithIncorrectNumberOfEncodings) { std::vector ssrcs = MAKE_VECTOR(kSsrcs3); - StreamParams sp = webrtc::CreateSimStreamParams("cname", ssrcs); + StreamParams sp = CreateSimStreamParams("cname", ssrcs); AddSendStream(sp); RtpParameters parameters = send_channel_->GetRtpSendParameters(last_ssrc_); @@ -7994,8 +7959,7 @@ TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersInvalidBitratePriority) { AddSendStream(); RtpParameters parameters = send_channel_->GetRtpSendParameters(last_ssrc_); EXPECT_EQ(1UL, parameters.encodings.size()); - EXPECT_EQ(webrtc::kDefaultBitratePriority, - parameters.encodings[0].bitrate_priority); + EXPECT_EQ(kDefaultBitratePriority, parameters.encodings[0].bitrate_priority); parameters.encodings[0].bitrate_priority = 0; EXPECT_FALSE( @@ -8011,8 +7975,7 @@ TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersPriorityOneStream) { AddSendStream(); RtpParameters parameters = send_channel_->GetRtpSendParameters(last_ssrc_); EXPECT_EQ(1UL, parameters.encodings.size()); - EXPECT_EQ(webrtc::kDefaultBitratePriority, - parameters.encodings[0].bitrate_priority); + EXPECT_EQ(kDefaultBitratePriority, parameters.encodings[0].bitrate_priority); // Change the value and set it on the VideoChannel. double new_bitrate_priority = 2.0; @@ -8047,7 +8010,7 @@ TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersPriorityOneStream) { TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersPrioritySimulcastStreams) { // Create the stream params with multiple ssrcs for simulcast. std::vector ssrcs = MAKE_VECTOR(kSsrcs3); - StreamParams stream_params = webrtc::CreateSimStreamParams("cname", ssrcs); + StreamParams stream_params = CreateSimStreamParams("cname", ssrcs); AddSendStream(stream_params); uint32_t primary_ssrc = stream_params.first_ssrc(); @@ -8059,15 +8022,13 @@ TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersPrioritySimulcastStreams) { EXPECT_TRUE( send_channel_->SetVideoSend(primary_ssrc, &options, &frame_forwarder)); send_channel_->SetSend(true); - frame_forwarder.IncomingCapturedFrame( - frame_source_.GetFrame(1920, 1080, VideoRotation::kVideoRotation_0, - webrtc::kNumMicrosecsPerSec / 30)); + frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame( + 1920, 1080, VideoRotation::kVideoRotation_0, kNumMicrosecsPerSec / 30)); // Get and set the rtp encoding parameters. RtpParameters parameters = send_channel_->GetRtpSendParameters(primary_ssrc); EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size()); - EXPECT_EQ(webrtc::kDefaultBitratePriority, - parameters.encodings[0].bitrate_priority); + EXPECT_EQ(kDefaultBitratePriority, parameters.encodings[0].bitrate_priority); // Change the value and set it on the VideoChannel. double new_bitrate_priority = 2.0; parameters.encodings[0].bitrate_priority = new_bitrate_priority; @@ -8125,7 +8086,7 @@ TEST_P(WebRtcVideoChannelScaleResolutionDownByTest, ScaleResolutionDownBy) { // Set up WebRtcVideoChannel for 3-layer simulcast. encoder_factory_->AddSupportedVideoCodecType(codec_name); VideoSenderParameters parameters; - Codec codec = webrtc::CreateVideoCodec(codec_name); + Codec codec = CreateVideoCodec(codec_name); // Codec ID does not matter, but must be valid. codec.id = 123; parameters.codecs.push_back(codec); @@ -8153,7 +8114,7 @@ TEST_P(WebRtcVideoChannelScaleResolutionDownByTest, ScaleResolutionDownBy) { // (See field trial set at the top of the test.) FakeFrameSource frame_source(test_params.resolution.width, test_params.resolution.height, - webrtc::kNumMicrosecsPerSec / 30); + kNumMicrosecsPerSec / 30); frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); // Ensure the scaling is correct. @@ -8257,7 +8218,7 @@ TEST_F(WebRtcVideoChannelTest, parameters.encodings[0].num_temporal_layers = 0; EXPECT_EQ(RTCErrorType::INVALID_RANGE, send_channel_->SetRtpSendParameters(last_ssrc_, parameters).type()); - parameters.encodings[0].num_temporal_layers = webrtc::kMaxTemporalStreams + 1; + parameters.encodings[0].num_temporal_layers = kMaxTemporalStreams + 1; EXPECT_EQ(RTCErrorType::INVALID_RANGE, send_channel_->SetRtpSendParameters(last_ssrc_, parameters).type()); } @@ -8411,7 +8372,7 @@ TEST_F(WebRtcVideoChannelTest, } TEST_F(WebRtcVideoChannelTest, FallbackForUnsetOrUnsupportedScalabilityMode) { - const absl::InlinedVector + const absl::InlinedVector kSupportedModes = {ScalabilityMode::kL1T1, ScalabilityMode::kL1T2, ScalabilityMode::kL1T3}; @@ -8439,7 +8400,7 @@ TEST_F(WebRtcVideoChannelTest, FallbackForUnsetOrUnsupportedScalabilityMode) { // Verify that the new value is propagated down to the encoder. // Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly. const std::optional kDefaultScalabilityMode = - webrtc::ScalabilityModeFromString(kDefaultScalabilityModeStr); + ScalabilityModeFromString(kDefaultScalabilityModeStr); EXPECT_EQ(2, stream->num_encoder_reconfigurations()); VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy(); EXPECT_EQ(kNumSimulcastStreams, encoder_config.number_of_streams); @@ -8480,7 +8441,7 @@ TEST_F(WebRtcVideoChannelTest, FallbackForUnsetOrUnsupportedScalabilityMode) { TEST_F( WebRtcVideoChannelTest, NoLayeringValueUsedIfModeIsUnsetOrUnsupportedByH265AndDefaultUnsupported) { - const absl::InlinedVector + const absl::InlinedVector kSupportedModes = {ScalabilityMode::kL1T1, ScalabilityMode::kL1T3}; encoder_factory_->AddSupportedVideoCodec( @@ -8510,7 +8471,7 @@ TEST_F( // Verify that the new value is propagated down to the encoder. // Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly. const std::optional kDefaultScalabilityMode = - webrtc::ScalabilityModeFromString(webrtc::kNoLayeringScalabilityModeStr); + ScalabilityModeFromString(kNoLayeringScalabilityModeStr); EXPECT_EQ(2, stream->num_encoder_reconfigurations()); VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy(); EXPECT_EQ(kNumSimulcastStreams, encoder_config.number_of_streams); @@ -8535,10 +8496,10 @@ TEST_F( EXPECT_THAT( parameters.encodings, ElementsAre(Field(&RtpEncodingParameters::scalability_mode, - webrtc::kNoLayeringScalabilityModeStr), + kNoLayeringScalabilityModeStr), Field(&RtpEncodingParameters::scalability_mode, "L1T3"), Field(&RtpEncodingParameters::scalability_mode, - webrtc::kNoLayeringScalabilityModeStr))); + kNoLayeringScalabilityModeStr))); // No parameters changed, encoder should not be reconfigured. EXPECT_TRUE(send_channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); @@ -8777,7 +8738,7 @@ TEST_F(WebRtcVideoChannelTest, // Verify that the new value is propagated down to the encoder. // Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly. const std::optional kDefaultScalabilityMode = - webrtc::ScalabilityModeFromString(kDefaultScalabilityModeStr); + ScalabilityModeFromString(kDefaultScalabilityModeStr); EXPECT_EQ(2, stream->num_encoder_reconfigurations()); VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy(); EXPECT_EQ(1u, encoder_config.number_of_streams); @@ -9084,7 +9045,7 @@ TEST_F(WebRtcVideoChannelTest, BandwidthAboveTotalMaxBitrateGivenToMaxLayer) { // Set max bandwidth equal to total max bitrate. send_parameters_.max_bandwidth_bps = - webrtc::GetTotalMaxBitrate(stream->GetVideoStreams()).bps(); + GetTotalMaxBitrate(stream->GetVideoStreams()).bps(); ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps); ASSERT_TRUE(send_channel_->SetSenderParameters(send_parameters_)); @@ -9095,7 +9056,7 @@ TEST_F(WebRtcVideoChannelTest, BandwidthAboveTotalMaxBitrateGivenToMaxLayer) { // Set max bandwidth above the total max bitrate. send_parameters_.max_bandwidth_bps = - webrtc::GetTotalMaxBitrate(stream->GetVideoStreams()).bps() + 1; + GetTotalMaxBitrate(stream->GetVideoStreams()).bps() + 1; ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps); ASSERT_TRUE(send_channel_->SetSenderParameters(send_parameters_)); @@ -9103,7 +9064,7 @@ TEST_F(WebRtcVideoChannelTest, BandwidthAboveTotalMaxBitrateGivenToMaxLayer) { // max should be given to the highest layer. EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size()); EXPECT_EQ(send_parameters_.max_bandwidth_bps, - webrtc::GetTotalMaxBitrate(stream->GetVideoStreams()).bps()); + GetTotalMaxBitrate(stream->GetVideoStreams()).bps()); EXPECT_EQ(kDefault[2].max_bitrate_bps + 1, stream->GetVideoStreams()[2].max_bitrate_bps); @@ -9132,7 +9093,7 @@ TEST_F(WebRtcVideoChannelTest, // Set max bandwidth above the total max bitrate. send_parameters_.max_bandwidth_bps = - webrtc::GetTotalMaxBitrate(stream->GetVideoStreams()).bps() + 1; + GetTotalMaxBitrate(stream->GetVideoStreams()).bps() + 1; ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps); ASSERT_TRUE(send_channel_->SetSenderParameters(send_parameters_)); @@ -9149,7 +9110,7 @@ TEST_F(WebRtcVideoChannelTest, SetMixedCodecSimulcastStreamConfig) { ScopedKeyValueConfig field_trials(field_trials_, "WebRTC-MixedCodecSimulcast/Enabled/"); - StreamParams sp = webrtc::CreateSimStreamParams("cname", {123, 456, 789}); + StreamParams sp = CreateSimStreamParams("cname", {123, 456, 789}); std::vector rid_descriptions; rid_descriptions.emplace_back("f", RidDirection::kSend); @@ -9196,22 +9157,22 @@ TEST_F(WebRtcVideoChannelTest, SetMixedCodecSimulcastStreamConfig) { #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) TEST_F(WebRtcVideoChannelTest, SetMixedCodecSimulcastWithDifferentConfigSettingsSizes) { - webrtc::test::ScopedKeyValueConfig field_trials( + test::ScopedKeyValueConfig field_trials( field_trials_, "WebRTC-MixedCodecSimulcast/Enabled/"); AddSendStream(); - webrtc::VideoSenderParameters parameters; - webrtc::Codec vp8 = GetEngineCodec("VP8"); + VideoSenderParameters parameters; + Codec vp8 = GetEngineCodec("VP8"); parameters.codecs.push_back(vp8); // `codec_settings_list.size()` is 1 after this in the EXPECT_TRUE(send_channel_->SetSenderParameters(parameters)); // It sets 2 sizes of config ssrc. - webrtc::StreamParams sp = CreateSimStreamParams("cname", {123, 456}); - std::vector rid_descriptions2; - rid_descriptions2.emplace_back("f", webrtc::RidDirection::kSend); - rid_descriptions2.emplace_back("h", webrtc::RidDirection::kSend); + StreamParams sp = CreateSimStreamParams("cname", {123, 456}); + std::vector rid_descriptions2; + rid_descriptions2.emplace_back("f", RidDirection::kSend); + rid_descriptions2.emplace_back("h", RidDirection::kSend); sp.set_rids(rid_descriptions2); // `WebRtcVideoSendStream::SetCodec` test for different sizes @@ -9267,7 +9228,7 @@ TEST_F(WebRtcVideoChannelTest, DefaultMinAndMaxBitratePropagatedToEncoder) { // FakeVideoSendStream calls CreateEncoderStreams, test that the vector of // VideoStreams are created appropriately. EXPECT_EQ(1u, stream->GetVideoStreams().size()); - EXPECT_EQ(webrtc::kDefaultMinVideoBitrateBps, + EXPECT_EQ(kDefaultMinVideoBitrateBps, stream->GetVideoStreams()[0].min_bitrate_bps); EXPECT_GT(stream->GetVideoStreams()[0].max_bitrate_bps, stream->GetVideoStreams()[0].min_bitrate_bps); @@ -9281,7 +9242,7 @@ TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersSetsMinBitrateForFirstActiveStream) { // Create the stream params with multiple ssrcs for simulcast. std::vector ssrcs = MAKE_VECTOR(kSsrcs3); - StreamParams stream_params = webrtc::CreateSimStreamParams("cname", ssrcs); + StreamParams stream_params = CreateSimStreamParams("cname", ssrcs); FakeVideoSendStream* fake_video_send_stream = AddSendStream(stream_params); uint32_t primary_ssrc = stream_params.first_ssrc(); @@ -9293,9 +9254,8 @@ TEST_F(WebRtcVideoChannelTest, EXPECT_TRUE( send_channel_->SetVideoSend(primary_ssrc, &options, &frame_forwarder)); send_channel_->SetSend(true); - frame_forwarder.IncomingCapturedFrame( - frame_source_.GetFrame(1920, 1080, VideoRotation::kVideoRotation_0, - webrtc::kNumMicrosecsPerSec / 30)); + frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame( + 1920, 1080, VideoRotation::kVideoRotation_0, kNumMicrosecsPerSec / 30)); // Check that all encodings are initially active. RtpParameters parameters = send_channel_->GetRtpSendParameters(primary_ssrc); @@ -9447,10 +9407,10 @@ TEST_F(WebRtcVideoChannelTest, GetRtpReceiveFmtpSprop) { TEST_F(WebRtcVideoChannelTest, DISABLED_GetRtpReceiveFmtpSprop) { #endif VideoReceiverParameters parameters; - Codec kH264sprop1 = webrtc::CreateVideoCodec(101, "H264"); + Codec kH264sprop1 = CreateVideoCodec(101, "H264"); kH264sprop1.SetParam(kH264FmtpSpropParameterSets, "uvw"); parameters.codecs.push_back(kH264sprop1); - Codec kH264sprop2 = webrtc::CreateVideoCodec(102, "H264"); + Codec kH264sprop2 = CreateVideoCodec(102, "H264"); kH264sprop2.SetParam(kH264FmtpSpropParameterSets, "xyz"); parameters.codecs.push_back(kH264sprop2); EXPECT_TRUE(receive_channel_->SetReceiverParameters(parameters)); @@ -9679,8 +9639,7 @@ TEST_F(WebRtcVideoChannelTest, GenerateKeyFrameSinglecast) { } TEST_F(WebRtcVideoChannelTest, GenerateKeyFrameSimulcast) { - StreamParams stream_params = - webrtc::CreateSimStreamParams("cname", {123, 456, 789}); + StreamParams stream_params = CreateSimStreamParams("cname", {123, 456, 789}); std::vector rids = {"f", "h", "q"}; std::vector rid_descriptions; @@ -9783,12 +9742,12 @@ class WebRtcVideoChannelSimulcastTest : public ::testing::Test { RTC_DCHECK(num_configured_streams <= ssrcs.size()); ssrcs.resize(num_configured_streams); - AddSendStream(webrtc::CreateSimStreamParams("cname", ssrcs)); + AddSendStream(CreateSimStreamParams("cname", ssrcs)); // Send a full-size frame to trigger a stream reconfiguration to use all // expected simulcast layers. FrameForwarder frame_forwarder; FakeFrameSource frame_source(capture_width, capture_height, - webrtc::kNumMicrosecsPerSec / 30); + kNumMicrosecsPerSec / 30); VideoOptions options; if (screenshare) @@ -9816,7 +9775,7 @@ class WebRtcVideoChannelSimulcastTest : public ::testing::Test { expected_streams = factory->CreateEncoderStreams( field_trials_, capture_width, capture_height, encoder_config); if (screenshare && conference_mode) { - for (const webrtc::VideoStream& expected_stream : expected_streams) { + for (const VideoStream& expected_stream : expected_streams) { // Never scale screen content. EXPECT_EQ(expected_stream.width, checked_cast(capture_width)); EXPECT_EQ(expected_stream.height, @@ -9828,7 +9787,7 @@ class WebRtcVideoChannelSimulcastTest : public ::testing::Test { expected_stream.width = capture_width; expected_stream.height = capture_height; expected_stream.max_framerate = kDefaultVideoMaxFramerate; - expected_stream.min_bitrate_bps = webrtc::kDefaultMinVideoBitrateBps; + expected_stream.min_bitrate_bps = kDefaultMinVideoBitrateBps; expected_stream.target_bitrate_bps = expected_stream.max_bitrate_bps = GetMaxDefaultBitrateBps(capture_width, capture_height); expected_stream.max_qp = kDefaultVideoMaxQpVpx; @@ -9916,34 +9875,31 @@ class WebRtcVideoChannelSimulcastTest : public ::testing::Test { }; TEST_F(WebRtcVideoChannelSimulcastTest, SetSendCodecsWith2SimulcastStreams) { - VerifySimulcastSettings(webrtc::CreateVideoCodec("VP8"), 640, 360, 2, 2, - false, true); + VerifySimulcastSettings(CreateVideoCodec("VP8"), 640, 360, 2, 2, false, true); } TEST_F(WebRtcVideoChannelSimulcastTest, SetSendCodecsWith3SimulcastStreams) { - VerifySimulcastSettings(webrtc::CreateVideoCodec("VP8"), 1280, 720, 3, 3, - false, true); + VerifySimulcastSettings(CreateVideoCodec("VP8"), 1280, 720, 3, 3, false, + true); } // Test that we normalize send codec format size in simulcast. TEST_F(WebRtcVideoChannelSimulcastTest, SetSendCodecsWithOddSizeInSimulcast) { - VerifySimulcastSettings(webrtc::CreateVideoCodec("VP8"), 541, 271, 2, 2, - false, true); + VerifySimulcastSettings(CreateVideoCodec("VP8"), 541, 271, 2, 2, false, true); } TEST_F(WebRtcVideoChannelSimulcastTest, SetSendCodecsForScreenshare) { - VerifySimulcastSettings(webrtc::CreateVideoCodec("VP8"), 1280, 720, 3, 3, - true, false); + VerifySimulcastSettings(CreateVideoCodec("VP8"), 1280, 720, 3, 3, true, + false); } TEST_F(WebRtcVideoChannelSimulcastTest, SetSendCodecsForSimulcastScreenshare) { - VerifySimulcastSettings(webrtc::CreateVideoCodec("VP8"), 1280, 720, 3, 2, - true, true); + VerifySimulcastSettings(CreateVideoCodec("VP8"), 1280, 720, 3, 2, true, true); } TEST_F(WebRtcVideoChannelSimulcastTest, SimulcastScreenshareWithoutConference) { - VerifySimulcastSettings(webrtc::CreateVideoCodec("VP8"), 1280, 720, 3, 3, - true, false); + VerifySimulcastSettings(CreateVideoCodec("VP8"), 1280, 720, 3, 3, true, + false); } TEST_F(WebRtcVideoChannelBaseTest, GetSources) { @@ -9987,7 +9943,7 @@ TEST_F(WebRtcVideoChannelBaseTest, GetSources) { } TEST_F(WebRtcVideoChannelTest, SetsRidsOnSendStream) { - StreamParams sp = webrtc::CreateSimStreamParams("cname", {123, 456, 789}); + StreamParams sp = CreateSimStreamParams("cname", {123, 456, 789}); std::vector rids = {"f", "h", "q"}; std::vector rid_descriptions; @@ -10040,7 +9996,7 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecast) { FakeVideoSendStream* stream = AddSendStream(); FrameForwarder frame_forwarder; - FakeFrameSource frame_source(1280, 720, webrtc::kNumMicrosecsPerSec / 30); + FakeFrameSource frame_source(1280, 720, kNumMicrosecsPerSec / 30); EXPECT_TRUE( send_channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder)); @@ -10098,7 +10054,7 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSinglecastScaling) { FakeVideoSendStream* stream = AddSendStream(); FrameForwarder frame_forwarder; - FakeFrameSource frame_source(1280, 720, webrtc::kNumMicrosecsPerSec / 30); + FakeFrameSource frame_source(1280, 720, kNumMicrosecsPerSec / 30); EXPECT_TRUE( send_channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder)); @@ -10174,7 +10130,7 @@ TEST_F(WebRtcVideoChannelTest, ScaleResolutionDownToSimulcast) { FakeVideoSendStream* stream = SetUpSimulcast(true, /*with_rtx=*/false); FrameForwarder frame_forwarder; - FakeFrameSource frame_source(1280, 720, webrtc::kNumMicrosecsPerSec / 30); + FakeFrameSource frame_source(1280, 720, kNumMicrosecsPerSec / 30); EXPECT_TRUE( send_channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder)); From ca02246fbf495a056fe4f9b2dddc2fd787942104 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Mon, 12 May 2025 03:11:13 -0700 Subject: [PATCH 117/292] Roll chromium_revision 4effae884f..fed22da4c6 (1458473:1458742) Change log: https://chromium.googlesource.com/chromium/src/+log/4effae884f..fed22da4c6 Full diff: https://chromium.googlesource.com/chromium/src/+/4effae884f..fed22da4c6 Changed dependencies * siso_version: git_revision:17b491ae74e86312a96239ff88e852acee83c135..git_revision:04b22e60e2aae70fa9d51141c91bf4554d8c5239 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/92d42d9ad2..b8a12c0da5 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/b128eca85f..c41670cc43 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/30c00d0f4d..f00a25ff94 * src/third_party/androidx/cipd: 1jSxEkUFGSfsQTosCWjXFmS0h3Lk_BRPBBLwBNosOVYC..wfVeehXfOqVwC9E4j9U8ghXvt7eeTMgZsdBUBB5fVRkC * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/61c101f683..2f8af1b7f2 * src/third_party/libunwind/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git/+log/4acde988c5..09f6f7b168 * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/6ffe623d28..11cd406514 * src/third_party/r8/cipd: UDiEQW04Q7U0lSlAQAik0xbIV-AdjAnCHl7GcecroU4C..QhYGRVpYYKZmt3f_Zb2HoJ9LIBEnWaeeLXRNei47Z30C * src/third_party/siso/cipd: git_revision:17b491ae74e86312a96239ff88e852acee83c135..git_revision:04b22e60e2aae70fa9d51141c91bf4554d8c5239 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/b9d3fbbda7..718569c346 DEPS diff: https://chromium.googlesource.com/chromium/src/+/4effae884f..fed22da4c6/DEPS No update to Clang. BUG=None Change-Id: I77c6c95ccd71a3f7e5233ac4699033dc025b2e09 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390900 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44589} --- DEPS | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/DEPS b/DEPS index a067a94538..9e76b0734c 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '4effae884f53c80ac1b2a41c2b3aa6b27b0a31a3', + 'chromium_revision': 'fed22da4c6801e618c8e0948bb276c87f643cdce', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -53,7 +53,7 @@ vars = { # reclient CIPD package version 'reclient_version': 're_client_version:0.177.1.e58c0145-gomaip', # siso CIPD package version. - 'siso_version': 'git_revision:17b491ae74e86312a96239ff88e852acee83c135', + 'siso_version': 'git_revision:04b22e60e2aae70fa9d51141c91bf4554d8c5239', # ninja CIPD package. 'ninja_package': 'infra/3pp/tools/ninja/', @@ -75,7 +75,7 @@ deps = { 'src/base': 'https://chromium.googlesource.com/chromium/src/base@994477767d99fcf1e3b34a25ae523c7eb7fd608a', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@92d42d9ad23dbee90f88c54bd7c53f347cb7040a', + 'https://chromium.googlesource.com/chromium/src/build@b8a12c0da5aabb81c840bad1fa5e8a4889a25f23', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@077a66f30fcf281b066fafb6dfc60818c238efb6', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@b128eca85f82e669877c6df05d834456d225f3f4', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@c41670cc43e5f3b0736263453f95fdd485ed85a3', 'condition': 'checkout_ios', }, 'src/testing': 'https://chromium.googlesource.com/chromium/src/testing@ab12b4b4c9bf4ea505ef5d0f4ed63dbf10cead00', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@30c00d0f4dae87cb4709312008d098a98de9ede5', + 'https://chromium.googlesource.com/chromium/src/third_party@f00a25ff945b6995bfb53d62ca7fd1787798ed7d', 'src/buildtools/linux64': { 'packages': [ @@ -331,7 +331,7 @@ deps = { 'src/third_party/llvm-libc/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@4a2940b40b394ca57312aa9bbc8af430fe9a5340', 'src/third_party/libunwind/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@4acde988c52e6a325e3ce2ac761284f295f2ab39', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@09f6f7b1685888b16c0fe52a7a68f9dff8b8d60e', 'src/third_party/test_fonts/test_fonts': { 'dep_type': 'gcs', @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@61c101f683f673a7917430bfee10f9e667b3a546', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@2f8af1b7f2ac17793e01bec1b2654ca12ed21966', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '6ffe623d283575c00047cf9abf90c3659505a592', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '11cd406514a8212d3568448b2207d2925d5ca202', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -632,7 +632,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/r8', - 'version': 'UDiEQW04Q7U0lSlAQAik0xbIV-AdjAnCHl7GcecroU4C', + 'version': 'QhYGRVpYYKZmt3f_Zb2HoJ9LIBEnWaeeLXRNei47Z30C', }, ], 'condition': 'checkout_android', @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@b9d3fbbda7b3606e4954991b008fa1e839f1ad90', + 'https://chromium.googlesource.com/chromium/src/tools@718569c3464aef1f18235ef679510fa0d12df4ac', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': '1jSxEkUFGSfsQTosCWjXFmS0h3Lk_BRPBBLwBNosOVYC', + 'version': 'wfVeehXfOqVwC9E4j9U8ghXvt7eeTMgZsdBUBB5fVRkC', }, ], 'condition': 'checkout_android and non_git_source', From 992005252ac3d060fb16d827c8e3471961ba9e4e Mon Sep 17 00:00:00 2001 From: Harald Alvestrand Date: Mon, 12 May 2025 09:31:25 +0000 Subject: [PATCH 118/292] Remove ability to have clockrate zero on a Codec. This was used as a wildcard, but apparently this was only relied on in unit tests. CreateAudioCodec() defaults to 8000 when 0 is passed; there were too many call sites to update. Bug: webrtc:416695360 Change-Id: Icde819563f0414d024422f1eb5c6b0846a0222ed Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390700 Reviewed-by: Evan Shrubsole Commit-Queue: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44590} --- call/payload_type_picker.cc | 2 +- media/base/codec.cc | 19 +++++++++++++++---- media/base/codec.h | 5 ++++- media/base/codec_comparators.cc | 16 ++++++---------- media/base/codec_comparators_unittest.cc | 19 +++++++++---------- media/base/codec_unittest.cc | 4 ++-- media/engine/webrtc_voice_engine_unittest.cc | 8 ++++---- pc/peer_connection_media_unittest.cc | 8 +++++--- pc/webrtc_sdp.cc | 2 +- 9 files changed, 47 insertions(+), 36 deletions(-) diff --git a/call/payload_type_picker.cc b/call/payload_type_picker.cc index c5bfa71269..3ce57cc502 100644 --- a/call/payload_type_picker.cc +++ b/call/payload_type_picker.cc @@ -169,7 +169,7 @@ PayloadTypePicker::PayloadTypePicker() { // Includes data to reduce collisions (and thus reassignments) // TODO(bugs.webrtc.org/400630582): Delete this, it's only for test // stability. - {{"reserved-do-not-use", 0, 0}, 102}, + {{"reserved-do-not-use", 1, 0}, 102}, {{kCnCodecName, 16000, 1}, 105}, {{kCnCodecName, 32000, 1}, 106}, {{kOpusCodecName, diff --git a/media/base/codec.cc b/media/base/codec.cc index b0b6a81cb9..de4184bfbb 100644 --- a/media/base/codec.cc +++ b/media/base/codec.cc @@ -110,9 +110,16 @@ Codec::Codec(Type type, name(name), clockrate(clockrate), bitrate(0), - channels(channels) {} + channels(channels) { + RTC_DCHECK_GT(clockrate, 0); +} -Codec::Codec(Type type) : Codec(type, kIdNotSet, "", 0) {} +Codec::Codec(Type type) + : Codec(type, + kIdNotSet, + "", + type == Type::kVideo ? kDefaultVideoClockRateHz + : kDefaultAudioClockRateHz) {} Codec::Codec(const webrtc::SdpAudioFormat& c) : Codec(Type::kAudio, kIdNotSet, c.name, c.clockrate_hz, c.num_channels) { @@ -285,7 +292,8 @@ std::string Codec::ToString() const { } Codec CreateAudioRtxCodec(int rtx_payload_type, int associated_payload_type) { - Codec rtx_codec = CreateAudioCodec(rtx_payload_type, kRtxCodecName, 0, 1); + Codec rtx_codec = CreateAudioCodec(rtx_payload_type, kRtxCodecName, + kDefaultAudioClockRateHz, 1); rtx_codec.SetParam(kCodecParamAssociatedPayloadType, associated_payload_type); return rtx_codec; } @@ -393,7 +401,10 @@ Codec CreateAudioCodec(int id, const std::string& name, int clockrate, size_t channels) { - return Codec(Codec::Type::kAudio, id, name, clockrate, channels); + return Codec(Codec::Type::kAudio, id, name, + // TODO: bugs.webrtc.org/416695360 - remove mapping when + // all call sites are updated to not use zero clock rate. + clockrate == 0 ? kDefaultAudioClockRateHz : clockrate, channels); } Codec CreateAudioCodec(const webrtc::SdpAudioFormat& c) { diff --git a/media/base/codec.h b/media/base/codec.h index 029b93c6ae..9566f37e73 100644 --- a/media/base/codec.h +++ b/media/base/codec.h @@ -16,7 +16,6 @@ #include #include -#include "absl/base/macros.h" #include "absl/container/inlined_vector.h" #include "absl/strings/str_format.h" #include "absl/strings/string_view.h" @@ -68,6 +67,8 @@ class FeedbackParams { std::vector params_; }; +constexpr int kDefaultVideoClockRateHz = 90'000; +constexpr int kDefaultAudioClockRateHz = 8'000; struct RTC_EXPORT Codec { enum class Type { @@ -172,6 +173,8 @@ struct RTC_EXPORT Codec { std::string ToString() const; + // Default constructor, for initialization. + Codec() : Codec(Type::kAudio, kIdNotSet, "", kDefaultAudioClockRateHz) {} Codec& operator=(const Codec& c); Codec& operator=(Codec&& c); diff --git a/media/base/codec_comparators.cc b/media/base/codec_comparators.cc index 4209d8cfa7..02ed96ac7c 100644 --- a/media/base/codec_comparators.cc +++ b/media/base/codec_comparators.cc @@ -313,22 +313,18 @@ bool MatchesWithCodecRules(const Codec& left_codec, const Codec& right_codec) { auto matches_type_specific = [&]() { switch (left_codec.type) { case Codec::Type::kAudio: - // If a nonzero clockrate is specified, it must match the actual - // clockrate. If a nonzero bitrate is specified, it must match the + // If a nonzero bitrate is specified, it must match the // actual bitrate, unless the codec is VBR (0), where we just force the // supplied value. The number of channels must match exactly, with the // exception that channels=0 is treated synonymously as channels=1, per // RFC 4566 section 6: " [The channels] parameter is OPTIONAL and may be // omitted if the number of channels is one." // Preference is ignored. - // TODO(juberti): Treat a zero clockrate as 8000Hz, the RTP default - // clockrate. - return ((right_codec.clockrate == 0 /*&& clockrate == 8000*/) || - left_codec.clockrate == right_codec.clockrate) && - (right_codec.bitrate == 0 || left_codec.bitrate <= 0 || - left_codec.bitrate == right_codec.bitrate) && - ((right_codec.channels < 2 && left_codec.channels < 2) || - left_codec.channels == right_codec.channels); + return ((left_codec.clockrate == right_codec.clockrate) && + (right_codec.bitrate == 0 || left_codec.bitrate <= 0 || + left_codec.bitrate == right_codec.bitrate) && + ((right_codec.channels < 2 && left_codec.channels < 2) || + left_codec.channels == right_codec.channels)); case Codec::Type::kVideo: return IsSameCodecSpecific(left_codec.name, left_codec.params, diff --git a/media/base/codec_comparators_unittest.cc b/media/base/codec_comparators_unittest.cc index b7dde12dec..67b630aebc 100644 --- a/media/base/codec_comparators_unittest.cc +++ b/media/base/codec_comparators_unittest.cc @@ -342,7 +342,6 @@ TEST(CodecTest, TestCodecMatches) { EXPECT_TRUE(c0.Matches(CreateAudioCodec(34, "", 44100, 1))); EXPECT_TRUE(c0.Matches(CreateAudioCodec(34, "", 44100, 0))); EXPECT_TRUE(c0.Matches(CreateAudioCodec(34, "", 44100, 0))); - EXPECT_TRUE(c0.Matches(CreateAudioCodec(34, "", 0, 0))); EXPECT_FALSE(c0.Matches(CreateAudioCodec(96, "A", 44100, 1))); EXPECT_FALSE(c0.Matches(CreateAudioCodec(96, "", 44100, 1))); EXPECT_FALSE(c0.Matches(CreateAudioCodec(95, "", 55100, 1))); @@ -352,15 +351,15 @@ TEST(CodecTest, TestCodecMatches) { // Test a codec with a dynamic payload type. Codec c1 = CreateAudioCodec(96, "A", 44100, 1); - EXPECT_TRUE(c1.Matches(CreateAudioCodec(96, "A", 0, 0))); - EXPECT_TRUE(c1.Matches(CreateAudioCodec(97, "A", 0, 0))); - EXPECT_TRUE(c1.Matches(CreateAudioCodec(96, "a", 0, 0))); - EXPECT_TRUE(c1.Matches(CreateAudioCodec(97, "a", 0, 0))); - EXPECT_TRUE(c1.Matches(CreateAudioCodec(35, "a", 0, 0))); - EXPECT_TRUE(c1.Matches(CreateAudioCodec(42, "a", 0, 0))); - EXPECT_TRUE(c1.Matches(CreateAudioCodec(65, "a", 0, 0))); - EXPECT_FALSE(c1.Matches(CreateAudioCodec(95, "A", 0, 0))); - EXPECT_FALSE(c1.Matches(CreateAudioCodec(34, "A", 0, 0))); + EXPECT_TRUE(c1.Matches(CreateAudioCodec(96, "A", 44100, 0))); + EXPECT_TRUE(c1.Matches(CreateAudioCodec(97, "A", 44100, 0))); + EXPECT_TRUE(c1.Matches(CreateAudioCodec(96, "a", 44100, 0))); + EXPECT_TRUE(c1.Matches(CreateAudioCodec(97, "a", 44100, 0))); + EXPECT_TRUE(c1.Matches(CreateAudioCodec(35, "a", 44100, 0))); + EXPECT_TRUE(c1.Matches(CreateAudioCodec(42, "a", 44100, 0))); + EXPECT_TRUE(c1.Matches(CreateAudioCodec(65, "a", 44100, 0))); + EXPECT_FALSE(c1.Matches(CreateAudioCodec(95, "A", 44100, 0))); + EXPECT_FALSE(c1.Matches(CreateAudioCodec(34, "A", 44100, 0))); EXPECT_FALSE(c1.Matches(CreateAudioCodec(96, "", 44100, 2))); EXPECT_FALSE(c1.Matches(CreateAudioCodec(96, "A", 55100, 1))); diff --git a/media/base/codec_unittest.cc b/media/base/codec_unittest.cc index 3a1bfef713..231198daa5 100644 --- a/media/base/codec_unittest.cc +++ b/media/base/codec_unittest.cc @@ -65,7 +65,7 @@ TEST(CodecTest, TestCodecOperators) { EXPECT_TRUE(c0 != c1); TestCodec c5; - TestCodec c6(Codec::kIdNotSet, "", 0); + TestCodec c6(Codec::kIdNotSet, "", kDefaultAudioClockRateHz); EXPECT_TRUE(c5 == c6); } @@ -83,7 +83,7 @@ TEST(CodecTest, TestAudioCodecOperators) { EXPECT_NE(c0, c4); EXPECT_NE(c0, c5); - Codec c8 = CreateAudioCodec(0, "", 0, 0); + Codec c8 = CreateAudioCodec(0, "", kDefaultAudioClockRateHz, 0); Codec c9 = c0; EXPECT_EQ(c9, c0); diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc index c4d1a1d02d..c6c44448e9 100644 --- a/media/engine/webrtc_voice_engine_unittest.cc +++ b/media/engine/webrtc_voice_engine_unittest.cc @@ -4083,8 +4083,8 @@ TEST(WebRtcVoiceEngineTest, CollectRecvCodecs) { // Rather than just ASSERTing that there are enough codecs, ensure that we // can check the actual values safely, to provide better test results. auto get_codec = [&codecs](size_t index) -> const webrtc::Codec& { - static const webrtc::Codec missing_codec = - webrtc::CreateAudioCodec(0, "", 0, 0); + static const webrtc::Codec missing_codec = webrtc::CreateAudioCodec( + 0, "", webrtc::kDefaultAudioClockRateHz, 0); if (codecs.size() > index) return codecs[index]; return missing_codec; @@ -4172,8 +4172,8 @@ TEST(WebRtcVoiceEngineTest, CollectRecvCodecsWithLatePtAssignment) { // Rather than just ASSERTing that there are enough codecs, ensure that we // can check the actual values safely, to provide better test results. auto get_codec = [&codecs](size_t index) -> const webrtc::Codec& { - static const webrtc::Codec missing_codec = - webrtc::CreateAudioCodec(0, "", 0, 0); + static const webrtc::Codec missing_codec = webrtc::CreateAudioCodec( + 0, "", webrtc::kDefaultAudioClockRateHz, 0); if (codecs.size() > index) return codecs[index]; return missing_codec; diff --git a/pc/peer_connection_media_unittest.cc b/pc/peer_connection_media_unittest.cc index fec693074d..e762cf6345 100644 --- a/pc/peer_connection_media_unittest.cc +++ b/pc/peer_connection_media_unittest.cc @@ -1468,8 +1468,9 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, SetCodecPreferencesAudioMissingRecvCodec) { auto fake_engine = std::make_unique(); auto send_codecs = fake_engine->voice().LegacySendCodecs(); - send_codecs.push_back( - CreateAudioCodec(send_codecs.back().id + 1, "send_only_codec", 0, 1)); + send_codecs.push_back(CreateAudioCodec(send_codecs.back().id + 1, + "send_only_codec", + kDefaultAudioClockRateHz, 1)); fake_engine->SetAudioSendCodecs(send_codecs); auto caller = CreatePeerConnectionWithAudio(std::move(fake_engine)); @@ -1902,7 +1903,8 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto fake_engine = std::make_unique(); std::vector audio_codecs; - audio_codecs.emplace_back(CreateAudioCodec(100, "foo", 0, 1)); + audio_codecs.emplace_back( + CreateAudioCodec(100, "foo", kDefaultAudioClockRateHz, 1)); audio_codecs.emplace_back(CreateAudioRtxCodec(101, 100)); fake_engine->SetAudioCodecs(audio_codecs); diff --git a/pc/webrtc_sdp.cc b/pc/webrtc_sdp.cc index eb6a07d0a8..8b179dece9 100644 --- a/pc/webrtc_sdp.cc +++ b/pc/webrtc_sdp.cc @@ -2841,7 +2841,7 @@ Codec GetCodecWithPayloadType(MediaType type, return *codec; // Return empty codec with `payload_type`. if (type == MediaType::AUDIO) { - return CreateAudioCodec(payload_type, "", 0, 0); + return CreateAudioCodec(payload_type, "", kDefaultAudioClockRateHz, 0); } else { return CreateVideoCodec(payload_type, ""); } From 5d2f62af597407635a78cce10435435240f3010f Mon Sep 17 00:00:00 2001 From: Emil Vardar Date: Sun, 27 Apr 2025 14:28:12 +0000 Subject: [PATCH 119/292] Add picture pair provider base class. MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The different picture pair providers should inherit this class to create encoded/decoded picture pairs. Bug: webrtc:358039777 Change-Id: Ib622db93d94a5c64c86de9da1c5b72c78f2c1c1e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/387861 Reviewed-by: Fanny Linderborg Reviewed-by: Erik Språng Commit-Queue: Emil Vardar (xWF) Cr-Commit-Position: refs/heads/main@{#44591} --- .../corruption_detection/evaluation/BUILD.gn | 11 ++++ .../evaluation/picture_pair_provider.h | 58 +++++++++++++++++++ 2 files changed, 69 insertions(+) create mode 100644 video/corruption_detection/evaluation/picture_pair_provider.h diff --git a/video/corruption_detection/evaluation/BUILD.gn b/video/corruption_detection/evaluation/BUILD.gn index 7b95a3b207..20ad2dcad4 100644 --- a/video/corruption_detection/evaluation/BUILD.gn +++ b/video/corruption_detection/evaluation/BUILD.gn @@ -8,6 +8,17 @@ import("../../../webrtc.gni") +rtc_library("picture_pair_provider") { + testonly = true + sources = [ "picture_pair_provider.h" ] + deps = [ + ":test_clip", + "../../../api/units:data_rate", + "../../../api/video:video_frame", + "../../../test:fileutils", + ] +} + rtc_library("test_clip") { testonly = true sources = [ diff --git a/video/corruption_detection/evaluation/picture_pair_provider.h b/video/corruption_detection/evaluation/picture_pair_provider.h new file mode 100644 index 0000000000..df04295f78 --- /dev/null +++ b/video/corruption_detection/evaluation/picture_pair_provider.h @@ -0,0 +1,58 @@ +/* + * Copyright 2025 The WebRTC project authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef VIDEO_CORRUPTION_DETECTION_EVALUATION_PICTURE_PAIR_PROVIDER_H_ +#define VIDEO_CORRUPTION_DETECTION_EVALUATION_PICTURE_PAIR_PROVIDER_H_ + +#include + +#include "api/units/data_rate.h" +#include "api/video/video_frame.h" +#include "test/testsupport/file_utils.h" +#include "video/corruption_detection/evaluation/test_clip.h" + +namespace webrtc { + +struct OriginalCompressedPicturePair { + const VideoFrame original_image; + // The corresponding compressed image, obtained through encoding and decoding + // with the QP value = `frame_average_qp`. + const VideoFrame compressed_image; + const int frame_average_qp = 0; +}; + +// Opens and reads one frame at a time from a raw video. Encodes and decodes +// this frame (to obtain a compressed frame) based on the provided bitrate. +// The original and compressed frame is returned with the `GetNextPicturePair()` +// method together with the corresponding average QP value for that frame. +class PicturePairProvider { + public: + PicturePairProvider() = default; + virtual ~PicturePairProvider() = default; + + // Configures the provider such that `GetNextPicturePair()` can provide an + // original and compressed frame. + // Inputs: + // `clip` indicates the test clip's path, codec_mode, resolution and + // framerate. + // `bitrate` the maximum bitrate allowed for encoding a raw video. + virtual bool Configure(const TestClip& clip, DataRate bitrate) = 0; + + // Encodes and decodes the next frame based on the parameters given in the + // `Configure()`. + // Returns: + // `OriginalCompressedPicturePair` structure, with the original and + // compressed frames and the mean QP of the frame in focus. + virtual std::optional GetNextPicturePair() = 0; +}; + +} // namespace webrtc + +#endif // VIDEO_CORRUPTION_DETECTION_EVALUATION_PICTURE_PAIR_PROVIDER_H_ From 3d7b1b16fbd8a576b0c35fccfef550e81be1a3ce Mon Sep 17 00:00:00 2001 From: Harald Alvestrand Date: Mon, 12 May 2025 10:42:02 +0000 Subject: [PATCH 120/292] Cleanup CreateAudioCodec to reject zero by changing all usages of zero to kDefaultAudioClockrateHz Bug: webrtc:416695360 Change-Id: I5e909287c76c62f416e37e716897422985e034df Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390608 Reviewed-by: Evan Shrubsole Commit-Queue: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44592} --- media/base/codec.cc | 5 +- pc/peer_connection_media_unittest.cc | 93 ++++++++++++++++++---------- 2 files changed, 62 insertions(+), 36 deletions(-) diff --git a/media/base/codec.cc b/media/base/codec.cc index de4184bfbb..f97f082fd9 100644 --- a/media/base/codec.cc +++ b/media/base/codec.cc @@ -401,10 +401,7 @@ Codec CreateAudioCodec(int id, const std::string& name, int clockrate, size_t channels) { - return Codec(Codec::Type::kAudio, id, name, - // TODO: bugs.webrtc.org/416695360 - remove mapping when - // all call sites are updated to not use zero clock rate. - clockrate == 0 ? kDefaultAudioClockRateHz : clockrate, channels); + return Codec(Codec::Type::kAudio, id, name, clockrate, channels); } Codec CreateAudioCodec(const webrtc::SdpAudioFormat& c) { diff --git a/pc/peer_connection_media_unittest.cc b/pc/peer_connection_media_unittest.cc index e762cf6345..b47febc085 100644 --- a/pc/peer_connection_media_unittest.cc +++ b/pc/peer_connection_media_unittest.cc @@ -1203,14 +1203,17 @@ TEST_P(PeerConnectionMediaTest, SetRemoteDescriptionFailsWithDuplicateMids) { // fmtp line is modified to refer to the correct payload type. TEST_P(PeerConnectionMediaTest, RedFmtpPayloadTypeReassigned) { std::vector caller_fake_codecs; - caller_fake_codecs.push_back(CreateAudioCodec(100, "foo", 0, 1)); + caller_fake_codecs.push_back( + CreateAudioCodec(100, "foo", kDefaultAudioClockRateHz, 1)); auto caller_fake_engine = std::make_unique(); caller_fake_engine->SetAudioCodecs(caller_fake_codecs); auto caller = CreatePeerConnectionWithAudio(std::move(caller_fake_engine)); std::vector callee_fake_codecs; - callee_fake_codecs.push_back(CreateAudioCodec(120, "foo", 0, 1)); - callee_fake_codecs.push_back(CreateAudioCodec(121, kRedCodecName, 0, 1)); + callee_fake_codecs.push_back( + CreateAudioCodec(120, "foo", kDefaultAudioClockRateHz, 1)); + callee_fake_codecs.push_back( + CreateAudioCodec(121, kRedCodecName, kDefaultAudioClockRateHz, 1)); callee_fake_codecs.back().SetParam(kCodecParamNotInNameValueFormat, "120/120"); auto callee_fake_engine = std::make_unique(); @@ -1245,15 +1248,19 @@ TEST_P(PeerConnectionMediaTest, RedFmtpPayloadTypeReassigned) { // Test that RED without fmtp does match RED without fmtp. TEST_P(PeerConnectionMediaTest, RedFmtpPayloadTypeNoFmtpMatchNoFmtp) { std::vector caller_fake_codecs; - caller_fake_codecs.push_back(CreateAudioCodec(100, "foo", 0, 1)); - caller_fake_codecs.push_back(CreateAudioCodec(101, kRedCodecName, 0, 1)); + caller_fake_codecs.push_back( + CreateAudioCodec(100, "foo", kDefaultAudioClockRateHz, 1)); + caller_fake_codecs.push_back( + CreateAudioCodec(101, kRedCodecName, kDefaultAudioClockRateHz, 1)); auto caller_fake_engine = std::make_unique(); caller_fake_engine->SetAudioCodecs(caller_fake_codecs); auto caller = CreatePeerConnectionWithAudio(std::move(caller_fake_engine)); std::vector callee_fake_codecs; - callee_fake_codecs.push_back(CreateAudioCodec(120, "foo", 0, 1)); - callee_fake_codecs.push_back(CreateAudioCodec(121, kRedCodecName, 0, 1)); + callee_fake_codecs.push_back( + CreateAudioCodec(120, "foo", kDefaultAudioClockRateHz, 1)); + callee_fake_codecs.push_back( + CreateAudioCodec(121, kRedCodecName, kDefaultAudioClockRateHz, 1)); auto callee_fake_engine = std::make_unique(); callee_fake_engine->SetAudioCodecs(callee_fake_codecs); auto callee = CreatePeerConnectionWithAudio(std::move(callee_fake_engine)); @@ -1285,15 +1292,19 @@ TEST_P(PeerConnectionMediaTest, RedFmtpPayloadTypeNoFmtpMatchNoFmtp) { // Test that RED without fmtp does not match RED with fmtp. TEST_P(PeerConnectionMediaTest, RedFmtpPayloadTypeNoFmtpNoMatchFmtp) { std::vector caller_fake_codecs; - caller_fake_codecs.push_back(CreateAudioCodec(100, "foo", 0, 1)); - caller_fake_codecs.push_back(CreateAudioCodec(101, kRedCodecName, 0, 1)); + caller_fake_codecs.push_back( + CreateAudioCodec(100, "foo", kDefaultAudioClockRateHz, 1)); + caller_fake_codecs.push_back( + CreateAudioCodec(101, kRedCodecName, kDefaultAudioClockRateHz, 1)); auto caller_fake_engine = std::make_unique(); caller_fake_engine->SetAudioCodecs(caller_fake_codecs); auto caller = CreatePeerConnectionWithAudio(std::move(caller_fake_engine)); std::vector callee_fake_codecs; - callee_fake_codecs.push_back(CreateAudioCodec(120, "foo", 0, 1)); - callee_fake_codecs.push_back(CreateAudioCodec(121, kRedCodecName, 0, 1)); + callee_fake_codecs.push_back( + CreateAudioCodec(120, "foo", kDefaultAudioClockRateHz, 1)); + callee_fake_codecs.push_back( + CreateAudioCodec(121, kRedCodecName, kDefaultAudioClockRateHz, 1)); callee_fake_codecs.back().SetParam(kCodecParamNotInNameValueFormat, "120/120"); auto callee_fake_engine = std::make_unique(); @@ -1329,8 +1340,10 @@ TEST_P(PeerConnectionMediaTest, RedFmtpPayloadTypeNoFmtpNoMatchFmtp) { // Test that RED with fmtp must match base codecs. TEST_P(PeerConnectionMediaTest, RedFmtpPayloadTypeMustMatchBaseCodecs) { std::vector caller_fake_codecs; - caller_fake_codecs.push_back(CreateAudioCodec(100, "foo", 0, 1)); - caller_fake_codecs.push_back(CreateAudioCodec(101, kRedCodecName, 0, 1)); + caller_fake_codecs.push_back( + CreateAudioCodec(100, "foo", kDefaultAudioClockRateHz, 1)); + caller_fake_codecs.push_back( + CreateAudioCodec(101, kRedCodecName, kDefaultAudioClockRateHz, 1)); caller_fake_codecs.back().SetParam(kCodecParamNotInNameValueFormat, "100/100"); auto caller_fake_engine = std::make_unique(); @@ -1338,9 +1351,12 @@ TEST_P(PeerConnectionMediaTest, RedFmtpPayloadTypeMustMatchBaseCodecs) { auto caller = CreatePeerConnectionWithAudio(std::move(caller_fake_engine)); std::vector callee_fake_codecs; - callee_fake_codecs.push_back(CreateAudioCodec(120, "foo", 0, 1)); - callee_fake_codecs.push_back(CreateAudioCodec(121, kRedCodecName, 0, 1)); - callee_fake_codecs.push_back(CreateAudioCodec(122, "bar", 0, 1)); + callee_fake_codecs.push_back( + CreateAudioCodec(120, "foo", kDefaultAudioClockRateHz, 1)); + callee_fake_codecs.push_back( + CreateAudioCodec(121, kRedCodecName, kDefaultAudioClockRateHz, 1)); + callee_fake_codecs.push_back( + CreateAudioCodec(122, "bar", kDefaultAudioClockRateHz, 1)); callee_fake_codecs.back().SetParam(kCodecParamNotInNameValueFormat, "122/122"); auto callee_fake_engine = std::make_unique(); @@ -1361,9 +1377,12 @@ TEST_P(PeerConnectionMediaTest, RedFmtpPayloadTypeMustMatchBaseCodecs) { // which is not supported. TEST_P(PeerConnectionMediaTest, RedFmtpPayloadMixed) { std::vector caller_fake_codecs; - caller_fake_codecs.push_back(CreateAudioCodec(100, "foo", 0, 1)); - caller_fake_codecs.push_back(CreateAudioCodec(102, "bar", 0, 1)); - caller_fake_codecs.push_back(CreateAudioCodec(101, kRedCodecName, 0, 1)); + caller_fake_codecs.push_back( + CreateAudioCodec(100, "foo", kDefaultAudioClockRateHz, 1)); + caller_fake_codecs.push_back( + CreateAudioCodec(102, "bar", kDefaultAudioClockRateHz, 1)); + caller_fake_codecs.push_back( + CreateAudioCodec(101, kRedCodecName, kDefaultAudioClockRateHz, 1)); caller_fake_codecs.back().SetParam(kCodecParamNotInNameValueFormat, "100/102"); auto caller_fake_engine = std::make_unique(); @@ -1371,8 +1390,10 @@ TEST_P(PeerConnectionMediaTest, RedFmtpPayloadMixed) { auto caller = CreatePeerConnectionWithAudio(std::move(caller_fake_engine)); std::vector callee_fake_codecs; - callee_fake_codecs.push_back(CreateAudioCodec(120, "foo", 0, 1)); - callee_fake_codecs.push_back(CreateAudioCodec(121, kRedCodecName, 0, 1)); + callee_fake_codecs.push_back( + CreateAudioCodec(120, "foo", kDefaultAudioClockRateHz, 1)); + callee_fake_codecs.push_back( + CreateAudioCodec(121, kRedCodecName, kDefaultAudioClockRateHz, 1)); callee_fake_codecs.back().SetParam(kCodecParamNotInNameValueFormat, "120/120"); auto callee_fake_engine = std::make_unique(); @@ -1393,8 +1414,10 @@ TEST_P(PeerConnectionMediaTest, RedFmtpPayloadMixed) { // redundancy. TEST_P(PeerConnectionMediaTest, RedFmtpPayloadDifferentRedundancy) { std::vector caller_fake_codecs; - caller_fake_codecs.push_back(CreateAudioCodec(100, "foo", 0, 1)); - caller_fake_codecs.push_back(CreateAudioCodec(101, kRedCodecName, 0, 1)); + caller_fake_codecs.push_back( + CreateAudioCodec(100, "foo", kDefaultAudioClockRateHz, 1)); + caller_fake_codecs.push_back( + CreateAudioCodec(101, kRedCodecName, kDefaultAudioClockRateHz, 1)); caller_fake_codecs.back().SetParam(kCodecParamNotInNameValueFormat, "100/100"); auto caller_fake_engine = std::make_unique(); @@ -1402,8 +1425,10 @@ TEST_P(PeerConnectionMediaTest, RedFmtpPayloadDifferentRedundancy) { auto caller = CreatePeerConnectionWithAudio(std::move(caller_fake_engine)); std::vector callee_fake_codecs; - callee_fake_codecs.push_back(CreateAudioCodec(120, "foo", 0, 1)); - callee_fake_codecs.push_back(CreateAudioCodec(121, kRedCodecName, 0, 1)); + callee_fake_codecs.push_back( + CreateAudioCodec(120, "foo", kDefaultAudioClockRateHz, 1)); + callee_fake_codecs.push_back( + CreateAudioCodec(121, kRedCodecName, kDefaultAudioClockRateHz, 1)); callee_fake_codecs.back().SetParam(kCodecParamNotInNameValueFormat, "120/120/120"); auto callee_fake_engine = std::make_unique(); @@ -1510,10 +1535,11 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto audio_codecs = fake_engine->voice().LegacySendCodecs(); audio_codecs.push_back( CreateAudioRtxCodec(audio_codecs.back().id + 1, audio_codecs.back().id)); - audio_codecs.push_back( - CreateAudioCodec(audio_codecs.back().id + 1, kRedCodecName, 0, 1)); - audio_codecs.push_back( - CreateAudioCodec(audio_codecs.back().id + 1, kUlpfecCodecName, 0, 1)); + audio_codecs.push_back(CreateAudioCodec( + audio_codecs.back().id + 1, kRedCodecName, kDefaultAudioClockRateHz, 1)); + audio_codecs.push_back(CreateAudioCodec(audio_codecs.back().id + 1, + kUlpfecCodecName, + kDefaultAudioClockRateHz, 1)); fake_engine->SetAudioCodecs(audio_codecs); auto caller = CreatePeerConnectionWithAudio(std::move(fake_engine)); @@ -1862,7 +1888,8 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto fake_engine = std::make_unique(); std::vector audio_codecs; - audio_codecs.emplace_back(CreateAudioCodec(100, "foo", 0, 1)); + audio_codecs.emplace_back( + CreateAudioCodec(100, "foo", kDefaultAudioClockRateHz, 1)); audio_codecs.emplace_back(CreateAudioRtxCodec(101, 100)); fake_engine->SetAudioCodecs(audio_codecs); @@ -1949,7 +1976,8 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto fake_engine = std::make_unique(); std::vector audio_codecs; - audio_codecs.emplace_back(CreateAudioCodec(100, "foo", 0, 1)); + audio_codecs.emplace_back( + CreateAudioCodec(100, "foo", kDefaultAudioClockRateHz, 1)); audio_codecs.emplace_back(CreateAudioRtxCodec(101, 100)); fake_engine->SetAudioCodecs(audio_codecs); @@ -1995,7 +2023,8 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, auto fake_engine = std::make_unique(); std::vector audio_codecs; - audio_codecs.emplace_back(CreateAudioCodec(100, "foo", 0, 1)); + audio_codecs.emplace_back( + CreateAudioCodec(100, "foo", kDefaultAudioClockRateHz, 1)); fake_engine->SetAudioRecvCodecs(audio_codecs); auto caller = CreatePeerConnectionWithAudio(std::move(fake_engine)); From 44825581b5ec2fab212335e3b8408b21166bbdbb Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Fri, 9 May 2025 10:30:29 +0000 Subject: [PATCH 121/292] [cleanup] Fix redundant webrtc name specifier This CL was uploaded by git cl split. R=hta@webrtc.org No-Iwyu: LSC Bug: webrtc:42232595 Change-Id: I36c2e42a06de0859fa05c531bbcd5bdaa18b6013 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390580 Reviewed-by: Harald Alvestrand Auto-Submit: Evan Shrubsole Commit-Queue: Evan Shrubsole Cr-Commit-Position: refs/heads/main@{#44593} --- api/test/fake_frame_decryptor.cc | 4 +- api/test/fake_frame_encryptor.cc | 7 ++-- ...me_perf_dashboard_metrics_exporter_test.cc | 9 ++--- .../metrics_set_proto_file_exporter.cc | 39 +++++++++---------- .../metrics_set_proto_file_exporter_test.cc | 16 ++++---- .../schedulable_network_node_builder.cc | 8 ++-- api/test/pclf/peer_configurer.cc | 8 ++-- 7 files changed, 42 insertions(+), 49 deletions(-) diff --git a/api/test/fake_frame_decryptor.cc b/api/test/fake_frame_decryptor.cc index ad635467f3..0e1da2c3f2 100644 --- a/api/test/fake_frame_decryptor.cc +++ b/api/test/fake_frame_decryptor.cc @@ -25,7 +25,7 @@ FakeFrameDecryptor::FakeFrameDecryptor(uint8_t fake_key, : fake_key_(fake_key), expected_postfix_byte_(expected_postfix_byte) {} FakeFrameDecryptor::Result FakeFrameDecryptor::Decrypt( - webrtc::MediaType /* media_type */, + MediaType /* media_type */, const std::vector& /* csrcs */, ArrayView /* additional_data */, ArrayView encrypted_frame, @@ -47,7 +47,7 @@ FakeFrameDecryptor::Result FakeFrameDecryptor::Decrypt( } size_t FakeFrameDecryptor::GetMaxPlaintextByteSize( - webrtc::MediaType /* media_type */, + MediaType /* media_type */, size_t encrypted_frame_size) { return encrypted_frame_size - 1; } diff --git a/api/test/fake_frame_encryptor.cc b/api/test/fake_frame_encryptor.cc index b988245b99..d35756e065 100644 --- a/api/test/fake_frame_encryptor.cc +++ b/api/test/fake_frame_encryptor.cc @@ -22,7 +22,7 @@ FakeFrameEncryptor::FakeFrameEncryptor(uint8_t fake_key, uint8_t postfix_byte) : fake_key_(fake_key), postfix_byte_(postfix_byte) {} // FrameEncryptorInterface implementation -int FakeFrameEncryptor::Encrypt(webrtc::MediaType /* media_type */, +int FakeFrameEncryptor::Encrypt(MediaType /* media_type */, uint32_t /* ssrc */, ArrayView /* additional_data */, ArrayView frame, @@ -42,9 +42,8 @@ int FakeFrameEncryptor::Encrypt(webrtc::MediaType /* media_type */, return static_cast(FakeEncryptionStatus::OK); } -size_t FakeFrameEncryptor::GetMaxCiphertextByteSize( - webrtc::MediaType /* media_type */, - size_t frame_size) { +size_t FakeFrameEncryptor::GetMaxCiphertextByteSize(MediaType /* media_type */, + size_t frame_size) { return frame_size + 1; } diff --git a/api/test/metrics/chrome_perf_dashboard_metrics_exporter_test.cc b/api/test/metrics/chrome_perf_dashboard_metrics_exporter_test.cc index ae1ab84b30..7177ed6c06 100644 --- a/api/test/metrics/chrome_perf_dashboard_metrics_exporter_test.cc +++ b/api/test/metrics/chrome_perf_dashboard_metrics_exporter_test.cc @@ -55,14 +55,11 @@ class ChromePerfDashboardMetricsExporterTest : public Test { ~ChromePerfDashboardMetricsExporterTest() override = default; void SetUp() override { - temp_filename_ = webrtc::test::TempFilename( - webrtc::test::OutputPath(), - "chrome_perf_dashboard_metrics_exporter_test"); + temp_filename_ = test::TempFilename( + test::OutputPath(), "chrome_perf_dashboard_metrics_exporter_test"); } - void TearDown() override { - ASSERT_TRUE(webrtc::test::RemoveFile(temp_filename_)); - } + void TearDown() override { ASSERT_TRUE(test::RemoveFile(temp_filename_)); } std::string temp_filename_; }; diff --git a/api/test/metrics/metrics_set_proto_file_exporter.cc b/api/test/metrics/metrics_set_proto_file_exporter.cc index 49364846e8..a26ae7ac35 100644 --- a/api/test/metrics/metrics_set_proto_file_exporter.cc +++ b/api/test/metrics/metrics_set_proto_file_exporter.cc @@ -30,42 +30,41 @@ namespace test { namespace { #if WEBRTC_ENABLE_PROTOBUF -webrtc::test_metrics::Unit ToProtoUnit(Unit unit) { +test_metrics::Unit ToProtoUnit(Unit unit) { switch (unit) { case Unit::kMilliseconds: - return webrtc::test_metrics::Unit::MILLISECONDS; + return test_metrics::Unit::MILLISECONDS; case Unit::kPercent: - return webrtc::test_metrics::Unit::PERCENT; + return test_metrics::Unit::PERCENT; case Unit::kBytes: - return webrtc::test_metrics::Unit::BYTES; + return test_metrics::Unit::BYTES; case Unit::kKilobitsPerSecond: - return webrtc::test_metrics::Unit::KILOBITS_PER_SECOND; + return test_metrics::Unit::KILOBITS_PER_SECOND; case Unit::kHertz: - return webrtc::test_metrics::Unit::HERTZ; + return test_metrics::Unit::HERTZ; case Unit::kUnitless: - return webrtc::test_metrics::Unit::UNITLESS; + return test_metrics::Unit::UNITLESS; case Unit::kCount: - return webrtc::test_metrics::Unit::COUNT; + return test_metrics::Unit::COUNT; } } -webrtc::test_metrics::ImprovementDirection ToProtoImprovementDirection( +test_metrics::ImprovementDirection ToProtoImprovementDirection( ImprovementDirection direction) { switch (direction) { case ImprovementDirection::kBiggerIsBetter: - return webrtc::test_metrics::ImprovementDirection::BIGGER_IS_BETTER; + return test_metrics::ImprovementDirection::BIGGER_IS_BETTER; case ImprovementDirection::kNeitherIsBetter: - return webrtc::test_metrics::ImprovementDirection::NEITHER_IS_BETTER; + return test_metrics::ImprovementDirection::NEITHER_IS_BETTER; case ImprovementDirection::kSmallerIsBetter: - return webrtc::test_metrics::ImprovementDirection::SMALLER_IS_BETTER; + return test_metrics::ImprovementDirection::SMALLER_IS_BETTER; } } -void SetTimeSeries( - const Metric::TimeSeries& time_series, - webrtc::test_metrics::Metric::TimeSeries* proto_time_series) { +void SetTimeSeries(const Metric::TimeSeries& time_series, + test_metrics::Metric::TimeSeries* proto_time_series) { for (const Metric::TimeSeries::Sample& sample : time_series.samples) { - webrtc::test_metrics::Metric::TimeSeries::Sample* proto_sample = + test_metrics::Metric::TimeSeries::Sample* proto_sample = proto_time_series->add_samples(); proto_sample->set_value(sample.value); proto_sample->set_timestamp_us(sample.timestamp.us()); @@ -76,7 +75,7 @@ void SetTimeSeries( } void SetStats(const Metric::Stats& stats, - webrtc::test_metrics::Metric::Stats* proto_stats) { + test_metrics::Metric::Stats* proto_stats) { if (stats.mean.has_value()) { proto_stats->set_mean(*stats.mean); } @@ -92,7 +91,7 @@ void SetStats(const Metric::Stats& stats, } bool WriteMetricsToFile(const std::string& path, - const webrtc::test_metrics::MetricsSet& metrics_set) { + const test_metrics::MetricsSet& metrics_set) { std::string data; bool ok = metrics_set.SerializeToString(&data); if (!ok) { @@ -135,12 +134,12 @@ MetricsSetProtoFileExporter::Options::Options( bool MetricsSetProtoFileExporter::Export(ArrayView metrics) { #if WEBRTC_ENABLE_PROTOBUF - webrtc::test_metrics::MetricsSet metrics_set; + test_metrics::MetricsSet metrics_set; for (const auto& [key, value] : options_.metadata) { metrics_set.mutable_metadata()->insert({key, value}); } for (const Metric& metric : metrics) { - webrtc::test_metrics::Metric* metric_proto = metrics_set.add_metrics(); + test_metrics::Metric* metric_proto = metrics_set.add_metrics(); metric_proto->set_name(metric.name); metric_proto->set_unit(ToProtoUnit(metric.unit)); metric_proto->set_improvement_direction( diff --git a/api/test/metrics/metrics_set_proto_file_exporter_test.cc b/api/test/metrics/metrics_set_proto_file_exporter_test.cc index 7eee6c3d2b..cc9ecde8d2 100644 --- a/api/test/metrics/metrics_set_proto_file_exporter_test.cc +++ b/api/test/metrics/metrics_set_proto_file_exporter_test.cc @@ -31,7 +31,7 @@ namespace { using ::testing::Eq; using ::testing::Test; -namespace proto = ::webrtc::test_metrics; +namespace proto = test_metrics; std::string ReadFileAsString(const std::string& filename) { std::ifstream infile(filename, std::ios_base::binary); @@ -66,13 +66,11 @@ class MetricsSetProtoFileExporterTest : public Test { ~MetricsSetProtoFileExporterTest() override = default; void SetUp() override { - temp_filename_ = webrtc::test::TempFilename( - webrtc::test::OutputPath(), "metrics_set_proto_file_exporter_test"); + temp_filename_ = test::TempFilename(test::OutputPath(), + "metrics_set_proto_file_exporter_test"); } - void TearDown() override { - ASSERT_TRUE(webrtc::test::RemoveFile(temp_filename_)); - } + void TearDown() override { ASSERT_TRUE(test::RemoveFile(temp_filename_)); } std::string temp_filename_; }; @@ -103,7 +101,7 @@ TEST_F(MetricsSetProtoFileExporterTest, MetricsAreExportedCorrectly) { .mean = 30.0, .stddev = 10.0, .min = 20.0, .max = 40.0}}; ASSERT_TRUE(exporter.Export(std::vector{metric1, metric2})); - webrtc::test_metrics::MetricsSet actual_metrics_set; + test_metrics::MetricsSet actual_metrics_set; actual_metrics_set.ParseFromString(ReadFileAsString(temp_filename_)); EXPECT_THAT(actual_metrics_set.metrics().size(), Eq(2)); @@ -152,7 +150,7 @@ TEST_F(MetricsSetProtoFileExporterTest, NoMetricsSetMetadata) { MetricsSetProtoFileExporter::Options options(temp_filename_); MetricsSetProtoFileExporter exporter(options); ASSERT_TRUE(exporter.Export(std::vector{})); - webrtc::test_metrics::MetricsSet actual_metrics_set; + test_metrics::MetricsSet actual_metrics_set; actual_metrics_set.ParseFromString(ReadFileAsString(temp_filename_)); EXPECT_EQ(actual_metrics_set.metadata_size(), 0); } @@ -162,7 +160,7 @@ TEST_F(MetricsSetProtoFileExporterTest, MetricsSetMetadata) { temp_filename_, {{"a_metadata_key", "a_metadata_value"}}); MetricsSetProtoFileExporter exporter(options); ASSERT_TRUE(exporter.Export(std::vector{})); - webrtc::test_metrics::MetricsSet actual_metrics_set; + test_metrics::MetricsSet actual_metrics_set; actual_metrics_set.ParseFromString(ReadFileAsString(temp_filename_)); EXPECT_EQ(actual_metrics_set.metadata_size(), 1); EXPECT_EQ(actual_metrics_set.metadata().at("a_metadata_key"), diff --git a/api/test/network_emulation/schedulable_network_node_builder.cc b/api/test/network_emulation/schedulable_network_node_builder.cc index 2b2bf39d86..f6117d9189 100644 --- a/api/test/network_emulation/schedulable_network_node_builder.cc +++ b/api/test/network_emulation/schedulable_network_node_builder.cc @@ -24,18 +24,18 @@ namespace webrtc { SchedulableNetworkNodeBuilder::SchedulableNetworkNodeBuilder( - webrtc::NetworkEmulationManager& net, + NetworkEmulationManager& net, network_behaviour::NetworkConfigSchedule schedule) : net_(net), schedule_(std::move(schedule)), - start_condition_([](webrtc::Timestamp) { return true; }) {} + start_condition_([](Timestamp) { return true; }) {} void SchedulableNetworkNodeBuilder::set_start_condition( - absl::AnyInvocable start_condition) { + absl::AnyInvocable start_condition) { start_condition_ = std::move(start_condition); } -webrtc::EmulatedNetworkNode* SchedulableNetworkNodeBuilder::Build( +EmulatedNetworkNode* SchedulableNetworkNodeBuilder::Build( std::optional random_seed) { uint64_t seed = random_seed.has_value() ? *random_seed : static_cast(TimeNanos()); diff --git a/api/test/pclf/peer_configurer.cc b/api/test/pclf/peer_configurer.cc index 7da73512d5..9077be94bf 100644 --- a/api/test/pclf/peer_configurer.cc +++ b/api/test/pclf/peer_configurer.cc @@ -92,17 +92,17 @@ PeerConfigurer* PeerConfigurer::SetVideoDecoderFactory( return this; } PeerConfigurer* PeerConfigurer::SetAudioEncoderFactory( - scoped_refptr audio_encoder_factory) { + scoped_refptr audio_encoder_factory) { components_->pcf_dependencies->audio_encoder_factory = audio_encoder_factory; return this; } PeerConfigurer* PeerConfigurer::SetAudioDecoderFactory( - scoped_refptr audio_decoder_factory) { + scoped_refptr audio_decoder_factory) { components_->pcf_dependencies->audio_decoder_factory = audio_decoder_factory; return this; } PeerConfigurer* PeerConfigurer::SetAsyncDnsResolverFactory( - std::unique_ptr + std::unique_ptr async_dns_resolver_factory) { components_->pc_dependencies->async_dns_resolver_factory = std::move(async_dns_resolver_factory); @@ -187,7 +187,7 @@ PeerConfigurer* PeerConfigurer::SetAudioProcessing( return this; } PeerConfigurer* PeerConfigurer::SetAudioMixer( - scoped_refptr audio_mixer) { + scoped_refptr audio_mixer) { components_->pcf_dependencies->audio_mixer = audio_mixer; return this; } From 8114be79ebf04c537700c929c491ca41b7d8ebf8 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Per=20=C3=85hgren?= Date: Thu, 24 Apr 2025 20:43:44 +0000 Subject: [PATCH 122/292] Re-add removed call-by-reference in the EchoCanceller3 factory Bug: None Change-Id: I4416b77c8f5a44aa741f6bc9d0bcd39f8b233e5c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/388440 Commit-Queue: Sam Zackrisson Reviewed-by: Sam Zackrisson Cr-Commit-Position: refs/heads/main@{#44594} --- api/audio/echo_canceller3_factory.cc | 4 ++-- api/audio/echo_canceller3_factory.h | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/api/audio/echo_canceller3_factory.cc b/api/audio/echo_canceller3_factory.cc index 6e3a54a783..8a87f0a648 100644 --- a/api/audio/echo_canceller3_factory.cc +++ b/api/audio/echo_canceller3_factory.cc @@ -22,11 +22,11 @@ namespace webrtc { EchoCanceller3Factory::EchoCanceller3Factory() {} -EchoCanceller3Factory::EchoCanceller3Factory(const EchoCanceller3Config config) +EchoCanceller3Factory::EchoCanceller3Factory(const EchoCanceller3Config& config) : config_(config), multichannel_config_(std::nullopt) {} EchoCanceller3Factory::EchoCanceller3Factory( - const EchoCanceller3Config config, + const EchoCanceller3Config& config, std::optional multichannel_config) : config_(config), multichannel_config_(multichannel_config) {} diff --git a/api/audio/echo_canceller3_factory.h b/api/audio/echo_canceller3_factory.h index e26f298930..756fd4682e 100644 --- a/api/audio/echo_canceller3_factory.h +++ b/api/audio/echo_canceller3_factory.h @@ -29,12 +29,12 @@ class RTC_EXPORT EchoCanceller3Factory : public EchoControlFactory { // Factory producing EchoCanceller3 instances with the specified // configuration. - explicit EchoCanceller3Factory(const EchoCanceller3Config config); + explicit EchoCanceller3Factory(const EchoCanceller3Config& config); // Factory producing EchoCanceller3 instances with the specified // configuration and multichannel configuration. EchoCanceller3Factory( - const EchoCanceller3Config config, + const EchoCanceller3Config& config, std::optional multichannel_config); // Creates an EchoCanceller3 with a specified channel count and sampling rate. From 72b9eb1de04ddb56f5c3e3ae8b0d1a50847fef5e Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Thu, 8 May 2025 15:53:05 +0200 Subject: [PATCH 123/292] [android] Allow to configure field trials per PeerConnectionFactory MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Provide webrtc::Environment to create AudioProcessing and AudioDeviceModule through java layer Reland of https://webrtc-review.googlesource.com/c/src/+/388620 Bug: webrtc:369904700, webrtc:413413572 Change-Id: Id1ac4bb94a4a80d788a409a235c8ea96d568f170 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390322 Reviewed-by: Zoé Lepaul Commit-Queue: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#44595} --- sdk/android/BUILD.gn | 34 ++++++++++++ .../org/webrtc/AudioProcessingFactory.java | 11 +++- sdk/android/api/org/webrtc/Environment.java | 54 +++++++++++++++++++ .../api/org/webrtc/PeerConnectionFactory.java | 51 ++++++++++-------- .../org/webrtc/audio/AudioDeviceModule.java | 9 +++- .../org/webrtc/PeerConnectionFactoryTest.java | 27 ++++++++++ sdk/android/src/jni/environment.cc | 42 +++++++++++++++ .../src/jni/pc/peer_connection_factory.cc | 11 ++-- 8 files changed, 213 insertions(+), 26 deletions(-) create mode 100644 sdk/android/api/org/webrtc/Environment.java create mode 100644 sdk/android/src/jni/environment.cc diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn index d9ca1e008b..5ce345fb62 100644 --- a/sdk/android/BUILD.gn +++ b/sdk/android/BUILD.gn @@ -117,6 +117,7 @@ if (is_android) { ":base_jni", ":builtin_audio_codecs_jni", ":default_video_codec_factory_jni", + ":environment_jni", ":java_audio_device_module_jni", ":peerconnection_jni", ":video_jni", @@ -182,6 +183,15 @@ if (is_android) { ] } + rtc_android_library("environment_java") { + visibility = [ "*" ] + sources = [ "api/org/webrtc/Environment.java" ] + deps = [ + ":generated_environment_jni_java", + "//third_party/androidx:androidx_annotation_annotation_java", + ] + } + rtc_android_library("audio_api_java") { visibility = [ "*" ] sources = [ @@ -330,6 +340,7 @@ if (is_android) { ":base_java", ":builtin_audio_codecs_java", ":default_video_codec_factory_java", + ":environment_java", ":generated_base_jni_java", ":generated_peerconnection_jni_java", @@ -616,6 +627,21 @@ if (current_os == "linux" || is_android) { ] } + rtc_library("environment_jni") { + visibility = [ "*" ] + allow_poison = [ "environment_construction" ] + sources = [ "src/jni/environment.cc" ] + deps = [ + ":generated_environment_jni", + ":native_api_jni", + "../../api:field_trials", + "../../api:field_trials_view", + "../../api/environment", + "../../api/environment:environment_factory", + "//third_party/jni_zero", + ] + } + rtc_library("builtin_audio_codecs_jni") { visibility = [ "*" ] allow_poison = [ "audio_codecs" ] @@ -778,6 +804,7 @@ if (current_os == "linux" || is_android) { deps = [ ":base_jni", + ":environment_jni", ":generated_external_classes_jni", ":generated_external_classes_jni", ":generated_peerconnection_jni", @@ -805,6 +832,7 @@ if (current_os == "linux" || is_android) { "../../api/audio:builtin_audio_processing_builder", "../../api/audio_codecs:audio_codecs_api", "../../api/crypto:options", + "../../api/environment", "../../api/neteq:neteq_api", "../../api/rtc_event_log:rtc_event_log_factory", "../../api/task_queue:default_task_queue_factory", @@ -1319,6 +1347,12 @@ if (current_os == "linux" || is_android) { jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h" } + generate_jni("generated_environment_jni") { + sources = [ "api/org/webrtc/Environment.java" ] + namespace = "webrtc::jni" + jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h" + } + generate_jni("generated_metrics_jni") { sources = [ "api/org/webrtc/Metrics.java" ] namespace = "webrtc::jni" diff --git a/sdk/android/api/org/webrtc/AudioProcessingFactory.java b/sdk/android/api/org/webrtc/AudioProcessingFactory.java index bd8fdb8989..e0aa25b87f 100644 --- a/sdk/android/api/org/webrtc/AudioProcessingFactory.java +++ b/sdk/android/api/org/webrtc/AudioProcessingFactory.java @@ -16,5 +16,14 @@ public interface AudioProcessingFactory { * Dynamically allocates a webrtc::AudioProcessing instance and returns a pointer to it. * The caller takes ownership of the object. */ - public long createNative(); + // TODO: bugs.webrtc.org/369904700 - Remove when implementations switch to + // another variant. + @Deprecated + public default long createNative() { + return 0; + } + + public default long createNative(long webrtcEnvRef) { + return createNative(); + } } diff --git a/sdk/android/api/org/webrtc/Environment.java b/sdk/android/api/org/webrtc/Environment.java new file mode 100644 index 0000000000..10749ec4e2 --- /dev/null +++ b/sdk/android/api/org/webrtc/Environment.java @@ -0,0 +1,54 @@ +/* + * Copyright 2025 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +package org.webrtc; + +import androidx.annotation.Nullable; + +/** Wrapper for the webrtc::Environment native class. */ +public final class Environment implements AutoCloseable { + private final long webrtcEnv; + + /** Builder for {@link Environment}. */ + public static class Builder { + public Builder setFieldTrials(String fieldTrials) { + this.fieldTrials = fieldTrials; + return this; + } + + public Environment build() { + return new Environment(this.fieldTrials); + } + + private @Nullable String fieldTrials; + } + + public static Builder builder() { + return new Builder(); + } + + /** Returns non-owning non-null native pointer to the webrtc::Environment */ + public long ref() { + return webrtcEnv; + } + + @Override + public void close() { + nativeFree(webrtcEnv); + } + + private Environment(@Nullable String fieldTrials) { + this.webrtcEnv = nativeCreate(fieldTrials); + } + + private static native long nativeCreate(@Nullable String fieldTrials); + + private static native void nativeFree(long webrtcEnv); +} diff --git a/sdk/android/api/org/webrtc/PeerConnectionFactory.java b/sdk/android/api/org/webrtc/PeerConnectionFactory.java index 2b2ea155c1..eac3bb02a0 100644 --- a/sdk/android/api/org/webrtc/PeerConnectionFactory.java +++ b/sdk/android/api/org/webrtc/PeerConnectionFactory.java @@ -15,9 +15,6 @@ import androidx.annotation.Nullable; import java.util.List; import org.webrtc.Logging.Severity; -import org.webrtc.MediaStreamTrack; -import org.webrtc.PeerConnection; -import org.webrtc.RtpCapabilities; import org.webrtc.audio.AudioDeviceModule; import org.webrtc.audio.JavaAudioDeviceModule; @@ -166,6 +163,7 @@ boolean getDisableNetworkMonitor() { public static class Builder { @Nullable private Options options; + private Environment.Builder envBuilder = Environment.builder(); @Nullable private AudioDeviceModule audioDeviceModule; private AudioEncoderFactoryFactory audioEncoderFactoryFactory = new BuiltinAudioEncoderFactoryFactory(); @@ -186,6 +184,11 @@ public Builder setOptions(Options options) { return this; } + public Builder setFieldTrials(String fieldTrials) { + envBuilder.setFieldTrials(fieldTrials); + return this; + } + public Builder setAudioDeviceModule(AudioDeviceModule audioDeviceModule) { this.audioDeviceModule = audioDeviceModule; return this; @@ -261,24 +264,30 @@ public Builder setNetEqFactoryFactory(NetEqFactoryFactory neteqFactoryFactory) { public PeerConnectionFactory createPeerConnectionFactory() { checkInitializeHasBeenCalled(); - if (audioDeviceModule == null) { - audioDeviceModule = JavaAudioDeviceModule.builder(ContextUtils.getApplicationContext()) - .createAudioDeviceModule(); + try (Environment env = envBuilder.build()) { + if (audioDeviceModule == null) { + audioDeviceModule = JavaAudioDeviceModule.builder(ContextUtils.getApplicationContext()) + .createAudioDeviceModule(); + } + return nativeCreatePeerConnectionFactory( + ContextUtils.getApplicationContext(), + options, + env.ref(), + audioDeviceModule.getNative(env.ref()), + audioEncoderFactoryFactory.createNativeAudioEncoderFactory(), + audioDecoderFactoryFactory.createNativeAudioDecoderFactory(), + videoEncoderFactory, + videoDecoderFactory, + audioProcessingFactory == null ? 0 : audioProcessingFactory.createNative(env.ref()), + fecControllerFactoryFactory == null ? 0 : fecControllerFactoryFactory.createNative(), + networkControllerFactoryFactory == null + ? 0 + : networkControllerFactoryFactory.createNativeNetworkControllerFactory(), + networkStatePredictorFactoryFactory == null + ? 0 + : networkStatePredictorFactoryFactory.createNativeNetworkStatePredictorFactory(), + neteqFactoryFactory == null ? 0 : neteqFactoryFactory.createNativeNetEqFactory()); } - return nativeCreatePeerConnectionFactory(ContextUtils.getApplicationContext(), options, - audioDeviceModule.getNativeAudioDeviceModulePointer(), - audioEncoderFactoryFactory.createNativeAudioEncoderFactory(), - audioDecoderFactoryFactory.createNativeAudioDecoderFactory(), videoEncoderFactory, - videoDecoderFactory, - audioProcessingFactory == null ? 0 : audioProcessingFactory.createNative(), - fecControllerFactoryFactory == null ? 0 : fecControllerFactoryFactory.createNative(), - networkControllerFactoryFactory == null - ? 0 - : networkControllerFactoryFactory.createNativeNetworkControllerFactory(), - networkStatePredictorFactoryFactory == null - ? 0 - : networkStatePredictorFactoryFactory.createNativeNetworkStatePredictorFactory(), - neteqFactoryFactory == null ? 0 : neteqFactoryFactory.createNativeNetEqFactory()); } } @@ -604,7 +613,7 @@ private void onSignalingThreadReady() { private static native void nativeStopInternalTracingCapture(); private static native PeerConnectionFactory nativeCreatePeerConnectionFactory(Context context, - Options options, long nativeAudioDeviceModule, long audioEncoderFactory, + Options options, long webrtcEnvRef, long nativeAudioDeviceModule, long audioEncoderFactory, long audioDecoderFactory, VideoEncoderFactory encoderFactory, VideoDecoderFactory decoderFactory, long nativeAudioProcessor, long nativeFecControllerFactory, long nativeNetworkControllerFactory, diff --git a/sdk/android/api/org/webrtc/audio/AudioDeviceModule.java b/sdk/android/api/org/webrtc/audio/AudioDeviceModule.java index 5a0bf5c74d..ba5097d6c8 100644 --- a/sdk/android/api/org/webrtc/audio/AudioDeviceModule.java +++ b/sdk/android/api/org/webrtc/audio/AudioDeviceModule.java @@ -22,7 +22,14 @@ public interface AudioDeviceModule { * Returns a C++ pointer to a webrtc::AudioDeviceModule. Caller does _not_ take ownership and * lifetime is handled through the release() call. */ - long getNativeAudioDeviceModulePointer(); + @Deprecated + default long getNativeAudioDeviceModulePointer() { + return 0; + } + + public default long getNative(long webrtcEnvRef) { + return getNativeAudioDeviceModulePointer(); + } /** * Release resources for this AudioDeviceModule, including native resources. The object should not diff --git a/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionFactoryTest.java b/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionFactoryTest.java index a28b9c25ff..1e36fef760 100644 --- a/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionFactoryTest.java +++ b/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionFactoryTest.java @@ -62,4 +62,31 @@ public void testInitializeWithTracerAndShutdown() { .createInitializationOptions()); PeerConnectionFactory.shutdownInternalTracer(); } + + // Tests that the JNI glue between Java and C++ does not crash when creating a + // PeerConnectionFactory. + @Test + @SmallTest + public void testCreation() throws Exception { + PeerConnectionFactory.initialize( + PeerConnectionFactory.InitializationOptions.builder( + InstrumentationRegistry.getTargetContext()) + .setNativeLibraryName(TestConstants.NATIVE_LIBRARY) + .createInitializationOptions()); + + PeerConnectionFactory factory = PeerConnectionFactory.builder().createPeerConnectionFactory(); + } + + @Test + @SmallTest + public void testCreationWithFieldTrials() throws Exception { + PeerConnectionFactory.initialize( + PeerConnectionFactory.InitializationOptions.builder( + InstrumentationRegistry.getTargetContext()) + .setNativeLibraryName(TestConstants.NATIVE_LIBRARY) + .createInitializationOptions()); + + PeerConnectionFactory factory = + PeerConnectionFactory.builder().setFieldTrials("").createPeerConnectionFactory(); + } } diff --git a/sdk/android/src/jni/environment.cc b/sdk/android/src/jni/environment.cc new file mode 100644 index 0000000000..fab50ebd5d --- /dev/null +++ b/sdk/android/src/jni/environment.cc @@ -0,0 +1,42 @@ +/* + * Copyright 2025 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ +#include "api/environment/environment.h" + +#include + +#include +#include + +#include "api/environment/environment_factory.h" +#include "api/field_trials.h" +#include "api/field_trials_view.h" +#include "sdk/android/generated_environment_jni/Environment_jni.h" +#include "sdk/android/native_api/jni/java_types.h" +#include "third_party/jni_zero/jni_zero.h" + +namespace webrtc::jni { + +void JNI_Environment_Free(JNIEnv* /*env*/, jlong webrtcEnv) { + delete reinterpret_cast(webrtcEnv); +} + +jlong JNI_Environment_Create( + JNIEnv* env, + const jni_zero::JavaParamRef& fieldTrials) { + std::unique_ptr field_trials; + if (!IsNull(env, fieldTrials)) { + field_trials = + std::make_unique(JavaToNativeString(env, fieldTrials)); + } + return NativeToJavaPointer( + new Environment(CreateEnvironment(std::move(field_trials)))); +} + +} // namespace webrtc::jni diff --git a/sdk/android/src/jni/pc/peer_connection_factory.cc b/sdk/android/src/jni/pc/peer_connection_factory.cc index a09859afbe..fbab65f66e 100644 --- a/sdk/android/src/jni/pc/peer_connection_factory.cc +++ b/sdk/android/src/jni/pc/peer_connection_factory.cc @@ -29,6 +29,7 @@ #include "api/audio_codecs/audio_encoder_factory.h" #include "api/audio_options.h" #include "api/enable_media.h" +#include "api/environment/environment.h" #include "api/fec_controller.h" #include "api/media_stream_interface.h" #include "api/neteq/neteq_factory.h" @@ -256,6 +257,7 @@ ScopedJavaLocalRef CreatePeerConnectionFactoryForJava( JNIEnv* jni, const jni_zero::JavaParamRef& jcontext, const jni_zero::JavaParamRef& joptions, + const Environment& env, scoped_refptr audio_device_module, scoped_refptr audio_encoder_factory, scoped_refptr audio_decoder_factory, @@ -292,8 +294,8 @@ ScopedJavaLocalRef CreatePeerConnectionFactoryForJava( JavaToNativePeerConnectionFactoryOptions(jni, joptions); PeerConnectionFactoryDependencies dependencies; - // TODO(bugs.webrtc.org/13145): Also add socket_server.get() to the - // dependencies. + dependencies.env = env; + dependencies.socket_factory = socket_server.get(); dependencies.network_thread = network_thread.get(); dependencies.worker_thread = worker_thread.get(); dependencies.signaling_thread = signaling_thread.get(); @@ -347,6 +349,7 @@ JNI_PeerConnectionFactory_CreatePeerConnectionFactory( JNIEnv* jni, const jni_zero::JavaParamRef& jcontext, const jni_zero::JavaParamRef& joptions, + jlong webrtc_env_ref, jlong native_audio_device_module, jlong native_audio_encoder_factory, jlong native_audio_decoder_factory, @@ -357,10 +360,12 @@ JNI_PeerConnectionFactory_CreatePeerConnectionFactory( jlong native_network_controller_factory, jlong native_network_state_predictor_factory, jlong native_neteq_factory) { + const Environment* env = reinterpret_cast(webrtc_env_ref); + RTC_CHECK(env != nullptr); scoped_refptr audio_processor( reinterpret_cast(native_audio_processor)); return CreatePeerConnectionFactoryForJava( - jni, jcontext, joptions, + jni, jcontext, joptions, *env, scoped_refptr( reinterpret_cast(native_audio_device_module)), TakeOwnershipOfRefPtr(native_audio_encoder_factory), From d4aa34c26fede6e59ebfa8409c44fdce5469e7f5 Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Thu, 8 May 2025 09:22:02 -0700 Subject: [PATCH 124/292] IWYU common_video/ using find common_video/ -name "*.h" -o -name "*.cc" | xargs tools_webrtc/iwyu/apply-include-cleaner followed by tools_webrtc/gn_check_autofix.py -C out/Default/ and git cl format libyuv include paths were manually prefixed with the third_party path in PS3 and should probably be rewired everywhere to go via webrtc_libyuv.h (which might need to move to a different location) BUG=webrtc:42226242 Change-Id: I8aec94cd6ad25772e82276abde5c1bef920f03e6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/383080 Reviewed-by: Jeremy Leconte Reviewed-by: Sergey Silkin Commit-Queue: Philipp Hancke Cr-Commit-Position: refs/heads/main@{#44596} --- common_video/BUILD.gn | 3 +++ common_video/bitrate_adjuster.cc | 4 ++++ common_video/bitrate_adjuster_unittest.cc | 3 +++ .../corruption_detection_converters_unittest.cc | 1 + common_video/frame_rate_estimator.cc | 4 ++++ common_video/frame_rate_estimator_unittest.cc | 3 ++- common_video/framerate_controller.cc | 3 +++ common_video/framerate_controller_unittest.cc | 2 +- .../generic_frame_descriptor/generic_frame_info.cc | 5 ++--- .../generic_frame_descriptor/generic_frame_info.h | 2 +- common_video/h264/h264_bitstream_parser.cc | 4 ++++ common_video/h264/h264_bitstream_parser.h | 1 + common_video/h264/h264_bitstream_parser_unittest.cc | 3 +++ common_video/h264/h264_common.cc | 5 +++++ common_video/h264/h264_common.h | 1 + common_video/h264/pps_parser.cc | 2 ++ common_video/h264/pps_parser_unittest.cc | 4 ++++ common_video/h264/sps_parser.cc | 3 +++ common_video/h264/sps_parser.h | 2 ++ common_video/h264/sps_parser_unittest.cc | 6 +++++- common_video/h264/sps_vui_rewriter.cc | 3 +++ common_video/h264/sps_vui_rewriter.h | 1 + common_video/h264/sps_vui_rewriter_unittest.cc | 6 +++++- common_video/h265/h265_bitstream_parser.h | 2 +- common_video/h265/h265_bitstream_parser_unittest.cc | 4 ++++ common_video/h265/h265_common.cc | 6 ++++++ common_video/h265/h265_common.h | 5 +++-- common_video/h265/h265_pps_parser.cc | 6 ++++-- common_video/h265/h265_pps_parser.h | 2 ++ common_video/h265/h265_pps_parser_unittest.cc | 7 +++++-- common_video/h265/h265_sps_parser.h | 2 ++ common_video/h265/h265_sps_parser_unittest.cc | 7 ++++++- common_video/h265/h265_vps_parser.cc | 6 ++++-- common_video/h265/h265_vps_parser.h | 2 ++ common_video/h265/h265_vps_parser_unittest.cc | 7 +++---- common_video/include/video_frame_buffer_pool.h | 1 + common_video/libyuv/libyuv_unittest.cc | 13 ++++++++++--- common_video/libyuv/webrtc_libyuv.cc | 11 ++++++++++- common_video/test/utilities.cc | 6 +++++- common_video/test/utilities.h | 3 ++- common_video/video_frame_buffer.cc | 6 ++++++ common_video/video_frame_buffer_pool.cc | 12 ++++++++++++ common_video/video_frame_buffer_pool_unittest.cc | 1 - common_video/video_frame_unittest.cc | 9 ++++++++- 44 files changed, 159 insertions(+), 30 deletions(-) diff --git a/common_video/BUILD.gn b/common_video/BUILD.gn index 1a69762a65..d8bc46c19e 100644 --- a/common_video/BUILD.gn +++ b/common_video/BUILD.gn @@ -144,6 +144,7 @@ if (rtc_include_tests && !build_with_chromium) { ":corruption_detection_converters", ":frame_instrumentation_data", "../api/transport/rtp:corruption_detection_message", + "../rtc_base:checks", "../test:test_support", ] } @@ -176,8 +177,10 @@ if (rtc_include_tests && !build_with_chromium) { deps = [ ":common_video", ":corruption_detection_converters_unittest", + "../api:array_view", "../api:scoped_refptr", "../api/units:time_delta", + "../api/units:timestamp", "../api/video:video_frame", "../api/video:video_frame_i010", "../api/video:video_rtp_headers", diff --git a/common_video/bitrate_adjuster.cc b/common_video/bitrate_adjuster.cc index 8f3ddeb532..66f31d9e0b 100644 --- a/common_video/bitrate_adjuster.cc +++ b/common_video/bitrate_adjuster.cc @@ -12,8 +12,12 @@ #include #include +#include +#include +#include #include "rtc_base/logging.h" +#include "rtc_base/synchronization/mutex.h" #include "rtc_base/time_utils.h" namespace webrtc { diff --git a/common_video/bitrate_adjuster_unittest.cc b/common_video/bitrate_adjuster_unittest.cc index 8660533281..088e71fd89 100644 --- a/common_video/bitrate_adjuster_unittest.cc +++ b/common_video/bitrate_adjuster_unittest.cc @@ -10,6 +10,9 @@ #include "common_video/include/bitrate_adjuster.h" +#include +#include + #include "api/units/time_delta.h" #include "rtc_base/fake_clock.h" #include "test/gtest.h" diff --git a/common_video/corruption_detection_converters_unittest.cc b/common_video/corruption_detection_converters_unittest.cc index 8b3372e653..c84fb597c9 100644 --- a/common_video/corruption_detection_converters_unittest.cc +++ b/common_video/corruption_detection_converters_unittest.cc @@ -15,6 +15,7 @@ #include "api/transport/rtp/corruption_detection_message.h" #include "common_video/frame_instrumentation_data.h" +#include "rtc_base/checks.h" #include "test/gmock.h" #include "test/gtest.h" diff --git a/common_video/frame_rate_estimator.cc b/common_video/frame_rate_estimator.cc index c83f268c6c..09a13bda94 100644 --- a/common_video/frame_rate_estimator.cc +++ b/common_video/frame_rate_estimator.cc @@ -10,6 +10,10 @@ #include "common_video/frame_rate_estimator.h" +#include + +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" #include "rtc_base/time_utils.h" namespace webrtc { diff --git a/common_video/frame_rate_estimator_unittest.cc b/common_video/frame_rate_estimator_unittest.cc index d11a7d4140..f4d37cd97e 100644 --- a/common_video/frame_rate_estimator_unittest.cc +++ b/common_video/frame_rate_estimator_unittest.cc @@ -10,8 +10,9 @@ #include "common_video/frame_rate_estimator.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" #include "system_wrappers/include/clock.h" -#include "test/gmock.h" #include "test/gtest.h" namespace webrtc { diff --git a/common_video/framerate_controller.cc b/common_video/framerate_controller.cc index 771a369bca..5a1326d3c6 100644 --- a/common_video/framerate_controller.cc +++ b/common_video/framerate_controller.cc @@ -10,7 +10,10 @@ #include "common_video/framerate_controller.h" +#include +#include #include +#include #include "rtc_base/time_utils.h" diff --git a/common_video/framerate_controller_unittest.cc b/common_video/framerate_controller_unittest.cc index a0a323c654..4b0abdef4e 100644 --- a/common_video/framerate_controller_unittest.cc +++ b/common_video/framerate_controller_unittest.cc @@ -10,7 +10,7 @@ #include "common_video/framerate_controller.h" -#include +#include #include "rtc_base/time_utils.h" #include "test/gtest.h" diff --git a/common_video/generic_frame_descriptor/generic_frame_info.cc b/common_video/generic_frame_descriptor/generic_frame_info.cc index af66bbaf67..168304e981 100644 --- a/common_video/generic_frame_descriptor/generic_frame_info.cc +++ b/common_video/generic_frame_descriptor/generic_frame_info.cc @@ -9,9 +9,8 @@ */ #include "common_video/generic_frame_descriptor/generic_frame_info.h" -#include - -#include "rtc_base/checks.h" +#include "absl/strings/string_view.h" +#include "api/transport/rtp/dependency_descriptor.h" namespace webrtc { diff --git a/common_video/generic_frame_descriptor/generic_frame_info.h b/common_video/generic_frame_descriptor/generic_frame_info.h index 30db423e37..45037b706e 100644 --- a/common_video/generic_frame_descriptor/generic_frame_info.h +++ b/common_video/generic_frame_descriptor/generic_frame_info.h @@ -12,7 +12,7 @@ #define COMMON_VIDEO_GENERIC_FRAME_DESCRIPTOR_GENERIC_FRAME_INFO_H_ #include -#include +#include #include #include "absl/container/inlined_vector.h" diff --git a/common_video/h264/h264_bitstream_parser.cc b/common_video/h264/h264_bitstream_parser.cc index 1d2d0496a5..d6a2e48603 100644 --- a/common_video/h264/h264_bitstream_parser.cc +++ b/common_video/h264/h264_bitstream_parser.cc @@ -12,9 +12,13 @@ #include #include +#include #include +#include "api/array_view.h" #include "common_video/h264/h264_common.h" +#include "common_video/h264/pps_parser.h" +#include "common_video/h264/sps_parser.h" #include "rtc_base/bitstream_reader.h" #include "rtc_base/logging.h" diff --git a/common_video/h264/h264_bitstream_parser.h b/common_video/h264/h264_bitstream_parser.h index e308fe1c10..2dafa86185 100644 --- a/common_video/h264/h264_bitstream_parser.h +++ b/common_video/h264/h264_bitstream_parser.h @@ -15,6 +15,7 @@ #include +#include "api/array_view.h" #include "api/video_codecs/bitstream_parser.h" #include "common_video/h264/pps_parser.h" #include "common_video/h264/sps_parser.h" diff --git a/common_video/h264/h264_bitstream_parser_unittest.cc b/common_video/h264/h264_bitstream_parser_unittest.cc index 00186a6a34..233486481f 100644 --- a/common_video/h264/h264_bitstream_parser_unittest.cc +++ b/common_video/h264/h264_bitstream_parser_unittest.cc @@ -10,6 +10,9 @@ #include "common_video/h264/h264_bitstream_parser.h" +#include +#include + #include "test/gtest.h" namespace webrtc { diff --git a/common_video/h264/h264_common.cc b/common_video/h264/h264_common.cc index 5444fcd6eb..2533bab965 100644 --- a/common_video/h264/h264_common.cc +++ b/common_video/h264/h264_common.cc @@ -10,7 +10,12 @@ #include "common_video/h264/h264_common.h" +#include #include +#include + +#include "api/array_view.h" +#include "rtc_base/buffer.h" namespace webrtc { namespace H264 { diff --git a/common_video/h264/h264_common.h b/common_video/h264/h264_common.h index e197fb4407..68c9504107 100644 --- a/common_video/h264/h264_common.h +++ b/common_video/h264/h264_common.h @@ -16,6 +16,7 @@ #include +#include "api/array_view.h" #include "rtc_base/buffer.h" #include "rtc_base/system/rtc_export.h" diff --git a/common_video/h264/pps_parser.cc b/common_video/h264/pps_parser.cc index 9d9d04a2c2..64a49feda8 100644 --- a/common_video/h264/pps_parser.cc +++ b/common_video/h264/pps_parser.cc @@ -12,9 +12,11 @@ #include #include +#include #include #include "absl/numeric/bits.h" +#include "api/array_view.h" #include "common_video/h264/h264_common.h" #include "rtc_base/bitstream_reader.h" #include "rtc_base/checks.h" diff --git a/common_video/h264/pps_parser_unittest.cc b/common_video/h264/pps_parser_unittest.cc index 287909f533..e3a9c08fd8 100644 --- a/common_video/h264/pps_parser_unittest.cc +++ b/common_video/h264/pps_parser_unittest.cc @@ -10,8 +10,12 @@ #include "common_video/h264/pps_parser.h" +#include +#include +#include #include +#include "api/array_view.h" #include "common_video/h264/h264_common.h" #include "rtc_base/bit_buffer.h" #include "rtc_base/buffer.h" diff --git a/common_video/h264/sps_parser.cc b/common_video/h264/sps_parser.cc index 197fac9573..cf99700a71 100644 --- a/common_video/h264/sps_parser.cc +++ b/common_video/h264/sps_parser.cc @@ -10,9 +10,12 @@ #include "common_video/h264/sps_parser.h" +#include #include +#include #include +#include "api/array_view.h" #include "common_video/h264/h264_common.h" #include "rtc_base/bitstream_reader.h" diff --git a/common_video/h264/sps_parser.h b/common_video/h264/sps_parser.h index a8a6675e0a..b24194c6e9 100644 --- a/common_video/h264/sps_parser.h +++ b/common_video/h264/sps_parser.h @@ -11,8 +11,10 @@ #ifndef COMMON_VIDEO_H264_SPS_PARSER_H_ #define COMMON_VIDEO_H264_SPS_PARSER_H_ +#include #include +#include "api/array_view.h" #include "rtc_base/bitstream_reader.h" #include "rtc_base/system/rtc_export.h" diff --git a/common_video/h264/sps_parser_unittest.cc b/common_video/h264/sps_parser_unittest.cc index 0f20eb6a31..d2396d016a 100644 --- a/common_video/h264/sps_parser_unittest.cc +++ b/common_video/h264/sps_parser_unittest.cc @@ -10,8 +10,12 @@ #include "common_video/h264/sps_parser.h" +#include +#include +#include + +#include "api/array_view.h" #include "common_video/h264/h264_common.h" -#include "rtc_base/arraysize.h" #include "rtc_base/bit_buffer.h" #include "rtc_base/buffer.h" #include "test/gtest.h" diff --git a/common_video/h264/sps_vui_rewriter.cc b/common_video/h264/sps_vui_rewriter.cc index 0684b0d86e..497f0932f6 100644 --- a/common_video/h264/sps_vui_rewriter.cc +++ b/common_video/h264/sps_vui_rewriter.cc @@ -15,13 +15,16 @@ #include #include +#include #include +#include "api/array_view.h" #include "api/video/color_space.h" #include "common_video/h264/h264_common.h" #include "common_video/h264/sps_parser.h" #include "rtc_base/bit_buffer.h" #include "rtc_base/bitstream_reader.h" +#include "rtc_base/buffer.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "system_wrappers/include/metrics.h" diff --git a/common_video/h264/sps_vui_rewriter.h b/common_video/h264/sps_vui_rewriter.h index 8cdc04e5ef..665142f3a0 100644 --- a/common_video/h264/sps_vui_rewriter.h +++ b/common_video/h264/sps_vui_rewriter.h @@ -17,6 +17,7 @@ #include +#include "api/array_view.h" #include "api/video/color_space.h" #include "common_video/h264/sps_parser.h" #include "rtc_base/buffer.h" diff --git a/common_video/h264/sps_vui_rewriter_unittest.cc b/common_video/h264/sps_vui_rewriter_unittest.cc index 2b0f06d5f7..b9c6c86bf5 100644 --- a/common_video/h264/sps_vui_rewriter_unittest.cc +++ b/common_video/h264/sps_vui_rewriter_unittest.cc @@ -10,11 +10,15 @@ #include "common_video/h264/sps_vui_rewriter.h" +#include #include -#include +#include +#include +#include "api/array_view.h" #include "api/video/color_space.h" #include "common_video/h264/h264_common.h" +#include "common_video/h264/sps_parser.h" #include "rtc_base/bit_buffer.h" #include "rtc_base/buffer.h" #include "rtc_base/logging.h" diff --git a/common_video/h265/h265_bitstream_parser.h b/common_video/h265/h265_bitstream_parser.h index 7c337f8be4..20eb8dfb74 100644 --- a/common_video/h265/h265_bitstream_parser.h +++ b/common_video/h265/h265_bitstream_parser.h @@ -15,8 +15,8 @@ #include #include -#include +#include "api/array_view.h" #include "api/video_codecs/bitstream_parser.h" #include "common_video/h265/h265_pps_parser.h" #include "common_video/h265/h265_sps_parser.h" diff --git a/common_video/h265/h265_bitstream_parser_unittest.cc b/common_video/h265/h265_bitstream_parser_unittest.cc index e359c5c2ae..754300b8c7 100644 --- a/common_video/h265/h265_bitstream_parser_unittest.cc +++ b/common_video/h265/h265_bitstream_parser_unittest.cc @@ -10,6 +10,10 @@ #include "common_video/h265/h265_bitstream_parser.h" +#include +#include + +#include "api/array_view.h" #include "common_video/h265/h265_common.h" #include "test/gmock.h" #include "test/gtest.h" diff --git a/common_video/h265/h265_common.cc b/common_video/h265/h265_common.cc index cb5557dc0f..2705d52773 100644 --- a/common_video/h265/h265_common.cc +++ b/common_video/h265/h265_common.cc @@ -10,7 +10,13 @@ #include "common_video/h265/h265_common.h" +#include +#include + +#include "api/array_view.h" #include "common_video/h264/h264_common.h" +#include "common_video/h265/h265_inline.h" +#include "rtc_base/buffer.h" namespace webrtc { namespace H265 { diff --git a/common_video/h265/h265_common.h b/common_video/h265/h265_common.h index 7bba7f84a7..9ce9fbee58 100644 --- a/common_video/h265/h265_common.h +++ b/common_video/h265/h265_common.h @@ -11,10 +11,11 @@ #ifndef COMMON_VIDEO_H265_H265_COMMON_H_ #define COMMON_VIDEO_H265_H265_COMMON_H_ -#include +#include +#include #include -#include "common_video/h265/h265_inline.h" +#include "api/array_view.h" #include "rtc_base/buffer.h" #include "rtc_base/system/rtc_export.h" diff --git a/common_video/h265/h265_pps_parser.cc b/common_video/h265/h265_pps_parser.cc index 52c1026f1e..c94e362ec9 100644 --- a/common_video/h265/h265_pps_parser.cc +++ b/common_video/h265/h265_pps_parser.cc @@ -10,13 +10,15 @@ #include "common_video/h265/h265_pps_parser.h" -#include +#include #include #include +#include "api/array_view.h" #include "common_video/h265/h265_common.h" -#include "rtc_base/bit_buffer.h" +#include "common_video/h265/h265_sps_parser.h" #include "rtc_base/bitstream_reader.h" +#include "rtc_base/checks.h" #include "rtc_base/logging.h" #define IN_RANGE_OR_RETURN_NULL(val, min, max) \ diff --git a/common_video/h265/h265_pps_parser.h b/common_video/h265/h265_pps_parser.h index ab4fd9a481..2cb3f67dab 100644 --- a/common_video/h265/h265_pps_parser.h +++ b/common_video/h265/h265_pps_parser.h @@ -11,6 +11,8 @@ #ifndef COMMON_VIDEO_H265_H265_PPS_PARSER_H_ #define COMMON_VIDEO_H265_H265_PPS_PARSER_H_ +#include +#include #include #include "api/array_view.h" diff --git a/common_video/h265/h265_pps_parser_unittest.cc b/common_video/h265/h265_pps_parser_unittest.cc index ff0058b2cf..688e9e98d7 100644 --- a/common_video/h265/h265_pps_parser_unittest.cc +++ b/common_video/h265/h265_pps_parser_unittest.cc @@ -11,12 +11,15 @@ #include "common_video/h265/h265_pps_parser.h" #include +#include +#include +#include +#include "api/array_view.h" #include "common_video/h265/h265_common.h" -#include "rtc_base/arraysize.h" +#include "common_video/h265/h265_sps_parser.h" #include "rtc_base/bit_buffer.h" #include "rtc_base/buffer.h" -#include "rtc_base/checks.h" #include "test/gtest.h" namespace webrtc { diff --git a/common_video/h265/h265_sps_parser.h b/common_video/h265/h265_sps_parser.h index 7830477462..1bccf6adcc 100644 --- a/common_video/h265/h265_sps_parser.h +++ b/common_video/h265/h265_sps_parser.h @@ -11,6 +11,8 @@ #ifndef COMMON_VIDEO_H265_H265_SPS_PARSER_H_ #define COMMON_VIDEO_H265_H265_SPS_PARSER_H_ +#include +#include #include #include diff --git a/common_video/h265/h265_sps_parser_unittest.cc b/common_video/h265/h265_sps_parser_unittest.cc index c51b7ab1c2..1d3e1b2859 100644 --- a/common_video/h265/h265_sps_parser_unittest.cc +++ b/common_video/h265/h265_sps_parser_unittest.cc @@ -10,8 +10,13 @@ #include "common_video/h265/h265_sps_parser.h" +#include +#include +#include +#include + +#include "api/array_view.h" #include "common_video/h265/h265_common.h" -#include "rtc_base/arraysize.h" #include "rtc_base/bit_buffer.h" #include "rtc_base/buffer.h" #include "test/gtest.h" diff --git a/common_video/h265/h265_vps_parser.cc b/common_video/h265/h265_vps_parser.cc index 5b3fc6b12f..8ebe5f96a1 100644 --- a/common_video/h265/h265_vps_parser.cc +++ b/common_video/h265/h265_vps_parser.cc @@ -10,10 +10,12 @@ #include "common_video/h265/h265_vps_parser.h" +#include +#include + +#include "api/array_view.h" #include "common_video/h265/h265_common.h" -#include "rtc_base/bit_buffer.h" #include "rtc_base/bitstream_reader.h" -#include "rtc_base/logging.h" namespace webrtc { diff --git a/common_video/h265/h265_vps_parser.h b/common_video/h265/h265_vps_parser.h index 619ae2e211..e890e4e64f 100644 --- a/common_video/h265/h265_vps_parser.h +++ b/common_video/h265/h265_vps_parser.h @@ -11,6 +11,8 @@ #ifndef COMMON_VIDEO_H265_H265_VPS_PARSER_H_ #define COMMON_VIDEO_H265_H265_VPS_PARSER_H_ +#include +#include #include #include "api/array_view.h" diff --git a/common_video/h265/h265_vps_parser_unittest.cc b/common_video/h265/h265_vps_parser_unittest.cc index fd2e8a8f52..740d676035 100644 --- a/common_video/h265/h265_vps_parser_unittest.cc +++ b/common_video/h265/h265_vps_parser_unittest.cc @@ -10,10 +10,9 @@ #include "common_video/h265/h265_vps_parser.h" -#include "common_video/h265/h265_common.h" -#include "rtc_base/arraysize.h" -#include "rtc_base/bit_buffer.h" -#include "rtc_base/buffer.h" +#include +#include + #include "test/gtest.h" namespace webrtc { diff --git a/common_video/include/video_frame_buffer_pool.h b/common_video/include/video_frame_buffer_pool.h index 2eabe910ea..77df050686 100644 --- a/common_video/include/video_frame_buffer_pool.h +++ b/common_video/include/video_frame_buffer_pool.h @@ -23,6 +23,7 @@ #include "api/video/i422_buffer.h" #include "api/video/i444_buffer.h" #include "api/video/nv12_buffer.h" +#include "api/video/video_frame_buffer.h" #include "rtc_base/race_checker.h" namespace webrtc { diff --git a/common_video/libyuv/libyuv_unittest.cc b/common_video/libyuv/libyuv_unittest.cc index b0e97943ea..fed19fd2e1 100644 --- a/common_video/libyuv/libyuv_unittest.cc +++ b/common_video/libyuv/libyuv_unittest.cc @@ -8,13 +8,17 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "third_party/libyuv/include/libyuv.h" - #include #include +#include +#include +#include #include +#include +#include +#include "api/scoped_refptr.h" #include "api/video/i010_buffer.h" #include "api/video/i210_buffer.h" #include "api/video/i410_buffer.h" @@ -23,12 +27,15 @@ #include "api/video/i444_buffer.h" #include "api/video/nv12_buffer.h" #include "api/video/video_frame.h" +#include "api/video/video_frame_buffer.h" +#include "api/video/video_rotation.h" #include "common_video/libyuv/include/webrtc_libyuv.h" -#include "rtc_base/logging.h" #include "test/frame_utils.h" #include "test/gmock.h" #include "test/gtest.h" #include "test/testsupport/file_utils.h" +#include "third_party/libyuv/include/libyuv/convert.h" +#include "third_party/libyuv/include/libyuv/rotate.h" namespace webrtc { diff --git a/common_video/libyuv/webrtc_libyuv.cc b/common_video/libyuv/webrtc_libyuv.cc index 8524bcbb11..916acafe4f 100644 --- a/common_video/libyuv/webrtc_libyuv.cc +++ b/common_video/libyuv/webrtc_libyuv.cc @@ -10,12 +10,21 @@ #include "common_video/libyuv/include/webrtc_libyuv.h" +#include #include +#include "api/scoped_refptr.h" #include "api/video/i420_buffer.h" +#include "api/video/video_frame.h" +#include "api/video/video_frame_buffer.h" #include "common_video/include/video_frame_buffer.h" #include "rtc_base/checks.h" -#include "third_party/libyuv/include/libyuv.h" +#include "third_party/libyuv/include/libyuv/compare.h" +#include "third_party/libyuv/include/libyuv/convert.h" +#include "third_party/libyuv/include/libyuv/convert_from.h" +#include "third_party/libyuv/include/libyuv/planar_functions.h" +#include "third_party/libyuv/include/libyuv/scale.h" +#include "third_party/libyuv/include/libyuv/video_common.h" namespace webrtc { diff --git a/common_video/test/utilities.cc b/common_video/test/utilities.cc index c2a3266dc1..e4bfe6e5e9 100644 --- a/common_video/test/utilities.cc +++ b/common_video/test/utilities.cc @@ -9,7 +9,11 @@ */ #include "common_video/test/utilities.h" -#include +#include + +#include "api/rtp_packet_infos.h" +#include "api/video/color_space.h" +#include "api/video/hdr_metadata.h" namespace webrtc { diff --git a/common_video/test/utilities.h b/common_video/test/utilities.h index 7e15cf947c..eb7cddf99a 100644 --- a/common_video/test/utilities.h +++ b/common_video/test/utilities.h @@ -11,10 +11,11 @@ #ifndef COMMON_VIDEO_TEST_UTILITIES_H_ #define COMMON_VIDEO_TEST_UTILITIES_H_ -#include +#include #include "api/rtp_packet_infos.h" #include "api/video/color_space.h" +#include "api/video/hdr_metadata.h" namespace webrtc { diff --git a/common_video/video_frame_buffer.cc b/common_video/video_frame_buffer.cc index e88cd0f3f0..0b043b6e03 100644 --- a/common_video/video_frame_buffer.cc +++ b/common_video/video_frame_buffer.cc @@ -9,9 +9,15 @@ */ #include "common_video/include/video_frame_buffer.h" +#include +#include + #include "api/make_ref_counted.h" +#include "api/scoped_refptr.h" #include "api/video/i420_buffer.h" +#include "api/video/video_frame_buffer.h" #include "rtc_base/checks.h" +#include "rtc_base/ref_counted_object.h" #include "third_party/libyuv/include/libyuv/convert.h" namespace webrtc { diff --git a/common_video/video_frame_buffer_pool.cc b/common_video/video_frame_buffer_pool.cc index d7349e1924..07d3fd00c8 100644 --- a/common_video/video_frame_buffer_pool.cc +++ b/common_video/video_frame_buffer_pool.cc @@ -10,10 +10,22 @@ #include "common_video/include/video_frame_buffer_pool.h" +#include #include #include "api/make_ref_counted.h" +#include "api/scoped_refptr.h" +#include "api/video/i010_buffer.h" +#include "api/video/i210_buffer.h" +#include "api/video/i410_buffer.h" +#include "api/video/i420_buffer.h" +#include "api/video/i422_buffer.h" +#include "api/video/i444_buffer.h" +#include "api/video/nv12_buffer.h" +#include "api/video/video_frame_buffer.h" #include "rtc_base/checks.h" +#include "rtc_base/race_checker.h" +#include "rtc_base/ref_counted_object.h" namespace webrtc { diff --git a/common_video/video_frame_buffer_pool_unittest.cc b/common_video/video_frame_buffer_pool_unittest.cc index 0d8a37bbd2..6c3feb85a3 100644 --- a/common_video/video_frame_buffer_pool_unittest.cc +++ b/common_video/video_frame_buffer_pool_unittest.cc @@ -15,7 +15,6 @@ #include "api/scoped_refptr.h" #include "api/video/i420_buffer.h" -#include "api/video/video_frame_buffer.h" #include "test/gtest.h" namespace webrtc { diff --git a/common_video/video_frame_unittest.cc b/common_video/video_frame_unittest.cc index 21ddf8d85e..a10f322805 100644 --- a/common_video/video_frame_unittest.cc +++ b/common_video/video_frame_unittest.cc @@ -13,6 +13,12 @@ #include #include +#include +#include +#include +#include + +#include "api/scoped_refptr.h" #include "api/video/i010_buffer.h" #include "api/video/i210_buffer.h" #include "api/video/i410_buffer.h" @@ -20,7 +26,8 @@ #include "api/video/i422_buffer.h" #include "api/video/i444_buffer.h" #include "api/video/nv12_buffer.h" -#include "rtc_base/time_utils.h" +#include "api/video/video_frame_buffer.h" +#include "api/video/video_rotation.h" #include "test/fake_texture_frame.h" #include "test/frame_utils.h" #include "test/gtest.h" From bcde0f6c885751a4356227844d5ef65bba13874d Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Fri, 9 May 2025 13:58:42 -0700 Subject: [PATCH 125/292] Reland "document footgun behavior of codec munging and setCodecPreferences" This is a reland of commit e7162754c0229d10c903525be3f900edb184dfb7 using VP9 instead of AV1. Original change's description: > document footgun behavior of codec munging and setCodecPreferences > > which drops the munged codec > > BUG=webrtc:396344370 > > Change-Id: I4980c6980ad7df29d4d1451777a61f90dac73f74 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389440 > Reviewed-by: Evan Shrubsole > Reviewed-by: Harald Alvestrand > Commit-Queue: Philipp Hancke > Cr-Commit-Position: refs/heads/main@{#44555} Bug: webrtc:396344370 Change-Id: I078c22d110d4adbf10641b1b4ba180cfc9b83ddd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390544 Reviewed-by: Fredrik Solenberg Reviewed-by: Harald Alvestrand Commit-Queue: Philipp Hancke Cr-Commit-Position: refs/heads/main@{#44597} --- pc/peer_connection_integrationtest.cc | 106 +++++++++++++++++++++++++- 1 file changed, 104 insertions(+), 2 deletions(-) diff --git a/pc/peer_connection_integrationtest.cc b/pc/peer_connection_integrationtest.cc index 67d9652227..b083a75fbb 100644 --- a/pc/peer_connection_integrationtest.cc +++ b/pc/peer_connection_integrationtest.cc @@ -18,6 +18,7 @@ #include #include +#include #include #include #include @@ -4779,6 +4780,107 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan, EXPECT_TRUE(caller()->SetLocalDescriptionAndSendSdpMessage(std::move(offer))); } +TEST_F(PeerConnectionIntegrationTestUnifiedPlan, + MungeOfferCodecAndReOfferWorksWithSetCodecPreferencesIsAFootgun) { + ASSERT_TRUE(CreatePeerConnectionWrappers()); + ConnectFakeSignaling(); + caller()->AddVideoTrack(); + bool has_munged = false; + auto munger = + [&has_munged](std::unique_ptr& sdp) { + auto video = GetFirstVideoContentDescription(sdp->description()); + auto codecs = video->codecs(); + std::optional replacement_codec; + for (auto&& codec : codecs) { + if (codec.name == "VP9") { + replacement_codec = codec; + break; + } + } + if (replacement_codec) { + for (auto&& codec : codecs) { + if (codec.name == "VP8") { + RTC_LOG(LS_INFO) << "Remapping VP8 codec " << codec << " to VP9"; + codec.name = replacement_codec->name; + codec.params = replacement_codec->params; + has_munged = true; + break; + } + } + video->set_codecs(codecs); + } else { + RTC_LOG(LS_INFO) << "Skipping munge, no VP9 codec found "; + } + }; + caller()->SetGeneratedSdpMunger(munger); + caller()->CreateAndSetAndSignalOffer(); + ASSERT_THAT( + WaitUntil([&] { return SignalingStateStable(); }, ::testing::IsTrue()), + IsRtcOk()); + caller()->SetGeneratedSdpMunger(nullptr); + + if (!has_munged) { + GTEST_SKIP() << "SDP munging did not replace codec, skipping."; + } + + // Note currently negotiated codecs and count VP8 and VP9. + auto codecs_after_munge = caller() + ->pc() + ->local_description() + ->description() + ->contents()[0] + .media_description() + ->codecs(); + size_t vp8_munge = 0; + size_t vp9_munge = 0; + for (const auto& codec : codecs_after_munge) { + if (codec.name == "VP8") + vp8_munge++; + else if (codec.name == "VP9") + vp9_munge++; + } + // We should have replaced VP8 with VP9. + EXPECT_EQ(vp8_munge, 0u); + EXPECT_GE(vp9_munge, 2u); + + // Call setCodecPreferences. + std::vector codecs = + caller() + ->pc_factory() + ->GetRtpReceiverCapabilities(webrtc::MediaType::VIDEO) + .codecs; + auto transceivers = caller()->pc()->GetTransceivers(); + ASSERT_EQ(transceivers.size(), 1u); + transceivers[0]->SetCodecPreferences(codecs); + + auto offer = caller()->CreateOfferAndWait(); + ASSERT_NE(offer, nullptr); + // The offer should be acceptable. + EXPECT_TRUE(caller()->SetLocalDescriptionAndSendSdpMessage(std::move(offer))); + + auto codecs_after_scp = caller() + ->pc() + ->local_description() + ->description() + ->contents()[0] + .media_description() + ->codecs(); + size_t vp8_scp = 0; + size_t vp9_scp = 0; + for (const auto& codec : codecs_after_scp) { + if (codec.name == "VP8") + vp8_scp++; + else if (codec.name == "VP9") + vp9_scp++; + } + // The SDP munging modification was reverted by sCP. + // This is a footgun but please do not mix such munging with + // setCodecPreferences, munging is on the way out. + EXPECT_EQ(vp8_scp, 1u); + EXPECT_GE(vp9_scp, 1u); + EXPECT_GT(vp9_munge, vp9_scp); +} + TEST_F(PeerConnectionIntegrationTestUnifiedPlan, SensibleRtxWithDuplicateCodecs) { ASSERT_TRUE(CreatePeerConnectionWrappers()); @@ -4823,7 +4925,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan, // associated RTX codec. std::unique_ptr answer = caller()->CreateAnswerForTest(); - ASSERT_THAT(answer, NotNull()); + ASSERT_NE(answer, nullptr); RTC_LOG(LS_ERROR) << "Answer is " << *answer; ASSERT_THAT(answer->description()->contents().size(), Eq(1)); auto codecs = @@ -4882,7 +4984,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan, caller()->AddVideoTrack(); auto offer2 = caller()->CreateOfferAndWait(); // Observe that packetization is raw on BOTH media sections. - ASSERT_THAT(offer2, NotNull()); + ASSERT_NE(offer2, nullptr); EXPECT_EQ(offer2->description()->contents().size(), 2U); for (const auto& content : offer2->description()->contents()) { for (const auto& codec : content.media_description()->codecs()) { From 1e71d089aa63e0f82451d1983863a06110e694a7 Mon Sep 17 00:00:00 2001 From: Tom Van Goethem Date: Mon, 12 May 2025 15:45:26 +0000 Subject: [PATCH 126/292] Fix metrics for outcome of SDP munging The enum of SdpMungingOutcome become out of sync with the enum defined in the metric. This CL fixes that. Bug: chromium:40567530 Change-Id: Ibcbd5de1001e52ac64aea801071427e96e02e796 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390980 Commit-Queue: Philipp Hancke Auto-Submit: Tom Van Goethem Reviewed-by: Harald Alvestrand Reviewed-by: Philipp Hancke Cr-Commit-Position: refs/heads/main@{#44598} --- api/uma_metrics.h | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/api/uma_metrics.h b/api/uma_metrics.h index e6f722ca43..93c8c8d0ca 100644 --- a/api/uma_metrics.h +++ b/api/uma_metrics.h @@ -231,9 +231,8 @@ enum SdpMungingType { // and if the should be accepted or rejected. Keep in sync with // SdpMungingOutcome from tools/metrics/histograms/metadata/web_rtc/enums.xml enum class SdpMungingOutcome { - kNoMunge = 0, - kAccepted = 1, - kRejected = 2, + kAccepted = 0, + kRejected = 1, kMaxValue = kRejected, }; From 4eaf80f562748a26dce91bcfa7d6cb17985bf47f Mon Sep 17 00:00:00 2001 From: Sunggook Chue Date: Fri, 9 May 2025 16:07:46 -0700 Subject: [PATCH 127/292] Desktop Capture Performance Improvement via HMONITOR Caching MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The GetHmonitorFromDeviceIndex API previously made repeated calls to the EnumDisplayDevicesW OS API, which is an expensive kernel operation. Specifically, EnumDisplayDevicesW was invoked twice—once in IsScreenValid and again in GetScreenRect. GetHmonitorFromDeviceIndex alone cost is 1000 counts meanwhile its parent ProcessFrame() itself is around 4000 counts, which includes memcpy (2000 counts) and other d3d11 operation (1000). This change removes the redundant call to IsScreenValid, no calling the expensive EnumDisplayDevicesW, as GetScreenRect already performs the necessary validation. Additionally, the HMONITOR value is now cached for each WgcCaptureSession used in screen capture. After this CL, GetHmonitorFromDeviceIndex cost is almost 0 as it isn't shown on the traces anymore. Bug: chromium:416088620 Change-Id: I1a9076ac9c797c3cb12f420dc60c73f174fe4f9f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390180 Reviewed-by: Ilya Nikolaevskiy Reviewed-by: Alexander Cooper Commit-Queue: Sunggook Chue Cr-Commit-Position: refs/heads/main@{#44599} --- .../win/screen_capture_utils.cc | 11 +++------- .../win/screen_capture_utils.h | 14 ++++++++----- .../win/screen_capturer_win_gdi.cc | 7 +++++-- .../win/screen_capturer_win_gdi.h | 2 +- .../win/screen_capturer_win_magnifier.cc | 4 +++- .../win/screen_capturer_win_magnifier.h | 2 +- .../win/wgc_capture_session.cc | 21 ++++++++++--------- .../desktop_capture/win/wgc_capture_session.h | 5 +++++ 8 files changed, 38 insertions(+), 28 deletions(-) diff --git a/modules/desktop_capture/win/screen_capture_utils.cc b/modules/desktop_capture/win/screen_capture_utils.cc index 30120db3f4..8ac8bfcd0c 100644 --- a/modules/desktop_capture/win/screen_capture_utils.cc +++ b/modules/desktop_capture/win/screen_capture_utils.cc @@ -69,12 +69,7 @@ bool GetHmonitorFromDeviceIndex(const DesktopCapturer::SourceId device_index, return true; } - std::wstring device_key; - if (!IsScreenValid(device_index, &device_key)) { - return false; - } - - DesktopRect screen_rect = GetScreenRect(device_index, device_key); + DesktopRect screen_rect = GetScreenRect(device_index, std::nullopt); if (screen_rect.is_empty()) { return false; } @@ -169,7 +164,7 @@ DesktopVector GetDpiForMonitor(HMONITOR monitor) { } DesktopRect GetScreenRect(const DesktopCapturer::SourceId screen, - const std::wstring& device_key) { + const std::optional& device_key) { if (screen == kFullDesktopScreenId) { return GetFullscreenRect(); } @@ -185,7 +180,7 @@ DesktopRect GetScreenRect(const DesktopCapturer::SourceId screen, // sure we are capturing the same device when devices are added or removed. // DeviceKey is documented as reserved, but it actually contains the registry // key for the device and is unique for each monitor, while DeviceID is not. - if (device_key != device.DeviceKey) { + if (device_key.has_value() && *device_key != device.DeviceKey) { return DesktopRect(); } diff --git a/modules/desktop_capture/win/screen_capture_utils.h b/modules/desktop_capture/win/screen_capture_utils.h index 71c79b9ab3..68407f5fda 100644 --- a/modules/desktop_capture/win/screen_capture_utils.h +++ b/modules/desktop_capture/win/screen_capture_utils.h @@ -61,7 +61,9 @@ DesktopVector GetDpiForMonitor(HMONITOR monitor); // Returns true if `screen` is a valid screen. The screen device key is // returned through `device_key` if the screen is valid. The device key can be // used in GetScreenRect to verify the screen matches the previously obtained -// id. +// id. It calls the EnumDisplayDevices API to check if the screen is valid but +// EnumDisplayDevices is quite slow so the caller of this function should +// be aware of the performance impact. bool IsScreenValid(DesktopCapturer::SourceId screen, std::wstring* device_key); // Get the rect of the entire system in system coordinate system. I.e. the @@ -69,10 +71,12 @@ bool IsScreenValid(DesktopCapturer::SourceId screen, std::wstring* device_key); DesktopRect GetFullscreenRect(); // Get the rect of the screen identified by `screen`, relative to the primary -// display's top-left. If the screen device key does not match `device_key`, or -// the screen does not exist, or any error happens, an empty rect is returned. -RTC_EXPORT DesktopRect GetScreenRect(DesktopCapturer::SourceId screen, - const std::wstring& device_key); +// display's top-left. If the optional screen device key exists, and does not +// match `device_key`, or the screen does not exist, or any error happens, +// an empty rect is returned. +RTC_EXPORT DesktopRect +GetScreenRect(DesktopCapturer::SourceId screen, + const std::optional& device_key); } // namespace webrtc diff --git a/modules/desktop_capture/win/screen_capturer_win_gdi.cc b/modules/desktop_capture/win/screen_capturer_win_gdi.cc index 24716813b5..ae6fb1da5e 100644 --- a/modules/desktop_capture/win/screen_capturer_win_gdi.cc +++ b/modules/desktop_capture/win/screen_capturer_win_gdi.cc @@ -112,9 +112,12 @@ bool ScreenCapturerWinGdi::GetSourceList(SourceList* sources) { } bool ScreenCapturerWinGdi::SelectSource(SourceId id) { - bool valid = IsScreenValid(id, ¤t_device_key_); - if (valid) + std::wstring device_key; + bool valid = IsScreenValid(id, &device_key); + if (valid) { current_screen_id_ = id; + current_device_key_ = device_key; + } return valid; } diff --git a/modules/desktop_capture/win/screen_capturer_win_gdi.h b/modules/desktop_capture/win/screen_capturer_win_gdi.h index 7c3977ed42..6e103aea4d 100644 --- a/modules/desktop_capture/win/screen_capturer_win_gdi.h +++ b/modules/desktop_capture/win/screen_capturer_win_gdi.h @@ -61,7 +61,7 @@ class ScreenCapturerWinGdi : public DesktopCapturer { Callback* callback_ = nullptr; std::unique_ptr shared_memory_factory_; SourceId current_screen_id_ = kFullDesktopScreenId; - std::wstring current_device_key_; + std::optional current_device_key_; ScopedThreadDesktop desktop_; diff --git a/modules/desktop_capture/win/screen_capturer_win_magnifier.cc b/modules/desktop_capture/win/screen_capturer_win_magnifier.cc index 9a3c9d5f53..6d2384eb00 100644 --- a/modules/desktop_capture/win/screen_capturer_win_magnifier.cc +++ b/modules/desktop_capture/win/screen_capturer_win_magnifier.cc @@ -135,7 +135,9 @@ bool ScreenCapturerWinMagnifier::GetSourceList(SourceList* sources) { } bool ScreenCapturerWinMagnifier::SelectSource(SourceId id) { - if (IsScreenValid(id, ¤t_device_key_)) { + std::wstring device_key; + if (IsScreenValid(id, &device_key)) { + current_device_key_ = device_key; current_screen_id_ = id; return true; } diff --git a/modules/desktop_capture/win/screen_capturer_win_magnifier.h b/modules/desktop_capture/win/screen_capturer_win_magnifier.h index 07c5b1e9e6..80cf6d410b 100644 --- a/modules/desktop_capture/win/screen_capturer_win_magnifier.h +++ b/modules/desktop_capture/win/screen_capturer_win_magnifier.h @@ -104,7 +104,7 @@ class ScreenCapturerWinMagnifier : public DesktopCapturer { Callback* callback_ = nullptr; std::unique_ptr shared_memory_factory_; ScreenId current_screen_id_ = kFullDesktopScreenId; - std::wstring current_device_key_; + std::optional current_device_key_; HWND excluded_window_ = NULL; // Queue of the frames buffers. diff --git a/modules/desktop_capture/win/wgc_capture_session.cc b/modules/desktop_capture/win/wgc_capture_session.cc index 100f28a0df..63a6a15229 100644 --- a/modules/desktop_capture/win/wgc_capture_session.cc +++ b/modules/desktop_capture/win/wgc_capture_session.cc @@ -13,8 +13,6 @@ #include #include #include -#include -#include #include #include @@ -469,18 +467,21 @@ HRESULT WgcCaptureSession::ProcessFrame() { DesktopFrame* current_frame = queue_.current_frame(); DesktopFrame* previous_frame = queue_.previous_frame(); - HMONITOR monitor; if (is_window_source_) { // If the captured window moves to another screen, the HMONITOR associated // with the captured window will change. Therefore, we need to get the value // of HMONITOR per frame. - monitor = MonitorFromWindow(reinterpret_cast(source_id_), - /*dwFlags=*/MONITOR_DEFAULTTONEAREST); + monitor_ = ::MonitorFromWindow(reinterpret_cast(source_id_), + /*dwFlags=*/MONITOR_DEFAULTTONEAREST); } else { - if (!GetHmonitorFromDeviceIndex(source_id_, &monitor)) { - RTC_LOG(LS_ERROR) << "Failed to get HMONITOR from device index."; - d3d_context->Unmap(mapped_texture_.Get(), 0); - return E_FAIL; + if (!monitor_.has_value()) { + HMONITOR monitor; + if (!GetHmonitorFromDeviceIndex(source_id_, &monitor)) { + RTC_LOG(LS_ERROR) << "Failed to get HMONITOR from device index."; + d3d_context->Unmap(mapped_texture_.Get(), 0); + return E_FAIL; + } + monitor_ = monitor; } } @@ -490,7 +491,7 @@ HRESULT WgcCaptureSession::ProcessFrame() { // 1, 1.5, 2.5, etc. DEVICE_SCALE_FACTOR device_scale_factor = DEVICE_SCALE_FACTOR_INVALID; HRESULT scale_factor_hr = - GetScaleFactorForMonitor(monitor, &device_scale_factor); + GetScaleFactorForMonitor(monitor_.value(), &device_scale_factor); RTC_LOG_IF(LS_ERROR, FAILED(scale_factor_hr)) << "Failed to get scale factor for monitor: " << scale_factor_hr; if (device_scale_factor != DEVICE_SCALE_FACTOR_INVALID) { diff --git a/modules/desktop_capture/win/wgc_capture_session.h b/modules/desktop_capture/win/wgc_capture_session.h index 5d3f056945..ca88b061a1 100644 --- a/modules/desktop_capture/win/wgc_capture_session.h +++ b/modules/desktop_capture/win/wgc_capture_session.h @@ -155,6 +155,11 @@ class WgcCaptureSession final { // The unique id to represent a Source of current DesktopCapturer. intptr_t source_id_; + // The monitor that is being captured when the target source_id is a + // screen. For window sources, it can't be used because the window can move + // around around the different monitors. + std::optional monitor_; + // The source type of the capture session. It can be either a window or a // screen. bool is_window_source_; From e732a85ce27ecead1b5ba47febfac4f86ba01306 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Mon, 12 May 2025 13:02:09 -0700 Subject: [PATCH 128/292] Roll chromium_revision fed22da4c6..b286fc6108 (1458742:1459047) Change log: https://chromium.googlesource.com/chromium/src/+log/fed22da4c6..b286fc6108 Full diff: https://chromium.googlesource.com/chromium/src/+/fed22da4c6..b286fc6108 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/994477767d..6a00af3991 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/b8a12c0da5..a395577776 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/c41670cc43..2683c99fea * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/ab12b4b4c9..6c46adee24 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/f00a25ff94..ca5dcf4ab6 * src/third_party/androidx/cipd: wfVeehXfOqVwC9E4j9U8ghXvt7eeTMgZsdBUBB5fVRkC..PAuqhSDBsIGL9f8sVxtknszgmnEYv3cgSAU7aUM1AjMC * src/third_party/breakpad/breakpad: https://chromium.googlesource.com/breakpad/breakpad.git/+log/b319c12317..9eeece27b7 * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/0c5acc073d..77017288fa * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/977c37458f..7d18f85450 * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/2f8af1b7f2..9a481d3d37 * src/third_party/libyuv: https://chromium.googlesource.com/libyuv/libyuv.git/+log/4db2af62da..61bdaee13a * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/11cd406514..bff34086ee * src/third_party/turbine/cipd: scfGptWnO9bwzbg-jr0mcnVO3NG5KQJvlAQd_JSD5QUC..PbV7UFdzFIl6b_4lNwsj4VnlvnoULNAZRDwsndTGXTsC * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/718569c346..5abd2ec618 * src/tools/luci-go: git_revision:4486e4bb411a38fa21ba426d9a89052402fe610f..git_revision:6704cef9341f59c9c6b3b996345b28e737bbb69d * src/tools/luci-go: git_revision:4486e4bb411a38fa21ba426d9a89052402fe610f..git_revision:6704cef9341f59c9c6b3b996345b28e737bbb69d * src/tools/luci-go: git_revision:4486e4bb411a38fa21ba426d9a89052402fe610f..git_revision:6704cef9341f59c9c6b3b996345b28e737bbb69d DEPS diff: https://chromium.googlesource.com/chromium/src/+/fed22da4c6..b286fc6108/DEPS No update to Clang. BUG=None Change-Id: I2185e8bd82929becdc4d0ad4492058e0c7ec5a3b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391080 Commit-Queue: Autoroller Bot-Commit: Autoroller Cr-Commit-Position: refs/heads/main@{#44600} --- DEPS | 36 ++++++++++++++++++------------------ 1 file changed, 18 insertions(+), 18 deletions(-) diff --git a/DEPS b/DEPS index 9e76b0734c..13376ac1ad 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': 'fed22da4c6801e618c8e0948bb276c87f643cdce', + 'chromium_revision': 'b286fc6108fe8c811662a3ab54976879eb049896', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@994477767d99fcf1e3b34a25ae523c7eb7fd608a', + 'https://chromium.googlesource.com/chromium/src/base@6a00af39918e5bc9cd3f30757727956f414da8fa', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@b8a12c0da5aabb81c840bad1fa5e8a4889a25f23', + 'https://chromium.googlesource.com/chromium/src/build@a39557777609c2a95f80f278cda71ba88cd8f8e5', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@077a66f30fcf281b066fafb6dfc60818c238efb6', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@c41670cc43e5f3b0736263453f95fdd485ed85a3', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@2683c99feaa8692aa0e236655bb0472e3b33c7ba', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@ab12b4b4c9bf4ea505ef5d0f4ed63dbf10cead00', + 'https://chromium.googlesource.com/chromium/src/testing@6c46adee24de6c3d759caa969be3e2ee5bc2e4cf', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@f00a25ff945b6995bfb53d62ca7fd1787798ed7d', + 'https://chromium.googlesource.com/chromium/src/third_party@ca5dcf4ab61398bda71fce44fe748a29c010cb9d', 'src/buildtools/linux64': { 'packages': [ @@ -482,9 +482,9 @@ deps = { 'src/third_party/boringssl/src': 'https://boringssl.googlesource.com/boringssl.git@136284f8548bc7fb43e99e7f69e03fab57168e8b', 'src/third_party/breakpad/breakpad': - 'https://chromium.googlesource.com/breakpad/breakpad.git@b319c12317824bfd0e92828db20bbfe0ec6e31f8', + 'https://chromium.googlesource.com/breakpad/breakpad.git@9eeece27b727d2ffac4b7042bb69d7c140e58312', 'src/third_party/catapult': - 'https://chromium.googlesource.com/catapult.git@0c5acc073dd14893f2a75a51819ad81627098551', + 'https://chromium.googlesource.com/catapult.git@77017288fa6b829b2d31aa8195799ea78e594128', 'src/third_party/ced/src': { 'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5', }, @@ -497,13 +497,13 @@ deps = { 'src/third_party/crc32c/src': 'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558', 'src/third_party/depot_tools': - 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@977c37458fda472d8822a8b57e4a83a7bc747471', + 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@7d18f854503a26cb29540012327ad3f78926de96', 'src/third_party/ffmpeg': 'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@dcdd0fa51b65a0b1688ff6b8f0cc81908f09ded2', 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@2f8af1b7f2ac17793e01bec1b2654ca12ed21966', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@9a481d3d37829273029b1173814e3e83c8a1e2aa', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -597,13 +597,13 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '11cd406514a8212d3568448b2207d2925d5ca202', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'bff34086ee3abf81659b5184c243f4cf2f799992', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': 'https://chromium.googlesource.com/webm/libvpx.git@6bb288633613c613c87d23b32e5f1d23102b1a43', 'src/third_party/libyuv': - 'https://chromium.googlesource.com/libyuv/libyuv.git@4db2af62dab48895226be6b52737247e898ebe36', + 'https://chromium.googlesource.com/libyuv/libyuv.git@61bdaee13a701d2b52c6dc943ccc5c888077a591', 'src/third_party/lss': { 'url': 'https://chromium.googlesource.com/linux-syscall-support.git@ed31caa60f20a4f6569883b2d752ef7522de51e0', 'condition': 'checkout_android or checkout_linux', @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@718569c3464aef1f18235ef679510fa0d12df4ac', + 'https://chromium.googlesource.com/chromium/src/tools@5abd2ec6187f43f7b7d634a27949cc1c1dda722e', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'wfVeehXfOqVwC9E4j9U8ghXvt7eeTMgZsdBUBB5fVRkC', + 'version': 'PAuqhSDBsIGL9f8sVxtknszgmnEYv3cgSAU7aUM1AjMC', }, ], 'condition': 'checkout_android and non_git_source', @@ -777,7 +777,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/turbine', - 'version': 'scfGptWnO9bwzbg-jr0mcnVO3NG5KQJvlAQd_JSD5QUC', + 'version': 'PbV7UFdzFIl6b_4lNwsj4VnlvnoULNAZRDwsndTGXTsC', }, ], 'condition': 'checkout_android', @@ -793,15 +793,15 @@ deps = { 'packages': [ { 'package': 'infra/tools/luci/cas/${{platform}}', - 'version': 'git_revision:4486e4bb411a38fa21ba426d9a89052402fe610f', + 'version': 'git_revision:6704cef9341f59c9c6b3b996345b28e737bbb69d', }, { 'package': 'infra/tools/luci/isolate/${{platform}}', - 'version': 'git_revision:4486e4bb411a38fa21ba426d9a89052402fe610f', + 'version': 'git_revision:6704cef9341f59c9c6b3b996345b28e737bbb69d', }, { 'package': 'infra/tools/luci/swarming/${{platform}}', - 'version': 'git_revision:4486e4bb411a38fa21ba426d9a89052402fe610f', + 'version': 'git_revision:6704cef9341f59c9c6b3b996345b28e737bbb69d', } ], 'dep_type': 'cipd', From d605e1d37da6f9e7d20dd2715027c16be590e5cd Mon Sep 17 00:00:00 2001 From: webrtc-version-updater Date: Mon, 12 May 2025 21:03:44 -0700 Subject: [PATCH 129/292] Update WebRTC code version (2025-05-13T04:03:41). Bug: None Change-Id: I9d392954dd4bece8973136d6c0626ec7859e91db Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391101 Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Cr-Commit-Position: refs/heads/main@{#44601} --- call/version.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/call/version.cc b/call/version.cc index 156e478d32..64ada0b45e 100644 --- a/call/version.cc +++ b/call/version.cc @@ -13,7 +13,7 @@ namespace webrtc { // The timestamp is always in UTC. -const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-10T04:08:17"; +const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-13T04:03:41"; void LoadWebRTCVersionInRegister() { // Using volatile to instruct the compiler to not optimize `p` away even From dafde4b473e8625209a25705ece2dfd55863be7a Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Mon, 12 May 2025 23:02:41 -0700 Subject: [PATCH 130/292] Roll chromium_revision b286fc6108..80cb2d193d (1459047:1459248) Change log: https://chromium.googlesource.com/chromium/src/+log/b286fc6108..80cb2d193d Full diff: https://chromium.googlesource.com/chromium/src/+/b286fc6108..80cb2d193d Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/6a00af3991..70d4d5cf23 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/a395577776..a7f3c053b0 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/2683c99fea..83a5db416b * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/6c46adee24..3d355ecb9a * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/ca5dcf4ab6..70731bd9b3 * src/third_party/androidx/cipd: PAuqhSDBsIGL9f8sVxtknszgmnEYv3cgSAU7aUM1AjMC..77JFcazE3nHDyZZ4RYND-IEyes3SQTICM2Kz231QD60C * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/136284f854..2a514a51ba * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/77017288fa..83c00a37f4 * src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/948a46574c..b5f72d2ab6 * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/7d18f85450..14bfda1708 * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/9a481d3d37..7b1679b14d * src/third_party/libc++abi/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git/+log/1efb5e6d7c..5a49db9990 * src/third_party/libunwind/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git/+log/09f6f7b168..2bd5f3cae1 * src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/4a2940b40b..a0ab545a0e * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/bff34086ee..4402fd7953 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/5abd2ec618..b40180e2fc * src/tools/luci-go: git_revision:6704cef9341f59c9c6b3b996345b28e737bbb69d..git_revision:45d1c0a0168f06a2bbde9eca9a03087ed1da523e * src/tools/luci-go: git_revision:6704cef9341f59c9c6b3b996345b28e737bbb69d..git_revision:45d1c0a0168f06a2bbde9eca9a03087ed1da523e * src/tools/luci-go: git_revision:6704cef9341f59c9c6b3b996345b28e737bbb69d..git_revision:45d1c0a0168f06a2bbde9eca9a03087ed1da523e DEPS diff: https://chromium.googlesource.com/chromium/src/+/b286fc6108..80cb2d193d/DEPS No update to Clang. BUG=None Change-Id: If0c7e12ff69a3bf603255a0dfb98f767dabe8dea Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391103 Commit-Queue: Autoroller Bot-Commit: Autoroller Cr-Commit-Position: refs/heads/main@{#44602} --- DEPS | 40 ++++++++++++++++++++-------------------- 1 file changed, 20 insertions(+), 20 deletions(-) diff --git a/DEPS b/DEPS index 13376ac1ad..e037952ab5 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': 'b286fc6108fe8c811662a3ab54976879eb049896', + 'chromium_revision': '80cb2d193db7515d8f2a457cbdac343bf0167588', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@6a00af39918e5bc9cd3f30757727956f414da8fa', + 'https://chromium.googlesource.com/chromium/src/base@70d4d5cf23c155e8a9cb4c5c8b4e751e8763707e', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@a39557777609c2a95f80f278cda71ba88cd8f8e5', + 'https://chromium.googlesource.com/chromium/src/build@a7f3c053b0f19d21089b7ceb3e62f525c0d02171', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@077a66f30fcf281b066fafb6dfc60818c238efb6', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@2683c99feaa8692aa0e236655bb0472e3b33c7ba', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@83a5db416b63068413e15dab161f92b679b5f5a5', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@6c46adee24de6c3d759caa969be3e2ee5bc2e4cf', + 'https://chromium.googlesource.com/chromium/src/testing@3d355ecb9aee09b8b4cfe6ee0ce63b98412965ec', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@ca5dcf4ab61398bda71fce44fe748a29c010cb9d', + 'https://chromium.googlesource.com/chromium/src/third_party@70731bd9b3ace2d70fc063b27182decf1ca272b8', 'src/buildtools/linux64': { 'packages': [ @@ -323,15 +323,15 @@ deps = { 'src/third_party/clang-format/script': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@37f6e68a107df43b7d7e044fd36a13cbae3413f2', 'src/third_party/compiler-rt/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@948a46574ce380c306ae4f01d2a630d93847b88f', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@b5f72d2ab6190b6cea33512880e6cbfe3648d513', 'src/third_party/libc++/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@a01c02c9d4acbdae3b7e8a2f3ee58579a9c29f96', 'src/third_party/libc++abi/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@1efb5e6d7c5eee01624f3730a935285405c9cd22', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@5a49db9990ee2ecee2bc7340be9756c0455bde6f', 'src/third_party/llvm-libc/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@4a2940b40b394ca57312aa9bbc8af430fe9a5340', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@a0ab545a0eaa7a8cd7154025d1902904e1800cfc', 'src/third_party/libunwind/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@09f6f7b1685888b16c0fe52a7a68f9dff8b8d60e', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@2bd5f3cae13e8ad6727d0a77a2ec9cdc6b06becb', 'src/third_party/test_fonts/test_fonts': { 'dep_type': 'gcs', @@ -480,11 +480,11 @@ deps = { }, 'src/third_party/boringssl/src': - 'https://boringssl.googlesource.com/boringssl.git@136284f8548bc7fb43e99e7f69e03fab57168e8b', + 'https://boringssl.googlesource.com/boringssl.git@2a514a51baebd5a232fc64f7b082f7a8b28cd29d', 'src/third_party/breakpad/breakpad': 'https://chromium.googlesource.com/breakpad/breakpad.git@9eeece27b727d2ffac4b7042bb69d7c140e58312', 'src/third_party/catapult': - 'https://chromium.googlesource.com/catapult.git@77017288fa6b829b2d31aa8195799ea78e594128', + 'https://chromium.googlesource.com/catapult.git@83c00a37f40a93932204f684172a13c62a379e7e', 'src/third_party/ced/src': { 'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5', }, @@ -497,13 +497,13 @@ deps = { 'src/third_party/crc32c/src': 'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558', 'src/third_party/depot_tools': - 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@7d18f854503a26cb29540012327ad3f78926de96', + 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@14bfda17088cb03d7bc0ba7df6cac2699e051e14', 'src/third_party/ffmpeg': 'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@dcdd0fa51b65a0b1688ff6b8f0cc81908f09ded2', 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@9a481d3d37829273029b1173814e3e83c8a1e2aa', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@7b1679b14d0292b5373bb270cf8c6390ae93dd9f', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'bff34086ee3abf81659b5184c243f4cf2f799992', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '4402fd7953d5dedf65659e99ab80404f6e94a004', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@5abd2ec6187f43f7b7d634a27949cc1c1dda722e', + 'https://chromium.googlesource.com/chromium/src/tools@b40180e2fcc6ef2334b3d0c463509d968de96ac8', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'PAuqhSDBsIGL9f8sVxtknszgmnEYv3cgSAU7aUM1AjMC', + 'version': '77JFcazE3nHDyZZ4RYND-IEyes3SQTICM2Kz231QD60C', }, ], 'condition': 'checkout_android and non_git_source', @@ -793,15 +793,15 @@ deps = { 'packages': [ { 'package': 'infra/tools/luci/cas/${{platform}}', - 'version': 'git_revision:6704cef9341f59c9c6b3b996345b28e737bbb69d', + 'version': 'git_revision:45d1c0a0168f06a2bbde9eca9a03087ed1da523e', }, { 'package': 'infra/tools/luci/isolate/${{platform}}', - 'version': 'git_revision:6704cef9341f59c9c6b3b996345b28e737bbb69d', + 'version': 'git_revision:45d1c0a0168f06a2bbde9eca9a03087ed1da523e', }, { 'package': 'infra/tools/luci/swarming/${{platform}}', - 'version': 'git_revision:6704cef9341f59c9c6b3b996345b28e737bbb69d', + 'version': 'git_revision:45d1c0a0168f06a2bbde9eca9a03087ed1da523e', } ], 'dep_type': 'cipd', From 887a6b9f7fb88291413ccc454651cb5e571740b2 Mon Sep 17 00:00:00 2001 From: Daniel White Date: Tue, 6 May 2025 14:00:00 -0400 Subject: [PATCH 131/292] [iOS] Update RTCCameraPreviewView to conform to iOS 17 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit In iOS 17, the video orientation API is deprecated. This CL updates the RTCCameraPreviewView file to use the video rotation angle API instead. This file is being updated now to support Chrome for iOS effort to raise its minimum deployment target to iOS 17. Bug: chromium:348649285 Change-Id: Icf72911e9a3431a9a91fefc3d85f9ea02597c5f4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390080 Reviewed-by: Mirko Bonadei Commit-Queue: Mirko Bonadei Reviewed-by: Kári Helgason Cr-Commit-Position: refs/heads/main@{#44603} --- sdk/objc/helpers/RTCCameraPreviewView.m | 70 +++++++++++++++++-------- 1 file changed, 49 insertions(+), 21 deletions(-) diff --git a/sdk/objc/helpers/RTCCameraPreviewView.m b/sdk/objc/helpers/RTCCameraPreviewView.m index 92c3cd2635..77d5e8823d 100644 --- a/sdk/objc/helpers/RTCCameraPreviewView.m +++ b/sdk/objc/helpers/RTCCameraPreviewView.m @@ -80,6 +80,54 @@ - (void)orientationChanged:(NSNotification *)notification { } - (void)setCorrectVideoOrientation { + if (@available(iOS 17, *)) { + [self modifyVideoAngle]; + return; + } + + [self modifyVideoOrientation]; +} + +#pragma mark - Private + +- (void)addOrientationObserver { + [[NSNotificationCenter defaultCenter] + addObserver:self + selector:@selector(orientationChanged:) + name:UIDeviceOrientationDidChangeNotification + object:nil]; +} + +- (void)removeOrientationObserver { + [[NSNotificationCenter defaultCenter] + removeObserver:self + name:UIDeviceOrientationDidChangeNotification + object:nil]; +} + +- (AVCaptureVideoPreviewLayer *)previewLayer { + return (AVCaptureVideoPreviewLayer *)self.layer; +} + +- (void)modifyVideoAngle API_AVAILABLE(ios(17.0)) { + AVCaptureDeviceInput* captureSessionInput = + _captureSession.inputs.firstObject; + AVCaptureDevice* camera = captureSessionInput.device; + AVCaptureVideoPreviewLayer* previewLayer = [self previewLayer]; + AVCaptureConnection* videoConnection = previewLayer.connection; + AVCaptureDeviceRotationCoordinator* rotationCoordiantor = + [[AVCaptureDeviceRotationCoordinator alloc] + initWithDevice:camera + previewLayer:previewLayer]; + CGFloat angle = + rotationCoordiantor.videoRotationAngleForHorizonLevelCapture; + if ([videoConnection isVideoRotationAngleSupported:angle]) { + [videoConnection setVideoRotationAngle:angle]; + } +} + +#if defined(__IPHONE_17_0) && __IPHONE_OS_VERSION_MAX_ALLOWED >= __IPHONE_17_0 +- (void)modifyVideoOrientation { // Get current device orientation. UIDeviceOrientation deviceOrientation = [UIDevice currentDevice].orientation; AVCaptureVideoPreviewLayer *previewLayer = [self previewLayer]; @@ -104,26 +152,6 @@ - (void)setCorrectVideoOrientation { // orientation. } } - -#pragma mark - Private - -- (void)addOrientationObserver { - [[NSNotificationCenter defaultCenter] - addObserver:self - selector:@selector(orientationChanged:) - name:UIDeviceOrientationDidChangeNotification - object:nil]; -} - -- (void)removeOrientationObserver { - [[NSNotificationCenter defaultCenter] - removeObserver:self - name:UIDeviceOrientationDidChangeNotification - object:nil]; -} - -- (AVCaptureVideoPreviewLayer *)previewLayer { - return (AVCaptureVideoPreviewLayer *)self.layer; -} +#endif @end From 0f6ee2821f60475fc4414e7e0432dd5a61e3d272 Mon Sep 17 00:00:00 2001 From: Harald Alvestrand Date: Tue, 13 May 2025 07:47:10 +0000 Subject: [PATCH 132/292] Remove unused "recv" variables in VideoSendChannel These were leftovers from the send/recv split. Bug: None Change-Id: Ia2efccc7a20c438e85aececbf5ed4f7d427392ae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390402 Commit-Queue: Harald Alvestrand Reviewed-by: Evan Shrubsole Cr-Commit-Position: refs/heads/main@{#44604} --- media/engine/webrtc_video_engine.cc | 12 +----------- media/engine/webrtc_video_engine.h | 16 ---------------- 2 files changed, 1 insertion(+), 27 deletions(-) diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc index 6f1d2f51ef..ea3c6a7c34 100644 --- a/media/engine/webrtc_video_engine.cc +++ b/media/engine/webrtc_video_engine.cc @@ -88,7 +88,6 @@ #include "modules/rtp_rtcp/include/rtp_header_extension_map.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" -#include "modules/rtp_rtcp/source/rtp_util.h" #include "modules/video_coding/svc/scalability_mode_util.h" #include "rtc_base/checks.h" #include "rtc_base/dscp.h" @@ -843,7 +842,7 @@ WebRtcVideoEngine::CreateSendChannel( VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) { return std::make_unique( call, config, options, crypto_options, encoder_factory_.get(), - decoder_factory_.get(), video_bitrate_allocator_factory); + video_bitrate_allocator_factory); } std::unique_ptr WebRtcVideoEngine::CreateReceiveChannel(Call* call, @@ -925,7 +924,6 @@ WebRtcVideoSendChannel::WebRtcVideoSendChannel( const VideoOptions& options, const CryptoOptions& crypto_options, VideoEncoderFactory* encoder_factory, - VideoDecoderFactory* decoder_factory, VideoBitrateAllocatorFactory* bitrate_allocator_factory) : MediaChannelUtil(call->network_thread(), config.enable_dscp), worker_thread_(call->worker_thread()), @@ -935,7 +933,6 @@ WebRtcVideoSendChannel::WebRtcVideoSendChannel( default_sink_(nullptr), video_config_(config.video), encoder_factory_(encoder_factory), - decoder_factory_(decoder_factory), bitrate_allocator_factory_(bitrate_allocator_factory), default_send_options_(options), last_send_stats_log_ms_(-1), @@ -946,13 +943,6 @@ WebRtcVideoSendChannel::WebRtcVideoSendChannel( crypto_options_(crypto_options) { RTC_DCHECK_RUN_ON(&thread_checker_); rtcp_receiver_report_ssrc_ = kDefaultRtcpReceiverReportSsrc; - // Crash if MapCodecs fails. - recv_codecs_ = MapCodecs(GetPayloadTypesAndDefaultCodecs( - decoder_factory_, /*is_decoder_factory=*/true, - /*include_rtx=*/true, call_->trials())) - .value(); - recv_flexfec_payload_type_ = - recv_codecs_.empty() ? 0 : recv_codecs_.front().flexfec_payload_type; } WebRtcVideoSendChannel::~WebRtcVideoSendChannel() { diff --git a/media/engine/webrtc_video_engine.h b/media/engine/webrtc_video_engine.h index 5d6bd6a847..e8cfa264bc 100644 --- a/media/engine/webrtc_video_engine.h +++ b/media/engine/webrtc_video_engine.h @@ -25,7 +25,6 @@ #include "absl/functional/any_invocable.h" #include "absl/strings/string_view.h" -#include "api/array_view.h" #include "api/crypto/crypto_options.h" #include "api/crypto/frame_decryptor_interface.h" #include "api/crypto/frame_encryptor_interface.h" @@ -164,7 +163,6 @@ class WebRtcVideoSendChannel : public MediaChannelUtil, const VideoOptions& options, const CryptoOptions& crypto_options, VideoEncoderFactory* encoder_factory, - VideoDecoderFactory* decoder_factory, VideoBitrateAllocatorFactory* bitrate_allocator_factory); ~WebRtcVideoSendChannel() override; @@ -308,15 +306,6 @@ class WebRtcVideoSendChannel : public MediaChannelUtil, bool ValidateSendSsrcAvailability(const StreamParams& sp) const RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_); - // Populates `rtx_associated_payload_types`, `raw_payload_types` and - // `decoders` based on codec settings provided by `recv_codecs`. - // `recv_codecs` must be non-empty and all other parameters must be empty. - static void ExtractCodecInformation( - ArrayView recv_codecs, - std::map& rtx_associated_payload_types, - std::set& raw_payload_types, - std::vector& decoders); - // Wrapper for the sender part. class WebRtcVideoSendStream { public: @@ -505,13 +494,8 @@ class WebRtcVideoSendChannel : public MediaChannelUtil, RTC_GUARDED_BY(thread_checker_); VideoEncoderFactory* const encoder_factory_ RTC_GUARDED_BY(thread_checker_); - VideoDecoderFactory* const decoder_factory_ RTC_GUARDED_BY(thread_checker_); VideoBitrateAllocatorFactory* const bitrate_allocator_factory_ RTC_GUARDED_BY(thread_checker_); - std::vector recv_codecs_ RTC_GUARDED_BY(thread_checker_); - RtpHeaderExtensionMap recv_rtp_extension_map_ RTC_GUARDED_BY(thread_checker_); - std::vector recv_rtp_extensions_ - RTC_GUARDED_BY(thread_checker_); // See reason for keeping track of the FlexFEC payload type separately in // comment in WebRtcVideoChannel::ChangedReceiverParameters. int recv_flexfec_payload_type_ RTC_GUARDED_BY(thread_checker_); From 1e9de04049e38b18952c27fefefddce93bdf9843 Mon Sep 17 00:00:00 2001 From: Jeremy Leconte Date: Tue, 13 May 2025 11:10:03 +0200 Subject: [PATCH 133/292] Make apply-include-cleaner call the python script apply_include_cleaner.py. Change-Id: Idb87e8fbc7582b2a7577c4dd1f2feea701c7759d Bug: None Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390921 Commit-Queue: Harald Alvestrand Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44605} --- tools_webrtc/iwyu/apply-include-cleaner | 155 +----------------------- 1 file changed, 1 insertion(+), 154 deletions(-) diff --git a/tools_webrtc/iwyu/apply-include-cleaner b/tools_webrtc/iwyu/apply-include-cleaner index cec4dac31e..437f318248 100755 --- a/tools_webrtc/iwyu/apply-include-cleaner +++ b/tools_webrtc/iwyu/apply-include-cleaner @@ -1,156 +1,3 @@ #!/usr/bin/env bash -# -# Run the include-cleaner tool (iwyu replacement) on a file in the webrtc source -# directory. -# -# -# In order to handle include paths correctly, you need to provide -# a compile DB (aka compile_commands.json). -# You can create it in one of the following ways: -# "gn gen --export-compile-commands path/to/out" -# "tools/clang/scripts/generate_compdb.py -p path/to/out > compile_commands.json" -# If "out/Default" exists, the script will attempt to generate it for you. -# -# clang-include-cleaner is built as part of the "clangd" package in our -# LLVM build. -# Example .gclient file: -# solutions = [ -# { -# "name": "src", -# "url": "https://webrtc.googlesource.com/src.git", -# "deps_file": "DEPS", -# "managed": False, -# "custom_deps": {}, -# "custom_vars" : { -# "checkout_clangd": True, -# "download_remoteexec_cfg" : True, -# } -# }, -# ] - -CLEANER=third_party/llvm-build/Release+Asserts/bin/clang-include-cleaner -if [ ! -x $CLEANER ]; then - echo "clang-include-cleaner not found" - echo -n "Add '\"checkout_clangd\": True' to 'custom_vars' in your" - echo ".gclient file and run 'gclient sync'." - exit 1 -fi - -# Debug level, also controlled by the "-d" argument. -# Set this to 1 to get more debug information. -# Set this to 2 to also get a dump of the iwyu tool output. -DEBUG=0 - -set -e -if [ $DEBUG -gt 0 ]; then - set -x -fi - -error() { - echo "$*" >&2 - exit 1 -} - -WORKDIR=out/Default - -usage() { - echo "Usage: $0 [-r] file.cc [file2.cc ...]" - echo "Runs the include-cleaner tool on a list of files" - echo "Arguments:" - echo " -n : Just print changes, don't do them" - echo " -c : Just return non-zero exit code if there are changes, don't do them" - echo " -r : Remove non-required includes from .h file" - echo " -d : Set debug level to " - echo " -w : Specify the workdir (out/Default if not specified)" - echo " -h : Print this help message" - echo "" -} - -COMMAND=" --edit" -INCLUDE_ARGS="" -GMOCK_INCLUDES="--extra-arg=-I../../third_party/googletest/src/googlemock/include/" -GTEST_INCLUDES="--extra-arg=-I../../third_party/googletest/src/googletest/include/" -CHECK_MODE=false - -while getopts 'd:rncw:h' opts; do - case "${opts}" in - n) COMMAND=" --print=changes" ;; - c) COMMAND=" --print=changes" ; CHECK_MODE=true ;; - r) INCLUDE_ARGS=" --remove" ;; - d) DEBUG=${OPTARG};if [ $DEBUG -gt 0 ]; then set -x; fi ;; - w) WORKDIR=${OPTARG} ;; - h) usage; exit 1 ;; - *) error "Unexpected option ${opts}" ;; - esac -done -shift $(expr $OPTIND - 1 ) - -if [[ -z "$COMPILE_COMMANDS" ]]; then - if [ -d "$WORKDIR" ]; then - if [ ! -f "$WORKDIR/compile_commands.json" ]; then - echo "Generating compile commands file" - tools/clang/scripts/generate_compdb.py -p $WORKDIR > $WORKDIR/compile_commands.json - fi - COMPILE_COMMANDS="$WORKDIR/compile_commands.json" - else - usage - error "Could not generate $WORKDIR/compile_commands.json. Specify the workdir with -w if other than out/Default." - fi -fi - -if [ $# -eq 0 ]; then - usage - error "No files provided as arguments. To get a list of files in a commit: git diff-tree --no-commit-id --name-only -r HEAD" -fi - -for FILE in "$@" -do - if [ -z $FILE ] || [ ! -f $FILE ]; then - usage - error "File $FILE is not found" - fi -done - -HAS_OUTPUT=false -for FILE in "$@" -do - OUTPUT=$($CLEANER -p $WORKDIR $INCLUDE_ARGS $GMOCK_INCLUDES $GTEST_INCLUDES $COMMAND $FILE) - - # include-cleaner does not support custom mappings for certain deps - # this ensures that the gtest/gmock deps it inserts are replaced - # with the right paths for those includes. - # Since sed inplace argument acts differently between GNU/BSD based systems - # we handle this here. - case "$(uname -s)" in - Linux*) INPLACE_ARG=( -i );; - Darwin*) INPLACE_ARG=( -i '' );; - *) INPLACE_ARG=( -i ) - esac - IWYU_MAPPING=( "\"gmock\/gmock\.h\":\"test\/gmock\.h\"" - "\"gtest\/gtest\.h\":\"test\/gtest\.h\"" - "\:\"rtc_base\/net_helpers\.h\"" ) - - for mapping in "${IWYU_MAPPING[@]}" ; do - KEY="${mapping%%:*}" - VALUE="${mapping##*:}" - if grep -q "#include ${VALUE}" $FILE; then - OUTPUT=$(echo "$OUTPUT" | sed "/+ ${KEY}/d") - sed "${INPLACE_ARG[@]}" -e "/#include ${KEY}/d" $FILE - else - sed "${INPLACE_ARG[@]}" -e "s@^#include ${KEY}@#include ${VALUE}@g" $FILE - fi - done - - echo "${OUTPUT}" - HAS_OUTPUT=$HAS_OUTPUT || [[ ! -z $OUTPUT ]] -done - -echo "Finished. Check diff, compile, gn gen --check (tools_webrtc/gn_check_autofix.py can fix most of the issues)" -echo "and git cl format before uploading." - -# Return a non-zero exit code if running with "CHECK_MODE" -# and there are changes to apply. -if $CHECK_MODE && [[ ! -z $OUTPUT ]]; then - exit 1 -fi +tools_webrtc/iwyu/apply_include_cleaner.py "$@" From 2f21adc141ea795562b9fb00f7e772f27f2ae27d Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Fri, 9 May 2025 11:54:45 +0000 Subject: [PATCH 134/292] WebRTC Rusty Base64 Bug: webrtc:416445288 Change-Id: I6eb49d95aa8000812b5299d25198a79a8c069ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390400 Commit-Queue: Harald Alvestrand Reviewed-by: Mirko Bonadei Reviewed-by: Harald Alvestrand Auto-Submit: Evan Shrubsole Cr-Commit-Position: refs/heads/main@{#44606} --- .gitignore | 1 + rtc_base/BUILD.gn | 60 +++++++++++++++++++++++++++++++++ rtc_base/DEPS | 3 ++ rtc_base/base64.cc | 4 +++ rtc_base/base64.h | 11 +++---- rtc_base/base64.rs | 73 +++++++++++++++++++++++++++++++++++++++++ rtc_base/base64_rust.cc | 41 +++++++++++++++++++++++ webrtc.gni | 3 ++ 8 files changed, 189 insertions(+), 7 deletions(-) create mode 100644 rtc_base/base64.rs create mode 100644 rtc_base/base64_rust.cc diff --git a/.gitignore b/.gitignore index 9f8caa9a88..2fb77ea2a8 100644 --- a/.gitignore +++ b/.gitignore @@ -73,3 +73,4 @@ /xcodebuild /.vscode !webrtc/* +rust-project.json diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn index 430060fc3d..1766edd83e 100644 --- a/rtc_base/BUILD.gn +++ b/rtc_base/BUILD.gn @@ -1795,6 +1795,66 @@ rtc_library("base64") { ] } +if (rtc_rusty_base64) { + import("//build/config/rust.gni") + import("//build/rust/rust_static_library.gni") + + rtc_source_set("base64_header") { + sources = [ "base64.h" ] + deps = [ + "../api:array_view", + "//third_party/abseil-cpp/absl/strings:string_view", + ] + } + + rust_static_library("base64_rust_bridge") { + allow_unsafe = true # Needed for FFI with cxx crate. + crate_root = "base64.rs" + sources = [ "base64.rs" ] + cxx_bindings = [ "base64.rs" ] + deps = [ + ":base64_header", + "//third_party/rust/base64/v0_22:lib", + ] + visibility = [ ":rusty_base64" ] + } + + rtc_library("rusty_base64") { + sources = [ "base64_rust.cc" ] + deps = [ + ":base64_header", + ":base64_rust_bridge", + "//build/rust:cxx_cppdeps", + "//third_party/abseil-cpp/absl/strings:string_view", + ] + } + + if (rtc_include_tests) { + rtc_test("rusty_base64_unittest") { + sources = [ "base64_unittest.cc" ] + deps = [ + ":base64_header", + ":rusty_base64", + "../test:test_main", + "../test:test_support", + "//third_party/abseil-cpp/absl/strings:string_view", + ] + } + + if (rtc_enable_google_benchmarks) { + rtc_test("rusty_base64_benchmark") { + sources = [ "base64_benchmark.cc" ] + deps = [ + ":base64_header", + ":rusty_base64", + "../test:benchmark_main", + "//third_party/google_benchmark", + ] + } + } + } +} + if (rtc_include_tests) { if (rtc_enable_google_benchmarks) { rtc_test("base64_benchmark") { diff --git a/rtc_base/DEPS b/rtc_base/DEPS index 0e7237d329..9e8908ceb2 100644 --- a/rtc_base/DEPS +++ b/rtc_base/DEPS @@ -41,4 +41,7 @@ specific_include_rules = { "base64_benchmark\.cc": [ "+benchmark", ], + "base64_rust\.cc": [ + "+third_party/rust/chromium_crates_io/vendor/cxx-v1/include/cxx.h", + ], } diff --git a/rtc_base/base64.cc b/rtc_base/base64.cc index eaa4d2bfd6..e9fc5d05c8 100644 --- a/rtc_base/base64.cc +++ b/rtc_base/base64.cc @@ -29,6 +29,10 @@ bool IsStrictBase64(absl::string_view data) { } } // namespace +std::string Base64Encode(absl::string_view data) { + return absl::Base64Escape(data); +} + std::optional Base64Decode(absl::string_view data, Base64DecodeOptions options) { // absl::Base64Unescape is forgiving. Return nullopt if the input is not diff --git a/rtc_base/base64.h b/rtc_base/base64.h index 24c1d97e0c..9a83d8b07d 100644 --- a/rtc_base/base64.h +++ b/rtc_base/base64.h @@ -15,22 +15,19 @@ #include #include -#include "absl/strings/escaping.h" #include "absl/strings/string_view.h" #include "api/array_view.h" namespace webrtc { +std::string Base64Encode(absl::string_view data); + inline std::string Base64Encode(ArrayView data) { - return absl::Base64Escape(absl::string_view( + return Base64Encode(absl::string_view( reinterpret_cast(data.data()), data.size())); } -inline std::string Base64Encode(absl::string_view data) { - return absl::Base64Escape(data); -} - -enum class Base64DecodeOptions { +enum class Base64DecodeOptions : uint8_t { kStrict, // Matches https://infra.spec.whatwg.org/#forgiving-base64-decode. kForgiving, diff --git a/rtc_base/base64.rs b/rtc_base/base64.rs new file mode 100644 index 0000000000..f9397dd457 --- /dev/null +++ b/rtc_base/base64.rs @@ -0,0 +1,73 @@ +/* + * Copyright (c) 2025 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +use base64::alphabet; +use base64::engine::general_purpose; +use base64::engine::DecodePaddingMode; +use base64::Engine; +use cxx::CxxString; +use std::pin::Pin; + +#[cxx::bridge(namespace = "webrtc")] +mod ffi { + #[repr(u8)] + enum Base64DecodeOptions { + kStrict, + kForgiving, + } + + extern "C++" { + include!("rtc_base/base64.h"); + type Base64DecodeOptions; + } + + extern "Rust" { + fn rs_base64_encode(data: &[u8]) -> String; + fn rs_base64_decode( + data: &[u8], + options: Base64DecodeOptions, + output: Pin<&mut CxxString>, + ) -> bool; + } +} + +fn rs_base64_encode(data: &[u8]) -> String { + general_purpose::STANDARD.encode(data) +} + +const FORGIVING_ENGINE: general_purpose::GeneralPurpose = general_purpose::GeneralPurpose::new( + &alphabet::STANDARD, + general_purpose::GeneralPurposeConfig::new() + .with_decode_padding_mode(DecodePaddingMode::Indifferent), +); + +fn rs_base64_decode( + data: &[u8], + options: ffi::Base64DecodeOptions, + output: Pin<&mut CxxString>, +) -> bool { + let result = match options { + ffi::Base64DecodeOptions::kStrict => general_purpose::STANDARD.decode(data), + ffi::Base64DecodeOptions::kForgiving => { + let data_without_whitespace: Vec = + data.iter().filter(|&c| !c.is_ascii_whitespace()).copied().collect(); + FORGIVING_ENGINE.decode(data_without_whitespace) + } + _ => unreachable!(), + }; + + match result { + Ok(vec) => { + output.push_bytes(vec.as_slice()); + true + } + Err(_) => false, + } +} diff --git a/rtc_base/base64_rust.cc b/rtc_base/base64_rust.cc new file mode 100644 index 0000000000..30061046c9 --- /dev/null +++ b/rtc_base/base64_rust.cc @@ -0,0 +1,41 @@ +/* + * Copyright (c) 2025 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include +#include +#include + +#include "absl/strings/string_view.h" +#include "rtc_base/base64.h" +#include "rtc_base/base64.rs.h" +#include "third_party/rust/chromium_crates_io/vendor/cxx-v1/include/cxx.h" + +namespace webrtc { + +std::string Base64Encode(absl::string_view data) { + rust::Slice input_slice( + reinterpret_cast(data.data()), data.size()); + rust::Vec output_slice; + rust::String str = rs_base64_encode(input_slice); + return std::string(str); +} + +std::optional Base64Decode(absl::string_view data, + Base64DecodeOptions options) { + rust::Slice input_slice( + reinterpret_cast(data.data()), data.size()); + std::string output; + if (!rs_base64_decode(input_slice, options, output)) { + return std::nullopt; + } + return output; +} + +} // namespace webrtc diff --git a/webrtc.gni b/webrtc.gni index 11ab193ec7..450476539c 100644 --- a/webrtc.gni +++ b/webrtc.gni @@ -334,6 +334,9 @@ declare_args() { # Set this to true to disable webrtc metrics. rtc_disable_metrics = false + + # Enables an experimental rust version of base64 for building and testing. + rtc_rusty_base64 = true } declare_args() { From 3f48fd60708b38c98e4c3ad2308775a12ffed693 Mon Sep 17 00:00:00 2001 From: Daniel White Date: Tue, 13 May 2025 07:12:58 -0700 Subject: [PATCH 135/292] Revert "[iOS] Update RTCCameraPreviewView to conform to iOS 17" MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This reverts commit 887a6b9f7fb88291413ccc454651cb5e571740b2. Reason for revert: The build guard is incorrect. Bug: chromium:348649285 Original change's description: > [iOS] Update RTCCameraPreviewView to conform to iOS 17 > > In iOS 17, the video orientation API is deprecated. This CL updates the > RTCCameraPreviewView file to use the video rotation angle API instead. > > This file is being updated now to support Chrome for iOS effort to raise > its minimum deployment target to iOS 17. > > Bug: chromium:348649285 > Change-Id: Icf72911e9a3431a9a91fefc3d85f9ea02597c5f4 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390080 > Reviewed-by: Mirko Bonadei > Commit-Queue: Mirko Bonadei > Reviewed-by: Kári Helgason > Cr-Commit-Position: refs/heads/main@{#44603} Bug: chromium:348649285 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I69482894536b6d8b67e733e2745f73ed34236662 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391220 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com Reviewed-by: Mirko Bonadei Commit-Queue: Mirko Bonadei Owners-Override: Mirko Bonadei Cr-Commit-Position: refs/heads/main@{#44607} --- sdk/objc/helpers/RTCCameraPreviewView.m | 70 ++++++++----------------- 1 file changed, 21 insertions(+), 49 deletions(-) diff --git a/sdk/objc/helpers/RTCCameraPreviewView.m b/sdk/objc/helpers/RTCCameraPreviewView.m index 77d5e8823d..92c3cd2635 100644 --- a/sdk/objc/helpers/RTCCameraPreviewView.m +++ b/sdk/objc/helpers/RTCCameraPreviewView.m @@ -80,54 +80,6 @@ - (void)orientationChanged:(NSNotification *)notification { } - (void)setCorrectVideoOrientation { - if (@available(iOS 17, *)) { - [self modifyVideoAngle]; - return; - } - - [self modifyVideoOrientation]; -} - -#pragma mark - Private - -- (void)addOrientationObserver { - [[NSNotificationCenter defaultCenter] - addObserver:self - selector:@selector(orientationChanged:) - name:UIDeviceOrientationDidChangeNotification - object:nil]; -} - -- (void)removeOrientationObserver { - [[NSNotificationCenter defaultCenter] - removeObserver:self - name:UIDeviceOrientationDidChangeNotification - object:nil]; -} - -- (AVCaptureVideoPreviewLayer *)previewLayer { - return (AVCaptureVideoPreviewLayer *)self.layer; -} - -- (void)modifyVideoAngle API_AVAILABLE(ios(17.0)) { - AVCaptureDeviceInput* captureSessionInput = - _captureSession.inputs.firstObject; - AVCaptureDevice* camera = captureSessionInput.device; - AVCaptureVideoPreviewLayer* previewLayer = [self previewLayer]; - AVCaptureConnection* videoConnection = previewLayer.connection; - AVCaptureDeviceRotationCoordinator* rotationCoordiantor = - [[AVCaptureDeviceRotationCoordinator alloc] - initWithDevice:camera - previewLayer:previewLayer]; - CGFloat angle = - rotationCoordiantor.videoRotationAngleForHorizonLevelCapture; - if ([videoConnection isVideoRotationAngleSupported:angle]) { - [videoConnection setVideoRotationAngle:angle]; - } -} - -#if defined(__IPHONE_17_0) && __IPHONE_OS_VERSION_MAX_ALLOWED >= __IPHONE_17_0 -- (void)modifyVideoOrientation { // Get current device orientation. UIDeviceOrientation deviceOrientation = [UIDevice currentDevice].orientation; AVCaptureVideoPreviewLayer *previewLayer = [self previewLayer]; @@ -152,6 +104,26 @@ - (void)modifyVideoOrientation { // orientation. } } -#endif + +#pragma mark - Private + +- (void)addOrientationObserver { + [[NSNotificationCenter defaultCenter] + addObserver:self + selector:@selector(orientationChanged:) + name:UIDeviceOrientationDidChangeNotification + object:nil]; +} + +- (void)removeOrientationObserver { + [[NSNotificationCenter defaultCenter] + removeObserver:self + name:UIDeviceOrientationDidChangeNotification + object:nil]; +} + +- (AVCaptureVideoPreviewLayer *)previewLayer { + return (AVCaptureVideoPreviewLayer *)self.layer; +} @end From 6b11987da4067f88039dd85b907fa634df3313b3 Mon Sep 17 00:00:00 2001 From: Tommi Date: Tue, 13 May 2025 14:20:15 +0200 Subject: [PATCH 136/292] Add ACMResampler::Resample10Msec() with support for views. MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit A follow up change will remove the version with raw pointers. Bug: chromium:335805780 Change-Id: I8f5a18b152222daf5ce0abfb230f3e38a4e9e25e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369741 Reviewed-by: Per Åhgren Commit-Queue: Tomas Gunnarsson Cr-Commit-Position: refs/heads/main@{#44608} --- .../resampler/include/push_resampler.h | 6 +- modules/audio_coding/acm2/acm_resampler.cc | 24 ++- modules/audio_coding/acm2/acm_resampler.h | 8 +- .../audio_coding/acm2/audio_coding_module.cc | 173 ++++++++++++------ modules/audio_coding/test/opus_test.cc | 9 +- 5 files changed, 144 insertions(+), 76 deletions(-) diff --git a/common_audio/resampler/include/push_resampler.h b/common_audio/resampler/include/push_resampler.h index 394e96b133..20d40423d1 100644 --- a/common_audio/resampler/include/push_resampler.h +++ b/common_audio/resampler/include/push_resampler.h @@ -11,6 +11,7 @@ #ifndef COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_ #define COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_ +#include #include #include @@ -32,10 +33,11 @@ class PushResampler final { ~PushResampler(); // Returns the total number of samples provided in destination (e.g. 32 kHz, - // 2 channel audio gives 640 samples). + // 2 channel audio gives 640 samples) - never a negative value. int Resample(InterleavedView src, InterleavedView dst); // For when a deinterleaved/mono channel already exists and we can skip the - // deinterleaved operation. + // deinterleaved operation. The return value is the number of resampled + // samples. int Resample(MonoView src, MonoView dst); private: diff --git a/modules/audio_coding/acm2/acm_resampler.cc b/modules/audio_coding/acm2/acm_resampler.cc index 1425bc94df..346ec68377 100644 --- a/modules/audio_coding/acm2/acm_resampler.cc +++ b/modules/audio_coding/acm2/acm_resampler.cc @@ -39,8 +39,21 @@ int ACMResampler::Resample10Msec(const int16_t* in_audio, SampleRateToDefaultChannelSize(out_freq_hz), num_audio_channels); RTC_DCHECK_GE(out_capacity_samples, dst.size()); + return Resample10Msec(src, in_freq_hz, dst, out_freq_hz); +} + +int ACMResampler::Resample10Msec(InterleavedView src, + int in_freq_hz, + InterleavedView dst, + int out_freq_hz) { + RTC_DCHECK_EQ(src.num_channels(), dst.num_channels()); if (in_freq_hz == out_freq_hz) { - if (out_capacity_samples < src.data().size()) { + // TODO(tommi): Could this `if()` be replaced with a stricter check? + // RTC_CHECK_EQ(dst.size(), src.size()); + // Eventually change the method to never return -1 and not have to + // expect that at higher level call sites. The Resample() method below + // can never return -1 + if (dst.size() < src.size()) { RTC_DCHECK_NOTREACHED(); return -1; } @@ -50,14 +63,9 @@ int ACMResampler::Resample10Msec(const int16_t* in_audio, } int out_length = resampler_.Resample(src, dst); - if (out_length == -1) { - RTC_LOG(LS_ERROR) << "Resample(" << in_audio << ", " << src.data().size() - << ", " << out_audio << ", " << out_capacity_samples - << ") failed."; - return -1; - } + RTC_CHECK_GE(out_length, 0); RTC_DCHECK_EQ(out_length, dst.size()); - RTC_DCHECK_EQ(out_length / num_audio_channels, dst.samples_per_channel()); + RTC_DCHECK_EQ(out_length / dst.num_channels(), dst.samples_per_channel()); return static_cast(dst.samples_per_channel()); } diff --git a/modules/audio_coding/acm2/acm_resampler.h b/modules/audio_coding/acm2/acm_resampler.h index 244e215df5..abf08f063c 100644 --- a/modules/audio_coding/acm2/acm_resampler.h +++ b/modules/audio_coding/acm2/acm_resampler.h @@ -17,6 +17,7 @@ #include #include "api/audio/audio_frame.h" +#include "api/audio/audio_view.h" #include "common_audio/resampler/include/push_resampler.h" namespace webrtc { @@ -27,7 +28,7 @@ class ACMResampler { ACMResampler(); ~ACMResampler(); - // TODO: b/335805780 - Change to accept InterleavedView<>. + // TODO: b/335805780 - Remove and use only the InterleavedView<> version int Resample10Msec(const int16_t* in_audio, int in_freq_hz, int out_freq_hz, @@ -35,6 +36,11 @@ class ACMResampler { size_t out_capacity_samples, int16_t* out_audio); + int Resample10Msec(InterleavedView src, + int src_freq_hz, + InterleavedView dst, + int dst_freq_hz); + private: PushResampler resampler_; }; diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc index 21ec232eed..6f69012755 100644 --- a/modules/audio_coding/acm2/audio_coding_module.cc +++ b/modules/audio_coding/acm2/audio_coding_module.cc @@ -20,6 +20,7 @@ #include "absl/strings/string_view.h" #include "api/array_view.h" +#include "api/audio/audio_view.h" #include "api/audio_codecs/audio_encoder.h" #include "api/function_view.h" #include "modules/audio_coding/acm2/acm_remixing.h" @@ -122,6 +123,13 @@ class AudioCodingModuleImpl final : public AudioCodingModule { bool HaveValidEncoder(absl::string_view caller_name) const RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_); + // Updates or checks `expected_in_ts_` and `expected_codec_ts` based on the + // timestamps in `in_frame`. If no audio frame has been received, the fields + // are just set. For subsequent frames, the expected timestamps are checked + // for consistency. + void SetInputTimestamps(const AudioFrame& in_frame) + RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_); + // Preprocessing of input audio, including resampling and down-mixing if // required, before pushing audio into encoder's buffer. // @@ -137,6 +145,20 @@ class AudioCodingModuleImpl final : public AudioCodingModule { const AudioFrame** ptr_out) RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_); + // Called from `PreprocessToAddData` when no resampling or downmixing is + // required. Returns a pointer to an output audio_frame. If timestamps are as + // expected the return value will point to `in_frame`, otherwise the data will + // have been copied into `preprocess_frame_` and the returned pointer points + // to `preprocess_frame_`. + const AudioFrame* AddDataNoPreProcess(const AudioFrame& in_frame) + RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_); + + // Resamples `source` into the `preprocess_frame_` buffer. + // `sample_rate` refers to the sample rate of `source`. + bool ResampleToPreProcessFrame(InterleavedView source, + int sample_rate) + RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_); + // Change required states after starting to receive the codec corresponding // to `index`. int UpdateUponReceivingCodec(int index); @@ -410,6 +432,23 @@ int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame, return 0; } +void AudioCodingModuleImpl::SetInputTimestamps(const AudioFrame& in_frame) { + if (!first_10ms_data_) { + expected_in_ts_ = in_frame.timestamp_; + expected_codec_ts_ = in_frame.timestamp_; + first_10ms_data_ = true; + } else if (in_frame.timestamp_ != expected_in_ts_) { + RTC_LOG(LS_WARNING) << "Unexpected input timestamp: " << in_frame.timestamp_ + << ", expected: " << expected_in_ts_; + expected_codec_ts_ += + (in_frame.timestamp_ - expected_in_ts_) * + static_cast( + static_cast(encoder_stack_->SampleRateHz()) / + static_cast(in_frame.sample_rate_hz_)); + expected_in_ts_ = in_frame.timestamp_; + } +} + // Perform a resampling and down-mix if required. We down-mix only if // encoder is mono and input is stereo. In case of dual-streaming, both // encoders has to be mono for down-mix to take place. @@ -418,6 +457,8 @@ int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame, // TODO(yujo): Make this more efficient for muted frames. int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame, const AudioFrame** ptr_out) { + SetInputTimestamps(in_frame); + const bool resample = in_frame.sample_rate_hz_ != encoder_stack_->SampleRateHz(); @@ -428,82 +469,53 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame, const bool down_mix = in_frame.num_channels_ == 2 && encoder_stack_->NumChannels() == 1; - if (!first_10ms_data_) { - expected_in_ts_ = in_frame.timestamp_; - expected_codec_ts_ = in_frame.timestamp_; - first_10ms_data_ = true; - } else if (in_frame.timestamp_ != expected_in_ts_) { - RTC_LOG(LS_WARNING) << "Unexpected input timestamp: " << in_frame.timestamp_ - << ", expected: " << expected_in_ts_; - expected_codec_ts_ += - (in_frame.timestamp_ - expected_in_ts_) * - static_cast( - static_cast(encoder_stack_->SampleRateHz()) / - static_cast(in_frame.sample_rate_hz_)); - expected_in_ts_ = in_frame.timestamp_; - } - if (!down_mix && !resample) { // No preprocessing is required. - if (expected_in_ts_ == expected_codec_ts_) { - // If we've never resampled, we can use the input frame as-is - *ptr_out = &in_frame; - } else { - // Otherwise we'll need to alter the timestamp. Since in_frame is const, - // we'll have to make a copy of it. - preprocess_frame_.CopyFrom(in_frame); - preprocess_frame_.timestamp_ = expected_codec_ts_; - *ptr_out = &preprocess_frame_; - } - - expected_in_ts_ += static_cast(in_frame.samples_per_channel_); - expected_codec_ts_ += static_cast(in_frame.samples_per_channel_); + *ptr_out = AddDataNoPreProcess(in_frame); return 0; } + // Some pre-processing will be required, so we'll use the internal buffer. *ptr_out = &preprocess_frame_; - preprocess_frame_.num_channels_ = in_frame.num_channels_; + preprocess_frame_.timestamp_ = expected_codec_ts_; preprocess_frame_.samples_per_channel_ = in_frame.samples_per_channel_; + + // Temporary buffer in case both downmixing and resampling is required. std::array audio; - const int16_t* src_ptr_audio; + // When resampling is needed, this view will represent the buffer to resample. + InterleavedView resample_src_audio; + if (down_mix) { - // If a resampling is required, the output of a down-mix is written into a - // local buffer, otherwise, it will be written to the output frame. - int16_t* dest_ptr_audio = - resample ? audio.data() : preprocess_frame_.mutable_data(); - RTC_DCHECK_GE(audio.size(), preprocess_frame_.samples_per_channel_); - RTC_DCHECK_GE(audio.size(), in_frame.samples_per_channel_); - DownMixFrame(in_frame, - ArrayView(dest_ptr_audio, - preprocess_frame_.samples_per_channel_)); - preprocess_frame_.num_channels_ = 1; - - // Set the input of the resampler to the down-mixed signal. - src_ptr_audio = audio.data(); + RTC_DCHECK_GE(audio.size(), in_frame.samples_per_channel()); + preprocess_frame_.num_channels_ = 1; // We always downmix to mono. + // If a resampling is also required, the output of a down-mix is written + // into a local buffer, otherwise, it will be written to the output frame. + auto downmixed = + resample + ? InterleavedView(audio.data(), + in_frame.samples_per_channel(), 1) + : preprocess_frame_.mutable_data(in_frame.samples_per_channel(), 1); + DownMixFrame(in_frame, downmixed.AsMono()); + if (resample) { + // Set the input for the resampler to the down-mixed signal. + resample_src_audio = downmixed; + } } else { - // Set the input of the resampler to the original data. - src_ptr_audio = in_frame.data(); + preprocess_frame_.num_channels_ = in_frame.num_channels_; + if (resample) { + // Set the input of the resampler to the original data. + resample_src_audio = in_frame.data_view(); + } } - preprocess_frame_.timestamp_ = expected_codec_ts_; - preprocess_frame_.sample_rate_hz_ = in_frame.sample_rate_hz_; - // If it is required, we have to do a resampling. - if (resample) { - // The result of the resampler is written to output frame. - int16_t* dest_ptr_audio = preprocess_frame_.mutable_data(); + RTC_DCHECK(resample_src_audio.empty() || resample); + preprocess_frame_.SetSampleRateAndChannelSize(encoder_stack_->SampleRateHz()); - int samples_per_channel = resampler_.Resample10Msec( - src_ptr_audio, in_frame.sample_rate_hz_, encoder_stack_->SampleRateHz(), - preprocess_frame_.num_channels_, AudioFrame::kMaxDataSizeSamples, - dest_ptr_audio); - - if (samples_per_channel < 0) { - RTC_LOG(LS_ERROR) << "Cannot add 10 ms audio, resampling failed"; + if (resample) { + if (!ResampleToPreProcessFrame(resample_src_audio, + in_frame.sample_rate_hz())) { return -1; } - preprocess_frame_.samples_per_channel_ = - static_cast(samples_per_channel); - preprocess_frame_.sample_rate_hz_ = encoder_stack_->SampleRateHz(); } expected_codec_ts_ += @@ -513,6 +525,45 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame, return 0; } +const AudioFrame* AudioCodingModuleImpl::AddDataNoPreProcess( + const AudioFrame& in_frame) { + const AudioFrame* ret = nullptr; + // No preprocessing is required. + if (expected_in_ts_ == expected_codec_ts_) { + // Timestamps as expected, we can use the input frame as-is. + ret = &in_frame; + } else { + // Otherwise we'll need to alter the timestamp. Since in_frame is const, + // we'll have to make a copy of it. + preprocess_frame_.CopyFrom(in_frame); + preprocess_frame_.timestamp_ = expected_codec_ts_; + ret = &preprocess_frame_; + } + + expected_in_ts_ += static_cast(in_frame.samples_per_channel_); + expected_codec_ts_ += static_cast(in_frame.samples_per_channel_); + + return ret; +} + +bool AudioCodingModuleImpl::ResampleToPreProcessFrame( + InterleavedView source, + int sample_rate) { + int samples_per_channel = resampler_.Resample10Msec( + source, sample_rate, + preprocess_frame_.mutable_data(preprocess_frame_.samples_per_channel(), + preprocess_frame_.num_channels()), + preprocess_frame_.sample_rate_hz()); + + // TODO(tommi): Resample10Msec() should never return an error. Remove this + // check and change this method to return void. + if (samples_per_channel < 0) { + RTC_LOG(LS_ERROR) << "Cannot add 10 ms audio, resampling failed"; + } + + return samples_per_channel >= 0; +} + ///////////////////////////////////////// // (FEC) Forward Error Correction (codec internal) // diff --git a/modules/audio_coding/test/opus_test.cc b/modules/audio_coding/test/opus_test.cc index 8bde843e8e..e908a807ed 100644 --- a/modules/audio_coding/test/opus_test.cc +++ b/modules/audio_coding/test/opus_test.cc @@ -17,6 +17,7 @@ #include #include "api/audio/audio_frame.h" +#include "api/audio/audio_view.h" #include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/environment/environment_factory.h" @@ -265,10 +266,10 @@ void OpusTest::Run(TestPackStereo* channel, } // If input audio is sampled at 32 kHz, resampling to 48 kHz is required. - EXPECT_EQ(480, resampler_.Resample10Msec( - audio_frame.data(), audio_frame.sample_rate_hz_, 48000, - channels, kBufferSizeSamples - written_samples, - &audio[written_samples])); + InterleavedView dst(&audio[written_samples], 480, channels); + EXPECT_EQ(480, resampler_.Resample10Msec(audio_frame.data_view(), + audio_frame.sample_rate_hz_, dst, + 48000)); written_samples += 480 * channels; // Sometimes we need to loop over the audio vector to produce the right From 23cb9ddb3226cdb91c6447dc62a4156f55214331 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Tue, 13 May 2025 07:11:11 -0700 Subject: [PATCH 137/292] Roll chromium_revision 80cb2d193d..86f00502dd (1459248:1459405) Change log: https://chromium.googlesource.com/chromium/src/+log/80cb2d193d..86f00502dd Full diff: https://chromium.googlesource.com/chromium/src/+/80cb2d193d..86f00502dd Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/70d4d5cf23..336cce9dcf * src/build: https://chromium.googlesource.com/chromium/src/build/+log/a7f3c053b0..9c19bf6fd8 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/83a5db416b..feb5fd2c3d * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/3d355ecb9a..3d4f61f5e0 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/70731bd9b3..c212732796 * src/third_party/androidx/cipd: 77JFcazE3nHDyZZ4RYND-IEyes3SQTICM2Kz231QD60C..NgN4G3kZkU23fj174TE4RsZAis3Apk2LfPTjb8wllxwC * src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/9c1ffdc7fb..1f21600f67 * src/third_party/googletest/src: https://chromium.googlesource.com/external/github.com/google/googletest.git/+log/90a4152114..8b8ef3ff0d * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/b40180e2fc..b0e95ba77d DEPS diff: https://chromium.googlesource.com/chromium/src/+/80cb2d193d..86f00502dd/DEPS No update to Clang. BUG=None Change-Id: I97f92d3d4e608d4db762e13c04e835c5c18c2ede Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391200 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44609} --- DEPS | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/DEPS b/DEPS index e037952ab5..5c84f06c22 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '80cb2d193db7515d8f2a457cbdac343bf0167588', + 'chromium_revision': '86f00502ddde0c74aa718eba555dc4961c1c325d', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@70d4d5cf23c155e8a9cb4c5c8b4e751e8763707e', + 'https://chromium.googlesource.com/chromium/src/base@336cce9dcf88c86858f4ff87f446d697952e48ad', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@a7f3c053b0f19d21089b7ceb3e62f525c0d02171', + 'https://chromium.googlesource.com/chromium/src/build@9c19bf6fd8ddb59c6cbb1c3498cd2100b5983113', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@077a66f30fcf281b066fafb6dfc60818c238efb6', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@83a5db416b63068413e15dab161f92b679b5f5a5', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@feb5fd2c3d8e44a6d1f41942dc455a2424fc8212', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@3d355ecb9aee09b8b4cfe6ee0ce63b98412965ec', + 'https://chromium.googlesource.com/chromium/src/testing@3d4f61f5e05d19364fe407b6706702d85e26f489', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@70731bd9b3ace2d70fc063b27182decf1ca272b8', + 'https://chromium.googlesource.com/chromium/src/third_party@c2127327968c62d37d7c039e218b935f6a93637c', 'src/buildtools/linux64': { 'packages': [ @@ -525,7 +525,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/googletest/src': - 'https://chromium.googlesource.com/external/github.com/google/googletest.git@90a41521142c978131f38c6da07b4eb96a9f1ff6', + 'https://chromium.googlesource.com/external/github.com/google/googletest.git@8b8ef3ff0d1f7a1739e0d7c3dd46ab37104fcd9a', 'src/third_party/icu': { 'url': 'https://chromium.googlesource.com/chromium/deps/icu.git@b929596baebf0ab4ac7ec07f38365db4c50a559d', }, @@ -583,7 +583,7 @@ deps = { 'src/third_party/libFuzzer/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@e31b99917861f891308269c36a32363b120126bb', 'src/third_party/fuzztest/src': - 'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@9c1ffdc7fb3d8db64e42c7a9662d8b9906625b77', + 'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@1f21600f675e2bb8972acf57bfae6ff3416d37bb', 'src/third_party/libjpeg_turbo': 'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@e14cbfaa85529d47f9f55b0f104a579c1061f9ad', 'src/third_party/libsrtp': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@b40180e2fcc6ef2334b3d0c463509d968de96ac8', + 'https://chromium.googlesource.com/chromium/src/tools@b0e95ba77dbdf0db4946f1316664fdc82e4bef30', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': '77JFcazE3nHDyZZ4RYND-IEyes3SQTICM2Kz231QD60C', + 'version': 'NgN4G3kZkU23fj174TE4RsZAis3Apk2LfPTjb8wllxwC', }, ], 'condition': 'checkout_android and non_git_source', From 83dac84ea01ec66634804fe88ebedca9ea9859b8 Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Fri, 9 May 2025 14:56:51 +0200 Subject: [PATCH 138/292] Delete deprecated EnableMediaWithDefaultsAndTimeController Custom clock for testing can be injected via PeerConnectionFactoryDependencies::env Bug: None Change-Id: I35a9de6d974f12524ca7335434cd94aca9ee0570 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390680 Reviewed-by: Harald Alvestrand Commit-Queue: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#44610} --- api/BUILD.gn | 11 ------- api/test/create_time_controller.cc | 47 ------------------------------ api/test/create_time_controller.h | 9 ------ 3 files changed, 67 deletions(-) diff --git a/api/BUILD.gn b/api/BUILD.gn index 125941ac45..52df41182f 100644 --- a/api/BUILD.gn +++ b/api/BUILD.gn @@ -1550,20 +1550,9 @@ if (rtc_include_tests) { ] deps = [ - ":enable_media_with_defaults", - ":libjingle_peerconnection_api", ":time_controller", - "../call", - "../call:call_interfaces", - "../media:media_engine", - "../pc:media_factory", - "../rtc_base:checks", - "../system_wrappers", "../test/time_controller", - "environment", - "environment:environment_factory", "units:timestamp", - "//third_party/abseil-cpp/absl/base:nullability", ] } diff --git a/api/test/create_time_controller.cc b/api/test/create_time_controller.cc index 9754f70ea8..2c50599a9c 100644 --- a/api/test/create_time_controller.cc +++ b/api/test/create_time_controller.cc @@ -11,21 +11,9 @@ #include "api/test/create_time_controller.h" #include -#include -#include "absl/base/nullability.h" -#include "api/enable_media_with_defaults.h" -#include "api/environment/environment.h" -#include "api/environment/environment_factory.h" -#include "api/peer_connection_interface.h" #include "api/test/time_controller.h" #include "api/units/timestamp.h" -#include "call/call.h" -#include "call/call_config.h" -#include "media/base/media_engine.h" -#include "pc/media_factory.h" -#include "rtc_base/checks.h" -#include "system_wrappers/include/clock.h" #include "test/time_controller/simulated_time_controller.h" namespace webrtc { @@ -35,39 +23,4 @@ std::unique_ptr CreateSimulatedTimeController() { Timestamp::Seconds(10000)); } -void EnableMediaWithDefaultsAndTimeController( - TimeController& time_controller, - PeerConnectionFactoryDependencies& deps) { - class TimeControllerBasedFactory : public MediaFactory { - public: - TimeControllerBasedFactory( - Clock* absl_nonnull clock, - absl_nonnull std::unique_ptr media_factory) - : clock_(clock), media_factory_(std::move(media_factory)) {} - - std::unique_ptr CreateCall(CallConfig config) override { - EnvironmentFactory env_factory(config.env); - env_factory.Set(clock_); - - config.env = env_factory.Create(); - return media_factory_->CreateCall(std::move(config)); - } - - std::unique_ptr CreateMediaEngine( - const Environment& env, - PeerConnectionFactoryDependencies& dependencies) override { - return media_factory_->CreateMediaEngine(env, dependencies); - } - - private: - Clock* absl_nonnull clock_; - absl_nonnull std::unique_ptr media_factory_; - }; - - EnableMediaWithDefaults(deps); - RTC_CHECK(deps.media_factory); - deps.media_factory = std::make_unique( - time_controller.GetClock(), std::move(deps.media_factory)); -} - } // namespace webrtc diff --git a/api/test/create_time_controller.h b/api/test/create_time_controller.h index 5572ef0989..3e632593f1 100644 --- a/api/test/create_time_controller.h +++ b/api/test/create_time_controller.h @@ -12,7 +12,6 @@ #include -#include "api/peer_connection_interface.h" #include "api/test/time_controller.h" namespace webrtc { @@ -20,14 +19,6 @@ namespace webrtc { // Creates a time controller that runs in simulated time. std::unique_ptr CreateSimulatedTimeController(); -// Adjusts media `deps` to use clock `time_controller` provides, fills media -// related dependencies, and enables media support for a PeerConnectionFactory -// created from `deps`. -[[deprecated("Inject Clock through the deps.env instead.")]] -void EnableMediaWithDefaultsAndTimeController( - TimeController& time_controller, - PeerConnectionFactoryDependencies& deps); - } // namespace webrtc #endif // API_TEST_CREATE_TIME_CONTROLLER_H_ From 8cf10372bebdc6ccfaf4e47bebc567208fdd30f0 Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Mon, 12 May 2025 10:54:50 +0200 Subject: [PATCH 139/292] Delete deprecated PeerConnectionFactoryDependendencies::audio_processing Bug: webrtc:369904700 Change-Id: If16c4c32d1012d3102a817746a61c185672cc0f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389800 Reviewed-by: Harald Alvestrand Commit-Queue: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#44611} --- api/enable_media.cc | 11 ++++------ api/enable_media_with_defaults.cc | 7 ++---- api/peer_connection_interface.cc | 9 ++------ api/peer_connection_interface.h | 3 --- pc/peer_connection_factory_unittest.cc | 22 +------------------ .../RTCPeerConnectionFactory.mm | 6 ----- 6 files changed, 9 insertions(+), 49 deletions(-) diff --git a/api/enable_media.cc b/api/enable_media.cc index 25b1bafc9c..81cb890185 100644 --- a/api/enable_media.cc +++ b/api/enable_media.cc @@ -41,13 +41,10 @@ class MediaFactoryImpl : public MediaFactory { std::unique_ptr CreateMediaEngine( const Environment& env, PeerConnectionFactoryDependencies& deps) override { - absl_nullable scoped_refptr audio_processing = - deps.audio_processing_builder != nullptr - ? std::move(deps.audio_processing_builder)->Build(env) -#pragma clang diagnostic push -#pragma clang diagnostic ignored "-Wdeprecated-declarations" - : std::move(deps.audio_processing); -#pragma clang diagnostic pop + absl_nullable scoped_refptr audio_processing; + if (deps.audio_processing_builder != nullptr) { + audio_processing = std::move(deps.audio_processing_builder)->Build(env); + } auto audio_engine = std::make_unique( env, std::move(deps.adm), std::move(deps.audio_encoder_factory), diff --git a/api/enable_media_with_defaults.cc b/api/enable_media_with_defaults.cc index 83d6f172f9..e55e1710e3 100644 --- a/api/enable_media_with_defaults.cc +++ b/api/enable_media_with_defaults.cc @@ -30,11 +30,8 @@ void EnableMediaWithDefaults(PeerConnectionFactoryDependencies& deps) { if (deps.audio_decoder_factory == nullptr) { deps.audio_decoder_factory = CreateBuiltinAudioDecoderFactory(); } -#pragma clang diagnostic push -#pragma clang diagnostic ignored "-Wdeprecated-declarations" - if (deps.audio_processing == nullptr && -#pragma clang diagnostic pop - deps.audio_processing_builder == nullptr) { + + if (deps.audio_processing_builder == nullptr) { deps.audio_processing_builder = std::make_unique(); } diff --git a/api/peer_connection_interface.cc b/api/peer_connection_interface.cc index 947ecc62a6..1af738a9d9 100644 --- a/api/peer_connection_interface.cc +++ b/api/peer_connection_interface.cc @@ -45,21 +45,16 @@ PeerConnectionDependencies::PeerConnectionDependencies( PeerConnectionObserver* observer_in) : observer(observer_in) {} -// TODO(bugs.webrtc.org/12598: remove pragma once async_resolver_factory -// is removed from PeerConnectionDependencies -#pragma clang diagnostic push -#pragma clang diagnostic ignored "-Wdeprecated-declarations" PeerConnectionDependencies::PeerConnectionDependencies( PeerConnectionDependencies&&) = default; -#pragma clang diagnostic pop PeerConnectionDependencies::~PeerConnectionDependencies() = default; PeerConnectionFactoryDependencies::PeerConnectionFactoryDependencies() = default; -// TODO: bugs.webrtc.org/369904700 - remove pragma once `audio_processing` -// is removed from PeerConnectionFactoryDependencies. +// Allow move constructor to move deprecated members. Pragma can be removed +// when there are no deprecated depedencies at the moment. #pragma clang diagnostic push #pragma clang diagnostic ignored "-Wdeprecated-declarations" PeerConnectionFactoryDependencies::PeerConnectionFactoryDependencies( diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h index b02e055029..1cd3379cba 100644 --- a/api/peer_connection_interface.h +++ b/api/peer_connection_interface.h @@ -1472,9 +1472,6 @@ struct RTC_EXPORT PeerConnectionFactoryDependencies final { scoped_refptr audio_encoder_factory; scoped_refptr audio_decoder_factory; scoped_refptr audio_mixer; - // TODO: bugs.webrtc.org/369904700 - Delete `audio_processing` in favor - // of `audio_processing_builder`. - [[deprecated]] scoped_refptr audio_processing; std::unique_ptr audio_processing_builder; std::unique_ptr audio_frame_processor; std::unique_ptr video_encoder_factory; diff --git a/pc/peer_connection_factory_unittest.cc b/pc/peer_connection_factory_unittest.cc index bde4d89d27..ec81afa1f6 100644 --- a/pc/peer_connection_factory_unittest.cc +++ b/pc/peer_connection_factory_unittest.cc @@ -764,7 +764,7 @@ TEST(PeerConnectionFactoryDependenciesTest, UsesPacketSocketFactory) { } TEST(PeerConnectionFactoryDependenciesTest, - CreatesAudioProcessingWithProvidedFactory) { + CreatesAudioProcessingWithProvidedBuilder) { auto ap_factory = std::make_unique(); auto audio_processing = make_ref_counted>(); // Validate that provided audio_processing is used by expecting that a request @@ -783,25 +783,5 @@ TEST(PeerConnectionFactoryDependenciesTest, pcf->StartAecDump(nullptr, 24'242); } -TEST(PeerConnectionFactoryDependenciesTest, UsesAudioProcessingWhenProvided) { - // Test legacy way of providing audio_processing. - // TODO: bugs.webrtc.org/369904700 - Delete this test when webrtc users no - // longer set PeerConnectionFactoryDependencies::audio_processing. - auto audio_processing = make_ref_counted>(); - EXPECT_CALL(*audio_processing, CreateAndAttachAecDump(A(), 24'242, _)); - - PeerConnectionFactoryDependencies pcf_dependencies; - pcf_dependencies.adm = FakeAudioCaptureModule::Create(); -#pragma clang diagnostic push -#pragma clang diagnostic ignored "-Wdeprecated-declarations" - pcf_dependencies.audio_processing = std::move(audio_processing); -#pragma clang diagnostic pop - EnableMediaWithDefaults(pcf_dependencies); - - scoped_refptr pcf = - CreateModularPeerConnectionFactory(std::move(pcf_dependencies)); - pcf->StartAecDump(nullptr, 24'242); -} - } // namespace } // namespace webrtc diff --git a/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm b/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm index 54618fba63..250ba40cd7 100644 --- a/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm +++ b/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm @@ -238,12 +238,6 @@ - (instancetype)initWithNoMedia { - (instancetype)initWithMediaAndDependencies: (webrtc::PeerConnectionFactoryDependencies &)dependencies { -#pragma clang diagnostic push -#pragma clang diagnostic ignored "-Wdeprecated-declarations" - // audio_processing_builder should be used instead in new code. - RTC_CHECK(dependencies.audio_processing == nullptr); -#pragma clang diagnostic pop - #ifndef WEBRTC_EXCLUDE_AUDIO_PROCESSING_MODULE if (dependencies.audio_processing_builder == nullptr) { dependencies.audio_processing_builder = From 66fdc4f27672f68ab303dd35c662007635aa4256 Mon Sep 17 00:00:00 2001 From: Sunggook Chue Date: Mon, 12 May 2025 18:17:01 -0700 Subject: [PATCH 140/292] Reset current screen ID when IsScreenValid fails. The current implementation doesn't reset the screen ID-related member variables when assigning a new source ID fails. This can result in undefined behavior, including the possibility of displaying incorrect content in the shared window or screen. This fix ensures the screen ID is reset to invalid values when validation fails. Bug: chromium:347991512 Change-Id: I17e4a6b76caf436e926fb4968144875db03291e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391100 Reviewed-by: Ilya Nikolaevskiy Reviewed-by: Alexander Cooper Commit-Queue: Sunggook Chue Cr-Commit-Position: refs/heads/main@{#44612} --- .../desktop_capture/win/screen_capturer_win_gdi.cc | 3 +++ .../win/screen_capturer_win_magnifier.cc | 12 +++++++----- 2 files changed, 10 insertions(+), 5 deletions(-) diff --git a/modules/desktop_capture/win/screen_capturer_win_gdi.cc b/modules/desktop_capture/win/screen_capturer_win_gdi.cc index ae6fb1da5e..9e65b7fa41 100644 --- a/modules/desktop_capture/win/screen_capturer_win_gdi.cc +++ b/modules/desktop_capture/win/screen_capturer_win_gdi.cc @@ -117,6 +117,9 @@ bool ScreenCapturerWinGdi::SelectSource(SourceId id) { if (valid) { current_screen_id_ = id; current_device_key_ = device_key; + } else { + current_screen_id_ = kFullDesktopScreenId; + current_device_key_ = std::nullopt; } return valid; } diff --git a/modules/desktop_capture/win/screen_capturer_win_magnifier.cc b/modules/desktop_capture/win/screen_capturer_win_magnifier.cc index 6d2384eb00..bb445129fc 100644 --- a/modules/desktop_capture/win/screen_capturer_win_magnifier.cc +++ b/modules/desktop_capture/win/screen_capturer_win_magnifier.cc @@ -136,13 +136,15 @@ bool ScreenCapturerWinMagnifier::GetSourceList(SourceList* sources) { bool ScreenCapturerWinMagnifier::SelectSource(SourceId id) { std::wstring device_key; - if (IsScreenValid(id, &device_key)) { - current_device_key_ = device_key; + bool valid = IsScreenValid(id, &device_key); + if (valid) { current_screen_id_ = id; - return true; + current_device_key_ = device_key; + } else { + current_screen_id_ = kFullDesktopScreenId; + current_device_key_ = std::nullopt; } - - return false; + return valid; } void ScreenCapturerWinMagnifier::SetExcludedWindow(WindowId excluded_window) { From 89012b7b73a4a5e8a72538ae08f376ed28e81462 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Tue, 13 May 2025 11:04:08 -0700 Subject: [PATCH 141/292] Roll chromium_revision 86f00502dd..73f6bd2d9c (1459405:1459577) Change log: https://chromium.googlesource.com/chromium/src/+log/86f00502dd..73f6bd2d9c Full diff: https://chromium.googlesource.com/chromium/src/+/86f00502dd..73f6bd2d9c Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/336cce9dcf..1245a2ae7c * src/build: https://chromium.googlesource.com/chromium/src/build/+log/9c19bf6fd8..f70a1fd60d * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/feb5fd2c3d..5309e7c44d * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/3d4f61f5e0..2cc288b951 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/c212732796..f10274682c * src/third_party/androidx/cipd: NgN4G3kZkU23fj174TE4RsZAis3Apk2LfPTjb8wllxwC..TIfHUjgo7_UN847vl82ZF4Pku0QVun5z2gYKRVxOL4EC * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/4402fd7953..060d748587 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/b0e95ba77d..39772975bf DEPS diff: https://chromium.googlesource.com/chromium/src/+/86f00502dd..73f6bd2d9c/DEPS No update to Clang. BUG=None Change-Id: I699e215c544f9322cc0e1a5d2e72f26c6eb53fbe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391161 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44613} --- DEPS | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/DEPS b/DEPS index 5c84f06c22..686eba468b 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '86f00502ddde0c74aa718eba555dc4961c1c325d', + 'chromium_revision': '73f6bd2d9c91d78e0435c846427b7b49515c67ed', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@336cce9dcf88c86858f4ff87f446d697952e48ad', + 'https://chromium.googlesource.com/chromium/src/base@1245a2ae7c0240d17922b20e68047f0d99481547', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@9c19bf6fd8ddb59c6cbb1c3498cd2100b5983113', + 'https://chromium.googlesource.com/chromium/src/build@f70a1fd60d242ef6268e59d610db848b273de1c7', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@077a66f30fcf281b066fafb6dfc60818c238efb6', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@feb5fd2c3d8e44a6d1f41942dc455a2424fc8212', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@5309e7c44d9e8738160bd2644b3915edd1f3d147', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@3d4f61f5e05d19364fe407b6706702d85e26f489', + 'https://chromium.googlesource.com/chromium/src/testing@2cc288b9518ffaae475d2a65717cdd6d90c3e8ec', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@c2127327968c62d37d7c039e218b935f6a93637c', + 'https://chromium.googlesource.com/chromium/src/third_party@f10274682ce14b084b2353fd8b95b0e8b8a40d4d', 'src/buildtools/linux64': { 'packages': [ @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '4402fd7953d5dedf65659e99ab80404f6e94a004', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '060d748587295617a22efdc6e6336997deb250cb', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@b0e95ba77dbdf0db4946f1316664fdc82e4bef30', + 'https://chromium.googlesource.com/chromium/src/tools@39772975bf409ef6170ed8acc7de1942479e2feb', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'NgN4G3kZkU23fj174TE4RsZAis3Apk2LfPTjb8wllxwC', + 'version': 'TIfHUjgo7_UN847vl82ZF4Pku0QVun5z2gYKRVxOL4EC', }, ], 'condition': 'checkout_android and non_git_source', From 6f90b4c26eda396966d71e84905af9c1bfacc7c9 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Tue, 13 May 2025 15:03:52 -0700 Subject: [PATCH 142/292] Roll chromium_revision 73f6bd2d9c..1754f51ab7 (1459577:1459743) Change log: https://chromium.googlesource.com/chromium/src/+log/73f6bd2d9c..1754f51ab7 Full diff: https://chromium.googlesource.com/chromium/src/+/73f6bd2d9c..1754f51ab7 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/1245a2ae7c..7ac8b768f3 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/f70a1fd60d..526c6cbd6a * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/5309e7c44d..1a4df0abd3 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/2cc288b951..3fc05696bf * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/f10274682c..feff0f7aa3 * src/third_party/android_build_tools/error_prone/cipd: FLtgIzcLepAc87BaSXIELZIJPT4yxnr8K3MmFaEHrK8C..bd18Z8-1Y85m4C8xYA3z1J_VIhAaE2qRbkrBzn4_200C * src/third_party/android_build_tools/lint/cipd: QltgzNfGb4l4ekv1_GjIWtuew1hrBof1WVy2B8zrkYwC..7xZFl5M3w3LIJvhKsJYDQoTh5JZ3vr3I_-VWYXiGJC0C * src/third_party/android_build_tools/manifest_merger/cipd: hVPp69VTDru05sisN9HcrOf67Tk_Mhu6P6bsR24gOrEC..Ym1eWXywKFRvxsgK0yC4bFj9SwZViDqGEPb_w3mAKqMC * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/2a514a51ba..89973806bc * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/83c00a37f4..5da8152ef9 * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/7b1679b14d..9fe4f187ec * src/third_party/kotlin_stdlib/cipd: aHCAN6w_ZZCJL7MOPlDbEieRsiUs6ftT33fCI4bbwVwC..eYKSu9agsqusPORDTTSPHcg632V_j9kIvvufVJiULXIC * src/third_party/libunwind/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git/+log/2bd5f3cae1..8575f4ae4f * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/060d748587..9b47377deb * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/39772975bf..6ef9fa6f6a DEPS diff: https://chromium.googlesource.com/chromium/src/+/73f6bd2d9c..1754f51ab7/DEPS No update to Clang. BUG=None Change-Id: I1c75307fefd50bae807756f49a87707baa502639 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391263 Commit-Queue: Autoroller Bot-Commit: Autoroller Cr-Commit-Position: refs/heads/main@{#44614} --- DEPS | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) diff --git a/DEPS b/DEPS index 686eba468b..619bc175fc 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '73f6bd2d9c91d78e0435c846427b7b49515c67ed', + 'chromium_revision': '1754f51ab7ace5eb256cdc915b5a962c840ba5af', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@1245a2ae7c0240d17922b20e68047f0d99481547', + 'https://chromium.googlesource.com/chromium/src/base@7ac8b768f3161dfff6a9c66587f7fef5ffa28443', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@f70a1fd60d242ef6268e59d610db848b273de1c7', + 'https://chromium.googlesource.com/chromium/src/build@526c6cbd6a32e5952e0cc09ab35d33ea4d883907', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@077a66f30fcf281b066fafb6dfc60818c238efb6', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@5309e7c44d9e8738160bd2644b3915edd1f3d147', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@1a4df0abd3385b3b735eca08d31ec867f5045f88', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@2cc288b9518ffaae475d2a65717cdd6d90c3e8ec', + 'https://chromium.googlesource.com/chromium/src/testing@3fc05696bfb2f482f61765391c64fab56a86b085', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@f10274682ce14b084b2353fd8b95b0e8b8a40d4d', + 'https://chromium.googlesource.com/chromium/src/third_party@feff0f7aa3506bcd11a1db1320701a1b57b5cd86', 'src/buildtools/linux64': { 'packages': [ @@ -331,7 +331,7 @@ deps = { 'src/third_party/llvm-libc/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@a0ab545a0eaa7a8cd7154025d1902904e1800cfc', 'src/third_party/libunwind/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@2bd5f3cae13e8ad6727d0a77a2ec9cdc6b06becb', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@8575f4ae4fcf8892938bd9766cf1a5c90a0ed04e', 'src/third_party/test_fonts/test_fonts': { 'dep_type': 'gcs', @@ -427,7 +427,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_build_tools/error_prone', - 'version': 'FLtgIzcLepAc87BaSXIELZIJPT4yxnr8K3MmFaEHrK8C', + 'version': 'bd18Z8-1Y85m4C8xYA3z1J_VIhAaE2qRbkrBzn4_200C', }, ], 'condition': 'checkout_android', @@ -449,7 +449,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_build_tools/lint', - 'version': 'QltgzNfGb4l4ekv1_GjIWtuew1hrBof1WVy2B8zrkYwC', + 'version': '7xZFl5M3w3LIJvhKsJYDQoTh5JZ3vr3I_-VWYXiGJC0C', }, ], 'condition': 'checkout_android and non_git_source', @@ -480,11 +480,11 @@ deps = { }, 'src/third_party/boringssl/src': - 'https://boringssl.googlesource.com/boringssl.git@2a514a51baebd5a232fc64f7b082f7a8b28cd29d', + 'https://boringssl.googlesource.com/boringssl.git@89973806bc2ef652189e157f2736a7d32229c404', 'src/third_party/breakpad/breakpad': 'https://chromium.googlesource.com/breakpad/breakpad.git@9eeece27b727d2ffac4b7042bb69d7c140e58312', 'src/third_party/catapult': - 'https://chromium.googlesource.com/catapult.git@83c00a37f40a93932204f684172a13c62a379e7e', + 'https://chromium.googlesource.com/catapult.git@5da8152ef93c63ea38e38c3ce1d9791b7e529b4d', 'src/third_party/ced/src': { 'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5', }, @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@7b1679b14d0292b5373bb270cf8c6390ae93dd9f', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@9fe4f187ec4747e17d57fd53167117282a00a55c', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -563,7 +563,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/kotlin_stdlib', - 'version': 'aHCAN6w_ZZCJL7MOPlDbEieRsiUs6ftT33fCI4bbwVwC', + 'version': 'eYKSu9agsqusPORDTTSPHcg632V_j9kIvvufVJiULXIC', }, ], 'condition': 'checkout_android', @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '060d748587295617a22efdc6e6336997deb250cb', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '9b47377deba086351a83567e04cc06f8c18996d0', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@39772975bf409ef6170ed8acc7de1942479e2feb', + 'https://chromium.googlesource.com/chromium/src/tools@6ef9fa6f6a128a79f97f1c8ea0fdbb260a852817', 'src/third_party/espresso': { 'packages': [ @@ -706,7 +706,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_build_tools/manifest_merger', - 'version': 'hVPp69VTDru05sisN9HcrOf67Tk_Mhu6P6bsR24gOrEC', + 'version': 'Ym1eWXywKFRvxsgK0yC4bFj9SwZViDqGEPb_w3mAKqMC', }, ], 'condition': 'checkout_android', From e245291d4135c7eda28c24d1a19d9e3ee315a98c Mon Sep 17 00:00:00 2001 From: Tom Anderson Date: Tue, 13 May 2025 21:42:14 +0000 Subject: [PATCH 143/292] [X11] Fix memory leak when presenting "Share your screen" dialog Fixed: chromium:40686790 Change-Id: Ia49ff5433c47908f1dac6ea12cb5e13e7376a850 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391020 Reviewed-by: Alexander Cooper Reviewed-by: Mark Foltz Commit-Queue: Thomas Anderson Cr-Commit-Position: refs/heads/main@{#44615} --- .../linux/x11/x_server_pixel_buffer.cc | 13 ++++++++----- .../linux/x11/x_server_pixel_buffer.h | 1 + 2 files changed, 9 insertions(+), 5 deletions(-) diff --git a/modules/desktop_capture/linux/x11/x_server_pixel_buffer.cc b/modules/desktop_capture/linux/x11/x_server_pixel_buffer.cc index e0b334dab1..7ab01cae04 100644 --- a/modules/desktop_capture/linux/x11/x_server_pixel_buffer.cc +++ b/modules/desktop_capture/linux/x11/x_server_pixel_buffer.cc @@ -164,6 +164,10 @@ void XServerPixelBuffer::Release() { void XServerPixelBuffer::ReleaseSharedMemorySegment() { if (!shm_segment_info_) return; + if (xshm_attached_) { + XShmDetach(display_, shm_segment_info_); + xshm_attached_ = false; + } if (shm_segment_info_->shmaddr != nullptr) shmdt(shm_segment_info_->shmaddr); if (shm_segment_info_->shmid != -1) @@ -211,7 +215,6 @@ void XServerPixelBuffer::InitShm(const XWindowAttributes& attributes) { return; } - bool using_shm = false; shm_segment_info_ = new XShmSegmentInfo; shm_segment_info_->shmid = -1; shm_segment_info_->shmaddr = nullptr; @@ -230,11 +233,11 @@ void XServerPixelBuffer::InitShm(const XWindowAttributes& attributes) { x_shm_image_->data = shm_segment_info_->shmaddr; XErrorTrap error_trap(display_); - using_shm = XShmAttach(display_, shm_segment_info_); + xshm_attached_ = XShmAttach(display_, shm_segment_info_); XSync(display_, False); if (error_trap.GetLastErrorAndDisable() != 0) - using_shm = false; - if (using_shm) { + xshm_attached_ = false; + if (xshm_attached_) { RTC_LOG(LS_VERBOSE) << "Using X shared memory segment " << shm_segment_info_->shmid; } @@ -245,7 +248,7 @@ void XServerPixelBuffer::InitShm(const XWindowAttributes& attributes) { } } - if (!using_shm) { + if (!xshm_attached_) { RTC_LOG(LS_WARNING) << "Not using shared memory. Performance may be degraded."; ReleaseSharedMemorySegment(); diff --git a/modules/desktop_capture/linux/x11/x_server_pixel_buffer.h b/modules/desktop_capture/linux/x11/x_server_pixel_buffer.h index 38af3a3e76..1ccf8af971 100644 --- a/modules/desktop_capture/linux/x11/x_server_pixel_buffer.h +++ b/modules/desktop_capture/linux/x11/x_server_pixel_buffer.h @@ -80,6 +80,7 @@ class XServerPixelBuffer { XImage* x_shm_image_ = nullptr; Pixmap shm_pixmap_ = 0; GC shm_gc_ = nullptr; + bool xshm_attached_ = false; bool xshm_get_image_succeeded_ = false; std::vector icc_profile_; }; From d1873d6135b3f3d24959a703eade3fe33ed52750 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Tue, 13 May 2025 21:04:05 -0700 Subject: [PATCH 144/292] Roll chromium_revision 1754f51ab7..274243632d (1459743:1459868) Change log: https://chromium.googlesource.com/chromium/src/+log/1754f51ab7..274243632d Full diff: https://chromium.googlesource.com/chromium/src/+/1754f51ab7..274243632d Changed dependencies * siso_version: git_revision:04b22e60e2aae70fa9d51141c91bf4554d8c5239..git_revision:30ea210712239b874febcadf661f118fc2c0f6d7 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/526c6cbd6a..e42f899ae7 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/1a4df0abd3..d6c29dd3c5 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/3fc05696bf..e93dc3dea2 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/feff0f7aa3..3dcdf2820d * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/5da8152ef9..fd624be2fa * src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/b5f72d2ab6..d33b54c621 * src/third_party/fuzztest/src: https://chromium.googlesource.com/external/github.com/google/fuzztest.git/+log/1f21600f67..f03aafb751 * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/9fe4f187ec..94198a4a9c * src/third_party/libc++abi/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git/+log/5a49db9990..069652e9cb * src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/a0ab545a0e..27ee073016 * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/9b47377deb..63f1ba4f68 * src/third_party/r8/d8/cipd: bA3htCoEd_EArHekDGQSNpmBzQrcby2ioG6SFyl3AtwC..QhYGRVpYYKZmt3f_Zb2HoJ9LIBEnWaeeLXRNei47Z30C * src/third_party/siso/cipd: git_revision:04b22e60e2aae70fa9d51141c91bf4554d8c5239..git_revision:30ea210712239b874febcadf661f118fc2c0f6d7 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/6ef9fa6f6a..635b64025b DEPS diff: https://chromium.googlesource.com/chromium/src/+/1754f51ab7..274243632d/DEPS No update to Clang. BUG=None Change-Id: Icdb0ab5f9aca61f4d6391789ec784e70ad3e96d0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391340 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44616} --- DEPS | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/DEPS b/DEPS index 619bc175fc..bae4a3a766 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '1754f51ab7ace5eb256cdc915b5a962c840ba5af', + 'chromium_revision': '274243632d0c927460706ca04f3c3e605d43b9c2', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -53,7 +53,7 @@ vars = { # reclient CIPD package version 'reclient_version': 're_client_version:0.177.1.e58c0145-gomaip', # siso CIPD package version. - 'siso_version': 'git_revision:04b22e60e2aae70fa9d51141c91bf4554d8c5239', + 'siso_version': 'git_revision:30ea210712239b874febcadf661f118fc2c0f6d7', # ninja CIPD package. 'ninja_package': 'infra/3pp/tools/ninja/', @@ -75,7 +75,7 @@ deps = { 'src/base': 'https://chromium.googlesource.com/chromium/src/base@7ac8b768f3161dfff6a9c66587f7fef5ffa28443', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@526c6cbd6a32e5952e0cc09ab35d33ea4d883907', + 'https://chromium.googlesource.com/chromium/src/build@e42f899ae75d20881f14e06fd324041312fe6a83', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@077a66f30fcf281b066fafb6dfc60818c238efb6', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@1a4df0abd3385b3b735eca08d31ec867f5045f88', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@d6c29dd3c53ab982e2fcdadb464a42f820be1de7', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@3fc05696bfb2f482f61765391c64fab56a86b085', + 'https://chromium.googlesource.com/chromium/src/testing@e93dc3dea230f64eab7e97113e02b942af35962e', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@feff0f7aa3506bcd11a1db1320701a1b57b5cd86', + 'https://chromium.googlesource.com/chromium/src/third_party@3dcdf2820d8a9fa70f479fa0862e31e492d19105', 'src/buildtools/linux64': { 'packages': [ @@ -323,13 +323,13 @@ deps = { 'src/third_party/clang-format/script': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@37f6e68a107df43b7d7e044fd36a13cbae3413f2', 'src/third_party/compiler-rt/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@b5f72d2ab6190b6cea33512880e6cbfe3648d513', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@d33b54c621474e2f2642af28d42a15227b1602cb', 'src/third_party/libc++/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@a01c02c9d4acbdae3b7e8a2f3ee58579a9c29f96', 'src/third_party/libc++abi/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@5a49db9990ee2ecee2bc7340be9756c0455bde6f', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@069652e9cbca8e3b5e2371057a206f91f2f710f6', 'src/third_party/llvm-libc/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@a0ab545a0eaa7a8cd7154025d1902904e1800cfc', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@27ee0730164e6f101985f057c6ad23a9ba7710f6', 'src/third_party/libunwind/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@8575f4ae4fcf8892938bd9766cf1a5c90a0ed04e', @@ -484,7 +484,7 @@ deps = { 'src/third_party/breakpad/breakpad': 'https://chromium.googlesource.com/breakpad/breakpad.git@9eeece27b727d2ffac4b7042bb69d7c140e58312', 'src/third_party/catapult': - 'https://chromium.googlesource.com/catapult.git@5da8152ef93c63ea38e38c3ce1d9791b7e529b4d', + 'https://chromium.googlesource.com/catapult.git@fd624be2faba11f03d2cf94e095b86707758f062', 'src/third_party/ced/src': { 'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5', }, @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@9fe4f187ec4747e17d57fd53167117282a00a55c', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@94198a4a9cd62d176f50cdfae9b8d4db6a052011', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -583,7 +583,7 @@ deps = { 'src/third_party/libFuzzer/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@e31b99917861f891308269c36a32363b120126bb', 'src/third_party/fuzztest/src': - 'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@1f21600f675e2bb8972acf57bfae6ff3416d37bb', + 'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@f03aafb7516050ea73f617bf969f03eac641aefc', 'src/third_party/libjpeg_turbo': 'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@e14cbfaa85529d47f9f55b0f104a579c1061f9ad', 'src/third_party/libsrtp': @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '9b47377deba086351a83567e04cc06f8c18996d0', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '63f1ba4f684943307b6d99d0669b1d180bc3b6f5', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -645,7 +645,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/r8', - 'version': 'bA3htCoEd_EArHekDGQSNpmBzQrcby2ioG6SFyl3AtwC', + 'version': 'QhYGRVpYYKZmt3f_Zb2HoJ9LIBEnWaeeLXRNei47Z30C', }, ], 'condition': 'checkout_android', @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@6ef9fa6f6a128a79f97f1c8ea0fdbb260a852817', + 'https://chromium.googlesource.com/chromium/src/tools@635b64025b17f341f6106d520736829bf76e544f', 'src/third_party/espresso': { 'packages': [ From 6f4dad8679ea0fd18899048ecbafc3276f31882e Mon Sep 17 00:00:00 2001 From: webrtc-version-updater Date: Tue, 13 May 2025 21:08:10 -0700 Subject: [PATCH 145/292] Update WebRTC code version (2025-05-14T04:07:42). Bug: None Change-Id: I7135233b7fc14f1e8f575ec6cb81e5426787aca6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391163 Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Cr-Commit-Position: refs/heads/main@{#44617} --- call/version.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/call/version.cc b/call/version.cc index 64ada0b45e..e1f43fe5b1 100644 --- a/call/version.cc +++ b/call/version.cc @@ -13,7 +13,7 @@ namespace webrtc { // The timestamp is always in UTC. -const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-13T04:03:41"; +const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-14T04:07:42"; void LoadWebRTCVersionInRegister() { // Using volatile to instruct the compiler to not optimize `p` away even From 3fe1ba3f5200c4fd04a7dc8e9714cd62984544f4 Mon Sep 17 00:00:00 2001 From: Daniel White Date: Tue, 13 May 2025 11:14:44 -0400 Subject: [PATCH 146/292] [iOS] Update RTCCameraPreviewView to conform to iOS 17 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit In iOS 17, the video orientation API is deprecated. This CL updates the RTCCameraPreviewView file to use the video rotation angle API instead. This file is being updated now to support Chrome for iOS effort to raise its minimum deployment target to iOS 17. In addition, this CL is a follow-up to the linked reverted CL that fixes the incorrect build guard that gated the deprecated code. Change-Id: I82feadbdca227259856948f0b683478ec52535cb Bug: chromium:348649285 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391240 Reviewed-by: Mirko Bonadei Commit-Queue: Kári Helgason Reviewed-by: Kári Helgason Cr-Commit-Position: refs/heads/main@{#44618} --- sdk/objc/helpers/RTCCameraPreviewView.m | 72 +++++++++++++++++-------- 1 file changed, 50 insertions(+), 22 deletions(-) diff --git a/sdk/objc/helpers/RTCCameraPreviewView.m b/sdk/objc/helpers/RTCCameraPreviewView.m index 92c3cd2635..0ebdcb5d3c 100644 --- a/sdk/objc/helpers/RTCCameraPreviewView.m +++ b/sdk/objc/helpers/RTCCameraPreviewView.m @@ -80,6 +80,54 @@ - (void)orientationChanged:(NSNotification *)notification { } - (void)setCorrectVideoOrientation { + if (@available(iOS 17, *)) { + [self modifyVideoAngle]; + return; + } + + [self modifyVideoOrientation]; +} + +#pragma mark - Private + +- (void)addOrientationObserver { + [[NSNotificationCenter defaultCenter] + addObserver:self + selector:@selector(orientationChanged:) + name:UIDeviceOrientationDidChangeNotification + object:nil]; +} + +- (void)removeOrientationObserver { + [[NSNotificationCenter defaultCenter] + removeObserver:self + name:UIDeviceOrientationDidChangeNotification + object:nil]; +} + +- (AVCaptureVideoPreviewLayer *)previewLayer { + return (AVCaptureVideoPreviewLayer *)self.layer; +} + +- (void)modifyVideoAngle API_AVAILABLE(ios(17.0)) { + AVCaptureDeviceInput* captureSessionInput = + _captureSession.inputs.firstObject; + AVCaptureDevice* camera = captureSessionInput.device; + AVCaptureVideoPreviewLayer* previewLayer = [self previewLayer]; + AVCaptureConnection* videoConnection = previewLayer.connection; + AVCaptureDeviceRotationCoordinator* rotationCoordiantor = + [[AVCaptureDeviceRotationCoordinator alloc] + initWithDevice:camera + previewLayer:previewLayer]; + CGFloat angle = + rotationCoordiantor.videoRotationAngleForHorizonLevelCapture; + if ([videoConnection isVideoRotationAngleSupported:angle]) { + [videoConnection setVideoRotationAngle:angle]; + } +} + +#if !defined(__IPHONE_17_0) || __IPHONE_OS_VERSION_MIN_REQUIRED < __IPHONE_17_0 +- (void)modifyVideoOrientation { // Get current device orientation. UIDeviceOrientation deviceOrientation = [UIDevice currentDevice].orientation; AVCaptureVideoPreviewLayer *previewLayer = [self previewLayer]; @@ -104,26 +152,6 @@ - (void)setCorrectVideoOrientation { // orientation. } } +#endif -#pragma mark - Private - -- (void)addOrientationObserver { - [[NSNotificationCenter defaultCenter] - addObserver:self - selector:@selector(orientationChanged:) - name:UIDeviceOrientationDidChangeNotification - object:nil]; -} - -- (void)removeOrientationObserver { - [[NSNotificationCenter defaultCenter] - removeObserver:self - name:UIDeviceOrientationDidChangeNotification - object:nil]; -} - -- (AVCaptureVideoPreviewLayer *)previewLayer { - return (AVCaptureVideoPreviewLayer *)self.layer; -} - -@end +@end \ No newline at end of file From 1b7ddd4cf9916da45f07c08909b6b08e1c27773d Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Tue, 13 May 2025 15:39:25 +0200 Subject: [PATCH 147/292] Propagate field trials into AudioDeviceIOS MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Bug: webrtc:413413572 Change-Id: I96fb90f4bf0304045fc3141c96955fd3f2df043a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389220 Commit-Queue: Danil Chapovalov Reviewed-by: Henrik Andreassson Reviewed-by: Kári Helgason Cr-Commit-Position: refs/heads/main@{#44619} --- modules/audio_device/audio_device_impl.cc | 10 ++-- modules/audio_device/audio_device_impl.h | 2 +- .../audio_device/include/test_audio_device.cc | 2 +- sdk/BUILD.gn | 16 ++++-- .../RTCPeerConnectionFactory.mm | 21 ++++---- ...nectionFactoryBuilder+DefaultComponents.mm | 4 +- sdk/objc/native/api/audio_device_module.h | 20 ++++--- sdk/objc/native/api/audio_device_module.mm | 52 +++++++++++-------- .../native/api/objc_audio_device_module.h | 5 +- .../native/api/objc_audio_device_module.mm | 8 +-- sdk/objc/native/src/audio/audio_device_ios.h | 4 ++ sdk/objc/native/src/audio/audio_device_ios.mm | 8 +-- .../src/audio/audio_device_module_ios.h | 5 +- .../src/audio/audio_device_module_ios.mm | 22 +++++--- sdk/objc/native/src/objc_audio_device.h | 5 +- sdk/objc/native/src/objc_audio_device.mm | 15 +++--- .../unittests/RTCAudioDeviceModule_xctest.mm | 6 ++- sdk/objc/unittests/RTCAudioDevice_xctest.mm | 8 ++- 18 files changed, 132 insertions(+), 81 deletions(-) diff --git a/modules/audio_device/audio_device_impl.cc b/modules/audio_device/audio_device_impl.cc index da98848b96..585d63776a 100644 --- a/modules/audio_device/audio_device_impl.cc +++ b/modules/audio_device/audio_device_impl.cc @@ -102,7 +102,7 @@ AudioDeviceModuleImpl::Create(const Environment& env, AudioLayer audio_layer) { } // Create the platform-dependent implementation. - if (audio_device->CreatePlatformSpecificObjects() == -1) { + if (audio_device->CreatePlatformSpecificObjects(env) == -1) { return nullptr; } @@ -166,7 +166,8 @@ int32_t AudioDeviceModuleImpl::CheckPlatform() { return 0; } -int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects() { +int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects( + [[maybe_unused]] const Environment& env) { RTC_LOG(LS_INFO) << __FUNCTION__; if (audio_device_ != nullptr) { RTC_LOG(LS_INFO) << "Reusing provided audio device"; @@ -241,10 +242,11 @@ int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects() { // iOS ADM implementation. #if defined(WEBRTC_IOS) if (audio_layer == kPlatformDefaultAudio) { - audio_device_.reset(new ios_adm::AudioDeviceIOS( + audio_device_ = std::make_unique( + env, /*bypass_voice_processing=*/false, /*muted_speech_event_handler=*/nullptr, - /*render_error_handler=*/nullptr)); + /*render_error_handler=*/nullptr); RTC_LOG(LS_INFO) << "iPhone Audio APIs will be utilized."; } // END #if defined(WEBRTC_IOS) diff --git a/modules/audio_device/audio_device_impl.h b/modules/audio_device/audio_device_impl.h index 609daf56cf..c7a229d2d9 100644 --- a/modules/audio_device/audio_device_impl.h +++ b/modules/audio_device/audio_device_impl.h @@ -61,7 +61,7 @@ class AudioDeviceModuleImpl : public AudioDeviceModuleForTest { ~AudioDeviceModuleImpl() override; int32_t CheckPlatform(); - int32_t CreatePlatformSpecificObjects(); + int32_t CreatePlatformSpecificObjects(const Environment& env); int32_t AttachAudioBuffer(); // Retrieve the currently utilized audio layer diff --git a/modules/audio_device/include/test_audio_device.cc b/modules/audio_device/include/test_audio_device.cc index f6d5386271..3b58ff21da 100644 --- a/modules/audio_device/include/test_audio_device.cc +++ b/modules/audio_device/include/test_audio_device.cc @@ -453,7 +453,7 @@ scoped_refptr TestAudioDeviceModule::Create( } // Create the platform-dependent implementation. - if (audio_device->CreatePlatformSpecificObjects() == -1) { + if (audio_device->CreatePlatformSpecificObjects(env) == -1) { return nullptr; } diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn index c0c8eb4796..c443527044 100644 --- a/sdk/BUILD.gn +++ b/sdk/BUILD.gn @@ -231,6 +231,10 @@ if (is_ios || is_mac) { rtc_library("native_api_audio_device_module") { visibility = [ "*" ] + # TODO: bugs.webrtc.org/413413572 - Remove the poison when users of this + # api provide Environment explicitly. + allow_poison = [ "environment_construction" ] + sources = [ "objc/native/api/audio_device_module.h", "objc/native/api/audio_device_module.mm", @@ -240,7 +244,10 @@ if (is_ios || is_mac) { ":audio_device", ":audio_device_module_error_handler", "../api:make_ref_counted", + "../api:scoped_refptr", "../api/audio:audio_device", + "../api/environment", + "../api/environment:environment_factory", "../modules/audio_device:audio_device_generic", "../rtc_base:checks", "../rtc_base:logging", @@ -306,8 +313,8 @@ if (is_ios || is_mac) { "../api:scoped_refptr", "../api:sequence_checker", "../api/audio:audio_device", + "../api/environment", "../api/task_queue", - "../api/task_queue:default_task_queue_factory", "../api/task_queue:pending_task_safety_flag", "../modules/audio_device:audio_device_buffer", "../modules/audio_device:audio_device_config", @@ -485,7 +492,6 @@ if (is_ios || is_mac) { rtc_library("audio_device_objc") { visibility = [ "*" ] - allow_poison = [ "environment_construction" ] sources = [ "objc/native/src/objc_audio_device.h", "objc/native/src/objc_audio_device.mm", @@ -501,8 +507,8 @@ if (is_ios || is_mac) { "../api:scoped_refptr", "../api:sequence_checker", "../api/audio:audio_device", + "../api/environment", "../api/task_queue", - "../api/task_queue:default_task_queue_factory", "../modules/audio_device:audio_device_buffer", "../rtc_base:buffer", "../rtc_base:checks", @@ -528,7 +534,6 @@ if (is_ios || is_mac) { rtc_library("objc_audio_device_module") { visibility = [ "*" ] - allow_poison = [ "environment_construction" ] sources = [ "objc/native/api/objc_audio_device_module.h", "objc/native/api/objc_audio_device_module.mm", @@ -538,7 +543,9 @@ if (is_ios || is_mac) { ":audio_device_api_objc", ":audio_device_objc", "../api:make_ref_counted", + "../api:scoped_refptr", "../api/audio:audio_device", + "../api/environment", "../rtc_base:logging", ] if (is_mac) { @@ -1209,6 +1216,7 @@ if (is_ios || is_mac) { "../api/audio:audio_processing", "../api/audio_codecs:builtin_audio_decoder_factory", "../api/audio_codecs:builtin_audio_encoder_factory", + "../api/environment", "../api/environment:environment_factory", "../api/task_queue:default_task_queue_factory", "../api/video:video_frame", diff --git a/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm b/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm index 250ba40cd7..5df145e34a 100644 --- a/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm +++ b/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm @@ -62,14 +62,6 @@ @implementation RTC_OBJC_TYPE (RTCPeerConnectionFactory) { @synthesize nativeFactory = _nativeFactory; -- (webrtc::scoped_refptr)audioDeviceModule { -#if defined(WEBRTC_IOS) - return webrtc::CreateAudioDeviceModule(); -#else - return nullptr; -#endif -} - - (instancetype)init { webrtc::PeerConnectionFactoryDependencies dependencies; dependencies.audio_encoder_factory = @@ -80,7 +72,10 @@ - (instancetype)init { [[RTC_OBJC_TYPE(RTCVideoEncoderFactoryH264) alloc] init]); dependencies.video_decoder_factory = webrtc::ObjCToNativeVideoDecoderFactory( [[RTC_OBJC_TYPE(RTCVideoDecoderFactoryH264) alloc] init]); - dependencies.adm = [self audioDeviceModule]; + dependencies.env = webrtc::CreateEnvironment(); +#ifdef WEBRTC_IOS + dependencies.adm = webrtc::CreateAudioDeviceModule(*dependencies.env); +#endif return [self initWithMediaAndDependencies:dependencies]; } @@ -105,6 +100,7 @@ - (instancetype)init { return [self initWithNoMedia]; #else webrtc::PeerConnectionFactoryDependencies dependencies; + dependencies.env = webrtc::CreateEnvironment(); dependencies.audio_encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory(); dependencies.audio_decoder_factory = @@ -118,9 +114,12 @@ - (instancetype)init { webrtc::ObjCToNativeVideoDecoderFactory(decoderFactory); } if (audioDevice) { - dependencies.adm = webrtc::CreateAudioDeviceModule(audioDevice); + dependencies.adm = + webrtc::CreateAudioDeviceModule(*dependencies.env, audioDevice); +#ifdef WEBRTC_IOS } else { - dependencies.adm = [self audioDeviceModule]; + dependencies.adm = webrtc::CreateAudioDeviceModule(*dependencies.env); +#endif } return [self initWithMediaAndDependencies:dependencies]; #endif diff --git a/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.mm b/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.mm index e5eeba5992..3e18fd1c57 100644 --- a/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.mm +++ b/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.mm @@ -19,6 +19,7 @@ #include "sdk/objc/native/api/video_encoder_factory.h" #if defined(WEBRTC_IOS) +#include "api/environment/environment_factory.h" #import "sdk/objc/native/api/audio_device_module.h" #endif @@ -42,7 +43,8 @@ + (RTCPeerConnectionFactoryBuilder *)defaultBuilder { [builder setVideoDecoderFactory:std::move(videoDecoderFactory)]; #if defined(WEBRTC_IOS) - [builder setAudioDeviceModule:webrtc::CreateAudioDeviceModule()]; + [builder setAudioDeviceModule:webrtc::CreateAudioDeviceModule( + webrtc::CreateEnvironment())]; #endif return builder; } diff --git a/sdk/objc/native/api/audio_device_module.h b/sdk/objc/native/api/audio_device_module.h index 34f99e85c8..5a8944c674 100644 --- a/sdk/objc/native/api/audio_device_module.h +++ b/sdk/objc/native/api/audio_device_module.h @@ -14,6 +14,8 @@ #include #include "api/audio/audio_device.h" +#include "api/environment/environment.h" +#include "api/scoped_refptr.h" #include "sdk/objc/native/api/audio_device_module_error_handler.h" namespace webrtc { @@ -23,20 +25,26 @@ namespace webrtc { // Warning: Setting `bypass_voice_processing` will have unpredictable // consequences for the audio path in the device. It is not advisable to use in // most scenarios. -webrtc::scoped_refptr CreateAudioDeviceModule( +scoped_refptr CreateAudioDeviceModule( + const Environment& env, bool bypass_voice_processing = false); -// If `muted_speech_event_handler` is exist, audio unit will catch speech -// activity while muted. -webrtc::scoped_refptr CreateMutedDetectAudioDeviceModule( - AudioDeviceModule::MutedSpeechEventHandler muted_speech_event_handler, +[[deprecated("Pass `env` explicitly instead of relying on the default")]] +scoped_refptr CreateAudioDeviceModule( bool bypass_voice_processing = false); // If `muted_speech_event_handler` is exist, audio unit will catch speech // activity while muted. // Provide `error_handler` to receive callbacks on errors such as microphone // init failed or playout start failied. -webrtc::scoped_refptr CreateMutedDetectAudioDeviceModule( +scoped_refptr CreateMutedDetectAudioDeviceModule( + const Environment& env, + AudioDeviceModule::MutedSpeechEventHandler muted_speech_event_handler, + ADMErrorHandler error_handler, + bool bypass_voice_processing = false); + +[[deprecated("Pass `env` explicitly instead of relying on the default")]] +scoped_refptr CreateMutedDetectAudioDeviceModule( AudioDeviceModule::MutedSpeechEventHandler muted_speech_event_handler, ADMErrorHandler error_handler, bool bypass_voice_processing = false); diff --git a/sdk/objc/native/api/audio_device_module.mm b/sdk/objc/native/api/audio_device_module.mm index 898886b592..8325179c94 100644 --- a/sdk/objc/native/api/audio_device_module.mm +++ b/sdk/objc/native/api/audio_device_module.mm @@ -10,49 +10,57 @@ #include "audio_device_module.h" +#include "api/environment/environment.h" +#include "api/environment/environment_factory.h" #include "api/make_ref_counted.h" #include "rtc_base/logging.h" - #include "sdk/objc/native/src/audio/audio_device_module_ios.h" +#ifndef WEBRTC_IOS +#error This file shouldn't be compiled on platforms other than IOS. +#endif + namespace webrtc { -webrtc::scoped_refptr CreateAudioDeviceModule( +scoped_refptr CreateAudioDeviceModule( + const Environment& env, bool bypass_voice_processing) { + RTC_DLOG(LS_INFO) << __FUNCTION__; + return make_ref_counted( + env, + bypass_voice_processing, + /*muted_speech_event_handler=*/nullptr, + /*error_handler=*/nullptr); +} + +scoped_refptr CreateAudioDeviceModule( bool bypass_voice_processing) { RTC_DLOG(LS_INFO) << __FUNCTION__; -#if defined(WEBRTC_IOS) - return webrtc::make_ref_counted( + return make_ref_counted( + CreateEnvironment(), bypass_voice_processing, /*muted_speech_event_handler=*/nullptr, /*error_handler=*/nullptr); -#else - RTC_LOG(LS_ERROR) - << "current platform is not supported => this module will self destruct!"; - return nullptr; -#endif } -webrtc::scoped_refptr CreateMutedDetectAudioDeviceModule( +scoped_refptr CreateMutedDetectAudioDeviceModule( + const Environment& env, AudioDeviceModule::MutedSpeechEventHandler muted_speech_event_handler, + ADMErrorHandler error_handler, bool bypass_voice_processing) { RTC_DLOG(LS_INFO) << __FUNCTION__; - return CreateMutedDetectAudioDeviceModule(muted_speech_event_handler, - /*error_handler=*/nullptr, - bypass_voice_processing); + return make_ref_counted( + env, bypass_voice_processing, muted_speech_event_handler, error_handler); } -webrtc::scoped_refptr CreateMutedDetectAudioDeviceModule( +scoped_refptr CreateMutedDetectAudioDeviceModule( AudioDeviceModule::MutedSpeechEventHandler muted_speech_event_handler, ADMErrorHandler error_handler, bool bypass_voice_processing) { RTC_DLOG(LS_INFO) << __FUNCTION__; -#if defined(WEBRTC_IOS) - return webrtc::make_ref_counted( - bypass_voice_processing, muted_speech_event_handler, error_handler); -#else - RTC_LOG(LS_ERROR) - << "current platform is not supported => this module will self destruct!"; - return nullptr; -#endif + return make_ref_counted( + CreateEnvironment(), + bypass_voice_processing, + muted_speech_event_handler, + error_handler); } } // namespace webrtc diff --git a/sdk/objc/native/api/objc_audio_device_module.h b/sdk/objc/native/api/objc_audio_device_module.h index a4387df9ee..be1b344866 100644 --- a/sdk/objc/native/api/objc_audio_device_module.h +++ b/sdk/objc/native/api/objc_audio_device_module.h @@ -12,11 +12,14 @@ #define SDK_OBJC_NATIVE_API_OBJC_AUDIO_DEVICE_MODULE_H_ #include "api/audio/audio_device.h" +#include "api/environment/environment.h" +#include "api/scoped_refptr.h" #import "components/audio/RTCAudioDevice.h" namespace webrtc { -webrtc::scoped_refptr CreateAudioDeviceModule( +scoped_refptr CreateAudioDeviceModule( + const Environment& env, id audio_device); } // namespace webrtc diff --git a/sdk/objc/native/api/objc_audio_device_module.mm b/sdk/objc/native/api/objc_audio_device_module.mm index cedb38f661..3eb78ece9b 100644 --- a/sdk/objc/native/api/objc_audio_device_module.mm +++ b/sdk/objc/native/api/objc_audio_device_module.mm @@ -10,6 +10,7 @@ #include "objc_audio_device_module.h" +#include "api/environment/environment.h" #include "api/make_ref_counted.h" #include "rtc_base/logging.h" @@ -17,11 +18,10 @@ namespace webrtc { -webrtc::scoped_refptr CreateAudioDeviceModule( - id audio_device) { +scoped_refptr CreateAudioDeviceModule( + const Environment& env, id audio_device) { RTC_DLOG(LS_INFO) << __FUNCTION__; - return webrtc::make_ref_counted( - audio_device); + return make_ref_counted(env, audio_device); } } // namespace webrtc diff --git a/sdk/objc/native/src/audio/audio_device_ios.h b/sdk/objc/native/src/audio/audio_device_ios.h index cc70ee7f2f..7dcffb07d5 100644 --- a/sdk/objc/native/src/audio/audio_device_ios.h +++ b/sdk/objc/native/src/audio/audio_device_ios.h @@ -14,6 +14,7 @@ #include #include +#include "api/environment/environment.h" #include "api/scoped_refptr.h" #include "api/sequence_checker.h" #include "api/task_queue/pending_task_safety_flag.h" @@ -56,6 +57,7 @@ class AudioDeviceIOS : public AudioDeviceGeneric, public VoiceProcessingAudioUnitObserver { public: explicit AudioDeviceIOS( + const Environment& env, bool bypass_voice_processing, AudioDeviceModule::MutedSpeechEventHandler muted_speech_event_handler, AudioDeviceIOSRenderErrorHandler render_error_handler); @@ -220,6 +222,8 @@ class AudioDeviceIOS : public AudioDeviceGeneric, // Resets thread-checkers before a call is restarted. void PrepareForNewStart(); + const Environment env_; + // Determines whether voice processing should be enabled or disabled. const bool bypass_voice_processing_; diff --git a/sdk/objc/native/src/audio/audio_device_ios.mm b/sdk/objc/native/src/audio/audio_device_ios.mm index c7e8b62b83..53dec77c75 100644 --- a/sdk/objc/native/src/audio/audio_device_ios.mm +++ b/sdk/objc/native/src/audio/audio_device_ios.mm @@ -17,6 +17,7 @@ #include #include "api/array_view.h" +#include "api/environment/environment.h" #include "api/task_queue/pending_task_safety_flag.h" #include "helpers.h" #include "modules/audio_device/fine_audio_buffer.h" @@ -25,7 +26,6 @@ #include "rtc_base/thread.h" #include "rtc_base/thread_annotations.h" #include "rtc_base/time_utils.h" -#include "system_wrappers/include/field_trial.h" #include "system_wrappers/include/metrics.h" #import "base/RTCLogging.h" @@ -96,10 +96,12 @@ static void LogDeviceInfo() { #endif // !defined(NDEBUG) AudioDeviceIOS::AudioDeviceIOS( + const Environment& env, bool bypass_voice_processing, AudioDeviceModule::MutedSpeechEventHandler muted_speech_event_handler, AudioDeviceIOSRenderErrorHandler render_error_handler) - : bypass_voice_processing_(bypass_voice_processing), + : env_(env), + bypass_voice_processing_(bypass_voice_processing), muted_speech_event_handler_(muted_speech_event_handler), render_error_handler_(render_error_handler), disregard_next_render_error_(false), @@ -579,7 +581,7 @@ static void LogDeviceInfo() { is_interrupted_); is_interrupted_ = false; if (!audio_unit_) return; - if (webrtc::field_trial::IsEnabled("WebRTC-Audio-iOS-Holding")) { + if (env_.field_trials().IsEnabled("WebRTC-Audio-iOS-Holding")) { // Work around an issue where audio does not restart properly after an // interruption by restarting the audio unit when the interruption ends. if (audio_unit_->GetState() == VoiceProcessingAudioUnit::kStarted) { diff --git a/sdk/objc/native/src/audio/audio_device_module_ios.h b/sdk/objc/native/src/audio/audio_device_module_ios.h index 394e1ff9bd..5ff50621b3 100644 --- a/sdk/objc/native/src/audio/audio_device_module_ios.h +++ b/sdk/objc/native/src/audio/audio_device_module_ios.h @@ -14,7 +14,6 @@ #include #include "api/audio/audio_device.h" -#include "api/task_queue/task_queue_factory.h" #include "audio_device_ios.h" #include "modules/audio_device/audio_device_buffer.h" #include "rtc_base/checks.h" @@ -31,6 +30,7 @@ class AudioDeviceModuleIOS : public AudioDeviceModule { int32_t AttachAudioBuffer(); explicit AudioDeviceModuleIOS( + const Environment& env, bool bypass_voice_processing, MutedSpeechEventHandler muted_speech_event_handler, ADMErrorHandler error_handler); @@ -137,11 +137,12 @@ class AudioDeviceModuleIOS : public AudioDeviceModule { #endif // WEBRTC_IOS private: void ReportError(ADMError error) const; + + const Environment env_; const bool bypass_voice_processing_; MutedSpeechEventHandler muted_speech_event_handler_; ADMErrorHandler error_handler_; bool initialized_ = false; - const std::unique_ptr task_queue_factory_; std::unique_ptr audio_device_; std::unique_ptr audio_device_buffer_; }; diff --git a/sdk/objc/native/src/audio/audio_device_module_ios.mm b/sdk/objc/native/src/audio/audio_device_module_ios.mm index 3b338f2399..7420d05ebd 100644 --- a/sdk/objc/native/src/audio/audio_device_module_ios.mm +++ b/sdk/objc/native/src/audio/audio_device_module_ios.mm @@ -10,7 +10,9 @@ #include "audio_device_module_ios.h" -#include "api/task_queue/default_task_queue_factory.h" +#include + +#include "api/environment/environment.h" #include "modules/audio_device/audio_device_config.h" #include "modules/audio_device/audio_device_generic.h" #include "rtc_base/checks.h" @@ -42,13 +44,14 @@ namespace ios_adm { AudioDeviceModuleIOS::AudioDeviceModuleIOS( + const Environment& env, bool bypass_voice_processing, MutedSpeechEventHandler muted_speech_event_handler, ADMErrorHandler error_handler) - : bypass_voice_processing_(bypass_voice_processing), + : env_(env), + bypass_voice_processing_(bypass_voice_processing), muted_speech_event_handler_(muted_speech_event_handler), - error_handler_(error_handler), - task_queue_factory_(CreateDefaultTaskQueueFactory()) { + error_handler_(error_handler) { RTC_LOG(LS_INFO) << "current platform is IOS"; RTC_LOG(LS_INFO) << "iPhone Audio APIs will be utilized."; } @@ -87,10 +90,13 @@ AudioDeviceIOSRenderErrorHandler error_handler = ^(OSStatus error) { ReportError(kRecordingDeviceFailed); }; - audio_device_buffer_.reset( - new webrtc::AudioDeviceBuffer(task_queue_factory_.get())); - audio_device_.reset(new ios_adm::AudioDeviceIOS( - bypass_voice_processing_, muted_speech_event_handler_, error_handler)); + audio_device_buffer_ = + std::make_unique(&env_.task_queue_factory()); + audio_device_ = + std::make_unique(env_, + bypass_voice_processing_, + muted_speech_event_handler_, + error_handler); RTC_CHECK(audio_device_); this->AttachAudioBuffer(); diff --git a/sdk/objc/native/src/objc_audio_device.h b/sdk/objc/native/src/objc_audio_device.h index 27eb0af88c..d8edde6f65 100644 --- a/sdk/objc/native/src/objc_audio_device.h +++ b/sdk/objc/native/src/objc_audio_device.h @@ -16,6 +16,7 @@ #import "components/audio/RTCAudioDevice.h" #include "api/audio/audio_device.h" +#include "api/environment/environment.h" #include "modules/audio_device/audio_device_buffer.h" #include "rtc_base/thread.h" @@ -30,7 +31,7 @@ namespace objc_adm { class ObjCAudioDeviceModule : public AudioDeviceModule { public: explicit ObjCAudioDeviceModule( - id audio_device); + const Environment& env, id audio_device); ~ObjCAudioDeviceModule() override; // Retrieve the currently utilized audio layer @@ -219,7 +220,7 @@ class ObjCAudioDeviceModule : public AudioDeviceModule { private: id audio_device_; - const std::unique_ptr task_queue_factory_; + const Environment env_; // AudioDeviceBuffer is a buffer to consume audio recorded by `RTCAudioDevice` // and provide audio to be played via `RTCAudioDevice`. diff --git a/sdk/objc/native/src/objc_audio_device.mm b/sdk/objc/native/src/objc_audio_device.mm index da595ba059..d17dd63811 100644 --- a/sdk/objc/native/src/objc_audio_device.mm +++ b/sdk/objc/native/src/objc_audio_device.mm @@ -9,12 +9,12 @@ */ #include "objc_audio_device.h" -#include "objc_audio_device_delegate.h" + +#include #import "components/audio/RTCAudioDevice.h" #include "modules/audio_device/fine_audio_buffer.h" - -#include "api/task_queue/default_task_queue_factory.h" +#include "objc_audio_device_delegate.h" #include "rtc_base/logging.h" #include "rtc_base/numerics/safe_minmax.h" #include "rtc_base/time_utils.h" @@ -49,9 +49,8 @@ namespace objc_adm { ObjCAudioDeviceModule::ObjCAudioDeviceModule( - id audio_device) - : audio_device_(audio_device), - task_queue_factory_(CreateDefaultTaskQueueFactory()) { + const Environment& env, id audio_device) + : audio_device_(audio_device), env_(env) { RTC_DLOG_F(LS_VERBOSE) << ""; RTC_DCHECK(audio_device_); thread_checker_.Detach(); @@ -82,8 +81,8 @@ io_record_thread_checker_.Detach(); thread_ = Thread::Current(); - audio_device_buffer_.reset( - new webrtc::AudioDeviceBuffer(task_queue_factory_.get())); + audio_device_buffer_ = + std::make_unique(&env_.task_queue_factory()); if (![audio_device_ isInitialized]) { if (audio_device_delegate_ == nil) { diff --git a/sdk/objc/unittests/RTCAudioDeviceModule_xctest.mm b/sdk/objc/unittests/RTCAudioDeviceModule_xctest.mm index e0ca410937..b87460c0ee 100644 --- a/sdk/objc/unittests/RTCAudioDeviceModule_xctest.mm +++ b/sdk/objc/unittests/RTCAudioDeviceModule_xctest.mm @@ -16,6 +16,7 @@ #import "sdk/objc/native/api/audio_device_module.h" #endif +#include "api/environment/environment_factory.h" #include "api/scoped_refptr.h" typedef int32_t (^NeedMorePlayDataBlock)(const size_t nSamples, @@ -156,7 +157,8 @@ - (void)setUp { _testEnabled = true; #endif - audioDeviceModule = webrtc::CreateAudioDeviceModule(); + audioDeviceModule = + webrtc::CreateAudioDeviceModule(webrtc::CreateEnvironment()); XCTAssertEqual(0, audioDeviceModule->Init()); XCTAssertEqual( 0, audioDeviceModule->GetPlayoutAudioParameters(&playoutParameters)); @@ -264,7 +266,7 @@ - (void)testStartPlayoutOnTwoInstances { // Create and initialize a second/extra ADM instance. The default ADM is // created by the test harness. webrtc::scoped_refptr secondAudioDeviceModule = - webrtc::CreateAudioDeviceModule(); + webrtc::CreateAudioDeviceModule(webrtc::CreateEnvironment()); XCTAssertNotEqual(secondAudioDeviceModule.get(), nullptr); XCTAssertEqual(0, secondAudioDeviceModule->Init()); diff --git a/sdk/objc/unittests/RTCAudioDevice_xctest.mm b/sdk/objc/unittests/RTCAudioDevice_xctest.mm index 5bdb90c3b8..f117832d22 100644 --- a/sdk/objc/unittests/RTCAudioDevice_xctest.mm +++ b/sdk/objc/unittests/RTCAudioDevice_xctest.mm @@ -12,6 +12,8 @@ #include +#include "api/environment/environment.h" +#include "api/environment/environment_factory.h" #include "api/task_queue/default_task_queue_factory.h" #import "sdk/objc/components/audio/RTCAudioSession+Private.h" @@ -45,8 +47,10 @@ - (void)setUp { _testEnabled = true; #endif - _audioDeviceModule = webrtc::CreateAudioDeviceModule(); + webrtc::Environment env = webrtc::CreateEnvironment(); + _audioDeviceModule = webrtc::CreateAudioDeviceModule(env); _audio_device.reset(new webrtc::ios_adm::AudioDeviceIOS( + env, /*bypass_voice_processing=*/false, /*muted_speech_event_handler=*/nullptr, /*render_error_handler=*/nullptr)); @@ -149,6 +153,7 @@ - (void)testMuteSpeechHandlerCalledWithStartedWhenSpeechActivityHasStarted { }; _audio_device.reset(new webrtc::ios_adm::AudioDeviceIOS( + webrtc::CreateEnvironment(), /*bypass_voice_processing=*/false, /*muted_speech_event_handler=*/muted_speech_event_handler, /*render_error_handler=*/nullptr)); @@ -169,6 +174,7 @@ - (void)testMuteSpeechHandlerCalledWithEndedWhenSpeechActivityHasEnded { }; _audio_device.reset(new webrtc::ios_adm::AudioDeviceIOS( + webrtc::CreateEnvironment(), /*bypass_voice_processing=*/false, /*muted_speech_event_handler=*/muted_speech_event_handler, /*render_error_handler=*/nullptr)); From 41557fb9818b23a6f686ac86fa37ae7543d373eb Mon Sep 17 00:00:00 2001 From: Gustaf Ullberg Date: Wed, 14 May 2025 10:50:30 +0200 Subject: [PATCH 148/292] Allow multiple different biquads in CascadedBiquadFilter As a bonus this CL also makes sure that cascaded_biquad_filter_unittest is run at all. Bug: webrtc:417439718 Change-Id: Ica11c2eccfdf70aedd11b55c5a84c9aac04c653f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391181 Reviewed-by: Sam Zackrisson Commit-Queue: Gustaf Ullberg Cr-Commit-Position: refs/heads/main@{#44620} --- modules/audio_processing/BUILD.gn | 1 + .../aec3/adaptive_fir_filter.cc | 9 ++- .../aec3/adaptive_fir_filter_unittest.cc | 31 +++++++-- .../aec3/subtractor_unittest.cc | 31 +++++++-- modules/audio_processing/high_pass_filter.cc | 50 ++++++++------ modules/audio_processing/utility/BUILD.gn | 2 + .../utility/cascaded_biquad_filter.cc | 12 +++- .../utility/cascaded_biquad_filter.h | 3 +- .../cascaded_biquad_filter_unittest.cc | 66 ++++++++++++++----- 9 files changed, 149 insertions(+), 56 deletions(-) diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn index c8b657fe72..66979a7c99 100644 --- a/modules/audio_processing/BUILD.gn +++ b/modules/audio_processing/BUILD.gn @@ -388,6 +388,7 @@ if (rtc_include_tests) { "capture_levels_adjuster", "capture_levels_adjuster:capture_levels_adjuster_unittests", "test/conversational_speech:unittest", + "utility:cascaded_biquad_filter_unittest", "utility:legacy_delay_estimator_unittest", "utility:pffft_wrapper_unittest", "vad:vad_unittests", diff --git a/modules/audio_processing/aec3/adaptive_fir_filter.cc b/modules/audio_processing/aec3/adaptive_fir_filter.cc index 22c3be2b80..e6e1de47b4 100644 --- a/modules/audio_processing/aec3/adaptive_fir_filter.cc +++ b/modules/audio_processing/aec3/adaptive_fir_filter.cc @@ -10,7 +10,15 @@ #include "modules/audio_processing/aec3/adaptive_fir_filter.h" +#include +#include +#include + // Defines WEBRTC_ARCH_X86_FAMILY, used below. +#include "api/array_view.h" +#include "modules/audio_processing/aec3/aec3_common.h" +#include "modules/audio_processing/aec3/render_buffer.h" +#include "modules/audio_processing/logging/apm_data_dumper.h" #include "rtc_base/system/arch.h" #if defined(WEBRTC_HAS_NEON) @@ -22,7 +30,6 @@ #include #include -#include #include "modules/audio_processing/aec3/fft_data.h" #include "rtc_base/checks.h" diff --git a/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc b/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc index ac973bed8f..0d5eef645a 100644 --- a/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc +++ b/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc @@ -14,9 +14,24 @@ #include #include +#include +#include +#include #include +#include #include - +#include +#include + +#include "api/array_view.h" +#include "api/audio/echo_canceller3_config.h" +#include "modules/audio_processing/aec3/aec3_common.h" +#include "modules/audio_processing/aec3/block.h" +#include "modules/audio_processing/aec3/delay_estimate.h" +#include "modules/audio_processing/aec3/echo_path_variability.h" +#include "modules/audio_processing/aec3/fft_data.h" +#include "modules/audio_processing/aec3/subtractor_output.h" +#include "rtc_base/checks.h" #include "rtc_base/system/arch.h" #if defined(WEBRTC_ARCH_X86_FAMILY) #include @@ -32,7 +47,6 @@ #include "modules/audio_processing/logging/apm_data_dumper.h" #include "modules/audio_processing/test/echo_canceller_test_tools.h" #include "modules/audio_processing/utility/cascaded_biquad_filter.h" -#include "rtc_base/arraysize.h" #include "rtc_base/numerics/safe_minmax.h" #include "rtc_base/random.h" #include "rtc_base/strings/string_builder.h" @@ -498,9 +512,9 @@ TEST_P(AdaptiveFirFilterMultiChannel, FilterAndAdapt) { num_capture_channels); std::array E2_coarse; // [B,A] = butter(2,100/8000,'high') - constexpr CascadedBiQuadFilter::BiQuadCoefficients - kHighPassFilterCoefficients = {{0.97261f, -1.94523f, 0.97261f}, - {-1.94448f, 0.94598f}}; + constexpr std::array + kHighPassFilterCoefficients = {CascadedBiQuadFilter::BiQuadCoefficients{ + {0.97261f, -1.94523f, 0.97261f}, {-1.94448f, 0.94598f}}}; for (auto& Y2_ch : Y2) { Y2_ch.fill(0.f); } @@ -521,9 +535,12 @@ TEST_P(AdaptiveFirFilterMultiChannel, FilterAndAdapt) { num_render_channels); for (size_t ch = 0; ch < num_render_channels; ++ch) { x_hp_filter[ch] = std::make_unique( - kHighPassFilterCoefficients, 1); + ArrayView( + kHighPassFilterCoefficients)); } - CascadedBiQuadFilter y_hp_filter(kHighPassFilterCoefficients, 1); + CascadedBiQuadFilter y_hp_filter( + (ArrayView( + kHighPassFilterCoefficients))); SCOPED_TRACE(ProduceDebugText(num_render_channels, delay_samples)); const size_t num_blocks_to_process = diff --git a/modules/audio_processing/aec3/subtractor_unittest.cc b/modules/audio_processing/aec3/subtractor_unittest.cc index 1b53273304..d3485327b5 100644 --- a/modules/audio_processing/aec3/subtractor_unittest.cc +++ b/modules/audio_processing/aec3/subtractor_unittest.cc @@ -11,16 +11,31 @@ #include "modules/audio_processing/aec3/subtractor.h" #include +#include +#include #include #include +#include #include +#include +#include +#include "api/array_view.h" +#include "api/audio/echo_canceller3_config.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" +#include "modules/audio_processing/aec3/aec3_common.h" +#include "modules/audio_processing/aec3/aec3_fft.h" #include "modules/audio_processing/aec3/aec_state.h" +#include "modules/audio_processing/aec3/block.h" +#include "modules/audio_processing/aec3/delay_estimate.h" +#include "modules/audio_processing/aec3/echo_path_variability.h" #include "modules/audio_processing/aec3/render_delay_buffer.h" +#include "modules/audio_processing/aec3/render_signal_analyzer.h" +#include "modules/audio_processing/aec3/subtractor_output.h" #include "modules/audio_processing/test/echo_canceller_test_tools.h" #include "modules/audio_processing/utility/cascaded_biquad_filter.h" +#include "rtc_base/checks.h" #include "rtc_base/random.h" #include "rtc_base/strings/string_builder.h" #include "test/gtest.h" @@ -83,20 +98,22 @@ std::vector RunSubtractorTest( } // [B,A] = butter(2,100/8000,'high') - constexpr CascadedBiQuadFilter::BiQuadCoefficients - kHighPassFilterCoefficients = {{0.97261f, -1.94523f, 0.97261f}, - {-1.94448f, 0.94598f}}; + constexpr std::array + kHighPassFilterCoefficients = {CascadedBiQuadFilter::BiQuadCoefficients{ + {0.97261f, -1.94523f, 0.97261f}, {-1.94448f, 0.94598f}}}; std::vector> x_hp_filter( num_render_channels); for (size_t ch = 0; ch < num_render_channels; ++ch) { - x_hp_filter[ch] = - std::make_unique(kHighPassFilterCoefficients, 1); + x_hp_filter[ch] = std::make_unique( + ArrayView( + kHighPassFilterCoefficients)); } std::vector> y_hp_filter( num_capture_channels); for (size_t ch = 0; ch < num_capture_channels; ++ch) { - y_hp_filter[ch] = - std::make_unique(kHighPassFilterCoefficients, 1); + y_hp_filter[ch] = std::make_unique( + ArrayView( + kHighPassFilterCoefficients)); } for (int block_num = 0; block_num < num_blocks_to_process; ++block_num) { diff --git a/modules/audio_processing/high_pass_filter.cc b/modules/audio_processing/high_pass_filter.cc index 9d8659ee8c..a6c2154b50 100644 --- a/modules/audio_processing/high_pass_filter.cc +++ b/modules/audio_processing/high_pass_filter.cc @@ -10,47 +10,57 @@ #include "modules/audio_processing/high_pass_filter.h" +#include +#include +#include + #include "api/array_view.h" #include "modules/audio_processing/audio_buffer.h" +#include "modules/audio_processing/utility/cascaded_biquad_filter.h" #include "rtc_base/checks.h" namespace webrtc { namespace { // [B,A] = butter(2,100/8000,'high') -constexpr CascadedBiQuadFilter::BiQuadCoefficients +constexpr std::array kHighPassFilterCoefficients16kHz = { - {0.972613898f, -1.945227797f, 0.972613898f}, - {-1.944477658f, 0.945977936f}}; + CascadedBiQuadFilter::BiQuadCoefficients{ + {0.972613898f, -1.945227797f, 0.972613898f}, + {-1.944477658f, 0.945977936f}}}; // [B,A] = butter(2,100/16000,'high') -constexpr CascadedBiQuadFilter::BiQuadCoefficients +constexpr std::array kHighPassFilterCoefficients32kHz = { - {0.986211925f, -1.972423849f, 0.986211925f}, - {-1.972233729f, 0.972613969f}}; + CascadedBiQuadFilter::BiQuadCoefficients{ + {0.986211925f, -1.972423849f, 0.986211925f}, + {-1.972233729f, 0.972613969f}}}; // [B,A] = butter(2,100/24000,'high') -constexpr CascadedBiQuadFilter::BiQuadCoefficients +constexpr std::array kHighPassFilterCoefficients48kHz = { - {0.990786698f, -1.981573396f, 0.990786698f}, - {-1.981488509f, 0.981658283f}}; - -constexpr size_t kNumberOfHighPassBiQuads = 1; + CascadedBiQuadFilter::BiQuadCoefficients{ + {0.990786698f, -1.981573396f, 0.990786698f}, + {-1.981488509f, 0.981658283f}}}; -const CascadedBiQuadFilter::BiQuadCoefficients& ChooseCoefficients( +ArrayView ChooseCoefficients( int sample_rate_hz) { switch (sample_rate_hz) { case 16000: - return kHighPassFilterCoefficients16kHz; + return ArrayView( + kHighPassFilterCoefficients16kHz); case 32000: - return kHighPassFilterCoefficients32kHz; + return ArrayView( + kHighPassFilterCoefficients32kHz); case 48000: - return kHighPassFilterCoefficients48kHz; + return ArrayView( + kHighPassFilterCoefficients48kHz); default: RTC_DCHECK_NOTREACHED(); } RTC_DCHECK_NOTREACHED(); - return kHighPassFilterCoefficients16kHz; + return ArrayView( + kHighPassFilterCoefficients16kHz); } } // namespace @@ -58,10 +68,9 @@ const CascadedBiQuadFilter::BiQuadCoefficients& ChooseCoefficients( HighPassFilter::HighPassFilter(int sample_rate_hz, size_t num_channels) : sample_rate_hz_(sample_rate_hz) { filters_.resize(num_channels); - const auto& coefficients = ChooseCoefficients(sample_rate_hz_); + auto coefficients = ChooseCoefficients(sample_rate_hz_); for (size_t k = 0; k < filters_.size(); ++k) { - filters_[k].reset( - new CascadedBiQuadFilter(coefficients, kNumberOfHighPassBiQuads)); + filters_[k].reset(new CascadedBiQuadFilter(coefficients)); } } @@ -109,8 +118,7 @@ void HighPassFilter::Reset(size_t num_channels) { } const auto& coefficients = ChooseCoefficients(sample_rate_hz_); for (size_t k = old_num_channels; k < filters_.size(); ++k) { - filters_[k].reset( - new CascadedBiQuadFilter(coefficients, kNumberOfHighPassBiQuads)); + filters_[k].reset(new CascadedBiQuadFilter(coefficients)); } } } diff --git a/modules/audio_processing/utility/BUILD.gn b/modules/audio_processing/utility/BUILD.gn index 4851e77b03..8dfb870cc9 100644 --- a/modules/audio_processing/utility/BUILD.gn +++ b/modules/audio_processing/utility/BUILD.gn @@ -51,6 +51,8 @@ if (rtc_include_tests) { deps = [ ":cascaded_biquad_filter", "../../../test:test_support", + "//api:array_view", + "//rtc_base:checks", "//testing/gtest", ] } diff --git a/modules/audio_processing/utility/cascaded_biquad_filter.cc b/modules/audio_processing/utility/cascaded_biquad_filter.cc index 19faadfc96..d56b62f26f 100644 --- a/modules/audio_processing/utility/cascaded_biquad_filter.cc +++ b/modules/audio_processing/utility/cascaded_biquad_filter.cc @@ -10,7 +10,11 @@ #include "modules/audio_processing/utility/cascaded_biquad_filter.h" #include +#include +#include +#include +#include "api/array_view.h" #include "rtc_base/checks.h" namespace webrtc { @@ -58,9 +62,11 @@ void CascadedBiQuadFilter::BiQuad::BiQuad::Reset() { } CascadedBiQuadFilter::CascadedBiQuadFilter( - const CascadedBiQuadFilter::BiQuadCoefficients& coefficients, - size_t num_biquads) - : biquads_(num_biquads, BiQuad(coefficients)) {} + ArrayView coefficients) { + for (const auto& single_biquad_coefficients : coefficients) { + biquads_.push_back(BiQuad(single_biquad_coefficients)); + } +} CascadedBiQuadFilter::CascadedBiQuadFilter( const std::vector& biquad_params) { diff --git a/modules/audio_processing/utility/cascaded_biquad_filter.h b/modules/audio_processing/utility/cascaded_biquad_filter.h index 799b439416..7050f98105 100644 --- a/modules/audio_processing/utility/cascaded_biquad_filter.h +++ b/modules/audio_processing/utility/cascaded_biquad_filter.h @@ -52,8 +52,7 @@ class CascadedBiQuadFilter { }; CascadedBiQuadFilter( - const CascadedBiQuadFilter::BiQuadCoefficients& coefficients, - size_t num_biquads); + ArrayView coefficients); explicit CascadedBiQuadFilter( const std::vector& biquad_params); ~CascadedBiQuadFilter(); diff --git a/modules/audio_processing/utility/cascaded_biquad_filter_unittest.cc b/modules/audio_processing/utility/cascaded_biquad_filter_unittest.cc index ff7022dba4..b83f4aa803 100644 --- a/modules/audio_processing/utility/cascaded_biquad_filter_unittest.cc +++ b/modules/audio_processing/utility/cascaded_biquad_filter_unittest.cc @@ -10,8 +10,12 @@ #include "modules/audio_processing/utility/cascaded_biquad_filter.h" +#include +#include #include +#include "api/array_view.h" +#include "rtc_base/checks.h" #include "test/gtest.h" namespace webrtc { @@ -20,17 +24,23 @@ namespace { // Coefficients for a second order Butterworth high-pass filter with cutoff // frequency 100 Hz. -const CascadedBiQuadFilter::BiQuadCoefficients kHighPassFilterCoefficients = { - {0.97261f, -1.94523f, 0.97261f}, - {-1.94448f, 0.94598f}}; +constexpr std::array + kHighPassFilterCoefficients = {CascadedBiQuadFilter::BiQuadCoefficients{ + {0.97261f, -1.94523f, 0.97261f}, + {-1.94448f, 0.94598f}}}; -const CascadedBiQuadFilter::BiQuadCoefficients kTransparentCoefficients = { - {1.f, 0.f, 0.f}, - {0.f, 0.f}}; +constexpr std::array + kTransparentCoefficients = { + CascadedBiQuadFilter::BiQuadCoefficients{{1.f, 0.f, 0.f}, {0.f, 0.f}}}; -const CascadedBiQuadFilter::BiQuadCoefficients kBlockingCoefficients = { - {0.f, 0.f, 0.f}, - {0.f, 0.f}}; +constexpr std::array + kBlockingCoefficients = { + CascadedBiQuadFilter::BiQuadCoefficients{{0.f, 0.f, 0.f}, {0.f, 0.f}}}; + +constexpr std::array + kCascadedCoefficients = { + CascadedBiQuadFilter::BiQuadCoefficients{{-1.f, 0.f, 0.f}, {0.f, 0.f}}, + CascadedBiQuadFilter::BiQuadCoefficients{{0.f, 0.f, -1.f}, {0.f, 0.f}}}; std::vector CreateInputWithIncreasingValues(size_t vector_length) { std::vector v(vector_length); @@ -47,7 +57,9 @@ std::vector CreateInputWithIncreasingValues(size_t vector_length) { TEST(CascadedBiquadFilter, BlockingConfiguration) { std::vector values = CreateInputWithIncreasingValues(1000); - CascadedBiQuadFilter filter(kBlockingCoefficients, 1); + CascadedBiQuadFilter filter( + (ArrayView( + kBlockingCoefficients))); filter.Process(values); EXPECT_EQ(std::vector(1000, 0.f), values); @@ -62,17 +74,21 @@ TEST(CascadedBiquadFilter, HighPassConfiguration) { values[k] = 1.f; } - CascadedBiQuadFilter filter(kHighPassFilterCoefficients, 2); + CascadedBiQuadFilter filter( + (ArrayView( + kHighPassFilterCoefficients))); filter.Process(values); for (size_t k = values.size() / 2; k < values.size(); ++k) { - EXPECT_NEAR(0.f, values[k], 1e-4); + EXPECT_NEAR(0.f, values[k], 1e-2); } } // Verifies that the reset functionality works as intended. TEST(CascadedBiquadFilter, HighPassConfigurationResetFunctionality) { - CascadedBiQuadFilter filter(kHighPassFilterCoefficients, 2); + CascadedBiQuadFilter filter( + (ArrayView( + kHighPassFilterCoefficients))); std::vector values1(100, 1.f); filter.Process(values1); @@ -94,12 +110,30 @@ TEST(CascadedBiquadFilter, TransparentConfiguration) { const std::vector input = CreateInputWithIncreasingValues(1000); std::vector output(input.size()); - CascadedBiQuadFilter filter(kTransparentCoefficients, 1); + CascadedBiQuadFilter filter( + (ArrayView( + kTransparentCoefficients))); filter.Process(input, output); EXPECT_EQ(input, output); } +// Verifies that cascading multiple biquads works as expected. +TEST(CascadedBiquadFilter, CascadedConfiguration) { + const std::vector input = CreateInputWithIncreasingValues(1000); + std::vector output(input.size()); + + CascadedBiQuadFilter filter( + (ArrayView( + kCascadedCoefficients))); + filter.Process(input, output); + + // The cascaded biquad is expected to result in a delay of two samples. + for (size_t i = 2; i < output.size(); ++i) { + EXPECT_EQ(output[i], input[i - 2]); + } +} + #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) // Verifies that the check of the lengths for the input and output works for the // non-in-place call. @@ -107,7 +141,9 @@ TEST(CascadedBiquadFilterDeathTest, InputSizeCheckVerification) { const std::vector input = CreateInputWithIncreasingValues(10); std::vector output(input.size() - 1); - CascadedBiQuadFilter filter(kTransparentCoefficients, 1); + CascadedBiQuadFilter filter( + (ArrayView( + kTransparentCoefficients))); EXPECT_DEATH(filter.Process(input, output), ""); } #endif From 46ac2af1fa8a1cc89fcb11e36270ee33ae612d85 Mon Sep 17 00:00:00 2001 From: Diep Bui Date: Wed, 14 May 2025 10:35:24 +0000 Subject: [PATCH 149/292] Clean up unused PaceAtLossBasedEstimate field trial Bug: webrtc:12707 Change-Id: Iae943dc63589e117a01cd097893aaaa429f77d32 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391420 Reviewed-by: Per Kjellander Commit-Queue: Diep Bui Cr-Commit-Position: refs/heads/main@{#44621} --- .../goog_cc/loss_based_bwe_v2.cc | 9 ----- .../goog_cc/loss_based_bwe_v2.h | 1 - .../goog_cc/loss_based_bwe_v2_test.cc | 36 ------------------- .../goog_cc/send_side_bandwidth_estimation.cc | 5 --- .../goog_cc/send_side_bandwidth_estimation.h | 1 - 5 files changed, 52 deletions(-) diff --git a/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc b/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc index 25978593ca..92150715cb 100644 --- a/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc +++ b/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc @@ -469,8 +469,6 @@ std::optional LossBasedBweV2::CreateConfig( FieldTrialParameter padding_duration("PaddingDuration", TimeDelta::Zero()); FieldTrialParameter bound_best_candidate("BoundBestCandidate", false); - FieldTrialParameter pace_at_loss_based_estimate( - "PaceAtLossBasedEstimate", false); FieldTrialParameter median_sending_rate_factor( "MedianSendingRateFactor", 2.0); if (key_value_config) { @@ -513,7 +511,6 @@ std::optional LossBasedBweV2::CreateConfig( &use_byte_loss_rate, &padding_duration, &bound_best_candidate, - &pace_at_loss_based_estimate, &median_sending_rate_factor}, key_value_config->Lookup("WebRTC-Bwe-LossBasedBweV2")); } @@ -579,7 +576,6 @@ std::optional LossBasedBweV2::CreateConfig( config.use_byte_loss_rate = use_byte_loss_rate.Get(); config.padding_duration = padding_duration.Get(); config.bound_best_candidate = bound_best_candidate.Get(); - config.pace_at_loss_based_estimate = pace_at_loss_based_estimate.Get(); config.median_sending_rate_factor = median_sending_rate_factor.Get(); return config; } @@ -1216,11 +1212,6 @@ bool LossBasedBweV2::CanKeepIncreasingState(DataRate estimate) const { last_padding_info_.padding_rate < estimate; } -bool LossBasedBweV2::PaceAtLossBasedEstimate() const { - return config_->pace_at_loss_based_estimate && - loss_based_result_.state != LossBasedState::kDelayBasedEstimate; -} - DataRate LossBasedBweV2::GetMedianSendingRate() const { std::vector sending_rates; for (const Observation& observation : observations_) { diff --git a/modules/congestion_controller/goog_cc/loss_based_bwe_v2.h b/modules/congestion_controller/goog_cc/loss_based_bwe_v2.h index 5102f463f9..b088420f19 100644 --- a/modules/congestion_controller/goog_cc/loss_based_bwe_v2.h +++ b/modules/congestion_controller/goog_cc/loss_based_bwe_v2.h @@ -75,7 +75,6 @@ class LossBasedBweV2 { void UpdateBandwidthEstimate(ArrayView packet_results, DataRate delay_based_estimate, bool in_alr); - bool PaceAtLossBasedEstimate() const; // For unit testing only. void SetBandwidthEstimate(DataRate bandwidth_estimate); diff --git a/modules/congestion_controller/goog_cc/loss_based_bwe_v2_test.cc b/modules/congestion_controller/goog_cc/loss_based_bwe_v2_test.cc index a061eb6a3a..0550e4acd3 100644 --- a/modules/congestion_controller/goog_cc/loss_based_bwe_v2_test.cc +++ b/modules/congestion_controller/goog_cc/loss_based_bwe_v2_test.cc @@ -1881,42 +1881,6 @@ TEST_F(LossBasedBweV2Test, UseByteLossRateDoesNotIgnoreLossSpikeOnSendBurst) { kDelayBasedEstimate); } -TEST_F(LossBasedBweV2Test, PaceAtLossBasedEstimate) { - ExplicitKeyValueConfig key_value_config(ShortObservationConfig( - "PaceAtLossBasedEstimate:true,PaddingDuration:1000ms")); - LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config); - loss_based_bandwidth_estimator.SetBandwidthEstimate( - DataRate::KilobitsPerSec(1000)); - loss_based_bandwidth_estimator.UpdateBandwidthEstimate( - CreatePacketResultsWithReceivedPackets( - /*first_packet_timestamp=*/Timestamp::Zero()), - /*delay_based_estimate=*/DataRate::KilobitsPerSec(1000), - /*in_alr=*/false); - EXPECT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state, - LossBasedState::kDelayBasedEstimate); - EXPECT_FALSE(loss_based_bandwidth_estimator.PaceAtLossBasedEstimate()); - - loss_based_bandwidth_estimator.UpdateBandwidthEstimate( - CreatePacketResultsWith100pLossRate( - /*first_packet_timestamp=*/Timestamp::Zero() + - kObservationDurationLowerBound), - /*delay_based_estimate=*/DataRate::KilobitsPerSec(1000), - /*in_alr=*/false); - EXPECT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state, - LossBasedState::kDecreasing); - EXPECT_TRUE(loss_based_bandwidth_estimator.PaceAtLossBasedEstimate()); - - loss_based_bandwidth_estimator.UpdateBandwidthEstimate( - CreatePacketResultsWithReceivedPackets( - /*first_packet_timestamp=*/Timestamp::Zero() + - kObservationDurationLowerBound * 2), - /*delay_based_estimate=*/DataRate::KilobitsPerSec(1000), - /*in_alr=*/false); - EXPECT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state, - LossBasedState::kIncreaseUsingPadding); - EXPECT_TRUE(loss_based_bandwidth_estimator.PaceAtLossBasedEstimate()); -} - TEST_F(LossBasedBweV2Test, EstimateDoesNotBackOffDueToPacketReorderingBetweenFeedback) { ExplicitKeyValueConfig key_value_config(ShortObservationConfig("")); diff --git a/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc b/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc index 7fe27bf910..c42e0152a7 100644 --- a/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc +++ b/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc @@ -693,9 +693,4 @@ bool SendSideBandwidthEstimation::LossBasedBandwidthEstimatorV2ReadyForUse() return loss_based_bandwidth_estimator_v2_->IsReady(); } -bool SendSideBandwidthEstimation::PaceAtLossBasedEstimate() const { - return LossBasedBandwidthEstimatorV2ReadyForUse() && - loss_based_bandwidth_estimator_v2_->PaceAtLossBasedEstimate(); -} - } // namespace webrtc diff --git a/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h b/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h index 9a56a280ab..72123a7308 100644 --- a/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h +++ b/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h @@ -128,7 +128,6 @@ class SendSideBandwidthEstimation { BandwidthUsage delay_detector_state, std::optional probe_bitrate, bool in_alr); - bool PaceAtLossBasedEstimate() const; private: friend class GoogCcStatePrinter; From f6bc203d3209111ca5cfb99a813ad293a4acd10a Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Wed, 14 May 2025 07:13:09 -0700 Subject: [PATCH 150/292] Roll chromium_revision 274243632d..4388280c46 (1459868:1460056) Change log: https://chromium.googlesource.com/chromium/src/+log/274243632d..4388280c46 Full diff: https://chromium.googlesource.com/chromium/src/+/274243632d..4388280c46 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/7ac8b768f3..7678e0c1d4 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/e42f899ae7..bf6d0908ba * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/d6c29dd3c5..58fb9d9896 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/e93dc3dea2..c9999c4687 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/3dcdf2820d..7be58989a8 * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/14bfda1708..a9cc320bf7 * src/third_party/googletest/src: https://chromium.googlesource.com/external/github.com/google/googletest.git/+log/8b8ef3ff0d..9f79a9597a * src/third_party/test_fonts/test_fonts: f26f29c9d3bfae588207bbc9762de8d142e58935c62a86f67332819b15203b35..a28b222b79851716f8358d2800157d9ffe117b3545031ae51f69b7e1e1b9a969 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/635b64025b..1a55eea0c7 DEPS diff: https://chromium.googlesource.com/chromium/src/+/274243632d..4388280c46/DEPS No update to Clang. BUG=None Change-Id: I8336669926acea00bc844f0f3e033bf2b72eb0ee Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391500 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44622} --- DEPS | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) diff --git a/DEPS b/DEPS index bae4a3a766..09afcfe0fa 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '274243632d0c927460706ca04f3c3e605d43b9c2', + 'chromium_revision': '4388280c464a89e79135070b78a0dc5851a069a1', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@7ac8b768f3161dfff6a9c66587f7fef5ffa28443', + 'https://chromium.googlesource.com/chromium/src/base@7678e0c1d4b7d1820a95eeaa17f61bcfd70b1dd3', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@e42f899ae75d20881f14e06fd324041312fe6a83', + 'https://chromium.googlesource.com/chromium/src/build@bf6d0908ba43a7e35178e96b6d085c5e7d1f382f', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@077a66f30fcf281b066fafb6dfc60818c238efb6', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@d6c29dd3c53ab982e2fcdadb464a42f820be1de7', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@58fb9d989636de6cfd016d08b19da0e92c2cbcb4', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@e93dc3dea230f64eab7e97113e02b942af35962e', + 'https://chromium.googlesource.com/chromium/src/testing@c9999c468704c1586181105de23637e77f3dc1ba', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@3dcdf2820d8a9fa70f479fa0862e31e492d19105', + 'https://chromium.googlesource.com/chromium/src/third_party@7be58989a8f50d31d1687113e547d556d09a191e', 'src/buildtools/linux64': { 'packages': [ @@ -339,10 +339,10 @@ deps = { 'bucket': 'chromium-fonts', 'objects': [ { - 'object_name': 'f26f29c9d3bfae588207bbc9762de8d142e58935c62a86f67332819b15203b35', - 'sha256sum': 'f26f29c9d3bfae588207bbc9762de8d142e58935c62a86f67332819b15203b35', - 'size_bytes': 32750602, - 'generation': 1717109450425063, + 'object_name': 'a28b222b79851716f8358d2800157d9ffe117b3545031ae51f69b7e1e1b9a969', + 'sha256sum': 'a28b222b79851716f8358d2800157d9ffe117b3545031ae51f69b7e1e1b9a969', + 'size_bytes': 33233004, + 'generation': 1746454415712520, }, ], }, @@ -497,7 +497,7 @@ deps = { 'src/third_party/crc32c/src': 'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558', 'src/third_party/depot_tools': - 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@14bfda17088cb03d7bc0ba7df6cac2699e051e14', + 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@a9cc320bf761323906c614f7040cb1dc22fcc858', 'src/third_party/ffmpeg': 'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@dcdd0fa51b65a0b1688ff6b8f0cc81908f09ded2', 'src/third_party/flatbuffers/src': @@ -525,7 +525,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/googletest/src': - 'https://chromium.googlesource.com/external/github.com/google/googletest.git@8b8ef3ff0d1f7a1739e0d7c3dd46ab37104fcd9a', + 'https://chromium.googlesource.com/external/github.com/google/googletest.git@9f79a9597ad9b5394e5c620ebf76824f77ffbde4', 'src/third_party/icu': { 'url': 'https://chromium.googlesource.com/chromium/deps/icu.git@b929596baebf0ab4ac7ec07f38365db4c50a559d', }, @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@635b64025b17f341f6106d520736829bf76e544f', + 'https://chromium.googlesource.com/chromium/src/tools@1a55eea0c72d2a09d815816f9bb92b2a2c664f0d', 'src/third_party/espresso': { 'packages': [ From ca335e682ddbb284acd0437201d734801e89dc5b Mon Sep 17 00:00:00 2001 From: Gustaf Ullberg Date: Wed, 14 May 2025 13:52:13 +0200 Subject: [PATCH 151/292] Simplify (less verbose) filter notation Bug: webrtc:417439718 Change-Id: I43681a1e0c3926993f81c957f93a6b0a285dedfd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391440 Commit-Queue: Gustaf Ullberg Reviewed-by: Sam Zackrisson Cr-Commit-Position: refs/heads/main@{#44623} --- .../aec3/adaptive_fir_filter_unittest.cc | 4 ++-- .../audio_processing/aec3/subtractor_unittest.cc | 4 ++-- modules/audio_processing/high_pass_filter.cc | 15 ++++++--------- .../utility/cascaded_biquad_filter_unittest.cc | 14 +++++--------- 4 files changed, 15 insertions(+), 22 deletions(-) diff --git a/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc b/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc index 0d5eef645a..387a0744c1 100644 --- a/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc +++ b/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc @@ -513,8 +513,8 @@ TEST_P(AdaptiveFirFilterMultiChannel, FilterAndAdapt) { std::array E2_coarse; // [B,A] = butter(2,100/8000,'high') constexpr std::array - kHighPassFilterCoefficients = {CascadedBiQuadFilter::BiQuadCoefficients{ - {0.97261f, -1.94523f, 0.97261f}, {-1.94448f, 0.94598f}}}; + kHighPassFilterCoefficients = { + {{{0.97261f, -1.94523f, 0.97261f}, {-1.94448f, 0.94598f}}}}; for (auto& Y2_ch : Y2) { Y2_ch.fill(0.f); } diff --git a/modules/audio_processing/aec3/subtractor_unittest.cc b/modules/audio_processing/aec3/subtractor_unittest.cc index d3485327b5..d940df47ef 100644 --- a/modules/audio_processing/aec3/subtractor_unittest.cc +++ b/modules/audio_processing/aec3/subtractor_unittest.cc @@ -99,8 +99,8 @@ std::vector RunSubtractorTest( // [B,A] = butter(2,100/8000,'high') constexpr std::array - kHighPassFilterCoefficients = {CascadedBiQuadFilter::BiQuadCoefficients{ - {0.97261f, -1.94523f, 0.97261f}, {-1.94448f, 0.94598f}}}; + kHighPassFilterCoefficients = { + {{{0.97261f, -1.94523f, 0.97261f}, {-1.94448f, 0.94598f}}}}; std::vector> x_hp_filter( num_render_channels); for (size_t ch = 0; ch < num_render_channels; ++ch) { diff --git a/modules/audio_processing/high_pass_filter.cc b/modules/audio_processing/high_pass_filter.cc index a6c2154b50..276b5a2c53 100644 --- a/modules/audio_processing/high_pass_filter.cc +++ b/modules/audio_processing/high_pass_filter.cc @@ -25,23 +25,20 @@ namespace { // [B,A] = butter(2,100/8000,'high') constexpr std::array kHighPassFilterCoefficients16kHz = { - CascadedBiQuadFilter::BiQuadCoefficients{ - {0.972613898f, -1.945227797f, 0.972613898f}, - {-1.944477658f, 0.945977936f}}}; + {{{0.972613898f, -1.945227797f, 0.972613898f}, + {-1.944477658f, 0.945977936f}}}}; // [B,A] = butter(2,100/16000,'high') constexpr std::array kHighPassFilterCoefficients32kHz = { - CascadedBiQuadFilter::BiQuadCoefficients{ - {0.986211925f, -1.972423849f, 0.986211925f}, - {-1.972233729f, 0.972613969f}}}; + {{{0.986211925f, -1.972423849f, 0.986211925f}, + {-1.972233729f, 0.972613969f}}}}; // [B,A] = butter(2,100/24000,'high') constexpr std::array kHighPassFilterCoefficients48kHz = { - CascadedBiQuadFilter::BiQuadCoefficients{ - {0.990786698f, -1.981573396f, 0.990786698f}, - {-1.981488509f, 0.981658283f}}}; + {{{0.990786698f, -1.981573396f, 0.990786698f}, + {-1.981488509f, 0.981658283f}}}}; ArrayView ChooseCoefficients( int sample_rate_hz) { diff --git a/modules/audio_processing/utility/cascaded_biquad_filter_unittest.cc b/modules/audio_processing/utility/cascaded_biquad_filter_unittest.cc index b83f4aa803..cf213787fc 100644 --- a/modules/audio_processing/utility/cascaded_biquad_filter_unittest.cc +++ b/modules/audio_processing/utility/cascaded_biquad_filter_unittest.cc @@ -25,22 +25,18 @@ namespace { // Coefficients for a second order Butterworth high-pass filter with cutoff // frequency 100 Hz. constexpr std::array - kHighPassFilterCoefficients = {CascadedBiQuadFilter::BiQuadCoefficients{ - {0.97261f, -1.94523f, 0.97261f}, - {-1.94448f, 0.94598f}}}; + kHighPassFilterCoefficients = { + {{{0.97261f, -1.94523f, 0.97261f}, {-1.94448f, 0.94598f}}}}; constexpr std::array - kTransparentCoefficients = { - CascadedBiQuadFilter::BiQuadCoefficients{{1.f, 0.f, 0.f}, {0.f, 0.f}}}; + kTransparentCoefficients = {{{{1.f, 0.f, 0.f}, {0.f, 0.f}}}}; constexpr std::array - kBlockingCoefficients = { - CascadedBiQuadFilter::BiQuadCoefficients{{0.f, 0.f, 0.f}, {0.f, 0.f}}}; + kBlockingCoefficients = {{{{0.f, 0.f, 0.f}, {0.f, 0.f}}}}; constexpr std::array kCascadedCoefficients = { - CascadedBiQuadFilter::BiQuadCoefficients{{-1.f, 0.f, 0.f}, {0.f, 0.f}}, - CascadedBiQuadFilter::BiQuadCoefficients{{0.f, 0.f, -1.f}, {0.f, 0.f}}}; + {{{-1.f, 0.f, 0.f}, {0.f, 0.f}}, {{0.f, 0.f, -1.f}, {0.f, 0.f}}}}; std::vector CreateInputWithIncreasingValues(size_t vector_length) { std::vector v(vector_length); From 697bf7666c739991da2de39dbb4e912eeb7bb729 Mon Sep 17 00:00:00 2001 From: Daniel White Date: Wed, 14 May 2025 12:08:41 -0400 Subject: [PATCH 152/292] [iOS] Fast-follow for Deprecated API Test Failure In a previous CL, the RTCCameraPreviewView.m file's use of the AVCaptureVideoOrientation API was encapsulted in a private function then gated behind a compile-time guard. However, the invocation was not also gated behind this same compile-time guard. This caused the WebRTC waterfall bots to fail. In this CL, the invocation of the private function that uses the deprecated API is also gated behind the compile-time flag. Change-Id: Ib94747fbe452b07b181c59387664bc1303545286 Bug: chromium:348649285 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391560 Reviewed-by: Jeremy Leconte Owners-Override: Jeremy Leconte Reviewed-by: Mirko Bonadei Commit-Queue: Mirko Bonadei Cr-Commit-Position: refs/heads/main@{#44624} --- sdk/objc/helpers/RTCCameraPreviewView.m | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sdk/objc/helpers/RTCCameraPreviewView.m b/sdk/objc/helpers/RTCCameraPreviewView.m index 0ebdcb5d3c..3314322b99 100644 --- a/sdk/objc/helpers/RTCCameraPreviewView.m +++ b/sdk/objc/helpers/RTCCameraPreviewView.m @@ -82,10 +82,12 @@ - (void)orientationChanged:(NSNotification *)notification { - (void)setCorrectVideoOrientation { if (@available(iOS 17, *)) { [self modifyVideoAngle]; - return; } - - [self modifyVideoOrientation]; +#if !defined(__IPHONE_17_0) || __IPHONE_OS_VERSION_MIN_REQUIRED < __IPHONE_17_0 + else { + [self modifyVideoOrientation]; + } +#endif } #pragma mark - Private From bf4ae747eff112cfa69e6bcbc326646bd42bf769 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Wed, 14 May 2025 15:06:36 -0700 Subject: [PATCH 153/292] Roll chromium_revision 4388280c46..4ad2db3c40 (1460056:1460388) Change log: https://chromium.googlesource.com/chromium/src/+log/4388280c46..4ad2db3c40 Full diff: https://chromium.googlesource.com/chromium/src/+/4388280c46..4ad2db3c40 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/7678e0c1d4..2e5431a6db * src/build: https://chromium.googlesource.com/chromium/src/build/+log/bf6d0908ba..0121a4c952 * src/buildtools: https://chromium.googlesource.com/chromium/src/buildtools/+log/077a66f30f..e9b36e9ffb * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/58fb9d9896..96b8ccee2c * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/c9999c4687..e0bf8ba12f * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/7be58989a8..efebb45477 * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/89973806bc..62956948e4 * src/third_party/breakpad/breakpad: https://chromium.googlesource.com/breakpad/breakpad.git/+log/9eeece27b7..fad8247731 * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/fd624be2fa..bbb14e4656 * src/third_party/freetype/src: https://chromium.googlesource.com/chromium/src/third_party/freetype2.git/+log/79912716e7..8a152c824a * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/94198a4a9c..708b5c0615 * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/63f1ba4f68..6690560f58 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/1a55eea0c7..2b4f2ff382 DEPS diff: https://chromium.googlesource.com/chromium/src/+/4388280c46..4ad2db3c40/DEPS No update to Clang. BUG=None Change-Id: Ie21d8b2e6cad4e1f9b7554e145ee177fd1aa42da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391601 Commit-Queue: Autoroller Bot-Commit: Autoroller Cr-Commit-Position: refs/heads/main@{#44625} --- DEPS | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/DEPS b/DEPS index 09afcfe0fa..b458d6258a 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '4388280c464a89e79135070b78a0dc5851a069a1', + 'chromium_revision': '4ad2db3c40b73c832a17c04293d7583634041f91', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,24 +73,24 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@7678e0c1d4b7d1820a95eeaa17f61bcfd70b1dd3', + 'https://chromium.googlesource.com/chromium/src/base@2e5431a6db254f0cad1071ee2e88c0c0be219985', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@bf6d0908ba43a7e35178e96b6d085c5e7d1f382f', + 'https://chromium.googlesource.com/chromium/src/build@0121a4c952cbf1556b17f68fed7c84c680dc940c', 'src/buildtools': - 'https://chromium.googlesource.com/chromium/src/buildtools@077a66f30fcf281b066fafb6dfc60818c238efb6', + 'https://chromium.googlesource.com/chromium/src/buildtools@e9b36e9ffb02b268d194c3f82f89952d09d0c714', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. 'src/examples/androidtests/third_party/gradle': { 'url': 'https://chromium.googlesource.com/external/github.com/gradle/gradle.git@f2d1fb54a951d8b11d25748e4711bec8d128d7e3', 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@58fb9d989636de6cfd016d08b19da0e92c2cbcb4', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@96b8ccee2c00d8355f2831f59e462d9925692134', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@c9999c468704c1586181105de23637e77f3dc1ba', + 'https://chromium.googlesource.com/chromium/src/testing@e0bf8ba12f5a76589729ee1b6d0476f337809cb3', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@7be58989a8f50d31d1687113e547d556d09a191e', + 'https://chromium.googlesource.com/chromium/src/third_party@efebb4547796b4a29d2f3be63d8d254f99a10560', 'src/buildtools/linux64': { 'packages': [ @@ -480,11 +480,11 @@ deps = { }, 'src/third_party/boringssl/src': - 'https://boringssl.googlesource.com/boringssl.git@89973806bc2ef652189e157f2736a7d32229c404', + 'https://boringssl.googlesource.com/boringssl.git@62956948e445c5bc67d3f299e0003c34a3030523', 'src/third_party/breakpad/breakpad': - 'https://chromium.googlesource.com/breakpad/breakpad.git@9eeece27b727d2ffac4b7042bb69d7c140e58312', + 'https://chromium.googlesource.com/breakpad/breakpad.git@fad82477312f786994151f01c352813fabf924c0', 'src/third_party/catapult': - 'https://chromium.googlesource.com/catapult.git@fd624be2faba11f03d2cf94e095b86707758f062', + 'https://chromium.googlesource.com/catapult.git@bbb14e46563c14a055b394f18b117e504e98d4ab', 'src/third_party/ced/src': { 'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5', }, @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@94198a4a9cd62d176f50cdfae9b8d4db6a052011', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@708b5c0615d00637bea1babef94bfeb69f0ab5a3', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -511,7 +511,7 @@ deps = { 'condition': 'checkout_linux', }, 'src/third_party/freetype/src': - 'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@79912716e783b4c9e47c35907ae080e2ec02fbc6', + 'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@8a152c824ae08fc3459df5c87e10770fc47f80b1', 'src/third_party/harfbuzz-ng/src': 'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@9f83bbbe64654b45ba5bb06927ff36c2e7588495', 'src/third_party/google_benchmark/src': { @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '63f1ba4f684943307b6d99d0669b1d180bc3b6f5', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '6690560f58167a82a9302a6df05303324f0e91bb', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@1a55eea0c72d2a09d815816f9bb92b2a2c664f0d', + 'https://chromium.googlesource.com/chromium/src/tools@2b4f2ff3824124542c33739f816406caed7fa2e8', 'src/third_party/espresso': { 'packages': [ From 6c876beeae5ebf638fc30748bd6a98f46baeb44b Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Wed, 14 May 2025 21:05:52 -0700 Subject: [PATCH 154/292] Roll chromium_revision 4ad2db3c40..35e96510ff (1460388:1460497) Change log: https://chromium.googlesource.com/chromium/src/+log/4ad2db3c40..35e96510ff Full diff: https://chromium.googlesource.com/chromium/src/+/4ad2db3c40..35e96510ff Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/2e5431a6db..53335694d5 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/0121a4c952..289b4d8c67 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/96b8ccee2c..e1277e5176 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/e0bf8ba12f..e7241882c3 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/efebb45477..c1d0b85060 * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/62956948e4..02bc0949e5 * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/bbb14e4656..2fa93b2332 * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/a9cc320bf7..c273e3eb62 * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/708b5c0615..3792c9842a * src/third_party/llvm-build/Release+Asserts: Linux_x64/clang-llvmorg-21-init-9266-g09006611-2.tar.xz,Linux_x64/clang-tidy-llvmorg-21-init-9266-g09006611-2.tar.xz,Linux_x64/clangd-llvmorg-21-init-9266-g09006611-2.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-2.tar.xz,Linux_x64/llvmobjdump-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac/clang-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac/clang-mac-runtime-library-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac/clang-tidy-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac/clangd-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac/llvm-code-coverage-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac_arm64/clang-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac_arm64/clang-tidy-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac_arm64/clangd-llvmorg-21-init-9266-g09006611-2.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-2.tar.xz,Win/clang-llvmorg-21-init-9266-g09006611-2.tar.xz,Win/clang-tidy-llvmorg-21-init-9266-g09006611-2.tar.xz,Win/clang-win-runtime-library-llvmorg-21-init-9266-g09006611-2.tar.xz,Win/clangd-llvmorg-21-init-9266-g09006611-2.tar.xz,Win/llvm-code-coverage-llvmorg-21-init-9266-g09006611-2.tar.xz,Win/llvmobjdump-llvmorg-21-init-9266-g09006611-2.tar.xz..Linux_x64/clang-llvmorg-21-init-11777-gfd3fecfc-1.tar.xz,Linux_x64/clang-tidy-llvmorg-21-init-11777-gfd3fecfc-1.tar.xz,Linux_x64/clangd-llvmorg-21-init-11777-gfd3fecfc-1.tar.xz,Linux_x64/llvm-code-coverage-llvmorg-21-init-11777-gfd3fecfc-1.tar.xz,Linux_x64/llvmobjdump-llvmorg-21-init-11777-gfd3fecfc-1.tar.xz,Mac/clang-llvmorg-21-init-11777-gfd3fecfc-1.tar.xz,Mac/clang-mac-runtime-library-llvmorg-21-init-11777-gfd3fecfc-1.tar.xz,Mac/clang-tidy-llvmorg-21-init-11777-gfd3fecfc-1.tar.xz,Mac/clangd-llvmorg-21-init-11777-gfd3fecfc-1.tar.xz,Mac/llvm-code-coverage-llvmorg-21-init-11777-gfd3fecfc-1.tar.xz,Mac_arm64/clang-llvmorg-21-init-11777-gfd3fecfc-1.tar.xz,Mac_arm64/clang-tidy-llvmorg-21-init-11777-gfd3fecfc-1.tar.xz,Mac_arm64/clangd-llvmorg-21-init-11777-gfd3fecfc-1.tar.xz,Mac_arm64/llvm-code-coverage-llvmorg-21-init-11777-gfd3fecfc-1.tar.xz,Win/clang-llvmorg-21-init-11777-gfd3fecfc-1.tar.xz,Win/clang-tidy-llvmorg-21-init-11777-gfd3fecfc-1.tar.xz,Win/clang-win-runtime-library-llvmorg-21-init-11777-gfd3fecfc-1.tar.xz,Win/clangd-llvmorg-21-init-11777-gfd3fecfc-1.tar.xz,Win/llvm-code-coverage-llvmorg-21-init-11777-gfd3fecfc-1.tar.xz,Win/llvmobjdump-llvmorg-21-init-11777-gfd3fecfc-1.tar.xz * src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/27ee073016..9aefa1cede * src/third_party/rust-toolchain: Linux_x64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Mac/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Mac_arm64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz,Win/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz..Linux_x64/rust-toolchain-4a0969e06dbeaaa43914d2d00b2e843d49aa3886-1-llvmorg-21-init-11777-gfd3fecfc.tar.xz,Mac/rust-toolchain-4a0969e06dbeaaa43914d2d00b2e843d49aa3886-1-llvmorg-21-init-11777-gfd3fecfc.tar.xz,Mac_arm64/rust-toolchain-4a0969e06dbeaaa43914d2d00b2e843d49aa3886-1-llvmorg-21-init-11777-gfd3fecfc.tar.xz,Win/rust-toolchain-4a0969e06dbeaaa43914d2d00b2e843d49aa3886-1-llvmorg-21-init-11777-gfd3fecfc.tar.xz * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/2b4f2ff382..18608d93c9 DEPS diff: https://chromium.googlesource.com/chromium/src/+/4ad2db3c40..35e96510ff/DEPS Clang version changed llvmorg-21-init-9266-g09006611:llvmorg-21-init-11777-gfd3fecfc Details: https://chromium.googlesource.com/chromium/src/+/4ad2db3c40..35e96510ff/tools/clang/scripts/update.py BUG=None Change-Id: Iccbabb42bebfc767be671a6548cc59198ed8be9c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391403 Commit-Queue: Autoroller Bot-Commit: Autoroller Cr-Commit-Position: refs/heads/main@{#44626} --- DEPS | 216 +++++++++++++++++++++++++++++------------------------------ 1 file changed, 108 insertions(+), 108 deletions(-) diff --git a/DEPS b/DEPS index b458d6258a..0570ecd00e 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '4ad2db3c40b73c832a17c04293d7583634041f91', + 'chromium_revision': '35e96510ff321fc81e1dee2e666be935fee27fb3', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@2e5431a6db254f0cad1071ee2e88c0c0be219985', + 'https://chromium.googlesource.com/chromium/src/base@53335694d5aa1a2016df233a7ad30ad668d47a2a', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@0121a4c952cbf1556b17f68fed7c84c680dc940c', + 'https://chromium.googlesource.com/chromium/src/build@289b4d8c67486acccce147ae049e3582b2e63ae3', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@e9b36e9ffb02b268d194c3f82f89952d09d0c714', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@96b8ccee2c00d8355f2831f59e462d9925692134', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@e1277e5176140de22a409f34c66e82a7d6493dce', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@e0bf8ba12f5a76589729ee1b6d0476f337809cb3', + 'https://chromium.googlesource.com/chromium/src/testing@e7241882c3beb9d0f1e69a84a6c16e7f728db721', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@efebb4547796b4a29d2f3be63d8d254f99a10560', + 'https://chromium.googlesource.com/chromium/src/third_party@c1d0b85060486cde0fa11e0eab490f67d6bbd5d8', 'src/buildtools/linux64': { 'packages': [ @@ -142,143 +142,143 @@ deps = { 'objects': [ { # The Android libclang_rt.builtins libraries are currently only included in the Linux clang package. - 'object_name': 'Linux_x64/clang-llvmorg-21-init-9266-g09006611-2.tar.xz', - 'sha256sum': 'dee83e0a8b35564c277a7902a7872a77f4170b4380f9053acd3e1cb9a502a69c', - 'size_bytes': 55555048, - 'generation': 1746105739973602, + 'object_name': 'Linux_x64/clang-llvmorg-21-init-11777-gfd3fecfc-1.tar.xz', + 'sha256sum': '5f69279b3697166facfc354634157e0a8a32fa6e36864200ad8a8f85add3f3f6', + 'size_bytes': 55654416, + 'generation': 1747138963230938, 'condition': '(host_os == "linux" or checkout_android) and non_git_source', }, { - 'object_name': 'Linux_x64/clang-tidy-llvmorg-21-init-9266-g09006611-2.tar.xz', - 'sha256sum': 'bb8684fb985a35cc680bda9161735c409e6409106497f4ec0253ff9e07dd6912', - 'size_bytes': 13555996, - 'generation': 1746105739988254, + 'object_name': 'Linux_x64/clang-tidy-llvmorg-21-init-11777-gfd3fecfc-1.tar.xz', + 'sha256sum': '9c9cd089b46f36232d7553d03a0b30cf509f5e42b2113fe8172ba14f905b91df', + 'size_bytes': 13597708, + 'generation': 1747138963497696, 'condition': 'host_os == "linux" and checkout_clang_tidy and non_git_source', }, { - 'object_name': 'Linux_x64/clangd-llvmorg-21-init-9266-g09006611-2.tar.xz', - 'sha256sum': '5074d634984fa526b868126ead41c013de337edf36b369422005cb4a0edb95dc', - 'size_bytes': 13762680, - 'generation': 1746105740081588, + 'object_name': 'Linux_x64/clangd-llvmorg-21-init-11777-gfd3fecfc-1.tar.xz', + 'sha256sum': 'cf580450a46e262d899c2d48fff2b6ea8f6691a6bcfc0c280f87d8b6da088e5f', + 'size_bytes': 13854788, + 'generation': 1747138963598582, 'condition': 'host_os == "linux" and checkout_clangd and non_git_source', }, { - 'object_name': 'Linux_x64/llvm-code-coverage-llvmorg-21-init-9266-g09006611-2.tar.xz', - 'sha256sum': 'fa46a9d107f515d0a16c41ae5751a2a4be7be34a550f315a1506fcfb17327936', - 'size_bytes': 2294512, - 'generation': 1746105740273331, + 'object_name': 'Linux_x64/llvm-code-coverage-llvmorg-21-init-11777-gfd3fecfc-1.tar.xz', + 'sha256sum': '1a3488e55d62bda48d09b69fb4e5267ac64247258dc51fd9dabdb7fe5d66da4c', + 'size_bytes': 2297996, + 'generation': 1747138964068040, 'condition': 'host_os == "linux" and checkout_clang_coverage_tools and non_git_source', }, { - 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'object_name': 'Mac_arm64/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz', - 'sha256sum': '1aec99f479ff28cefe44ed739844833e016a1da255cf3c17d79e59a273246615', - 'size_bytes': 101605468, - 'generation': 1745271339727037, + 'object_name': 'Mac_arm64/rust-toolchain-4a0969e06dbeaaa43914d2d00b2e843d49aa3886-1-llvmorg-21-init-11777-gfd3fecfc.tar.xz', + 'sha256sum': 'd975e322a2e5c680b54f27a6545c63c0de2c5367ca6ffefda04de491d6b76553', + 'size_bytes': 102206348, + 'generation': 1747160501743827, 'condition': 'host_os == "mac" and host_cpu == "arm64"', }, { - 'object_name': 'Win/rust-toolchain-c8f94230282a8e8c1148f3e657f0199aad909228-1-llvmorg-21-init-9266-g09006611.tar.xz', - 'sha256sum': 'b291520613a3ebc415e4576a7fa31d840a5ebf4ab9be6e9dc5d90062dc001c1e', - 'size_bytes': 193280372, - 'generation': 1745271341223097, + 'object_name': 'Win/rust-toolchain-4a0969e06dbeaaa43914d2d00b2e843d49aa3886-1-llvmorg-21-init-11777-gfd3fecfc.tar.xz', + 'sha256sum': '25c51b4c2e0e8b4e974973f0357b12ecb645a0a7a136f9e0a6604e87c21acc7a', + 'size_bytes': 193582440, + 'generation': 1747160503376785, 'condition': 'host_os == "win"', }, ], @@ -329,7 +329,7 @@ deps = { 'src/third_party/libc++abi/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@069652e9cbca8e3b5e2371057a206f91f2f710f6', 'src/third_party/llvm-libc/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@27ee0730164e6f101985f057c6ad23a9ba7710f6', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@9aefa1cede964571c0b0ef41460cc5c3bbc6b4b8', 'src/third_party/libunwind/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@8575f4ae4fcf8892938bd9766cf1a5c90a0ed04e', @@ -480,11 +480,11 @@ deps = { }, 'src/third_party/boringssl/src': - 'https://boringssl.googlesource.com/boringssl.git@62956948e445c5bc67d3f299e0003c34a3030523', + 'https://boringssl.googlesource.com/boringssl.git@02bc0949e5cac0e1ee82c6f365f5a6c3cfd0cfa9', 'src/third_party/breakpad/breakpad': 'https://chromium.googlesource.com/breakpad/breakpad.git@fad82477312f786994151f01c352813fabf924c0', 'src/third_party/catapult': - 'https://chromium.googlesource.com/catapult.git@bbb14e46563c14a055b394f18b117e504e98d4ab', + 'https://chromium.googlesource.com/catapult.git@2fa93b2332b7a27c3066bef8d7ba9de2a5e1ab2e', 'src/third_party/ced/src': { 'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5', }, @@ -497,13 +497,13 @@ deps = { 'src/third_party/crc32c/src': 'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558', 'src/third_party/depot_tools': - 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@a9cc320bf761323906c614f7040cb1dc22fcc858', + 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@c273e3eb6202ef710a9cbf48fc99437a56a13631', 'src/third_party/ffmpeg': 'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@dcdd0fa51b65a0b1688ff6b8f0cc81908f09ded2', 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@708b5c0615d00637bea1babef94bfeb69f0ab5a3', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@3792c9842ab0a149237174633aacf1aed3d50575', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@2b4f2ff3824124542c33739f816406caed7fa2e8', + 'https://chromium.googlesource.com/chromium/src/tools@18608d93c9e76e96e6109a3979d7f507b212bf8b', 'src/third_party/espresso': { 'packages': [ From a86e045f89ce16865854e51d1e326b8ace39e629 Mon Sep 17 00:00:00 2001 From: webrtc-version-updater Date: Wed, 14 May 2025 21:06:40 -0700 Subject: [PATCH 155/292] Update WebRTC code version (2025-05-15T04:06:06). Bug: None Change-Id: I4b01e2ac296b7a4b594dc49290fce4c94a7a537a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391603 Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Cr-Commit-Position: refs/heads/main@{#44627} --- call/version.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/call/version.cc b/call/version.cc index e1f43fe5b1..3473abdf62 100644 --- a/call/version.cc +++ b/call/version.cc @@ -13,7 +13,7 @@ namespace webrtc { // The timestamp is always in UTC. -const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-14T04:07:42"; +const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-15T04:06:05"; void LoadWebRTCVersionInRegister() { // Using volatile to instruct the compiler to not optimize `p` away even From 55b5f0e429a572aabb28ab3b078f19a4365c1028 Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Tue, 13 May 2025 14:54:05 -0700 Subject: [PATCH 156/292] IWYU modules/video_capture/ without platform-specific subdirectories first. Done using ls modules/video_capture/*.{h,cc} | xargs tools_webrtc/iwyu/apply-include-cleaner | xargs tools_webrtc/iwyu/apply-include-cleaner followed by tools_webrtc/gn_check_autofix.py -C out/Default/ and git cl format Manual changes: - video_capture_impl.cc: use C++ headers Bug: webrtc:42226242 Change-Id: I2477021f0cfd9045cae4c5a789c4983a194ef7ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391262 Reviewed-by: Ilya Nikolaevskiy Reviewed-by: Danil Chapovalov Commit-Queue: Philipp Hancke Cr-Commit-Position: refs/heads/main@{#44628} --- modules/video_capture/BUILD.gn | 2 ++ .../video_capture/raw_video_sink_interface.h | 4 ++++ modules/video_capture/video_capture.h | 4 ++++ modules/video_capture/video_capture_defines.h | 3 ++- .../video_capture/video_capture_factory.cc | 2 ++ modules/video_capture/video_capture_factory.h | 1 - .../video_capture_factory_null.cc | 2 ++ modules/video_capture/video_capture_impl.cc | 19 +++++++++++++++---- modules/video_capture/video_capture_impl.h | 2 ++ .../video_capture/video_capture_options.cc | 3 +++ 10 files changed, 36 insertions(+), 6 deletions(-) diff --git a/modules/video_capture/BUILD.gn b/modules/video_capture/BUILD.gn index e5e74c9fc3..c3e0e5af71 100644 --- a/modules/video_capture/BUILD.gn +++ b/modules/video_capture/BUILD.gn @@ -28,6 +28,7 @@ rtc_library("video_capture_module") { ] deps = [ + "../../api:ref_count", "../../api:scoped_refptr", "../../api:sequence_checker", "../../api/video:video_frame", @@ -55,6 +56,7 @@ if (!build_with_chromium || is_linux || is_chromeos) { visibility = [ "*" ] deps = [ ":video_capture_module", + "../../api:make_ref_counted", "../../api:scoped_refptr", "../../api:sequence_checker", "../../media:video_common", diff --git a/modules/video_capture/raw_video_sink_interface.h b/modules/video_capture/raw_video_sink_interface.h index 094e9e20bd..a22ca9811d 100644 --- a/modules/video_capture/raw_video_sink_interface.h +++ b/modules/video_capture/raw_video_sink_interface.h @@ -14,6 +14,10 @@ #ifndef MODULES_VIDEO_CAPTURE_RAW_VIDEO_SINK_INTERFACE_H_ #define MODULES_VIDEO_CAPTURE_RAW_VIDEO_SINK_INTERFACE_H_ +#include +#include + +#include "api/video/video_rotation.h" #include "modules/video_capture/video_capture_defines.h" namespace webrtc { diff --git a/modules/video_capture/video_capture.h b/modules/video_capture/video_capture.h index fe5c09bd87..7707608dbc 100644 --- a/modules/video_capture/video_capture.h +++ b/modules/video_capture/video_capture.h @@ -11,6 +11,10 @@ #ifndef MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_H_ #define MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_H_ +#include + +#include "api/ref_count.h" +#include "api/video/video_frame.h" #include "api/video/video_rotation.h" #include "api/video/video_sink_interface.h" #include "modules/video_capture/raw_video_sink_interface.h" diff --git a/modules/video_capture/video_capture_defines.h b/modules/video_capture/video_capture_defines.h index 63534600a9..f814e4e7f5 100644 --- a/modules/video_capture/video_capture_defines.h +++ b/modules/video_capture/video_capture_defines.h @@ -11,7 +11,8 @@ #ifndef MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_DEFINES_H_ #define MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_DEFINES_H_ -#include "api/video/video_frame.h" +#include + #include "common_video/libyuv/include/webrtc_libyuv.h" namespace webrtc { diff --git a/modules/video_capture/video_capture_factory.cc b/modules/video_capture/video_capture_factory.cc index 8354067f93..d7fa0fa26e 100644 --- a/modules/video_capture/video_capture_factory.cc +++ b/modules/video_capture/video_capture_factory.cc @@ -10,6 +10,8 @@ #include "modules/video_capture/video_capture_factory.h" +#include "api/scoped_refptr.h" +#include "modules/video_capture/video_capture.h" #include "modules/video_capture/video_capture_impl.h" namespace webrtc { diff --git a/modules/video_capture/video_capture_factory.h b/modules/video_capture/video_capture_factory.h index c24eaf559f..e669470e35 100644 --- a/modules/video_capture/video_capture_factory.h +++ b/modules/video_capture/video_capture_factory.h @@ -16,7 +16,6 @@ #include "api/scoped_refptr.h" #include "modules/video_capture/video_capture.h" -#include "modules/video_capture/video_capture_defines.h" #include "rtc_base/system/rtc_export.h" namespace webrtc { diff --git a/modules/video_capture/video_capture_factory_null.cc b/modules/video_capture/video_capture_factory_null.cc index 77b7f4b75e..4dbc7b9f5c 100644 --- a/modules/video_capture/video_capture_factory_null.cc +++ b/modules/video_capture/video_capture_factory_null.cc @@ -8,6 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include "api/scoped_refptr.h" +#include "modules/video_capture/video_capture.h" #include "modules/video_capture/video_capture_impl.h" namespace webrtc { diff --git a/modules/video_capture/video_capture_impl.cc b/modules/video_capture/video_capture_impl.cc index 06f9bd3165..f12631c274 100644 --- a/modules/video_capture/video_capture_impl.cc +++ b/modules/video_capture/video_capture_impl.cc @@ -10,17 +10,28 @@ #include "modules/video_capture/video_capture_impl.h" -#include -#include +#include +#include +#include +#include "api/scoped_refptr.h" +#include "api/sequence_checker.h" #include "api/video/i420_buffer.h" -#include "api/video/video_frame_buffer.h" +#include "api/video/video_frame.h" +#include "api/video/video_rotation.h" +#include "api/video/video_sink_interface.h" #include "common_video/libyuv/include/webrtc_libyuv.h" +#include "libyuv/convert.h" +#include "libyuv/rotate.h" +#include "modules/video_capture/raw_video_sink_interface.h" #include "modules/video_capture/video_capture_config.h" +#include "modules/video_capture/video_capture_defines.h" +#include "rtc_base/checks.h" #include "rtc_base/logging.h" +#include "rtc_base/race_checker.h" +#include "rtc_base/synchronization/mutex.h" #include "rtc_base/time_utils.h" #include "rtc_base/trace_event.h" -#include "third_party/libyuv/include/libyuv.h" namespace webrtc { namespace videocapturemodule { diff --git a/modules/video_capture/video_capture_impl.h b/modules/video_capture/video_capture_impl.h index 3ec832f871..d3ac5fe5ef 100644 --- a/modules/video_capture/video_capture_impl.h +++ b/modules/video_capture/video_capture_impl.h @@ -23,12 +23,14 @@ #include "api/video/video_frame.h" #include "api/video/video_rotation.h" #include "api/video/video_sink_interface.h" +#include "modules/video_capture/raw_video_sink_interface.h" #include "modules/video_capture/video_capture.h" #include "modules/video_capture/video_capture_config.h" #include "modules/video_capture/video_capture_defines.h" #include "rtc_base/race_checker.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/system/rtc_export.h" +#include "rtc_base/thread_annotations.h" namespace webrtc { diff --git a/modules/video_capture/video_capture_options.cc b/modules/video_capture/video_capture_options.cc index 64cd6216e7..dd102df9db 100644 --- a/modules/video_capture/video_capture_options.cc +++ b/modules/video_capture/video_capture_options.cc @@ -10,6 +10,9 @@ #include "modules/video_capture/video_capture_options.h" +#include "api/make_ref_counted.h" +#include "api/scoped_refptr.h" + #if defined(WEBRTC_USE_PIPEWIRE) #include "modules/video_capture/linux/pipewire_session.h" #endif From dff5f9c27f4bfad368e8db0eb7dfd25391fe8bcc Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Wed, 14 May 2025 17:15:30 -0700 Subject: [PATCH 157/292] Count tests (run, disabled, skipped) and export as metrics which allows looking at graphs and shows trends in test usage. Bug: None No-IWYU: cross-platform false positive and stdlib false positive Change-Id: I7b71bd515b986f45873f03be74019131e13dd797 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390301 Commit-Queue: Philipp Hancke Reviewed-by: Christoffer Dewerin Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44629} --- test/BUILD.gn | 2 ++ test/test_main_lib.cc | 31 +++++++++++++++++++++++---- test/testsupport/file_utils.cc | 39 +++++++++++++++++++--------------- test/testsupport/file_utils.h | 6 ++++-- 4 files changed, 55 insertions(+), 23 deletions(-) diff --git a/test/BUILD.gn b/test/BUILD.gn index ad3012d1dc..eb8bd0ddf5 100644 --- a/test/BUILD.gn +++ b/test/BUILD.gn @@ -587,12 +587,14 @@ if (rtc_include_tests) { deps = [ ":field_trial", + ":fileutils", ":perf_test", ":resources_dir_flag", ":test_flags", ":test_support", "../api/test/metrics:chrome_perf_dashboard_metrics_exporter", "../api/test/metrics:global_metrics_logger_and_exporter", + "../api/test/metrics:metric", "../api/test/metrics:metrics_exporter", "../api/test/metrics:metrics_set_proto_file_exporter", "../api/test/metrics:print_result_proxy_metrics_exporter", diff --git a/test/test_main_lib.cc b/test/test_main_lib.cc index 8aa63c395d..5348d1c0be 100644 --- a/test/test_main_lib.cc +++ b/test/test_main_lib.cc @@ -10,10 +10,8 @@ #include "test/test_main_lib.h" -#include - #include -#include +#include #include #include #include @@ -25,11 +23,11 @@ #include "absl/strings/string_view.h" #include "api/test/metrics/chrome_perf_dashboard_metrics_exporter.h" #include "api/test/metrics/global_metrics_logger_and_exporter.h" +#include "api/test/metrics/metric.h" #include "api/test/metrics/metrics_exporter.h" #include "api/test/metrics/metrics_set_proto_file_exporter.h" #include "api/test/metrics/print_result_proxy_metrics_exporter.h" #include "api/test/metrics/stdout_metrics_exporter.h" -#include "rtc_base/checks.h" #include "rtc_base/event_tracer.h" #include "rtc_base/logging.h" #include "rtc_base/ssl_adapter.h" @@ -38,6 +36,7 @@ #include "system_wrappers/include/metrics.h" #include "test/gtest.h" #include "test/test_flags.h" +#include "test/testsupport/file_utils.h" #include "test/testsupport/perf_test.h" #include "test/testsupport/resources_dir_flag.h" @@ -217,6 +216,30 @@ class TestMainImpl : public TestMain { exporters.push_back( std::make_unique()); } + // Log number of tests that should be run, are disabled or skipped and total + // number. + int total_test_count = + ::testing::UnitTest::GetInstance()->total_test_count(); + int test_to_run_count = + ::testing::UnitTest::GetInstance()->test_to_run_count(); + int disabled_test_count = + ::testing::UnitTest::GetInstance()->disabled_test_count(); + int skipped_test_count = + ::testing::UnitTest::GetInstance()->skipped_test_count(); + absl::string_view test_suite_name = test::FileName(argv[0]); + test::GetGlobalMetricsLogger()->LogSingleValueMetric( + "TotalTestCount", test_suite_name, total_test_count, test::Unit::kCount, + test::ImprovementDirection::kBiggerIsBetter); + test::GetGlobalMetricsLogger()->LogSingleValueMetric( + "RunTestCount", test_suite_name, test_to_run_count, test::Unit::kCount, + test::ImprovementDirection::kBiggerIsBetter); + test::GetGlobalMetricsLogger()->LogSingleValueMetric( + "DisabledTestCount", test_suite_name, disabled_test_count, + test::Unit::kCount, test::ImprovementDirection::kSmallerIsBetter); + test::GetGlobalMetricsLogger()->LogSingleValueMetric( + "SkippedTestCount", test_suite_name, skipped_test_count, + test::Unit::kCount, test::ImprovementDirection::kSmallerIsBetter); + test::ExportPerfMetric(*test::GetGlobalMetricsLogger(), std::move(exporters)); if (!absl::GetFlag(FLAGS_export_perf_results_new_api)) { diff --git a/test/testsupport/file_utils.cc b/test/testsupport/file_utils.cc index e41046c7f9..3148863536 100644 --- a/test/testsupport/file_utils.cc +++ b/test/testsupport/file_utils.cc @@ -10,6 +10,21 @@ #include "test/testsupport/file_utils.h" +#include +#include +#include +#include +#include +#include + +#include "absl/base/attributes.h" +#include "absl/strings/string_view.h" +#include "rtc_base/checks.h" +#include "rtc_base/crypto_random.h" +#include "rtc_base/string_utils.h" +#include "rtc_base/strings/string_builder.h" +#include "test/testsupport/file_utils_override.h" + #if defined(WEBRTC_POSIX) #include #endif @@ -38,29 +53,12 @@ #define S_ISDIR(mode) (((mode) & S_IFMT) == S_IFDIR) #endif -#include -#include - -#include -#include -#include -#include -#include -#include - #if defined(WEBRTC_IOS) #include "test/testsupport/ios_file_utils.h" #elif defined(WEBRTC_MAC) #include "test/testsupport/mac_file_utils.h" #endif -#include "absl/strings/string_view.h" -#include "rtc_base/checks.h" -#include "rtc_base/crypto_random.h" -#include "rtc_base/string_utils.h" -#include "rtc_base/strings/string_builder.h" -#include "test/testsupport/file_utils_override.h" - namespace webrtc { namespace test { @@ -82,6 +80,13 @@ std::string DirName(absl::string_view path) { return std::string(path.substr(0, path.find_last_of(kPathDelimiter))); } +absl::string_view FileName(absl::string_view path) { + if (path.find_last_of(kPathDelimiter) == absl::string_view::npos) { + return path; + } + return path.substr(path.find_last_of(kPathDelimiter) + 1); +} + bool FileExists(absl::string_view file_name) { struct stat file_info = {0}; return stat(std::string(file_name).c_str(), &file_info) == 0; diff --git a/test/testsupport/file_utils.h b/test/testsupport/file_utils.h index b0652eafd6..4e5b268c35 100644 --- a/test/testsupport/file_utils.h +++ b/test/testsupport/file_utils.h @@ -8,11 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include - #ifndef TEST_TESTSUPPORT_FILE_UTILS_H_ #define TEST_TESTSUPPORT_FILE_UTILS_H_ +#include #include #include #include @@ -107,6 +106,9 @@ bool DirExists(absl::string_view directory_name); // Strips the rightmost path segment from a path. std::string DirName(absl::string_view path); +// Returns the rightmost path segment from a path. +absl::string_view FileName(absl::string_view path); + // File size of the supplied file in bytes. Will return 0 if the file is // empty or if the file does not exist/is readable. size_t GetFileSize(absl::string_view filename); From d78a854c567e72c9686e2aef16bd2f4ea8bcdb29 Mon Sep 17 00:00:00 2001 From: Harald Alvestrand Date: Wed, 14 May 2025 21:35:20 -0700 Subject: [PATCH 158/292] Revert "Use only PayloadTypeSuggester for PT assigment" This reverts commit e12bd20849d40501e17c33b74adce2aaf6c2a755. Reason for revert: b/409355977 - sometimes caused PT clashes Bug: webrtc:360058654 Original change's description: > Use only PayloadTypeSuggester for PT assigment > > This merges two relands: > Reland "Stop using the UsedPayloadTypes class" > This reverts commit ebb67395c1a7c81ec592f45d2abfdaf7f8f3e722. > Reland "Use only one kind of codec list merging in CodecVendor." > This reverts commit 4aa2908fee6a0d286449172565604a04ebfdea7c. > > The UsedPayloadTypes class has been removed. > > Bug: webrtc:360058654 > Change-Id: I72b8e4f336e37e9a276c6a449482c3b7f46fdc65 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/385223 > Reviewed-by: Evan Shrubsole > Commit-Queue: Harald Alvestrand > Cr-Commit-Position: refs/heads/main@{#44547} No-IWYU: Rollback Bug: webrtc:360058654 Change-Id: Ifd62f359b457f8a88208fb4c082f6bcac3275e8f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391700 Commit-Queue: Harald Alvestrand Bot-Commit: rubber-stamper@appspot.gserviceaccount.com Reviewed-by: Mirko Bonadei Cr-Commit-Position: refs/heads/main@{#44630} --- pc/codec_vendor.cc | 211 ++++++++++++++++++++++++++++++++------------- pc/codec_vendor.h | 3 + pc/used_ids.h | 38 ++++++++ 3 files changed, 193 insertions(+), 59 deletions(-) diff --git a/pc/codec_vendor.cc b/pc/codec_vendor.cc index e45516c767..42d52f6487 100644 --- a/pc/codec_vendor.cc +++ b/pc/codec_vendor.cc @@ -39,6 +39,7 @@ #include "pc/rtp_media_utils.h" #include "pc/session_description.h" #include "pc/typed_codec_vendor.h" +#include "pc/used_ids.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/string_encode.h" @@ -53,6 +54,11 @@ namespace webrtc { namespace { +using webrtc::PayloadTypeSuggester; +using webrtc::RTCError; +using webrtc::RTCErrorOr; +using webrtc::RtpTransceiverDirection; + bool IsRtxCodec(const RtpCodecCapability& capability) { return absl::EqualsIgnoreCase(capability.name, kRtxCodecName); } @@ -69,7 +75,8 @@ bool IsComfortNoiseCodec(const Codec& codec) { std::optional FindMatchingCodec(const CodecList& codecs1, const CodecList& codecs2, const Codec& codec_to_match) { - return FindMatchingCodec(codecs1.codecs(), codecs2.codecs(), codec_to_match); + return webrtc::FindMatchingCodec(codecs1.codecs(), codecs2.codecs(), + codec_to_match); } void StripCNCodecs(CodecList& audio_codecs) { @@ -98,7 +105,7 @@ const Codec* GetAssociatedCodecForRtx(const CodecList& codec_list, } int associated_pt; - if (!FromString(associated_pt_str, &associated_pt)) { + if (!webrtc::FromString(associated_pt_str, &associated_pt)) { RTC_LOG(LS_WARNING) << "Couldn't convert payload type " << associated_pt_str << " of RTX codec " << rtx_codec.id << " to an integer."; @@ -129,14 +136,14 @@ const Codec* GetAssociatedCodecForRed(const CodecList& codec_list, return nullptr; } - std::vector redundant_payloads = split(fmtp, '/'); + std::vector redundant_payloads = webrtc::split(fmtp, '/'); if (redundant_payloads.size() < 2) { return nullptr; } absl::string_view associated_pt_str = redundant_payloads[0]; int associated_pt; - if (!FromString(associated_pt_str, &associated_pt)) { + if (!webrtc::FromString(associated_pt_str, &associated_pt)) { RTC_LOG(LS_WARNING) << "Couldn't convert first payload type " << associated_pt_str << " of RED codec " << red_codec.id << " to an integer."; @@ -175,21 +182,7 @@ RTCError MergeCodecs(const CodecList& reference_codecs, return suggestion.MoveError(); } codec.id = suggestion.value(); - // The rewrite of the parameter may have turned the codec into - // one that is already present. - bool skip = false; - for (const Codec& present_codec : offered_codecs) { - if (present_codec.id == codec.id) { - RTC_DCHECK(MatchesWithReferenceAttributes(present_codec, codec)) - << "Codec mismatch: present " << present_codec << ", new " - << codec; - skip = true; - break; - } - } - if (!skip) { - offered_codecs.push_back(codec); - } + offered_codecs.push_back(codec); } } @@ -239,7 +232,7 @@ RTCError MergeCodecs(const CodecList& reference_codecs, } std::string red_param = absl::StrCat(matching_codec->id); red_codec.params[kCodecParamNotInNameValueFormat] = - StrJoin(std::vector{red_param, red_param}, "/"); + webrtc::StrJoin(std::vector{red_param, red_param}, "/"); } RTCErrorOr suggestion = pt_suggester.SuggestPayloadType(mid, red_codec); @@ -247,23 +240,82 @@ RTCError MergeCodecs(const CodecList& reference_codecs, return suggestion.MoveError(); } red_codec.id = suggestion.value(); - // The rewrite of the parameter may have turned the RED codec into - // one that is already present. - bool skip = false; - for (const Codec& present_codec : offered_codecs) { - if (present_codec.id == red_codec.id) { - RTC_DCHECK(MatchesWithReferenceAttributes(present_codec, red_codec)); - skip = true; - break; - } + offered_codecs.push_back(red_codec); + } + } + offered_codecs.CheckConsistency(); + return RTCError::OK(); +} + +// Adds all codecs from `reference_codecs` to `offered_codecs` that don't +// already exist in `offered_codecs` and ensure the payload types don't +// collide. +// OLD VERSION - uses UsedPayloadTypes +void MergeCodecs(const CodecList& reference_codecs, + CodecList& offered_codecs, + UsedPayloadTypes* used_pltypes) { + // Add all new codecs that are not RTX/RED codecs. + // The two-pass splitting of the loops means preferring payload types + // of actual codecs with respect to collisions. + for (const Codec& reference_codec : reference_codecs) { + if (reference_codec.GetResiliencyType() != Codec::ResiliencyType::kRtx && + reference_codec.GetResiliencyType() != Codec::ResiliencyType::kRed && + !FindMatchingCodec(reference_codecs, offered_codecs, reference_codec)) { + Codec codec = reference_codec; + used_pltypes->FindAndSetIdUsed(&codec); + offered_codecs.push_back(codec); + } + } + + // Add all new RTX or RED codecs. + for (const Codec& reference_codec : reference_codecs) { + if (reference_codec.GetResiliencyType() == Codec::ResiliencyType::kRtx && + !FindMatchingCodec(reference_codecs, offered_codecs, reference_codec)) { + Codec rtx_codec = reference_codec; + const Codec* associated_codec = + GetAssociatedCodecForRtx(reference_codecs, rtx_codec); + if (!associated_codec) { + continue; } - if (!skip) { - offered_codecs.push_back(red_codec); + // Find a codec in the offered list that matches the reference codec. + // Its payload type may be different than the reference codec. + std::optional matching_codec = FindMatchingCodec( + reference_codecs, offered_codecs, *associated_codec); + if (!matching_codec) { + RTC_LOG(LS_WARNING) + << "Couldn't find matching " << associated_codec->name << " codec."; + continue; } + + rtx_codec.params[kCodecParamAssociatedPayloadType] = + absl::StrCat(matching_codec->id); + used_pltypes->FindAndSetIdUsed(&rtx_codec); + offered_codecs.push_back(rtx_codec); + } else if (reference_codec.GetResiliencyType() == + Codec::ResiliencyType::kRed && + !FindMatchingCodec(reference_codecs, offered_codecs, + reference_codec)) { + Codec red_codec = reference_codec; + const Codec* associated_codec = + GetAssociatedCodecForRed(reference_codecs, red_codec); + if (associated_codec) { + std::optional matching_codec = FindMatchingCodec( + reference_codecs, offered_codecs, *associated_codec); + if (!matching_codec) { + RTC_LOG(LS_WARNING) << "Couldn't find matching " + << associated_codec->name << " codec."; + continue; + } + + red_codec.params[kCodecParamNotInNameValueFormat] = + absl::StrCat(matching_codec->id) + "/" + + absl::StrCat(matching_codec->id); + } + used_pltypes->FindAndSetIdUsed(&red_codec); + offered_codecs.push_back(red_codec); } } offered_codecs.CheckConsistency(); - return RTCError::OK(); } // `codecs` is a full list of codecs with correct payload type mappings, which @@ -330,7 +382,7 @@ CodecList MatchCodecPreference( codec.params.find(kCodecParamNotInNameValueFormat); if (fmtp != codec.params.end()) { std::vector redundant_payloads = - split(fmtp->second, '/'); + webrtc::split(fmtp->second, '/'); if (!redundant_payloads.empty() && redundant_payloads[0] == id) { if (!red_was_added) { @@ -391,7 +443,7 @@ void NegotiateVideoCodecLevelsForOffer( for (const Codec& supported_codec : supported_codecs) { if (absl::EqualsIgnoreCase(supported_codec.name, kH265CodecName)) { std::optional supported_ptl = - ParseSdpForH265ProfileTierLevel(supported_codec.params); + webrtc::ParseSdpForH265ProfileTierLevel(supported_codec.params); if (supported_ptl.has_value()) { supported_h265_profiles[supported_ptl->profile] = supported_ptl->level; @@ -406,14 +458,14 @@ void NegotiateVideoCodecLevelsForOffer( for (auto& filtered_codec : filtered_codecs) { if (absl::EqualsIgnoreCase(filtered_codec.name, kH265CodecName)) { std::optional filtered_ptl = - ParseSdpForH265ProfileTierLevel(filtered_codec.params); + webrtc::ParseSdpForH265ProfileTierLevel(filtered_codec.params); if (filtered_ptl.has_value()) { auto it = supported_h265_profiles.find(filtered_ptl->profile); if (it != supported_h265_profiles.end() && filtered_ptl->level != it->second) { filtered_codec.params[kH265FmtpLevelId] = - H265LevelToString(it->second); + webrtc::H265LevelToString(it->second); } } } @@ -454,13 +506,13 @@ RTCError NegotiateCodecs(const CodecList& local_codecs, } } if (absl::EqualsIgnoreCase(ours.name, kH264CodecName)) { - H264GenerateProfileLevelIdForAnswer(ours.params, theirs->params, - &negotiated.params); + webrtc::H264GenerateProfileLevelIdForAnswer(ours.params, theirs->params, + &negotiated.params); } #ifdef RTC_ENABLE_H265 if (absl::EqualsIgnoreCase(ours.name, kH265CodecName)) { - H265GenerateProfileTierLevelForAnswer(ours.params, theirs->params, - &negotiated.params); + webrtc::H265GenerateProfileTierLevelForAnswer( + ours.params, theirs->params, &negotiated.params); NegotiateTxMode(ours, *theirs, &negotiated); } #endif @@ -482,7 +534,7 @@ RTCError NegotiateCodecs(const CodecList& local_codecs, continue; } int apt_value; - if (!FromString(apt_str, &apt_value)) { + if (!webrtc::FromString(apt_str, &apt_value)) { RTC_LOG(LS_WARNING) << "Unconvertable apt value"; continue; } @@ -582,11 +634,9 @@ RTCErrorOr> CodecVendor::GetNegotiatedCodecsForOffer( } // Add our codecs that are not in the current description. if (media_description_options.type == MediaType::AUDIO) { - MergeCodecs(audio_recv_codecs(), mid, codecs, pt_suggester); - MergeCodecs(audio_send_codecs(), mid, codecs, pt_suggester); + MergeCodecs(all_audio_codecs(), mid, codecs, pt_suggester); } else { - MergeCodecs(video_recv_codecs(), mid, codecs, pt_suggester); - MergeCodecs(video_send_codecs(), mid, codecs, pt_suggester); + MergeCodecs(all_video_codecs(), mid, codecs, pt_suggester); } CodecList filtered_codecs; CodecList supported_codecs = @@ -617,11 +667,20 @@ RTCErrorOr> CodecVendor::GetNegotiatedCodecsForOffer( const MediaContentDescription* mcd = current_content->media_description(); for (const Codec& codec : mcd->codecs()) { - if (FindMatchingCodec(mcd->codecs(), codecs.codecs(), codec)) { + if (webrtc::FindMatchingCodec(mcd->codecs(), codecs.codecs(), + codec)) { filtered_codecs.push_back(codec); } } } + // Note what PTs are already in use. + UsedPayloadTypes + used_pltypes; // Used to avoid pt collisions in filtered_codecs + for (auto& codec : filtered_codecs) { + // Note: This may change PTs. Doing so woud indicate an error, but + // UsedPayloadTypes doesn't offer a means to make the distinction. + used_pltypes.FindAndSetIdUsed(&codec); + } // Add other supported codecs. for (const Codec& codec : supported_codecs) { std::optional found_codec = @@ -648,11 +707,8 @@ RTCErrorOr> CodecVendor::GetNegotiatedCodecsForOffer( changed_referenced_codec->id); } } - auto pt_or_error = pt_suggester.SuggestPayloadType(mid, *found_codec); - if (!pt_or_error.ok()) { - return pt_or_error.MoveError(); - } - found_codec->id = pt_or_error.value(); + // Quick fix for b/395077842: Remap the codec if it collides. + used_pltypes.FindAndSetIdUsed(&(*found_codec)); filtered_codecs.push_back(*found_codec); } } @@ -707,11 +763,9 @@ RTCErrorOr CodecVendor::GetNegotiatedCodecsForAnswer( } // Add all our supported codecs if (media_description_options.type == MediaType::AUDIO) { - MergeCodecs(audio_recv_codecs(), mid, codecs, pt_suggester); - MergeCodecs(audio_send_codecs(), mid, codecs, pt_suggester); + MergeCodecs(all_audio_codecs(), mid, codecs, pt_suggester); } else { - MergeCodecs(video_recv_codecs(), mid, codecs, pt_suggester); - MergeCodecs(video_send_codecs(), mid, codecs, pt_suggester); + MergeCodecs(all_video_codecs(), mid, codecs, pt_suggester); } CodecList filtered_codecs; CodecList negotiated_codecs; @@ -740,8 +794,8 @@ RTCErrorOr CodecVendor::GetNegotiatedCodecsForAnswer( const MediaContentDescription* mcd = current_content->media_description(); for (const Codec& codec : mcd->codecs()) { - if (std::optional found_codec = - FindMatchingCodec(mcd->codecs(), codecs.codecs(), codec)) { + if (std::optional found_codec = webrtc::FindMatchingCodec( + mcd->codecs(), codecs.codecs(), codec)) { filtered_codecs.push_back(*found_codec); } } @@ -859,7 +913,8 @@ CodecList CodecVendor::GetVideoCodecsForAnswer( case RtpTransceiverDirection::kSendRecv: case RtpTransceiverDirection::kStopped: case RtpTransceiverDirection::kInactive: - return GetVideoCodecsForOffer(RtpTransceiverDirectionReversed(offer)); + return GetVideoCodecsForOffer( + webrtc::RtpTransceiverDirectionReversed(offer)); case RtpTransceiverDirection::kSendOnly: return video_send_codecs_.codecs(); case RtpTransceiverDirection::kRecvOnly: @@ -893,7 +948,8 @@ CodecList CodecVendor::GetAudioCodecsForAnswer( case RtpTransceiverDirection::kSendRecv: case RtpTransceiverDirection::kStopped: case RtpTransceiverDirection::kInactive: - return GetAudioCodecsForOffer(RtpTransceiverDirectionReversed(offer)); + return GetAudioCodecsForOffer( + webrtc::RtpTransceiverDirectionReversed(offer)); case RtpTransceiverDirection::kSendOnly: return audio_send_codecs_.codecs(); case RtpTransceiverDirection::kRecvOnly: @@ -902,6 +958,43 @@ CodecList CodecVendor::GetAudioCodecsForAnswer( RTC_CHECK_NOTREACHED(); } +CodecList CodecVendor::all_video_codecs() const { + CodecList all_codecs; + UsedPayloadTypes used_payload_types; + for (const Codec& codec : video_recv_codecs_.codecs()) { + Codec codec_mutable = codec; + used_payload_types.FindAndSetIdUsed(&codec_mutable); + all_codecs.push_back(codec_mutable); + } + + // Use MergeCodecs to merge the second half of our list as it already checks + // and fixes problems with duplicate payload types. + MergeCodecs(video_send_codecs_.codecs(), all_codecs, &used_payload_types); + + return all_codecs; +} + +CodecList CodecVendor::all_audio_codecs() const { + // Compute the audio codecs union. + CodecList codecs; + for (const Codec& send : audio_send_codecs_.codecs()) { + codecs.push_back(send); + if (!FindMatchingCodec(audio_send_codecs_.codecs(), + audio_recv_codecs_.codecs(), send)) { + // It doesn't make sense to have an RTX codec we support sending but not + // receiving. + RTC_DCHECK(send.GetResiliencyType() != Codec::ResiliencyType::kRtx); + } + } + for (const Codec& recv : audio_recv_codecs_.codecs()) { + if (!FindMatchingCodec(audio_recv_codecs_.codecs(), + audio_send_codecs_.codecs(), recv)) { + codecs.push_back(recv); + } + } + return codecs; +} + CodecList CodecVendor::audio_sendrecv_codecs() const { // Use NegotiateCodecs to merge our codec lists, since the operation is // essentially the same. Put send_codecs as the offered_codecs, which is the diff --git a/pc/codec_vendor.h b/pc/codec_vendor.h index 4a3a8d92b6..1472225a00 100644 --- a/pc/codec_vendor.h +++ b/pc/codec_vendor.h @@ -102,6 +102,9 @@ class CodecVendor { const RtpTransceiverDirection& offer, const RtpTransceiverDirection& answer) const; + CodecList all_video_codecs() const; + CodecList all_audio_codecs() const; + TypedCodecVendor audio_send_codecs_; TypedCodecVendor audio_recv_codecs_; diff --git a/pc/used_ids.h b/pc/used_ids.h index f9d825c957..fe80531006 100644 --- a/pc/used_ids.h +++ b/pc/used_ids.h @@ -14,7 +14,9 @@ #include #include "api/rtp_parameters.h" +#include "media/base/codec.h" #include "rtc_base/checks.h" +#include "rtc_base/logging.h" namespace webrtc { template @@ -86,6 +88,41 @@ class UsedIds { std::set id_set_; }; +// Helper class used for finding duplicate RTP payload types among audio, video +// and data codecs. When bundle is used the payload types may not collide. +class UsedPayloadTypes : public UsedIds { + public: + UsedPayloadTypes() + : UsedIds(kFirstDynamicPayloadTypeLowerRange, + kLastDynamicPayloadTypeUpperRange) {} + + // Check if a payload type is valid. The range [64-95] is forbidden + // when rtcp-mux is used. + static bool IsIdValid(Codec codec, bool rtcp_mux) { + if (rtcp_mux && (codec.id > kLastDynamicPayloadTypeLowerRange && + codec.id < kFirstDynamicPayloadTypeUpperRange)) { + return false; + } + return codec.id >= 0 && codec.id <= kLastDynamicPayloadTypeUpperRange; + } + + protected: + bool IsIdUsed(int new_id) override { + // Range marked for RTCP avoidance is "used". + if (new_id > kLastDynamicPayloadTypeLowerRange && + new_id < kFirstDynamicPayloadTypeUpperRange) + return true; + return UsedIds::IsIdUsed(new_id); + } + + private: + static const int kFirstDynamicPayloadTypeLowerRange = 35; + static const int kLastDynamicPayloadTypeLowerRange = 63; + + static const int kFirstDynamicPayloadTypeUpperRange = 96; + static const int kLastDynamicPayloadTypeUpperRange = 127; +}; + // Helper class used for finding duplicate RTP Header extension ids among // audio and video extensions. class UsedRtpHeaderExtensionIds : public UsedIds { @@ -153,6 +190,7 @@ class UsedRtpHeaderExtensionIds : public UsedIds { #ifdef WEBRTC_ALLOW_DEPRECATED_NAMESPACES namespace cricket { using ::webrtc::UsedIds; +using ::webrtc::UsedPayloadTypes; using ::webrtc::UsedRtpHeaderExtensionIds; } // namespace cricket #endif // WEBRTC_ALLOW_DEPRECATED_NAMESPACES From 60c61fc56bce2985b3a2130412029519a9ea4517 Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Wed, 14 May 2025 16:14:37 +0000 Subject: [PATCH 159/292] Allow to inject webrtc::Environment to create android AudioDeviceModule Deprecate api that doesn't propagate Environment. This would allow adm to share task queue factory with other components and use propagated field trials. Bug: webrtc:413413572 Change-Id: Ia70a93a636de57ce2764355f80b641c1019ef6b8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391300 Commit-Queue: Danil Chapovalov Reviewed-by: Henrik Andreassson Cr-Commit-Position: refs/heads/main@{#44631} --- examples/androidvoip/BUILD.gn | 2 +- .../androidvoip/jni/android_voip_client.cc | 8 +- modules/audio_device/BUILD.gn | 2 - modules/audio_device/audio_device_impl.cc | 4 - sdk/android/BUILD.gn | 29 +++++- .../audio_device_android.cc | 96 ++++++++++++++----- .../audio_device_android.h | 29 ++++-- .../audio_device/audio_device_unittest.cc | 21 ++-- .../peer_connection_factory_unittest.cc | 5 +- .../jni/audio_device/audio_device_module.cc | 32 +++++-- .../jni/audio_device/audio_device_module.h | 12 +++ .../audio_device/java_audio_device_module.cc | 15 +-- 12 files changed, 190 insertions(+), 65 deletions(-) diff --git a/examples/androidvoip/BUILD.gn b/examples/androidvoip/BUILD.gn index 05bccd7f77..4c8af525ec 100644 --- a/examples/androidvoip/BUILD.gn +++ b/examples/androidvoip/BUILD.gn @@ -57,6 +57,7 @@ if (is_android) { deps = [ ":generated_jni", "../../api/audio:builtin_audio_processing_builder", + "../../api/environment:environment_factory", "../../rtc_base:async_packet_socket", "../../rtc_base:async_udp_socket", "../../rtc_base:logging", @@ -69,7 +70,6 @@ if (is_android) { "//api/audio_codecs:audio_codecs_api", "//api/audio_codecs:builtin_audio_decoder_factory", "//api/audio_codecs:builtin_audio_encoder_factory", - "//api/task_queue:default_task_queue_factory", "//api/units:time_delta", "//api/voip:voip_api", "//api/voip:voip_engine_factory", diff --git a/examples/androidvoip/jni/android_voip_client.cc b/examples/androidvoip/jni/android_voip_client.cc index cea3d48582..ceef8e925b 100644 --- a/examples/androidvoip/jni/android_voip_client.cc +++ b/examples/androidvoip/jni/android_voip_client.cc @@ -23,7 +23,7 @@ #include "api/audio/builtin_audio_processing_builder.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" -#include "api/task_queue/default_task_queue_factory.h" +#include "api/environment/environment_factory.h" #include "api/units/time_delta.h" #include "api/voip/voip_base.h" #include "api/voip/voip_codec.h" @@ -127,9 +127,9 @@ void AndroidVoipClient::Init( webrtc::VoipEngineConfig config; config.encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory(); config.decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory(); - config.task_queue_factory = webrtc::CreateDefaultTaskQueueFactory(); - config.audio_device_module = - webrtc::CreateJavaAudioDeviceModule(env, application_context.obj()); + config.env = webrtc::CreateEnvironment(); + config.audio_device_module = webrtc::CreateJavaAudioDeviceModule( + env, *config.env, application_context.obj()); config.audio_processing_builder = std::make_unique(); diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn index 6ddd196126..9c9083d081 100644 --- a/modules/audio_device/BUILD.gn +++ b/modules/audio_device/BUILD.gn @@ -327,8 +327,6 @@ rtc_library("audio_device_impl") { "include/audio_device_data_observer.h", ] if (is_android) { - deps += [ "../../sdk/android:native_api_audio_device_module" ] - if (build_with_mozilla) { include_dirs += [ "/config/external/nspr", diff --git a/modules/audio_device/audio_device_impl.cc b/modules/audio_device/audio_device_impl.cc index 585d63776a..e64e09d563 100644 --- a/modules/audio_device/audio_device_impl.cc +++ b/modules/audio_device/audio_device_impl.cc @@ -33,10 +33,6 @@ #if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD) #include "modules/audio_device/win/audio_device_core_win.h" #endif -#elif defined(WEBRTC_ANDROID) -#include - -#include "sdk/android/native_api/audio_device_module/audio_device_android.h" #elif defined(WEBRTC_LINUX) #if defined(WEBRTC_ENABLE_LINUX_ALSA) #include "modules/audio_device/linux/audio_device_alsa_linux.h" diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn index 5ce345fb62..f778eeb318 100644 --- a/sdk/android/BUILD.gn +++ b/sdk/android/BUILD.gn @@ -868,10 +868,16 @@ if (current_os == "linux" || is_android) { visibility = [ "*" ] sources = [ "src/jni/audio_device/java_audio_device_module.cc" ] + # TODO: bugs.webrtc.org/413413572 - Remove poison when Environment is + # propagated + allow_poison = [ "environment_construction" ] + deps = [ ":base_jni", ":generated_java_audio_jni", ":java_audio_device_module", + "../../api/environment", + "../../api/environment:environment_factory", "//third_party/jni_zero", ] } @@ -1029,6 +1035,10 @@ if (current_os == "linux" || is_android) { rtc_library("native_api_audio_device_module") { visibility = [ "*" ] + # TODO: bugs.webrtc.org/413413572 - Remove poison when Environment is + # propagated + allow_poison = [ "environment_construction" ] + sources = [ "native_api/audio_device_module/audio_device_android.cc", "native_api/audio_device_module/audio_device_android.h", @@ -1042,6 +1052,8 @@ if (current_os == "linux" || is_android) { ":opensles_audio_device_module", "../../api:scoped_refptr", "../../api/audio:audio_device", + "../../api/environment", + "../../api/environment:environment_factory", "../../rtc_base:checks", "../../rtc_base:logging", "../../rtc_base:refcount", @@ -1218,6 +1230,10 @@ if (current_os == "linux" || is_android) { rtc_library("audio_device_module_base") { visibility = [ "*" ] + # TODO: bugs.webrtc.org/413413572 - Remove poison when Environment is + # propagated + allow_poison = [ "environment_construction" ] + sources = [ "src/jni/audio_device/audio_common.h", "src/jni/audio_device/audio_device_module.cc", @@ -1229,10 +1245,12 @@ if (current_os == "linux" || is_android) { ":generated_audio_device_module_base_jni", ":native_api_jni", "../../api:make_ref_counted", + "../../api:scoped_refptr", "../../api:sequence_checker", "../../api/audio:audio_device", + "../../api/environment", + "../../api/environment:environment_factory", "../../api/task_queue", - "../../api/task_queue:default_task_queue_factory", "../../modules/audio_device:audio_device_buffer", "../../rtc_base:checks", "../../rtc_base:logging", @@ -1243,6 +1261,10 @@ if (current_os == "linux" || is_android) { rtc_library("java_audio_device_module") { visibility = [ "*" ] + # TODO: bugs.webrtc.org/413413572 - Remove poison when + # 'audio_device_module_base' is no longer poisonous. + allow_poison = [ "environment_construction" ] + sources = [ "src/jni/audio_device/audio_record_jni.cc", "src/jni/audio_device/audio_record_jni.h", @@ -1300,6 +1322,11 @@ if (current_os == "linux" || is_android) { rtc_library("opensles_audio_device_module") { visibility = [ "*" ] + + # TODO: bugs.webrtc.org/413413572 - Remove poison when + # 'audio_device_module_base' is no longer poisonous. + allow_poison = [ "environment_construction" ] + sources = [ "src/jni/audio_device/opensles_common.cc", "src/jni/audio_device/opensles_common.h", diff --git a/sdk/android/native_api/audio_device_module/audio_device_android.cc b/sdk/android/native_api/audio_device_module/audio_device_android.cc index 2ee8b74d41..eee261682c 100644 --- a/sdk/android/native_api/audio_device_module/audio_device_android.cc +++ b/sdk/android/native_api/audio_device_module/audio_device_android.cc @@ -15,6 +15,8 @@ #include #include +#include "api/environment/environment.h" +#include "api/environment/environment_factory.h" #include "api/scoped_refptr.h" #include "rtc_base/logging.h" #include "rtc_base/ref_count.h" @@ -56,8 +58,9 @@ void GetDefaultAudioParameters(JNIEnv* env, } // namespace #if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO) -webrtc::scoped_refptr CreateAAudioAudioDeviceModule( +scoped_refptr CreateAAudioAudioDeviceModule( JNIEnv* env, + const Environment& webrtc_env, jobject application_context) { RTC_DLOG(LS_INFO) << __FUNCTION__; // Get default audio input/output parameters. @@ -67,15 +70,17 @@ webrtc::scoped_refptr CreateAAudioAudioDeviceModule( &output_parameters); // Create ADM from AAudioRecorder and AAudioPlayer. return CreateAudioDeviceModuleFromInputAndOutput( - AudioDeviceModule::kAndroidAAudioAudio, false /* use_stereo_input */, - false /* use_stereo_output */, + webrtc_env, AudioDeviceModule::kAndroidAAudioAudio, + /*is_stereo_playout_supported=*/false, + /*is_stereo_record_supported=*/false, jni::kLowLatencyModeDelayEstimateInMilliseconds, std::make_unique(input_parameters), std::make_unique(output_parameters)); } -webrtc::scoped_refptr +scoped_refptr CreateJavaInputAndAAudioOutputAudioDeviceModule(JNIEnv* env, + const Environment& webrtc_env, jobject application_context) { RTC_DLOG(LS_INFO) << __FUNCTION__; // Get default audio input/output parameters. @@ -93,15 +98,17 @@ CreateJavaInputAndAAudioOutputAudioDeviceModule(JNIEnv* env, j_audio_manager)); return CreateAudioDeviceModuleFromInputAndOutput( - AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio, - false /* use_stereo_input */, false /* use_stereo_output */, + webrtc_env, AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio, + /*is_stereo_playout_supported=*/false, + /*is_stereo_record_supported=*/false, jni::kLowLatencyModeDelayEstimateInMilliseconds, std::move(audio_input), std::make_unique(output_parameters)); } #endif -webrtc::scoped_refptr CreateJavaAudioDeviceModule( +scoped_refptr CreateJavaAudioDeviceModule( JNIEnv* env, + const Environment& webrtc_env, jobject application_context) { RTC_DLOG(LS_INFO) << __FUNCTION__; // Get default audio input/output parameters. @@ -122,14 +129,16 @@ webrtc::scoped_refptr CreateJavaAudioDeviceModule( jni::AudioTrackJni::CreateJavaWebRtcAudioTrack(env, j_context, j_audio_manager)); return CreateAudioDeviceModuleFromInputAndOutput( - AudioDeviceModule::kAndroidJavaAudio, false /* use_stereo_input */, - false /* use_stereo_output */, + webrtc_env, AudioDeviceModule::kAndroidJavaAudio, + /*is_stereo_playout_supported=*/false, + /*is_stereo_record_supported=*/false, jni::kHighLatencyModeDelayEstimateInMilliseconds, std::move(audio_input), std::move(audio_output)); } -webrtc::scoped_refptr CreateOpenSLESAudioDeviceModule( +scoped_refptr CreateOpenSLESAudioDeviceModule( JNIEnv* env, + const Environment& webrtc_env, jobject application_context) { RTC_DLOG(LS_INFO) << __FUNCTION__; // Get default audio input/output parameters. @@ -138,21 +147,23 @@ webrtc::scoped_refptr CreateOpenSLESAudioDeviceModule( GetDefaultAudioParameters(env, application_context, &input_parameters, &output_parameters); // Create ADM from OpenSLESRecorder and OpenSLESPlayer. - webrtc::scoped_refptr engine_manager( + scoped_refptr engine_manager( new jni::OpenSLEngineManager()); auto audio_input = std::make_unique(input_parameters, engine_manager); auto audio_output = std::make_unique( output_parameters, std::move(engine_manager)); return CreateAudioDeviceModuleFromInputAndOutput( - AudioDeviceModule::kAndroidOpenSLESAudio, false /* use_stereo_input */, - false /* use_stereo_output */, + webrtc_env, AudioDeviceModule::kAndroidOpenSLESAudio, + /*is_stereo_playout_supported=*/false, + /*is_stereo_record_supported=*/false, jni::kLowLatencyModeDelayEstimateInMilliseconds, std::move(audio_input), std::move(audio_output)); } -webrtc::scoped_refptr +scoped_refptr CreateJavaInputAndOpenSLESOutputAudioDeviceModule(JNIEnv* env, + const Environment& webrtc_env, jobject application_context) { RTC_DLOG(LS_INFO) << __FUNCTION__; // Get default audio input/output parameters. @@ -169,21 +180,23 @@ CreateJavaInputAndOpenSLESOutputAudioDeviceModule(JNIEnv* env, jni::AudioRecordJni::CreateJavaWebRtcAudioRecord(env, j_context, j_audio_manager)); - webrtc::scoped_refptr engine_manager( + scoped_refptr engine_manager( new jni::OpenSLEngineManager()); auto audio_output = std::make_unique( output_parameters, std::move(engine_manager)); return CreateAudioDeviceModuleFromInputAndOutput( - AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio, - false /* use_stereo_input */, false /* use_stereo_output */, + webrtc_env, AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio, + /*is_stereo_playout_supported=*/false, + /*is_stereo_record_supported=*/false, jni::kLowLatencyModeDelayEstimateInMilliseconds, std::move(audio_input), std::move(audio_output)); } -webrtc::scoped_refptr CreateAndroidAudioDeviceModule( +scoped_refptr CreateAndroidAudioDeviceModule( + const Environment& webrtc_env, AudioDeviceModule::AudioLayer audio_layer) { auto env = AttachCurrentThreadIfNeeded(); - auto j_context = webrtc::GetAppContext(env); + auto j_context = GetAppContext(env); // Select best possible combination of audio layers. if (audio_layer == AudioDeviceModule::kPlatformDefaultAudio) { #if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO) @@ -209,23 +222,23 @@ webrtc::scoped_refptr CreateAndroidAudioDeviceModule( switch (audio_layer) { case AudioDeviceModule::kAndroidJavaAudio: // Java audio for both input and output audio. - return CreateJavaAudioDeviceModule(env, j_context.obj()); + return CreateJavaAudioDeviceModule(env, webrtc_env, j_context.obj()); case AudioDeviceModule::kAndroidOpenSLESAudio: // OpenSL ES based audio for both input and output audio. - return CreateOpenSLESAudioDeviceModule(env, j_context.obj()); + return CreateOpenSLESAudioDeviceModule(env, webrtc_env, j_context.obj()); case AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio: // Java audio for input and OpenSL ES for output audio (i.e. mixed APIs). // This combination provides low-latency output audio and at the same // time support for HW AEC using the AudioRecord Java API. - return CreateJavaInputAndOpenSLESOutputAudioDeviceModule(env, + return CreateJavaInputAndOpenSLESOutputAudioDeviceModule(env, webrtc_env, j_context.obj()); #if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO) case AudioDeviceModule::kAndroidAAudioAudio: // AAudio based audio for both input and output. - return CreateAAudioAudioDeviceModule(env, j_context.obj()); + return CreateAAudioAudioDeviceModule(env, webrtc_env, j_context.obj()); case AudioDeviceModule::kAndroidJavaInputAndAAudioOutputAudio: // Java audio for input and AAudio for output audio (i.e. mixed APIs). - return CreateJavaInputAndAAudioOutputAudioDeviceModule(env, + return CreateJavaInputAndAAudioOutputAudioDeviceModule(env, webrtc_env, j_context.obj()); #endif default: @@ -233,4 +246,39 @@ webrtc::scoped_refptr CreateAndroidAudioDeviceModule( } } +#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO) +scoped_refptr CreateAAudioAudioDeviceModule( + JNIEnv* env, + jobject application_context) { + return CreateAAudioAudioDeviceModule(env, CreateEnvironment(), + application_context); +} +#endif + +scoped_refptr CreateJavaAudioDeviceModule( + JNIEnv* env, + jobject application_context) { + return CreateJavaAudioDeviceModule(env, CreateEnvironment(), + application_context); +} + +scoped_refptr CreateOpenSLESAudioDeviceModule( + JNIEnv* env, + jobject application_context) { + return CreateOpenSLESAudioDeviceModule(env, CreateEnvironment(), + application_context); +} + +scoped_refptr +CreateJavaInputAndOpenSLESOutputAudioDeviceModule(JNIEnv* env, + jobject application_context) { + return CreateJavaInputAndOpenSLESOutputAudioDeviceModule( + env, CreateEnvironment(), application_context); +} + +scoped_refptr CreateAndroidAudioDeviceModule( + AudioDeviceModule::AudioLayer audio_layer) { + return CreateAndroidAudioDeviceModule(CreateEnvironment(), audio_layer); +} + } // namespace webrtc diff --git a/sdk/android/native_api/audio_device_module/audio_device_android.h b/sdk/android/native_api/audio_device_module/audio_device_android.h index 4bfd7e850c..88ecbb0d7e 100644 --- a/sdk/android/native_api/audio_device_module/audio_device_android.h +++ b/sdk/android/native_api/audio_device_module/audio_device_android.h @@ -14,32 +14,49 @@ #include #include "api/audio/audio_device.h" +#include "api/environment/environment.h" +#include "api/scoped_refptr.h" namespace webrtc { +scoped_refptr CreateJavaAudioDeviceModule( + JNIEnv* env, + const Environment& webrtc_env, + jobject application_context); + +scoped_refptr CreateAndroidAudioDeviceModule( + const Environment& webrtc_env, + AudioDeviceModule::AudioLayer audio_layer); + #if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO) -webrtc::scoped_refptr CreateAAudioAudioDeviceModule( +[[deprecated("bugs.webrtc.org/413413572")]] +scoped_refptr CreateAAudioAudioDeviceModule( JNIEnv* env, jobject application_context); #endif -webrtc::scoped_refptr CreateJavaAudioDeviceModule( +[[deprecated("bugs.webrtc.org/413413572")]] +scoped_refptr CreateJavaAudioDeviceModule( JNIEnv* env, jobject application_context); -webrtc::scoped_refptr CreateOpenSLESAudioDeviceModule( +[[deprecated("bugs.webrtc.org/413413572")]] +scoped_refptr CreateOpenSLESAudioDeviceModule( JNIEnv* env, jobject application_context); -webrtc::scoped_refptr +[[deprecated("bugs.webrtc.org/413413572")]] +scoped_refptr CreateJavaInputAndOpenSLESOutputAudioDeviceModule(JNIEnv* env, jobject application_context); -webrtc::scoped_refptr +[[deprecated("bugs.webrtc.org/413413572")]] +scoped_refptr CreateJavaInputAndAAudioOutputAudioDeviceModule(JNIEnv* env, jobject application_context); -webrtc::scoped_refptr CreateAndroidAudioDeviceModule( +[[deprecated("bugs.webrtc.org/413413572")]] +scoped_refptr CreateAndroidAudioDeviceModule( AudioDeviceModule::AudioLayer audio_layer); } // namespace webrtc diff --git a/sdk/android/native_unittests/audio_device/audio_device_unittest.cc b/sdk/android/native_unittests/audio_device/audio_device_unittest.cc index 878f8ae5d3..e8e5ad7c6b 100644 --- a/sdk/android/native_unittests/audio_device/audio_device_unittest.cc +++ b/sdk/android/native_unittests/audio_device/audio_device_unittest.cc @@ -14,6 +14,8 @@ #include #include +#include "api/environment/environment.h" +#include "api/environment/environment_factory.h" #include "api/scoped_refptr.h" #include "modules/audio_device/include/mock_audio_transport.h" #include "rtc_base/arraysize.h" @@ -467,7 +469,8 @@ class AudioDeviceTest : public ::testing::Test { // Creates an audio device using a default audio layer. jni_ = AttachCurrentThreadIfNeeded(); context_ = GetAppContext(jni_); - audio_device_ = CreateJavaAudioDeviceModule(jni_, context_.obj()); + audio_device_ = + CreateJavaAudioDeviceModule(jni_, webrtc_env_, context_.obj()); EXPECT_NE(audio_device_.get(), nullptr); EXPECT_EQ(0, audio_device_->Init()); audio_manager_ = GetAudioManager(jni_, context_); @@ -491,7 +494,7 @@ class AudioDeviceTest : public ::testing::Test { } void SetActiveAudioLayer(AudioDeviceModule::AudioLayer audio_layer) { - audio_device_ = CreateAndroidAudioDeviceModule(audio_layer); + audio_device_ = CreateAndroidAudioDeviceModule(webrtc_env_, audio_layer); EXPECT_NE(audio_device_.get(), nullptr); EXPECT_EQ(0, audio_device_->Init()); UpdateParameters(); @@ -542,7 +545,7 @@ class AudioDeviceTest : public ::testing::Test { int TestDelayOnAudioLayer( const AudioDeviceModule::AudioLayer& layer_to_test) { webrtc::scoped_refptr audio_device; - audio_device = CreateAndroidAudioDeviceModule(layer_to_test); + audio_device = CreateAndroidAudioDeviceModule(webrtc_env_, layer_to_test); EXPECT_NE(audio_device.get(), nullptr); uint16_t playout_delay; EXPECT_EQ(0, audio_device->PlayoutDelay(&playout_delay)); @@ -552,7 +555,7 @@ class AudioDeviceTest : public ::testing::Test { AudioDeviceModule::AudioLayer TestActiveAudioLayer( const AudioDeviceModule::AudioLayer& layer_to_test) { webrtc::scoped_refptr audio_device; - audio_device = CreateAndroidAudioDeviceModule(layer_to_test); + audio_device = CreateAndroidAudioDeviceModule(webrtc_env_, layer_to_test); EXPECT_NE(audio_device.get(), nullptr); AudioDeviceModule::AudioLayer active; EXPECT_EQ(0, audio_device->ActiveAudioLayer(&active)); @@ -667,6 +670,7 @@ class AudioDeviceTest : public ::testing::Test { } JNIEnv* jni_; + const Environment webrtc_env_ = CreateEnvironment(); ScopedJavaLocalRef context_; webrtc::Event test_is_done_; webrtc::scoped_refptr audio_device_; @@ -1157,19 +1161,20 @@ TEST_F(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) { // TODO(https://crbug.com/webrtc/15537): test randomly fails. TEST(JavaAudioDeviceTest, DISABLED_TestRunningTwoAdmsSimultaneously) { JNIEnv* jni = AttachCurrentThreadIfNeeded(); + const Environment webrtc_env = CreateEnvironment(); ScopedJavaLocalRef context = GetAppContext(jni); // Create and start the first ADM. - webrtc::scoped_refptr adm_1 = - CreateJavaAudioDeviceModule(jni, context.obj()); + scoped_refptr adm_1 = + CreateJavaAudioDeviceModule(jni, webrtc_env, context.obj()); EXPECT_EQ(0, adm_1->Init()); EXPECT_EQ(0, adm_1->InitRecording()); EXPECT_EQ(0, adm_1->StartRecording()); // Create and start a second ADM. Expect this to fail due to the microphone // already being in use. - webrtc::scoped_refptr adm_2 = - CreateJavaAudioDeviceModule(jni, context.obj()); + scoped_refptr adm_2 = + CreateJavaAudioDeviceModule(jni, webrtc_env, context.obj()); int32_t err = adm_2->Init(); err |= adm_2->InitRecording(); err |= adm_2->StartRecording(); diff --git a/sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc b/sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc index 0e243c527e..f5d8470ddb 100644 --- a/sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc +++ b/sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc @@ -16,6 +16,7 @@ #include #include "api/enable_media_with_defaults.h" +#include "api/environment/environment_factory.h" #include "api/peer_connection_interface.h" #include "api/rtc_event_log/rtc_event_log_factory.h" #include "api/scoped_refptr.h" @@ -53,8 +54,10 @@ webrtc::scoped_refptr CreateTestPCF( pcf_deps.worker_thread = worker_thread; pcf_deps.signaling_thread = signaling_thread; pcf_deps.event_log_factory = std::make_unique(); + pcf_deps.env = CreateEnvironment(); - pcf_deps.adm = CreateJavaAudioDeviceModule(jni, GetAppContext(jni).obj()); + pcf_deps.adm = + CreateJavaAudioDeviceModule(jni, *pcf_deps.env, GetAppContext(jni).obj()); pcf_deps.video_encoder_factory = std::make_unique(); pcf_deps.video_decoder_factory = diff --git a/sdk/android/src/jni/audio_device/audio_device_module.cc b/sdk/android/src/jni/audio_device/audio_device_module.cc index 4d64e5ff35..732e8d4607 100644 --- a/sdk/android/src/jni/audio_device/audio_device_module.cc +++ b/sdk/android/src/jni/audio_device/audio_device_module.cc @@ -13,10 +13,10 @@ #include #include +#include "api/environment/environment.h" +#include "api/environment/environment_factory.h" #include "api/make_ref_counted.h" #include "api/sequence_checker.h" -#include "api/task_queue/default_task_queue_factory.h" -#include "api/task_queue/task_queue_factory.h" #include "modules/audio_device/audio_device_buffer.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" @@ -54,17 +54,18 @@ class AndroidAudioDeviceModule : public AudioDeviceModule { NUM_STATUSES = 4 }; - AndroidAudioDeviceModule(AudioDeviceModule::AudioLayer audio_layer, + AndroidAudioDeviceModule(const Environment& env, + AudioDeviceModule::AudioLayer audio_layer, bool is_stereo_playout_supported, bool is_stereo_record_supported, uint16_t playout_delay_ms, std::unique_ptr audio_input, std::unique_ptr audio_output) - : audio_layer_(audio_layer), + : env_(env), + audio_layer_(audio_layer), is_stereo_playout_supported_(is_stereo_playout_supported), is_stereo_record_supported_(is_stereo_record_supported), playout_delay_ms_(playout_delay_ms), - task_queue_factory_(CreateDefaultTaskQueueFactory()), input_(std::move(audio_input)), output_(std::move(audio_output)), initialized_(false) { @@ -92,7 +93,7 @@ class AndroidAudioDeviceModule : public AudioDeviceModule { RTC_DLOG(LS_INFO) << __FUNCTION__; RTC_DCHECK(thread_checker_.IsCurrent()); audio_device_buffer_ = - std::make_unique(task_queue_factory_.get()); + std::make_unique(&env_.task_queue_factory()); AttachAudioBuffer(); if (initialized_) { return 0; @@ -589,11 +590,11 @@ class AndroidAudioDeviceModule : public AudioDeviceModule { private: SequenceChecker thread_checker_; + const Environment env_; const AudioDeviceModule::AudioLayer audio_layer_; const bool is_stereo_playout_supported_; const bool is_stereo_record_supported_; const uint16_t playout_delay_ms_; - const std::unique_ptr task_queue_factory_; const std::unique_ptr input_; const std::unique_ptr output_; std::unique_ptr audio_device_buffer_; @@ -656,7 +657,22 @@ scoped_refptr CreateAudioDeviceModuleFromInputAndOutput( std::unique_ptr audio_output) { RTC_DLOG(LS_INFO) << __FUNCTION__; return make_ref_counted( - audio_layer, is_stereo_playout_supported, is_stereo_record_supported, + CreateEnvironment(), audio_layer, is_stereo_playout_supported, + is_stereo_record_supported, playout_delay_ms, std::move(audio_input), + std::move(audio_output)); +} + +scoped_refptr CreateAudioDeviceModuleFromInputAndOutput( + const Environment& env, + AudioDeviceModule::AudioLayer audio_layer, + bool is_stereo_playout_supported, + bool is_stereo_record_supported, + uint16_t playout_delay_ms, + std::unique_ptr audio_input, + std::unique_ptr audio_output) { + RTC_DLOG(LS_INFO) << __FUNCTION__; + return make_ref_counted( + env, audio_layer, is_stereo_playout_supported, is_stereo_record_supported, playout_delay_ms, std::move(audio_input), std::move(audio_output)); } diff --git a/sdk/android/src/jni/audio_device/audio_device_module.h b/sdk/android/src/jni/audio_device/audio_device_module.h index cbf71ed473..f60432543e 100644 --- a/sdk/android/src/jni/audio_device/audio_device_module.h +++ b/sdk/android/src/jni/audio_device/audio_device_module.h @@ -15,6 +15,8 @@ #include #include "api/audio/audio_device.h" +#include "api/environment/environment.h" +#include "api/scoped_refptr.h" #include "sdk/android/native_api/jni/scoped_java_ref.h" namespace webrtc { @@ -95,6 +97,7 @@ bool IsLowLatencyOutputSupported(JNIEnv* env, const JavaRef& j_context); // Glue together an audio input and audio output to get an AudioDeviceModule. +[[deprecated("bugs.webrtc.org/413413572")]] scoped_refptr CreateAudioDeviceModuleFromInputAndOutput( AudioDeviceModule::AudioLayer audio_layer, bool is_stereo_playout_supported, @@ -103,6 +106,15 @@ scoped_refptr CreateAudioDeviceModuleFromInputAndOutput( std::unique_ptr audio_input, std::unique_ptr audio_output); +scoped_refptr CreateAudioDeviceModuleFromInputAndOutput( + const Environment& env, + AudioDeviceModule::AudioLayer audio_layer, + bool is_stereo_playout_supported, + bool is_stereo_record_supported, + uint16_t playout_delay_ms, + std::unique_ptr audio_input, + std::unique_ptr audio_output); + } // namespace jni } // namespace webrtc diff --git a/sdk/android/src/jni/audio_device/java_audio_device_module.cc b/sdk/android/src/jni/audio_device/java_audio_device_module.cc index ea0c659558..7f58163a7f 100644 --- a/sdk/android/src/jni/audio_device/java_audio_device_module.cc +++ b/sdk/android/src/jni/audio_device/java_audio_device_module.cc @@ -10,6 +10,8 @@ #include +#include "api/environment/environment.h" +#include "api/environment/environment_factory.h" #include "sdk/android/generated_java_audio_jni/JavaAudioDeviceModule_jni.h" #include "sdk/android/src/jni/audio_device/audio_record_jni.h" #include "sdk/android/src/jni/audio_device/audio_track_jni.h" @@ -29,6 +31,7 @@ static jlong JNI_JavaAudioDeviceModule_CreateAudioDeviceModule( int output_sample_rate, jboolean j_use_stereo_input, jboolean j_use_stereo_output) { + const Environment webrtc_env = CreateEnvironment(); AudioParameters input_parameters; AudioParameters output_parameters; GetAudioParameters(env, j_context, j_audio_manager, input_sample_rate, @@ -40,12 +43,12 @@ static jlong JNI_JavaAudioDeviceModule_CreateAudioDeviceModule( j_webrtc_audio_record); auto audio_output = std::make_unique(env, output_parameters, j_webrtc_audio_track); - return jlongFromPointer(CreateAudioDeviceModuleFromInputAndOutput( - AudioDeviceModule::kAndroidJavaAudio, - j_use_stereo_input, j_use_stereo_output, - kHighLatencyModeDelayEstimateInMilliseconds, - std::move(audio_input), std::move(audio_output)) - .release()); + return NativeToJavaPointer( + CreateAudioDeviceModuleFromInputAndOutput( + webrtc_env, AudioDeviceModule::kAndroidJavaAudio, j_use_stereo_input, + j_use_stereo_output, kHighLatencyModeDelayEstimateInMilliseconds, + std::move(audio_input), std::move(audio_output)) + .release()); } } // namespace jni From 2b5eccc4f2eed4b5567817f1fa1fef38aca7f199 Mon Sep 17 00:00:00 2001 From: Ilya Nikolaevskiy Date: Thu, 15 May 2025 01:53:34 -0700 Subject: [PATCH 160/292] Revert "IWYU modules/video_capture/" This reverts commit 55b5f0e429a572aabb28ab3b078f19a4365c1028. Reason for revert: breaks downstream projects. Bug: webrtc:42226242 Original change's description: > IWYU modules/video_capture/ > > without platform-specific subdirectories first. Done > using > ls modules/video_capture/*.{h,cc} | xargs tools_webrtc/iwyu/apply-include-cleaner | xargs tools_webrtc/iwyu/apply-include-cleaner > followed by > tools_webrtc/gn_check_autofix.py -C out/Default/ > and git cl format > > Manual changes: > - video_capture_impl.cc: use C++ headers > > Bug: webrtc:42226242 > Change-Id: I2477021f0cfd9045cae4c5a789c4983a194ef7ef > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391262 > Reviewed-by: Ilya Nikolaevskiy > Reviewed-by: Danil Chapovalov > Commit-Queue: Philipp Hancke > Cr-Commit-Position: refs/heads/main@{#44628} Bug: webrtc:42226242 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: If01a94c3699bc7d4bdadd23fef0416685544b191 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391720 Reviewed-by: Jeremy Leconte Owners-Override: Ilya Nikolaevskiy Bot-Commit: rubber-stamper@appspot.gserviceaccount.com Commit-Queue: Ilya Nikolaevskiy Cr-Commit-Position: refs/heads/main@{#44632} --- modules/video_capture/BUILD.gn | 2 -- .../video_capture/raw_video_sink_interface.h | 4 ---- modules/video_capture/video_capture.h | 4 ---- modules/video_capture/video_capture_defines.h | 3 +-- .../video_capture/video_capture_factory.cc | 2 -- modules/video_capture/video_capture_factory.h | 1 + .../video_capture_factory_null.cc | 2 -- modules/video_capture/video_capture_impl.cc | 19 ++++--------------- modules/video_capture/video_capture_impl.h | 2 -- .../video_capture/video_capture_options.cc | 3 --- 10 files changed, 6 insertions(+), 36 deletions(-) diff --git a/modules/video_capture/BUILD.gn b/modules/video_capture/BUILD.gn index c3e0e5af71..e5e74c9fc3 100644 --- a/modules/video_capture/BUILD.gn +++ b/modules/video_capture/BUILD.gn @@ -28,7 +28,6 @@ rtc_library("video_capture_module") { ] deps = [ - "../../api:ref_count", "../../api:scoped_refptr", "../../api:sequence_checker", "../../api/video:video_frame", @@ -56,7 +55,6 @@ if (!build_with_chromium || is_linux || is_chromeos) { visibility = [ "*" ] deps = [ ":video_capture_module", - "../../api:make_ref_counted", "../../api:scoped_refptr", "../../api:sequence_checker", "../../media:video_common", diff --git a/modules/video_capture/raw_video_sink_interface.h b/modules/video_capture/raw_video_sink_interface.h index a22ca9811d..094e9e20bd 100644 --- a/modules/video_capture/raw_video_sink_interface.h +++ b/modules/video_capture/raw_video_sink_interface.h @@ -14,10 +14,6 @@ #ifndef MODULES_VIDEO_CAPTURE_RAW_VIDEO_SINK_INTERFACE_H_ #define MODULES_VIDEO_CAPTURE_RAW_VIDEO_SINK_INTERFACE_H_ -#include -#include - -#include "api/video/video_rotation.h" #include "modules/video_capture/video_capture_defines.h" namespace webrtc { diff --git a/modules/video_capture/video_capture.h b/modules/video_capture/video_capture.h index 7707608dbc..fe5c09bd87 100644 --- a/modules/video_capture/video_capture.h +++ b/modules/video_capture/video_capture.h @@ -11,10 +11,6 @@ #ifndef MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_H_ #define MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_H_ -#include - -#include "api/ref_count.h" -#include "api/video/video_frame.h" #include "api/video/video_rotation.h" #include "api/video/video_sink_interface.h" #include "modules/video_capture/raw_video_sink_interface.h" diff --git a/modules/video_capture/video_capture_defines.h b/modules/video_capture/video_capture_defines.h index f814e4e7f5..63534600a9 100644 --- a/modules/video_capture/video_capture_defines.h +++ b/modules/video_capture/video_capture_defines.h @@ -11,8 +11,7 @@ #ifndef MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_DEFINES_H_ #define MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_DEFINES_H_ -#include - +#include "api/video/video_frame.h" #include "common_video/libyuv/include/webrtc_libyuv.h" namespace webrtc { diff --git a/modules/video_capture/video_capture_factory.cc b/modules/video_capture/video_capture_factory.cc index d7fa0fa26e..8354067f93 100644 --- a/modules/video_capture/video_capture_factory.cc +++ b/modules/video_capture/video_capture_factory.cc @@ -10,8 +10,6 @@ #include "modules/video_capture/video_capture_factory.h" -#include "api/scoped_refptr.h" -#include "modules/video_capture/video_capture.h" #include "modules/video_capture/video_capture_impl.h" namespace webrtc { diff --git a/modules/video_capture/video_capture_factory.h b/modules/video_capture/video_capture_factory.h index e669470e35..c24eaf559f 100644 --- a/modules/video_capture/video_capture_factory.h +++ b/modules/video_capture/video_capture_factory.h @@ -16,6 +16,7 @@ #include "api/scoped_refptr.h" #include "modules/video_capture/video_capture.h" +#include "modules/video_capture/video_capture_defines.h" #include "rtc_base/system/rtc_export.h" namespace webrtc { diff --git a/modules/video_capture/video_capture_factory_null.cc b/modules/video_capture/video_capture_factory_null.cc index 4dbc7b9f5c..77b7f4b75e 100644 --- a/modules/video_capture/video_capture_factory_null.cc +++ b/modules/video_capture/video_capture_factory_null.cc @@ -8,8 +8,6 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "api/scoped_refptr.h" -#include "modules/video_capture/video_capture.h" #include "modules/video_capture/video_capture_impl.h" namespace webrtc { diff --git a/modules/video_capture/video_capture_impl.cc b/modules/video_capture/video_capture_impl.cc index f12631c274..06f9bd3165 100644 --- a/modules/video_capture/video_capture_impl.cc +++ b/modules/video_capture/video_capture_impl.cc @@ -10,28 +10,17 @@ #include "modules/video_capture/video_capture_impl.h" -#include -#include -#include +#include +#include -#include "api/scoped_refptr.h" -#include "api/sequence_checker.h" #include "api/video/i420_buffer.h" -#include "api/video/video_frame.h" -#include "api/video/video_rotation.h" -#include "api/video/video_sink_interface.h" +#include "api/video/video_frame_buffer.h" #include "common_video/libyuv/include/webrtc_libyuv.h" -#include "libyuv/convert.h" -#include "libyuv/rotate.h" -#include "modules/video_capture/raw_video_sink_interface.h" #include "modules/video_capture/video_capture_config.h" -#include "modules/video_capture/video_capture_defines.h" -#include "rtc_base/checks.h" #include "rtc_base/logging.h" -#include "rtc_base/race_checker.h" -#include "rtc_base/synchronization/mutex.h" #include "rtc_base/time_utils.h" #include "rtc_base/trace_event.h" +#include "third_party/libyuv/include/libyuv.h" namespace webrtc { namespace videocapturemodule { diff --git a/modules/video_capture/video_capture_impl.h b/modules/video_capture/video_capture_impl.h index d3ac5fe5ef..3ec832f871 100644 --- a/modules/video_capture/video_capture_impl.h +++ b/modules/video_capture/video_capture_impl.h @@ -23,14 +23,12 @@ #include "api/video/video_frame.h" #include "api/video/video_rotation.h" #include "api/video/video_sink_interface.h" -#include "modules/video_capture/raw_video_sink_interface.h" #include "modules/video_capture/video_capture.h" #include "modules/video_capture/video_capture_config.h" #include "modules/video_capture/video_capture_defines.h" #include "rtc_base/race_checker.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/system/rtc_export.h" -#include "rtc_base/thread_annotations.h" namespace webrtc { diff --git a/modules/video_capture/video_capture_options.cc b/modules/video_capture/video_capture_options.cc index dd102df9db..64cd6216e7 100644 --- a/modules/video_capture/video_capture_options.cc +++ b/modules/video_capture/video_capture_options.cc @@ -10,9 +10,6 @@ #include "modules/video_capture/video_capture_options.h" -#include "api/make_ref_counted.h" -#include "api/scoped_refptr.h" - #if defined(WEBRTC_USE_PIPEWIRE) #include "modules/video_capture/linux/pipewire_session.h" #endif From cc6e26341db3c6f52b2641c411cec3666887bb30 Mon Sep 17 00:00:00 2001 From: Ilya Nikolaevskiy Date: Thu, 15 May 2025 11:40:22 +0200 Subject: [PATCH 161/292] Reland "IWYU modules/video_capture/" This reverts commit 2b5eccc4f2eed4b5567817f1fa1fef38aca7f199. Reason for revert: Fixing includes Bug: webrtc:42226242 Original change's description: > Revert "IWYU modules/video_capture/" > > This reverts commit 55b5f0e429a572aabb28ab3b078f19a4365c1028. > > Reason for revert: breaks downstream projects. > > Bug: webrtc:42226242 > Original change's description: > > IWYU modules/video_capture/ > > > > without platform-specific subdirectories first. Done > > using > > ls modules/video_capture/*.{h,cc} | xargs tools_webrtc/iwyu/apply-include-cleaner | xargs tools_webrtc/iwyu/apply-include-cleaner > > followed by > > tools_webrtc/gn_check_autofix.py -C out/Default/ > > and git cl format > > > > Manual changes: > > - video_capture_impl.cc: use C++ headers > > > > Bug: webrtc:42226242 > > Change-Id: I2477021f0cfd9045cae4c5a789c4983a194ef7ef > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391262 > > Reviewed-by: Ilya Nikolaevskiy > > Reviewed-by: Danil Chapovalov > > Commit-Queue: Philipp Hancke > > Cr-Commit-Position: refs/heads/main@{#44628} > > Bug: webrtc:42226242 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Change-Id: If01a94c3699bc7d4bdadd23fef0416685544b191 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391720 > Reviewed-by: Jeremy Leconte > Owners-Override: Ilya Nikolaevskiy > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com > Commit-Queue: Ilya Nikolaevskiy > Cr-Commit-Position: refs/heads/main@{#44632} Bug: webrtc:42226242 Change-Id: I6acbb4abecbc293ff9acca31e864ba72b952b5b8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391722 Commit-Queue: Ilya Nikolaevskiy Reviewed-by: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#44633} --- modules/video_capture/BUILD.gn | 2 ++ .../video_capture/raw_video_sink_interface.h | 4 ++++ modules/video_capture/video_capture.h | 4 ++++ modules/video_capture/video_capture_defines.h | 3 ++- .../video_capture/video_capture_factory.cc | 2 ++ modules/video_capture/video_capture_factory.h | 1 - .../video_capture_factory_null.cc | 2 ++ modules/video_capture/video_capture_impl.cc | 19 +++++++++++++++---- modules/video_capture/video_capture_impl.h | 2 ++ .../video_capture/video_capture_options.cc | 3 +++ 10 files changed, 36 insertions(+), 6 deletions(-) diff --git a/modules/video_capture/BUILD.gn b/modules/video_capture/BUILD.gn index e5e74c9fc3..c3e0e5af71 100644 --- a/modules/video_capture/BUILD.gn +++ b/modules/video_capture/BUILD.gn @@ -28,6 +28,7 @@ rtc_library("video_capture_module") { ] deps = [ + "../../api:ref_count", "../../api:scoped_refptr", "../../api:sequence_checker", "../../api/video:video_frame", @@ -55,6 +56,7 @@ if (!build_with_chromium || is_linux || is_chromeos) { visibility = [ "*" ] deps = [ ":video_capture_module", + "../../api:make_ref_counted", "../../api:scoped_refptr", "../../api:sequence_checker", "../../media:video_common", diff --git a/modules/video_capture/raw_video_sink_interface.h b/modules/video_capture/raw_video_sink_interface.h index 094e9e20bd..a22ca9811d 100644 --- a/modules/video_capture/raw_video_sink_interface.h +++ b/modules/video_capture/raw_video_sink_interface.h @@ -14,6 +14,10 @@ #ifndef MODULES_VIDEO_CAPTURE_RAW_VIDEO_SINK_INTERFACE_H_ #define MODULES_VIDEO_CAPTURE_RAW_VIDEO_SINK_INTERFACE_H_ +#include +#include + +#include "api/video/video_rotation.h" #include "modules/video_capture/video_capture_defines.h" namespace webrtc { diff --git a/modules/video_capture/video_capture.h b/modules/video_capture/video_capture.h index fe5c09bd87..7707608dbc 100644 --- a/modules/video_capture/video_capture.h +++ b/modules/video_capture/video_capture.h @@ -11,6 +11,10 @@ #ifndef MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_H_ #define MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_H_ +#include + +#include "api/ref_count.h" +#include "api/video/video_frame.h" #include "api/video/video_rotation.h" #include "api/video/video_sink_interface.h" #include "modules/video_capture/raw_video_sink_interface.h" diff --git a/modules/video_capture/video_capture_defines.h b/modules/video_capture/video_capture_defines.h index 63534600a9..f814e4e7f5 100644 --- a/modules/video_capture/video_capture_defines.h +++ b/modules/video_capture/video_capture_defines.h @@ -11,7 +11,8 @@ #ifndef MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_DEFINES_H_ #define MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_DEFINES_H_ -#include "api/video/video_frame.h" +#include + #include "common_video/libyuv/include/webrtc_libyuv.h" namespace webrtc { diff --git a/modules/video_capture/video_capture_factory.cc b/modules/video_capture/video_capture_factory.cc index 8354067f93..d7fa0fa26e 100644 --- a/modules/video_capture/video_capture_factory.cc +++ b/modules/video_capture/video_capture_factory.cc @@ -10,6 +10,8 @@ #include "modules/video_capture/video_capture_factory.h" +#include "api/scoped_refptr.h" +#include "modules/video_capture/video_capture.h" #include "modules/video_capture/video_capture_impl.h" namespace webrtc { diff --git a/modules/video_capture/video_capture_factory.h b/modules/video_capture/video_capture_factory.h index c24eaf559f..e669470e35 100644 --- a/modules/video_capture/video_capture_factory.h +++ b/modules/video_capture/video_capture_factory.h @@ -16,7 +16,6 @@ #include "api/scoped_refptr.h" #include "modules/video_capture/video_capture.h" -#include "modules/video_capture/video_capture_defines.h" #include "rtc_base/system/rtc_export.h" namespace webrtc { diff --git a/modules/video_capture/video_capture_factory_null.cc b/modules/video_capture/video_capture_factory_null.cc index 77b7f4b75e..4dbc7b9f5c 100644 --- a/modules/video_capture/video_capture_factory_null.cc +++ b/modules/video_capture/video_capture_factory_null.cc @@ -8,6 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include "api/scoped_refptr.h" +#include "modules/video_capture/video_capture.h" #include "modules/video_capture/video_capture_impl.h" namespace webrtc { diff --git a/modules/video_capture/video_capture_impl.cc b/modules/video_capture/video_capture_impl.cc index 06f9bd3165..eace423634 100644 --- a/modules/video_capture/video_capture_impl.cc +++ b/modules/video_capture/video_capture_impl.cc @@ -10,17 +10,28 @@ #include "modules/video_capture/video_capture_impl.h" -#include -#include +#include +#include +#include +#include "api/scoped_refptr.h" +#include "api/sequence_checker.h" #include "api/video/i420_buffer.h" -#include "api/video/video_frame_buffer.h" +#include "api/video/video_frame.h" +#include "api/video/video_rotation.h" +#include "api/video/video_sink_interface.h" #include "common_video/libyuv/include/webrtc_libyuv.h" +#include "modules/video_capture/raw_video_sink_interface.h" #include "modules/video_capture/video_capture_config.h" +#include "modules/video_capture/video_capture_defines.h" +#include "rtc_base/checks.h" #include "rtc_base/logging.h" +#include "rtc_base/race_checker.h" +#include "rtc_base/synchronization/mutex.h" #include "rtc_base/time_utils.h" #include "rtc_base/trace_event.h" -#include "third_party/libyuv/include/libyuv.h" +#include "third_party/libyuv/include/libyuv/convert.h" +#include "third_party/libyuv/include/libyuv/rotate.h" namespace webrtc { namespace videocapturemodule { diff --git a/modules/video_capture/video_capture_impl.h b/modules/video_capture/video_capture_impl.h index 3ec832f871..d3ac5fe5ef 100644 --- a/modules/video_capture/video_capture_impl.h +++ b/modules/video_capture/video_capture_impl.h @@ -23,12 +23,14 @@ #include "api/video/video_frame.h" #include "api/video/video_rotation.h" #include "api/video/video_sink_interface.h" +#include "modules/video_capture/raw_video_sink_interface.h" #include "modules/video_capture/video_capture.h" #include "modules/video_capture/video_capture_config.h" #include "modules/video_capture/video_capture_defines.h" #include "rtc_base/race_checker.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/system/rtc_export.h" +#include "rtc_base/thread_annotations.h" namespace webrtc { diff --git a/modules/video_capture/video_capture_options.cc b/modules/video_capture/video_capture_options.cc index 64cd6216e7..dd102df9db 100644 --- a/modules/video_capture/video_capture_options.cc +++ b/modules/video_capture/video_capture_options.cc @@ -10,6 +10,9 @@ #include "modules/video_capture/video_capture_options.h" +#include "api/make_ref_counted.h" +#include "api/scoped_refptr.h" + #if defined(WEBRTC_USE_PIPEWIRE) #include "modules/video_capture/linux/pipewire_session.h" #endif From 4d52fa953c5ea7b1e16d961ea045a1ec19a40cac Mon Sep 17 00:00:00 2001 From: Fredrik Solenberg Date: Wed, 14 May 2025 19:43:45 +0200 Subject: [PATCH 162/292] Remove CpuInfo from system_wrappers 1. Create rtc_base/cpu_info.* with cpu_info namespace 2. Move implementation of system_wrappers/ CpuInfo::DetectNumberOfCores into cpu_info:: 3. Keep forwarding method in system_wrappers/ CpuInfo until downstream has been updated. Bug: webrtc:42228262 Change-Id: I74454e3c1807fffb12e2b7dcf4d7614164615ec7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391060 Reviewed-by: Harald Alvestrand Auto-Submit: Fredrik Solenberg Commit-Queue: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44634} --- call/BUILD.gn | 1 + call/call.cc | 4 +- modules/video_coding/BUILD.gn | 1 + .../test/videocodec_test_fixture_impl.cc | 4 +- rtc_base/BUILD.gn | 12 +++ rtc_base/cpu_info.cc | 80 +++++++++++++++++++ rtc_base/cpu_info.h | 27 +++++++ system_wrappers/BUILD.gn | 1 + system_wrappers/include/cpu_info.h | 1 + system_wrappers/source/cpu_info.cc | 56 +------------ test/pc/e2e/BUILD.gn | 1 + test/pc/e2e/peer_connection_quality_test.cc | 5 +- video/BUILD.gn | 1 + video/video_analyzer.cc | 5 +- 14 files changed, 135 insertions(+), 64 deletions(-) create mode 100644 rtc_base/cpu_info.cc create mode 100644 rtc_base/cpu_info.h diff --git a/call/BUILD.gn b/call/BUILD.gn index 073b4099e6..cf4ad08d15 100644 --- a/call/BUILD.gn +++ b/call/BUILD.gn @@ -333,6 +333,7 @@ rtc_library("call") { "../modules/video_coding:nack_requester", "../rtc_base:checks", "../rtc_base:copy_on_write_buffer", + "../rtc_base:cpu_info", "../rtc_base:event_tracer", "../rtc_base:logging", "../rtc_base:macromagic", diff --git a/call/call.cc b/call/call.cc index ace69e717a..d6d0e4b2d3 100644 --- a/call/call.cc +++ b/call/call.cc @@ -80,6 +80,7 @@ #include "modules/video_coding/nack_requester.h" #include "rtc_base/checks.h" #include "rtc_base/copy_on_write_buffer.h" +#include "rtc_base/cpu_info.h" #include "rtc_base/logging.h" #include "rtc_base/network/sent_packet.h" #include "rtc_base/strings/string_builder.h" @@ -90,7 +91,6 @@ #include "rtc_base/time_utils.h" #include "rtc_base/trace_event.h" #include "system_wrappers/include/clock.h" -#include "system_wrappers/include/cpu_info.h" #include "system_wrappers/include/metrics.h" #include "video/call_stats2.h" #include "video/config/video_encoder_config.h" @@ -718,7 +718,7 @@ Call::Call(CallConfig config, config.decode_metronome, worker_thread_) : nullptr), - num_cpu_cores_(CpuInfo::DetectNumberOfCores()), + num_cpu_cores_(cpu_info::DetectNumberOfCores()), call_stats_(new CallStats(&env_.clock(), worker_thread_)), bitrate_allocator_(new BitrateAllocator( this, diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn index 63661464a0..6314840119 100644 --- a/modules/video_coding/BUILD.gn +++ b/modules/video_coding/BUILD.gn @@ -981,6 +981,7 @@ if (rtc_include_tests) { "../../media:media_constants", "../../media:rtc_audio_video", "../../rtc_base:checks", + "../../rtc_base:cpu_info", "../../rtc_base:logging", "../../rtc_base:rtc_base_tests_utils", "../../rtc_base:stringutils", diff --git a/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc b/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc index 6d8f2eecdf..7372508ecb 100644 --- a/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc +++ b/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc @@ -65,6 +65,7 @@ #include "modules/video_coding/codecs/vp9/svc_config.h" #include "modules/video_coding/utility/ivf_file_writer.h" #include "rtc_base/checks.h" +#include "rtc_base/cpu_info.h" #include "rtc_base/cpu_time.h" #include "rtc_base/logging.h" #include "rtc_base/strings/string_builder.h" @@ -72,7 +73,6 @@ #include "rtc_base/system_time.h" #include "rtc_base/task_queue_for_test.h" #include "rtc_base/time_utils.h" -#include "system_wrappers/include/cpu_info.h" #include "system_wrappers/include/sleep.h" #include "test/gtest.h" #include "test/testsupport/file_utils.h" @@ -296,7 +296,7 @@ void VideoCodecTestFixtureImpl::Config::SetCodecSettings( } size_t VideoCodecTestFixtureImpl::Config::NumberOfCores() const { - return use_single_core ? 1 : CpuInfo::DetectNumberOfCores(); + return use_single_core ? 1 : cpu_info::DetectNumberOfCores(); } size_t VideoCodecTestFixtureImpl::Config::NumberOfTemporalLayers() const { diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn index 1766edd83e..dc6a50798a 100644 --- a/rtc_base/BUILD.gn +++ b/rtc_base/BUILD.gn @@ -1855,6 +1855,18 @@ if (rtc_rusty_base64) { } } +rtc_library("cpu_info") { + visibility = [ "*" ] + sources = [ + "cpu_info.cc", + "cpu_info.h", + ] + deps = [ + ":checks", + ":logging", + ] +} + if (rtc_include_tests) { if (rtc_enable_google_benchmarks) { rtc_test("base64_benchmark") { diff --git a/rtc_base/cpu_info.cc b/rtc_base/cpu_info.cc new file mode 100644 index 0000000000..99a5bc8f8a --- /dev/null +++ b/rtc_base/cpu_info.cc @@ -0,0 +1,80 @@ +/* + * Copyright (c) 2025 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "rtc_base/cpu_info.h" + +#include + +#if defined(WEBRTC_WIN) +#include +#elif defined(WEBRTC_LINUX) +#include +#elif defined(WEBRTC_MAC) +#include +#elif defined(WEBRTC_FUCHSIA) +#include +#endif + +#include "rtc_base/checks.h" +#include "rtc_base/logging.h" + +namespace { + +uint32_t DetectNumberOfCores() { + int number_of_cores = 0; + +#if defined(WEBRTC_WIN) + SYSTEM_INFO si; + GetNativeSystemInfo(&si); + number_of_cores = static_cast(si.dwNumberOfProcessors); +#elif defined(WEBRTC_LINUX) || defined(WEBRTC_ANDROID) + number_of_cores = static_cast(sysconf(_SC_NPROCESSORS_ONLN)); + if (number_of_cores <= 0) { + RTC_LOG(LS_ERROR) << "Failed to get number of cores"; + number_of_cores = 1; + } +#elif defined(WEBRTC_MAC) || defined(WEBRTC_IOS) + int name[] = {CTL_HW, HW_AVAILCPU}; + size_t size = sizeof(number_of_cores); + if (0 != sysctl(name, 2, &number_of_cores, &size, NULL, 0)) { + RTC_LOG(LS_ERROR) << "Failed to get number of cores"; + number_of_cores = 1; + } +#elif defined(WEBRTC_FUCHSIA) + number_of_cores = zx_system_get_num_cpus(); +#else + RTC_LOG(LS_ERROR) << "No function to get number of cores"; + number_of_cores = 1; +#endif + + RTC_LOG(LS_INFO) << "Available number of cores: " << number_of_cores; + + RTC_CHECK_GT(number_of_cores, 0); + return static_cast(number_of_cores); +} + +} // namespace + +namespace webrtc { + +namespace cpu_info { + +uint32_t DetectNumberOfCores() { + // Statically cache the number of system cores available since if the process + // is running in a sandbox, we may only be able to read the value once (before + // the sandbox is initialized) and not thereafter. + // For more information see crbug.com/176522. + static const uint32_t logical_cpus = ::DetectNumberOfCores(); + return logical_cpus; +} + +} // namespace cpu_info + +} // namespace webrtc diff --git a/rtc_base/cpu_info.h b/rtc_base/cpu_info.h new file mode 100644 index 0000000000..9a4f569f3f --- /dev/null +++ b/rtc_base/cpu_info.h @@ -0,0 +1,27 @@ +/* + * Copyright (c) 2025 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef RTC_BASE_CPU_INFO_H_ +#define RTC_BASE_CPU_INFO_H_ + +#include + +namespace webrtc { + +namespace cpu_info { + +// Returned number of cores is always >= 1. +uint32_t DetectNumberOfCores(); + +} // namespace cpu_info + +} // namespace webrtc + +#endif // RTC_BASE_CPU_INFO_H_ diff --git a/system_wrappers/BUILD.gn b/system_wrappers/BUILD.gn index 786c469689..bb22946ffd 100644 --- a/system_wrappers/BUILD.gn +++ b/system_wrappers/BUILD.gn @@ -35,6 +35,7 @@ rtc_library("system_wrappers") { "../api/units:timestamp", "../modules:module_api_public", "../rtc_base:checks", + "../rtc_base:cpu_info", "../rtc_base:logging", "../rtc_base:safe_conversions", "../rtc_base:timeutils", diff --git a/system_wrappers/include/cpu_info.h b/system_wrappers/include/cpu_info.h index ab546c7214..6bc99b6f2d 100644 --- a/system_wrappers/include/cpu_info.h +++ b/system_wrappers/include/cpu_info.h @@ -17,6 +17,7 @@ namespace webrtc { class CpuInfo { public: + // Wrapper kept while updating dependencies. static uint32_t DetectNumberOfCores(); private: diff --git a/system_wrappers/source/cpu_info.cc b/system_wrappers/source/cpu_info.cc index 8f72653269..90bf72077a 100644 --- a/system_wrappers/source/cpu_info.cc +++ b/system_wrappers/source/cpu_info.cc @@ -12,64 +12,12 @@ #include -#if defined(WEBRTC_WIN) -#include -#elif defined(WEBRTC_LINUX) -#include -#elif defined(WEBRTC_MAC) -#include -#elif defined(WEBRTC_FUCHSIA) -#include -#endif - -#include "rtc_base/checks.h" -#include "rtc_base/logging.h" - -namespace internal { -static int DetectNumberOfCores() { - int number_of_cores; - -#if defined(WEBRTC_WIN) - SYSTEM_INFO si; - GetNativeSystemInfo(&si); - number_of_cores = static_cast(si.dwNumberOfProcessors); -#elif defined(WEBRTC_LINUX) || defined(WEBRTC_ANDROID) - number_of_cores = static_cast(sysconf(_SC_NPROCESSORS_ONLN)); - if (number_of_cores <= 0) { - RTC_LOG(LS_ERROR) << "Failed to get number of cores"; - number_of_cores = 1; - } -#elif defined(WEBRTC_MAC) || defined(WEBRTC_IOS) - int name[] = {CTL_HW, HW_AVAILCPU}; - size_t size = sizeof(number_of_cores); - if (0 != sysctl(name, 2, &number_of_cores, &size, NULL, 0)) { - RTC_LOG(LS_ERROR) << "Failed to get number of cores"; - number_of_cores = 1; - } -#elif defined(WEBRTC_FUCHSIA) - number_of_cores = zx_system_get_num_cpus(); -#else - RTC_LOG(LS_ERROR) << "No function to get number of cores"; - number_of_cores = 1; -#endif - - RTC_LOG(LS_INFO) << "Available number of cores: " << number_of_cores; - - RTC_CHECK_GT(number_of_cores, 0); - return number_of_cores; -} -} // namespace internal +#include "rtc_base/cpu_info.h" namespace webrtc { uint32_t CpuInfo::DetectNumberOfCores() { - // Statically cache the number of system cores available since if the process - // is running in a sandbox, we may only be able to read the value once (before - // the sandbox is initialized) and not thereafter. - // For more information see crbug.com/176522. - static const uint32_t logical_cpus = - static_cast(::internal::DetectNumberOfCores()); - return logical_cpus; + return cpu_info::DetectNumberOfCores(); } } // namespace webrtc diff --git a/test/pc/e2e/BUILD.gn b/test/pc/e2e/BUILD.gn index 578b9922a0..7899fc1465 100644 --- a/test/pc/e2e/BUILD.gn +++ b/test/pc/e2e/BUILD.gn @@ -264,6 +264,7 @@ if (!build_with_chromium) { "../../../pc:pc_test_utils", "../../../pc:sdp_utils", "../../../rtc_base:checks", + "../../../rtc_base:cpu_info", "../../../rtc_base:gunit_helpers", "../../../rtc_base:logging", "../../../rtc_base:macromagic", diff --git a/test/pc/e2e/peer_connection_quality_test.cc b/test/pc/e2e/peer_connection_quality_test.cc index 764a1bce0f..f6e7cf2b57 100644 --- a/test/pc/e2e/peer_connection_quality_test.cc +++ b/test/pc/e2e/peer_connection_quality_test.cc @@ -51,13 +51,13 @@ #include "media/base/media_constants.h" #include "pc/test/mock_peer_connection_observers.h" #include "rtc_base/checks.h" +#include "rtc_base/cpu_info.h" #include "rtc_base/logging.h" #include "rtc_base/strings/string_builder.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/task_queue_for_test.h" #include "rtc_base/task_utils/repeating_task.h" #include "rtc_base/thread.h" -#include "system_wrappers/include/cpu_info.h" #include "system_wrappers/include/field_trial.h" #include "test/field_trial.h" #include "test/gtest.h" @@ -329,8 +329,7 @@ void PeerConnectionE2EQualityTest::Run(RunParams run_params) { [this]() { StartVideo(bob_video_sources_); }), bob_remote_audio_config, run_params.echo_emulation_config); - int num_cores = CpuInfo::DetectNumberOfCores(); - RTC_DCHECK_GE(num_cores, 1); + int num_cores = cpu_info::DetectNumberOfCores(); int video_analyzer_threads = num_cores - kPeerConnectionUsedThreads - kFrameworkUsedThreads; diff --git a/video/BUILD.gn b/video/BUILD.gn index 04ae5c23de..26be31300f 100644 --- a/video/BUILD.gn +++ b/video/BUILD.gn @@ -614,6 +614,7 @@ if (rtc_include_tests) { "../modules/video_coding:video_coding_utility", "../rtc_base:checks", "../rtc_base:copy_on_write_buffer", + "../rtc_base:cpu_info", "../rtc_base:logging", "../rtc_base:macromagic", "../rtc_base:platform_thread", diff --git a/video/video_analyzer.cc b/video/video_analyzer.cc index 159bd8ca2e..f25fc3be72 100644 --- a/video/video_analyzer.cc +++ b/video/video_analyzer.cc @@ -49,6 +49,7 @@ #include "modules/video_coding/codecs/vp9/include/vp9_globals.h" #include "rtc_base/checks.h" #include "rtc_base/copy_on_write_buffer.h" +#include "rtc_base/cpu_info.h" #include "rtc_base/cpu_time.h" #include "rtc_base/logging.h" #include "rtc_base/memory_usage.h" @@ -59,7 +60,6 @@ #include "rtc_base/task_utils/repeating_task.h" #include "rtc_base/time_utils.h" #include "system_wrappers/include/clock.h" -#include "system_wrappers/include/cpu_info.h" #include "test/gtest.h" #include "test/layer_filtering_transport.h" #include "test/rtp_file_reader.h" @@ -169,8 +169,7 @@ VideoAnalyzer::VideoAnalyzer(test::LayerFilteringTransport* transport, // Also, don't allocate more than kMaxComparisonThreads, even if there are // spare cores. - uint32_t num_cores = CpuInfo::DetectNumberOfCores(); - RTC_DCHECK_GE(num_cores, 1); + uint32_t num_cores = cpu_info::DetectNumberOfCores(); static const uint32_t kMinCoresLeft = 4; static const uint32_t kMaxComparisonThreads = 8; From 205c24ed95f5016eb1fe68a15a49f6294b20f2f1 Mon Sep 17 00:00:00 2001 From: Ilya Nikolaevskiy Date: Thu, 15 May 2025 12:57:30 +0200 Subject: [PATCH 163/292] Fix -Wshadow compile error in chromium import for screen_capturer_fuchsia.cc Bug: none Change-Id: I4392e79fec83efcbe3ee220860388b5a10330e8d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391800 Reviewed-by: Harald Alvestrand Commit-Queue: Ilya Nikolaevskiy Cr-Commit-Position: refs/heads/main@{#44635} --- modules/desktop_capture/screen_capturer_fuchsia.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/modules/desktop_capture/screen_capturer_fuchsia.cc b/modules/desktop_capture/screen_capturer_fuchsia.cc index 7294290ef8..5fb9e39058 100644 --- a/modules/desktop_capture/screen_capturer_fuchsia.cc +++ b/modules/desktop_capture/screen_capturer_fuchsia.cc @@ -387,7 +387,7 @@ void ScreenCapturerFuchsia::SetupBuffers() { const zx::vmo& virt_mem = buffer_collection_info_.buffers()[buffer_index].vmo(); virtual_memory_mapped_addrs_[buffer_index] = nullptr; - auto status = zx::vmar::root_self()->map( + status = zx::vmar::root_self()->map( ZX_VM_PERM_READ, /*vmar_offset*/ 0, virt_mem, /*vmo_offset*/ 0, virt_mem_bytes, reinterpret_cast( From 42e086f52dba30c905d7b24225d2e1ad8534f03d Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Thu, 15 May 2025 11:49:39 +0000 Subject: [PATCH 164/292] [cleanup] Replace uses of NULL with nullptr Generated using clang-tidy, tools/clang/scripts/build_clang_tools_extra.py \ --fetch out/Default clang-tidy clang-apply-replacements ninja -C out/Default gn gen out/Default --export-compile-commands cd out/Default tools/clang/third_party/llvm/clang-tools-extra/clang-tidy/tool/run-clang-tidy.py -p . \ -clang-tidy-binary out/Default/tools/clang/third_party/llvm/build/bin/clang-tidy \ -clang-apply-replacements-binary \ out/Default/tools/clang/third_party/llvm/build/bin/clang-apply-replacements \ -checks='-*,modernize-use-nullptr' \ -fix This CL was uploaded by git cl split. Bug: webrtc:42232595 Change-Id: I1ec03e2466ce56813213f73a149944486997cea4 No-Iwyu: LSC Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391841 Auto-Submit: Evan Shrubsole Reviewed-by: Danil Chapovalov Commit-Queue: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#44636} --- modules/rtp_rtcp/source/nack_rtx_unittest.cc | 2 +- modules/rtp_rtcp/source/receive_statistics_unittest.cc | 8 ++++---- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/modules/rtp_rtcp/source/nack_rtx_unittest.cc index 90e75a3f45..8f2265dc02 100644 --- a/modules/rtp_rtcp/source/nack_rtx_unittest.cc +++ b/modules/rtp_rtcp/source/nack_rtx_unittest.cc @@ -76,7 +76,7 @@ class RtxLoopBackTransport : public Transport { consecutive_drop_end_(0), rtx_ssrc_(rtx_ssrc), count_rtx_ssrc_(0), - module_(NULL) {} + module_(nullptr) {} void SetSendModule(RtpRtcpInterface* rtpRtcpModule) { module_ = rtpRtcpModule; diff --git a/modules/rtp_rtcp/source/receive_statistics_unittest.cc b/modules/rtp_rtcp/source/receive_statistics_unittest.cc index 492eaae29d..5eb552be45 100644 --- a/modules/rtp_rtcp/source/receive_statistics_unittest.cc +++ b/modules/rtp_rtcp/source/receive_statistics_unittest.cc @@ -132,7 +132,7 @@ TEST_P(ReceiveStatisticsTest, TwoIncomingSsrcs) { StreamStatistician* statistician = receive_statistics_->GetStatistician(kSsrc1); - ASSERT_TRUE(statistician != NULL); + ASSERT_TRUE(statistician != nullptr); EXPECT_GT(statistician->BitrateReceived(), 0u); StreamDataCounters counters = statistician->GetReceiveStreamDataCounters(); EXPECT_EQ(176u, counters.transmitted.payload_bytes); @@ -141,7 +141,7 @@ TEST_P(ReceiveStatisticsTest, TwoIncomingSsrcs) { EXPECT_EQ(2u, counters.transmitted.packets); statistician = receive_statistics_->GetStatistician(kSsrc2); - ASSERT_TRUE(statistician != NULL); + ASSERT_TRUE(statistician != nullptr); EXPECT_GT(statistician->BitrateReceived(), 0u); counters = statistician->GetReceiveStreamDataCounters(); EXPECT_EQ(576u, counters.transmitted.payload_bytes); @@ -237,7 +237,7 @@ TEST_P(ReceiveStatisticsTest, ActiveStatisticians) { EXPECT_EQ(1u, receive_statistics_->RtcpReportBlocks(3).size()); StreamStatistician* statistician = receive_statistics_->GetStatistician(kSsrc1); - ASSERT_TRUE(statistician != NULL); + ASSERT_TRUE(statistician != nullptr); StreamDataCounters counters = statistician->GetReceiveStreamDataCounters(); EXPECT_EQ(176u, counters.transmitted.payload_bytes); EXPECT_EQ(24u, counters.transmitted.header_bytes); @@ -260,7 +260,7 @@ TEST_P(ReceiveStatisticsTest, GetReceiveStreamDataCounters) { receive_statistics_->OnRtpPacket(packet1_); StreamStatistician* statistician = receive_statistics_->GetStatistician(kSsrc1); - ASSERT_TRUE(statistician != NULL); + ASSERT_TRUE(statistician != nullptr); StreamDataCounters counters = statistician->GetReceiveStreamDataCounters(); EXPECT_TRUE(counters.first_packet_time.IsFinite()); From f5588e2f194d7ec90e39dfa0e15f34653e1bb6fb Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Thu, 15 May 2025 11:53:44 +0000 Subject: [PATCH 165/292] [cleanup] Replace uses of NULL with nullptr Generated using clang-tidy, tools/clang/scripts/build_clang_tools_extra.py \ --fetch out/Default clang-tidy clang-apply-replacements ninja -C out/Default gn gen out/Default --export-compile-commands cd out/Default tools/clang/third_party/llvm/clang-tools-extra/clang-tidy/tool/run-clang-tidy.py -p . \ -clang-tidy-binary out/Default/tools/clang/third_party/llvm/build/bin/clang-tidy \ -clang-apply-replacements-binary \ out/Default/tools/clang/third_party/llvm/build/bin/clang-apply-replacements \ -checks='-*,modernize-use-nullptr' \ -fix This CL was uploaded by git cl split. Bug: webrtc:42232595 Change-Id: I356bf20cda6046c5b6d761e8d745d58f08307b99 No-Iwyu: LSC Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391863 Auto-Submit: Evan Shrubsole Commit-Queue: Mirko Bonadei Reviewed-by: Mirko Bonadei Cr-Commit-Position: refs/heads/main@{#44637} --- rtc_tools/frame_analyzer/video_quality_analysis_unittest.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/rtc_tools/frame_analyzer/video_quality_analysis_unittest.cc b/rtc_tools/frame_analyzer/video_quality_analysis_unittest.cc index 2b18388e81..7ad8ec1a03 100644 --- a/rtc_tools/frame_analyzer/video_quality_analysis_unittest.cc +++ b/rtc_tools/frame_analyzer/video_quality_analysis_unittest.cc @@ -138,7 +138,7 @@ TEST(VideoQualityAnalysisTest, std::string log_filename = TempFilename(test::OutputPath(), "log.log"); FILE* logfile = fopen(log_filename.c_str(), "w"); - ASSERT_TRUE(logfile != NULL); + ASSERT_TRUE(logfile != nullptr); result.max_repeated_frames = 1; result.max_skipped_frames = 0; From 4837af0b12cfab7cc2d556e4083ad0528d98d0d2 Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Thu, 15 May 2025 13:52:00 +0200 Subject: [PATCH 166/292] Poison global field trials To ensure most of the WebRTC non-test code query propagated field trials rather than use global field trials string. Bug: webrtc:42220378 Change-Id: I22cd767e5aab30b7598f42e334121ef2d7641ff2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/384620 Reviewed-by: Harald Alvestrand Commit-Queue: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#44638} --- api/BUILD.gn | 1 - api/transport/BUILD.gn | 1 + media/BUILD.gn | 3 --- modules/audio_coding/BUILD.gn | 1 - modules/audio_device/BUILD.gn | 1 - modules/audio_processing/agc/BUILD.gn | 1 - modules/audio_processing/agc2/BUILD.gn | 1 - modules/audio_processing/agc2/input_volume_controller.cc | 8 ++++++-- modules/audio_processing/ns/BUILD.gn | 1 - modules/congestion_controller/goog_cc/BUILD.gn | 1 - modules/congestion_controller/rtp/BUILD.gn | 1 - modules/pacing/BUILD.gn | 1 - p2p/BUILD.gn | 1 - rtc_base/BUILD.gn | 2 -- rtc_base/experiments/BUILD.gn | 4 ++-- rtc_base/experiments/quality_scaling_experiment.cc | 5 ++++- sdk/BUILD.gn | 1 - sdk/android/BUILD.gn | 1 - system_wrappers/BUILD.gn | 1 + video/BUILD.gn | 3 --- 20 files changed, 14 insertions(+), 25 deletions(-) diff --git a/api/BUILD.gn b/api/BUILD.gn index 52df41182f..6e32258531 100644 --- a/api/BUILD.gn +++ b/api/BUILD.gn @@ -319,7 +319,6 @@ rtc_library("libjingle_peerconnection_api") { ":data_channel_event_observer_interface", ":dtls_transport_interface", ":fec_controller_api", - ":field_trials", ":field_trials_view", ":frame_transformer_interface", ":ice_transport_interface", diff --git a/api/transport/BUILD.gn b/api/transport/BUILD.gn index 1dfce618d6..624788b855 100644 --- a/api/transport/BUILD.gn +++ b/api/transport/BUILD.gn @@ -62,6 +62,7 @@ rtc_library("network_control") { rtc_library("field_trial_based_config") { visibility = [ "*" ] + allow_poison = [ "environment_construction" ] sources = [ "field_trial_based_config.cc", "field_trial_based_config.h", diff --git a/media/BUILD.gn b/media/BUILD.gn index 3386c58c22..0ba26c7e82 100644 --- a/media/BUILD.gn +++ b/media/BUILD.gn @@ -174,7 +174,6 @@ rtc_library("video_adapter") { "../rtc_base:timeutils", "../rtc_base/synchronization:mutex", "../rtc_base/system:rtc_export", - "../system_wrappers:field_trial", ] } @@ -545,7 +544,6 @@ rtc_library("rtc_internal_video_codecs") { "../rtc_base:checks", "../rtc_base:logging", "../rtc_base/system:rtc_export", - "../system_wrappers:field_trial", "../test:fake_video_codecs", "//third_party/abseil-cpp/absl/container:inlined_vector", "//third_party/abseil-cpp/absl/strings", @@ -779,7 +777,6 @@ rtc_library("rtc_data_sctp_transport_factory") { deps += [ ":rtc_data_dcsctp_transport", "../system_wrappers", - "../system_wrappers:field_trial", ] } } diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn index 31a21d3a7d..d27943394a 100644 --- a/modules/audio_coding/BUILD.gn +++ b/modules/audio_coding/BUILD.gn @@ -435,7 +435,6 @@ rtc_library("audio_network_adaptor") { "../../rtc_base:timeutils", "../../rtc_base/system:file_wrapper", "../../system_wrappers", - "../../system_wrappers:field_trial", "//third_party/abseil-cpp/absl/algorithm:container", "//third_party/abseil-cpp/absl/strings:string_view", ] diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn index 9c9083d081..1bae2579b7 100644 --- a/modules/audio_device/BUILD.gn +++ b/modules/audio_device/BUILD.gn @@ -291,7 +291,6 @@ rtc_library("audio_device_impl") { "../../rtc_base/system:file_wrapper", "../../rtc_base/task_utils:repeating_task", "../../system_wrappers", - "../../system_wrappers:field_trial", "../../system_wrappers:metrics", "../utility", "//third_party/abseil-cpp/absl/base:core_headers", diff --git a/modules/audio_processing/agc/BUILD.gn b/modules/audio_processing/agc/BUILD.gn index 754b807879..61a450922f 100644 --- a/modules/audio_processing/agc/BUILD.gn +++ b/modules/audio_processing/agc/BUILD.gn @@ -34,7 +34,6 @@ rtc_library("agc") { "../../../rtc_base:gtest_prod", "../../../rtc_base:logging", "../../../rtc_base:safe_minmax", - "../../../system_wrappers:field_trial", "../../../system_wrappers:metrics", "../agc2:clipping_predictor", "../agc2:gain_map", diff --git a/modules/audio_processing/agc2/BUILD.gn b/modules/audio_processing/agc2/BUILD.gn index 3808a92ca2..86609504b3 100644 --- a/modules/audio_processing/agc2/BUILD.gn +++ b/modules/audio_processing/agc2/BUILD.gn @@ -216,7 +216,6 @@ rtc_library("input_volume_controller") { "../../../rtc_base:gtest_prod", "../../../rtc_base:logging", "../../../rtc_base:safe_minmax", - "../../../system_wrappers:field_trial", "../../../system_wrappers:metrics", ] } diff --git a/modules/audio_processing/agc2/input_volume_controller.cc b/modules/audio_processing/agc2/input_volume_controller.cc index 557c1e67d1..20fd6dc548 100644 --- a/modules/audio_processing/agc2/input_volume_controller.cc +++ b/modules/audio_processing/agc2/input_volume_controller.cc @@ -12,15 +12,19 @@ #include #include +#include +#include +#include -#include "api/array_view.h" +#include "api/audio/audio_processing.h" +#include "modules/audio_processing/agc2/clipping_predictor.h" #include "modules/audio_processing/agc2/gain_map_internal.h" #include "modules/audio_processing/agc2/input_volume_stats_reporter.h" +#include "modules/audio_processing/audio_buffer.h" #include "modules/audio_processing/include/audio_frame_view.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/numerics/safe_minmax.h" -#include "system_wrappers/include/field_trial.h" #include "system_wrappers/include/metrics.h" namespace webrtc { diff --git a/modules/audio_processing/ns/BUILD.gn b/modules/audio_processing/ns/BUILD.gn index 90aedfabdb..e90ef0e06a 100644 --- a/modules/audio_processing/ns/BUILD.gn +++ b/modules/audio_processing/ns/BUILD.gn @@ -60,7 +60,6 @@ rtc_static_library("ns") { "../../../rtc_base:safe_minmax", "../../../rtc_base/system:arch", "../../../system_wrappers", - "../../../system_wrappers:field_trial", "../../../system_wrappers:metrics", "../utility:cascaded_biquad_filter", ] diff --git a/modules/congestion_controller/goog_cc/BUILD.gn b/modules/congestion_controller/goog_cc/BUILD.gn index 326168679d..f970a33b94 100644 --- a/modules/congestion_controller/goog_cc/BUILD.gn +++ b/modules/congestion_controller/goog_cc/BUILD.gn @@ -79,7 +79,6 @@ rtc_library("alr_detector") { deps = [ "../../../api:field_trials_view", "../../../api/rtc_event_log", - "../../../api/transport:field_trial_based_config", "../../../logging:rtc_event_pacing", "../../../rtc_base:checks", "../../../rtc_base:safe_conversions", diff --git a/modules/congestion_controller/rtp/BUILD.gn b/modules/congestion_controller/rtp/BUILD.gn index 302c3dee01..f9e230ebb9 100644 --- a/modules/congestion_controller/rtp/BUILD.gn +++ b/modules/congestion_controller/rtp/BUILD.gn @@ -55,7 +55,6 @@ rtc_library("transport_feedback") { "../../../rtc_base/synchronization:mutex", "../../../rtc_base/system:no_unique_address", "../../../system_wrappers", - "../../../system_wrappers:field_trial", "../../rtp_rtcp:ntp_time_util", "../../rtp_rtcp:rtp_rtcp_format", "//third_party/abseil-cpp/absl/algorithm:container", diff --git a/modules/pacing/BUILD.gn b/modules/pacing/BUILD.gn index 21d529ff04..9faa33a254 100644 --- a/modules/pacing/BUILD.gn +++ b/modules/pacing/BUILD.gn @@ -40,7 +40,6 @@ rtc_library("pacing") { "../../api/rtc_event_log", "../../api/task_queue:pending_task_safety_flag", "../../api/task_queue:task_queue", - "../../api/transport:field_trial_based_config", "../../api/transport:network_control", "../../api/units:data_rate", "../../api/units:data_size", diff --git a/p2p/BUILD.gn b/p2p/BUILD.gn index a34e893ef0..6971391435 100644 --- a/p2p/BUILD.gn +++ b/p2p/BUILD.gn @@ -194,7 +194,6 @@ rtc_library("connection") { "../api:sequence_checker", "../api/task_queue", "../api/transport:enums", - "../api/transport:field_trial_based_config", "../api/transport:stun_types", "../api/units:time_delta", "../api/units:timestamp", diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn index dc6a50798a..58addef7aa 100644 --- a/rtc_base/BUILD.gn +++ b/rtc_base/BUILD.gn @@ -986,7 +986,6 @@ rtc_library("threading") { "../api/transport:ecn_marking", "../api/units:time_delta", "../api/units:timestamp", - "../system_wrappers:field_trial", "./network:ecn_marking", "synchronization:mutex", "system:no_unique_address", @@ -1287,7 +1286,6 @@ rtc_library("async_udp_socket") { "../api:sequence_checker", "../api/units:time_delta", "../api/units:timestamp", - "../system_wrappers:field_trial", "network:received_packet", "network:sent_packet", "system:no_unique_address", diff --git a/rtc_base/experiments/BUILD.gn b/rtc_base/experiments/BUILD.gn index 9f5573c0a1..a9f5c7d5cf 100644 --- a/rtc_base/experiments/BUILD.gn +++ b/rtc_base/experiments/BUILD.gn @@ -62,9 +62,10 @@ rtc_library("quality_scaling_experiment") { "quality_scaling_experiment.h", ] deps = [ + "..:checks", "..:logging", "../../api:field_trials_view", - "../../api/transport:field_trial_based_config", + "../../api/video:video_frame", "../../api/video_codecs:video_codecs_api", "../../rtc_base/experiments:field_trial_parser", "//third_party/abseil-cpp/absl/strings", @@ -92,7 +93,6 @@ rtc_library("balanced_degradation_settings") { "..:logging", "../../api:field_trials_view", "../../api/video_codecs:video_codecs_api", - "../../system_wrappers:field_trial", ] } diff --git a/rtc_base/experiments/quality_scaling_experiment.cc b/rtc_base/experiments/quality_scaling_experiment.cc index be563639e0..30a80d3b64 100644 --- a/rtc_base/experiments/quality_scaling_experiment.cc +++ b/rtc_base/experiments/quality_scaling_experiment.cc @@ -11,11 +11,14 @@ #include +#include #include #include "absl/strings/match.h" #include "api/field_trials_view.h" -#include "api/transport/field_trial_based_config.h" +#include "api/video/video_codec_type.h" +#include "api/video_codecs/video_encoder.h" +#include "rtc_base/checks.h" #include "rtc_base/experiments/field_trial_parser.h" #include "rtc_base/logging.h" diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn index c443527044..417efe6423 100644 --- a/sdk/BUILD.gn +++ b/sdk/BUILD.gn @@ -326,7 +326,6 @@ if (is_ios || is_mac) { "../rtc_base:refcount", "../rtc_base:threading", "../rtc_base:timeutils", - "../system_wrappers:field_trial", "../system_wrappers:metrics", ] diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn index f778eeb318..f1e75262b3 100644 --- a/sdk/android/BUILD.gn +++ b/sdk/android/BUILD.gn @@ -620,7 +620,6 @@ if (current_os == "linux" || is_android) { "../../rtc_base:refcount", "../../rtc_base:stringutils", "../../rtc_base:threading", - "../../system_wrappers:field_trial", "../../system_wrappers:metrics", "//third_party/abseil-cpp/absl/strings:string_view", "//third_party/jni_zero", diff --git a/system_wrappers/BUILD.gn b/system_wrappers/BUILD.gn index bb22946ffd..c708f81a73 100644 --- a/system_wrappers/BUILD.gn +++ b/system_wrappers/BUILD.gn @@ -80,6 +80,7 @@ rtc_library("system_wrappers") { rtc_library("field_trial") { visibility = [ "*" ] + poisonous = [ "environment_construction" ] public = [ "include/field_trial.h" ] sources = [ "source/field_trial.cc" ] if (rtc_exclude_field_trial_default) { diff --git a/video/BUILD.gn b/video/BUILD.gn index 26be31300f..dec17dde3d 100644 --- a/video/BUILD.gn +++ b/video/BUILD.gn @@ -182,7 +182,6 @@ rtc_library("video") { "../rtc_base/system:no_unique_address", "../rtc_base/task_utils:repeating_task", "../system_wrappers", - "../system_wrappers:field_trial", "../system_wrappers:metrics", "../video/config:encoder_config", "../video/corruption_detection:frame_instrumentation_evaluation", @@ -277,7 +276,6 @@ rtc_library("frame_cadence_adapter") { "../rtc_base/system:unused", "../rtc_base/task_utils:repeating_task", "../system_wrappers", - "../system_wrappers:field_trial", "../system_wrappers:metrics", "//third_party/abseil-cpp/absl/algorithm:container", "//third_party/abseil-cpp/absl/base:core_headers", @@ -319,7 +317,6 @@ rtc_library("video_stream_buffer_controller") { "../rtc_base/experiments:field_trial_parser", "../rtc_base/system:no_unique_address", "../system_wrappers", - "../system_wrappers:field_trial", "//third_party/abseil-cpp/absl/base:core_headers", "//third_party/abseil-cpp/absl/container:inlined_vector", "//third_party/abseil-cpp/absl/functional:bind_front", From 4ac9a99f0f3163b7137c734b5548f1d69921de53 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Thu, 15 May 2025 11:53:30 +0000 Subject: [PATCH 167/292] [cleanup] Replace uses of NULL with nullptr Generated using clang-tidy, tools/clang/scripts/build_clang_tools_extra.py \ --fetch out/Default clang-tidy clang-apply-replacements ninja -C out/Default gn gen out/Default --export-compile-commands cd out/Default tools/clang/third_party/llvm/clang-tools-extra/clang-tidy/tool/run-clang-tidy.py -p . \ -clang-tidy-binary out/Default/tools/clang/third_party/llvm/build/bin/clang-tidy \ -clang-apply-replacements-binary \ out/Default/tools/clang/third_party/llvm/build/bin/clang-apply-replacements \ -checks='-*,modernize-use-nullptr' \ -fix This CL was uploaded by git cl split. Bug: webrtc:42232595 Change-Id: I1520f776ba878d8b92b34c7f76d2712427a52c00 No-Iwyu: LSC Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391644 Reviewed-by: Mirko Bonadei Auto-Submit: Evan Shrubsole Commit-Queue: Mirko Bonadei Cr-Commit-Position: refs/heads/main@{#44639} --- rtc_base/firewall_socket_server.cc | 2 +- rtc_base/memory/aligned_malloc.cc | 10 +++++----- rtc_base/memory/aligned_malloc_unittest.cc | 4 ++-- rtc_base/openssl_adapter.cc | 2 +- rtc_base/openssl_stream_adapter.cc | 2 +- rtc_base/platform_thread.cc | 2 +- 6 files changed, 11 insertions(+), 11 deletions(-) diff --git a/rtc_base/firewall_socket_server.cc b/rtc_base/firewall_socket_server.cc index 1301081bdf..24b1ec8d11 100644 --- a/rtc_base/firewall_socket_server.cc +++ b/rtc_base/firewall_socket_server.cc @@ -110,7 +110,7 @@ class FirewallSocket : public AsyncSocketAdapter { << addr.ToSensitiveString() << " to " << GetLocalAddress().ToSensitiveString() << " denied"; } - return 0; + return nullptr; } private: diff --git a/rtc_base/memory/aligned_malloc.cc b/rtc_base/memory/aligned_malloc.cc index 7add079398..e0bae74dc3 100644 --- a/rtc_base/memory/aligned_malloc.cc +++ b/rtc_base/memory/aligned_malloc.cc @@ -41,10 +41,10 @@ bool ValidAlignment(size_t alignment) { void* GetRightAlign(const void* pointer, size_t alignment) { if (!pointer) { - return NULL; + return nullptr; } if (!ValidAlignment(alignment)) { - return NULL; + return nullptr; } uintptr_t start_pos = reinterpret_cast(pointer); return reinterpret_cast(GetRightAlign(start_pos, alignment)); @@ -52,10 +52,10 @@ void* GetRightAlign(const void* pointer, size_t alignment) { void* AlignedMalloc(size_t size, size_t alignment) { if (size == 0) { - return NULL; + return nullptr; } if (!ValidAlignment(alignment)) { - return NULL; + return nullptr; } // The memory is aligned towards the lowest address that so only @@ -83,7 +83,7 @@ void* AlignedMalloc(size_t size, size_t alignment) { } void AlignedFree(void* mem_block) { - if (mem_block == NULL) { + if (mem_block == nullptr) { return; } uintptr_t aligned_pos = reinterpret_cast(mem_block); diff --git a/rtc_base/memory/aligned_malloc_unittest.cc b/rtc_base/memory/aligned_malloc_unittest.cc index 8e750035f0..5c3f02d15c 100644 --- a/rtc_base/memory/aligned_malloc_unittest.cc +++ b/rtc_base/memory/aligned_malloc_unittest.cc @@ -26,7 +26,7 @@ namespace webrtc { bool CorrectUsage(size_t size, size_t alignment) { std::unique_ptr scoped( static_cast(AlignedMalloc(size, alignment))); - if (scoped.get() == NULL) { + if (scoped.get() == nullptr) { return false; } const uintptr_t scoped_address = reinterpret_cast(scoped.get()); @@ -39,7 +39,7 @@ TEST(AlignedMalloc, GetRightAlign) { const size_t left_misalignment = 1; std::unique_ptr scoped( static_cast(AlignedMalloc(size, alignment))); - EXPECT_TRUE(scoped.get() != NULL); + EXPECT_TRUE(scoped.get() != nullptr); const uintptr_t aligned_address = reinterpret_cast(scoped.get()); const uintptr_t misaligned_address = aligned_address - left_misalignment; const char* misaligned_ptr = diff --git a/rtc_base/openssl_adapter.cc b/rtc_base/openssl_adapter.cc index f804f08da3..b25e9fb027 100644 --- a/rtc_base/openssl_adapter.cc +++ b/rtc_base/openssl_adapter.cc @@ -102,7 +102,7 @@ static BIO* BIO_new_socket(webrtc::Socket* socket) { static int socket_new(BIO* b) { BIO_set_shutdown(b, 0); BIO_set_init(b, 1); - BIO_set_data(b, 0); + BIO_set_data(b, nullptr); return 1; } diff --git a/rtc_base/openssl_stream_adapter.cc b/rtc_base/openssl_stream_adapter.cc index 3a0a55884e..944567b43c 100644 --- a/rtc_base/openssl_stream_adapter.cc +++ b/rtc_base/openssl_stream_adapter.cc @@ -196,7 +196,7 @@ static BIO* BIO_new_stream(StreamInterface* stream) { static int stream_new(BIO* b) { BIO_set_shutdown(b, 0); BIO_set_init(b, 1); - BIO_set_data(b, 0); + BIO_set_data(b, nullptr); return 1; } diff --git a/rtc_base/platform_thread.cc b/rtc_base/platform_thread.cc index 53381a4d4e..7f11696b21 100644 --- a/rtc_base/platform_thread.cc +++ b/rtc_base/platform_thread.cc @@ -108,7 +108,7 @@ void* RunPlatformThread(void* param) { auto function = static_cast*>(param); (*function)(); delete function; - return 0; + return nullptr; } #endif // defined(WEBRTC_WIN) From b92c9139c4970b24347834dee7222099caa7d09f Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Thu, 15 May 2025 11:50:11 +0000 Subject: [PATCH 168/292] [cleanup] Replace uses of NULL with nullptr MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Generated using clang-tidy, tools/clang/scripts/build_clang_tools_extra.py \ --fetch out/Default clang-tidy clang-apply-replacements ninja -C out/Default gn gen out/Default --export-compile-commands cd out/Default tools/clang/third_party/llvm/clang-tools-extra/clang-tidy/tool/run-clang-tidy.py -p . \ -clang-tidy-binary out/Default/tools/clang/third_party/llvm/build/bin/clang-tidy \ -clang-apply-replacements-binary \ out/Default/tools/clang/third_party/llvm/build/bin/clang-apply-replacements \ -checks='-*,modernize-use-nullptr' \ -fix This CL was uploaded by git cl split. Bug: webrtc:42232595 Change-Id: I0fc15379919b3f7fc780846fa85b934d7766c608 No-Iwyu: LSC Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391842 Auto-Submit: Evan Shrubsole Commit-Queue: Jakob Ivarsson‎ Reviewed-by: Jakob Ivarsson‎ Cr-Commit-Position: refs/heads/main@{#44640} --- audio/channel_send.cc | 2 +- .../audio_coding/acm2/audio_coding_module.cc | 2 +- .../acm2/audio_coding_module_unittest.cc | 6 ++-- .../audio_coding/codecs/cng/cng_unittest.cc | 4 +-- .../audio_coding/codecs/g711/test/testG711.cc | 8 +++--- .../audio_coding/codecs/g722/test/testG722.cc | 6 ++-- .../audio_coding/codecs/opus/opus_fec_test.cc | 10 +++---- .../codecs/opus/opus_interface.cc | 18 ++++++------ .../codecs/opus/opus_speed_test.cc | 4 +-- .../audio_coding/codecs/opus/opus_unittest.cc | 20 ++++++------- .../red/audio_encoder_copy_red_unittest.cc | 6 ++-- .../codecs/tools/audio_codec_speed_test.cc | 2 +- .../neteq/audio_decoder_unittest.cc | 4 +-- .../neteq/audio_multi_vector_unittest.cc | 2 +- .../audio_coding/neteq/decoder_database.cc | 6 ++-- .../neteq/decoder_database_unittest.cc | 22 +++++++-------- .../neteq/dtmf_tone_generator_unittest.cc | 2 +- modules/audio_coding/neteq/expand_unittest.cc | 2 +- .../audio_coding/neteq/neteq_impl_unittest.cc | 7 +++-- .../neteq/packet_buffer_unittest.cc | 2 +- .../neteq/test/neteq_opus_quality_test.cc | 4 +-- .../audio_coding/neteq/test/result_sink.cc | 2 +- .../neteq/time_stretch_unittest.cc | 4 +-- .../neteq/timestamp_scaler_unittest.cc | 2 +- .../neteq/tools/rtp_file_source.cc | 2 +- modules/audio_coding/neteq/tools/rtpcat.cc | 4 +-- modules/audio_coding/test/Channel.cc | 6 ++-- modules/audio_coding/test/EncodeDecodeTest.cc | 2 +- modules/audio_coding/test/PCMFile.cc | 12 ++++---- modules/audio_coding/test/RTPFile.cc | 8 +++--- modules/audio_coding/test/TestAllCodecs.cc | 8 +++--- modules/audio_coding/test/TestRedFec.cc | 6 ++-- modules/audio_coding/test/TestStereo.cc | 14 +++++----- modules/audio_coding/test/opus_test.cc | 28 +++++++++---------- 34 files changed, 119 insertions(+), 118 deletions(-) diff --git a/audio/channel_send.cc b/audio/channel_send.cc index e953549ecf..2077d97718 100644 --- a/audio/channel_send.cc +++ b/audio/channel_send.cc @@ -549,7 +549,7 @@ ChannelSend::~ChannelSend() { frame_transformer_delegate_->Reset(); StopSend(); - int error = audio_coding_->RegisterTransportCallback(NULL); + int error = audio_coding_->RegisterTransportCallback(nullptr); RTC_DCHECK_EQ(0, error); // Delete the encoder task queue first to ensure that there are no running diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc index 6f69012755..373e3a34c1 100644 --- a/modules/audio_coding/acm2/audio_coding_module.cc +++ b/modules/audio_coding/acm2/audio_coding_module.cc @@ -221,7 +221,7 @@ AudioCodingModuleImpl::AudioCodingModuleImpl() previous_pltype_(255), first_10ms_data_(false), first_frame_(true), - packetization_callback_(NULL), + packetization_callback_(nullptr), codec_histogram_bins_log_(), number_of_consecutive_empty_packets_(0) { RTC_LOG(LS_INFO) << "Created"; diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc index 301da4aef4..9d5916931b 100644 --- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc +++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc @@ -666,7 +666,7 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test, audio_source_.reset(new test::InputAudioFile(input_file_name)); send_test_.reset(new test::AcmSendTestOldApi(audio_source_.get(), source_rate, kTestDurationMs)); - return send_test_.get() != NULL; + return send_test_.get() != nullptr; } // Registers a send codec in the test::AcmSendTest object. Returns true on @@ -758,7 +758,7 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test, std::unique_ptr NextPacket() override { auto packet = send_test_->NextPacket(); if (!packet) - return NULL; + return nullptr; VerifyPacket(packet.get()); // TODO(henrik.lundin) Save the packet to file as well. @@ -1338,7 +1338,7 @@ class AcmSwitchingOutputFrequencyOldApi : public ::testing::Test, // "manually". if (num_packets_++ > kTestNumPackets) { EXPECT_TRUE(has_toggled_); - return NULL; // Test ended. + return nullptr; // Test ended. } // Get the next packet from the source. diff --git a/modules/audio_coding/codecs/cng/cng_unittest.cc b/modules/audio_coding/codecs/cng/cng_unittest.cc index a486cc01b1..01b7fbfb1c 100644 --- a/modules/audio_coding/codecs/cng/cng_unittest.cc +++ b/modules/audio_coding/codecs/cng/cng_unittest.cc @@ -52,11 +52,11 @@ void CngTest::SetUp() { const std::string file_name = test::ResourcePath("audio_coding/testfile32kHz", "pcm"); input_file = fopen(file_name.c_str(), "rb"); - ASSERT_TRUE(input_file != NULL); + ASSERT_TRUE(input_file != nullptr); ASSERT_EQ(640, static_cast( fread(speech_data_, sizeof(int16_t), 640, input_file))); fclose(input_file); - input_file = NULL; + input_file = nullptr; } void CngTest::TestCngEncode(int sample_rate_hz, int quality) { diff --git a/modules/audio_coding/codecs/g711/test/testG711.cc b/modules/audio_coding/codecs/g711/test/testG711.cc index 6117045fb4..770836c337 100644 --- a/modules/audio_coding/codecs/g711/test/testG711.cc +++ b/modules/audio_coding/codecs/g711/test/testG711.cc @@ -38,7 +38,7 @@ int main(int argc, char* argv[]) { char inname[80], outname[40], bitname[40]; FILE* inp; FILE* outp; - FILE* bitp = NULL; + FILE* bitp = nullptr; int framecnt; bool endfile; @@ -94,17 +94,17 @@ int main(int argc, char* argv[]) { sscanf(argv[4], "%s", outname); if (argc == 6) { sscanf(argv[5], "%s", bitname); - if ((bitp = fopen(bitname, "wb")) == NULL) { + if ((bitp = fopen(bitname, "wb")) == nullptr) { printf(" G.711: Cannot read file %s.\n", bitname); exit(1); } } - if ((inp = fopen(inname, "rb")) == NULL) { + if ((inp = fopen(inname, "rb")) == nullptr) { printf(" G.711: Cannot read file %s.\n", inname); exit(1); } - if ((outp = fopen(outname, "wb")) == NULL) { + if ((outp = fopen(outname, "wb")) == nullptr) { printf(" G.711: Cannot write file %s.\n", outname); exit(1); } diff --git a/modules/audio_coding/codecs/g722/test/testG722.cc b/modules/audio_coding/codecs/g722/test/testG722.cc index 4f65e9938c..fcff08edd2 100644 --- a/modules/audio_coding/codecs/g722/test/testG722.cc +++ b/modules/audio_coding/codecs/g722/test/testG722.cc @@ -87,15 +87,15 @@ int main(int argc, char* argv[]) { sscanf(argv[3], "%s", outbit); sscanf(argv[4], "%s", outname); - if ((inp = fopen(inname, "rb")) == NULL) { + if ((inp = fopen(inname, "rb")) == nullptr) { printf(" G.722: Cannot read file %s.\n", inname); exit(1); } - if ((outbitp = fopen(outbit, "wb")) == NULL) { + if ((outbitp = fopen(outbit, "wb")) == nullptr) { printf(" G.722: Cannot write file %s.\n", outbit); exit(1); } - if ((outp = fopen(outname, "wb")) == NULL) { + if ((outp = fopen(outname, "wb")) == nullptr) { printf(" G.722: Cannot write file %s.\n", outname); exit(1); } diff --git a/modules/audio_coding/codecs/opus/opus_fec_test.cc b/modules/audio_coding/codecs/opus/opus_fec_test.cc index c48f486181..bf38347148 100644 --- a/modules/audio_coding/codecs/opus/opus_fec_test.cc +++ b/modules/audio_coding/codecs/opus/opus_fec_test.cc @@ -81,7 +81,7 @@ void OpusFecTest::SetUp() { in_filename_ = test::ResourcePath(get<2>(GetParam()), get<3>(GetParam())); FILE* fp = fopen(in_filename_.c_str(), "rb"); - ASSERT_FALSE(fp == NULL); + ASSERT_FALSE(fp == nullptr); // Obtain file size. fseek(fp, 0, SEEK_END); @@ -135,8 +135,8 @@ OpusFecTest::OpusFecTest() data_pointer_(0), max_bytes_(0), encoded_bytes_(0), - opus_encoder_(NULL), - opus_decoder_(NULL) {} + opus_encoder_(nullptr), + opus_decoder_(nullptr) {} void OpusFecTest::EncodeABlock() { int value = @@ -161,8 +161,8 @@ void OpusFecTest::DecodeABlock(bool lost_previous, bool lost_current) { } else { // Call decoder PLC. while (value_1 < static_cast(block_length_sample_)) { - int ret = WebRtcOpus_Decode(opus_decoder_, NULL, 0, &out_data_[value_1], - &audio_type); + int ret = WebRtcOpus_Decode(opus_decoder_, nullptr, 0, + &out_data_[value_1], &audio_type); EXPECT_EQ(ret, sampling_khz_ * 10); // Should return 10 ms of samples. value_1 += ret; } diff --git a/modules/audio_coding/codecs/opus/opus_interface.cc b/modules/audio_coding/codecs/opus/opus_interface.cc index 9dc579a4b7..f934cca073 100644 --- a/modules/audio_coding/codecs/opus/opus_interface.cc +++ b/modules/audio_coding/codecs/opus/opus_interface.cc @@ -381,10 +381,10 @@ int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int error; OpusDecInst* state; - if (inst != NULL) { + if (inst != nullptr) { // Create Opus decoder state. state = reinterpret_cast(calloc(1, sizeof(OpusDecInst))); - if (state == NULL) { + if (state == nullptr) { return -1; } @@ -418,10 +418,10 @@ int16_t WebRtcOpus_MultistreamDecoderCreate( int error; OpusDecInst* state; - if (inst != NULL) { + if (inst != nullptr) { // Create Opus decoder state. state = reinterpret_cast(calloc(1, sizeof(OpusDecInst))); - if (state == NULL) { + if (state == nullptr) { return -1; } @@ -532,7 +532,7 @@ static int DecodePlc(OpusDecInst* inst, int16_t* decoded) { FrameSizePerChannel(kWebRtcOpusPlcFrameSizeMs, inst->sample_rate_hz); decoded_samples = - DecodeNative(inst, NULL, 0, plc_samples, decoded, &audio_type, 0); + DecodeNative(inst, nullptr, 0, plc_samples, decoded, &audio_type, 0); if (decoded_samples < 0) { return -1; } @@ -678,7 +678,7 @@ int WebRtcOpus_NumSilkFrames(const uint8_t* payload) { // parsing the LP layer of an Opus packet, particularly the LBRR flag. int WebRtcOpus_PacketHasFec(const uint8_t* payload, size_t payload_length_bytes) { - if (payload == NULL || payload_length_bytes == 0) + if (payload == nullptr || payload_length_bytes == 0) return 0; // In CELT_ONLY mode, packets should not have FEC. @@ -699,7 +699,7 @@ int WebRtcOpus_PacketHasFec(const uint8_t* payload, // Parse packet to get the frames. But we only care about the first frame, // since we can only decode the FEC from the first one. if (opus_packet_parse(payload, static_cast(payload_length_bytes), - NULL, frame_data, frame_sizes, NULL) < 0) { + nullptr, frame_data, frame_sizes, nullptr) < 0) { return 0; } @@ -727,7 +727,7 @@ int WebRtcOpus_PacketHasFec(const uint8_t* payload, int WebRtcOpus_PacketHasVoiceActivity(const uint8_t* payload, size_t payload_length_bytes) { - if (payload == NULL || payload_length_bytes == 0) + if (payload == nullptr || payload_length_bytes == 0) return 0; // In CELT_ONLY mode we can not determine whether there is VAD. @@ -748,7 +748,7 @@ int WebRtcOpus_PacketHasVoiceActivity(const uint8_t* payload, // Parse packet to get the frames. int frames = opus_packet_parse(payload, static_cast(payload_length_bytes), - NULL, frame_data, frame_sizes, NULL); + nullptr, frame_data, frame_sizes, nullptr); if (frames < 0) return -1; diff --git a/modules/audio_coding/codecs/opus/opus_speed_test.cc b/modules/audio_coding/codecs/opus/opus_speed_test.cc index 62bf48ce4a..6bc015ff88 100644 --- a/modules/audio_coding/codecs/opus/opus_speed_test.cc +++ b/modules/audio_coding/codecs/opus/opus_speed_test.cc @@ -46,8 +46,8 @@ OpusSpeedTest::OpusSpeedTest() : AudioCodecSpeedTest(kOpusBlockDurationMs, kOpusSamplingKhz, kOpusSamplingKhz), - opus_encoder_(NULL), - opus_decoder_(NULL) {} + opus_encoder_(nullptr), + opus_decoder_(nullptr) {} void OpusSpeedTest::SetUp() { AudioCodecSpeedTest::SetUp(); diff --git a/modules/audio_coding/codecs/opus/opus_unittest.cc b/modules/audio_coding/codecs/opus/opus_unittest.cc index 674e50e0f4..af00347c3e 100644 --- a/modules/audio_coding/codecs/opus/opus_unittest.cc +++ b/modules/audio_coding/codecs/opus/opus_unittest.cc @@ -240,8 +240,8 @@ int OpusTest::EncodeDecode(WebRtcOpusEncInst* encoder, decoder_sample_rate_hz_ / encoder_sample_rate_hz_; while (total_dtx_len < output_samples_per_channel) { - int est_len = WebRtcOpus_DurationEst(decoder, NULL, 0); - int act_len = WebRtcOpus_Decode(decoder, NULL, 0, + int est_len = WebRtcOpus_DurationEst(decoder, nullptr, 0); + int act_len = WebRtcOpus_Decode(decoder, nullptr, 0, &output_audio[total_dtx_len * channels_], audio_type); EXPECT_EQ(est_len, act_len); @@ -476,7 +476,7 @@ TEST(OpusTest, OpusCreateFail) { WebRtcOpusDecInst* opus_decoder; // Test to see that an invalid pointer is caught. - EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(NULL, 1, 0, 48000)); + EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(nullptr, 1, 0, 48000)); // Invalid channel number. EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(&opus_encoder, 257, 0, 48000)); // Invalid applciation mode. @@ -484,7 +484,7 @@ TEST(OpusTest, OpusCreateFail) { // Invalid sample rate. EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(&opus_encoder, 1, 0, 12345)); - EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(NULL, 1, 48000)); + EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(nullptr, 1, 48000)); // Invalid channel number. EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(&opus_decoder, 257, 48000)); // Invalid sample rate. @@ -494,8 +494,8 @@ TEST(OpusTest, OpusCreateFail) { // Test failing Free. TEST(OpusTest, OpusFreeFail) { // Test to see that an invalid pointer is caught. - EXPECT_EQ(-1, WebRtcOpus_EncoderFree(NULL)); - EXPECT_EQ(-1, WebRtcOpus_DecoderFree(NULL)); + EXPECT_EQ(-1, WebRtcOpus_EncoderFree(nullptr)); + EXPECT_EQ(-1, WebRtcOpus_DecoderFree(nullptr)); } // Test normal Create and Free. @@ -504,8 +504,8 @@ TEST_P(OpusTest, OpusCreateFree) { use_multistream_, encoder_sample_rate_hz_); CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_, decoder_sample_rate_hz_); - EXPECT_TRUE(opus_encoder_ != NULL); - EXPECT_TRUE(opus_decoder_ != NULL); + EXPECT_TRUE(opus_encoder_ != nullptr); + EXPECT_TRUE(opus_decoder_ != nullptr); // Free encoder and decoder memory. EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_)); EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_)); @@ -838,8 +838,8 @@ TEST_P(OpusTest, OpusDecodePlc) { constexpr int kPlcDurationMs = 10; const int plc_samples = decoder_sample_rate_hz_ * kPlcDurationMs / 1000; int16_t* plc_buffer = new int16_t[plc_samples * channels_]; - EXPECT_EQ(plc_samples, - WebRtcOpus_Decode(opus_decoder_, NULL, 0, plc_buffer, &audio_type)); + EXPECT_EQ(plc_samples, WebRtcOpus_Decode(opus_decoder_, nullptr, 0, + plc_buffer, &audio_type)); // Free memory. delete[] plc_buffer; diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc index 1ba411beb0..cc791dde44 100644 --- a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc +++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc @@ -68,7 +68,7 @@ class AudioEncoderCopyRedTest : public ::testing::Test { void TearDown() override { red_.reset(); } void Encode() { - ASSERT_TRUE(red_.get() != NULL); + ASSERT_TRUE(red_.get() != nullptr); encoded_.Clear(); encoded_info_ = red_->Encode( timestamp_, ArrayView(audio_, num_audio_samples_10ms), @@ -650,9 +650,9 @@ TEST_F(AudioEncoderCopyRedDeathTest, WrongFrameSize) { TEST_F(AudioEncoderCopyRedDeathTest, NullSpeechEncoder) { test::ScopedKeyValueConfig field_trials; - AudioEncoderCopyRed* red = NULL; + AudioEncoderCopyRed* red = nullptr; AudioEncoderCopyRed::Config config; - config.speech_encoder = NULL; + config.speech_encoder = nullptr; RTC_EXPECT_DEATH( red = new AudioEncoderCopyRed(std::move(config), field_trials), "Speech encoder not provided."); diff --git a/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc b/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc index f039ab6587..6fb58911b9 100644 --- a/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc +++ b/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc @@ -40,7 +40,7 @@ AudioCodecSpeedTest::AudioCodecSpeedTest(int block_duration_ms, encoded_bytes_(0), encoding_time_ms_(0.0), decoding_time_ms_(0.0), - out_file_(NULL) {} + out_file_(nullptr) {} void AudioCodecSpeedTest::SetUp() { channels_ = get<0>(GetParam()); diff --git a/modules/audio_coding/neteq/audio_decoder_unittest.cc b/modules/audio_coding/neteq/audio_decoder_unittest.cc index 400035353b..d8b891298e 100644 --- a/modules/audio_coding/neteq/audio_decoder_unittest.cc +++ b/modules/audio_coding/neteq/audio_decoder_unittest.cc @@ -110,7 +110,7 @@ class AudioDecoderTest : public ::testing::Test { data_length_(0), channels_(1), payload_type_(17), - decoder_(NULL) {} + decoder_(nullptr) {} ~AudioDecoderTest() override {} @@ -123,7 +123,7 @@ class AudioDecoderTest : public ::testing::Test { void TearDown() override { delete decoder_; - decoder_ = NULL; + decoder_ = nullptr; } virtual void InitEncoder() {} diff --git a/modules/audio_coding/neteq/audio_multi_vector_unittest.cc b/modules/audio_coding/neteq/audio_multi_vector_unittest.cc index c7b6311867..61938476ee 100644 --- a/modules/audio_coding/neteq/audio_multi_vector_unittest.cc +++ b/modules/audio_coding/neteq/audio_multi_vector_unittest.cc @@ -120,7 +120,7 @@ TEST_P(AudioMultiVectorTest, PushBackInterleavedAndCopy) { // Try to copy to a NULL pointer. Nothing should happen. TEST_P(AudioMultiVectorTest, CopyToNull) { AudioMultiVector vec(num_channels_); - AudioMultiVector* vec_copy = NULL; + AudioMultiVector* vec_copy = nullptr; vec.PushBackInterleaved(array_interleaved_); vec.CopyTo(vec_copy); } diff --git a/modules/audio_coding/neteq/decoder_database.cc b/modules/audio_coding/neteq/decoder_database.cc index 070a07b165..dbeba60171 100644 --- a/modules/audio_coding/neteq/decoder_database.cc +++ b/modules/audio_coding/neteq/decoder_database.cc @@ -188,7 +188,7 @@ const DecoderDatabase::DecoderInfo* DecoderDatabase::GetDecoderInfo( DecoderMap::const_iterator it = decoders_.find(rtp_payload_type); if (it == decoders_.end()) { // Decoder not found. - return NULL; + return nullptr; } return &it->second; } @@ -221,7 +221,7 @@ int DecoderDatabase::SetActiveDecoder(uint8_t rtp_payload_type, AudioDecoder* DecoderDatabase::GetActiveDecoder() const { if (active_decoder_type_ < 0) { // No active decoder. - return NULL; + return nullptr; } return GetDecoder(active_decoder_type_); } @@ -246,7 +246,7 @@ int DecoderDatabase::SetActiveCngDecoder(uint8_t rtp_payload_type) { ComfortNoiseDecoder* DecoderDatabase::GetActiveCngDecoder() const { if (active_cng_decoder_type_ < 0) { // No active CNG decoder. - return NULL; + return nullptr; } if (!active_cng_decoder_) { active_cng_decoder_.reset(new ComfortNoiseDecoder); diff --git a/modules/audio_coding/neteq/decoder_database_unittest.cc b/modules/audio_coding/neteq/decoder_database_unittest.cc index 4718f559af..6df9ab3892 100644 --- a/modules/audio_coding/neteq/decoder_database_unittest.cc +++ b/modules/audio_coding/neteq/decoder_database_unittest.cc @@ -88,12 +88,12 @@ TEST(DecoderDatabase, GetDecoderInfo) { db.RegisterPayload(kPayloadType, SdpAudioFormat(kCodecName, 8000, 1))); const DecoderDatabase::DecoderInfo* info; info = db.GetDecoderInfo(kPayloadType); - ASSERT_TRUE(info != NULL); + ASSERT_TRUE(info != nullptr); EXPECT_TRUE(info->IsType("pcmu")); EXPECT_EQ(kCodecName, info->get_name()); EXPECT_EQ(decoder, db.GetDecoder(kPayloadType)); info = db.GetDecoderInfo(kPayloadType + 1); // Other payload type. - EXPECT_TRUE(info == NULL); // Should not be found. + EXPECT_TRUE(info == nullptr); // Should not be found. } TEST(DecoderDatabase, GetDecoder) { @@ -103,7 +103,7 @@ TEST(DecoderDatabase, GetDecoder) { EXPECT_EQ(DecoderDatabase::kOK, db.RegisterPayload(kPayloadType, SdpAudioFormat("l16", 8000, 1))); AudioDecoder* dec = db.GetDecoder(kPayloadType); - ASSERT_TRUE(dec != NULL); + ASSERT_TRUE(dec != nullptr); } TEST(DecoderDatabase, TypeTests) { @@ -193,40 +193,40 @@ TEST(DecoderDatabase, IF_ISAC(ActiveDecoders)) { ASSERT_EQ(DecoderDatabase::kOK, db.RegisterPayload(13, SdpAudioFormat("cn", 8000, 1))); // Verify that no decoders are active from the start. - EXPECT_EQ(NULL, db.GetActiveDecoder()); - EXPECT_EQ(NULL, db.GetActiveCngDecoder()); + EXPECT_EQ(nullptr, db.GetActiveDecoder()); + EXPECT_EQ(nullptr, db.GetActiveCngDecoder()); // Set active speech codec. bool changed; // Should be true when the active decoder changed. EXPECT_EQ(DecoderDatabase::kOK, db.SetActiveDecoder(0, &changed)); EXPECT_TRUE(changed); AudioDecoder* decoder = db.GetActiveDecoder(); - ASSERT_FALSE(decoder == NULL); // Should get a decoder here. + ASSERT_FALSE(decoder == nullptr); // Should get a decoder here. // Set the same again. Expect no change. EXPECT_EQ(DecoderDatabase::kOK, db.SetActiveDecoder(0, &changed)); EXPECT_FALSE(changed); decoder = db.GetActiveDecoder(); - ASSERT_FALSE(decoder == NULL); // Should get a decoder here. + ASSERT_FALSE(decoder == nullptr); // Should get a decoder here. // Change active decoder. EXPECT_EQ(DecoderDatabase::kOK, db.SetActiveDecoder(103, &changed)); EXPECT_TRUE(changed); decoder = db.GetActiveDecoder(); - ASSERT_FALSE(decoder == NULL); // Should get a decoder here. + ASSERT_FALSE(decoder == nullptr); // Should get a decoder here. // Remove the active decoder, and verify that the active becomes NULL. EXPECT_EQ(DecoderDatabase::kOK, db.Remove(103)); - EXPECT_EQ(NULL, db.GetActiveDecoder()); + EXPECT_EQ(nullptr, db.GetActiveDecoder()); // Set active CNG codec. EXPECT_EQ(DecoderDatabase::kOK, db.SetActiveCngDecoder(13)); ComfortNoiseDecoder* cng = db.GetActiveCngDecoder(); - ASSERT_FALSE(cng == NULL); // Should get a decoder here. + ASSERT_FALSE(cng == nullptr); // Should get a decoder here. // Remove the active CNG decoder, and verify that the active becomes NULL. EXPECT_EQ(DecoderDatabase::kOK, db.Remove(13)); - EXPECT_EQ(NULL, db.GetActiveCngDecoder()); + EXPECT_EQ(nullptr, db.GetActiveCngDecoder()); // Try to set non-existing codecs as active. EXPECT_EQ(DecoderDatabase::kDecoderNotFound, diff --git a/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc b/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc index 625610e501..10e4c21309 100644 --- a/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc +++ b/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc @@ -176,7 +176,7 @@ TEST(DtmfToneGenerator, TestErrors) { EXPECT_TRUE(tone_gen.initialized()); // NULL pointer to destination. EXPECT_EQ(DtmfToneGenerator::kParameterError, - tone_gen.Generate(kNumSamples, NULL)); + tone_gen.Generate(kNumSamples, nullptr)); } } // namespace webrtc diff --git a/modules/audio_coding/neteq/expand_unittest.cc b/modules/audio_coding/neteq/expand_unittest.cc index cb0dcaaa88..6e2c28327c 100644 --- a/modules/audio_coding/neteq/expand_unittest.cc +++ b/modules/audio_coding/neteq/expand_unittest.cc @@ -50,7 +50,7 @@ TEST(Expand, CreateUsingFactory) { ExpandFactory expand_factory; Expand* expand = expand_factory.Create(&bgn, &sync_buffer, &random_vector, &statistics, fs, channels); - EXPECT_TRUE(expand != NULL); + EXPECT_TRUE(expand != nullptr); delete expand; } diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc index 22413620fd..999f1952ca 100644 --- a/modules/audio_coding/neteq/neteq_impl_unittest.cc +++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc @@ -156,13 +156,14 @@ class NetEqImplTest : public ::testing::Test { new TimestampScaler(*deps.decoder_database.get())); neteq_.reset(new NetEqImpl(config_, std::move(deps))); - ASSERT_TRUE(neteq_ != NULL); + ASSERT_TRUE(neteq_ != nullptr); } void CreateInstance() { CreateInstance(CreateBuiltinAudioDecoderFactory()); } void UseNoMocks() { - ASSERT_TRUE(neteq_ == NULL) << "Must call UseNoMocks before CreateInstance"; + ASSERT_TRUE(neteq_ == nullptr) + << "Must call UseNoMocks before CreateInstance"; use_mock_decoder_database_ = false; use_mock_neteq_controller_ = false; use_mock_dtmf_buffer_ = false; @@ -568,7 +569,7 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) { // Check the timestamp for the last value in the sync buffer. This should // be one full frame length ahead of the RTP timestamp. const SyncBuffer* sync_buffer = neteq_->sync_buffer_for_test(); - ASSERT_TRUE(sync_buffer != NULL); + ASSERT_TRUE(sync_buffer != nullptr); EXPECT_EQ(rtp_header.timestamp + kPayloadLengthSamples, sync_buffer->end_timestamp()); diff --git a/modules/audio_coding/neteq/packet_buffer_unittest.cc b/modules/audio_coding/neteq/packet_buffer_unittest.cc index 5713a6d01d..873ded51fc 100644 --- a/modules/audio_coding/neteq/packet_buffer_unittest.cc +++ b/modules/audio_coding/neteq/packet_buffer_unittest.cc @@ -391,7 +391,7 @@ TEST(PacketBuffer, Failures) { EXPECT_EQ(PacketBuffer::kBufferEmpty, buffer.NextTimestamp(&temp_ts)); EXPECT_EQ(PacketBuffer::kBufferEmpty, buffer.NextHigherTimestamp(0, &temp_ts)); - EXPECT_EQ(NULL, buffer.PeekNextPacket()); + EXPECT_EQ(nullptr, buffer.PeekNextPacket()); EXPECT_FALSE(buffer.GetNextPacket()); // Discarding packets will not invoke mock_stats.PacketDiscarded() because the diff --git a/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc b/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc index e7496a7c3b..276adc106e 100644 --- a/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc +++ b/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc @@ -82,8 +82,8 @@ NetEqOpusQualityTest::NetEqOpusQualityTest() kOpusSamplingKhz, kOpusSamplingKhz, SdpAudioFormat("opus", 48000, 2)), - opus_encoder_(NULL), - repacketizer_(NULL), + opus_encoder_(nullptr), + repacketizer_(nullptr), sub_block_size_samples_( static_cast(kOpusBlockDurationMs * kOpusSamplingKhz)), bit_rate_kbps_(absl::GetFlag(FLAGS_bit_rate_kbps)), diff --git a/modules/audio_coding/neteq/test/result_sink.cc b/modules/audio_coding/neteq/test/result_sink.cc index 72dfc318a2..5310ca8a86 100644 --- a/modules/audio_coding/neteq/test/result_sink.cc +++ b/modules/audio_coding/neteq/test/result_sink.cc @@ -69,7 +69,7 @@ ResultSink::ResultSink(absl::string_view output_file) : output_fp_(nullptr), digest_(MessageDigestFactory::Create(DIGEST_SHA_1)) { if (!output_file.empty()) { output_fp_ = fopen(std::string(output_file).c_str(), "wb"); - EXPECT_TRUE(output_fp_ != NULL); + EXPECT_TRUE(output_fp_ != nullptr); } } diff --git a/modules/audio_coding/neteq/time_stretch_unittest.cc b/modules/audio_coding/neteq/time_stretch_unittest.cc index e53bd49724..029ab887e9 100644 --- a/modules/audio_coding/neteq/time_stretch_unittest.cc +++ b/modules/audio_coding/neteq/time_stretch_unittest.cc @@ -48,13 +48,13 @@ TEST(TimeStretch, CreateUsingFactory) { AccelerateFactory accelerate_factory; Accelerate* accelerate = accelerate_factory.Create(kSampleRate, kNumChannels, bgn); - EXPECT_TRUE(accelerate != NULL); + EXPECT_TRUE(accelerate != nullptr); delete accelerate; PreemptiveExpandFactory preemptive_expand_factory; PreemptiveExpand* preemptive_expand = preemptive_expand_factory.Create( kSampleRate, kNumChannels, bgn, kOverlapSamples); - EXPECT_TRUE(preemptive_expand != NULL); + EXPECT_TRUE(preemptive_expand != nullptr); delete preemptive_expand; } diff --git a/modules/audio_coding/neteq/timestamp_scaler_unittest.cc b/modules/audio_coding/neteq/timestamp_scaler_unittest.cc index f5d6411ead..b32928e0de 100644 --- a/modules/audio_coding/neteq/timestamp_scaler_unittest.cc +++ b/modules/audio_coding/neteq/timestamp_scaler_unittest.cc @@ -332,7 +332,7 @@ TEST(TimestampScaler, Failures) { uint32_t timestamp = 4711; // Some number. EXPECT_EQ(timestamp, scaler.ToInternal(timestamp, kRtpPayloadType)); - Packet* packet = NULL; + Packet* packet = nullptr; scaler.ToInternal(packet); // Should not crash. That's all we can test. EXPECT_CALL(db, Die()); // Called when database object is deleted. diff --git a/modules/audio_coding/neteq/tools/rtp_file_source.cc b/modules/audio_coding/neteq/tools/rtp_file_source.cc index 437b560543..4a83cf70fa 100644 --- a/modules/audio_coding/neteq/tools/rtp_file_source.cc +++ b/modules/audio_coding/neteq/tools/rtp_file_source.cc @@ -57,7 +57,7 @@ std::unique_ptr RtpFileSource::NextPacket() { while (true) { RtpPacket temp_packet; if (!rtp_reader_->NextPacket(&temp_packet)) { - return NULL; + return nullptr; } if (temp_packet.original_length == 0) { // May be an RTCP packet. diff --git a/modules/audio_coding/neteq/tools/rtpcat.cc b/modules/audio_coding/neteq/tools/rtpcat.cc index 1c126ff720..9e91490512 100644 --- a/modules/audio_coding/neteq/tools/rtpcat.cc +++ b/modules/audio_coding/neteq/tools/rtpcat.cc @@ -29,13 +29,13 @@ int main(int argc, char* argv[]) { std::unique_ptr output( RtpFileWriter::Create(RtpFileWriter::kRtpDump, argv[argc - 1])); - RTC_CHECK(output.get() != NULL) << "Cannot open output file."; + RTC_CHECK(output.get() != nullptr) << "Cannot open output file."; printf("Output RTP file: %s\n", argv[argc - 1]); for (int i = 1; i < argc - 1; i++) { std::unique_ptr input( RtpFileReader::Create(RtpFileReader::kRtpDump, argv[i])); - RTC_CHECK(input.get() != NULL) << "Cannot open input file " << argv[i]; + RTC_CHECK(input.get() != nullptr) << "Cannot open input file " << argv[i]; printf("Input RTP file: %s\n", argv[i]); webrtc::test::RtpPacket packet; diff --git a/modules/audio_coding/test/Channel.cc b/modules/audio_coding/test/Channel.cc index b3be9101b1..5d3c2192d3 100644 --- a/modules/audio_coding/test/Channel.cc +++ b/modules/audio_coding/test/Channel.cc @@ -117,7 +117,7 @@ void Channel::CalcStatistics(const RTPHeader& rtp_header, size_t payloadSize) { _lastPayloadType = rtp_header.payloadType; bool newPayload = true; - ACMTestPayloadStats* currentPayloadStr = NULL; + ACMTestPayloadStats* currentPayloadStr = nullptr; for (n = 0; n < MAX_NUM_PAYLOADS; n++) { if (rtp_header.payloadType == _payloadStats[n].payloadType) { newPayload = false; @@ -196,9 +196,9 @@ void Channel::CalcStatistics(const RTPHeader& rtp_header, size_t payloadSize) { } Channel::Channel(int16_t chID) - : _neteq(NULL), + : _neteq(nullptr), _seqNo(0), - _bitStreamFile(NULL), + _bitStreamFile(nullptr), _saveBitStream(false), _lastPayloadType(-1), _isStereo(false), diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc index 1de3da55bb..ab298064d4 100644 --- a/modules/audio_coding/test/EncodeDecodeTest.cc +++ b/modules/audio_coding/test/EncodeDecodeTest.cc @@ -61,7 +61,7 @@ int32_t TestPacketization::SendData( } Sender::Sender() - : _acm(NULL), _pcmFile(), _audioFrame(), _packetization(NULL) {} + : _acm(nullptr), _pcmFile(), _audioFrame(), _packetization(nullptr) {} void Sender::Setup(const Environment& env, AudioCodingModule* acm, diff --git a/modules/audio_coding/test/PCMFile.cc b/modules/audio_coding/test/PCMFile.cc index 513703e7e5..b6c0d4c5cc 100644 --- a/modules/audio_coding/test/PCMFile.cc +++ b/modules/audio_coding/test/PCMFile.cc @@ -28,7 +28,7 @@ namespace webrtc { #define MAX_FILE_NAME_LENGTH_BYTE 500 PCMFile::PCMFile() - : pcm_file_(NULL), + : pcm_file_(nullptr), samples_10ms_(160), frequency_(16000), end_of_file_(false), @@ -41,7 +41,7 @@ PCMFile::PCMFile() } PCMFile::PCMFile(uint32_t timestamp) - : pcm_file_(NULL), + : pcm_file_(nullptr), samples_10ms_(160), frequency_(16000), end_of_file_(false), @@ -63,7 +63,7 @@ int16_t PCMFile::ChooseFile(std::string* file_name, uint16_t* frequency_hz) { char tmp_name[MAX_FILE_NAME_LENGTH_BYTE]; - EXPECT_TRUE(fgets(tmp_name, MAX_FILE_NAME_LENGTH_BYTE, stdin) != NULL); + EXPECT_TRUE(fgets(tmp_name, MAX_FILE_NAME_LENGTH_BYTE, stdin) != nullptr); tmp_name[MAX_FILE_NAME_LENGTH_BYTE - 1] = '\0'; int16_t n = 0; @@ -101,7 +101,7 @@ int16_t PCMFile::ChooseFile(std::string* file_name, } printf("Enter the sampling frequency (in Hz) of the above file [%u]: ", *frequency_hz); - EXPECT_TRUE(fgets(tmp_name, 10, stdin) != NULL); + EXPECT_TRUE(fgets(tmp_name, 10, stdin) != nullptr); uint16_t tmp_frequency = (uint16_t)atoi(tmp_name); if (tmp_frequency > 0) { *frequency_hz = tmp_frequency; @@ -114,7 +114,7 @@ void PCMFile::Open(absl::string_view file_name, absl::string_view mode, bool auto_rewind) { if ((pcm_file_ = fopen(std::string(file_name).c_str(), - std::string(mode).c_str())) == NULL) { + std::string(mode).c_str())) == nullptr) { printf("Cannot open file %s.\n", std::string(file_name).c_str()); ADD_FAILURE() << "Unable to read file"; } @@ -208,7 +208,7 @@ void PCMFile::Write10MsData(const int16_t* playout_buffer, void PCMFile::Close() { fclose(pcm_file_); - pcm_file_ = NULL; + pcm_file_ = nullptr; blocks_read_ = 0; } diff --git a/modules/audio_coding/test/RTPFile.cc b/modules/audio_coding/test/RTPFile.cc index 24e7f86e35..adb12fc630 100644 --- a/modules/audio_coding/test/RTPFile.cc +++ b/modules/audio_coding/test/RTPFile.cc @@ -134,7 +134,7 @@ bool RTPBuffer::EndOfFile() const { void RTPFile::Open(absl::string_view filename, absl::string_view mode) { std::string filename_str = std::string(filename); if ((_rtpFile = fopen(filename_str.c_str(), std::string(mode).c_str())) == - NULL) { + nullptr) { printf("Cannot write file %s.\n", filename_str.c_str()); ADD_FAILURE() << "Unable to write file"; exit(1); @@ -142,9 +142,9 @@ void RTPFile::Open(absl::string_view filename, absl::string_view mode) { } void RTPFile::Close() { - if (_rtpFile != NULL) { + if (_rtpFile != nullptr) { fclose(_rtpFile); - _rtpFile = NULL; + _rtpFile = nullptr; } } @@ -165,7 +165,7 @@ void RTPFile::ReadHeader() { uint32_t start_sec, start_usec, source; uint16_t port, padding; char fileHeader[40]; - EXPECT_TRUE(fgets(fileHeader, 40, _rtpFile) != 0); + EXPECT_TRUE(fgets(fileHeader, 40, _rtpFile) != nullptr); EXPECT_EQ(1u, fread(&start_sec, 4, 1, _rtpFile)); start_sec = ntohl(start_sec); EXPECT_EQ(1u, fread(&start_usec, 4, 1, _rtpFile)); diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc index 786ae2b552..cbf04a43d0 100644 --- a/modules/audio_coding/test/TestAllCodecs.cc +++ b/modules/audio_coding/test/TestAllCodecs.cc @@ -56,7 +56,7 @@ namespace webrtc { // Class for simulating packet handling. TestPack::TestPack() - : neteq_(NULL), + : neteq_(nullptr), sequence_number_(0), timestamp_diff_(0), last_in_timestamp_(0), @@ -121,15 +121,15 @@ TestAllCodecs::TestAllCodecs() neteq_(DefaultNetEqFactory().Create(env_, NetEq::Config(), CreateBuiltinAudioDecoderFactory())), - channel_a_to_b_(NULL), + channel_a_to_b_(nullptr), test_count_(0), packet_size_samples_(0), packet_size_bytes_(0) {} TestAllCodecs::~TestAllCodecs() { - if (channel_a_to_b_ != NULL) { + if (channel_a_to_b_ != nullptr) { delete channel_a_to_b_; - channel_a_to_b_ = NULL; + channel_a_to_b_ = nullptr; } } diff --git a/modules/audio_coding/test/TestRedFec.cc b/modules/audio_coding/test/TestRedFec.cc index 7d93fee2d4..965eb69506 100644 --- a/modules/audio_coding/test/TestRedFec.cc +++ b/modules/audio_coding/test/TestRedFec.cc @@ -60,13 +60,13 @@ TestRedFec::TestRedFec() _neteq(DefaultNetEqFactory().Create(env_, NetEq::Config(), decoder_factory_)), - _channelA2B(NULL), + _channelA2B(nullptr), _testCntr(0) {} TestRedFec::~TestRedFec() { - if (_channelA2B != NULL) { + if (_channelA2B != nullptr) { delete _channelA2B; - _channelA2B = NULL; + _channelA2B = nullptr; } } diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc index 8625276cc2..78b0d8ca74 100644 --- a/modules/audio_coding/test/TestStereo.cc +++ b/modules/audio_coding/test/TestStereo.cc @@ -42,7 +42,7 @@ namespace webrtc { // Class for simulating packet handling TestPackStereo::TestPackStereo() - : neteq_(NULL), + : neteq_(nullptr), seq_no_(0), timestamp_diff_(0), last_in_timestamp_(0), @@ -120,16 +120,16 @@ TestStereo::TestStereo() neteq_(DefaultNetEqFactory().Create(env_, NetEq::Config(), CreateBuiltinAudioDecoderFactory())), - channel_a2b_(NULL), + channel_a2b_(nullptr), test_cntr_(0), pack_size_samp_(0), pack_size_bytes_(0), counter_(0) {} TestStereo::~TestStereo() { - if (channel_a2b_ != NULL) { + if (channel_a2b_ != nullptr) { delete channel_a2b_; - channel_a2b_ = NULL; + channel_a2b_ = nullptr; } } @@ -152,7 +152,7 @@ void TestStereo::Perform() { in_file_mono_->ReadStereo(false); // Create and initialize two ACMs, one for each side of a one-to-one call. - ASSERT_TRUE((acm_a_.get() != NULL) && (neteq_.get() != NULL)); + ASSERT_TRUE((acm_a_.get() != nullptr) && (neteq_.get() != nullptr)); neteq_->FlushBuffers(); neteq_->SetCodecs({{103, {"ISAC", 16000, 1}}, @@ -474,7 +474,7 @@ void TestStereo::RegisterSendCodec(char side, 0.875); // Set pointer to the ACM where to register the codec - AudioCodingModule* my_acm = NULL; + AudioCodingModule* my_acm = nullptr; switch (side) { case 'A': { my_acm = acm_a_.get(); @@ -489,7 +489,7 @@ void TestStereo::RegisterSendCodec(char side, default: break; } - ASSERT_TRUE(my_acm != NULL); + ASSERT_TRUE(my_acm != nullptr); auto encoder_factory = CreateBuiltinAudioEncoderFactory(); const int clockrate_hz = absl::EqualsIgnoreCase(codec_name, "g722") diff --git a/modules/audio_coding/test/opus_test.cc b/modules/audio_coding/test/opus_test.cc index e908a807ed..1f31e3c2a9 100644 --- a/modules/audio_coding/test/opus_test.cc +++ b/modules/audio_coding/test/opus_test.cc @@ -35,31 +35,31 @@ OpusTest::OpusTest() : neteq_(DefaultNetEqFactory().Create(CreateEnvironment(), NetEq::Config(), CreateBuiltinAudioDecoderFactory())), - channel_a2b_(NULL), + channel_a2b_(nullptr), counter_(0), payload_type_(255), rtp_timestamp_(0) {} OpusTest::~OpusTest() { - if (channel_a2b_ != NULL) { + if (channel_a2b_ != nullptr) { delete channel_a2b_; - channel_a2b_ = NULL; + channel_a2b_ = nullptr; } - if (opus_mono_encoder_ != NULL) { + if (opus_mono_encoder_ != nullptr) { WebRtcOpus_EncoderFree(opus_mono_encoder_); - opus_mono_encoder_ = NULL; + opus_mono_encoder_ = nullptr; } - if (opus_stereo_encoder_ != NULL) { + if (opus_stereo_encoder_ != nullptr) { WebRtcOpus_EncoderFree(opus_stereo_encoder_); - opus_stereo_encoder_ = NULL; + opus_stereo_encoder_ = nullptr; } - if (opus_mono_decoder_ != NULL) { + if (opus_mono_decoder_ != nullptr) { WebRtcOpus_DecoderFree(opus_mono_decoder_); - opus_mono_decoder_ = NULL; + opus_mono_decoder_ = nullptr; } - if (opus_stereo_decoder_ != NULL) { + if (opus_stereo_decoder_ != nullptr) { WebRtcOpus_DecoderFree(opus_stereo_decoder_); - opus_stereo_decoder_ = NULL; + opus_stereo_decoder_ = nullptr; } } @@ -93,7 +93,7 @@ void OpusTest::Perform() { WebRtcOpus_DecoderInit(opus_mono_decoder_); WebRtcOpus_DecoderInit(opus_stereo_decoder_); - ASSERT_TRUE(neteq_.get() != NULL); + ASSERT_TRUE(neteq_.get() != nullptr); neteq_->FlushBuffers(); // Register Opus stereo as receiving codec. @@ -316,7 +316,7 @@ void OpusTest::Run(TestPackStereo* channel, size_t total_plc_samples = 0; while (total_plc_samples < frame_length) { int ret = WebRtcOpus_Decode( - opus_mono_decoder_, NULL, 0, + opus_mono_decoder_, nullptr, 0, &out_audio[decoded_samples * channels], &audio_type); EXPECT_EQ(ret, kPlcSamples); decoded_samples += ret; @@ -336,7 +336,7 @@ void OpusTest::Run(TestPackStereo* channel, size_t total_plc_samples = 0; while (total_plc_samples < frame_length) { int ret = WebRtcOpus_Decode( - opus_stereo_decoder_, NULL, 0, + opus_stereo_decoder_, nullptr, 0, &out_audio[decoded_samples * channels], &audio_type); EXPECT_EQ(ret, kPlcSamples); decoded_samples += ret; From 71f23e648d707a857371df33b669be329eb6a52e Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Thu, 15 May 2025 11:51:42 +0000 Subject: [PATCH 169/292] [cleanup] Replace uses of NULL with nullptr Generated using clang-tidy, tools/clang/scripts/build_clang_tools_extra.py \ --fetch out/Default clang-tidy clang-apply-replacements ninja -C out/Default gn gen out/Default --export-compile-commands cd out/Default tools/clang/third_party/llvm/clang-tools-extra/clang-tidy/tool/run-clang-tidy.py -p . \ -clang-tidy-binary out/Default/tools/clang/third_party/llvm/build/bin/clang-tidy \ -clang-apply-replacements-binary \ out/Default/tools/clang/third_party/llvm/build/bin/clang-apply-replacements \ -checks='-*,modernize-use-nullptr' \ -fix This CL was uploaded by git cl split. Bug: webrtc:42232595 Change-Id: I781fa6e9bbe20ddef584d9a8d654a79594e5b8e2 No-Iwyu: LSC Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391803 Commit-Queue: Fanny Linderborg Auto-Submit: Evan Shrubsole Reviewed-by: Fanny Linderborg Cr-Commit-Position: refs/heads/main@{#44641} --- .../goog_cc/delay_based_bwe_unittest_helper.cc | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc b/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc index c93bf29c51..9e22ae8682 100644 --- a/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc +++ b/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc @@ -58,7 +58,7 @@ int64_t RtpStream::GenerateFrame(int64_t time_now_us, if (time_now_us < next_rtp_time_) { return next_rtp_time_; } - RTC_CHECK(packets != NULL); + RTC_CHECK(packets != nullptr); size_t bits_per_frame = (bitrate_bps_ + fps_ / 2) / fps_; size_t n_packets = std::max((bits_per_frame + 4 * kMtu) / (8 * kMtu), 1u); @@ -137,7 +137,7 @@ void StreamGenerator::SetBitrateBps(int bitrate_bps) { int64_t StreamGenerator::GenerateFrame(int64_t time_now_us, int64_t* next_sequence_number, std::vector* packets) { - RTC_CHECK(packets != NULL); + RTC_CHECK(packets != nullptr); RTC_CHECK(packets->empty()); RTC_CHECK_GT(capacity_, 0); auto it = From 35d053440b7ea9e9f1cbbe1816c8a185a44bafe1 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Thu, 15 May 2025 05:02:36 -0700 Subject: [PATCH 170/292] Roll chromium_revision 35e96510ff..66a9943b59 (1460497:1460639) Change log: https://chromium.googlesource.com/chromium/src/+log/35e96510ff..66a9943b59 Full diff: https://chromium.googlesource.com/chromium/src/+/35e96510ff..66a9943b59 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/53335694d5..a4805d687b * src/build: https://chromium.googlesource.com/chromium/src/build/+log/289b4d8c67..a4e74b4312 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/e1277e5176..e24185bdfb * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/e7241882c3..11317b92ec * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/c1d0b85060..faf790f4c1 * src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/d33b54c621..647003beca * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/c273e3eb62..f810dae272 * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/3792c9842a..828b2679ea * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/6690560f58..77388f6009 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/18608d93c9..e9e3e0175e DEPS diff: https://chromium.googlesource.com/chromium/src/+/35e96510ff..66a9943b59/DEPS No update to Clang. BUG=None Change-Id: I325bec4d0c13b6ea23f85d2522cb320e15f03410 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391880 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44642} --- DEPS | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/DEPS b/DEPS index 0570ecd00e..6c6a7f0f3c 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '35e96510ff321fc81e1dee2e666be935fee27fb3', + 'chromium_revision': '66a9943b59a237f7184cce1a9b3e45218c19da49', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@53335694d5aa1a2016df233a7ad30ad668d47a2a', + 'https://chromium.googlesource.com/chromium/src/base@a4805d687b1782f8ca734c851fd7b7c4f2f9822f', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@289b4d8c67486acccce147ae049e3582b2e63ae3', + 'https://chromium.googlesource.com/chromium/src/build@a4e74b4312dafbbb000f24000ab3b57120a5e18d', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@e9b36e9ffb02b268d194c3f82f89952d09d0c714', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@e1277e5176140de22a409f34c66e82a7d6493dce', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@e24185bdfb0339d724f676aa7c8fe50efd49f2d4', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@e7241882c3beb9d0f1e69a84a6c16e7f728db721', + 'https://chromium.googlesource.com/chromium/src/testing@11317b92ec4abfe891ff12e5583256aab2a68758', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@c1d0b85060486cde0fa11e0eab490f67d6bbd5d8', + 'https://chromium.googlesource.com/chromium/src/third_party@faf790f4c169c146a68006e1c2d69262faa2d52e', 'src/buildtools/linux64': { 'packages': [ @@ -323,7 +323,7 @@ deps = { 'src/third_party/clang-format/script': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@37f6e68a107df43b7d7e044fd36a13cbae3413f2', 'src/third_party/compiler-rt/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@d33b54c621474e2f2642af28d42a15227b1602cb', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@647003becabf611bc63c0e959e84060376d371be', 'src/third_party/libc++/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@a01c02c9d4acbdae3b7e8a2f3ee58579a9c29f96', 'src/third_party/libc++abi/src': @@ -497,13 +497,13 @@ deps = { 'src/third_party/crc32c/src': 'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558', 'src/third_party/depot_tools': - 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@c273e3eb6202ef710a9cbf48fc99437a56a13631', + 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@f810dae27200e4209bd0cba85bd2266ec0d55312', 'src/third_party/ffmpeg': 'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@dcdd0fa51b65a0b1688ff6b8f0cc81908f09ded2', 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@3792c9842ab0a149237174633aacf1aed3d50575', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@828b2679ea647472c0ddad9a11140f121120b73a', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '6690560f58167a82a9302a6df05303324f0e91bb', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '77388f6009c785fd1c2cfe8e3a3969e52967dcf3', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@18608d93c9e76e96e6109a3979d7f507b212bf8b', + 'https://chromium.googlesource.com/chromium/src/tools@e9e3e0175e8962443592097100fd74f52e244c74', 'src/third_party/espresso': { 'packages': [ From f5318fdea8afd5c4f0d80346f65bd8d340c7bce8 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Thu, 15 May 2025 11:52:37 +0000 Subject: [PATCH 171/292] [cleanup] Replace uses of NULL with nullptr Generated using clang-tidy, tools/clang/scripts/build_clang_tools_extra.py \ --fetch out/Default clang-tidy clang-apply-replacements ninja -C out/Default gn gen out/Default --export-compile-commands cd out/Default tools/clang/third_party/llvm/clang-tools-extra/clang-tidy/tool/run-clang-tidy.py -p . \ -clang-tidy-binary out/Default/tools/clang/third_party/llvm/build/bin/clang-tidy \ -clang-apply-replacements-binary \ out/Default/tools/clang/third_party/llvm/build/bin/clang-apply-replacements \ -checks='-*,modernize-use-nullptr' \ -fix This CL was uploaded by git cl split. Bug: webrtc:42232595 Change-Id: Ic8b00e28b090427ce7bd3eb1cb85fd05916147c0 No-Iwyu: LSC Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391643 Auto-Submit: Evan Shrubsole Reviewed-by: Philip Eliasson Commit-Queue: Philip Eliasson Cr-Commit-Position: refs/heads/main@{#44643} --- .../codecs/vp8/libvpx_vp8_decoder.cc | 22 +++---- .../codecs/vp8/libvpx_vp8_encoder.cc | 18 ++--- .../codecs/vp9/libvpx_vp9_decoder.cc | 2 +- .../video_coding/deprecated/frame_buffer.cc | 4 +- .../video_coding/deprecated/jitter_buffer.cc | 20 +++--- .../deprecated/jitter_buffer_unittest.cc | 66 +++++++++---------- modules/video_coding/deprecated/packet.cc | 2 +- modules/video_coding/deprecated/receiver.cc | 8 +-- .../video_coding/deprecated/session_info.cc | 8 +-- .../deprecated/stream_generator.cc | 2 +- modules/video_coding/generic_decoder.cc | 2 +- .../video_coding/packet_buffer_unittest.cc | 4 +- .../utility/simulcast_test_fixture_impl.cc | 30 ++++----- 13 files changed, 94 insertions(+), 94 deletions(-) diff --git a/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc b/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc index 8bb121ff0d..741c4628d2 100644 --- a/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc +++ b/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc @@ -123,9 +123,9 @@ LibvpxVp8Decoder::LibvpxVp8Decoder(const Environment& env) kIsArm ? env.field_trials().IsEnabled(kVp8PostProcArmFieldTrial) : true), buffer_pool_(false, 300 /* max_number_of_buffers*/), - decode_complete_callback_(NULL), + decode_complete_callback_(nullptr), inited_(false), - decoder_(NULL), + decoder_(nullptr), last_frame_width_(0), last_frame_height_(0), key_frame_required_(true), @@ -143,7 +143,7 @@ bool LibvpxVp8Decoder::Configure(const Settings& settings) { if (Release() < 0) { return false; } - if (decoder_ == NULL) { + if (decoder_ == nullptr) { decoder_ = new vpx_codec_ctx_t; memset(decoder_, 0, sizeof(*decoder_)); } @@ -183,10 +183,10 @@ int LibvpxVp8Decoder::Decode(const EncodedImage& input_image, if (!inited_) { return WEBRTC_VIDEO_CODEC_UNINITIALIZED; } - if (decode_complete_callback_ == NULL) { + if (decode_complete_callback_ == nullptr) { return WEBRTC_VIDEO_CODEC_UNINITIALIZED; } - if (input_image.data() == NULL && input_image.size() > 0) { + if (input_image.data() == nullptr && input_image.size() > 0) { return WEBRTC_VIDEO_CODEC_ERR_PARAMETER; } @@ -242,14 +242,14 @@ int LibvpxVp8Decoder::Decode(const EncodedImage& input_image, const uint8_t* buffer = input_image.data(); if (input_image.size() == 0) { - buffer = NULL; // Triggers full frame concealment. + buffer = nullptr; // Triggers full frame concealment. } - if (vpx_codec_decode(decoder_, buffer, input_image.size(), 0, + if (vpx_codec_decode(decoder_, buffer, input_image.size(), nullptr, kDecodeDeadlineRealtime)) { return WEBRTC_VIDEO_CODEC_ERROR; } - vpx_codec_iter_t iter = NULL; + vpx_codec_iter_t iter = nullptr; vpx_image_t* img = vpx_codec_get_frame(decoder_, &iter); int qp; vpx_codec_err_t vpx_ret = @@ -267,7 +267,7 @@ int LibvpxVp8Decoder::ReturnFrame(const vpx_image_t* img, uint32_t timestamp, int qp, const ColorSpace* explicit_color_space) { - if (img == NULL) { + if (img == nullptr) { // Decoder OK and NULL image => No show frame return WEBRTC_VIDEO_CODEC_NO_OUTPUT; } @@ -322,14 +322,14 @@ int LibvpxVp8Decoder::RegisterDecodeCompleteCallback( int LibvpxVp8Decoder::Release() { int ret_val = WEBRTC_VIDEO_CODEC_OK; - if (decoder_ != NULL) { + if (decoder_ != nullptr) { if (inited_) { if (vpx_codec_destroy(decoder_)) { ret_val = WEBRTC_VIDEO_CODEC_MEMORY; } } delete decoder_; - decoder_ = NULL; + decoder_ = nullptr; } buffer_pool_.Release(); inited_ = false; diff --git a/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc b/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc index 8730a47ffc..4557a4fab1 100644 --- a/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc +++ b/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc @@ -499,7 +499,7 @@ void LibvpxVp8Encoder::SetFecControllerOverride( // TODO(eladalon): s/inst/codec_settings/g. int LibvpxVp8Encoder::InitEncode(const VideoCodec* inst, const VideoEncoder::Settings& settings) { - if (inst == NULL) { + if (inst == nullptr) { return WEBRTC_VIDEO_CODEC_ERR_PARAMETER; } if (inst->maxFramerate < 1) { @@ -696,7 +696,7 @@ int LibvpxVp8Encoder::InitEncode(const VideoCodec* inst, // Actual pointer will be set in encode. Setting align to 1, as it // is meaningless (no memory allocation is done here). libvpx_->img_wrap(&raw_images_[0], pixel_format, inst->width, inst->height, 1, - NULL); + nullptr); // Note the order we use is different from webm, we have lowest resolution // at position 0 and they have highest resolution at position 0. @@ -1020,7 +1020,7 @@ int LibvpxVp8Encoder::Encode(const VideoFrame& frame, if (!inited_) return WEBRTC_VIDEO_CODEC_UNINITIALIZED; - if (encoded_complete_callback_ == NULL) + if (encoded_complete_callback_ == nullptr) return WEBRTC_VIDEO_CODEC_UNINITIALIZED; bool key_frame_requested = false; @@ -1236,15 +1236,15 @@ int LibvpxVp8Encoder::GetEncodedPartitions(const VideoFrame& input_image, int result = WEBRTC_VIDEO_CODEC_OK; for (size_t encoder_idx = 0; encoder_idx < encoders_.size(); ++encoder_idx, --stream_idx) { - vpx_codec_iter_t iter = NULL; + vpx_codec_iter_t iter = nullptr; encoded_images_[encoder_idx].set_size(0); encoded_images_[encoder_idx]._frameType = VideoFrameType::kVideoFrameDelta; CodecSpecificInfo codec_specific; - const vpx_codec_cx_pkt_t* pkt = NULL; + const vpx_codec_cx_pkt_t* pkt = nullptr; size_t encoded_size = 0; while ((pkt = libvpx_->codec_get_cx_data(&encoders_[encoder_idx], &iter)) != - NULL) { + nullptr) { if (pkt->kind == VPX_CODEC_CX_FRAME_PKT) { encoded_size += pkt->data.frame.sz; } @@ -1252,10 +1252,10 @@ int LibvpxVp8Encoder::GetEncodedPartitions(const VideoFrame& input_image, auto buffer = EncodedImageBuffer::Create(encoded_size); - iter = NULL; + iter = nullptr; size_t encoded_pos = 0; while ((pkt = libvpx_->codec_get_cx_data(&encoders_[encoder_idx], &iter)) != - NULL) { + nullptr) { switch (pkt->kind) { case VPX_CODEC_CX_FRAME_PKT: { RTC_CHECK_LE(encoded_pos + pkt->data.frame.sz, buffer->size()); @@ -1435,7 +1435,7 @@ void LibvpxVp8Encoder::MaybeUpdatePixelFormat(vpx_img_fmt fmt) { libvpx_->img_free(&img); // First image is wrapping the input frame, the rest are allocated. if (i == 0) { - libvpx_->img_wrap(&img, fmt, d_w, d_h, 1, NULL); + libvpx_->img_wrap(&img, fmt, d_w, d_h, 1, nullptr); } else { libvpx_->img_alloc(&img, fmt, d_w, d_h, kVp832ByteAlign); } diff --git a/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc b/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc index ee9b5e54f3..e6d9f96a8f 100644 --- a/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc +++ b/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc @@ -245,7 +245,7 @@ int LibvpxVp9Decoder::Decode(const EncodedImage& input_image, // `libvpx_buffer_pool_`. In practice libvpx keeps a few (~3-4) buffers alive // at a time. if (vpx_codec_decode(decoder_, buffer, - static_cast(input_image.size()), 0, + static_cast(input_image.size()), nullptr, VPX_DL_REALTIME)) { return WEBRTC_VIDEO_CODEC_ERROR; } diff --git a/modules/video_coding/deprecated/frame_buffer.cc b/modules/video_coding/deprecated/frame_buffer.cc index 5b1af328a5..0b7b2a9b7f 100644 --- a/modules/video_coding/deprecated/frame_buffer.cc +++ b/modules/video_coding/deprecated/frame_buffer.cc @@ -82,8 +82,8 @@ VCMFrameBufferEnum VCMFrameBuffer::InsertPacket(const VCMPacket& packet, int64_t timeInMs, const FrameData& frame_data) { TRACE_EVENT0("webrtc", "VCMFrameBuffer::InsertPacket"); - RTC_DCHECK(!(NULL == packet.dataPtr && packet.sizeBytes > 0)); - if (packet.dataPtr != NULL) { + RTC_DCHECK(!(nullptr == packet.dataPtr && packet.sizeBytes > 0)); + if (packet.dataPtr != nullptr) { _payloadType = packet.payloadType; } diff --git a/modules/video_coding/deprecated/jitter_buffer.cc b/modules/video_coding/deprecated/jitter_buffer.cc index 2e6bd97a9b..dea2437a69 100644 --- a/modules/video_coding/deprecated/jitter_buffer.cc +++ b/modules/video_coding/deprecated/jitter_buffer.cc @@ -56,7 +56,7 @@ void FrameList::InsertFrame(VCMFrameBuffer* frame) { VCMFrameBuffer* FrameList::PopFrame(uint32_t timestamp) { FrameList::iterator it = find(timestamp); if (it == end()) - return NULL; + return nullptr; VCMFrameBuffer* frame = it->second; erase(it); return frame; @@ -271,7 +271,7 @@ VCMEncodedFrame* VCMJitterBuffer::NextCompleteFrame(uint32_t max_wait_time_ms) { VCMEncodedFrame* VCMJitterBuffer::ExtractAndSetDecode(uint32_t timestamp) { MutexLock lock(&mutex_); if (!running_) { - return NULL; + return nullptr; } // Extract the frame with the desired timestamp. VCMFrameBuffer* frame = decodable_frames_.PopFrame(timestamp); @@ -281,7 +281,7 @@ VCMEncodedFrame* VCMJitterBuffer::ExtractAndSetDecode(uint32_t timestamp) { if (frame) continuous = last_decoded_state_.ContinuousFrame(frame); else - return NULL; + return nullptr; } // Frame pulled out from jitter buffer, update the jitter estimate. const bool retransmitted = (frame->GetNackCount() > 0); @@ -330,20 +330,20 @@ VCMFrameBufferEnum VCMJitterBuffer::GetFrame(const VCMPacket& packet, VCMFrameBuffer** frame, FrameList** frame_list) { *frame = incomplete_frames_.PopFrame(packet.timestamp); - if (*frame != NULL) { + if (*frame != nullptr) { *frame_list = &incomplete_frames_; return kNoError; } *frame = decodable_frames_.PopFrame(packet.timestamp); - if (*frame != NULL) { + if (*frame != nullptr) { *frame_list = &decodable_frames_; return kNoError; } - *frame_list = NULL; + *frame_list = nullptr; // No match, return empty frame. *frame = GetEmptyFrame(); - if (*frame == NULL) { + if (*frame == nullptr) { // No free frame! Try to reclaim some... RTC_LOG(LS_WARNING) << "Unable to get empty frame; Recycling."; bool found_key_frame = RecycleFramesUntilKeyFrame(); @@ -499,7 +499,7 @@ VCMFrameBufferEnum VCMJitterBuffer::InsertPacket(const VCMPacket& packet, case kOutOfBoundsPacket: case kDuplicatePacket: { // Put back the frame where it came from. - if (frame_list != NULL) { + if (frame_list != nullptr) { frame_list->InsertFrame(frame); } else { RecycleFrameBuffer(frame); @@ -686,7 +686,7 @@ VCMFrameBuffer* VCMJitterBuffer::NextFrame() const { return decodable_frames_.Front(); if (!incomplete_frames_.empty()) return incomplete_frames_.Front(); - return NULL; + return nullptr; } bool VCMJitterBuffer::UpdateNackList(uint16_t sequence_number) { @@ -772,7 +772,7 @@ void VCMJitterBuffer::DropPacketsFromNackList( VCMFrameBuffer* VCMJitterBuffer::GetEmptyFrame() { if (free_frames_.empty()) { if (!TryToIncreaseJitterBufferSize()) { - return NULL; + return nullptr; } } VCMFrameBuffer* frame = free_frames_.front(); diff --git a/modules/video_coding/deprecated/jitter_buffer_unittest.cc b/modules/video_coding/deprecated/jitter_buffer_unittest.cc index cc0eaacec7..5c7c265811 100644 --- a/modules/video_coding/deprecated/jitter_buffer_unittest.cc +++ b/modules/video_coding/deprecated/jitter_buffer_unittest.cc @@ -212,7 +212,7 @@ class TestRunningJitterBuffer : public ::testing::Test { VCMEncodedFrame* frame = jitter_buffer_->ExtractAndSetDecode(found_frame->RtpTimestamp()); - bool ret = (frame != NULL); + bool ret = (frame != nullptr); jitter_buffer_->ReleaseFrame(frame); return ret; } @@ -236,11 +236,11 @@ class TestJitterBufferNack : public TestRunningJitterBuffer { TEST_F(TestBasicJitterBuffer, StopRunning) { jitter_buffer_->Stop(); - EXPECT_TRUE(NULL == DecodeCompleteFrame()); + EXPECT_TRUE(nullptr == DecodeCompleteFrame()); jitter_buffer_->Start(); // No packets inserted. - EXPECT_TRUE(NULL == DecodeCompleteFrame()); + EXPECT_TRUE(nullptr == DecodeCompleteFrame()); } TEST_F(TestBasicJitterBuffer, SinglePacketFrame) { @@ -270,7 +270,7 @@ TEST_F(TestBasicJitterBuffer, DualPacketFrame) { jitter_buffer_->InsertPacket(*packet_, &retransmitted)); VCMEncodedFrame* frame_out = DecodeCompleteFrame(); // Should not be complete. - EXPECT_TRUE(frame_out == NULL); + EXPECT_TRUE(frame_out == nullptr); ++seq_num_; packet_->video_header.is_first_packet_in_frame = false; @@ -299,7 +299,7 @@ TEST_F(TestBasicJitterBuffer, 100PacketKeyFrame) { VCMEncodedFrame* frame_out = DecodeCompleteFrame(); // Frame should not be complete. - EXPECT_TRUE(frame_out == NULL); + EXPECT_TRUE(frame_out == nullptr); // Insert 98 frames. int loop = 0; @@ -340,7 +340,7 @@ TEST_F(TestBasicJitterBuffer, 100PacketDeltaFrame) { EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &retransmitted)); VCMEncodedFrame* frame_out = DecodeCompleteFrame(); - EXPECT_FALSE(frame_out == NULL); + EXPECT_FALSE(frame_out == nullptr); jitter_buffer_->ReleaseFrame(frame_out); ++seq_num_; @@ -355,7 +355,7 @@ TEST_F(TestBasicJitterBuffer, 100PacketDeltaFrame) { frame_out = DecodeCompleteFrame(); // Frame should not be complete. - EXPECT_TRUE(frame_out == NULL); + EXPECT_TRUE(frame_out == nullptr); packet_->video_header.is_first_packet_in_frame = false; // Insert 98 frames. @@ -401,7 +401,7 @@ TEST_F(TestBasicJitterBuffer, PacketReorderingReverseOrder) { VCMEncodedFrame* frame_out = DecodeCompleteFrame(); - EXPECT_TRUE(frame_out == NULL); + EXPECT_TRUE(frame_out == nullptr); // Insert 98 packets. int loop = 0; @@ -444,7 +444,7 @@ TEST_F(TestBasicJitterBuffer, FrameReordering2Frames2PacketsEach) { VCMEncodedFrame* frame_out = DecodeCompleteFrame(); - EXPECT_TRUE(frame_out == NULL); + EXPECT_TRUE(frame_out == nullptr); seq_num_++; packet_->video_header.is_first_packet_in_frame = false; @@ -456,7 +456,7 @@ TEST_F(TestBasicJitterBuffer, FrameReordering2Frames2PacketsEach) { // check that we fail to get frame since seqnum is not continuous frame_out = DecodeCompleteFrame(); - EXPECT_TRUE(frame_out == NULL); + EXPECT_TRUE(frame_out == nullptr); seq_num_ -= 3; timestamp_ -= 33 * 90; @@ -472,7 +472,7 @@ TEST_F(TestBasicJitterBuffer, FrameReordering2Frames2PacketsEach) { frame_out = DecodeCompleteFrame(); // It should not be complete. - EXPECT_TRUE(frame_out == NULL); + EXPECT_TRUE(frame_out == nullptr); seq_num_++; packet_->video_header.is_first_packet_in_frame = false; @@ -504,7 +504,7 @@ TEST_F(TestBasicJitterBuffer, TestReorderingWithPadding) { EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &retransmitted)); VCMEncodedFrame* frame_out = DecodeCompleteFrame(); - EXPECT_TRUE(frame_out != NULL); + EXPECT_TRUE(frame_out != nullptr); jitter_buffer_->ReleaseFrame(frame_out); // Now send in a complete delta frame (Frame C), but with a sequence number @@ -519,7 +519,7 @@ TEST_F(TestBasicJitterBuffer, TestReorderingWithPadding) { EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &retransmitted)); frame_out = DecodeCompleteFrame(); - EXPECT_TRUE(frame_out == NULL); + EXPECT_TRUE(frame_out == nullptr); // Now send in a complete delta frame (Frame B) that is continuous from A, but // doesn't fill the full gap to C. The rest of the gap is going to be padding. @@ -529,12 +529,12 @@ TEST_F(TestBasicJitterBuffer, TestReorderingWithPadding) { EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &retransmitted)); frame_out = DecodeCompleteFrame(); - EXPECT_TRUE(frame_out != NULL); + EXPECT_TRUE(frame_out != nullptr); jitter_buffer_->ReleaseFrame(frame_out); // But Frame C isn't continuous yet. frame_out = DecodeCompleteFrame(); - EXPECT_TRUE(frame_out == NULL); + EXPECT_TRUE(frame_out == nullptr); // Add in the padding. These are empty packets (data length is 0) with no // marker bit and matching the timestamp of Frame B. @@ -555,7 +555,7 @@ TEST_F(TestBasicJitterBuffer, TestReorderingWithPadding) { // But now Frame C should be ready! frame_out = DecodeCompleteFrame(); - EXPECT_TRUE(frame_out != NULL); + EXPECT_TRUE(frame_out != nullptr); jitter_buffer_->ReleaseFrame(frame_out); } @@ -574,7 +574,7 @@ TEST_F(TestBasicJitterBuffer, DuplicatePackets) { VCMEncodedFrame* frame_out = DecodeCompleteFrame(); - EXPECT_TRUE(frame_out == NULL); + EXPECT_TRUE(frame_out == nullptr); EXPECT_EQ(1, jitter_buffer_->num_packets()); EXPECT_EQ(0, jitter_buffer_->num_duplicated_packets()); @@ -593,7 +593,7 @@ TEST_F(TestBasicJitterBuffer, DuplicatePackets) { jitter_buffer_->InsertPacket(*packet_, &retransmitted)); frame_out = DecodeCompleteFrame(); - ASSERT_TRUE(frame_out != NULL); + ASSERT_TRUE(frame_out != nullptr); CheckOutFrame(frame_out, 2 * size_, false); EXPECT_EQ(VideoFrameType::kVideoFrameKey, frame_out->FrameType()); @@ -617,7 +617,7 @@ TEST_F(TestBasicJitterBuffer, DuplicatePreviousDeltaFramePacket) { jitter_buffer_->InsertPacket(*packet_, &retransmitted)); VCMEncodedFrame* frame_out = DecodeCompleteFrame(); - ASSERT_TRUE(frame_out != NULL); + ASSERT_TRUE(frame_out != nullptr); CheckOutFrame(frame_out, size_, false); EXPECT_EQ(VideoFrameType::kVideoFrameKey, frame_out->FrameType()); jitter_buffer_->ReleaseFrame(frame_out); @@ -646,7 +646,7 @@ TEST_F(TestBasicJitterBuffer, DuplicatePreviousDeltaFramePacket) { // Should be able to decode 3 delta frames, key frame already decoded. for (size_t i = 0; i < 3; ++i) { frame_out = DecodeCompleteFrame(); - ASSERT_TRUE(frame_out != NULL); + ASSERT_TRUE(frame_out != nullptr); CheckOutFrame(frame_out, size_, false); EXPECT_EQ(VideoFrameType::kVideoFrameDelta, frame_out->FrameType()); jitter_buffer_->ReleaseFrame(frame_out); @@ -886,7 +886,7 @@ TEST_F(TestBasicJitterBuffer, H264InsertStartCode) { VCMEncodedFrame* frame_out = DecodeCompleteFrame(); // Frame should not be complete. - EXPECT_TRUE(frame_out == NULL); + EXPECT_TRUE(frame_out == nullptr); seq_num_++; packet_->video_header.is_first_packet_in_frame = false; @@ -986,7 +986,7 @@ TEST_F(TestBasicJitterBuffer, DeltaFrame100PacketsWithSeqNumWrap) { VCMEncodedFrame* frame_out = DecodeCompleteFrame(); - EXPECT_TRUE(frame_out == NULL); + EXPECT_TRUE(frame_out == nullptr); int loop = 0; do { @@ -1000,7 +1000,7 @@ TEST_F(TestBasicJitterBuffer, DeltaFrame100PacketsWithSeqNumWrap) { frame_out = DecodeCompleteFrame(); - EXPECT_TRUE(frame_out == NULL); + EXPECT_TRUE(frame_out == nullptr); loop++; } while (loop < 98); @@ -1035,7 +1035,7 @@ TEST_F(TestBasicJitterBuffer, PacketReorderingReverseWithNegSeqNumWrap) { VCMEncodedFrame* frame_out = DecodeCompleteFrame(); // Should not be complete. - EXPECT_TRUE(frame_out == NULL); + EXPECT_TRUE(frame_out == nullptr); // Insert 98 frames. int loop = 0; @@ -1050,7 +1050,7 @@ TEST_F(TestBasicJitterBuffer, PacketReorderingReverseWithNegSeqNumWrap) { frame_out = DecodeCompleteFrame(); - EXPECT_TRUE(frame_out == NULL); + EXPECT_TRUE(frame_out == nullptr); loop++; } while (loop < 98); @@ -1161,7 +1161,7 @@ TEST_F(TestBasicJitterBuffer, TimestampWrap) { jitter_buffer_->InsertPacket(*packet_, &retransmitted)); VCMEncodedFrame* frame_out = DecodeCompleteFrame(); - EXPECT_TRUE(frame_out == NULL); + EXPECT_TRUE(frame_out == nullptr); seq_num_++; packet_->video_header.is_first_packet_in_frame = false; @@ -1187,7 +1187,7 @@ TEST_F(TestBasicJitterBuffer, TimestampWrap) { jitter_buffer_->InsertPacket(*packet_, &retransmitted)); frame_out = DecodeCompleteFrame(); - EXPECT_TRUE(frame_out == NULL); + EXPECT_TRUE(frame_out == nullptr); seq_num_++; packet_->video_header.is_first_packet_in_frame = false; @@ -1320,7 +1320,7 @@ TEST_F(TestBasicJitterBuffer, DeltaFrameWithMoreThanMaxNumberOfPackets) { // Insert the packet -> frame recycled. EXPECT_EQ(kSizeError, jitter_buffer_->InsertPacket(*packet_, &retransmitted)); - EXPECT_TRUE(NULL == DecodeCompleteFrame()); + EXPECT_TRUE(nullptr == DecodeCompleteFrame()); } TEST_F(TestBasicJitterBuffer, ExceedNumOfFrameWithSeqNumWrap) { @@ -1415,7 +1415,7 @@ TEST_F(TestBasicJitterBuffer, NextFrameWhenIncomplete) { EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &retransmitted)); VCMEncodedFrame* frame_out = DecodeCompleteFrame(); - EXPECT_TRUE(frame_out != NULL); + EXPECT_TRUE(frame_out != nullptr); jitter_buffer_->ReleaseFrame(frame_out); packet_->seqNum += 2; @@ -1638,7 +1638,7 @@ TEST_F(TestJitterBufferNack, NackListBuiltBeforeFirstDecode) { InsertFrame(VideoFrameType::kVideoFrameKey); stream_generator_->GenerateFrame(VideoFrameType::kVideoFrameDelta, 2, 0, clock_->TimeInMilliseconds()); - stream_generator_->NextPacket(NULL); // Drop packet. + stream_generator_->NextPacket(nullptr); // Drop packet. EXPECT_EQ(kIncomplete, InsertPacketAndPop(0)); EXPECT_TRUE(DecodeCompleteFrame()); bool extended = false; @@ -1734,7 +1734,7 @@ TEST_F(TestJitterBufferNack, NormalOperation) { if (stream_generator_->NextSequenceNumber() % 10 != 0) { EXPECT_EQ(kIncomplete, InsertPacketAndPop(0)); } else { - stream_generator_->NextPacket(NULL); // Drop packet + stream_generator_->NextPacket(nullptr); // Drop packet } } EXPECT_EQ(kIncomplete, InsertPacketAndPop(0)); @@ -1768,7 +1768,7 @@ TEST_F(TestJitterBufferNack, NormalOperationWrap) { EXPECT_EQ(kIncomplete, InsertPacketAndPop(0)); EXPECT_FALSE(request_key_frame); } else { - stream_generator_->NextPacket(NULL); // Drop packet + stream_generator_->NextPacket(nullptr); // Drop packet } } EXPECT_EQ(kIncomplete, InsertPacketAndPop(0)); @@ -1802,7 +1802,7 @@ TEST_F(TestJitterBufferNack, NormalOperationWrap2) { EXPECT_EQ(kCompleteSession, InsertPacketAndPop(0)); EXPECT_FALSE(request_key_frame); } else { - stream_generator_->NextPacket(NULL); // Drop packet + stream_generator_->NextPacket(nullptr); // Drop packet } stream_generator_->GenerateFrame(VideoFrameType::kVideoFrameDelta, 1, 0, clock_->TimeInMilliseconds()); diff --git a/modules/video_coding/deprecated/packet.cc b/modules/video_coding/deprecated/packet.cc index 6b55bc45d3..ca9bfa4eec 100644 --- a/modules/video_coding/deprecated/packet.cc +++ b/modules/video_coding/deprecated/packet.cc @@ -26,7 +26,7 @@ VCMPacket::VCMPacket() timestamp(0), ntp_time_ms_(0), seqNum(0), - dataPtr(NULL), + dataPtr(nullptr), sizeBytes(0), markerBit(false), timesNacked(-1), diff --git a/modules/video_coding/deprecated/receiver.cc b/modules/video_coding/deprecated/receiver.cc index c9b41ddd14..b42d5599c3 100644 --- a/modules/video_coding/deprecated/receiver.cc +++ b/modules/video_coding/deprecated/receiver.cc @@ -137,7 +137,7 @@ VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms, // Timing error => reset timing and flush the jitter buffer. jitter_buffer_.Flush(); timing_->Reset(); - return NULL; + return nullptr; } if (prefer_late_decoding) { @@ -158,7 +158,7 @@ VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms, // waiting as long as we're allowed to avoid busy looping, and then return // NULL. Next call to this function might return the frame. render_wait_event_->Wait(new_max_wait_time); - return NULL; + return nullptr; } // Wait until it's time to render. render_wait_event_->Wait(wait_time_ms); @@ -166,8 +166,8 @@ VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms, // Extract the frame from the jitter buffer and set the render time. VCMEncodedFrame* frame = jitter_buffer_.ExtractAndSetDecode(frame_timestamp); - if (frame == NULL) { - return NULL; + if (frame == nullptr) { + return nullptr; } frame->SetRenderTime(render_time_ms); TRACE_EVENT_ASYNC_STEP_INTO1("webrtc", "Video", frame->RtpTimestamp(), diff --git a/modules/video_coding/deprecated/session_info.cc b/modules/video_coding/deprecated/session_info.cc index c351425dec..0c937a4734 100644 --- a/modules/video_coding/deprecated/session_info.cc +++ b/modules/video_coding/deprecated/session_info.cc @@ -54,8 +54,8 @@ VCMSessionInfo::~VCMSessionInfo() {} void VCMSessionInfo::UpdateDataPointers(const uint8_t* old_base_ptr, const uint8_t* new_base_ptr) { for (PacketIterator it = packets_.begin(); it != packets_.end(); ++it) - if ((*it).dataPtr != NULL) { - RTC_DCHECK(old_base_ptr != NULL && new_base_ptr != NULL); + if ((*it).dataPtr != nullptr) { + RTC_DCHECK(old_base_ptr != nullptr && new_base_ptr != nullptr); (*it).dataPtr = new_base_ptr + ((*it).dataPtr - old_base_ptr); } } @@ -272,7 +272,7 @@ void VCMSessionInfo::ShiftSubsequentPackets(PacketIterator it, // Calculate the total move length and move the data pointers in advance. for (; it != packets_.end(); ++it) { shift_length += (*it).sizeBytes; - if ((*it).dataPtr != NULL) + if ((*it).dataPtr != nullptr) (*it).dataPtr += steps_to_shift; } memmove(first_packet_ptr + steps_to_shift, first_packet_ptr, shift_length); @@ -334,7 +334,7 @@ size_t VCMSessionInfo::DeletePacketData(PacketIterator start, for (PacketIterator it = start; it != packet_after_end; ++it) { bytes_to_delete += (*it).sizeBytes; (*it).sizeBytes = 0; - (*it).dataPtr = NULL; + (*it).dataPtr = nullptr; } if (bytes_to_delete > 0) ShiftSubsequentPackets(end, -static_cast(bytes_to_delete)); diff --git a/modules/video_coding/deprecated/stream_generator.cc b/modules/video_coding/deprecated/stream_generator.cc index 09be3382fe..2870991505 100644 --- a/modules/video_coding/deprecated/stream_generator.cc +++ b/modules/video_coding/deprecated/stream_generator.cc @@ -97,7 +97,7 @@ bool StreamGenerator::GetPacket(VCMPacket* packet, int index) { bool StreamGenerator::NextPacket(VCMPacket* packet) { if (packets_.empty()) return false; - if (packet != NULL) + if (packet != nullptr) *packet = packets_.front(); packets_.pop_front(); return true; diff --git a/modules/video_coding/generic_decoder.cc b/modules/video_coding/generic_decoder.cc index b3d6e51553..b6d14adf6b 100644 --- a/modules/video_coding/generic_decoder.cc +++ b/modules/video_coding/generic_decoder.cc @@ -292,7 +292,7 @@ void VCMDecodedFrameCallback::ClearTimestampMap() { } VCMGenericDecoder::VCMGenericDecoder(VideoDecoder* decoder) - : _callback(NULL), + : _callback(nullptr), decoder_(decoder), _last_keyframe_content_type(VideoContentType::UNSPECIFIED) { RTC_DCHECK(decoder_); diff --git a/modules/video_coding/packet_buffer_unittest.cc b/modules/video_coding/packet_buffer_unittest.cc index 17a0d75c3b..a75b57d6ca 100644 --- a/modules/video_coding/packet_buffer_unittest.cc +++ b/modules/video_coding/packet_buffer_unittest.cc @@ -859,9 +859,9 @@ TEST_F(PacketBufferH264FrameGap, TEST_F(PacketBufferH264FrameGap, DoesntCrashWhenTryToClearBefore1stPacket) { // Test scenario copied from the https://issues.chromium.org/370689424 - InsertH264(41087, kKeyFrame, kNotFirst, kNotLast, 123, 0, 0, false); + InsertH264(41087, kKeyFrame, kNotFirst, kNotLast, 123, nullptr, 0, false); packet_buffer_.ClearTo(30896); - InsertH264(32896, kKeyFrame, kFirst, kLast, 123, 0, 0, false); + InsertH264(32896, kKeyFrame, kFirst, kLast, 123, nullptr, 0, false); } } // namespace diff --git a/modules/video_coding/utility/simulcast_test_fixture_impl.cc b/modules/video_coding/utility/simulcast_test_fixture_impl.cc index 129d76368b..4dbc6b3da6 100644 --- a/modules/video_coding/utility/simulcast_test_fixture_impl.cc +++ b/modules/video_coding/utility/simulcast_test_fixture_impl.cc @@ -766,7 +766,7 @@ void SimulcastTestFixtureImpl::TestSpatioTemporalLayers333PatternEncoder() { bool expected_layer_sync[3] = {false, false, false}; // First frame: #0. - EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL)); + EXPECT_EQ(0, encoder_->Encode(*input_frame_, nullptr)); SetExpectedValues3(0, 0, 0, expected_temporal_idx); SetExpectedValues3(!is_h264, !is_h264, !is_h264, expected_layer_sync); VerifyTemporalIdxAndSyncForAllSpatialLayers( @@ -774,7 +774,7 @@ void SimulcastTestFixtureImpl::TestSpatioTemporalLayers333PatternEncoder() { // Next frame: #1. input_frame_->set_rtp_timestamp(input_frame_->rtp_timestamp() + 3000); - EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL)); + EXPECT_EQ(0, encoder_->Encode(*input_frame_, nullptr)); SetExpectedValues3(2, 2, 2, expected_temporal_idx); SetExpectedValues3(true, true, true, expected_layer_sync); VerifyTemporalIdxAndSyncForAllSpatialLayers( @@ -782,7 +782,7 @@ void SimulcastTestFixtureImpl::TestSpatioTemporalLayers333PatternEncoder() { // Next frame: #2. input_frame_->set_rtp_timestamp(input_frame_->rtp_timestamp() + 3000); - EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL)); + EXPECT_EQ(0, encoder_->Encode(*input_frame_, nullptr)); SetExpectedValues3(1, 1, 1, expected_temporal_idx); SetExpectedValues3(true, true, true, expected_layer_sync); VerifyTemporalIdxAndSyncForAllSpatialLayers( @@ -790,7 +790,7 @@ void SimulcastTestFixtureImpl::TestSpatioTemporalLayers333PatternEncoder() { // Next frame: #3. input_frame_->set_rtp_timestamp(input_frame_->rtp_timestamp() + 3000); - EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL)); + EXPECT_EQ(0, encoder_->Encode(*input_frame_, nullptr)); SetExpectedValues3(2, 2, 2, expected_temporal_idx); SetExpectedValues3(false, false, false, expected_layer_sync); VerifyTemporalIdxAndSyncForAllSpatialLayers( @@ -798,7 +798,7 @@ void SimulcastTestFixtureImpl::TestSpatioTemporalLayers333PatternEncoder() { // Next frame: #4. input_frame_->set_rtp_timestamp(input_frame_->rtp_timestamp() + 3000); - EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL)); + EXPECT_EQ(0, encoder_->Encode(*input_frame_, nullptr)); SetExpectedValues3(0, 0, 0, expected_temporal_idx); SetExpectedValues3(false, false, false, expected_layer_sync); VerifyTemporalIdxAndSyncForAllSpatialLayers( @@ -806,7 +806,7 @@ void SimulcastTestFixtureImpl::TestSpatioTemporalLayers333PatternEncoder() { // Next frame: #5. input_frame_->set_rtp_timestamp(input_frame_->rtp_timestamp() + 3000); - EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL)); + EXPECT_EQ(0, encoder_->Encode(*input_frame_, nullptr)); SetExpectedValues3(2, 2, 2, expected_temporal_idx); SetExpectedValues3(is_h264, is_h264, is_h264, expected_layer_sync); VerifyTemporalIdxAndSyncForAllSpatialLayers( @@ -836,7 +836,7 @@ void SimulcastTestFixtureImpl::TestSpatioTemporalLayers321PatternEncoder() { bool expected_layer_sync[3] = {false, false, false}; // First frame: #0. - EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL)); + EXPECT_EQ(0, encoder_->Encode(*input_frame_, nullptr)); SetExpectedValues3(0, 0, 255, expected_temporal_idx); SetExpectedValues3(true, true, false, expected_layer_sync); VerifyTemporalIdxAndSyncForAllSpatialLayers( @@ -844,7 +844,7 @@ void SimulcastTestFixtureImpl::TestSpatioTemporalLayers321PatternEncoder() { // Next frame: #1. input_frame_->set_rtp_timestamp(input_frame_->rtp_timestamp() + 3000); - EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL)); + EXPECT_EQ(0, encoder_->Encode(*input_frame_, nullptr)); SetExpectedValues3(2, 1, 255, expected_temporal_idx); SetExpectedValues3(true, true, false, expected_layer_sync); VerifyTemporalIdxAndSyncForAllSpatialLayers( @@ -852,7 +852,7 @@ void SimulcastTestFixtureImpl::TestSpatioTemporalLayers321PatternEncoder() { // Next frame: #2. input_frame_->set_rtp_timestamp(input_frame_->rtp_timestamp() + 3000); - EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL)); + EXPECT_EQ(0, encoder_->Encode(*input_frame_, nullptr)); SetExpectedValues3(1, 0, 255, expected_temporal_idx); SetExpectedValues3(true, false, false, expected_layer_sync); VerifyTemporalIdxAndSyncForAllSpatialLayers( @@ -860,7 +860,7 @@ void SimulcastTestFixtureImpl::TestSpatioTemporalLayers321PatternEncoder() { // Next frame: #3. input_frame_->set_rtp_timestamp(input_frame_->rtp_timestamp() + 3000); - EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL)); + EXPECT_EQ(0, encoder_->Encode(*input_frame_, nullptr)); SetExpectedValues3(2, 1, 255, expected_temporal_idx); SetExpectedValues3(false, false, false, expected_layer_sync); VerifyTemporalIdxAndSyncForAllSpatialLayers( @@ -868,7 +868,7 @@ void SimulcastTestFixtureImpl::TestSpatioTemporalLayers321PatternEncoder() { // Next frame: #4. input_frame_->set_rtp_timestamp(input_frame_->rtp_timestamp() + 3000); - EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL)); + EXPECT_EQ(0, encoder_->Encode(*input_frame_, nullptr)); SetExpectedValues3(0, 0, 255, expected_temporal_idx); SetExpectedValues3(false, false, false, expected_layer_sync); VerifyTemporalIdxAndSyncForAllSpatialLayers( @@ -876,7 +876,7 @@ void SimulcastTestFixtureImpl::TestSpatioTemporalLayers321PatternEncoder() { // Next frame: #5. input_frame_->set_rtp_timestamp(input_frame_->rtp_timestamp() + 3000); - EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL)); + EXPECT_EQ(0, encoder_->Encode(*input_frame_, nullptr)); SetExpectedValues3(2, 1, 255, expected_temporal_idx); SetExpectedValues3(false, true, false, expected_layer_sync); VerifyTemporalIdxAndSyncForAllSpatialLayers( @@ -910,7 +910,7 @@ void SimulcastTestFixtureImpl::TestStrideEncodeDecode() { plane_offset[kVPlane] = kColorV; CreateImage(input_buffer_, plane_offset); - EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL)); + EXPECT_EQ(0, encoder_->Encode(*input_frame_, nullptr)); // Change color. plane_offset[kYPlane] += 1; @@ -918,7 +918,7 @@ void SimulcastTestFixtureImpl::TestStrideEncodeDecode() { plane_offset[kVPlane] += 1; CreateImage(input_buffer_, plane_offset); input_frame_->set_rtp_timestamp(input_frame_->rtp_timestamp() + 3000); - EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL)); + EXPECT_EQ(0, encoder_->Encode(*input_frame_, nullptr)); EncodedImage encoded_frame; // Only encoding one frame - so will be a key frame. @@ -952,7 +952,7 @@ void SimulcastTestFixtureImpl::TestDecodeWidthHeightSet() { return EncodedImageCallback::Result( EncodedImageCallback::Result::OK, 0); })); - EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL)); + EXPECT_EQ(0, encoder_->Encode(*input_frame_, nullptr)); EXPECT_CALL(decoder_callback, Decoded(_, _, _)) .WillOnce( From e7f936d2821c7bf6aba427795cf44f575d7a7038 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Thu, 15 May 2025 11:50:44 +0000 Subject: [PATCH 172/292] [cleanup] Replace uses of NULL with nullptr Generated using clang-tidy, tools/clang/scripts/build_clang_tools_extra.py \ --fetch out/Default clang-tidy clang-apply-replacements ninja -C out/Default gn gen out/Default --export-compile-commands cd out/Default tools/clang/third_party/llvm/clang-tools-extra/clang-tidy/tool/run-clang-tidy.py -p . \ -clang-tidy-binary out/Default/tools/clang/third_party/llvm/build/bin/clang-tidy \ -clang-apply-replacements-binary \ out/Default/tools/clang/third_party/llvm/build/bin/clang-apply-replacements \ -checks='-*,modernize-use-nullptr' \ -fix This CL was uploaded by git cl split. Bug: webrtc:42232595 Change-Id: Ic5d702891a2431aaac94504c038630acdf836b37 No-Iwyu: LSC Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391801 Auto-Submit: Evan Shrubsole Commit-Queue: Tomas Gunnarsson Reviewed-by: Tomas Gunnarsson Cr-Commit-Position: refs/heads/main@{#44644} --- examples/peerconnection/client/conductor.cc | 4 +- examples/peerconnection/client/linux/main.cc | 4 +- .../peerconnection/client/linux/main_wnd.cc | 58 +++++++++---------- .../client/peer_connection_client.cc | 21 ++++--- examples/peerconnection/server/data_socket.cc | 6 +- examples/peerconnection/server/main.cc | 3 +- .../peerconnection/server/peer_channel.cc | 28 ++++----- 7 files changed, 64 insertions(+), 60 deletions(-) diff --git a/examples/peerconnection/client/conductor.cc b/examples/peerconnection/client/conductor.cc index 34aefe96f1..dc5ad214a0 100644 --- a/examples/peerconnection/client/conductor.cc +++ b/examples/peerconnection/client/conductor.cc @@ -340,7 +340,7 @@ void Conductor::OnPeerDisconnected(int id) { RTC_LOG(LS_INFO) << __FUNCTION__; if (id == peer_id_) { RTC_LOG(LS_INFO) << "Our peer disconnected"; - main_wnd_->QueueUIThreadCallback(PEER_CONNECTION_CLOSED, NULL); + main_wnd_->QueueUIThreadCallback(PEER_CONNECTION_CLOSED, nullptr); } else { // Refresh the list if we're showing it. if (main_wnd_->current_ui() == MainWindow::LIST_PEERS) @@ -457,7 +457,7 @@ void Conductor::OnMessageFromPeer(int peer_id, const std::string& message) { void Conductor::OnMessageSent(int err) { // Process the next pending message if any. - main_wnd_->QueueUIThreadCallback(SEND_MESSAGE_TO_PEER, NULL); + main_wnd_->QueueUIThreadCallback(SEND_MESSAGE_TO_PEER, nullptr); } void Conductor::OnServerConnectionFailure() { diff --git a/examples/peerconnection/client/linux/main.cc b/examples/peerconnection/client/linux/main.cc index 3278e9d0fd..5e3be8beb9 100644 --- a/examples/peerconnection/client/linux/main.cc +++ b/examples/peerconnection/client/linux/main.cc @@ -27,7 +27,7 @@ class CustomSocketServer : public webrtc::PhysicalSocketServer { public: explicit CustomSocketServer(GtkMainWnd* wnd) - : wnd_(wnd), conductor_(NULL), client_(NULL) {} + : wnd_(wnd), conductor_(nullptr), client_(nullptr) {} virtual ~CustomSocketServer() {} void SetMessageQueue(webrtc::Thread* queue) override { @@ -49,7 +49,7 @@ class CustomSocketServer : public webrtc::PhysicalSocketServer { gtk_main_iteration(); if (!wnd_->IsWindow() && !conductor_->connection_active() && - client_ != NULL && !client_->is_connected()) { + client_ != nullptr && !client_->is_connected()) { message_queue_->Quit(); } return webrtc::PhysicalSocketServer::Wait(webrtc::TimeDelta::Zero(), diff --git a/examples/peerconnection/client/linux/main_wnd.cc b/examples/peerconnection/client/linux/main_wnd.cc index c750365dbc..ef624f1a65 100644 --- a/examples/peerconnection/client/linux/main_wnd.cc +++ b/examples/peerconnection/client/linux/main_wnd.cc @@ -80,7 +80,7 @@ gboolean SimulateLastRowActivated(gpointer data) { GtkTreeModel* model = gtk_tree_view_get_model(tree_view); // "if iter is NULL, then the number of toplevel nodes is returned." - int rows = gtk_tree_model_iter_n_children(model, NULL); + int rows = gtk_tree_model_iter_n_children(model, nullptr); GtkTreePath* lastpath = gtk_tree_path_new_from_indices(rows - 1, -1); // Select the last item in the list @@ -154,13 +154,13 @@ GtkMainWnd::GtkMainWnd(const char* server, int port, bool autoconnect, bool autocall) - : window_(NULL), - draw_area_(NULL), - vbox_(NULL), - server_edit_(NULL), - port_edit_(NULL), - peer_list_(NULL), - callback_(NULL), + : window_(nullptr), + draw_area_(nullptr), + vbox_(nullptr), + server_edit_(nullptr), + port_edit_(nullptr), + peer_list_(nullptr), + callback_(nullptr), server_(server), autoconnect_(autoconnect), autocall_(autocall) { @@ -178,7 +178,7 @@ void GtkMainWnd::RegisterObserver(MainWndCallback* callback) { } bool GtkMainWnd::IsWindow() { - return window_ != NULL && GTK_IS_WINDOW(window_); + return window_ != nullptr && GTK_IS_WINDOW(window_); } void GtkMainWnd::MessageBox(const char* caption, @@ -226,7 +226,7 @@ void GtkMainWnd::QueueUIThreadCallback(int msg_id, void* data) { } bool GtkMainWnd::Create() { - RTC_DCHECK(window_ == NULL); + RTC_DCHECK(window_ == nullptr); window_ = gtk_window_new(GTK_WINDOW_TOPLEVEL); if (window_) { @@ -241,7 +241,7 @@ bool GtkMainWnd::Create() { SwitchToConnectUI(); } - return window_ != NULL; + return window_ != nullptr; } bool GtkMainWnd::Destroy() { @@ -249,7 +249,7 @@ bool GtkMainWnd::Destroy() { return false; gtk_widget_destroy(window_); - window_ = NULL; + window_ = nullptr; return true; } @@ -258,13 +258,13 @@ void GtkMainWnd::SwitchToConnectUI() { RTC_LOG(LS_INFO) << __FUNCTION__; RTC_DCHECK(IsWindow()); - RTC_DCHECK(vbox_ == NULL); + RTC_DCHECK(vbox_ == nullptr); gtk_container_set_border_width(GTK_CONTAINER(window_), 10); if (peer_list_) { gtk_widget_destroy(peer_list_); - peer_list_ = NULL; + peer_list_ = nullptr; } vbox_ = gtk_box_new(GTK_ORIENTATION_VERTICAL, 5); @@ -309,12 +309,12 @@ void GtkMainWnd::SwitchToPeerList(const Peers& peers) { gtk_container_set_border_width(GTK_CONTAINER(window_), 0); if (vbox_) { gtk_widget_destroy(vbox_); - vbox_ = NULL; - server_edit_ = NULL; - port_edit_ = NULL; + vbox_ = nullptr; + server_edit_ = nullptr; + port_edit_ = nullptr; } else if (draw_area_) { gtk_widget_destroy(draw_area_); - draw_area_ = NULL; + draw_area_ = nullptr; draw_buffer_.SetSize(0); } @@ -342,12 +342,12 @@ void GtkMainWnd::SwitchToPeerList(const Peers& peers) { void GtkMainWnd::SwitchToStreamingUI() { RTC_LOG(LS_INFO) << __FUNCTION__; - RTC_DCHECK(draw_area_ == NULL); + RTC_DCHECK(draw_area_ == nullptr); gtk_container_set_border_width(GTK_CONTAINER(window_), 0); if (peer_list_) { gtk_widget_destroy(peer_list_); - peer_list_ = NULL; + peer_list_ = nullptr; } draw_area_ = gtk_drawing_area_new(); @@ -359,12 +359,12 @@ void GtkMainWnd::SwitchToStreamingUI() { void GtkMainWnd::OnDestroyed(GtkWidget* widget, GdkEvent* event) { callback_->Close(); - window_ = NULL; - draw_area_ = NULL; - vbox_ = NULL; - server_edit_ = NULL; - port_edit_ = NULL; - peer_list_ = NULL; + window_ = nullptr; + draw_area_ = nullptr; + vbox_ = nullptr; + server_edit_ = nullptr; + port_edit_ = nullptr; + peer_list_ = nullptr; } void GtkMainWnd::OnClicked(GtkWidget* widget) { @@ -392,7 +392,7 @@ void GtkMainWnd::OnKeyPress(GtkWidget* widget, GdkEventKey* key) { case GDK_KEY_KP_Enter: case GDK_KEY_Return: if (vbox_) { - OnClicked(NULL); + OnClicked(nullptr); } else if (peer_list_) { // OnRowActivated will be called automatically when the user // presses enter. @@ -408,7 +408,7 @@ void GtkMainWnd::OnKeyPress(GtkWidget* widget, GdkEventKey* key) { void GtkMainWnd::OnRowActivated(GtkTreeView* tree_view, GtkTreePath* path, GtkTreeViewColumn* column) { - RTC_DCHECK(peer_list_ != NULL); + RTC_DCHECK(peer_list_ != nullptr); GtkTreeIter iter; GtkTreeModel* model; GtkTreeSelection* selection = @@ -428,7 +428,7 @@ void GtkMainWnd::OnRedraw() { VideoRenderer* remote_renderer = remote_renderer_.get(); if (remote_renderer && !remote_renderer->image().empty() && - draw_area_ != NULL) { + draw_area_ != nullptr) { if (width_ != remote_renderer->width() || height_ != remote_renderer->height()) { width_ = remote_renderer->width(); diff --git a/examples/peerconnection/client/peer_connection_client.cc b/examples/peerconnection/client/peer_connection_client.cc index b646c58226..6aa60e8746 100644 --- a/examples/peerconnection/client/peer_connection_client.cc +++ b/examples/peerconnection/client/peer_connection_client.cc @@ -27,20 +27,23 @@ constexpr webrtc::TimeDelta kReconnectDelay = webrtc::TimeDelta::Seconds(2); webrtc::Socket* CreateClientSocket(int family) { webrtc::Thread* thread = webrtc::Thread::Current(); - RTC_DCHECK(thread != NULL); + RTC_DCHECK(thread != nullptr); return thread->socketserver()->CreateSocket(family, SOCK_STREAM); } } // namespace PeerConnectionClient::PeerConnectionClient() - : callback_(NULL), resolver_(nullptr), state_(NOT_CONNECTED), my_id_(-1) {} + : callback_(nullptr), + resolver_(nullptr), + state_(NOT_CONNECTED), + my_id_(-1) {} PeerConnectionClient::~PeerConnectionClient() = default; void PeerConnectionClient::InitSocketSignals() { - RTC_DCHECK(control_socket_.get() != NULL); - RTC_DCHECK(hanging_get_.get() != NULL); + RTC_DCHECK(control_socket_.get() != nullptr); + RTC_DCHECK(hanging_get_.get() != nullptr); control_socket_->SignalCloseEvent.connect(this, &PeerConnectionClient::OnClose); hanging_get_->SignalCloseEvent.connect(this, &PeerConnectionClient::OnClose); @@ -249,7 +252,7 @@ bool PeerConnectionClient::GetHeaderValue(const std::string& data, size_t eoh, const char* header_pattern, size_t* value) { - RTC_DCHECK(value != NULL); + RTC_DCHECK(value != nullptr); size_t found = data.find(header_pattern); if (found != std::string::npos && found < eoh) { *value = atoi(&data[found + strlen(header_pattern)]); @@ -262,7 +265,7 @@ bool PeerConnectionClient::GetHeaderValue(const std::string& data, size_t eoh, const char* header_pattern, std::string* value) { - RTC_DCHECK(value != NULL); + RTC_DCHECK(value != nullptr); size_t found = data.find(header_pattern); if (found != std::string::npos && found < eoh) { size_t begin = found + strlen(header_pattern); @@ -412,9 +415,9 @@ bool PeerConnectionClient::ParseEntry(const std::string& entry, std::string* name, int* id, bool* connected) { - RTC_DCHECK(name != NULL); - RTC_DCHECK(id != NULL); - RTC_DCHECK(connected != NULL); + RTC_DCHECK(name != nullptr); + RTC_DCHECK(id != nullptr); + RTC_DCHECK(connected != nullptr); RTC_DCHECK(!entry.empty()); *connected = false; diff --git a/examples/peerconnection/server/data_socket.cc b/examples/peerconnection/server/data_socket.cc index 855ebd8c0c..75835aede3 100644 --- a/examples/peerconnection/server/data_socket.cc +++ b/examples/peerconnection/server/data_socket.cc @@ -197,7 +197,7 @@ bool DataSocket::ParseMethodAndPath(const char* begin, size_t len) { {"OPTIONS", 7, OPTIONS}, }; - const char* path = NULL; + const char* path = nullptr; for (size_t i = 0; i < ARRAYSIZE(supported_methods); ++i) { if (len > supported_methods[i].method_name_len && isspace(begin[supported_methods[i].method_name_len]) && @@ -246,7 +246,7 @@ bool DataSocket::ParseContentLengthAndType(const char* headers, size_t length) { while (headers[0] == ' ') ++headers; const char* type_end = strstr(headers, "\r\n"); - if (type_end == NULL) + if (type_end == nullptr) type_end = end; content_type_.assign(headers, type_end); } @@ -293,7 +293,7 @@ DataSocket* ListeningSocket::Accept() const { NativeSocket client = accept(socket_, reinterpret_cast(&addr), &size); if (client == INVALID_SOCKET) - return NULL; + return nullptr; return new DataSocket(client); } diff --git a/examples/peerconnection/server/main.cc b/examples/peerconnection/server/main.cc index 50b8c23401..501c7fdaee 100644 --- a/examples/peerconnection/server/main.cc +++ b/examples/peerconnection/server/main.cc @@ -109,7 +109,8 @@ int main(int argc, char* argv[]) { FD_SET((*i)->socket(), &socket_set); struct timeval timeout = {10, 0}; - if (select(FD_SETSIZE, &socket_set, NULL, NULL, &timeout) == SOCKET_ERROR) { + if (select(FD_SETSIZE, &socket_set, nullptr, nullptr, &timeout) == + SOCKET_ERROR) { printf("select failed\n"); break; } diff --git a/examples/peerconnection/server/peer_channel.cc b/examples/peerconnection/server/peer_channel.cc index f53820cc60..bb74124977 100644 --- a/examples/peerconnection/server/peer_channel.cc +++ b/examples/peerconnection/server/peer_channel.cc @@ -53,10 +53,10 @@ const size_t kMaxNameLength = 512; int ChannelMember::s_member_id_ = 0; ChannelMember::ChannelMember(DataSocket* socket) - : waiting_socket_(NULL), + : waiting_socket_(nullptr), id_(++s_member_id_), connected_(true), - timestamp_(time(NULL)) { + timestamp_(time(nullptr)) { RTC_DCHECK(socket); RTC_DCHECK_EQ(socket->method(), DataSocket::GET); RTC_DCHECK(socket->PathEquals("/sign_in")); @@ -76,7 +76,7 @@ bool ChannelMember::is_wait_request(DataSocket* ds) const { } bool ChannelMember::TimedOut() { - return waiting_socket_ == NULL && (time(NULL) - timestamp_) > 30; + return waiting_socket_ == nullptr && (time(nullptr) - timestamp_) > 30; } std::string ChannelMember::GetPeerIdHeader() const { @@ -119,8 +119,8 @@ void ChannelMember::ForwardRequestToPeer(DataSocket* ds, ChannelMember* peer) { void ChannelMember::OnClosing(DataSocket* ds) { if (ds == waiting_socket_) { - waiting_socket_ = NULL; - timestamp_ = time(NULL); + waiting_socket_ = nullptr; + timestamp_ = time(nullptr); } } @@ -136,8 +136,8 @@ void ChannelMember::QueueResponse(const std::string& status, if (!ok) { printf("Failed to deliver data to waiting socket\n"); } - waiting_socket_ = NULL; - timestamp_ = time(NULL); + waiting_socket_ = nullptr; + timestamp_ = time(nullptr); } else { QueuedResponse qr; qr.status = status; @@ -176,7 +176,7 @@ ChannelMember* PeerChannel::Lookup(DataSocket* ds) const { RTC_DCHECK(ds); if (ds->method() != DataSocket::GET && ds->method() != DataSocket::POST) - return NULL; + return nullptr; size_t i = 0; for (; i < ARRAYSIZE(kRequestPaths); ++i) { @@ -185,13 +185,13 @@ ChannelMember* PeerChannel::Lookup(DataSocket* ds) const { } if (i == ARRAYSIZE(kRequestPaths)) - return NULL; + return nullptr; std::string args(ds->request_arguments()); static const char kPeerId[] = "peer_id="; size_t found = args.find(kPeerId); if (found == std::string::npos) - return NULL; + return nullptr; int id = atoi(&args[found + ARRAYSIZE(kPeerId) - 1]); Members::const_iterator iter = members_.begin(); @@ -205,7 +205,7 @@ ChannelMember* PeerChannel::Lookup(DataSocket* ds) const { } } - return NULL; + return nullptr; } ChannelMember* PeerChannel::IsTargetedRequest(const DataSocket* ds) const { @@ -215,13 +215,13 @@ ChannelMember* PeerChannel::IsTargetedRequest(const DataSocket* ds) const { const std::string& path = ds->request_path(); size_t args = path.find('?'); if (args == std::string::npos) - return NULL; + return nullptr; size_t found; const char kTargetPeerIdParam[] = "to="; do { found = path.find(kTargetPeerIdParam, args); if (found == std::string::npos) - return NULL; + return nullptr; if (found == (args + 1) || path[found - 1] == '&') { found += ARRAYSIZE(kTargetPeerIdParam) - 1; break; @@ -235,7 +235,7 @@ ChannelMember* PeerChannel::IsTargetedRequest(const DataSocket* ds) const { return *i; } } - return NULL; + return nullptr; } bool PeerChannel::AddMember(DataSocket* ds) { From 410c1ba1885f069fa4bcfeeb79d7fceb75502575 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Thu, 15 May 2025 11:53:14 +0000 Subject: [PATCH 173/292] [cleanup] Replace uses of NULL with nullptr Generated using clang-tidy, tools/clang/scripts/build_clang_tools_extra.py \ --fetch out/Default clang-tidy clang-apply-replacements ninja -C out/Default gn gen out/Default --export-compile-commands cd out/Default tools/clang/third_party/llvm/clang-tools-extra/clang-tidy/tool/run-clang-tidy.py -p . \ -clang-tidy-binary out/Default/tools/clang/third_party/llvm/build/bin/clang-tidy \ -clang-apply-replacements-binary \ out/Default/tools/clang/third_party/llvm/build/bin/clang-apply-replacements \ -checks='-*,modernize-use-nullptr' \ -fix This CL was uploaded by git cl split. Bug: webrtc:42232595 Change-Id: I08a6bc1757e921564343a10e2d5b397357fe793c No-Iwyu: LSC Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391862 Reviewed-by: Harald Alvestrand Commit-Queue: Harald Alvestrand Auto-Submit: Evan Shrubsole Cr-Commit-Position: refs/heads/main@{#44645} --- pc/dtmf_sender_unittest.cc | 2 +- pc/external_hmac.cc | 4 ++-- pc/jsep_ice_candidate.cc | 2 +- pc/jsep_session_description.cc | 2 +- pc/jsep_session_description_unittest.cc | 16 ++++++------- pc/jsep_transport.cc | 2 +- pc/legacy_stats_collector.cc | 12 +++++----- pc/legacy_stats_collector_unittest.cc | 8 +++---- pc/media_session.cc | 2 +- pc/media_session_unittest.cc | 8 +++---- pc/media_stream.cc | 2 +- pc/media_stream_unittest.cc | 6 ++--- pc/peer_connection.cc | 2 +- pc/peer_connection_factory_unittest.cc | 6 ++--- pc/peer_connection_interface_unittest.cc | 4 ++-- pc/remote_audio_source.cc | 4 ++-- pc/sctp_utils.cc | 2 +- pc/session_description.cc | 20 ++++++++-------- pc/test/fake_audio_capture_module_unittest.cc | 2 +- pc/test/peer_connection_test_wrapper.cc | 2 +- pc/webrtc_sdp.cc | 24 +++++++++---------- pc/webrtc_sdp_unittest.cc | 8 +++---- 22 files changed, 70 insertions(+), 70 deletions(-) diff --git a/pc/dtmf_sender_unittest.cc b/pc/dtmf_sender_unittest.cc index d121809129..6c442e1419 100644 --- a/pc/dtmf_sender_unittest.cc +++ b/pc/dtmf_sender_unittest.cc @@ -306,7 +306,7 @@ TEST_F(DtmfSenderTest, InsertDtmfWhileSenderIsDeleted) { .clock = &fake_clock_}), webrtc::IsRtcOk()); // Delete the sender. - dtmf_ = NULL; + dtmf_ = nullptr; // The queue should be discontinued so no more tone callbacks. fake_clock_.AdvanceTime(webrtc::TimeDelta::Millis(200)); EXPECT_EQ(1U, observer_->tones().size()); diff --git a/pc/external_hmac.cc b/pc/external_hmac.cc index 6ee917d2ba..77f11a32d7 100644 --- a/pc/external_hmac.cc +++ b/pc/external_hmac.cc @@ -39,7 +39,7 @@ static const srtp_auth_test_case_t kExternalHmacTestCase0 = { const_cast(kExternalHmacTestCase0Data), // Data 10, // Octets in tag const_cast(kExternalHmacFakeTag), // Tag - NULL // Pointer to next + nullptr // Pointer to next // testcase }; @@ -75,7 +75,7 @@ srtp_err_status_t external_hmac_alloc(srtp_auth_t** a, // Allocate memory for auth and hmac_ctx_t structures. pointer = new uint8_t[(sizeof(ExternalHmacContext) + sizeof(srtp_auth_t))]; - if (pointer == NULL) + if (pointer == nullptr) return srtp_err_status_alloc_fail; // Set pointers diff --git a/pc/jsep_ice_candidate.cc b/pc/jsep_ice_candidate.cc index 0b0a282431..3245f45584 100644 --- a/pc/jsep_ice_candidate.cc +++ b/pc/jsep_ice_candidate.cc @@ -31,7 +31,7 @@ IceCandidateInterface* CreateIceCandidate(const std::string& sdp_mid, JsepIceCandidate* jsep_ice = new JsepIceCandidate(sdp_mid, sdp_mline_index); if (!jsep_ice->Initialize(sdp, error)) { delete jsep_ice; - return NULL; + return nullptr; } return jsep_ice; } diff --git a/pc/jsep_session_description.cc b/pc/jsep_session_description.cc index fe4bd79afc..48feb1012a 100644 --- a/pc/jsep_session_description.cc +++ b/pc/jsep_session_description.cc @@ -287,7 +287,7 @@ size_t JsepSessionDescription::number_of_mediasections() const { const IceCandidateCollection* JsepSessionDescription::candidates( size_t mediasection_index) const { if (mediasection_index >= candidate_collection_.size()) - return NULL; + return nullptr; return &candidate_collection_[mediasection_index]; } diff --git a/pc/jsep_session_description_unittest.cc b/pc/jsep_session_description_unittest.cc index 60247df538..789ece47fb 100644 --- a/pc/jsep_session_description_unittest.cc +++ b/pc/jsep_session_description_unittest.cc @@ -74,12 +74,12 @@ CreateCricketSessionDescription() { webrtc::CN_AUDIO, webrtc::TransportDescription( std::vector(), kCandidateUfragVoice, kCandidatePwdVoice, - webrtc::ICEMODE_FULL, webrtc::CONNECTIONROLE_NONE, NULL))); + webrtc::ICEMODE_FULL, webrtc::CONNECTIONROLE_NONE, nullptr))); desc->AddTransportInfo(webrtc::TransportInfo( webrtc::CN_VIDEO, webrtc::TransportDescription( std::vector(), kCandidateUfragVideo, kCandidatePwdVideo, - webrtc::ICEMODE_FULL, webrtc::CONNECTIONROLE_NONE, NULL))); + webrtc::ICEMODE_FULL, webrtc::CONNECTIONROLE_NONE, nullptr))); return desc; } @@ -173,10 +173,10 @@ TEST_F(JsepSessionDescriptionTest, AddCandidateWithoutMid) { JsepIceCandidate jsep_candidate("", 0, candidate_); EXPECT_TRUE(jsep_desc_->AddCandidate(&jsep_candidate)); const IceCandidateCollection* ice_candidates = jsep_desc_->candidates(0); - ASSERT_TRUE(ice_candidates != NULL); + ASSERT_TRUE(ice_candidates != nullptr); EXPECT_EQ(1u, ice_candidates->count()); const IceCandidateInterface* ice_candidate = ice_candidates->at(0); - ASSERT_TRUE(ice_candidate != NULL); + ASSERT_TRUE(ice_candidate != nullptr); candidate_.set_username(kCandidateUfragVoice); candidate_.set_password(kCandidatePwdVoice); EXPECT_TRUE(ice_candidate->candidate().IsEquivalent(candidate_)); @@ -193,10 +193,10 @@ TEST_F(JsepSessionDescriptionTest, AddAndRemoveCandidatesWithMid) { EXPECT_TRUE(jsep_desc_->AddCandidate(&jsep_candidate)); EXPECT_EQ(0u, jsep_desc_->candidates(0)->count()); const IceCandidateCollection* ice_candidates = jsep_desc_->candidates(1); - ASSERT_TRUE(ice_candidates != NULL); + ASSERT_TRUE(ice_candidates != nullptr); EXPECT_EQ(1u, ice_candidates->count()); const IceCandidateInterface* ice_candidate = ice_candidates->at(0); - ASSERT_TRUE(ice_candidate != NULL); + ASSERT_TRUE(ice_candidate != nullptr); candidate_.set_username(kCandidateUfragVideo); candidate_.set_password(kCandidatePwdVideo); EXPECT_TRUE(ice_candidate->candidate().IsEquivalent(candidate_)); @@ -216,10 +216,10 @@ TEST_F(JsepSessionDescriptionTest, AddCandidateAlreadyHasUfrag) { JsepIceCandidate jsep_candidate("audio", 0, candidate_); EXPECT_TRUE(jsep_desc_->AddCandidate(&jsep_candidate)); const IceCandidateCollection* ice_candidates = jsep_desc_->candidates(0); - ASSERT_TRUE(ice_candidates != NULL); + ASSERT_TRUE(ice_candidates != nullptr); EXPECT_EQ(1u, ice_candidates->count()); const IceCandidateInterface* ice_candidate = ice_candidates->at(0); - ASSERT_TRUE(ice_candidate != NULL); + ASSERT_TRUE(ice_candidate != nullptr); candidate_.set_username(kCandidateUfrag); candidate_.set_password(kCandidatePwd); EXPECT_TRUE(ice_candidate->candidate().IsEquivalent(candidate_)); diff --git a/pc/jsep_transport.cc b/pc/jsep_transport.cc index 3a9234faea..b1f179cef6 100644 --- a/pc/jsep_transport.cc +++ b/pc/jsep_transport.cc @@ -366,7 +366,7 @@ RTCError JsepTransport::VerifyCertificateFingerprint( } std::unique_ptr fp_tmp = SSLFingerprint::CreateUnique( fingerprint->algorithm, *certificate->identity()); - RTC_DCHECK(fp_tmp.get() != NULL); + RTC_DCHECK(fp_tmp.get() != nullptr); if (*fp_tmp == *fingerprint) { return RTCError::OK(); } diff --git a/pc/legacy_stats_collector.cc b/pc/legacy_stats_collector.cc index e77db34c9b..e871468183 100644 --- a/pc/legacy_stats_collector.cc +++ b/pc/legacy_stats_collector.cc @@ -576,7 +576,7 @@ double LegacyStatsCollector::GetTimeNow() { // to GetStats. void LegacyStatsCollector::AddStream(MediaStreamInterface* stream) { RTC_DCHECK_RUN_ON(pc_->signaling_thread()); - RTC_DCHECK(stream != NULL); + RTC_DCHECK(stream != nullptr); CreateTrackReports(stream->GetAudioTracks(), &reports_, &track_ids_); @@ -599,7 +599,7 @@ void LegacyStatsCollector::AddTrack(MediaStreamTrackInterface* track) { void LegacyStatsCollector::AddLocalAudioTrack(AudioTrackInterface* audio_track, uint32_t ssrc) { RTC_DCHECK_RUN_ON(pc_->signaling_thread()); - RTC_DCHECK(audio_track != NULL); + RTC_DCHECK(audio_track != nullptr); #if RTC_DCHECK_IS_ON for (const auto& track : local_audio_tracks_) RTC_DCHECK(track.first != audio_track || track.second != ssrc); @@ -621,7 +621,7 @@ void LegacyStatsCollector::AddLocalAudioTrack(AudioTrackInterface* audio_track, void LegacyStatsCollector::RemoveLocalAudioTrack( AudioTrackInterface* audio_track, uint32_t ssrc) { - RTC_DCHECK(audio_track != NULL); + RTC_DCHECK(audio_track != nullptr); local_audio_tracks_.erase( std::remove_if( local_audio_tracks_.begin(), local_audio_tracks_.end(), @@ -634,7 +634,7 @@ void LegacyStatsCollector::RemoveLocalAudioTrack( void LegacyStatsCollector::GetStats(MediaStreamTrackInterface* track, StatsReports* reports) { RTC_DCHECK_RUN_ON(pc_->signaling_thread()); - RTC_DCHECK(reports != NULL); + RTC_DCHECK(reports != nullptr); RTC_DCHECK(reports->empty()); Thread::ScopedDisallowBlockingCalls no_blocking_calls; @@ -1360,7 +1360,7 @@ void LegacyStatsCollector::UpdateStatsFromExistingLocalAudioTracks( uint32_t ssrc = it.second; StatsReport* report = GetReport(StatsReport::kStatsReportTypeSsrc, absl::StrCat(ssrc), StatsReport::kSend); - if (report == NULL) { + if (report == nullptr) { // This can happen if a local audio track is added to a stream on the // fly and the report has not been set up yet. Do nothing in this case. RTC_LOG(LS_ERROR) << "Stats report does not exist for ssrc " << ssrc; @@ -1383,7 +1383,7 @@ void LegacyStatsCollector::UpdateReportFromAudioTrack( StatsReport* report, bool has_remote_tracks) { RTC_DCHECK_RUN_ON(pc_->signaling_thread()); - RTC_DCHECK(track != NULL); + RTC_DCHECK(track != nullptr); // Don't overwrite report values if they're not available. int signal_level; diff --git a/pc/legacy_stats_collector_unittest.cc b/pc/legacy_stats_collector_unittest.cc index efe90c2395..f9af394c63 100644 --- a/pc/legacy_stats_collector_unittest.cc +++ b/pc/legacy_stats_collector_unittest.cc @@ -111,7 +111,7 @@ class FakeAudioTrack : public MediaStreamTrack { : MediaStreamTrack(id), processor_(make_ref_counted()) {} std::string kind() const override { return "audio"; } - AudioSourceInterface* GetSource() const override { return NULL; } + AudioSourceInterface* GetSource() const override { return nullptr; } void AddSink(AudioTrackSinkInterface* sink) override {} void RemoveSink(AudioTrackSinkInterface* sink) override {} bool GetSignalLevel(int* level) override { @@ -148,7 +148,7 @@ class FakeAudioTrackWithInitValue : MediaStreamTrack(id), processor_(make_ref_counted()) {} std::string kind() const override { return "audio"; } - AudioSourceInterface* GetSource() const override { return NULL; } + AudioSourceInterface* GetSource() const override { return nullptr; } void AddSink(AudioTrackSinkInterface* sink) override {} void RemoveSink(AudioTrackSinkInterface* sink) override {} bool GetSignalLevel(int* level) override { @@ -295,7 +295,7 @@ void CheckCertChainReports(const StatsReports& reports, size_t i = 0; while (true) { const StatsReport* report = FindReportById(reports, *certificate_id); - ASSERT_TRUE(report != NULL); + ASSERT_TRUE(report != nullptr); std::string der_base64; EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameDer, &der_base64)); @@ -1644,7 +1644,7 @@ TEST_P(StatsCollectorTrackTest, GetStatsFromLocalAudioTrack) { // we did not set it up. const StatsReport* remote_report = FindNthReportByType(reports, StatsReport::kStatsReportTypeRemoteSsrc, 1); - EXPECT_TRUE(remote_report == NULL); + EXPECT_TRUE(remote_report == nullptr); } // This test verifies that audio receive streams populate stats reports diff --git a/pc/media_session.cc b/pc/media_session.cc index 85e475c0ad..12a6c0f032 100644 --- a/pc/media_session.cc +++ b/pc/media_session.cc @@ -655,7 +655,7 @@ void SetMediaProtocol(bool secure_transport, MediaContentDescription* desc) { const TransportDescription* GetTransportDescription( const std::string& content_name, const SessionDescription* current_description) { - const TransportDescription* desc = NULL; + const TransportDescription* desc = nullptr; if (current_description) { const TransportInfo* info = current_description->GetTransportInfoByName(content_name); diff --git a/pc/media_session_unittest.cc b/pc/media_session_unittest.cc index c6deaa2623..f7a7b2209f 100644 --- a/pc/media_session_unittest.cc +++ b/pc/media_session_unittest.cc @@ -827,8 +827,8 @@ TEST_F(MediaSessionDescriptionFactoryTest, ASSERT_TRUE(offer.get()); const ContentInfo* ac = offer->GetContentByName("audio"); const ContentInfo* vc = offer->GetContentByName("video"); - ASSERT_TRUE(ac != NULL); - ASSERT_TRUE(vc == NULL); + ASSERT_TRUE(ac != nullptr); + ASSERT_TRUE(vc == nullptr); EXPECT_EQ(MediaProtocolType::kRtp, ac->type); const MediaContentDescription* acd = ac->media_description(); EXPECT_EQ(webrtc::MediaType::AUDIO, acd->type()); @@ -855,8 +855,8 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateAudioOfferWithRedForOpus) { ASSERT_TRUE(offer.get()); const ContentInfo* ac = offer->GetContentByName("audio"); const ContentInfo* vc = offer->GetContentByName("video"); - ASSERT_TRUE(ac != NULL); - ASSERT_TRUE(vc == NULL); + ASSERT_TRUE(ac != nullptr); + ASSERT_TRUE(vc == nullptr); EXPECT_EQ(MediaProtocolType::kRtp, ac->type); const MediaContentDescription* acd = ac->media_description(); EXPECT_EQ(webrtc::MediaType::AUDIO, acd->type()); diff --git a/pc/media_stream.cc b/pc/media_stream.cc index 7c62bff9d0..ceafb52c31 100644 --- a/pc/media_stream.cc +++ b/pc/media_stream.cc @@ -84,7 +84,7 @@ bool MediaStream::AddTrack(TrackVector* tracks, scoped_refptr track) { template bool MediaStream::RemoveTrack(TrackVector* tracks, scoped_refptr track) { - RTC_DCHECK(tracks != NULL); + RTC_DCHECK(tracks != nullptr); if (!track) return false; typename TrackVector::iterator it = FindTrack(tracks, track->id()); diff --git a/pc/media_stream_unittest.cc b/pc/media_stream_unittest.cc index 9647080f49..3a8820664b 100644 --- a/pc/media_stream_unittest.cc +++ b/pc/media_stream_unittest.cc @@ -56,16 +56,16 @@ class MediaStreamTest : public ::testing::Test { protected: virtual void SetUp() { stream_ = MediaStream::Create(kStreamId1); - ASSERT_TRUE(stream_.get() != NULL); + ASSERT_TRUE(stream_.get() != nullptr); video_track_ = VideoTrack::Create( kVideoTrackId, FakeVideoTrackSource::Create(), Thread::Current()); - ASSERT_TRUE(video_track_.get() != NULL); + ASSERT_TRUE(video_track_.get() != nullptr); EXPECT_EQ(MediaStreamTrackInterface::kLive, video_track_->state()); audio_track_ = AudioTrack::Create(kAudioTrackId, nullptr); - ASSERT_TRUE(audio_track_.get() != NULL); + ASSERT_TRUE(audio_track_.get() != nullptr); EXPECT_EQ(MediaStreamTrackInterface::kLive, audio_track_->state()); EXPECT_TRUE(stream_->AddTrack(video_track_)); diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc index fcb2975270..829fb10ff9 100644 --- a/pc/peer_connection.cc +++ b/pc/peer_connection.cc @@ -2551,7 +2551,7 @@ bool PeerConnection::ValidateBundleSettings( for (ContentInfos::const_iterator citer = contents.begin(); citer != contents.end(); ++citer) { const ContentInfo* content = (&*citer); - RTC_DCHECK(content != NULL); + RTC_DCHECK(content != nullptr); auto it = bundle_groups_by_mid.find(content->mid()); if (it != bundle_groups_by_mid.end() && !(content->rejected || content->bundle_only) && diff --git a/pc/peer_connection_factory_unittest.cc b/pc/peer_connection_factory_unittest.cc index ec81afa1f6..816360376f 100644 --- a/pc/peer_connection_factory_unittest.cc +++ b/pc/peer_connection_factory_unittest.cc @@ -162,7 +162,7 @@ class PeerConnectionFactoryTest : public ::testing::Test { OpenH264DecoderTemplateAdapter, Dav1dDecoderTemplateAdapter>>(), nullptr /* audio_mixer */, nullptr /* audio_processing */); - ASSERT_TRUE(factory_.get() != NULL); + ASSERT_TRUE(factory_.get() != nullptr); port_allocator_ = std::make_unique(CreateEnvironment(), socket_server_.get()); raw_port_allocator_ = port_allocator_.get(); @@ -641,10 +641,10 @@ TEST_F(PeerConnectionFactoryTest, LocalRendering) { FakeFrameSource frame_source(1280, 720, kNumMicrosecsPerSec / 30); - ASSERT_TRUE(source.get() != NULL); + ASSERT_TRUE(source.get() != nullptr); scoped_refptr track( factory_->CreateVideoTrack(source, "testlabel")); - ASSERT_TRUE(track.get() != NULL); + ASSERT_TRUE(track.get() != nullptr); FakeVideoTrackRenderer local_renderer(track.get()); EXPECT_EQ(0, local_renderer.num_rendered_frames()); diff --git a/pc/peer_connection_interface_unittest.cc b/pc/peer_connection_interface_unittest.cc index d95377cb93..4e6bc33395 100644 --- a/pc/peer_connection_interface_unittest.cc +++ b/pc/peer_connection_interface_unittest.cc @@ -2064,7 +2064,7 @@ TEST_P(PeerConnectionInterfaceTest, DISABLED_TestRejectSctpDataChannelInAnswer) RTCConfiguration rtc_config; CreatePeerConnection(rtc_config); - auto offer_channel = pc_->CreateDataChannelOrError("offer_channel", NULL); + auto offer_channel = pc_->CreateDataChannelOrError("offer_channel", nullptr); CreateOfferAsLocalDescription(); @@ -2474,7 +2474,7 @@ TEST_F(PeerConnectionInterfaceTestPlanB, CloseAndTestMethods) { pc_->RemoveStream(local_stream.get()); EXPECT_FALSE(pc_->AddStream(local_stream.get())); - EXPECT_FALSE(pc_->CreateDataChannelOrError("test", NULL).ok()); + EXPECT_FALSE(pc_->CreateDataChannelOrError("test", nullptr).ok()); EXPECT_TRUE(pc_->local_description() != nullptr); EXPECT_TRUE(pc_->remote_description() != nullptr); diff --git a/pc/remote_audio_source.cc b/pc/remote_audio_source.cc index ecebff5d04..706c7f410f 100644 --- a/pc/remote_audio_source.cc +++ b/pc/remote_audio_source.cc @@ -125,13 +125,13 @@ void RemoteAudioSource::SetVolume(double volume) { } void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) { - RTC_DCHECK(observer != NULL); + RTC_DCHECK(observer != nullptr); RTC_DCHECK(!absl::c_linear_search(audio_observers_, observer)); audio_observers_.push_back(observer); } void RemoteAudioSource::UnregisterAudioObserver(AudioObserver* observer) { - RTC_DCHECK(observer != NULL); + RTC_DCHECK(observer != nullptr); audio_observers_.remove(observer); } diff --git a/pc/sctp_utils.cc b/pc/sctp_utils.cc index bc248cd2c3..edf15721ea 100644 --- a/pc/sctp_utils.cc +++ b/pc/sctp_utils.cc @@ -197,7 +197,7 @@ bool WriteDataChannelOpenMessage(const std::string& label, } } - ByteBufferWriter buffer(NULL, 20 + label.length() + protocol.length()); + ByteBufferWriter buffer(nullptr, 20 + label.length() + protocol.length()); // TODO(tommi): Add error handling and check resulting length. buffer.WriteUInt8(DATA_CHANNEL_OPEN_MESSAGE_TYPE); buffer.WriteUInt8(channel_type); diff --git a/pc/session_description.cc b/pc/session_description.cc index 96ea3d9deb..29897769cd 100644 --- a/pc/session_description.cc +++ b/pc/session_description.cc @@ -45,7 +45,7 @@ const ContentInfo* FindContentInfoByName(const ContentInfos& contents, return &(*content); } } - return NULL; + return nullptr; } const ContentInfo* FindContentInfoByType(const ContentInfos& contents, @@ -70,7 +70,7 @@ ContentGroup& ContentGroup::operator=(ContentGroup&&) = default; ContentGroup::~ContentGroup() = default; const std::string* ContentGroup::FirstContentName() const { - return (!content_names_.empty()) ? &(*content_names_.begin()) : NULL; + return (!content_names_.empty()) ? &(*content_names_.begin()) : nullptr; } bool ContentGroup::HasContentName(absl::string_view content_name) const { @@ -127,8 +127,8 @@ ContentInfo* SessionDescription::GetContentByName(const std::string& name) { const MediaContentDescription* SessionDescription::GetContentDescriptionByName( absl::string_view name) const { const ContentInfo* cinfo = FindContentInfoByName(contents_, name); - if (cinfo == NULL) { - return NULL; + if (cinfo == nullptr) { + return nullptr; } return cinfo->media_description(); @@ -137,8 +137,8 @@ const MediaContentDescription* SessionDescription::GetContentDescriptionByName( MediaContentDescription* SessionDescription::GetContentDescriptionByName( absl::string_view name) { ContentInfo* cinfo = FindContentInfoByName(&contents_, name); - if (cinfo == NULL) { - return NULL; + if (cinfo == nullptr) { + return nullptr; } return cinfo->media_description(); @@ -150,7 +150,7 @@ const ContentInfo* SessionDescription::FirstContentByType( } const ContentInfo* SessionDescription::FirstContent() const { - return (contents_.empty()) ? NULL : &(*contents_.begin()); + return (contents_.empty()) ? nullptr : &(*contents_.begin()); } void SessionDescription::AddContent( @@ -222,7 +222,7 @@ const TransportInfo* SessionDescription::GetTransportInfoByName( return &(*iter); } } - return NULL; + return nullptr; } TransportInfo* SessionDescription::GetTransportInfoByName( @@ -233,7 +233,7 @@ TransportInfo* SessionDescription::GetTransportInfoByName( return &(*iter); } } - return NULL; + return nullptr; } void SessionDescription::RemoveGroupByName(const std::string& name) { @@ -264,7 +264,7 @@ const ContentGroup* SessionDescription::GetGroupByName( return &(*iter); } } - return NULL; + return nullptr; } std::vector SessionDescription::GetGroupsByName( diff --git a/pc/test/fake_audio_capture_module_unittest.cc b/pc/test/fake_audio_capture_module_unittest.cc index 8a81e88bbc..b4249166e5 100644 --- a/pc/test/fake_audio_capture_module_unittest.cc +++ b/pc/test/fake_audio_capture_module_unittest.cc @@ -36,7 +36,7 @@ class FakeAdmTest : public ::testing::Test, public webrtc::AudioTransport { void SetUp() override { fake_audio_capture_module_ = FakeAudioCaptureModule::Create(); - EXPECT_TRUE(fake_audio_capture_module_.get() != NULL); + EXPECT_TRUE(fake_audio_capture_module_.get() != nullptr); } // Callbacks inherited from webrtc::AudioTransport. diff --git a/pc/test/peer_connection_test_wrapper.cc b/pc/test/peer_connection_test_wrapper.cc index 94f9b6530e..37877d6387 100644 --- a/pc/test/peer_connection_test_wrapper.cc +++ b/pc/test/peer_connection_test_wrapper.cc @@ -359,7 +359,7 @@ void PeerConnectionTestWrapper::AddIceCandidate(const std::string& sdp_mid, int sdp_mline_index, const std::string& candidate) { std::unique_ptr owned_candidate( - webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, candidate, NULL)); + webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, candidate, nullptr)); EXPECT_TRUE(peer_connection_->AddIceCandidate(owned_candidate.get())); } diff --git a/pc/webrtc_sdp.cc b/pc/webrtc_sdp.cc index 8b179dece9..e0fc9afdb4 100644 --- a/pc/webrtc_sdp.cc +++ b/pc/webrtc_sdp.cc @@ -848,7 +848,7 @@ std::string SdpSerialize(const JsepSessionDescription& jdesc) { desc->GetGroupsByName(GROUP_TYPE_BUNDLE); for (const ContentGroup* group : groups) { std::string group_line = kAttrGroup; - RTC_DCHECK(group != NULL); + RTC_DCHECK(group != nullptr); for (const std::string& content_name : group->content_names()) { group_line.append(" "); group_line.append(content_name); @@ -970,7 +970,7 @@ bool SdpDeserialize(absl::string_view message, bool SdpDeserializeCandidate(absl::string_view message, JsepIceCandidate* jcandidate, SdpParseError* error) { - RTC_DCHECK(jcandidate != NULL); + RTC_DCHECK(jcandidate != nullptr); Candidate candidate; if (!ParseCandidate(message, &candidate, error, true)) { return false; @@ -995,7 +995,7 @@ bool ParseCandidate(absl::string_view message, Candidate* candidate, SdpParseError* error, bool is_raw) { - RTC_DCHECK(candidate != NULL); + RTC_DCHECK(candidate != nullptr); // Makes sure `message` contains only one line. absl::string_view first_line; @@ -1847,8 +1847,8 @@ bool GetParameter(const std::string& name, void BuildRtpmap(const MediaContentDescription* media_desc, const MediaType media_type, std::string* message) { - RTC_DCHECK(message != NULL); - RTC_DCHECK(media_desc != NULL); + RTC_DCHECK(message != nullptr); + RTC_DCHECK(media_desc != nullptr); StringBuilder os; if (media_type == MediaType::VIDEO) { for (const Codec& codec : media_desc->codecs()) { @@ -2227,7 +2227,7 @@ bool ParseSessionDescription(absl::string_view message, bool ParseGroupAttribute(absl::string_view line, SessionDescription* desc, SdpParseError* error) { - RTC_DCHECK(desc != NULL); + RTC_DCHECK(desc != nullptr); // RFC 5888 and draft-holmberg-mmusic-sdp-bundle-negotiation-00 // a=group:BUNDLE video voice @@ -2622,7 +2622,7 @@ bool ParseMediaDescription( SessionDescription* desc, std::vector>* candidates, SdpParseError* error) { - RTC_DCHECK(desc != NULL); + RTC_DCHECK(desc != nullptr); int mline_index = -1; int msid_signaling = desc->msid_signaling(); @@ -2964,9 +2964,9 @@ bool ParseContent(absl::string_view message, TransportDescription* transport, std::vector>* candidates, SdpParseError* error) { - RTC_DCHECK(media_desc != NULL); - RTC_DCHECK(content_name != NULL); - RTC_DCHECK(transport != NULL); + RTC_DCHECK(media_desc != nullptr); + RTC_DCHECK(content_name != nullptr); + RTC_DCHECK(transport != nullptr); if (media_type == MediaType::AUDIO) { MaybeCreateStaticPayloadAudioCodecs(payload_types, media_desc); @@ -3377,7 +3377,7 @@ bool ParseSsrcAttribute(absl::string_view line, SsrcInfoVec* ssrc_infos, int* msid_signaling, SdpParseError* error) { - RTC_DCHECK(ssrc_infos != NULL); + RTC_DCHECK(ssrc_infos != nullptr); // RFC 5576 // a=ssrc: // a=ssrc: : @@ -3449,7 +3449,7 @@ bool ParseSsrcAttribute(absl::string_view line, bool ParseSsrcGroupAttribute(absl::string_view line, SsrcGroupVec* ssrc_groups, SdpParseError* error) { - RTC_DCHECK(ssrc_groups != NULL); + RTC_DCHECK(ssrc_groups != nullptr); // RFC 5576 // a=ssrc-group: ... std::vector fields = diff --git a/pc/webrtc_sdp_unittest.cc b/pc/webrtc_sdp_unittest.cc index 2ce4765b66..3906027e6e 100644 --- a/pc/webrtc_sdp_unittest.cc +++ b/pc/webrtc_sdp_unittest.cc @@ -836,12 +836,12 @@ static SdpType kDummyType = SdpType::kOffer; static bool SdpDeserialize(const std::string& message, JsepSessionDescription* jdesc) { - return SdpDeserialize(message, jdesc, NULL); + return SdpDeserialize(message, jdesc, nullptr); } static bool SdpDeserializeCandidate(const std::string& message, JsepIceCandidate* candidate) { - return SdpDeserializeCandidate(message, candidate, NULL); + return SdpDeserializeCandidate(message, candidate, nullptr); } // Add some extra `newlines` to the `message` after `line`. @@ -1547,7 +1547,7 @@ class WebRtcSdpTest : public ::testing::Test { void AddIceOptions(const std::string& content_name, const std::vector& transport_options) { - ASSERT_TRUE(desc_.GetTransportInfoByName(content_name) != NULL); + ASSERT_TRUE(desc_.GetTransportInfoByName(content_name) != nullptr); TransportInfo transport_info = *(desc_.GetTransportInfoByName(content_name)); desc_.RemoveTransportInfoByName(content_name); @@ -1558,7 +1558,7 @@ class WebRtcSdpTest : public ::testing::Test { void SetIceUfragPwd(const std::string& content_name, const std::string& ice_ufrag, const std::string& ice_pwd) { - ASSERT_TRUE(desc_.GetTransportInfoByName(content_name) != NULL); + ASSERT_TRUE(desc_.GetTransportInfoByName(content_name) != nullptr); TransportInfo transport_info = *(desc_.GetTransportInfoByName(content_name)); desc_.RemoveTransportInfoByName(content_name); From cb926f95719ff4de23c0144291a0b04f9cc9e4c9 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Thu, 15 May 2025 11:51:12 +0000 Subject: [PATCH 174/292] [cleanup] Replace uses of NULL with nullptr Generated using clang-tidy, tools/clang/scripts/build_clang_tools_extra.py \ --fetch out/Default clang-tidy clang-apply-replacements ninja -C out/Default gn gen out/Default --export-compile-commands cd out/Default tools/clang/third_party/llvm/clang-tools-extra/clang-tidy/tool/run-clang-tidy.py -p . \ -clang-tidy-binary out/Default/tools/clang/third_party/llvm/build/bin/clang-tidy \ -clang-apply-replacements-binary \ out/Default/tools/clang/third_party/llvm/build/bin/clang-apply-replacements \ -checks='-*,modernize-use-nullptr' \ -fix This CL was uploaded by git cl split. Bug: webrtc:42232595 Change-Id: Idfca5164dd5e02c8d8a7307a024f354c41cce4cb No-Iwyu: LSC Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391802 Reviewed-by: Harald Alvestrand Commit-Queue: Harald Alvestrand Auto-Submit: Evan Shrubsole Reviewed-by: Henrik Andreassson Commit-Queue: Henrik Andreassson Cr-Commit-Position: refs/heads/main@{#44646} --- modules/audio_device/audio_device_impl.cc | 12 +- .../audio_device/dummy/file_audio_device.cc | 14 +- .../linux/audio_device_alsa_linux.cc | 64 ++++----- .../linux/audio_device_pulse_linux.cc | 132 +++++++++--------- .../linux/audio_mixer_manager_alsa_linux.cc | 116 +++++++-------- .../linux/audio_mixer_manager_pulse_linux.cc | 40 +++--- system_wrappers/source/field_trial.cc | 4 +- 7 files changed, 192 insertions(+), 190 deletions(-) diff --git a/modules/audio_device/audio_device_impl.cc b/modules/audio_device/audio_device_impl.cc index e64e09d563..c63a478c5f 100644 --- a/modules/audio_device/audio_device_impl.cc +++ b/modules/audio_device/audio_device_impl.cc @@ -652,16 +652,16 @@ int32_t AudioDeviceModuleImpl::PlayoutDeviceName( char guid[kAdmMaxGuidSize]) { RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << index << ", ...)"; CHECKinitialized_(); - if (name == NULL) { + if (name == nullptr) { return -1; } if (audio_device_->PlayoutDeviceName(index, name, guid) == -1) { return -1; } - if (name != NULL) { + if (name != nullptr) { RTC_LOG(LS_INFO) << "output: name = " << name; } - if (guid != NULL) { + if (guid != nullptr) { RTC_LOG(LS_INFO) << "output: guid = " << guid; } return 0; @@ -673,16 +673,16 @@ int32_t AudioDeviceModuleImpl::RecordingDeviceName( char guid[kAdmMaxGuidSize]) { RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << index << ", ...)"; CHECKinitialized_(); - if (name == NULL) { + if (name == nullptr) { return -1; } if (audio_device_->RecordingDeviceName(index, name, guid) == -1) { return -1; } - if (name != NULL) { + if (name != nullptr) { RTC_LOG(LS_INFO) << "output: name = " << name; } - if (guid != NULL) { + if (guid != nullptr) { RTC_LOG(LS_INFO) << "output: guid = " << guid; } return 0; diff --git a/modules/audio_device/dummy/file_audio_device.cc b/modules/audio_device/dummy/file_audio_device.cc index 9a365047ca..7c31d6bdd3 100644 --- a/modules/audio_device/dummy/file_audio_device.cc +++ b/modules/audio_device/dummy/file_audio_device.cc @@ -32,9 +32,9 @@ const size_t kRecordingBufferSize = FileAudioDevice::FileAudioDevice(absl::string_view inputFilename, absl::string_view outputFilename) - : _ptrAudioBuffer(NULL), - _recordingBuffer(NULL), - _playoutBuffer(NULL), + : _ptrAudioBuffer(nullptr), + _recordingBuffer(nullptr), + _playoutBuffer(nullptr), _recordingFramesLeft(0), _playoutFramesLeft(0), _recordingBufferSizeIn10MS(0), @@ -212,7 +212,7 @@ int32_t FileAudioDevice::StartPlayout() { RTC_LOG(LS_ERROR) << "Failed to open playout file: " << _outputFilename; _playing = false; delete[] _playoutBuffer; - _playoutBuffer = NULL; + _playoutBuffer = nullptr; return -1; } } @@ -244,7 +244,7 @@ int32_t FileAudioDevice::StopPlayout() { _playoutFramesLeft = 0; delete[] _playoutBuffer; - _playoutBuffer = NULL; + _playoutBuffer = nullptr; _outputFile.Close(); RTC_LOG(LS_INFO) << "Stopped playout capture to output file: " @@ -273,7 +273,7 @@ int32_t FileAudioDevice::StartRecording() { << _inputFilename; _recording = false; delete[] _recordingBuffer; - _recordingBuffer = NULL; + _recordingBuffer = nullptr; return -1; } } @@ -304,7 +304,7 @@ int32_t FileAudioDevice::StopRecording() { _recordingFramesLeft = 0; if (_recordingBuffer) { delete[] _recordingBuffer; - _recordingBuffer = NULL; + _recordingBuffer = nullptr; } _inputFile.Close(); diff --git a/modules/audio_device/linux/audio_device_alsa_linux.cc b/modules/audio_device/linux/audio_device_alsa_linux.cc index f5a3189371..e4895392cf 100644 --- a/modules/audio_device/linux/audio_device_alsa_linux.cc +++ b/modules/audio_device/linux/audio_device_alsa_linux.cc @@ -65,13 +65,13 @@ static const unsigned int ALSA_CAPTURE_WAIT_TIMEOUT = 5; // in ms #define FUNC_GET_DEVICE_NAME_FOR_AN_ENUM 2 AudioDeviceLinuxALSA::AudioDeviceLinuxALSA() - : _ptrAudioBuffer(NULL), + : _ptrAudioBuffer(nullptr), _inputDeviceIndex(0), _outputDeviceIndex(0), _inputDeviceIsSpecified(false), _outputDeviceIsSpecified(false), - _handleRecord(NULL), - _handlePlayout(NULL), + _handleRecord(nullptr), + _handlePlayout(nullptr), _recordingBuffersizeInFrame(0), _recordingPeriodSizeInFrame(0), _playoutBufferSizeInFrame(0), @@ -84,8 +84,8 @@ AudioDeviceLinuxALSA::AudioDeviceLinuxALSA() _playoutFreq(ALSA_PLAYOUT_FREQ), _recChannels(ALSA_CAPTURE_CH), _playChannels(ALSA_PLAYOUT_CH), - _recordingBuffer(NULL), - _playoutBuffer(NULL), + _recordingBuffer(nullptr), + _playoutBuffer(nullptr), _recordingFramesLeft(0), _playoutFramesLeft(0), _initialized(false), @@ -111,11 +111,11 @@ AudioDeviceLinuxALSA::~AudioDeviceLinuxALSA() { // Clean up the recording buffer and playout buffer. if (_recordingBuffer) { delete[] _recordingBuffer; - _recordingBuffer = NULL; + _recordingBuffer = nullptr; } if (_playoutBuffer) { delete[] _playoutBuffer; - _playoutBuffer = NULL; + _playoutBuffer = nullptr; } } @@ -154,7 +154,7 @@ AudioDeviceGeneric::InitStatus AudioDeviceLinuxALSA::Init() { } #if defined(WEBRTC_USE_X11) // Get X display handle for typing detection - _XDisplay = XOpenDisplay(NULL); + _XDisplay = XOpenDisplay(nullptr); if (!_XDisplay) { RTC_LOG(LS_WARNING) << "failed to open X display, typing detection will not work"; @@ -186,7 +186,7 @@ int32_t AudioDeviceLinuxALSA::Terminate() { #if defined(WEBRTC_USE_X11) if (_XDisplay) { XCloseDisplay(_XDisplay); - _XDisplay = NULL; + _XDisplay = nullptr; } #endif _initialized = false; @@ -616,13 +616,13 @@ int32_t AudioDeviceLinuxALSA::PlayoutDeviceName( char guid[kAdmMaxGuidSize]) { const uint16_t nDevices(PlayoutDevices()); - if ((index > (nDevices - 1)) || (name == NULL)) { + if ((index > (nDevices - 1)) || (name == nullptr)) { return -1; } memset(name, 0, kAdmMaxDeviceNameSize); - if (guid != NULL) { + if (guid != nullptr) { memset(guid, 0, kAdmMaxGuidSize); } @@ -635,13 +635,13 @@ int32_t AudioDeviceLinuxALSA::RecordingDeviceName( char guid[kAdmMaxGuidSize]) { const uint16_t nDevices(RecordingDevices()); - if ((index > (nDevices - 1)) || (name == NULL)) { + if ((index > (nDevices - 1)) || (name == nullptr)) { return -1; } memset(name, 0, kAdmMaxDeviceNameSize); - if (guid != NULL) { + if (guid != nullptr) { memset(guid, 0, kAdmMaxGuidSize); } @@ -759,9 +759,9 @@ int32_t AudioDeviceLinuxALSA::InitPlayoutLocked() { // Start by closing any existing wave-output devices // - if (_handlePlayout != NULL) { + if (_handlePlayout != nullptr) { LATE(snd_pcm_close)(_handlePlayout); - _handlePlayout = NULL; + _handlePlayout = nullptr; _playIsInitialized = false; if (errVal < 0) { RTC_LOG(LS_ERROR) << "Error closing current playout sound device, error: " @@ -793,7 +793,7 @@ int32_t AudioDeviceLinuxALSA::InitPlayoutLocked() { if (errVal < 0) { RTC_LOG(LS_ERROR) << "unable to open playback device: " << LATE(snd_strerror)(errVal) << " (" << errVal << ")"; - _handlePlayout = NULL; + _handlePlayout = nullptr; return -1; } @@ -817,7 +817,7 @@ int32_t AudioDeviceLinuxALSA::InitPlayoutLocked() { << LATE(snd_strerror)(errVal) << " (" << errVal << ")"; ErrorRecovery(errVal, _handlePlayout); errVal = LATE(snd_pcm_close)(_handlePlayout); - _handlePlayout = NULL; + _handlePlayout = nullptr; return -1; } @@ -846,7 +846,7 @@ int32_t AudioDeviceLinuxALSA::InitPlayoutLocked() { // Init varaibles used for play - if (_handlePlayout != NULL) { + if (_handlePlayout != nullptr) { _playIsInitialized = true; return 0; } else { @@ -881,9 +881,9 @@ int32_t AudioDeviceLinuxALSA::InitRecordingLocked() { // Start by closing any existing pcm-input devices // - if (_handleRecord != NULL) { + if (_handleRecord != nullptr) { errVal = LATE(snd_pcm_close)(_handleRecord); - _handleRecord = NULL; + _handleRecord = nullptr; _recIsInitialized = false; if (errVal < 0) { RTC_LOG(LS_ERROR) @@ -917,7 +917,7 @@ int32_t AudioDeviceLinuxALSA::InitRecordingLocked() { if (errVal < 0) { RTC_LOG(LS_ERROR) << "unable to open record device: " << LATE(snd_strerror)(errVal); - _handleRecord = NULL; + _handleRecord = nullptr; return -1; } @@ -959,7 +959,7 @@ int32_t AudioDeviceLinuxALSA::InitRecordingLocked() { << LATE(snd_strerror)(errVal) << " (" << errVal << ")"; ErrorRecovery(errVal, _handleRecord); errVal = LATE(snd_pcm_close)(_handleRecord); - _handleRecord = NULL; + _handleRecord = nullptr; return -1; } } @@ -987,7 +987,7 @@ int32_t AudioDeviceLinuxALSA::InitRecordingLocked() { _recordingBufferSizeIn10MS = LATE(snd_pcm_frames_to_bytes)(_handleRecord, _recordingFramesIn10MS); - if (_handleRecord != NULL) { + if (_handleRecord != nullptr) { // Mark recording side as initialized _recIsInitialized = true; return 0; @@ -1062,7 +1062,7 @@ int32_t AudioDeviceLinuxALSA::StopRecordingLocked() { return 0; } - if (_handleRecord == NULL) { + if (_handleRecord == nullptr) { return -1; } @@ -1075,7 +1075,7 @@ int32_t AudioDeviceLinuxALSA::StopRecordingLocked() { _recordingFramesLeft = 0; if (_recordingBuffer) { delete[] _recordingBuffer; - _recordingBuffer = NULL; + _recordingBuffer = nullptr; } // Stop and close pcm recording device. @@ -1100,7 +1100,7 @@ int32_t AudioDeviceLinuxALSA::StopRecordingLocked() { } // set the pcm input handle to NULL - _handleRecord = NULL; + _handleRecord = nullptr; return 0; } @@ -1167,7 +1167,7 @@ int32_t AudioDeviceLinuxALSA::StopPlayoutLocked() { return 0; } - if (_handlePlayout == NULL) { + if (_handlePlayout == nullptr) { return -1; } @@ -1178,7 +1178,7 @@ int32_t AudioDeviceLinuxALSA::StopPlayoutLocked() { _playoutFramesLeft = 0; delete[] _playoutBuffer; - _playoutBuffer = NULL; + _playoutBuffer = nullptr; // stop and close pcm playout device int errVal = LATE(snd_pcm_drop)(_handlePlayout); @@ -1193,7 +1193,7 @@ int32_t AudioDeviceLinuxALSA::StopPlayoutLocked() { // set the pcm input handle to NULL _playIsInitialized = false; - _handlePlayout = NULL; + _handlePlayout = nullptr; RTC_LOG(LS_VERBOSE) << "handle_playout is now set to NULL"; return 0; @@ -1261,7 +1261,7 @@ int32_t AudioDeviceLinuxALSA::GetDevicesInfo(const int32_t function, return 0; } - for (void** list = hints; *list != NULL; ++list) { + for (void** list = hints; *list != nullptr; ++list) { char* actualType = LATE(snd_device_name_get_hint)(*list, "IOID"); if (actualType) { // NULL means it's both. bool wrongType = (strcmp(actualType, type) != 0); @@ -1351,7 +1351,7 @@ int32_t AudioDeviceLinuxALSA::GetDevicesInfo(const int32_t function, } int32_t AudioDeviceLinuxALSA::InputSanityCheckAfterUnlockedPeriod() const { - if (_handleRecord == NULL) { + if (_handleRecord == nullptr) { RTC_LOG(LS_ERROR) << "input state has been modified during unlocked period"; return -1; } @@ -1359,7 +1359,7 @@ int32_t AudioDeviceLinuxALSA::InputSanityCheckAfterUnlockedPeriod() const { } int32_t AudioDeviceLinuxALSA::OutputSanityCheckAfterUnlockedPeriod() const { - if (_handlePlayout == NULL) { + if (_handlePlayout == nullptr) { RTC_LOG(LS_ERROR) << "output state has been modified during unlocked period"; return -1; diff --git a/modules/audio_device/linux/audio_device_pulse_linux.cc b/modules/audio_device/linux/audio_device_pulse_linux.cc index 8843fe727e..7acfc5d9b9 100644 --- a/modules/audio_device/linux/audio_device_pulse_linux.cc +++ b/modules/audio_device/linux/audio_device_pulse_linux.cc @@ -33,7 +33,7 @@ WebRTCPulseSymbolTable* GetPulseSymbolTable() { namespace webrtc { AudioDeviceLinuxPulse::AudioDeviceLinuxPulse() - : _ptrAudioBuffer(NULL), + : _ptrAudioBuffer(nullptr), _inputDeviceIndex(0), _outputDeviceIndex(0), _inputDeviceIsSpecified(false), @@ -55,28 +55,28 @@ AudioDeviceLinuxPulse::AudioDeviceLinuxPulse() _deviceIndex(-1), _numPlayDevices(0), _numRecDevices(0), - _playDeviceName(NULL), - _recDeviceName(NULL), - _playDisplayDeviceName(NULL), - _recDisplayDeviceName(NULL), - _playBuffer(NULL), + _playDeviceName(nullptr), + _recDeviceName(nullptr), + _playDisplayDeviceName(nullptr), + _recDisplayDeviceName(nullptr), + _playBuffer(nullptr), _playbackBufferSize(0), _playbackBufferUnused(0), _tempBufferSpace(0), - _recBuffer(NULL), + _recBuffer(nullptr), _recordBufferSize(0), _recordBufferUsed(0), - _tempSampleData(NULL), + _tempSampleData(nullptr), _tempSampleDataSize(0), _configuredLatencyPlay(0), _configuredLatencyRec(0), _paDeviceIndex(-1), _paStateChanged(false), - _paMainloop(NULL), - _paMainloopApi(NULL), - _paContext(NULL), - _recStream(NULL), - _playStream(NULL), + _paMainloop(nullptr), + _paMainloopApi(nullptr), + _paContext(nullptr), + _recStream(nullptr), + _playStream(nullptr), _recStreamFlags(0), _playStreamFlags(0) { RTC_DLOG(LS_INFO) << __FUNCTION__ << " created"; @@ -94,19 +94,19 @@ AudioDeviceLinuxPulse::~AudioDeviceLinuxPulse() { if (_recBuffer) { delete[] _recBuffer; - _recBuffer = NULL; + _recBuffer = nullptr; } if (_playBuffer) { delete[] _playBuffer; - _playBuffer = NULL; + _playBuffer = nullptr; } if (_playDeviceName) { delete[] _playDeviceName; - _playDeviceName = NULL; + _playDeviceName = nullptr; } if (_recDeviceName) { delete[] _recDeviceName; - _recDeviceName = NULL; + _recDeviceName = nullptr; } } @@ -151,7 +151,7 @@ AudioDeviceGeneric::InitStatus AudioDeviceLinuxPulse::Init() { #if defined(WEBRTC_USE_X11) // Get X display handle for typing detection - _XDisplay = XOpenDisplay(NULL); + _XDisplay = XOpenDisplay(nullptr); if (!_XDisplay) { RTC_LOG(LS_WARNING) << "failed to open X display, typing detection will not work"; @@ -208,7 +208,7 @@ int32_t AudioDeviceLinuxPulse::Terminate() { #if defined(WEBRTC_USE_X11) if (_XDisplay) { XCloseDisplay(_XDisplay); - _XDisplay = NULL; + _XDisplay = nullptr; } #endif @@ -238,7 +238,7 @@ int32_t AudioDeviceLinuxPulse::InitSpeaker() { // check if default device if (_outputDeviceIndex == 0) { uint16_t deviceIndex = 0; - GetDefaultDeviceInfo(false, NULL, deviceIndex); + GetDefaultDeviceInfo(false, nullptr, deviceIndex); _paDeviceIndex = deviceIndex; } else { // get the PA device index from @@ -275,7 +275,7 @@ int32_t AudioDeviceLinuxPulse::InitMicrophone() { // Check if default device if (_inputDeviceIndex == 0) { uint16_t deviceIndex = 0; - GetDefaultDeviceInfo(true, NULL, deviceIndex); + GetDefaultDeviceInfo(true, nullptr, deviceIndex); _paDeviceIndex = deviceIndex; } else { // Get the PA device index from @@ -642,7 +642,7 @@ int32_t AudioDeviceLinuxPulse::MinMicrophoneVolume(uint32_t& minVolume) const { int16_t AudioDeviceLinuxPulse::PlayoutDevices() { PaLock(); - pa_operation* paOperation = NULL; + pa_operation* paOperation = nullptr; _numPlayDevices = 1; // init to 1 to account for "default" // get the whole list of devices and update _numPlayDevices @@ -691,13 +691,13 @@ int32_t AudioDeviceLinuxPulse::PlayoutDeviceName( RTC_DCHECK(thread_checker_.IsCurrent()); const uint16_t nDevices = PlayoutDevices(); - if ((index > (nDevices - 1)) || (name == NULL)) { + if ((index > (nDevices - 1)) || (name == nullptr)) { return -1; } memset(name, 0, kAdmMaxDeviceNameSize); - if (guid != NULL) { + if (guid != nullptr) { memset(guid, 0, kAdmMaxGuidSize); } @@ -716,7 +716,7 @@ int32_t AudioDeviceLinuxPulse::PlayoutDeviceName( PlayoutDevices(); // clear device name and index - _playDisplayDeviceName = NULL; + _playDisplayDeviceName = nullptr; _deviceIndex = -1; return 0; @@ -729,13 +729,13 @@ int32_t AudioDeviceLinuxPulse::RecordingDeviceName( RTC_DCHECK(thread_checker_.IsCurrent()); const uint16_t nDevices(RecordingDevices()); - if ((index > (nDevices - 1)) || (name == NULL)) { + if ((index > (nDevices - 1)) || (name == nullptr)) { return -1; } memset(name, 0, kAdmMaxDeviceNameSize); - if (guid != NULL) { + if (guid != nullptr) { memset(guid, 0, kAdmMaxGuidSize); } @@ -754,7 +754,7 @@ int32_t AudioDeviceLinuxPulse::RecordingDeviceName( RecordingDevices(); // Clear device name and index - _recDisplayDeviceName = NULL; + _recDisplayDeviceName = nullptr; _deviceIndex = -1; return 0; @@ -763,7 +763,7 @@ int32_t AudioDeviceLinuxPulse::RecordingDeviceName( int16_t AudioDeviceLinuxPulse::RecordingDevices() { PaLock(); - pa_operation* paOperation = NULL; + pa_operation* paOperation = nullptr; _numRecDevices = 1; // Init to 1 to account for "default" // Get the whole list of devices and update _numRecDevices @@ -869,7 +869,7 @@ int32_t AudioDeviceLinuxPulse::InitPlayout() { { MutexLock lock(&mutex_); _playStream = - LATE(pa_stream_new)(_paContext, "playStream", &playSampleSpec, NULL); + LATE(pa_stream_new)(_paContext, "playStream", &playSampleSpec, nullptr); } if (!_playStream) { @@ -976,7 +976,7 @@ int32_t AudioDeviceLinuxPulse::InitRecording() { // Create a new rec stream _recStream = - LATE(pa_stream_new)(_paContext, "recStream", &recSampleSpec, NULL); + LATE(pa_stream_new)(_paContext, "recStream", &recSampleSpec, nullptr); if (!_recStream) { RTC_LOG(LS_ERROR) << "failed to create rec stream, err=" << LATE(pa_context_errno)(_paContext); @@ -1091,7 +1091,7 @@ int32_t AudioDeviceLinuxPulse::StopRecording() { return 0; } - if (_recStream == NULL) { + if (_recStream == nullptr) { return -1; } @@ -1104,10 +1104,10 @@ int32_t AudioDeviceLinuxPulse::StopRecording() { PaLock(); DisableReadCallback(); - LATE(pa_stream_set_overflow_callback)(_recStream, NULL, NULL); + LATE(pa_stream_set_overflow_callback)(_recStream, nullptr, nullptr); // Unset this here so that we don't get a TERMINATED callback - LATE(pa_stream_set_state_callback)(_recStream, NULL, NULL); + LATE(pa_stream_set_state_callback)(_recStream, nullptr, nullptr); if (LATE(pa_stream_get_state)(_recStream) != PA_STREAM_UNCONNECTED) { // Disconnect the stream @@ -1122,7 +1122,7 @@ int32_t AudioDeviceLinuxPulse::StopRecording() { } LATE(pa_stream_unref)(_recStream); - _recStream = NULL; + _recStream = nullptr; PaUnLock(); @@ -1131,7 +1131,7 @@ int32_t AudioDeviceLinuxPulse::StopRecording() { if (_recBuffer) { delete[] _recBuffer; - _recBuffer = NULL; + _recBuffer = nullptr; } return 0; @@ -1206,7 +1206,7 @@ int32_t AudioDeviceLinuxPulse::StopPlayout() { return 0; } - if (_playStream == NULL) { + if (_playStream == nullptr) { return -1; } @@ -1220,10 +1220,10 @@ int32_t AudioDeviceLinuxPulse::StopPlayout() { PaLock(); DisableWriteCallback(); - LATE(pa_stream_set_underflow_callback)(_playStream, NULL, NULL); + LATE(pa_stream_set_underflow_callback)(_playStream, nullptr, nullptr); // Unset this here so that we don't get a TERMINATED callback - LATE(pa_stream_set_state_callback)(_playStream, NULL, NULL); + LATE(pa_stream_set_state_callback)(_playStream, nullptr, nullptr); if (LATE(pa_stream_get_state)(_playStream) != PA_STREAM_UNCONNECTED) { // Disconnect the stream @@ -1238,7 +1238,7 @@ int32_t AudioDeviceLinuxPulse::StopPlayout() { } LATE(pa_stream_unref)(_playStream); - _playStream = NULL; + _playStream = nullptr; PaUnLock(); @@ -1247,7 +1247,7 @@ int32_t AudioDeviceLinuxPulse::StopPlayout() { if (_playBuffer) { delete[] _playBuffer; - _playBuffer = NULL; + _playBuffer = nullptr; } return 0; @@ -1437,7 +1437,7 @@ void AudioDeviceLinuxPulse::PaStreamStateCallbackHandler(pa_stream* p) { int32_t AudioDeviceLinuxPulse::CheckPulseAudioVersion() { PaLock(); - pa_operation* paOperation = NULL; + pa_operation* paOperation = nullptr; // get the server info and update deviceName paOperation = @@ -1455,7 +1455,7 @@ int32_t AudioDeviceLinuxPulse::CheckPulseAudioVersion() { int32_t AudioDeviceLinuxPulse::InitSamplingFrequency() { PaLock(); - pa_operation* paOperation = NULL; + pa_operation* paOperation = nullptr; // Get the server info and update sample_rate_hz_ paOperation = @@ -1474,7 +1474,7 @@ int32_t AudioDeviceLinuxPulse::GetDefaultDeviceInfo(bool recDevice, char tmpName[kAdmMaxDeviceNameSize] = {0}; // subtract length of "default: " uint16_t nameLen = kAdmMaxDeviceNameSize - 9; - char* pName = NULL; + char* pName = nullptr; if (name) { // Add "default: " @@ -1498,7 +1498,7 @@ int32_t AudioDeviceLinuxPulse::GetDefaultDeviceInfo(bool recDevice, PaLock(); - pa_operation* paOperation = NULL; + pa_operation* paOperation = nullptr; // Get the server info and update deviceName paOperation = @@ -1528,8 +1528,8 @@ int32_t AudioDeviceLinuxPulse::GetDefaultDeviceInfo(bool recDevice, } // Clear members - _playDisplayDeviceName = NULL; - _recDisplayDeviceName = NULL; + _playDisplayDeviceName = nullptr; + _recDisplayDeviceName = nullptr; _paDeviceIndex = -1; _deviceIndex = -1; _numPlayDevices = 0; @@ -1598,8 +1598,8 @@ int32_t AudioDeviceLinuxPulse::InitPulseAudio() { // Connect the context to a server (default) _paStateChanged = false; - retVal = - LATE(pa_context_connect)(_paContext, NULL, PA_CONTEXT_NOAUTOSPAWN, NULL); + retVal = LATE(pa_context_connect)(_paContext, nullptr, PA_CONTEXT_NOAUTOSPAWN, + nullptr); if (retVal != PA_OK) { RTC_LOG(LS_ERROR) << "failed to connect context, error=" << retVal; @@ -1671,7 +1671,7 @@ int32_t AudioDeviceLinuxPulse::TerminatePulseAudio() { } PaUnLock(); - _paContext = NULL; + _paContext = nullptr; // Stop the threaded main loop if (_paMainloop) { @@ -1683,7 +1683,7 @@ int32_t AudioDeviceLinuxPulse::TerminatePulseAudio() { LATE(pa_threaded_mainloop_free)(_paMainloop); } - _paMainloop = NULL; + _paMainloop = nullptr; RTC_LOG(LS_VERBOSE) << "PulseAudio terminated"; @@ -1733,7 +1733,7 @@ void AudioDeviceLinuxPulse::EnableWriteCallback() { } void AudioDeviceLinuxPulse::DisableWriteCallback() { - LATE(pa_stream_set_write_callback)(_playStream, NULL, NULL); + LATE(pa_stream_set_write_callback)(_playStream, nullptr, nullptr); } void AudioDeviceLinuxPulse::PaStreamWriteCallback(pa_stream* /*unused*/, @@ -1789,7 +1789,7 @@ void AudioDeviceLinuxPulse::PaStreamUnderflowCallbackHandler() { _playBufferAttr.prebuf = _playBufferAttr.tlength - _playBufferAttr.minreq; pa_operation* op = LATE(pa_stream_set_buffer_attr)( - _playStream, &_playBufferAttr, NULL, NULL); + _playStream, &_playBufferAttr, nullptr, nullptr); if (!op) { RTC_LOG(LS_ERROR) << "pa_stream_set_buffer_attr()"; return; @@ -1807,7 +1807,7 @@ void AudioDeviceLinuxPulse::EnableReadCallback() { } void AudioDeviceLinuxPulse::DisableReadCallback() { - LATE(pa_stream_set_read_callback)(_recStream, NULL, NULL); + LATE(pa_stream_set_read_callback)(_recStream, nullptr, nullptr); } void AudioDeviceLinuxPulse::PaStreamReadCallback(pa_stream* /*unused1*/, @@ -1991,7 +1991,7 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess() { RTC_LOG(LS_VERBOSE) << "_startPlay true, performing initial actions"; _startPlay = false; - _playDeviceName = NULL; + _playDeviceName = nullptr; // Set if not default device if (_outputDeviceIndex > 0) { @@ -2021,7 +2021,7 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess() { PaLock(); // NULL gives PA the choice of startup volume. - pa_cvolume* ptr_cvolume = NULL; + pa_cvolume* ptr_cvolume = nullptr; if (update_speaker_volume_at_startup_) { pa_cvolume cVolumes; ptr_cvolume = &cVolumes; @@ -2033,9 +2033,10 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess() { } // Connect the stream to a sink - if (LATE(pa_stream_connect_playback)( - _playStream, _playDeviceName, &_playBufferAttr, - (pa_stream_flags_t)_playStreamFlags, ptr_cvolume, NULL) != PA_OK) { + if (LATE(pa_stream_connect_playback)(_playStream, _playDeviceName, + &_playBufferAttr, + (pa_stream_flags_t)_playStreamFlags, + ptr_cvolume, nullptr) != PA_OK) { RTC_LOG(LS_ERROR) << "failed to connect play stream, err=" << LATE(pa_context_errno)(_paContext); } @@ -2057,7 +2058,7 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess() { // Clear device name if (_playDeviceName) { delete[] _playDeviceName; - _playDeviceName = NULL; + _playDeviceName = nullptr; } _playing = true; @@ -2081,7 +2082,7 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess() { PaLock(); if (LATE(pa_stream_write)( _playStream, (void*)&_playBuffer[_playbackBufferUnused], write, - NULL, (int64_t)0, PA_SEEK_RELATIVE) != PA_OK) { + nullptr, (int64_t)0, PA_SEEK_RELATIVE) != PA_OK) { _writeErrors++; if (_writeErrors > 10) { RTC_LOG(LS_ERROR) << "Playout error: _writeErrors=" << _writeErrors @@ -2125,7 +2126,8 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess() { RTC_LOG(LS_VERBOSE) << "will write"; PaLock(); if (LATE(pa_stream_write)(_playStream, (void*)&_playBuffer[0], write, - NULL, (int64_t)0, PA_SEEK_RELATIVE) != PA_OK) { + nullptr, (int64_t)0, + PA_SEEK_RELATIVE) != PA_OK) { _writeErrors++; if (_writeErrors > 10) { RTC_LOG(LS_ERROR) << "Playout error: _writeErrors=" << _writeErrors @@ -2160,7 +2162,7 @@ bool AudioDeviceLinuxPulse::RecThreadProcess() { if (_startRec) { RTC_LOG(LS_VERBOSE) << "_startRec true, performing initial actions"; - _recDeviceName = NULL; + _recDeviceName = nullptr; // Set if not default device if (_inputDeviceIndex > 0) { @@ -2199,7 +2201,7 @@ bool AudioDeviceLinuxPulse::RecThreadProcess() { // Clear device name if (_recDeviceName) { delete[] _recDeviceName; - _recDeviceName = NULL; + _recDeviceName = nullptr; } _startRec = false; @@ -2215,7 +2217,7 @@ bool AudioDeviceLinuxPulse::RecThreadProcess() { return true; } - _tempSampleData = NULL; + _tempSampleData = nullptr; _tempSampleDataSize = 0; PaLock(); diff --git a/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc b/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc index e7e7033173..76084c6bda 100644 --- a/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc +++ b/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc @@ -23,10 +23,10 @@ namespace webrtc { AudioMixerManagerLinuxALSA::AudioMixerManagerLinuxALSA() - : _outputMixerHandle(NULL), - _inputMixerHandle(NULL), - _outputMixerElement(NULL), - _inputMixerElement(NULL) { + : _outputMixerHandle(nullptr), + _inputMixerHandle(nullptr), + _outputMixerElement(nullptr), + _inputMixerElement(nullptr) { RTC_DLOG(LS_INFO) << __FUNCTION__ << " created"; memset(_outputMixerStr, 0, kAdmMaxDeviceNameSize); @@ -63,7 +63,7 @@ int32_t AudioMixerManagerLinuxALSA::CloseSpeakerLocked() { int errVal = 0; - if (_outputMixerHandle != NULL) { + if (_outputMixerHandle != nullptr) { RTC_LOG(LS_VERBOSE) << "Closing playout mixer"; LATE(snd_mixer_free)(_outputMixerHandle); if (errVal < 0) { @@ -80,8 +80,8 @@ int32_t AudioMixerManagerLinuxALSA::CloseSpeakerLocked() { RTC_LOG(LS_ERROR) << "Error snd_mixer_close(handleMixer) errVal=" << errVal; } - _outputMixerHandle = NULL; - _outputMixerElement = NULL; + _outputMixerHandle = nullptr; + _outputMixerElement = nullptr; } memset(_outputMixerStr, 0, kAdmMaxDeviceNameSize); @@ -98,7 +98,7 @@ int32_t AudioMixerManagerLinuxALSA::CloseMicrophoneLocked() { int errVal = 0; - if (_inputMixerHandle != NULL) { + if (_inputMixerHandle != nullptr) { RTC_LOG(LS_VERBOSE) << "Closing record mixer"; LATE(snd_mixer_free)(_inputMixerHandle); @@ -122,8 +122,8 @@ int32_t AudioMixerManagerLinuxALSA::CloseMicrophoneLocked() { } RTC_LOG(LS_VERBOSE) << "Closing record mixer 4"; - _inputMixerHandle = NULL; - _inputMixerElement = NULL; + _inputMixerHandle = nullptr; + _inputMixerElement = nullptr; } memset(_inputMixerStr, 0, kAdmMaxDeviceNameSize); @@ -140,7 +140,7 @@ int32_t AudioMixerManagerLinuxALSA::OpenSpeaker(char* deviceName) { // Close any existing output mixer handle // - if (_outputMixerHandle != NULL) { + if (_outputMixerHandle != nullptr) { RTC_LOG(LS_VERBOSE) << "Closing playout mixer"; LATE(snd_mixer_free)(_outputMixerHandle); @@ -159,8 +159,8 @@ int32_t AudioMixerManagerLinuxALSA::OpenSpeaker(char* deviceName) { << errVal; } } - _outputMixerHandle = NULL; - _outputMixerElement = NULL; + _outputMixerHandle = nullptr; + _outputMixerElement = nullptr; errVal = LATE(snd_mixer_open)(&_outputMixerHandle, 0); if (errVal < 0) { @@ -178,18 +178,18 @@ int32_t AudioMixerManagerLinuxALSA::OpenSpeaker(char* deviceName) { if (errVal < 0) { RTC_LOG(LS_ERROR) << "snd_mixer_attach(_outputMixerHandle, " << controlName << ") error: " << LATE(snd_strerror)(errVal); - _outputMixerHandle = NULL; + _outputMixerHandle = nullptr; return -1; } strcpy(_outputMixerStr, controlName); - errVal = LATE(snd_mixer_selem_register)(_outputMixerHandle, NULL, NULL); + errVal = LATE(snd_mixer_selem_register)(_outputMixerHandle, nullptr, nullptr); if (errVal < 0) { RTC_LOG(LS_ERROR) << "snd_mixer_selem_register(_outputMixerHandle, NULL, NULL), " "error: " << LATE(snd_strerror)(errVal); - _outputMixerHandle = NULL; + _outputMixerHandle = nullptr; return -1; } @@ -198,7 +198,7 @@ int32_t AudioMixerManagerLinuxALSA::OpenSpeaker(char* deviceName) { return -1; } - if (_outputMixerHandle != NULL) { + if (_outputMixerHandle != nullptr) { RTC_LOG(LS_VERBOSE) << "the output mixer device is now open (" << _outputMixerHandle << ")"; } @@ -216,7 +216,7 @@ int32_t AudioMixerManagerLinuxALSA::OpenMicrophone(char* deviceName) { // Close any existing input mixer handle // - if (_inputMixerHandle != NULL) { + if (_inputMixerHandle != nullptr) { RTC_LOG(LS_VERBOSE) << "Closing record mixer"; LATE(snd_mixer_free)(_inputMixerHandle); @@ -240,8 +240,8 @@ int32_t AudioMixerManagerLinuxALSA::OpenMicrophone(char* deviceName) { } RTC_LOG(LS_VERBOSE) << "Closing record mixer"; } - _inputMixerHandle = NULL; - _inputMixerElement = NULL; + _inputMixerHandle = nullptr; + _inputMixerElement = nullptr; errVal = LATE(snd_mixer_open)(&_inputMixerHandle, 0); if (errVal < 0) { @@ -260,19 +260,19 @@ int32_t AudioMixerManagerLinuxALSA::OpenMicrophone(char* deviceName) { RTC_LOG(LS_ERROR) << "snd_mixer_attach(_inputMixerHandle, " << controlName << ") error: " << LATE(snd_strerror)(errVal); - _inputMixerHandle = NULL; + _inputMixerHandle = nullptr; return -1; } strcpy(_inputMixerStr, controlName); - errVal = LATE(snd_mixer_selem_register)(_inputMixerHandle, NULL, NULL); + errVal = LATE(snd_mixer_selem_register)(_inputMixerHandle, nullptr, nullptr); if (errVal < 0) { RTC_LOG(LS_ERROR) << "snd_mixer_selem_register(_inputMixerHandle, NULL, NULL), " "error: " << LATE(snd_strerror)(errVal); - _inputMixerHandle = NULL; + _inputMixerHandle = nullptr; return -1; } // Load and find the proper mixer element @@ -280,7 +280,7 @@ int32_t AudioMixerManagerLinuxALSA::OpenMicrophone(char* deviceName) { return -1; } - if (_inputMixerHandle != NULL) { + if (_inputMixerHandle != nullptr) { RTC_LOG(LS_VERBOSE) << "the input mixer device is now open (" << _inputMixerHandle << ")"; } @@ -291,13 +291,13 @@ int32_t AudioMixerManagerLinuxALSA::OpenMicrophone(char* deviceName) { bool AudioMixerManagerLinuxALSA::SpeakerIsInitialized() const { RTC_DLOG(LS_INFO) << __FUNCTION__; - return (_outputMixerHandle != NULL); + return (_outputMixerHandle != nullptr); } bool AudioMixerManagerLinuxALSA::MicrophoneIsInitialized() const { RTC_DLOG(LS_INFO) << __FUNCTION__; - return (_inputMixerHandle != NULL); + return (_inputMixerHandle != nullptr); } int32_t AudioMixerManagerLinuxALSA::SetSpeakerVolume(uint32_t volume) { @@ -306,7 +306,7 @@ int32_t AudioMixerManagerLinuxALSA::SetSpeakerVolume(uint32_t volume) { MutexLock lock(&mutex_); - if (_outputMixerElement == NULL) { + if (_outputMixerElement == nullptr) { RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists"; return -1; } @@ -323,7 +323,7 @@ int32_t AudioMixerManagerLinuxALSA::SetSpeakerVolume(uint32_t volume) { } int32_t AudioMixerManagerLinuxALSA::SpeakerVolume(uint32_t& volume) const { - if (_outputMixerElement == NULL) { + if (_outputMixerElement == nullptr) { RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists"; return -1; } @@ -347,7 +347,7 @@ int32_t AudioMixerManagerLinuxALSA::SpeakerVolume(uint32_t& volume) const { int32_t AudioMixerManagerLinuxALSA::MaxSpeakerVolume( uint32_t& maxVolume) const { - if (_outputMixerElement == NULL) { + if (_outputMixerElement == nullptr) { RTC_LOG(LS_WARNING) << "no avilable output mixer element exists"; return -1; } @@ -373,7 +373,7 @@ int32_t AudioMixerManagerLinuxALSA::MaxSpeakerVolume( int32_t AudioMixerManagerLinuxALSA::MinSpeakerVolume( uint32_t& minVolume) const { - if (_outputMixerElement == NULL) { + if (_outputMixerElement == nullptr) { RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists"; return -1; } @@ -482,7 +482,7 @@ int32_t AudioMixerManagerLinuxALSA::MinSpeakerVolume( */ int32_t AudioMixerManagerLinuxALSA::SpeakerVolumeIsAvailable(bool& available) { - if (_outputMixerElement == NULL) { + if (_outputMixerElement == nullptr) { RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists"; return -1; } @@ -493,7 +493,7 @@ int32_t AudioMixerManagerLinuxALSA::SpeakerVolumeIsAvailable(bool& available) { } int32_t AudioMixerManagerLinuxALSA::SpeakerMuteIsAvailable(bool& available) { - if (_outputMixerElement == NULL) { + if (_outputMixerElement == nullptr) { RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists"; return -1; } @@ -509,7 +509,7 @@ int32_t AudioMixerManagerLinuxALSA::SetSpeakerMute(bool enable) { MutexLock lock(&mutex_); - if (_outputMixerElement == NULL) { + if (_outputMixerElement == nullptr) { RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists"; return -1; } @@ -535,7 +535,7 @@ int32_t AudioMixerManagerLinuxALSA::SetSpeakerMute(bool enable) { } int32_t AudioMixerManagerLinuxALSA::SpeakerMute(bool& enabled) const { - if (_outputMixerElement == NULL) { + if (_outputMixerElement == nullptr) { RTC_LOG(LS_WARNING) << "no avaliable output mixer exists"; return -1; } @@ -567,7 +567,7 @@ int32_t AudioMixerManagerLinuxALSA::SpeakerMute(bool& enabled) const { } int32_t AudioMixerManagerLinuxALSA::MicrophoneMuteIsAvailable(bool& available) { - if (_inputMixerElement == NULL) { + if (_inputMixerElement == nullptr) { RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists"; return -1; } @@ -582,7 +582,7 @@ int32_t AudioMixerManagerLinuxALSA::SetMicrophoneMute(bool enable) { MutexLock lock(&mutex_); - if (_inputMixerElement == NULL) { + if (_inputMixerElement == nullptr) { RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists"; return -1; } @@ -608,7 +608,7 @@ int32_t AudioMixerManagerLinuxALSA::SetMicrophoneMute(bool enable) { } int32_t AudioMixerManagerLinuxALSA::MicrophoneMute(bool& enabled) const { - if (_inputMixerElement == NULL) { + if (_inputMixerElement == nullptr) { RTC_LOG(LS_WARNING) << "no avaliable input mixer exists"; return -1; } @@ -640,7 +640,7 @@ int32_t AudioMixerManagerLinuxALSA::MicrophoneMute(bool& enabled) const { int32_t AudioMixerManagerLinuxALSA::MicrophoneVolumeIsAvailable( bool& available) { - if (_inputMixerElement == NULL) { + if (_inputMixerElement == nullptr) { RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists"; return -1; } @@ -657,7 +657,7 @@ int32_t AudioMixerManagerLinuxALSA::SetMicrophoneVolume(uint32_t volume) { MutexLock lock(&mutex_); - if (_inputMixerElement == NULL) { + if (_inputMixerElement == nullptr) { RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists"; return -1; } @@ -761,7 +761,7 @@ int32_t AudioMixerManagerLinuxALSA::SetMicrophoneVolume(uint32_t volume) { */ int32_t AudioMixerManagerLinuxALSA::MicrophoneVolume(uint32_t& volume) const { - if (_inputMixerElement == NULL) { + if (_inputMixerElement == nullptr) { RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists"; return -1; } @@ -785,7 +785,7 @@ int32_t AudioMixerManagerLinuxALSA::MicrophoneVolume(uint32_t& volume) const { int32_t AudioMixerManagerLinuxALSA::MaxMicrophoneVolume( uint32_t& maxVolume) const { - if (_inputMixerElement == NULL) { + if (_inputMixerElement == nullptr) { RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists"; return -1; } @@ -816,7 +816,7 @@ int32_t AudioMixerManagerLinuxALSA::MaxMicrophoneVolume( int32_t AudioMixerManagerLinuxALSA::MinMicrophoneVolume( uint32_t& minVolume) const { - if (_inputMixerElement == NULL) { + if (_inputMixerElement == nullptr) { RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists"; return -1; } @@ -848,14 +848,14 @@ int32_t AudioMixerManagerLinuxALSA::LoadMicMixerElement() const { if (errVal < 0) { RTC_LOG(LS_ERROR) << "snd_mixer_load(_inputMixerHandle), error: " << LATE(snd_strerror)(errVal); - _inputMixerHandle = NULL; + _inputMixerHandle = nullptr; return -1; } - snd_mixer_elem_t* elem = NULL; - snd_mixer_elem_t* micElem = NULL; + snd_mixer_elem_t* elem = nullptr; + snd_mixer_elem_t* micElem = nullptr; unsigned mixerIdx = 0; - const char* selemName = NULL; + const char* selemName = nullptr; // Find and store handles to the right mixer elements for (elem = LATE(snd_mixer_first_elem)(_inputMixerHandle); elem; @@ -879,13 +879,13 @@ int32_t AudioMixerManagerLinuxALSA::LoadMicMixerElement() const { } } - if (_inputMixerElement == NULL) { + if (_inputMixerElement == nullptr) { // We didn't find a Capture handle, use Mic. - if (micElem != NULL) { + if (micElem != nullptr) { _inputMixerElement = micElem; RTC_LOG(LS_VERBOSE) << "Using Mic as capture volume."; } else { - _inputMixerElement = NULL; + _inputMixerElement = nullptr; RTC_LOG(LS_ERROR) << "Could not find capture volume on the mixer."; return -1; @@ -900,15 +900,15 @@ int32_t AudioMixerManagerLinuxALSA::LoadSpeakerMixerElement() const { if (errVal < 0) { RTC_LOG(LS_ERROR) << "snd_mixer_load(_outputMixerHandle), error: " << LATE(snd_strerror)(errVal); - _outputMixerHandle = NULL; + _outputMixerHandle = nullptr; return -1; } - snd_mixer_elem_t* elem = NULL; - snd_mixer_elem_t* masterElem = NULL; - snd_mixer_elem_t* speakerElem = NULL; + snd_mixer_elem_t* elem = nullptr; + snd_mixer_elem_t* masterElem = nullptr; + snd_mixer_elem_t* speakerElem = nullptr; unsigned mixerIdx = 0; - const char* selemName = NULL; + const char* selemName = nullptr; // Find and store handles to the right mixer elements for (elem = LATE(snd_mixer_first_elem)(_outputMixerHandle); elem; @@ -938,15 +938,15 @@ int32_t AudioMixerManagerLinuxALSA::LoadSpeakerMixerElement() const { } // If we didn't find a PCM Handle, use Master or Speaker - if (_outputMixerElement == NULL) { - if (masterElem != NULL) { + if (_outputMixerElement == nullptr) { + if (masterElem != nullptr) { _outputMixerElement = masterElem; RTC_LOG(LS_VERBOSE) << "Using Master as output volume."; - } else if (speakerElem != NULL) { + } else if (speakerElem != nullptr) { _outputMixerElement = speakerElem; RTC_LOG(LS_VERBOSE) << "Using Speaker as output volume."; } else { - _outputMixerElement = NULL; + _outputMixerElement = nullptr; RTC_LOG(LS_ERROR) << "Could not find output volume in the mixer."; return -1; } diff --git a/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc b/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc index 91beee3c87..9f899158a9 100644 --- a/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc +++ b/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc @@ -43,10 +43,10 @@ class AutoPulseLock { AudioMixerManagerLinuxPulse::AudioMixerManagerLinuxPulse() : _paOutputDeviceIndex(-1), _paInputDeviceIndex(-1), - _paPlayStream(NULL), - _paRecStream(NULL), - _paMainloop(NULL), - _paContext(NULL), + _paPlayStream(nullptr), + _paRecStream(nullptr), + _paMainloop(nullptr), + _paContext(nullptr), _paVolume(0), _paMute(0), _paVolSteps(0), @@ -95,8 +95,8 @@ int32_t AudioMixerManagerLinuxPulse::Close() { CloseSpeaker(); CloseMicrophone(); - _paMainloop = NULL; - _paContext = NULL; + _paMainloop = nullptr; + _paContext = nullptr; _paObjectsSet = false; return 0; @@ -108,7 +108,7 @@ int32_t AudioMixerManagerLinuxPulse::CloseSpeaker() { // Reset the index to -1 _paOutputDeviceIndex = -1; - _paPlayStream = NULL; + _paPlayStream = nullptr; return 0; } @@ -119,7 +119,7 @@ int32_t AudioMixerManagerLinuxPulse::CloseMicrophone() { // Reset the index to -1 _paInputDeviceIndex = -1; - _paRecStream = NULL; + _paRecStream = nullptr; return 0; } @@ -226,10 +226,10 @@ int32_t AudioMixerManagerLinuxPulse::SetSpeakerVolume(uint32_t volume) { pa_cvolume cVolumes; LATE(pa_cvolume_set)(&cVolumes, spec->channels, volume); - pa_operation* paOperation = NULL; + pa_operation* paOperation = nullptr; paOperation = LATE(pa_context_set_sink_input_volume)( _paContext, LATE(pa_stream_get_index)(_paPlayStream), &cVolumes, - PaSetVolumeCallback, NULL); + PaSetVolumeCallback, nullptr); if (!paOperation) { setFailed = true; } @@ -346,10 +346,10 @@ int32_t AudioMixerManagerLinuxPulse::SetSpeakerMute(bool enable) { // We can only really mute if we have a connected stream AutoPulseLock auto_lock(_paMainloop); - pa_operation* paOperation = NULL; + pa_operation* paOperation = nullptr; paOperation = LATE(pa_context_set_sink_input_mute)( _paContext, LATE(pa_stream_get_index)(_paPlayStream), (int)enable, - PaSetVolumeCallback, NULL); + PaSetVolumeCallback, nullptr); if (!paOperation) { setFailed = true; } @@ -442,7 +442,7 @@ int32_t AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable( deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream); } - pa_operation* paOperation = NULL; + pa_operation* paOperation = nullptr; // Get info for this source // We want to know if the actual device can record in stereo @@ -487,7 +487,7 @@ int32_t AudioMixerManagerLinuxPulse::SetMicrophoneMute(bool enable) { } bool setFailed(false); - pa_operation* paOperation = NULL; + pa_operation* paOperation = nullptr; uint32_t deviceIndex = (uint32_t)_paInputDeviceIndex; @@ -503,7 +503,7 @@ int32_t AudioMixerManagerLinuxPulse::SetMicrophoneMute(bool enable) { // Set mute switch for the source paOperation = LATE(pa_context_set_source_mute_by_index)( - _paContext, deviceIndex, enable, PaSetVolumeCallback, NULL); + _paContext, deviceIndex, enable, PaSetVolumeCallback, nullptr); if (!paOperation) { setFailed = true; @@ -599,7 +599,7 @@ int32_t AudioMixerManagerLinuxPulse::SetMicrophoneVolume(uint32_t volume) { } bool setFailed(false); - pa_operation* paOperation = NULL; + pa_operation* paOperation = nullptr; // Get the number of channels for this source paOperation = LATE(pa_context_get_source_info_by_index)( @@ -613,7 +613,7 @@ int32_t AudioMixerManagerLinuxPulse::SetMicrophoneVolume(uint32_t volume) { // Set the volume for the source paOperation = LATE(pa_context_set_source_volume_by_index)( - _paContext, deviceIndex, &cVolumes, PaSetVolumeCallback, NULL); + _paContext, deviceIndex, &cVolumes, PaSetVolumeCallback, nullptr); if (!paOperation) { setFailed = true; @@ -807,7 +807,7 @@ void AudioMixerManagerLinuxPulse::WaitForOperationCompletion( } bool AudioMixerManagerLinuxPulse::GetSinkInputInfo() const { - pa_operation* paOperation = NULL; + pa_operation* paOperation = nullptr; AutoPulseLock auto_lock(_paMainloop); // Get info for this stream (sink input). @@ -820,7 +820,7 @@ bool AudioMixerManagerLinuxPulse::GetSinkInputInfo() const { } bool AudioMixerManagerLinuxPulse::GetSinkInfoByIndex(int device_index) const { - pa_operation* paOperation = NULL; + pa_operation* paOperation = nullptr; AutoPulseLock auto_lock(_paMainloop); paOperation = LATE(pa_context_get_sink_info_by_index)( @@ -831,7 +831,7 @@ bool AudioMixerManagerLinuxPulse::GetSinkInfoByIndex(int device_index) const { } bool AudioMixerManagerLinuxPulse::GetSourceInfoByIndex(int device_index) const { - pa_operation* paOperation = NULL; + pa_operation* paOperation = nullptr; AutoPulseLock auto_lock(_paMainloop); paOperation = LATE(pa_context_get_source_info_by_index)( diff --git a/system_wrappers/source/field_trial.cc b/system_wrappers/source/field_trial.cc index a1309edbfe..5efa01c9ef 100644 --- a/system_wrappers/source/field_trial.cc +++ b/system_wrappers/source/field_trial.cc @@ -28,7 +28,7 @@ namespace webrtc { namespace field_trial { -static const char* trials_init_string = NULL; +static const char* trials_init_string = nullptr; namespace { @@ -127,7 +127,7 @@ std::string FindFullName(absl::string_view name) { << name << " is not registered, see g3doc/field-trials.md."; #endif - if (trials_init_string == NULL) + if (trials_init_string == nullptr) return std::string(); absl::string_view trials_string(trials_init_string); From 8d54148f2aea78211654f4e5216a08f9a0ee0e5c Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Thu, 15 May 2025 11:49:22 +0000 Subject: [PATCH 175/292] [cleanup] Replace uses of NULL with nullptr Generated using clang-tidy, tools/clang/scripts/build_clang_tools_extra.py \ --fetch out/Default clang-tidy clang-apply-replacements ninja -C out/Default gn gen out/Default --export-compile-commands cd out/Default tools/clang/third_party/llvm/clang-tools-extra/clang-tidy/tool/run-clang-tidy.py -p . \ -clang-tidy-binary out/Default/tools/clang/third_party/llvm/build/bin/clang-tidy \ -clang-apply-replacements-binary \ out/Default/tools/clang/third_party/llvm/build/bin/clang-apply-replacements \ -checks='-*,modernize-use-nullptr' \ -fix This CL was uploaded by git cl split. Bug: webrtc:42232595 Change-Id: I458421abc8bb302782720f194e7eeace5aef8332 No-Iwyu: LSC Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391840 Reviewed-by: Artem Titov Auto-Submit: Evan Shrubsole Commit-Queue: Evan Shrubsole Cr-Commit-Position: refs/heads/main@{#44647} --- api/test/metrics/chrome_perf_dashboard_metrics_exporter.cc | 2 +- api/test/metrics/metrics_set_proto_file_exporter.cc | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/api/test/metrics/chrome_perf_dashboard_metrics_exporter.cc b/api/test/metrics/chrome_perf_dashboard_metrics_exporter.cc index b03185d4b0..5fe0b14c2b 100644 --- a/api/test/metrics/chrome_perf_dashboard_metrics_exporter.cc +++ b/api/test/metrics/chrome_perf_dashboard_metrics_exporter.cc @@ -75,7 +75,7 @@ ImproveDirection ToChromePerfDashboardImproveDirection( bool WriteMetricsToFile(const std::string& path, const std::string& data) { CreateDir(DirName(path)); FILE* output = fopen(path.c_str(), "wb"); - if (output == NULL) { + if (output == nullptr) { printf("Failed to write to %s.\n", path.c_str()); return false; } diff --git a/api/test/metrics/metrics_set_proto_file_exporter.cc b/api/test/metrics/metrics_set_proto_file_exporter.cc index a26ae7ac35..044828c120 100644 --- a/api/test/metrics/metrics_set_proto_file_exporter.cc +++ b/api/test/metrics/metrics_set_proto_file_exporter.cc @@ -101,7 +101,7 @@ bool WriteMetricsToFile(const std::string& path, CreateDir(DirName(path)); FILE* output = fopen(path.c_str(), "wb"); - if (output == NULL) { + if (output == nullptr) { RTC_LOG(LS_ERROR) << "Failed to write to " << path; return false; } From 8d0f13abbf3750f75eb0b57d0a4f1861119c3d6b Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Thu, 15 May 2025 11:49:54 +0000 Subject: [PATCH 176/292] [cleanup] Replace uses of NULL with nullptr Generated using clang-tidy, tools/clang/scripts/build_clang_tools_extra.py \ --fetch out/Default clang-tidy clang-apply-replacements ninja -C out/Default gn gen out/Default --export-compile-commands cd out/Default tools/clang/third_party/llvm/clang-tools-extra/clang-tidy/tool/run-clang-tidy.py -p . \ -clang-tidy-binary out/Default/tools/clang/third_party/llvm/build/bin/clang-tidy \ -clang-apply-replacements-binary \ out/Default/tools/clang/third_party/llvm/build/bin/clang-apply-replacements \ -checks='-*,modernize-use-nullptr' \ -fix This CL was uploaded by git cl split. Bug: webrtc:42232595 Change-Id: I6045071c0816a0606f429a5242fe4a622615a4a8 No-Iwyu: LSC Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391860 Commit-Queue: Per Kjellander Auto-Submit: Evan Shrubsole Reviewed-by: Per Kjellander Cr-Commit-Position: refs/heads/main@{#44648} --- api/transport/stun.cc | 20 ++++++------- api/transport/stun_unittest.cc | 54 +++++++++++++++++----------------- 2 files changed, 37 insertions(+), 37 deletions(-) diff --git a/api/transport/stun.cc b/api/transport/stun.cc index b5e74e49df..d688f0c4ff 100644 --- a/api/transport/stun.cc +++ b/api/transport/stun.cc @@ -732,7 +732,7 @@ StunAttribute* StunMessage::CreateAttribute(int type, size_t length) /*const*/ { return StunAttribute::Create(STUN_VALUE_BYTE_STRING, type, static_cast(length), this); } else { - return NULL; + return nullptr; } } @@ -742,7 +742,7 @@ const StunAttribute* StunMessage::GetAttribute(int type) const { return attr.get(); } } - return NULL; + return nullptr; } bool StunMessage::IsValidTransactionId(absl::string_view transaction_id) { @@ -828,7 +828,7 @@ StunAttribute* StunAttribute::Create(StunAttributeValueType value_type, case STUN_VALUE_UINT16_LIST: return new StunUInt16ListAttribute(type, length); default: - return NULL; + return nullptr; } } @@ -954,7 +954,7 @@ bool StunAddressAttribute::Write(ByteBufferWriter* buf) const { StunXorAddressAttribute::StunXorAddressAttribute(uint16_t type, const SocketAddress& addr) - : StunAddressAttribute(type, addr), owner_(NULL) {} + : StunAddressAttribute(type, addr), owner_(nullptr) {} StunXorAddressAttribute::StunXorAddressAttribute(uint16_t type, uint16_t length, @@ -1096,23 +1096,23 @@ bool StunUInt64Attribute::Write(ByteBufferWriter* buf) const { } StunByteStringAttribute::StunByteStringAttribute(uint16_t type) - : StunAttribute(type, 0), bytes_(NULL) {} + : StunAttribute(type, 0), bytes_(nullptr) {} StunByteStringAttribute::StunByteStringAttribute(uint16_t type, absl::string_view str) - : StunAttribute(type, 0), bytes_(NULL) { + : StunAttribute(type, 0), bytes_(nullptr) { CopyBytes(str); } StunByteStringAttribute::StunByteStringAttribute(uint16_t type, const void* bytes, size_t length) - : StunAttribute(type, 0), bytes_(NULL) { + : StunAttribute(type, 0), bytes_(nullptr) { CopyBytes(bytes, length); } StunByteStringAttribute::StunByteStringAttribute(uint16_t type, uint16_t length) - : StunAttribute(type, length), bytes_(NULL) {} + : StunAttribute(type, length), bytes_(nullptr) {} StunByteStringAttribute::~StunByteStringAttribute() { delete[] bytes_; @@ -1135,13 +1135,13 @@ void StunByteStringAttribute::CopyBytes(const void* bytes, size_t length) { } uint8_t StunByteStringAttribute::GetByte(size_t index) const { - RTC_DCHECK(bytes_ != NULL); + RTC_DCHECK(bytes_ != nullptr); RTC_DCHECK(index < length()); return bytes_[index]; } void StunByteStringAttribute::SetByte(size_t index, uint8_t value) { - RTC_DCHECK(bytes_ != NULL); + RTC_DCHECK(bytes_ != nullptr); RTC_DCHECK(index < length()); bytes_[index] = value; } diff --git a/api/transport/stun_unittest.cc b/api/transport/stun_unittest.cc index bcbb2badc6..a49f1ecd66 100644 --- a/api/transport/stun_unittest.cc +++ b/api/transport/stun_unittest.cc @@ -652,20 +652,20 @@ TEST_F(StunTest, ReadRfc5769RequestMessage) { const StunByteStringAttribute* software = msg.GetByteString(STUN_ATTR_SOFTWARE); - ASSERT_TRUE(software != NULL); + ASSERT_TRUE(software != nullptr); EXPECT_EQ(kRfc5769SampleMsgClientSoftware, software->string_view()); const StunByteStringAttribute* username = msg.GetByteString(STUN_ATTR_USERNAME); - ASSERT_TRUE(username != NULL); + ASSERT_TRUE(username != nullptr); EXPECT_EQ(kRfc5769SampleMsgUsername, username->string_view()); // Actual M-I value checked in a later test. - ASSERT_TRUE(msg.GetByteString(STUN_ATTR_MESSAGE_INTEGRITY) != NULL); + ASSERT_TRUE(msg.GetByteString(STUN_ATTR_MESSAGE_INTEGRITY) != nullptr); // Fingerprint checked in a later test, but double-check the value here. const StunUInt32Attribute* fingerprint = msg.GetUInt32(STUN_ATTR_FINGERPRINT); - ASSERT_TRUE(fingerprint != NULL); + ASSERT_TRUE(fingerprint != nullptr); EXPECT_EQ(0xe57a3bcf, fingerprint->value()); } @@ -679,17 +679,17 @@ TEST_F(StunTest, ReadRfc5769ResponseMessage) { const StunByteStringAttribute* software = msg.GetByteString(STUN_ATTR_SOFTWARE); - ASSERT_TRUE(software != NULL); + ASSERT_TRUE(software != nullptr); EXPECT_EQ(kRfc5769SampleMsgServerSoftware, software->string_view()); const StunAddressAttribute* mapped_address = msg.GetAddress(STUN_ATTR_XOR_MAPPED_ADDRESS); - ASSERT_TRUE(mapped_address != NULL); + ASSERT_TRUE(mapped_address != nullptr); EXPECT_EQ(kRfc5769SampleMsgMappedAddress, mapped_address->GetAddress()); // Actual M-I and fingerprint checked in later tests. - ASSERT_TRUE(msg.GetByteString(STUN_ATTR_MESSAGE_INTEGRITY) != NULL); - ASSERT_TRUE(msg.GetUInt32(STUN_ATTR_FINGERPRINT) != NULL); + ASSERT_TRUE(msg.GetByteString(STUN_ATTR_MESSAGE_INTEGRITY) != nullptr); + ASSERT_TRUE(msg.GetUInt32(STUN_ATTR_FINGERPRINT) != nullptr); } // Read the RFC5389 fields from the RFC5769 sample STUN response for IPv6. @@ -702,17 +702,17 @@ TEST_F(StunTest, ReadRfc5769ResponseMessageIPv6) { const StunByteStringAttribute* software = msg.GetByteString(STUN_ATTR_SOFTWARE); - ASSERT_TRUE(software != NULL); + ASSERT_TRUE(software != nullptr); EXPECT_EQ(kRfc5769SampleMsgServerSoftware, software->string_view()); const StunAddressAttribute* mapped_address = msg.GetAddress(STUN_ATTR_XOR_MAPPED_ADDRESS); - ASSERT_TRUE(mapped_address != NULL); + ASSERT_TRUE(mapped_address != nullptr); EXPECT_EQ(kRfc5769SampleMsgIPv6MappedAddress, mapped_address->GetAddress()); // Actual M-I and fingerprint checked in later tests. - ASSERT_TRUE(msg.GetByteString(STUN_ATTR_MESSAGE_INTEGRITY) != NULL); - ASSERT_TRUE(msg.GetUInt32(STUN_ATTR_FINGERPRINT) != NULL); + ASSERT_TRUE(msg.GetByteString(STUN_ATTR_MESSAGE_INTEGRITY) != nullptr); + ASSERT_TRUE(msg.GetUInt32(STUN_ATTR_FINGERPRINT) != nullptr); } // Read the RFC5389 fields from the RFC5769 sample STUN response with auth. @@ -725,20 +725,20 @@ TEST_F(StunTest, ReadRfc5769RequestMessageLongTermAuth) { const StunByteStringAttribute* username = msg.GetByteString(STUN_ATTR_USERNAME); - ASSERT_TRUE(username != NULL); + ASSERT_TRUE(username != nullptr); EXPECT_EQ(kRfc5769SampleMsgWithAuthUsername, username->string_view()); const StunByteStringAttribute* nonce = msg.GetByteString(STUN_ATTR_NONCE); - ASSERT_TRUE(nonce != NULL); + ASSERT_TRUE(nonce != nullptr); EXPECT_EQ(kRfc5769SampleMsgWithAuthNonce, nonce->string_view()); const StunByteStringAttribute* realm = msg.GetByteString(STUN_ATTR_REALM); - ASSERT_TRUE(realm != NULL); + ASSERT_TRUE(realm != nullptr); EXPECT_EQ(kRfc5769SampleMsgWithAuthRealm, realm->string_view()); // No fingerprint, actual M-I checked in later tests. - ASSERT_TRUE(msg.GetByteString(STUN_ATTR_MESSAGE_INTEGRITY) != NULL); - ASSERT_TRUE(msg.GetUInt32(STUN_ATTR_FINGERPRINT) == NULL); + ASSERT_TRUE(msg.GetByteString(STUN_ATTR_MESSAGE_INTEGRITY) != nullptr); + ASSERT_TRUE(msg.GetUInt32(STUN_ATTR_FINGERPRINT) == nullptr); } // The RFC3489 packet in this test is the same as @@ -778,7 +778,7 @@ TEST_F(StunTest, SetIPv6XorAddressAttributeOwner) { // Owner with a different transaction ID. StunMessage msg2(STUN_INVALID_MESSAGE_TYPE, "ABCDABCDABCD"); - StunXorAddressAttribute addr2(STUN_ATTR_XOR_MAPPED_ADDRESS, 20, NULL); + StunXorAddressAttribute addr2(STUN_ATTR_XOR_MAPPED_ADDRESS, 20, nullptr); addr2.SetIP(addr->ipaddr()); addr2.SetPort(addr->port()); addr2.SetOwner(&msg2); @@ -799,7 +799,7 @@ TEST_F(StunTest, SetIPv6XorAddressAttributeOwner) { addr2.SetIP(addr->ipaddr()); addr2.SetPort(addr->port()); // Try writing with no owner at all, should fail and write nothing. - addr2.SetOwner(NULL); + addr2.SetOwner(nullptr); ASSERT_EQ(addr2.ipaddr(), addr->ipaddr()); wrong_buf.Clear(); EXPECT_FALSE(addr2.Write(&wrong_buf)); @@ -825,7 +825,7 @@ TEST_F(StunTest, SetIPv4XorAddressAttributeOwner) { // Owner with a different transaction ID. StunMessage msg2(STUN_INVALID_MESSAGE_TYPE, "ABCDABCDABCD"); - StunXorAddressAttribute addr2(STUN_ATTR_XOR_MAPPED_ADDRESS, 20, NULL); + StunXorAddressAttribute addr2(STUN_ATTR_XOR_MAPPED_ADDRESS, 20, nullptr); addr2.SetIP(addr->ipaddr()); addr2.SetPort(addr->port()); addr2.SetOwner(&msg2); @@ -846,7 +846,7 @@ TEST_F(StunTest, SetIPv4XorAddressAttributeOwner) { ASSERT_EQ(addr->ipaddr(), addr2.ipaddr()); // However, no owner is still an error, should fail and write nothing. - addr2.SetOwner(NULL); + addr2.SetOwner(nullptr); ASSERT_EQ(addr2.ipaddr(), addr->ipaddr()); wrong_buf.Clear(); EXPECT_FALSE(addr2.Write(&wrong_buf)); @@ -1015,7 +1015,7 @@ TEST_F(StunTest, ReadByteStringAttribute) { CheckStunTransactionID(msg, kTestTransactionId2, kStunTransactionIdLength); const StunByteStringAttribute* username = msg.GetByteString(STUN_ATTR_USERNAME); - ASSERT_TRUE(username != NULL); + ASSERT_TRUE(username != nullptr); EXPECT_EQ(kTestUserName1, username->string_view()); } @@ -1028,7 +1028,7 @@ TEST_F(StunTest, ReadPaddedByteStringAttribute) { CheckStunTransactionID(msg, kTestTransactionId2, kStunTransactionIdLength); const StunByteStringAttribute* username = msg.GetByteString(STUN_ATTR_USERNAME); - ASSERT_TRUE(username != NULL); + ASSERT_TRUE(username != nullptr); EXPECT_EQ(kTestUserName2, username->string_view()); } @@ -1039,7 +1039,7 @@ TEST_F(StunTest, ReadErrorCodeAttribute) { CheckStunHeader(msg, STUN_BINDING_ERROR_RESPONSE, size); CheckStunTransactionID(msg, kTestTransactionId1, kStunTransactionIdLength); const StunErrorCodeAttribute* errorcode = msg.GetErrorCode(); - ASSERT_TRUE(errorcode != NULL); + ASSERT_TRUE(errorcode != nullptr); EXPECT_EQ(kTestErrorClass, errorcode->eclass()); EXPECT_EQ(kTestErrorNumber, errorcode->number()); EXPECT_EQ(kTestErrorReason, errorcode->reason()); @@ -1060,7 +1060,7 @@ TEST_F(StunTest, ReadMessageWithAUInt16ListAttribute) { size_t size = ReadStunMessage(&msg, kStunMessageWithUInt16ListAttribute); CheckStunHeader(msg, STUN_BINDING_REQUEST, size); const StunUInt16ListAttribute* types = msg.GetUnknownAttributes(); - ASSERT_TRUE(types != NULL); + ASSERT_TRUE(types != nullptr); EXPECT_EQ(3U, types->Size()); EXPECT_EQ(0x1U, types->GetType(0)); EXPECT_EQ(0x1000U, types->GetType(1)); @@ -1075,7 +1075,7 @@ TEST_F(StunTest, ReadMessageWithAnUnknownAttribute) { // Parsing should have succeeded and there should be a USERNAME attribute const StunByteStringAttribute* username = msg.GetByteString(STUN_ATTR_USERNAME); - ASSERT_TRUE(username != NULL); + ASSERT_TRUE(username != nullptr); EXPECT_EQ(kTestUserName2, username->string_view()); } @@ -1680,7 +1680,7 @@ TEST_F(StunTest, GoogMiscInfo) { CheckStunHeader(msg, STUN_BINDING_REQUEST, read_size); const StunUInt16ListAttribute* types = msg.GetUInt16List(STUN_ATTR_GOOG_MISC_INFO); - ASSERT_TRUE(types != NULL); + ASSERT_TRUE(types != nullptr); EXPECT_EQ(4U, types->Size()); EXPECT_EQ(0x1U, types->GetType(0)); EXPECT_EQ(0x0U, types->GetType(1)); From ca6be77a33e75087e29210415ddf2df96837da73 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Thu, 15 May 2025 11:52:57 +0000 Subject: [PATCH 177/292] [cleanup] Replace uses of NULL with nullptr Generated using clang-tidy, tools/clang/scripts/build_clang_tools_extra.py \ --fetch out/Default clang-tidy clang-apply-replacements ninja -C out/Default gn gen out/Default --export-compile-commands cd out/Default tools/clang/third_party/llvm/clang-tools-extra/clang-tidy/tool/run-clang-tidy.py -p . \ -clang-tidy-binary out/Default/tools/clang/third_party/llvm/build/bin/clang-tidy \ -clang-apply-replacements-binary \ out/Default/tools/clang/third_party/llvm/build/bin/clang-apply-replacements \ -checks='-*,modernize-use-nullptr' \ -fix This CL was uploaded by git cl split. Bug: webrtc:42232595 Change-Id: I2ecc7b78f814e589934aee139f6644e5f8ff2f22 No-Iwyu: LSC Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391861 Reviewed-by: Jonas Oreland Commit-Queue: Jonas Oreland Auto-Submit: Evan Shrubsole Cr-Commit-Position: refs/heads/main@{#44649} --- p2p/base/async_stun_tcp_socket_unittest.cc | 2 +- p2p/base/basic_packet_socket_factory.cc | 22 +-- p2p/base/connection.cc | 4 +- p2p/base/p2p_transport_channel.cc | 4 +- p2p/base/p2p_transport_channel_unittest.cc | 32 ++-- p2p/base/port.cc | 8 +- p2p/base/port_unittest.cc | 156 +++++++++--------- p2p/base/pseudo_tcp.cc | 2 +- p2p/base/pseudo_tcp_unittest.cc | 4 +- p2p/base/stun_port_unittest.cc | 6 +- p2p/base/stun_request.cc | 2 +- p2p/base/stun_request_unittest.cc | 6 +- p2p/base/tcp_port.cc | 18 +- p2p/base/transport_description_factory.cc | 10 +- .../transport_description_factory_unittest.cc | 66 ++++---- p2p/base/turn_port_unittest.cc | 28 ++-- p2p/client/basic_port_allocator.cc | 16 +- p2p/dtls/dtls_transport.cc | 2 +- p2p/dtls/dtls_transport_unittest.cc | 2 +- p2p/test/nat_types.cc | 2 +- p2p/test/nat_unittest.cc | 2 +- p2p/test/stun_server_unittest.cc | 14 +- p2p/test/turn_server.cc | 18 +- 23 files changed, 214 insertions(+), 212 deletions(-) diff --git a/p2p/base/async_stun_tcp_socket_unittest.cc b/p2p/base/async_stun_tcp_socket_unittest.cc index 2730dcd1f7..bdb8bbb7e7 100644 --- a/p2p/base/async_stun_tcp_socket_unittest.cc +++ b/p2p/base/async_stun_tcp_socket_unittest.cc @@ -97,7 +97,7 @@ class AsyncStunTCPSocketTest : public ::testing::Test, client, kClientAddr, listen_socket_->GetLocalAddress())); send_socket_->SignalSentPacket.connect( this, &AsyncStunTCPSocketTest::OnSentPacket); - ASSERT_TRUE(send_socket_.get() != NULL); + ASSERT_TRUE(send_socket_.get() != nullptr); vss_->ProcessMessagesUntilIdle(); } diff --git a/p2p/base/basic_packet_socket_factory.cc b/p2p/base/basic_packet_socket_factory.cc index 4604570ceb..785f80b330 100644 --- a/p2p/base/basic_packet_socket_factory.cc +++ b/p2p/base/basic_packet_socket_factory.cc @@ -47,12 +47,12 @@ AsyncPacketSocket* BasicPacketSocketFactory::CreateUdpSocket( // UDP sockets are simple. Socket* socket = socket_factory_->CreateSocket(address.family(), SOCK_DGRAM); if (!socket) { - return NULL; + return nullptr; } if (BindSocket(socket, address, min_port, max_port) < 0) { RTC_LOG(LS_ERROR) << "UDP bind failed with error " << socket->GetError(); delete socket; - return NULL; + return nullptr; } return new AsyncUDPSocket(socket); } @@ -65,23 +65,23 @@ AsyncListenSocket* BasicPacketSocketFactory::CreateServerTcpSocket( // Fail if TLS is required. if (opts & PacketSocketFactory::OPT_TLS) { RTC_LOG(LS_ERROR) << "TLS support currently is not available."; - return NULL; + return nullptr; } if (opts & PacketSocketFactory::OPT_TLS_FAKE) { RTC_LOG(LS_ERROR) << "Fake TLS not supported."; - return NULL; + return nullptr; } Socket* socket = socket_factory_->CreateSocket(local_address.family(), SOCK_STREAM); if (!socket) { - return NULL; + return nullptr; } if (BindSocket(socket, local_address, min_port, max_port) < 0) { RTC_LOG(LS_ERROR) << "TCP bind failed with error " << socket->GetError(); delete socket; - return NULL; + return nullptr; } RTC_CHECK(!(opts & PacketSocketFactory::OPT_STUN)); @@ -96,7 +96,7 @@ AsyncPacketSocket* BasicPacketSocketFactory::CreateClientTcpSocket( Socket* socket = socket_factory_->CreateSocket(local_address.family(), SOCK_STREAM); if (!socket) { - return NULL; + return nullptr; } if (BindSocket(socket, local_address, 0, 0) < 0) { @@ -109,7 +109,7 @@ AsyncPacketSocket* BasicPacketSocketFactory::CreateClientTcpSocket( } else { RTC_LOG(LS_ERROR) << "TCP bind failed with error " << socket->GetError(); delete socket; - return NULL; + return nullptr; } } @@ -134,7 +134,7 @@ AsyncPacketSocket* BasicPacketSocketFactory::CreateClientTcpSocket( // Using TLS, wrap the socket in an SSL adapter. SSLAdapter* ssl_adapter = SSLAdapter::Create(socket); if (!ssl_adapter) { - return NULL; + return nullptr; } if (tlsOpts & PacketSocketFactory::OPT_TLS_INSECURE) { @@ -149,7 +149,7 @@ AsyncPacketSocket* BasicPacketSocketFactory::CreateClientTcpSocket( if (ssl_adapter->StartSSL(remote_address.hostname().c_str()) != 0) { delete ssl_adapter; - return NULL; + return nullptr; } } else if (tlsOpts & PacketSocketFactory::OPT_TLS_FAKE) { @@ -160,7 +160,7 @@ AsyncPacketSocket* BasicPacketSocketFactory::CreateClientTcpSocket( if (socket->Connect(remote_address) < 0) { RTC_LOG(LS_ERROR) << "TCP connect failed with error " << socket->GetError(); delete socket; - return NULL; + return nullptr; } // Finally, wrap that socket in a TCP or STUN TCP packet socket. diff --git a/p2p/base/connection.cc b/p2p/base/connection.cc index 497df771a6..92403b4d78 100644 --- a/p2p/base/connection.cc +++ b/p2p/base/connection.cc @@ -741,8 +741,8 @@ void Connection::SendStunBindingResponse(const StunMessage* message) { // Retrieve the username from the `message`. const StunByteStringAttribute* username_attr = message->GetByteString(STUN_ATTR_USERNAME); - RTC_DCHECK(username_attr != NULL); - if (username_attr == NULL) { + RTC_DCHECK(username_attr != nullptr); + if (username_attr == nullptr) { // No valid username, skip the response. return; } diff --git a/p2p/base/p2p_transport_channel.cc b/p2p/base/p2p_transport_channel.cc index 93ab452f50..ad69f37445 100644 --- a/p2p/base/p2p_transport_channel.cc +++ b/p2p/base/p2p_transport_channel.cc @@ -1310,7 +1310,7 @@ void P2PTransportChannel::FinishAddingRemoteCandidate( } // Create connections to this remote candidate. - CreateConnections(new_remote_candidate, NULL); + CreateConnections(new_remote_candidate, nullptr); // Resort the connections list, which may have new elements. ice_controller_->OnImmediateSortAndSwitchRequest( @@ -1371,7 +1371,7 @@ bool P2PTransportChannel::CreateConnections(const Candidate& remote_candidate, } } - if ((origin_port != NULL) && !absl::c_linear_search(ports_, origin_port)) { + if ((origin_port != nullptr) && !absl::c_linear_search(ports_, origin_port)) { if (CreateConnection(origin_port, remote_candidate, origin_port)) created = true; } diff --git a/p2p/base/p2p_transport_channel_unittest.cc b/p2p/base/p2p_transport_channel_unittest.cc index b042942f92..acebe9a6c4 100644 --- a/p2p/base/p2p_transport_channel_unittest.cc +++ b/p2p/base/p2p_transport_channel_unittest.cc @@ -408,7 +408,7 @@ class P2PTransportChannelTestBase : public ::testing::Test, } ChannelData* GetChannelData(PacketTransportInternal* transport) { if (!HasTransport(transport)) - return NULL; + return nullptr; if (cd1_.ch_.get() == transport) return &cd1_; else @@ -572,7 +572,7 @@ class P2PTransportChannelTestBase : public ::testing::Test, } else if (endpoint == 1) { return &ep2_; } else { - return NULL; + return nullptr; } } BasicPortAllocator* GetAllocator(int endpoint) { @@ -988,12 +988,12 @@ class P2PTransportChannelTestBase : public ::testing::Test, static const Candidate* LocalCandidate(P2PTransportChannel* ch) { return (ch && ch->selected_connection()) ? &ch->selected_connection()->local_candidate() - : NULL; + : nullptr; } static const Candidate* RemoteCandidate(P2PTransportChannel* ch) { return (ch && ch->selected_connection()) ? &ch->selected_connection()->remote_candidate() - : NULL; + : nullptr; } Endpoint* GetEndpoint(PacketTransportInternal* transport) { if (ep1_.HasTransport(transport)) { @@ -1001,7 +1001,7 @@ class P2PTransportChannelTestBase : public ::testing::Test, } else if (ep2_.HasTransport(transport)) { return &ep2_; } else { - return NULL; + return nullptr; } } P2PTransportChannel* GetRemoteChannel(IceTransportInternal* ch) { @@ -1014,7 +1014,7 @@ class P2PTransportChannelTestBase : public ::testing::Test, else if (ch == ep2_ch2()) return ep1_ch2(); else - return NULL; + return nullptr; } std::list& GetPacketList(PacketTransportInternal* transport) { return GetChannelData(transport)->ch_packets_; @@ -1255,21 +1255,23 @@ const P2PTransportChannelMatrixTest::Result* /*OP*/ {LULU, LUSU, LUSU, LUSU, LUPU, LUSU, LUPU, LTPT, LTPT, LSRS}, /*CO*/ - {SULU, SUSU, SUSU, SUSU, SUPU, SUSU, SUPU, NULL, NULL, LSRS}, + {SULU, SUSU, SUSU, SUSU, SUPU, SUSU, SUPU, nullptr, nullptr, LSRS}, /*AD*/ - {SULU, SUSU, SUSU, SUSU, SUPU, SUSU, SUPU, NULL, NULL, LSRS}, + {SULU, SUSU, SUSU, SUSU, SUPU, SUSU, SUPU, nullptr, nullptr, LSRS}, /*PO*/ - {SULU, SUSU, SUSU, SUSU, RUPU, SUSU, RUPU, NULL, NULL, LSRS}, + {SULU, SUSU, SUSU, SUSU, RUPU, SUSU, RUPU, nullptr, nullptr, LSRS}, /*SY*/ - {PULU, PUSU, PUSU, PURU, PURU, PUSU, PURU, NULL, NULL, LSRS}, + {PULU, PUSU, PUSU, PURU, PURU, PUSU, PURU, nullptr, nullptr, LSRS}, /*2C*/ - {SULU, SUSU, SUSU, SUSU, SUPU, SUSU, SUPU, NULL, NULL, LSRS}, + {SULU, SUSU, SUSU, SUSU, SUPU, SUSU, SUPU, nullptr, nullptr, LSRS}, /*SC*/ - {PULU, PUSU, PUSU, PURU, PURU, PUSU, PURU, NULL, NULL, LSRS}, + {PULU, PUSU, PUSU, PURU, PURU, PUSU, PURU, nullptr, nullptr, LSRS}, /*!U*/ - {LTPT, NULL, NULL, NULL, NULL, NULL, NULL, LTPT, LTPT, LSRS}, + {LTPT, nullptr, nullptr, nullptr, nullptr, nullptr, nullptr, LTPT, LTPT, + LSRS}, /*!T*/ - {PTLT, NULL, NULL, NULL, NULL, NULL, NULL, PTLT, LTRT, LSRS}, + {PTLT, nullptr, nullptr, nullptr, nullptr, nullptr, nullptr, PTLT, LTRT, + LSRS}, /*HT*/ {LSRS, LSRS, LSRS, LSRS, LSRS, LSRS, LSRS, LSRS, LSRS, LSRS}, }; @@ -1828,7 +1830,7 @@ TEST_F(P2PTransportChannelTest, RemoteCandidatesWithoutUfragPwd) { ConfigureEndpoints(env, OPEN, OPEN, webrtc::kDefaultPortAllocatorFlags, webrtc::kDefaultPortAllocatorFlags); CreateChannels(env); - const Connection* selected_connection = NULL; + const Connection* selected_connection = nullptr; // Wait until the callee's connections are created. EXPECT_THAT( webrtc::WaitUntil( diff --git a/p2p/base/port.cc b/p2p/base/port.cc index e7feb8eecf..f488e61106 100644 --- a/p2p/base/port.cc +++ b/p2p/base/port.cc @@ -209,7 +209,7 @@ Connection* Port::GetConnection(const SocketAddress& remote_addr) { if (iter != connections_.end()) return iter->second; else - return NULL; + return nullptr; } void Port::AddAddress(const SocketAddress& address, @@ -401,8 +401,8 @@ bool Port::GetStunMessage(const char* data, // NOTE: This could clearly be optimized to avoid allocating any memory. // However, at the data rates we'll be looking at on the client side, // this probably isn't worth worrying about. - RTC_DCHECK(out_msg != NULL); - RTC_DCHECK(out_username != NULL); + RTC_DCHECK(out_msg != nullptr); + RTC_DCHECK(out_username != nullptr); out_username->clear(); // Don't bother parsing the packet if we can tell it's not STUN. @@ -609,7 +609,7 @@ bool Port::ParseStunUsername(const StunMessage* stun_msg, remote_ufrag->clear(); const StunByteStringAttribute* username_attr = stun_msg->GetByteString(STUN_ATTR_USERNAME); - if (username_attr == NULL) + if (username_attr == nullptr) return false; // RFRAG:LFRAG diff --git a/p2p/base/port_unittest.cc b/p2p/base/port_unittest.cc index 900296a708..72db2f8ca1 100644 --- a/p2p/base/port_unittest.cc +++ b/p2p/base/port_unittest.cc @@ -334,7 +334,7 @@ class TestChannel : public sigslot::has_slots<> { conn_->SignalDestroyed.disconnect(this); conn_ = nullptr; } - ASSERT_TRUE(remote_request_.get() != NULL); + ASSERT_TRUE(remote_request_.get() != nullptr); Candidate c = remote_candidate; c.set_address(remote_address_); conn_ = port_->CreateConnection(c, Port::ORIGIN_MESSAGE); @@ -375,9 +375,9 @@ class TestChannel : public sigslot::has_slots<> { msg->GetByteString(STUN_ATTR_MESSAGE_INTEGRITY); const StunUInt32Attribute* fingerprint_attr = msg->GetUInt32(STUN_ATTR_FINGERPRINT); - EXPECT_TRUE(priority_attr != NULL); - EXPECT_TRUE(mi_attr != NULL); - EXPECT_TRUE(fingerprint_attr != NULL); + EXPECT_TRUE(priority_attr != nullptr); + EXPECT_TRUE(mi_attr != nullptr); + EXPECT_TRUE(fingerprint_attr != nullptr); remote_address_ = addr; remote_request_ = CopyStunMessage(*msg); remote_frag_ = rf; @@ -994,7 +994,7 @@ void PortTest::TestConnectivity(absl::string_view name1, // Send a ping from src to dst. This may or may not make it. ch1.CreateConnection(GetCandidate(ch2.port())); - ASSERT_TRUE(ch1.conn() != NULL); + ASSERT_TRUE(ch1.conn() != nullptr); EXPECT_THAT( webrtc::WaitUntil( [&] { return ch1.conn()->connected(); }, IsTrue(), @@ -1017,7 +1017,7 @@ void PortTest::TestConnectivity(absl::string_view name1, // Send a ping from dst to src. ch2.AcceptConnection(GetCandidate(ch1.port())); - ASSERT_TRUE(ch2.conn() != NULL); + ASSERT_TRUE(ch2.conn() != nullptr); ch2.Ping(); EXPECT_THAT( webrtc::WaitUntil( @@ -1033,7 +1033,7 @@ void PortTest::TestConnectivity(absl::string_view name1, // Send a ping from dst to src. Again, this may or may not make it. ch2.CreateConnection(GetCandidate(ch1.port())); - ASSERT_TRUE(ch2.conn() != NULL); + ASSERT_TRUE(ch2.conn() != nullptr); ch2.Ping(); SIMULATED_WAIT(ch2.conn()->write_state() == Connection::STATE_WRITABLE, kShortTimeout, clock); @@ -1073,7 +1073,7 @@ void PortTest::TestConnectivity(absl::string_view name1, // Pick up the actual address and establish the connection. ch2.AcceptConnection(GetCandidate(ch1.port())); - ASSERT_TRUE(ch2.conn() != NULL); + ASSERT_TRUE(ch2.conn() != nullptr); ch2.Ping(); EXPECT_THAT( webrtc::WaitUntil( @@ -1108,8 +1108,8 @@ void PortTest::TestConnectivity(absl::string_view name1, } // Everything should be good, unless we know the situation is impossible. - ASSERT_TRUE(ch1.conn() != NULL); - ASSERT_TRUE(ch2.conn() != NULL); + ASSERT_TRUE(ch1.conn() != nullptr); + ASSERT_TRUE(ch2.conn() != nullptr); if (possible) { EXPECT_TRUE(ch1.conn()->receiving()); EXPECT_EQ(Connection::STATE_WRITABLE, ch1.conn()->write_state()); @@ -1138,15 +1138,15 @@ void PortTest::TestConnectivity(absl::string_view name1, class FakePacketSocketFactory : public PacketSocketFactory { public: FakePacketSocketFactory() - : next_udp_socket_(NULL), next_server_tcp_socket_(NULL) {} + : next_udp_socket_(nullptr), next_server_tcp_socket_(nullptr) {} ~FakePacketSocketFactory() override {} AsyncPacketSocket* CreateUdpSocket(const SocketAddress& address, uint16_t min_port, uint16_t max_port) override { - EXPECT_TRUE(next_udp_socket_ != NULL); + EXPECT_TRUE(next_udp_socket_ != nullptr); AsyncPacketSocket* result = next_udp_socket_; - next_udp_socket_ = NULL; + next_udp_socket_ = nullptr; return result; } @@ -1154,9 +1154,9 @@ class FakePacketSocketFactory : public PacketSocketFactory { uint16_t min_port, uint16_t max_port, int opts) override { - EXPECT_TRUE(next_server_tcp_socket_ != NULL); + EXPECT_TRUE(next_server_tcp_socket_ != nullptr); AsyncListenSocket* result = next_server_tcp_socket_; - next_server_tcp_socket_ = NULL; + next_server_tcp_socket_ = nullptr; return result; } @@ -1761,7 +1761,7 @@ TEST_F(PortTest, TestTcpNoDelay) { webrtc::IsRtcOk()); // Connect and send a ping from src to dst. ch1.CreateConnection(GetCandidate(ch2.port())); - ASSERT_TRUE(ch1.conn() != NULL); + ASSERT_TRUE(ch1.conn() != nullptr); EXPECT_THAT( webrtc::WaitUntil( [&] { return ch1.conn()->connected(); }, IsTrue(), @@ -1772,7 +1772,7 @@ TEST_F(PortTest, TestTcpNoDelay) { // Accept the connection. ch2.AcceptConnection(GetCandidate(ch1.port())); - ASSERT_TRUE(ch2.conn() != NULL); + ASSERT_TRUE(ch2.conn() != nullptr); option_value = -1; success = static_cast(ch1.conn()) @@ -1878,11 +1878,11 @@ void PortTest::TestCrossFamilyPorts(int type) { } Connection* c = ports[0]->CreateConnection(GetCandidate(ports[2].get()), Port::ORIGIN_MESSAGE); - EXPECT_TRUE(NULL == c); + EXPECT_TRUE(nullptr == c); EXPECT_EQ(0U, ports[0]->connections().size()); c = ports[0]->CreateConnection(GetCandidate(ports[1].get()), Port::ORIGIN_MESSAGE); - EXPECT_FALSE(NULL == c); + EXPECT_FALSE(nullptr == c); EXPECT_EQ(1U, ports[0]->connections().size()); // IPv6 Port, connects to IPv4 candidate and to IPv6 candidate. @@ -1892,11 +1892,11 @@ void PortTest::TestCrossFamilyPorts(int type) { } c = ports[2]->CreateConnection(GetCandidate(ports[0].get()), Port::ORIGIN_MESSAGE); - EXPECT_TRUE(NULL == c); + EXPECT_TRUE(nullptr == c); EXPECT_EQ(0U, ports[2]->connections().size()); c = ports[2]->CreateConnection(GetCandidate(ports[3].get()), Port::ORIGIN_MESSAGE); - EXPECT_FALSE(NULL == c); + EXPECT_FALSE(nullptr == c); EXPECT_EQ(1U, ports[2]->connections().size()); } @@ -1911,10 +1911,10 @@ TEST_F(PortTest, TestSkipCrossFamilyUdp) { void PortTest::ExpectPortsCanConnect(bool can_connect, Port* p1, Port* p2) { Connection* c = p1->CreateConnection(GetCandidate(p2), Port::ORIGIN_MESSAGE); if (can_connect) { - EXPECT_FALSE(NULL == c); + EXPECT_FALSE(nullptr == c); EXPECT_EQ(1U, p1->connections().size()); } else { - EXPECT_TRUE(NULL == c); + EXPECT_TRUE(nullptr == c); EXPECT_EQ(0U, p1->connections().size()); } } @@ -2043,27 +2043,27 @@ TEST_F(PortTest, TestSendStunMessage) { EXPECT_FALSE(msg->IsLegacy()); const StunByteStringAttribute* username_attr = msg->GetByteString(STUN_ATTR_USERNAME); - ASSERT_TRUE(username_attr != NULL); + ASSERT_TRUE(username_attr != nullptr); const StunUInt32Attribute* priority_attr = msg->GetUInt32(STUN_ATTR_PRIORITY); - ASSERT_TRUE(priority_attr != NULL); + ASSERT_TRUE(priority_attr != nullptr); EXPECT_EQ(kDefaultPrflxPriority, priority_attr->value()); EXPECT_EQ("rfrag:lfrag", username_attr->string_view()); - EXPECT_TRUE(msg->GetByteString(STUN_ATTR_MESSAGE_INTEGRITY) != NULL); + EXPECT_TRUE(msg->GetByteString(STUN_ATTR_MESSAGE_INTEGRITY) != nullptr); EXPECT_EQ(StunMessage::IntegrityStatus::kIntegrityOk, msg->ValidateMessageIntegrity("rpass")); const StunUInt64Attribute* ice_controlling_attr = msg->GetUInt64(STUN_ATTR_ICE_CONTROLLING); - ASSERT_TRUE(ice_controlling_attr != NULL); + ASSERT_TRUE(ice_controlling_attr != nullptr); EXPECT_EQ(lport->IceTiebreaker(), ice_controlling_attr->value()); - EXPECT_TRUE(msg->GetByteString(STUN_ATTR_ICE_CONTROLLED) == NULL); - EXPECT_TRUE(msg->GetByteString(STUN_ATTR_USE_CANDIDATE) != NULL); - EXPECT_TRUE(msg->GetUInt32(STUN_ATTR_FINGERPRINT) != NULL); + EXPECT_TRUE(msg->GetByteString(STUN_ATTR_ICE_CONTROLLED) == nullptr); + EXPECT_TRUE(msg->GetByteString(STUN_ATTR_USE_CANDIDATE) != nullptr); + EXPECT_TRUE(msg->GetUInt32(STUN_ATTR_FINGERPRINT) != nullptr); EXPECT_TRUE(StunMessage::ValidateFingerprint( reinterpret_cast(lport->last_stun_buf().data()), lport->last_stun_buf().size())); // Request should not include ping count. - ASSERT_TRUE(msg->GetUInt32(STUN_ATTR_RETRANSMIT_COUNT) == NULL); + ASSERT_TRUE(msg->GetUInt32(STUN_ATTR_RETRANSMIT_COUNT) == nullptr); // Save a copy of the BINDING-REQUEST for use below. std::unique_ptr request = CopyStunMessage(*msg); @@ -2072,7 +2072,7 @@ TEST_F(PortTest, TestSendStunMessage) { rconn->OnReadPacket( ReceivedIpPacket(lport->last_stun_buf(), SocketAddress(), std::nullopt)); msg = rport->last_stun_msg(); - ASSERT_TRUE(msg != NULL); + ASSERT_TRUE(msg != nullptr); EXPECT_EQ(STUN_BINDING_RESPONSE, msg->type()); // Received a BINDING-RESPONSE. lconn->OnReadPacket( @@ -2088,25 +2088,25 @@ TEST_F(PortTest, TestSendStunMessage) { EXPECT_FALSE(msg->IsLegacy()); const StunAddressAttribute* addr_attr = msg->GetAddress(STUN_ATTR_XOR_MAPPED_ADDRESS); - ASSERT_TRUE(addr_attr != NULL); + ASSERT_TRUE(addr_attr != nullptr); EXPECT_EQ(lport->Candidates()[0].address(), addr_attr->GetAddress()); - EXPECT_TRUE(msg->GetByteString(STUN_ATTR_MESSAGE_INTEGRITY) != NULL); + EXPECT_TRUE(msg->GetByteString(STUN_ATTR_MESSAGE_INTEGRITY) != nullptr); EXPECT_EQ(StunMessage::IntegrityStatus::kIntegrityOk, msg->ValidateMessageIntegrity("rpass")); - EXPECT_TRUE(msg->GetUInt32(STUN_ATTR_FINGERPRINT) != NULL); + EXPECT_TRUE(msg->GetUInt32(STUN_ATTR_FINGERPRINT) != nullptr); EXPECT_TRUE(StunMessage::ValidateFingerprint( reinterpret_cast(lport->last_stun_buf().data()), lport->last_stun_buf().size())); // No USERNAME or PRIORITY in ICE responses. - EXPECT_TRUE(msg->GetByteString(STUN_ATTR_USERNAME) == NULL); - EXPECT_TRUE(msg->GetByteString(STUN_ATTR_PRIORITY) == NULL); - EXPECT_TRUE(msg->GetByteString(STUN_ATTR_MAPPED_ADDRESS) == NULL); - EXPECT_TRUE(msg->GetByteString(STUN_ATTR_ICE_CONTROLLING) == NULL); - EXPECT_TRUE(msg->GetByteString(STUN_ATTR_ICE_CONTROLLED) == NULL); - EXPECT_TRUE(msg->GetByteString(STUN_ATTR_USE_CANDIDATE) == NULL); + EXPECT_TRUE(msg->GetByteString(STUN_ATTR_USERNAME) == nullptr); + EXPECT_TRUE(msg->GetByteString(STUN_ATTR_PRIORITY) == nullptr); + EXPECT_TRUE(msg->GetByteString(STUN_ATTR_MAPPED_ADDRESS) == nullptr); + EXPECT_TRUE(msg->GetByteString(STUN_ATTR_ICE_CONTROLLING) == nullptr); + EXPECT_TRUE(msg->GetByteString(STUN_ATTR_ICE_CONTROLLED) == nullptr); + EXPECT_TRUE(msg->GetByteString(STUN_ATTR_USE_CANDIDATE) == nullptr); // Response should not include ping count. - ASSERT_TRUE(msg->GetUInt32(STUN_ATTR_RETRANSMIT_COUNT) == NULL); + ASSERT_TRUE(msg->GetUInt32(STUN_ATTR_RETRANSMIT_COUNT) == nullptr); // Respond with a BINDING-ERROR-RESPONSE. This wouldn't happen in real life, // but we can do it here. @@ -2114,23 +2114,23 @@ TEST_F(PortTest, TestSendStunMessage) { request.get(), lport->Candidates()[0].address(), STUN_ERROR_SERVER_ERROR, STUN_ERROR_REASON_SERVER_ERROR); msg = rport->last_stun_msg(); - ASSERT_TRUE(msg != NULL); + ASSERT_TRUE(msg != nullptr); EXPECT_EQ(STUN_BINDING_ERROR_RESPONSE, msg->type()); EXPECT_FALSE(msg->IsLegacy()); const StunErrorCodeAttribute* error_attr = msg->GetErrorCode(); - ASSERT_TRUE(error_attr != NULL); + ASSERT_TRUE(error_attr != nullptr); EXPECT_EQ(STUN_ERROR_SERVER_ERROR, error_attr->code()); EXPECT_EQ(std::string(STUN_ERROR_REASON_SERVER_ERROR), error_attr->reason()); - EXPECT_TRUE(msg->GetByteString(STUN_ATTR_MESSAGE_INTEGRITY) != NULL); + EXPECT_TRUE(msg->GetByteString(STUN_ATTR_MESSAGE_INTEGRITY) != nullptr); EXPECT_EQ(StunMessage::IntegrityStatus::kIntegrityOk, msg->ValidateMessageIntegrity("rpass")); - EXPECT_TRUE(msg->GetUInt32(STUN_ATTR_FINGERPRINT) != NULL); + EXPECT_TRUE(msg->GetUInt32(STUN_ATTR_FINGERPRINT) != nullptr); EXPECT_TRUE(StunMessage::ValidateFingerprint( reinterpret_cast(lport->last_stun_buf().data()), lport->last_stun_buf().size())); // No USERNAME with ICE. - EXPECT_TRUE(msg->GetByteString(STUN_ATTR_USERNAME) == NULL); - EXPECT_TRUE(msg->GetByteString(STUN_ATTR_PRIORITY) == NULL); + EXPECT_TRUE(msg->GetByteString(STUN_ATTR_USERNAME) == nullptr); + EXPECT_TRUE(msg->GetByteString(STUN_ATTR_PRIORITY) == nullptr); // Testing STUN binding requests from rport --> lport, having ICE_CONTROLLED // and (incremented) RETRANSMIT_COUNT attributes. @@ -2147,14 +2147,14 @@ TEST_F(PortTest, TestSendStunMessage) { EXPECT_EQ(STUN_BINDING_REQUEST, msg->type()); const StunUInt64Attribute* ice_controlled_attr = msg->GetUInt64(STUN_ATTR_ICE_CONTROLLED); - ASSERT_TRUE(ice_controlled_attr != NULL); + ASSERT_TRUE(ice_controlled_attr != nullptr); EXPECT_EQ(rport->IceTiebreaker(), ice_controlled_attr->value()); - EXPECT_TRUE(msg->GetByteString(STUN_ATTR_USE_CANDIDATE) == NULL); + EXPECT_TRUE(msg->GetByteString(STUN_ATTR_USE_CANDIDATE) == nullptr); // Request should include ping count. const StunUInt32Attribute* retransmit_attr = msg->GetUInt32(STUN_ATTR_RETRANSMIT_COUNT); - ASSERT_TRUE(retransmit_attr != NULL); + ASSERT_TRUE(retransmit_attr != nullptr); EXPECT_EQ(2U, retransmit_attr->value()); // Respond with a BINDING-RESPONSE. @@ -2180,7 +2180,7 @@ TEST_F(PortTest, TestSendStunMessage) { // Response should include same ping count. retransmit_attr = msg->GetUInt32(STUN_ATTR_RETRANSMIT_COUNT); - ASSERT_TRUE(retransmit_attr != NULL); + ASSERT_TRUE(retransmit_attr != nullptr); EXPECT_EQ(2U, retransmit_attr->value()); } @@ -2307,10 +2307,10 @@ TEST_F(PortTest, TestUseCandidateAttribute) { IceMessage* msg = lport->last_stun_msg(); const StunUInt64Attribute* ice_controlling_attr = msg->GetUInt64(STUN_ATTR_ICE_CONTROLLING); - ASSERT_TRUE(ice_controlling_attr != NULL); + ASSERT_TRUE(ice_controlling_attr != nullptr); const StunByteStringAttribute* use_candidate_attr = msg->GetByteString(STUN_ATTR_USE_CANDIDATE); - ASSERT_TRUE(use_candidate_attr != NULL); + ASSERT_TRUE(use_candidate_attr != nullptr); } // Tests that when the network type changes, the network cost of the port will @@ -2403,7 +2403,7 @@ TEST_F(PortTest, TestNetworkInfoAttribute) { IceMessage* msg = lport->last_stun_msg(); const StunUInt32Attribute* network_info_attr = msg->GetUInt32(STUN_ATTR_GOOG_NETWORK_INFO); - ASSERT_TRUE(network_info_attr != NULL); + ASSERT_TRUE(network_info_attr != nullptr); uint32_t network_info = network_info_attr->value(); EXPECT_EQ(lnetwork_id, network_info >> 16); // Default network has unknown type and cost kNetworkCostUnknown. @@ -2423,7 +2423,7 @@ TEST_F(PortTest, TestNetworkInfoAttribute) { webrtc::IsRtcOk()); msg = rport->last_stun_msg(); network_info_attr = msg->GetUInt32(STUN_ATTR_GOOG_NETWORK_INFO); - ASSERT_TRUE(network_info_attr != NULL); + ASSERT_TRUE(network_info_attr != nullptr); network_info = network_info_attr->value(); EXPECT_EQ(rnetwork_id, network_info >> 16); EXPECT_EQ(webrtc::kNetworkCostHigh, network_info & 0xFFFF); @@ -2447,7 +2447,7 @@ TEST_F(PortTest, TestHandleStunMessage) { WriteStunMessage(*in_msg, buf.get()); EXPECT_TRUE(GetStunMessageFromBufferWriter(port.get(), buf.get(), addr, &out_msg, &username)); - EXPECT_TRUE(out_msg.get() != NULL); + EXPECT_TRUE(out_msg.get() != nullptr); EXPECT_EQ("lfrag", username); // BINDING-RESPONSE without username, with MESSAGE-INTEGRITY and FINGERPRINT. @@ -2459,7 +2459,7 @@ TEST_F(PortTest, TestHandleStunMessage) { WriteStunMessage(*in_msg, buf.get()); EXPECT_TRUE(GetStunMessageFromBufferWriter(port.get(), buf.get(), addr, &out_msg, &username)); - EXPECT_TRUE(out_msg.get() != NULL); + EXPECT_TRUE(out_msg.get() != nullptr); EXPECT_EQ("", username); // BINDING-ERROR-RESPONSE without username, with error, M-I, and FINGERPRINT. @@ -2471,9 +2471,9 @@ TEST_F(PortTest, TestHandleStunMessage) { WriteStunMessage(*in_msg, buf.get()); EXPECT_TRUE(GetStunMessageFromBufferWriter(port.get(), buf.get(), addr, &out_msg, &username)); - EXPECT_TRUE(out_msg.get() != NULL); + EXPECT_TRUE(out_msg.get() != nullptr); EXPECT_EQ("", username); - ASSERT_TRUE(out_msg->GetErrorCode() != NULL); + ASSERT_TRUE(out_msg->GetErrorCode() != nullptr); EXPECT_EQ(STUN_ERROR_SERVER_ERROR, out_msg->GetErrorCode()->code()); EXPECT_EQ(std::string(STUN_ERROR_REASON_SERVER_ERROR), out_msg->GetErrorCode()->reason()); @@ -2495,7 +2495,7 @@ TEST_F(PortTest, TestHandleStunMessageBadUsername) { WriteStunMessage(*in_msg, buf.get()); EXPECT_TRUE(GetStunMessageFromBufferWriter(port.get(), buf.get(), addr, &out_msg, &username)); - EXPECT_TRUE(out_msg.get() == NULL); + EXPECT_TRUE(out_msg.get() == nullptr); EXPECT_EQ("", username); EXPECT_EQ(STUN_ERROR_BAD_REQUEST, port->last_stun_error_code()); @@ -2506,7 +2506,7 @@ TEST_F(PortTest, TestHandleStunMessageBadUsername) { WriteStunMessage(*in_msg, buf.get()); EXPECT_TRUE(GetStunMessageFromBufferWriter(port.get(), buf.get(), addr, &out_msg, &username)); - EXPECT_TRUE(out_msg.get() == NULL); + EXPECT_TRUE(out_msg.get() == nullptr); EXPECT_EQ("", username); EXPECT_EQ(STUN_ERROR_UNAUTHORIZED, port->last_stun_error_code()); @@ -2517,7 +2517,7 @@ TEST_F(PortTest, TestHandleStunMessageBadUsername) { WriteStunMessage(*in_msg, buf.get()); EXPECT_TRUE(GetStunMessageFromBufferWriter(port.get(), buf.get(), addr, &out_msg, &username)); - EXPECT_TRUE(out_msg.get() == NULL); + EXPECT_TRUE(out_msg.get() == nullptr); EXPECT_EQ("", username); EXPECT_EQ(STUN_ERROR_UNAUTHORIZED, port->last_stun_error_code()); @@ -2528,7 +2528,7 @@ TEST_F(PortTest, TestHandleStunMessageBadUsername) { WriteStunMessage(*in_msg, buf.get()); EXPECT_TRUE(GetStunMessageFromBufferWriter(port.get(), buf.get(), addr, &out_msg, &username)); - EXPECT_TRUE(out_msg.get() == NULL); + EXPECT_TRUE(out_msg.get() == nullptr); EXPECT_EQ("", username); EXPECT_EQ(STUN_ERROR_UNAUTHORIZED, port->last_stun_error_code()); @@ -2539,7 +2539,7 @@ TEST_F(PortTest, TestHandleStunMessageBadUsername) { WriteStunMessage(*in_msg, buf.get()); EXPECT_TRUE(GetStunMessageFromBufferWriter(port.get(), buf.get(), addr, &out_msg, &username)); - EXPECT_TRUE(out_msg.get() == NULL); + EXPECT_TRUE(out_msg.get() == nullptr); EXPECT_EQ("", username); EXPECT_EQ(STUN_ERROR_UNAUTHORIZED, port->last_stun_error_code()); } @@ -2561,7 +2561,7 @@ TEST_F(PortTest, TestHandleStunMessageBadMessageIntegrity) { WriteStunMessage(*in_msg, buf.get()); EXPECT_TRUE(GetStunMessageFromBufferWriter(port.get(), buf.get(), addr, &out_msg, &username)); - EXPECT_TRUE(out_msg.get() == NULL); + EXPECT_TRUE(out_msg.get() == nullptr); EXPECT_EQ("", username); EXPECT_EQ(STUN_ERROR_BAD_REQUEST, port->last_stun_error_code()); @@ -2573,7 +2573,7 @@ TEST_F(PortTest, TestHandleStunMessageBadMessageIntegrity) { WriteStunMessage(*in_msg, buf.get()); EXPECT_TRUE(GetStunMessageFromBufferWriter(port.get(), buf.get(), addr, &out_msg, &username)); - EXPECT_TRUE(out_msg.get() == NULL); + EXPECT_TRUE(out_msg.get() == nullptr); EXPECT_EQ("", username); EXPECT_EQ(STUN_ERROR_UNAUTHORIZED, port->last_stun_error_code()); @@ -2777,7 +2777,7 @@ TEST_F(PortTest, TestHandleStunBindingIndication) { WriteStunMessage(*in_msg, buf.get()); EXPECT_TRUE(GetStunMessageFromBufferWriter(lport.get(), buf.get(), addr, &out_msg, &username)); - EXPECT_TRUE(out_msg.get() != NULL); + EXPECT_TRUE(out_msg.get() != nullptr); EXPECT_EQ(out_msg->type(), STUN_BINDING_INDICATION); EXPECT_EQ("", username); @@ -3165,7 +3165,7 @@ TEST_F(PortTest, TestWritableState) { // Send a ping from src to dst. ch1.CreateConnection(GetCandidate(ch2.port())); - ASSERT_TRUE(ch1.conn() != NULL); + ASSERT_TRUE(ch1.conn() != nullptr); EXPECT_EQ(Connection::STATE_WRITE_INIT, ch1.conn()->write_state()); // for TCP connect EXPECT_THAT( @@ -3258,7 +3258,7 @@ TEST_F(PortTest, TestWritableStateWithConfiguredThreshold) { // Send a ping from src to dst. ch1.CreateConnection(GetCandidate(ch2.port())); - ASSERT_TRUE(ch1.conn() != NULL); + ASSERT_TRUE(ch1.conn() != nullptr); ch1.Ping(); SIMULATED_WAIT(!ch2.remote_address().IsNil(), kShortTimeout, clock); @@ -3312,7 +3312,7 @@ TEST_F(PortTest, TestTimeoutForNeverWritable) { ch2.Start(); ch1.CreateConnection(GetCandidate(ch2.port())); - ASSERT_TRUE(ch1.conn() != NULL); + ASSERT_TRUE(ch1.conn() != nullptr); EXPECT_EQ(Connection::STATE_WRITE_INIT, ch1.conn()->write_state()); // Attempt to go directly to write timeout. @@ -3349,7 +3349,7 @@ TEST_F(PortTest, TestIceLiteConnectivity) { ASSERT_FALSE(ice_lite_port->Candidates().empty()); ch1.CreateConnection(GetCandidate(ice_lite_port.get())); - ASSERT_TRUE(ch1.conn() != NULL); + ASSERT_TRUE(ch1.conn() != nullptr); EXPECT_EQ(Connection::STATE_WRITE_INIT, ch1.conn()->write_state()); // Send ping from full mode client. @@ -3363,7 +3363,7 @@ TEST_F(PortTest, TestIceLiteConnectivity) { {.timeout = TimeDelta::Millis(kDefaultTimeout)}), webrtc::IsRtcOk()); IceMessage* msg = ice_full_port_ptr->last_stun_msg(); - EXPECT_TRUE(msg->GetByteString(STUN_ATTR_USE_CANDIDATE) == NULL); + EXPECT_TRUE(msg->GetByteString(STUN_ATTR_USE_CANDIDATE) == nullptr); // Respond with a BINDING-RESPONSE from litemode client. // NOTE: Ideally we should't create connection at this stage from lite @@ -3399,7 +3399,7 @@ TEST_F(PortTest, TestIceLiteConnectivity) { {.timeout = TimeDelta::Millis(kDefaultTimeout)}), webrtc::IsRtcOk()); msg = ice_full_port_ptr->last_stun_msg(); - EXPECT_TRUE(msg->GetByteString(STUN_ATTR_USE_CANDIDATE) != NULL); + EXPECT_TRUE(msg->GetByteString(STUN_ATTR_USE_CANDIDATE) != nullptr); ch1.Stop(); } @@ -3476,7 +3476,7 @@ TEST_P(GoogPingTest, TestGoogPingAnnounceEnable) { ASSERT_FALSE(port2->Candidates().empty()); ch1.CreateConnection(GetCandidate(port2.get())); - ASSERT_TRUE(ch1.conn() != NULL); + ASSERT_TRUE(ch1.conn() != nullptr); EXPECT_EQ(Connection::STATE_WRITE_INIT, ch1.conn()->write_state()); ch1.conn()->SetIceFieldTrials(&trials); @@ -3575,7 +3575,7 @@ TEST_F(PortTest, TestGoogPingUnsupportedVersionInStunBinding) { ASSERT_FALSE(port2->Candidates().empty()); ch1.CreateConnection(GetCandidate(port2.get())); - ASSERT_TRUE(ch1.conn() != NULL); + ASSERT_TRUE(ch1.conn() != nullptr); EXPECT_EQ(Connection::STATE_WRITE_INIT, ch1.conn()->write_state()); ch1.conn()->SetIceFieldTrials(&trials); @@ -3650,7 +3650,7 @@ TEST_F(PortTest, TestGoogPingUnsupportedVersionInStunBindingResponse) { ASSERT_FALSE(port2->Candidates().empty()); ch1.CreateConnection(GetCandidate(port2.get())); - ASSERT_TRUE(ch1.conn() != NULL); + ASSERT_TRUE(ch1.conn() != nullptr); EXPECT_EQ(Connection::STATE_WRITE_INIT, ch1.conn()->write_state()); ch1.conn()->SetIceFieldTrials(&trials); @@ -3854,7 +3854,7 @@ TEST_F(PortTest, TestErrorResponseMakesGoogPingFallBackToStunBinding) { ASSERT_FALSE(port2->Candidates().empty()); ch1.CreateConnection(GetCandidate(port2.get())); - ASSERT_TRUE(ch1.conn() != NULL); + ASSERT_TRUE(ch1.conn() != nullptr); EXPECT_EQ(Connection::STATE_WRITE_INIT, ch1.conn()->write_state()); ch1.conn()->SetIceFieldTrials(&trials); diff --git a/p2p/base/pseudo_tcp.cc b/p2p/base/pseudo_tcp.cc index bef1a51dbf..1470fd0b36 100644 --- a/p2p/base/pseudo_tcp.cc +++ b/p2p/base/pseudo_tcp.cc @@ -915,7 +915,7 @@ bool PseudoTcp::process(Segment& seg) { } else { uint32_t nOffset = seg.seq - m_rcv_nxt; - if (!m_rbuf.WriteOffset(seg.data, seg.len, nOffset, NULL)) { + if (!m_rbuf.WriteOffset(seg.data, seg.len, nOffset, nullptr)) { // Ignore incoming packets outside of the receive window. return false; } diff --git a/p2p/base/pseudo_tcp_unittest.cc b/p2p/base/pseudo_tcp_unittest.cc index 1c3ae2118f..cdb794a2ba 100644 --- a/p2p/base/pseudo_tcp_unittest.cc +++ b/p2p/base/pseudo_tcp_unittest.cc @@ -358,8 +358,8 @@ class PseudoTcpTestPingPong : public PseudoTcpTestBase { public: PseudoTcpTestPingPong() : iterations_remaining_(0), - sender_(NULL), - receiver_(NULL), + sender_(nullptr), + receiver_(nullptr), bytes_per_send_(0) {} void SetBytesPerSend(int bytes) { bytes_per_send_ = bytes; } void TestPingPong(int size, int iterations) { diff --git a/p2p/base/stun_port_unittest.cc b/p2p/base/stun_port_unittest.cc index f0d08739fd..809c435b78 100644 --- a/p2p/base/stun_port_unittest.cc +++ b/p2p/base/stun_port_unittest.cc @@ -225,7 +225,7 @@ class StunPortTestBase : public ::testing::Test, public sigslot::has_slots<> { socket_.reset(socket_factory()->CreateUdpSocket( webrtc::SocketAddress(kPrivateIP.ipaddr(), 0), 0, 0)); } - ASSERT_TRUE(socket_ != NULL); + ASSERT_TRUE(socket_ != nullptr); socket_->RegisterReceivedPacketCallback( [&](webrtc::AsyncPacketSocket* socket, const webrtc::ReceivedIpPacket& packet) { @@ -242,7 +242,7 @@ class StunPortTestBase : public ::testing::Test, public sigslot::has_slots<> { .ice_password = webrtc::CreateRandomString(22)}, socket_.get(), false, std::nullopt); stun_port_->set_server_addresses(stun_servers); - ASSERT_TRUE(stun_port_ != NULL); + ASSERT_TRUE(stun_port_ != nullptr); stun_port_->SetIceTiebreaker(kTiebreakerDefault); stun_port_->SignalPortComplete.connect(this, &StunPortTestBase::OnPortComplete); @@ -270,7 +270,7 @@ class StunPortTestBase : public ::testing::Test, public sigslot::has_slots<> { protected: static void SetUpTestSuite() { // Ensure the RNG is inited. - webrtc::InitRandom(NULL, 0); + webrtc::InitRandom(nullptr, 0); } void OnPortComplete(webrtc::Port* /* port */) { diff --git a/p2p/base/stun_request.cc b/p2p/base/stun_request.cc index 1ff8389ff1..1a31df5b45 100644 --- a/p2p/base/stun_request.cc +++ b/p2p/base/stun_request.cc @@ -256,7 +256,7 @@ StunRequest::StunRequest(StunRequestManager& manager, StunRequest::~StunRequest() {} int StunRequest::type() { - RTC_DCHECK(msg_ != NULL); + RTC_DCHECK(msg_ != nullptr); return msg_->type(); } diff --git a/p2p/base/stun_request_unittest.cc b/p2p/base/stun_request_unittest.cc index 14f497ab5c..945d29b82e 100644 --- a/p2p/base/stun_request_unittest.cc +++ b/p2p/base/stun_request_unittest.cc @@ -55,7 +55,7 @@ class StunRequestTest : public ::testing::Test { OnSendPacket(data, size, request); }), request_count_(0), - response_(NULL), + response_(nullptr), success_(false), failure_(false), timeout_(false) {} @@ -143,7 +143,7 @@ TEST_F(StunRequestTest, TestUnexpected) { manager_.Send(request); EXPECT_FALSE(manager_.CheckResponse(res.get())); - EXPECT_TRUE(response_ == NULL); + EXPECT_TRUE(response_ == nullptr); EXPECT_FALSE(success_); EXPECT_FALSE(failure_); EXPECT_FALSE(timeout_); @@ -187,7 +187,7 @@ TEST_F(StunRequestTest, TestTimeout) { SIMULATED_WAIT(false, STUN_TOTAL_TIMEOUT, fake_clock); EXPECT_FALSE(manager_.CheckResponse(res.get())); - EXPECT_TRUE(response_ == NULL); + EXPECT_TRUE(response_ == nullptr); EXPECT_FALSE(success_); EXPECT_FALSE(failure_); EXPECT_TRUE(timeout_); diff --git a/p2p/base/tcp_port.cc b/p2p/base/tcp_port.cc index 5717f0a2d0..625ff766fd 100644 --- a/p2p/base/tcp_port.cc +++ b/p2p/base/tcp_port.cc @@ -132,31 +132,31 @@ TCPPort::~TCPPort() { Connection* TCPPort::CreateConnection(const Candidate& address, CandidateOrigin origin) { if (!SupportsProtocol(address.protocol())) { - return NULL; + return nullptr; } if ((address.tcptype() == TCPTYPE_ACTIVE_STR && !address.is_prflx()) || (address.tcptype().empty() && address.address().port() == 0)) { // It's active only candidate, we should not try to create connections // for these candidates. - return NULL; + return nullptr; } // We can't accept TCP connections incoming on other ports if (origin == ORIGIN_OTHER_PORT) - return NULL; + return nullptr; // We don't know how to act as an ssl server yet if ((address.protocol() == SSLTCP_PROTOCOL_NAME) && (origin == ORIGIN_THIS_PORT)) { - return NULL; + return nullptr; } if (!IsCompatibleAddress(address.address())) { - return NULL; + return nullptr; } - TCPConnection* conn = NULL; + TCPConnection* conn = nullptr; if (AsyncPacketSocket* socket = GetIncoming(address.address(), true)) { // Incoming connection; we already created a socket and connected signals, // so we need to hand off the "read packet" responsibility to @@ -205,7 +205,7 @@ int TCPPort::SendTo(const void* data, const SocketAddress& addr, const AsyncSocketPacketOptions& options, bool payload) { - AsyncPacketSocket* socket = NULL; + AsyncPacketSocket* socket = nullptr; TCPConnection* conn = static_cast(GetConnection(addr)); // For Connection, this is the code path used by Ping() to establish @@ -313,7 +313,7 @@ void TCPPort::TryCreateServerSocket() { AsyncPacketSocket* TCPPort::GetIncoming(const SocketAddress& addr, bool remove) { - AsyncPacketSocket* socket = NULL; + AsyncPacketSocket* socket = nullptr; for (std::list::iterator it = incoming_.begin(); it != incoming_.end(); ++it) { if (it->addr == addr) { @@ -350,7 +350,7 @@ TCPConnection::TCPConnection(WeakPtr tcp_port, : Connection(std::move(tcp_port), 0, candidate), socket_(socket), error_(0), - outgoing_(socket == NULL), + outgoing_(socket == nullptr), connection_pending_(false), pretending_to_be_writable_(false), reconnection_timeout_(CONNECTION_WRITE_CONNECT_TIMEOUT) { diff --git a/p2p/base/transport_description_factory.cc b/p2p/base/transport_description_factory.cc index f016afa466..6391e89d0d 100644 --- a/p2p/base/transport_description_factory.cc +++ b/p2p/base/transport_description_factory.cc @@ -58,7 +58,7 @@ std::unique_ptr TransportDescriptionFactory::CreateOffer( // Fail if we can't create the fingerprint. // If we are the initiator set role to "actpass". if (!SetSecurityInfo(desc.get(), CONNECTIONROLE_ACTPASS)) { - return NULL; + return nullptr; } return desc; @@ -74,7 +74,7 @@ std::unique_ptr TransportDescriptionFactory::CreateAnswer( if (!offer) { RTC_LOG(LS_WARNING) << "Failed to create TransportDescription answer " "because offer is NULL"; - return NULL; + return nullptr; } auto desc = std::make_unique(); @@ -103,7 +103,7 @@ std::unique_ptr TransportDescriptionFactory::CreateAnswer( // We require DTLS, but the other side didn't offer it. Fail. RTC_LOG(LS_WARNING) << "Failed to create TransportDescription answer " "because of incompatible security settings"; - return NULL; + return nullptr; } // This may be a bundled section, fingerprint may legitimately be missing. return desc; @@ -130,10 +130,10 @@ std::unique_ptr TransportDescriptionFactory::CreateAnswer( RTC_LOG(LS_ERROR) << "Remote offer connection role is " << role << " which is a protocol violation"; RTC_DCHECK_NOTREACHED(); - return NULL; + return nullptr; } if (!SetSecurityInfo(desc.get(), role)) { - return NULL; + return nullptr; } return desc; } diff --git a/p2p/base/transport_description_factory_unittest.cc b/p2p/base/transport_description_factory_unittest.cc index 1324a4fc76..fe081185ce 100644 --- a/p2p/base/transport_description_factory_unittest.cc +++ b/p2p/base/transport_description_factory_unittest.cc @@ -60,7 +60,7 @@ class TransportDescriptionFactoryTest : public ::testing::Test { absl::string_view ice_ufrag, absl::string_view ice_pwd, absl::string_view dtls_alg) { - ASSERT_TRUE(desc != NULL); + ASSERT_TRUE(desc != nullptr); EXPECT_EQ(!opt.empty(), desc->HasOption(opt)); if (ice_ufrag.empty() && ice_pwd.empty()) { EXPECT_EQ(static_cast(webrtc::ICE_UFRAG_LENGTH), @@ -72,9 +72,9 @@ class TransportDescriptionFactoryTest : public ::testing::Test { EXPECT_EQ(ice_pwd, desc->ice_pwd); } if (dtls_alg.empty()) { - EXPECT_TRUE(desc->identity_fingerprint.get() == NULL); + EXPECT_TRUE(desc->identity_fingerprint.get() == nullptr); } else { - ASSERT_TRUE(desc->identity_fingerprint.get() != NULL); + ASSERT_TRUE(desc->identity_fingerprint.get() != nullptr); EXPECT_EQ(desc->identity_fingerprint->algorithm, dtls_alg); EXPECT_GT(desc->identity_fingerprint->digest.size(), 0U); } @@ -94,9 +94,9 @@ class TransportDescriptionFactoryTest : public ::testing::Test { webrtc::TransportOptions options; // The initial offer / answer exchange. std::unique_ptr offer = - f1_.CreateOffer(options, NULL, &ice_credentials_); - std::unique_ptr answer = - f2_.CreateAnswer(offer.get(), options, true, NULL, &ice_credentials_); + f1_.CreateOffer(options, nullptr, &ice_credentials_); + std::unique_ptr answer = f2_.CreateAnswer( + offer.get(), options, true, nullptr, &ice_credentials_); // Create an updated offer where we restart ice. options.ice_restart = true; @@ -109,7 +109,7 @@ class TransportDescriptionFactoryTest : public ::testing::Test { // |options.ice_restart == true| std::unique_ptr restart_answer = f2_.CreateAnswer( restart_offer.get(), options, true, answer.get(), &ice_credentials_); - ASSERT_TRUE(restart_answer.get() != NULL); + ASSERT_TRUE(restart_answer.get() != nullptr); VerifyUfragAndPasswordChanged(dtls, answer.get(), restart_answer.get()); } @@ -193,7 +193,7 @@ TEST_F(TransportDescriptionFactoryTest, TestOfferDtls) { ASSERT_TRUE( cert1_->GetSSLCertificate().GetSignatureDigestAlgorithm(&digest_alg)); std::unique_ptr desc = - f1_.CreateOffer(TransportOptions(), NULL, &ice_credentials_); + f1_.CreateOffer(TransportOptions(), nullptr, &ice_credentials_); CheckDesc(desc.get(), "", "", "", digest_alg); } @@ -201,8 +201,8 @@ TEST_F(TransportDescriptionFactoryTest, TestOfferDtls) { TEST_F(TransportDescriptionFactoryTest, TestOfferDtlsWithNoIdentity) { f1_.set_certificate(nullptr); std::unique_ptr desc = - f1_.CreateOffer(TransportOptions(), NULL, &ice_credentials_); - ASSERT_TRUE(desc.get() == NULL); + f1_.CreateOffer(TransportOptions(), nullptr, &ice_credentials_); + ASSERT_TRUE(desc.get() == nullptr); } // Test updating an offer with DTLS to pick ICE. @@ -212,8 +212,8 @@ TEST_F(TransportDescriptionFactoryTest, TestOfferDtlsReofferDtls) { ASSERT_TRUE( cert1_->GetSSLCertificate().GetSignatureDigestAlgorithm(&digest_alg)); std::unique_ptr old_desc = - f1_.CreateOffer(TransportOptions(), NULL, &ice_credentials_); - ASSERT_TRUE(old_desc.get() != NULL); + f1_.CreateOffer(TransportOptions(), nullptr, &ice_credentials_); + ASSERT_TRUE(old_desc.get() != nullptr); std::unique_ptr desc = f1_.CreateOffer(TransportOptions(), old_desc.get(), &ice_credentials_); CheckDesc(desc.get(), "", old_desc->ice_ufrag, old_desc->ice_pwd, digest_alg); @@ -224,12 +224,12 @@ TEST_F(TransportDescriptionFactoryTest, TestAnswerDefault) { ASSERT_TRUE( cert1_->GetSSLCertificate().GetSignatureDigestAlgorithm(&digest_alg)); std::unique_ptr offer = - f1_.CreateOffer(TransportOptions(), NULL, &ice_credentials_); - ASSERT_TRUE(offer.get() != NULL); + f1_.CreateOffer(TransportOptions(), nullptr, &ice_credentials_); + ASSERT_TRUE(offer.get() != nullptr); std::unique_ptr desc = f2_.CreateAnswer( - offer.get(), TransportOptions(), true, NULL, &ice_credentials_); + offer.get(), TransportOptions(), true, nullptr, &ice_credentials_); CheckDesc(desc.get(), "", "", "", digest_alg); - desc = f2_.CreateAnswer(offer.get(), TransportOptions(), true, NULL, + desc = f2_.CreateAnswer(offer.get(), TransportOptions(), true, nullptr, &ice_credentials_); CheckDesc(desc.get(), "", "", "", digest_alg); } @@ -240,14 +240,14 @@ TEST_F(TransportDescriptionFactoryTest, TestReanswer) { ASSERT_TRUE( cert1_->GetSSLCertificate().GetSignatureDigestAlgorithm(&digest_alg)); std::unique_ptr offer = - f1_.CreateOffer(TransportOptions(), NULL, &ice_credentials_); - ASSERT_TRUE(offer.get() != NULL); + f1_.CreateOffer(TransportOptions(), nullptr, &ice_credentials_); + ASSERT_TRUE(offer.get() != nullptr); std::unique_ptr old_desc = f2_.CreateAnswer( - offer.get(), TransportOptions(), true, NULL, &ice_credentials_); - ASSERT_TRUE(old_desc.get() != NULL); + offer.get(), TransportOptions(), true, nullptr, &ice_credentials_); + ASSERT_TRUE(old_desc.get() != nullptr); std::unique_ptr desc = f2_.CreateAnswer( offer.get(), TransportOptions(), true, old_desc.get(), &ice_credentials_); - ASSERT_TRUE(desc.get() != NULL); + ASSERT_TRUE(desc.get() != nullptr); CheckDesc(desc.get(), "", old_desc->ice_ufrag, old_desc->ice_pwd, digest_alg); } @@ -256,10 +256,10 @@ TEST_F(TransportDescriptionFactoryTest, TestAnswerDtlsToNoDtls) { f2_.SetInsecureForTesting(); f2_.set_certificate(nullptr); std::unique_ptr offer = - f1_.CreateOffer(TransportOptions(), NULL, &ice_credentials_); - ASSERT_TRUE(offer.get() != NULL); + f1_.CreateOffer(TransportOptions(), nullptr, &ice_credentials_); + ASSERT_TRUE(offer.get() != nullptr); std::unique_ptr desc = f2_.CreateAnswer( - offer.get(), TransportOptions(), true, NULL, &ice_credentials_); + offer.get(), TransportOptions(), true, nullptr, &ice_credentials_); CheckDesc(desc.get(), "", "", "", ""); } @@ -269,15 +269,15 @@ TEST_F(TransportDescriptionFactoryTest, TestAnswerNoDtlsToDtls) { f1_.SetInsecureForTesting(); f1_.set_certificate(nullptr); std::unique_ptr offer = - f1_.CreateOffer(TransportOptions(), NULL, &ice_credentials_); - ASSERT_TRUE(offer.get() != NULL); + f1_.CreateOffer(TransportOptions(), nullptr, &ice_credentials_); + ASSERT_TRUE(offer.get() != nullptr); // Normal case. std::unique_ptr desc = f2_.CreateAnswer( - offer.get(), TransportOptions(), true, NULL, &ice_credentials_); - ASSERT_TRUE(desc.get() == NULL); + offer.get(), TransportOptions(), true, nullptr, &ice_credentials_); + ASSERT_TRUE(desc.get() == nullptr); // Insecure case. f2_.SetInsecureForTesting(); - desc = f2_.CreateAnswer(offer.get(), TransportOptions(), true, NULL, + desc = f2_.CreateAnswer(offer.get(), TransportOptions(), true, nullptr, &ice_credentials_); CheckDesc(desc.get(), "", "", "", ""); } @@ -292,14 +292,14 @@ TEST_F(TransportDescriptionFactoryTest, TestAnswerDtlsToDtls) { cert2_->GetSSLCertificate().GetSignatureDigestAlgorithm(&digest_alg2)); std::unique_ptr offer = - f1_.CreateOffer(TransportOptions(), NULL, &ice_credentials_); - ASSERT_TRUE(offer.get() != NULL); + f1_.CreateOffer(TransportOptions(), nullptr, &ice_credentials_); + ASSERT_TRUE(offer.get() != nullptr); std::unique_ptr desc = f2_.CreateAnswer( - offer.get(), TransportOptions(), true, NULL, &ice_credentials_); + offer.get(), TransportOptions(), true, nullptr, &ice_credentials_); CheckDesc(desc.get(), "", "", "", digest_alg2); f2_.SetInsecureForTesting(); - desc = f2_.CreateAnswer(offer.get(), TransportOptions(), true, NULL, + desc = f2_.CreateAnswer(offer.get(), TransportOptions(), true, nullptr, &ice_credentials_); CheckDesc(desc.get(), "", "", "", digest_alg2); } diff --git a/p2p/base/turn_port_unittest.cc b/p2p/base/turn_port_unittest.cc index dea2713cfc..d8f4505665 100644 --- a/p2p/base/turn_port_unittest.cc +++ b/p2p/base/turn_port_unittest.cc @@ -163,7 +163,7 @@ static int GetFDCount() { struct dirent* dp; int fd_count = 0; DIR* dir = opendir("/proc/self/fd/"); - while ((dp = readdir(dir)) != NULL) { + while ((dp = readdir(dir)) != nullptr) { if (dp->d_name[0] == '.') continue; ++fd_count; @@ -344,7 +344,7 @@ class TurnPortTest : public ::testing::Test, if (!socket_) { socket_.reset(socket_factory()->CreateUdpSocket( SocketAddress(kLocalAddr1.ipaddr(), 0), 0, 0)); - ASSERT_TRUE(socket_ != NULL); + ASSERT_TRUE(socket_ != nullptr); socket_->RegisterReceivedPacketCallback( [&](AsyncPacketSocket* socket, const ReceivedIpPacket& packet) { OnSocketReadPacket(socket, packet); @@ -660,7 +660,7 @@ class TurnPortTest : public ::testing::Test, ASSERT_GE(turn_port_->Candidates().size(), 1U); Connection* conn1 = udp_port_->CreateConnection(turn_port_->Candidates()[0], Port::ORIGIN_MESSAGE); - ASSERT_TRUE(conn1 != NULL); + ASSERT_TRUE(conn1 != nullptr); conn1->Ping(0); SIMULATED_WAIT(!turn_unknown_address_, kSimulatedRtt * 2, fake_clock_); EXPECT_FALSE(turn_unknown_address_); @@ -670,7 +670,7 @@ class TurnPortTest : public ::testing::Test, // Send ping from TURN to UDP. Connection* conn2 = turn_port_->CreateConnection(udp_port_->Candidates()[0], Port::ORIGIN_MESSAGE); - ASSERT_TRUE(conn2 != NULL); + ASSERT_TRUE(conn2 != nullptr); ASSERT_THAT( WaitUntil([&] { return turn_create_permission_success_; }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt), @@ -711,7 +711,7 @@ class TurnPortTest : public ::testing::Test, // the TurnPort. turn_port_->set_timeout_delay(10 * 60 * 1000); - ASSERT_TRUE(conn2 != NULL); + ASSERT_TRUE(conn2 != nullptr); ASSERT_THAT( WaitUntil([&] { return turn_create_permission_success_; }, IsTrue(), {.timeout = TimeDelta::Millis(kSimulatedRtt), @@ -789,8 +789,8 @@ class TurnPortTest : public ::testing::Test, Port::ORIGIN_MESSAGE); Connection* conn2 = udp_port_->CreateConnection(turn_port_->Candidates()[0], Port::ORIGIN_MESSAGE); - ASSERT_TRUE(conn1 != NULL); - ASSERT_TRUE(conn2 != NULL); + ASSERT_TRUE(conn1 != nullptr); + ASSERT_TRUE(conn2 != nullptr); conn1->RegisterReceivedPacketCallback( [&](Connection* connection, const ReceivedIpPacket& packet) { turn_packets_.push_back( @@ -862,8 +862,8 @@ class TurnPortTest : public ::testing::Test, Port::ORIGIN_MESSAGE); Connection* conn2 = udp_port_->CreateConnection(turn_port_->Candidates()[0], Port::ORIGIN_MESSAGE); - ASSERT_TRUE(conn1 != NULL); - ASSERT_TRUE(conn2 != NULL); + ASSERT_TRUE(conn1 != nullptr); + ASSERT_TRUE(conn2 != nullptr); conn1->RegisterReceivedPacketCallback( [&](Connection* connection, const ReceivedIpPacket& packet) { turn_packets_.push_back( @@ -1466,8 +1466,8 @@ TEST_F(TurnPortTest, TestStopProcessingPacketsAfterClosed) { Port::ORIGIN_MESSAGE); Connection* conn2 = udp_port_->CreateConnection(turn_port_->Candidates()[0], Port::ORIGIN_MESSAGE); - ASSERT_TRUE(conn1 != NULL); - ASSERT_TRUE(conn2 != NULL); + ASSERT_TRUE(conn1 != nullptr); + ASSERT_TRUE(conn2 != nullptr); // Make sure conn2 is writable. conn2->Ping(0); EXPECT_THAT(WaitUntil([&] { return conn2->write_state(); }, @@ -1494,13 +1494,13 @@ TEST_F(TurnPortTest, TestCreateConnectionWhenSocketClosed) { // Create a connection. Connection* conn1 = turn_port_->CreateConnection(udp_port_->Candidates()[0], Port::ORIGIN_MESSAGE); - ASSERT_TRUE(conn1 != NULL); + ASSERT_TRUE(conn1 != nullptr); // Close the socket and create a connection again. turn_port_->OnSocketClose(turn_port_->socket(), 1); conn1 = turn_port_->CreateConnection(udp_port_->Candidates()[0], Port::ORIGIN_MESSAGE); - ASSERT_TRUE(conn1 == NULL); + ASSERT_TRUE(conn1 == nullptr); } // Tests that when a TCP socket is closed, the respective TURN connection will @@ -1660,7 +1660,7 @@ TEST_F(TurnPortTest, TestRefreshCreatePermissionRequest) { Connection* conn = turn_port_->CreateConnection(udp_port_->Candidates()[0], Port::ORIGIN_MESSAGE); - ASSERT_TRUE(conn != NULL); + ASSERT_TRUE(conn != nullptr); EXPECT_THAT( WaitUntil( [&] { return turn_create_permission_success_; }, IsTrue(), diff --git a/p2p/client/basic_port_allocator.cc b/p2p/client/basic_port_allocator.cc index 3e04332aa4..fa12319c6c 100644 --- a/p2p/client/basic_port_allocator.cc +++ b/p2p/client/basic_port_allocator.cc @@ -955,7 +955,7 @@ void BasicPortAllocatorSession::OnCandidateReady(Port* port, const Candidate& c) { RTC_DCHECK_RUN_ON(network_thread_); PortData* data = FindPort(port); - RTC_DCHECK(data != NULL); + RTC_DCHECK(data != nullptr); RTC_LOG(LS_INFO) << port->ToString() << ": Gathered candidate: " << c.ToSensitiveString(); // Discarding any candidate signal if port allocation status is @@ -1104,7 +1104,7 @@ void BasicPortAllocatorSession::OnPortComplete(Port* port) { RTC_LOG(LS_INFO) << port->ToString() << ": Port completed gathering candidates."; PortData* data = FindPort(port); - RTC_DCHECK(data != NULL); + RTC_DCHECK(data != nullptr); // Ignore any late signals. if (!data->inprogress()) { @@ -1122,7 +1122,7 @@ void BasicPortAllocatorSession::OnPortError(Port* port) { RTC_LOG(LS_INFO) << port->ToString() << ": Port encountered error while gathering candidates."; PortData* data = FindPort(port); - RTC_DCHECK(data != NULL); + RTC_DCHECK(data != nullptr); // We might have already given up on this port and stopped it. if (!data->inprogress()) { return; @@ -1209,7 +1209,7 @@ BasicPortAllocatorSession::PortData* BasicPortAllocatorSession::FindPort( return &*it; } } - return NULL; + return nullptr; } std::vector @@ -1270,7 +1270,7 @@ AllocationSequence::AllocationSequence( state_(kInit), flags_(flags), udp_socket_(), - udp_port_(NULL), + udp_port_(nullptr), phase_(0), port_allocation_complete_callback_( std::move(port_allocation_complete_callback)) {} @@ -1293,7 +1293,7 @@ void AllocationSequence::Init() { void AllocationSequence::Clear() { TRACE_EVENT0("webrtc", "AllocationSequence::Clear"); - udp_port_ = NULL; + udp_port_ = nullptr; relay_ports_.clear(); } @@ -1650,7 +1650,7 @@ void AllocationSequence::CreateTurnPort(const RelayServerConfig& config, continue; } } - RTC_DCHECK(port != NULL); + RTC_DCHECK(port != nullptr); port->SetIceTiebreaker(session_->allocator()->ice_tiebreaker()); session_->AddAllocatedPort(port.release(), this); } @@ -1692,7 +1692,7 @@ void AllocationSequence::OnReadPacket(AsyncPacketSocket* socket, void AllocationSequence::OnPortDestroyed(PortInterface* port) { if (udp_port_ == port) { - udp_port_ = NULL; + udp_port_ = nullptr; return; } diff --git a/p2p/dtls/dtls_transport.cc b/p2p/dtls/dtls_transport.cc index ab4f5de3ae..61a136fddb 100644 --- a/p2p/dtls/dtls_transport.cc +++ b/p2p/dtls/dtls_transport.cc @@ -166,7 +166,7 @@ bool StreamInterfaceChannel::OnPacketReceived(const char* data, size_t size) { if (packets_.size() > 0) { RTC_LOG(LS_WARNING) << "Packet already in queue."; } - bool ret = packets_.WriteBack(data, size, NULL); + bool ret = packets_.WriteBack(data, size, nullptr); if (!ret) { // Somehow we received another packet before the SSLStreamAdapter read the // previous one out of our temporary buffer. In this case, we'll log an diff --git a/p2p/dtls/dtls_transport_unittest.cc b/p2p/dtls/dtls_transport_unittest.cc index b8cfd06074..d7a2f7fdbf 100644 --- a/p2p/dtls/dtls_transport_unittest.cc +++ b/p2p/dtls/dtls_transport_unittest.cc @@ -369,7 +369,7 @@ class DtlsTestClient : public sigslot::has_slots<> { RTC_LOG(LS_INFO) << "Found DTLS ACK"; } else if (dtls_transport_->IsDtlsActive()) { if (IsRtpLeadByte(data[0])) { - ASSERT_TRUE(VerifyPacket(packet.payload(), NULL)); + ASSERT_TRUE(VerifyPacket(packet.payload(), nullptr)); } else if (packet_size_ && packet.payload().size() >= packet_size_) { ASSERT_TRUE(VerifyEncryptedPacket(data, packet.payload().size())); } diff --git a/p2p/test/nat_types.cc b/p2p/test/nat_types.cc index 6fbc973280..1b7504e2a3 100644 --- a/p2p/test/nat_types.cc +++ b/p2p/test/nat_types.cc @@ -54,7 +54,7 @@ NAT* NAT::Create(NATType type) { return new SymmetricNAT(); default: RTC_DCHECK_NOTREACHED(); - return 0; + return nullptr; } } diff --git a/p2p/test/nat_unittest.cc b/p2p/test/nat_unittest.cc index bb00fbacf5..b9dddc1da5 100644 --- a/p2p/test/nat_unittest.cc +++ b/p2p/test/nat_unittest.cc @@ -49,7 +49,7 @@ bool CheckReceive(webrtc::TestClient* client, bool should_receive, const char* buf, size_t size) { - return (should_receive) ? client->CheckNextPacket(buf, size, 0) + return (should_receive) ? client->CheckNextPacket(buf, size, nullptr) : client->CheckNoPacket(); } diff --git a/p2p/test/stun_server_unittest.cc b/p2p/test/stun_server_unittest.cc index 1a270d8b02..1a86a3059b 100644 --- a/p2p/test/stun_server_unittest.cc +++ b/p2p/test/stun_server_unittest.cc @@ -53,7 +53,7 @@ class StunServerTest : public ::testing::Test { } bool ReceiveFails() { return (client_->CheckNoPacket()); } StunMessage* Receive() { - StunMessage* msg = NULL; + StunMessage* msg = nullptr; std::unique_ptr packet = client_->NextPacket(TestClient::kTimeoutMs); if (packet) { @@ -78,13 +78,13 @@ TEST_F(StunServerTest, TestGood) { Send(req); StunMessage* msg = Receive(); - ASSERT_TRUE(msg != NULL); + ASSERT_TRUE(msg != nullptr); EXPECT_EQ(STUN_BINDING_RESPONSE, msg->type()); EXPECT_EQ(req.transaction_id(), msg->transaction_id()); const StunAddressAttribute* mapped_addr = msg->GetAddress(STUN_ATTR_MAPPED_ADDRESS); - EXPECT_TRUE(mapped_addr != NULL); + EXPECT_TRUE(mapped_addr != nullptr); EXPECT_EQ(1, mapped_addr->family()); EXPECT_EQ(client_addr.port(), mapped_addr->port()); @@ -99,13 +99,13 @@ TEST_F(StunServerTest, TestGoodXorMappedAddr) { Send(req); StunMessage* msg = Receive(); - ASSERT_TRUE(msg != NULL); + ASSERT_TRUE(msg != nullptr); EXPECT_EQ(STUN_BINDING_RESPONSE, msg->type()); EXPECT_EQ(req.transaction_id(), msg->transaction_id()); const StunAddressAttribute* mapped_addr = msg->GetAddress(STUN_ATTR_XOR_MAPPED_ADDRESS); - EXPECT_TRUE(mapped_addr != NULL); + EXPECT_TRUE(mapped_addr != nullptr); EXPECT_EQ(1, mapped_addr->family()); EXPECT_EQ(client_addr.port(), mapped_addr->port()); @@ -120,13 +120,13 @@ TEST_F(StunServerTest, TestNoXorMappedAddr) { Send(req); StunMessage* msg = Receive(); - ASSERT_TRUE(msg != NULL); + ASSERT_TRUE(msg != nullptr); EXPECT_EQ(STUN_BINDING_RESPONSE, msg->type()); EXPECT_EQ(req.transaction_id(), msg->transaction_id()); const StunAddressAttribute* mapped_addr = msg->GetAddress(STUN_ATTR_XOR_MAPPED_ADDRESS); - EXPECT_TRUE(mapped_addr == NULL); + EXPECT_TRUE(mapped_addr == nullptr); delete msg; } diff --git a/p2p/test/turn_server.cc b/p2p/test/turn_server.cc index 0c49e80a7d..57fb0fd271 100644 --- a/p2p/test/turn_server.cc +++ b/p2p/test/turn_server.cc @@ -89,8 +89,8 @@ static void InitErrorResponse(int code, TurnServer::TurnServer(TaskQueueBase* thread) : thread_(thread), nonce_key_(CreateRandomString(kNonceKeySize)), - auth_hook_(NULL), - redirect_hook_(NULL), + auth_hook_(nullptr), + redirect_hook_(nullptr), enable_otu_nonce_(false) {} TurnServer::~TurnServer() { @@ -152,7 +152,7 @@ void TurnServer::AcceptConnection(Socket* server_socket) { // Check if someone is trying to connect to us. SocketAddress accept_addr; Socket* accepted_socket = server_socket->Accept(&accept_addr); - if (accepted_socket != NULL) { + if (accepted_socket != nullptr) { const ServerSocketInfo& info = server_listen_sockets_[server_socket]; if (info.ssl_adapter_factory) { SSLAdapter* ssl_adapter = @@ -222,7 +222,7 @@ void TurnServer::HandleStunMessage(TurnServerConnection* conn, return; } - if (redirect_hook_ != NULL && msg.type() == STUN_ALLOCATE_REQUEST) { + if (redirect_hook_ != nullptr && msg.type() == STUN_ALLOCATE_REQUEST) { SocketAddress address; if (redirect_hook_->ShouldRedirect(conn->src(), &address)) { SendErrorResponseWithAlternateServer(conn, &msg, address); @@ -276,7 +276,7 @@ bool TurnServer::GetKey(const StunMessage* msg, std::string* key) { return false; } - return (auth_hook_ != NULL && + return (auth_hook_ != nullptr && auth_hook_->GetKey(std::string(username_attr->string_view()), realm_, key)); } @@ -429,9 +429,9 @@ TurnServerAllocation* TurnServer::CreateAllocation(TurnServerConnection* conn, AsyncPacketSocket* external_socket = (external_socket_factory_) ? external_socket_factory_->CreateUdpSocket(external_addr_, 0, 0) - : NULL; + : nullptr; if (!external_socket) { - return NULL; + return nullptr; } // The Allocation takes ownership of the socket. @@ -589,7 +589,7 @@ std::string TurnServerAllocation::ToString() const { void TurnServerAllocation::HandleTurnMessage(const TurnMessage* msg) { RTC_DCHECK_RUN_ON(thread_); - RTC_DCHECK(msg != NULL); + RTC_DCHECK(msg != nullptr); switch (msg->type()) { case STUN_ALLOCATE_REQUEST: HandleAllocateRequest(msg); @@ -619,7 +619,7 @@ void TurnServerAllocation::HandleAllocateRequest(const TurnMessage* msg) { transaction_id_ = msg->transaction_id(); const StunByteStringAttribute* username_attr = msg->GetByteString(STUN_ATTR_USERNAME); - RTC_DCHECK(username_attr != NULL); + RTC_DCHECK(username_attr != nullptr); username_ = std::string(username_attr->string_view()); // Figure out the lifetime and start the allocation timer. From 0a13f56063f96b02b585e93cc2595f3ec7066da8 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Thu, 15 May 2025 11:52:21 +0000 Subject: [PATCH 178/292] [cleanup] Replace uses of NULL with nullptr Generated using clang-tidy, tools/clang/scripts/build_clang_tools_extra.py \ --fetch out/Default clang-tidy clang-apply-replacements ninja -C out/Default gn gen out/Default --export-compile-commands cd out/Default tools/clang/third_party/llvm/clang-tools-extra/clang-tidy/tool/run-clang-tidy.py -p . \ -clang-tidy-binary out/Default/tools/clang/third_party/llvm/build/bin/clang-tidy \ -clang-apply-replacements-binary \ out/Default/tools/clang/third_party/llvm/build/bin/clang-apply-replacements \ -checks='-*,modernize-use-nullptr' \ -fix This CL was uploaded by git cl split. Bug: webrtc:42232595 Change-Id: I976f0f78bee83f934d4317f758d9d1f23fa5c096 No-Iwyu: LSC Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391821 Commit-Queue: Ilya Nikolaevskiy Reviewed-by: Ilya Nikolaevskiy Auto-Submit: Evan Shrubsole Cr-Commit-Position: refs/heads/main@{#44650} --- modules/video_capture/device_info_impl.cc | 2 +- modules/video_capture/linux/video_capture_v4l2.cc | 6 +++--- .../video_capture/test/video_capture_unittest.cc | 14 +++++++------- modules/video_capture/video_capture_impl.cc | 10 +++++----- 4 files changed, 16 insertions(+), 16 deletions(-) diff --git a/modules/video_capture/device_info_impl.cc b/modules/video_capture/device_info_impl.cc index dc49d8abb9..1d3e3ee89e 100644 --- a/modules/video_capture/device_info_impl.cc +++ b/modules/video_capture/device_info_impl.cc @@ -31,7 +31,7 @@ namespace webrtc { namespace videocapturemodule { DeviceInfoImpl::DeviceInfoImpl() - : _lastUsedDeviceName(NULL), _lastUsedDeviceNameLength(0) {} + : _lastUsedDeviceName(nullptr), _lastUsedDeviceNameLength(0) {} DeviceInfoImpl::~DeviceInfoImpl(void) { MutexLock lock(&_apiLock); diff --git a/modules/video_capture/linux/video_capture_v4l2.cc b/modules/video_capture/linux/video_capture_v4l2.cc index 33aa452a07..243656e782 100644 --- a/modules/video_capture/linux/video_capture_v4l2.cc +++ b/modules/video_capture/linux/video_capture_v4l2.cc @@ -55,7 +55,7 @@ VideoCaptureModuleV4L2::VideoCaptureModuleV4L2() _deviceFd(-1), _buffersAllocatedByDevice(-1), _captureStarted(false), - _pool(NULL) {} + _pool(nullptr) {} int32_t VideoCaptureModuleV4L2::Init(const char* deviceUniqueIdUTF8) { RTC_DCHECK_RUN_ON(&api_checker_); @@ -366,7 +366,7 @@ bool VideoCaptureModuleV4L2::AllocateVideoBuffers() { return false; } - _pool[i].start = mmap(NULL, buffer.length, PROT_READ | PROT_WRITE, + _pool[i].start = mmap(nullptr, buffer.length, PROT_READ | PROT_WRITE, MAP_SHARED, _deviceFd, buffer.m.offset); if (MAP_FAILED == _pool[i].start) { @@ -420,7 +420,7 @@ bool VideoCaptureModuleV4L2::CaptureProcess() { timeout.tv_usec = 0; // _deviceFd written only in StartCapture, when this thread isn't running. - retVal = select(_deviceFd + 1, &rSet, NULL, NULL, &timeout); + retVal = select(_deviceFd + 1, &rSet, nullptr, nullptr, &timeout); { MutexLock lock(&capture_lock_); diff --git a/modules/video_capture/test/video_capture_unittest.cc b/modules/video_capture/test/video_capture_unittest.cc index 2019f0bf4e..e8165d9060 100644 --- a/modules/video_capture/test/video_capture_unittest.cc +++ b/modules/video_capture/test/video_capture_unittest.cc @@ -156,7 +156,7 @@ class VideoCaptureTest : public ::testing::Test { webrtc::scoped_refptr module( VideoCaptureFactory::Create(unique_name)); - if (module.get() == NULL) + if (module.get() == nullptr) return nullptr; EXPECT_FALSE(module->CaptureStarted()); @@ -193,7 +193,7 @@ TEST_F(VideoCaptureTest, MAYBE_CreateDelete) { TestVideoCaptureCallback capture_observer; webrtc::scoped_refptr module( OpenVideoCaptureDevice(0, &capture_observer)); - ASSERT_TRUE(module.get() != NULL); + ASSERT_TRUE(module.get() != nullptr); VideoCaptureCapability capability; #ifndef WEBRTC_MAC @@ -237,7 +237,7 @@ TEST_F(VideoCaptureTest, MAYBE_Capabilities) { webrtc::scoped_refptr module( OpenVideoCaptureDevice(0, &capture_observer)); - ASSERT_TRUE(module.get() != NULL); + ASSERT_TRUE(module.get() != nullptr); int number_of_capabilities = device_info_->NumberOfCapabilities(module->CurrentDeviceName()); @@ -301,7 +301,7 @@ TEST_F(VideoCaptureTest, DISABLED_TestTwoCameras) { TestVideoCaptureCallback capture_observer1; webrtc::scoped_refptr module1( OpenVideoCaptureDevice(0, &capture_observer1)); - ASSERT_TRUE(module1.get() != NULL); + ASSERT_TRUE(module1.get() != nullptr); VideoCaptureCapability capability1; #ifndef WEBRTC_MAC device_info_->GetCapability(module1->CurrentDeviceName(), 0, capability1); @@ -316,7 +316,7 @@ TEST_F(VideoCaptureTest, DISABLED_TestTwoCameras) { TestVideoCaptureCallback capture_observer2; webrtc::scoped_refptr module2( OpenVideoCaptureDevice(1, &capture_observer2)); - ASSERT_TRUE(module1.get() != NULL); + ASSERT_TRUE(module1.get() != nullptr); VideoCaptureCapability capability2; #ifndef WEBRTC_MAC @@ -353,7 +353,7 @@ TEST_F(VideoCaptureTest, MAYBE_ConcurrentAccess) { TestVideoCaptureCallback capture_observer1; webrtc::scoped_refptr module1( OpenVideoCaptureDevice(0, &capture_observer1)); - ASSERT_TRUE(module1.get() != NULL); + ASSERT_TRUE(module1.get() != nullptr); VideoCaptureCapability capability; device_info_->GetCapability(module1->CurrentDeviceName(), 0, capability); capture_observer1.SetExpectedCapability(capability); @@ -361,7 +361,7 @@ TEST_F(VideoCaptureTest, MAYBE_ConcurrentAccess) { TestVideoCaptureCallback capture_observer2; webrtc::scoped_refptr module2( OpenVideoCaptureDevice(0, &capture_observer2)); - ASSERT_TRUE(module2.get() != NULL); + ASSERT_TRUE(module2.get() != nullptr); capture_observer2.SetExpectedCapability(capability); // Starting module1 should work. diff --git a/modules/video_capture/video_capture_impl.cc b/modules/video_capture/video_capture_impl.cc index eace423634..6e205a215f 100644 --- a/modules/video_capture/video_capture_impl.cc +++ b/modules/video_capture/video_capture_impl.cc @@ -84,12 +84,12 @@ int32_t VideoCaptureImpl::RotationInDegrees(VideoRotation rotation, } VideoCaptureImpl::VideoCaptureImpl() - : _deviceUniqueId(NULL), + : _deviceUniqueId(nullptr), _requestedCapability(), _lastProcessTimeNanos(TimeNanos()), _lastFrameRateCallbackTimeNanos(TimeNanos()), - _dataCallBack(NULL), - _rawDataCallBack(NULL), + _dataCallBack(nullptr), + _rawDataCallBack(nullptr), _lastProcessFrameTimeNanos(TimeNanos()), _rotateFrame(kVideoRotation_0), apply_rotation_(false) { @@ -123,8 +123,8 @@ void VideoCaptureImpl::RegisterCaptureDataCallback( void VideoCaptureImpl::DeRegisterCaptureDataCallback() { MutexLock lock(&api_lock_); - _dataCallBack = NULL; - _rawDataCallBack = NULL; + _dataCallBack = nullptr; + _rawDataCallBack = nullptr; } int32_t VideoCaptureImpl::DeliverCapturedFrame(VideoFrame& captureFrame) { RTC_CHECK_RUNS_SERIALIZED(&capture_checker_); From 691c150f9232901f75368bf741281f5a8f5ddefa Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Thu, 15 May 2025 11:03:00 -0700 Subject: [PATCH 179/292] Roll chromium_revision 66a9943b59..73401d12be (1460639:1460851) Change log: https://chromium.googlesource.com/chromium/src/+log/66a9943b59..73401d12be Full diff: https://chromium.googlesource.com/chromium/src/+/66a9943b59..73401d12be Changed dependencies * fuchsia_version: version:28.20250508.3.1..version:28.20250515.3.1 * src/base: https://chromium.googlesource.com/chromium/src/base/+log/a4805d687b..0db7dd0468 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/a4e74b4312..19cc7359e9 * src/buildtools/linux64: git_revision:487f8353f15456474437df32bb186187b0940b45..git_revision:0c25d1bbde6ef17e1e6d3938011f56a00c02e457 * src/buildtools/mac: git_revision:487f8353f15456474437df32bb186187b0940b45..git_revision:0c25d1bbde6ef17e1e6d3938011f56a00c02e457 * src/buildtools/win: git_revision:487f8353f15456474437df32bb186187b0940b45..git_revision:0c25d1bbde6ef17e1e6d3938011f56a00c02e457 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/e24185bdfb..48336cc678 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/11317b92ec..ff8a487cbe * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/faf790f4c1..eefd1d7c5e * src/third_party/freetype/src: https://chromium.googlesource.com/chromium/src/third_party/freetype2.git/+log/8a152c824a..7172bd11ba * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/828b2679ea..8aaac1117a * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/77388f6009..964a1a3958 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/e9e3e0175e..8ea6108a88 DEPS diff: https://chromium.googlesource.com/chromium/src/+/66a9943b59..73401d12be/DEPS No update to Clang. BUG=None Change-Id: I8ef70338d4071ea57e204ff592cac6e37e030a01 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391960 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44651} --- DEPS | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/DEPS b/DEPS index 6c6a7f0f3c..2851d5f6d3 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '66a9943b59a237f7184cce1a9b3e45218c19da49', + 'chromium_revision': '73401d12bed0a0ab3c34967bef1a4b1b7b18cc74', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -36,7 +36,7 @@ vars = { # By default, download the fuchsia sdk from the public sdk directory. 'fuchsia_sdk_cipd_prefix': 'fuchsia/sdk/core/', - 'fuchsia_version': 'version:28.20250508.3.1', + 'fuchsia_version': 'version:28.20250515.3.1', # By default, download the fuchsia images from the fuchsia GCS bucket. 'fuchsia_images_bucket': 'fuchsia', 'checkout_fuchsia': False, @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@a4805d687b1782f8ca734c851fd7b7c4f2f9822f', + 'https://chromium.googlesource.com/chromium/src/base@0db7dd046837490ee1d8c138ecf0c69ba4164407', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@a4e74b4312dafbbb000f24000ab3b57120a5e18d', + 'https://chromium.googlesource.com/chromium/src/build@19cc7359e9c645518c19c5c8bd0fb291cee0f156', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@e9b36e9ffb02b268d194c3f82f89952d09d0c714', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,19 +84,19 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@e24185bdfb0339d724f676aa7c8fe50efd49f2d4', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@48336cc678c8cee577f5961b85bb3c3da9b48d7f', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@11317b92ec4abfe891ff12e5583256aab2a68758', + 'https://chromium.googlesource.com/chromium/src/testing@ff8a487cbe6f55fd5cb5a792a33b21f02162a337', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@faf790f4c169c146a68006e1c2d69262faa2d52e', + 'https://chromium.googlesource.com/chromium/src/third_party@eefd1d7c5ee7d0a0f54b908a5aba609866dbfa4e', 'src/buildtools/linux64': { 'packages': [ { 'package': 'gn/gn/linux-${{arch}}', - 'version': 'git_revision:487f8353f15456474437df32bb186187b0940b45', + 'version': 'git_revision:0c25d1bbde6ef17e1e6d3938011f56a00c02e457', } ], 'dep_type': 'cipd', @@ -106,7 +106,7 @@ deps = { 'packages': [ { 'package': 'gn/gn/mac-${{arch}}', - 'version': 'git_revision:487f8353f15456474437df32bb186187b0940b45', + 'version': 'git_revision:0c25d1bbde6ef17e1e6d3938011f56a00c02e457', } ], 'dep_type': 'cipd', @@ -116,7 +116,7 @@ deps = { 'packages': [ { 'package': 'gn/gn/windows-amd64', - 'version': 'git_revision:487f8353f15456474437df32bb186187b0940b45', + 'version': 'git_revision:0c25d1bbde6ef17e1e6d3938011f56a00c02e457', } ], 'dep_type': 'cipd', @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@828b2679ea647472c0ddad9a11140f121120b73a', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@8aaac1117a87e52316daf1d9ea83ce4bb5c42976', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -511,7 +511,7 @@ deps = { 'condition': 'checkout_linux', }, 'src/third_party/freetype/src': - 'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@8a152c824ae08fc3459df5c87e10770fc47f80b1', + 'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@7172bd11badd468f6a86dba0b1769d624ead885c', 'src/third_party/harfbuzz-ng/src': 'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@9f83bbbe64654b45ba5bb06927ff36c2e7588495', 'src/third_party/google_benchmark/src': { @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '77388f6009c785fd1c2cfe8e3a3969e52967dcf3', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '964a1a39589933fd299bfde714b9cf5d0b5fb293', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@e9e3e0175e8962443592097100fd74f52e244c74', + 'https://chromium.googlesource.com/chromium/src/tools@8ea6108a8828992bf53a55a8e83ab41ce07bf209', 'src/third_party/espresso': { 'packages': [ From 09d832b1be7539e782d1205377126b502accf42b Mon Sep 17 00:00:00 2001 From: Fredrik Solenberg Date: Thu, 15 May 2025 21:32:31 +0200 Subject: [PATCH 180/292] Move RtpToNtpEstimator to rtc_base/ Bug: webrtc:42228262 Change-Id: I7814f13009f38b5546e68ec6753c5437baa770e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391580 Commit-Queue: Fredrik Solenberg Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44652} --- modules/rtp_rtcp/BUILD.gn | 1 + .../include/remote_ntp_time_estimator.h | 2 +- .../source/remote_ntp_time_estimator.cc | 2 +- rtc_base/BUILD.gn | 19 +++++++++++++++++++ .../rtp_to_ntp_estimator.cc | 11 +++++------ .../rtp_to_ntp_estimator.h | 10 ++++------ .../rtp_to_ntp_estimator_unittest.cc | 6 ++++-- system_wrappers/BUILD.gn | 3 --- video/BUILD.gn | 2 ++ video/rtp_streams_synchronizer2.cc | 2 +- video/stream_synchronization.h | 2 +- video/stream_synchronization_unittest.cc | 2 +- 12 files changed, 40 insertions(+), 22 deletions(-) rename {system_wrappers/source => rtc_base}/rtp_to_ntp_estimator.cc (96%) rename {system_wrappers/include => rtc_base}/rtp_to_ntp_estimator.h (90%) rename {system_wrappers/source => rtc_base}/rtp_to_ntp_estimator_unittest.cc (98%) diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn index 92c9954dcb..2a680f5567 100644 --- a/modules/rtp_rtcp/BUILD.gn +++ b/modules/rtp_rtcp/BUILD.gn @@ -339,6 +339,7 @@ rtc_library("rtp_rtcp") { "../../rtc_base:random", "../../rtc_base:rate_limiter", "../../rtc_base:rtc_numerics", + "../../rtc_base:rtp_to_ntp_estimator", "../../rtc_base:safe_conversions", "../../rtc_base:safe_minmax", "../../rtc_base:threading", diff --git a/modules/rtp_rtcp/include/remote_ntp_time_estimator.h b/modules/rtp_rtcp/include/remote_ntp_time_estimator.h index 9c8404b38e..5b08d799a1 100644 --- a/modules/rtp_rtcp/include/remote_ntp_time_estimator.h +++ b/modules/rtp_rtcp/include/remote_ntp_time_estimator.h @@ -18,8 +18,8 @@ #include "api/units/time_delta.h" #include "api/units/timestamp.h" #include "rtc_base/numerics/moving_percentile_filter.h" +#include "rtc_base/rtp_to_ntp_estimator.h" #include "system_wrappers/include/ntp_time.h" -#include "system_wrappers/include/rtp_to_ntp_estimator.h" namespace webrtc { diff --git a/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc b/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc index 3faca87ca8..af6e692581 100644 --- a/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc +++ b/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc @@ -17,9 +17,9 @@ #include "api/units/timestamp.h" #include "modules/rtp_rtcp/source/ntp_time_util.h" #include "rtc_base/logging.h" +#include "rtc_base/rtp_to_ntp_estimator.h" #include "system_wrappers/include/clock.h" #include "system_wrappers/include/ntp_time.h" -#include "system_wrappers/include/rtp_to_ntp_estimator.h" namespace webrtc { diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn index 58addef7aa..4af30800ed 100644 --- a/rtc_base/BUILD.gn +++ b/rtc_base/BUILD.gn @@ -1762,6 +1762,21 @@ rtc_library("rtc_base_tests_utils") { } } +rtc_library("rtp_to_ntp_estimator") { + visibility = [ "*" ] + sources = [ + "rtp_to_ntp_estimator.cc", + "rtp_to_ntp_estimator.h", + ] + deps = [ + ":checks", + ":logging", + ":rtc_numerics", + ":safe_conversions", + "../system_wrappers", + ] +} + rtc_library("task_queue_for_test") { testonly = true @@ -2190,6 +2205,7 @@ if (rtc_include_tests) { "rolling_accumulator_unittest.cc", "rtc_certificate_generator_unittest.cc", "rtc_certificate_unittest.cc", + "rtp_to_ntp_estimator_unittest.cc", "test_client_unittest.cc", "thread_unittest.cc", "unique_id_generator_unittest.cc", @@ -2216,11 +2232,13 @@ if (rtc_include_tests) { ":network_constants", ":network_route", ":null_socket_server", + ":random", ":refcount", ":rolling_accumulator", ":rtc_base_tests_utils", ":rtc_certificate_generator", ":rtc_event", + ":rtp_to_ntp_estimator", ":safe_conversions", ":socket", ":socket_adapters", @@ -2249,6 +2267,7 @@ if (rtc_include_tests) { "../api/task_queue:pending_task_safety_flag", "../api/task_queue:task_queue_test", "../api/units:time_delta", + "../system_wrappers", "../test:field_trial", "../test:fileutils", "../test:rtc_expect_death", diff --git a/system_wrappers/source/rtp_to_ntp_estimator.cc b/rtc_base/rtp_to_ntp_estimator.cc similarity index 96% rename from system_wrappers/source/rtp_to_ntp_estimator.cc rename to rtc_base/rtp_to_ntp_estimator.cc index 549967c0da..d5f41f2f83 100644 --- a/system_wrappers/source/rtp_to_ntp_estimator.cc +++ b/rtc_base/rtp_to_ntp_estimator.cc @@ -8,17 +8,16 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "system_wrappers/include/rtp_to_ntp_estimator.h" - -#include +#include "rtc_base/rtp_to_ntp_estimator.h" #include -#include +#include +#include +#include -#include "api/array_view.h" -#include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/numerics/safe_conversions.h" +#include "system_wrappers/include/ntp_time.h" namespace webrtc { namespace { diff --git a/system_wrappers/include/rtp_to_ntp_estimator.h b/rtc_base/rtp_to_ntp_estimator.h similarity index 90% rename from system_wrappers/include/rtp_to_ntp_estimator.h rename to rtc_base/rtp_to_ntp_estimator.h index 2f3698176f..a279def51f 100644 --- a/system_wrappers/include/rtp_to_ntp_estimator.h +++ b/rtc_base/rtp_to_ntp_estimator.h @@ -8,15 +8,13 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef SYSTEM_WRAPPERS_INCLUDE_RTP_TO_NTP_ESTIMATOR_H_ -#define SYSTEM_WRAPPERS_INCLUDE_RTP_TO_NTP_ESTIMATOR_H_ - -#include +#ifndef RTC_BASE_RTP_TO_NTP_ESTIMATOR_H_ +#define RTC_BASE_RTP_TO_NTP_ESTIMATOR_H_ +#include #include #include -#include "rtc_base/checks.h" #include "rtc_base/numerics/sequence_number_unwrapper.h" #include "system_wrappers/include/ntp_time.h" @@ -69,4 +67,4 @@ class RtpToNtpEstimator { }; } // namespace webrtc -#endif // SYSTEM_WRAPPERS_INCLUDE_RTP_TO_NTP_ESTIMATOR_H_ +#endif // RTC_BASE_RTP_TO_NTP_ESTIMATOR_H_ diff --git a/system_wrappers/source/rtp_to_ntp_estimator_unittest.cc b/rtc_base/rtp_to_ntp_estimator_unittest.cc similarity index 98% rename from system_wrappers/source/rtp_to_ntp_estimator_unittest.cc rename to rtc_base/rtp_to_ntp_estimator_unittest.cc index 81effd9452..e74624dc8c 100644 --- a/system_wrappers/source/rtp_to_ntp_estimator_unittest.cc +++ b/rtc_base/rtp_to_ntp_estimator_unittest.cc @@ -8,11 +8,13 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "system_wrappers/include/rtp_to_ntp_estimator.h" +#include "rtc_base/rtp_to_ntp_estimator.h" -#include +#include +#include #include "rtc_base/random.h" +#include "system_wrappers/include/ntp_time.h" #include "test/gtest.h" namespace webrtc { diff --git a/system_wrappers/BUILD.gn b/system_wrappers/BUILD.gn index c708f81a73..3b23716960 100644 --- a/system_wrappers/BUILD.gn +++ b/system_wrappers/BUILD.gn @@ -19,12 +19,10 @@ rtc_library("system_wrappers") { "include/cpu_features_wrapper.h", "include/cpu_info.h", "include/ntp_time.h", - "include/rtp_to_ntp_estimator.h", "include/sleep.h", "source/clock.cc", "source/cpu_features.cc", "source/cpu_info.cc", - "source/rtp_to_ntp_estimator.cc", "source/sleep.cc", ] @@ -136,7 +134,6 @@ if (rtc_include_tests && !build_with_chromium) { "source/metrics_default_unittest.cc", "source/metrics_unittest.cc", "source/ntp_time_unittest.cc", - "source/rtp_to_ntp_estimator_unittest.cc", ] deps = [ diff --git a/video/BUILD.gn b/video/BUILD.gn index dec17dde3d..bf77b139ca 100644 --- a/video/BUILD.gn +++ b/video/BUILD.gn @@ -166,6 +166,7 @@ rtc_library("video") { "../rtc_base:rate_tracker", "../rtc_base:rtc_event", "../rtc_base:rtc_numerics", + "../rtc_base:rtp_to_ntp_estimator", "../rtc_base:safe_conversions", "../rtc_base:sample_counter", "../rtc_base:stringutils", @@ -1026,6 +1027,7 @@ if (rtc_include_tests) { "../rtc_base:rtc_base_tests_utils", "../rtc_base:rtc_event", "../rtc_base:rtc_numerics", + "../rtc_base:rtp_to_ntp_estimator", "../rtc_base:safe_conversions", "../rtc_base:stringutils", "../rtc_base:task_queue_for_test", diff --git a/video/rtp_streams_synchronizer2.cc b/video/rtp_streams_synchronizer2.cc index cbc1a59124..f3c3875dc9 100644 --- a/video/rtp_streams_synchronizer2.cc +++ b/video/rtp_streams_synchronizer2.cc @@ -19,11 +19,11 @@ #include "call/syncable.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" +#include "rtc_base/rtp_to_ntp_estimator.h" #include "rtc_base/task_utils/repeating_task.h" #include "rtc_base/time_utils.h" #include "rtc_base/trace_event.h" #include "system_wrappers/include/ntp_time.h" -#include "system_wrappers/include/rtp_to_ntp_estimator.h" #include "video/stream_synchronization.h" namespace webrtc { diff --git a/video/stream_synchronization.h b/video/stream_synchronization.h index 61073cb4b2..e34036f0b3 100644 --- a/video/stream_synchronization.h +++ b/video/stream_synchronization.h @@ -13,7 +13,7 @@ #include -#include "system_wrappers/include/rtp_to_ntp_estimator.h" +#include "rtc_base/rtp_to_ntp_estimator.h" namespace webrtc { diff --git a/video/stream_synchronization_unittest.cc b/video/stream_synchronization_unittest.cc index 0d1d379e4f..e99e7f99e1 100644 --- a/video/stream_synchronization_unittest.cc +++ b/video/stream_synchronization_unittest.cc @@ -13,9 +13,9 @@ #include #include +#include "rtc_base/rtp_to_ntp_estimator.h" #include "system_wrappers/include/clock.h" #include "system_wrappers/include/ntp_time.h" -#include "system_wrappers/include/rtp_to_ntp_estimator.h" #include "test/gtest.h" namespace webrtc { From e30e216756cb9bdee35074b8020a78e26c981d94 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Thu, 15 May 2025 17:03:46 -0700 Subject: [PATCH 181/292] Roll chromium_revision 73401d12be..a208c76a1b (1460851:1461082) Change log: https://chromium.googlesource.com/chromium/src/+log/73401d12be..a208c76a1b Full diff: https://chromium.googlesource.com/chromium/src/+/73401d12be..a208c76a1b Changed dependencies * reclient_version: re_client_version:0.177.1.e58c0145-gomaip..re_client_version:0.178.0.5ee9d3e8-gomaip * src/base: https://chromium.googlesource.com/chromium/src/base/+log/0db7dd0468..75762d452e * src/build: https://chromium.googlesource.com/chromium/src/build/+log/19cc7359e9..524367bd1e * src/buildtools/reclient: re_client_version:0.177.1.e58c0145-gomaip..re_client_version:0.178.0.5ee9d3e8-gomaip * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/48336cc678..a71b3cb45c * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/ff8a487cbe..19995fa38d * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/eefd1d7c5e..5c9c7bff04 * src/third_party/android_deps/autorolled/cipd: TZHdcfZjA4uMdOqeaQ-7MG_Pi3P1vgW-fN5TrDzqfVsC..PY5tSZDTL7W8p6Yi4BuTUezpSGjWKlBrmJKhcuwdcx0C * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/02bc0949e5..99eba5614c * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/2fa93b2332..ffd1c50cbd * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/8aaac1117a..23167060ae * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/964a1a3958..109a2ebe2b * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/8ea6108a88..ecedd4647b DEPS diff: https://chromium.googlesource.com/chromium/src/+/73401d12be..a208c76a1b/DEPS No update to Clang. BUG=None Change-Id: I58e568900ae6c1fe6e9ce1d05043c72a1751a014 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392041 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44653} --- DEPS | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) diff --git a/DEPS b/DEPS index 2851d5f6d3..89084e88bf 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '73401d12bed0a0ab3c34967bef1a4b1b7b18cc74', + 'chromium_revision': 'a208c76a1b6362ac1b194c8253ee8b2b845cc6b1', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -51,7 +51,7 @@ vars = { # RBE instance to use for running remote builds 'rbe_instance': 'projects/rbe-webrtc-developer/instances/default_instance', # reclient CIPD package version - 'reclient_version': 're_client_version:0.177.1.e58c0145-gomaip', + 'reclient_version': 're_client_version:0.178.0.5ee9d3e8-gomaip', # siso CIPD package version. 'siso_version': 'git_revision:30ea210712239b874febcadf661f118fc2c0f6d7', @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@0db7dd046837490ee1d8c138ecf0c69ba4164407', + 'https://chromium.googlesource.com/chromium/src/base@75762d452eaa1d43c34921f88dc6698ec08c2f32', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@19cc7359e9c645518c19c5c8bd0fb291cee0f156', + 'https://chromium.googlesource.com/chromium/src/build@524367bd1ea6a622c7d304907af0c07476f7774f', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@e9b36e9ffb02b268d194c3f82f89952d09d0c714', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@48336cc678c8cee577f5961b85bb3c3da9b48d7f', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@a71b3cb45c04b1c1baa034882d0157f52425ae3e', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@ff8a487cbe6f55fd5cb5a792a33b21f02162a337', + 'https://chromium.googlesource.com/chromium/src/testing@19995fa38d7748c46ccc3f90b793bd0926a2089b', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@eefd1d7c5ee7d0a0f54b908a5aba609866dbfa4e', + 'https://chromium.googlesource.com/chromium/src/third_party@5c9c7bff045e1f0bfb26d5471406fc4875584a56', 'src/buildtools/linux64': { 'packages': [ @@ -480,11 +480,11 @@ deps = { }, 'src/third_party/boringssl/src': - 'https://boringssl.googlesource.com/boringssl.git@02bc0949e5cac0e1ee82c6f365f5a6c3cfd0cfa9', + 'https://boringssl.googlesource.com/boringssl.git@99eba5614c519092e3d797a95dade2ac2660d5d9', 'src/third_party/breakpad/breakpad': 'https://chromium.googlesource.com/breakpad/breakpad.git@fad82477312f786994151f01c352813fabf924c0', 'src/third_party/catapult': - 'https://chromium.googlesource.com/catapult.git@2fa93b2332b7a27c3066bef8d7ba9de2a5e1ab2e', + 'https://chromium.googlesource.com/catapult.git@ffd1c50cbd19e3f0c5de2fd1d20b43adc04b6e85', 'src/third_party/ced/src': { 'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5', }, @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@8aaac1117a87e52316daf1d9ea83ce4bb5c42976', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@23167060ae192ce2eef0e581018b37d34c2baf5a', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '964a1a39589933fd299bfde714b9cf5d0b5fb293', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '109a2ebe2b65215b8ba3ff0d6a3a8361220a0992', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@8ea6108a8828992bf53a55a8e83ab41ce07bf209', + 'https://chromium.googlesource.com/chromium/src/tools@ecedd4647b4622ea134d8d3fd08f19f9710433d9', 'src/third_party/espresso': { 'packages': [ @@ -827,7 +827,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_deps/autorolled', - 'version': 'TZHdcfZjA4uMdOqeaQ-7MG_Pi3P1vgW-fN5TrDzqfVsC', + 'version': 'PY5tSZDTL7W8p6Yi4BuTUezpSGjWKlBrmJKhcuwdcx0C', }, ], 'condition': 'checkout_android and non_git_source', From 795fbba5767e5bcbd949639a0f8117278474520f Mon Sep 17 00:00:00 2001 From: webrtc-version-updater Date: Thu, 15 May 2025 21:11:18 -0700 Subject: [PATCH 182/292] Update WebRTC code version (2025-05-16T04:10:38). Bug: None Change-Id: Ib4ce0fa1f16aa3894d7246f3fbc6e3d32ea9c4e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392081 Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Cr-Commit-Position: refs/heads/main@{#44654} --- call/version.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/call/version.cc b/call/version.cc index 3473abdf62..3758d817ce 100644 --- a/call/version.cc +++ b/call/version.cc @@ -13,7 +13,7 @@ namespace webrtc { // The timestamp is always in UTC. -const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-15T04:06:05"; +const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-16T04:10:38"; void LoadWebRTCVersionInRegister() { // Using volatile to instruct the compiler to not optimize `p` away even From e76eb76482d3759bdaf2a938f45e02266c6142c4 Mon Sep 17 00:00:00 2001 From: Yury Yarashevich Date: Thu, 15 May 2025 15:15:52 +0200 Subject: [PATCH 183/292] Log libraries json to facilitate debugging of occasional CI failures. When gn prints warnings in output there is obscure json parse exception. Bug: None Change-Id: Icfb434fdfbfdd9c6a41a23f75f6d63deaaa9114d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391900 Reviewed-by: Mirko Bonadei Commit-Queue: Mirko Bonadei Cr-Commit-Position: refs/heads/main@{#44655} --- tools_webrtc/libs/generate_licenses.py | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/tools_webrtc/libs/generate_licenses.py b/tools_webrtc/libs/generate_licenses.py index 73fd19ae86..0faddeabd4 100755 --- a/tools_webrtc/libs/generate_licenses.py +++ b/tools_webrtc/libs/generate_licenses.py @@ -196,7 +196,12 @@ def _run_gn(buildfile_dir, target): return output_json def _get_third_party_libraries(self, buildfile_dir, target): - output = json.loads(LicenseBuilder._run_gn(buildfile_dir, target)) + license_json = LicenseBuilder._run_gn(buildfile_dir, target) + try: + output = json.loads(license_json) + except: + logging.error("unable to parse license_json = '%s'", license_json) + raise libraries = set() for described_target in list(output.values()): third_party_libs = (self._parse_library(dep) From 4e2cc762ec4e854755642f2d9a63a722ed67720e Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Thu, 15 May 2025 11:51:54 +0000 Subject: [PATCH 184/292] [cleanup] Replace uses of NULL with nullptr Generated using clang-tidy, tools/clang/scripts/build_clang_tools_extra.py \ --fetch out/Default clang-tidy clang-apply-replacements ninja -C out/Default gn gen out/Default --export-compile-commands cd out/Default tools/clang/third_party/llvm/clang-tools-extra/clang-tidy/tool/run-clang-tidy.py -p . \ -clang-tidy-binary out/Default/tools/clang/third_party/llvm/build/bin/clang-tidy \ -clang-apply-replacements-binary \ out/Default/tools/clang/third_party/llvm/build/bin/clang-apply-replacements \ -checks='-*,modernize-use-nullptr' \ -fix This CL was uploaded by git cl split. Bug: webrtc:42232595 Change-Id: I26d5e2e4711c17eec406011df7349c563bcaafec No-Iwyu: LSC Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391804 Commit-Queue: Evan Shrubsole Reviewed-by: Harald Alvestrand Auto-Submit: Evan Shrubsole Cr-Commit-Position: refs/heads/main@{#44656} --- modules/desktop_capture/cropping_window_capturer.cc | 2 +- modules/desktop_capture/desktop_region_unittest.cc | 4 ++-- .../linux/x11/mouse_cursor_monitor_x11.cc | 8 ++++---- .../desktop_capture/linux/x11/screen_capturer_x11.cc | 6 +++--- modules/desktop_capture/linux/x11/shared_x_display.cc | 2 +- .../desktop_capture/linux/x11/x_server_pixel_buffer.cc | 10 +++++----- modules/desktop_capture/screen_capturer_unittest.cc | 2 +- 7 files changed, 17 insertions(+), 17 deletions(-) diff --git a/modules/desktop_capture/cropping_window_capturer.cc b/modules/desktop_capture/cropping_window_capturer.cc index 5e0faaade9..d1cc1c8313 100644 --- a/modules/desktop_capture/cropping_window_capturer.cc +++ b/modules/desktop_capture/cropping_window_capturer.cc @@ -22,7 +22,7 @@ namespace webrtc { CroppingWindowCapturer::CroppingWindowCapturer( const DesktopCaptureOptions& options) : options_(options), - callback_(NULL), + callback_(nullptr), window_capturer_(DesktopCapturer::CreateRawWindowCapturer(options)), selected_window_(kNullWindowId), excluded_window_(kNullWindowId) {} diff --git a/modules/desktop_capture/desktop_region_unittest.cc b/modules/desktop_capture/desktop_region_unittest.cc index b8bd78e990..0cd45e9c6b 100644 --- a/modules/desktop_capture/desktop_region_unittest.cc +++ b/modules/desktop_capture/desktop_region_unittest.cc @@ -47,7 +47,7 @@ void CompareRegion(const DesktopRegion& region, // Verify that regions are empty when created. TEST(DesktopRegionTest, Empty) { DesktopRegion r; - CompareRegion(r, NULL, 0); + CompareRegion(r, nullptr, 0); } // Verify that empty rectangles are ignored. @@ -55,7 +55,7 @@ TEST(DesktopRegionTest, AddEmpty) { DesktopRegion r; DesktopRect rect = DesktopRect::MakeXYWH(1, 2, 0, 0); r.AddRect(rect); - CompareRegion(r, NULL, 0); + CompareRegion(r, nullptr, 0); } // Verify that regions with a single rectangles are handled properly. diff --git a/modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.cc b/modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.cc index d9c7635c1d..064b98fa15 100644 --- a/modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.cc +++ b/modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.cc @@ -69,7 +69,7 @@ MouseCursorMonitorX11::MouseCursorMonitorX11( const DesktopCaptureOptions& options, Window window) : x_display_(options.x_display()), - callback_(NULL), + callback_(nullptr), mode_(SHAPE_AND_POSITION), window_(window), have_xfixes_(false), @@ -231,10 +231,10 @@ MouseCursorMonitor* MouseCursorMonitorX11::CreateForWindow( const DesktopCaptureOptions& options, WindowId window) { if (!options.x_display()) - return NULL; + return nullptr; window = GetTopLevelWindow(options.x_display()->display(), window); if (window == None) - return NULL; + return nullptr; return new MouseCursorMonitorX11(options, window); } @@ -242,7 +242,7 @@ MouseCursorMonitor* MouseCursorMonitorX11::CreateForScreen( const DesktopCaptureOptions& options, ScreenId screen) { if (!options.x_display()) - return NULL; + return nullptr; return new MouseCursorMonitorX11( options, DefaultRootWindow(options.x_display()->display())); } diff --git a/modules/desktop_capture/linux/x11/screen_capturer_x11.cc b/modules/desktop_capture/linux/x11/screen_capturer_x11.cc index 0702c755a1..0384a0373b 100644 --- a/modules/desktop_capture/linux/x11/screen_capturer_x11.cc +++ b/modules/desktop_capture/linux/x11/screen_capturer_x11.cc @@ -67,8 +67,8 @@ bool ScreenCapturerX11::Init(const DesktopCaptureOptions& options) { return false; } - gc_ = XCreateGC(display(), root_window_, 0, NULL); - if (gc_ == NULL) { + gc_ = XCreateGC(display(), root_window_, 0, nullptr); + if (gc_ == nullptr) { RTC_LOG(LS_ERROR) << "Unable to get graphics context"; DeinitXlib(); return false; @@ -133,7 +133,7 @@ void ScreenCapturerX11::InitXDamage() { } // Create an XFixes server-side region to collate damage into. - damage_region_ = XFixesCreateRegion(display(), 0, 0); + damage_region_ = XFixesCreateRegion(display(), nullptr, 0); if (!damage_region_) { XDamageDestroy(display(), damage_handle_); RTC_LOG(LS_ERROR) << "Unable to create XFixes region."; diff --git a/modules/desktop_capture/linux/x11/shared_x_display.cc b/modules/desktop_capture/linux/x11/shared_x_display.cc index aee9183c53..60101b9aa4 100644 --- a/modules/desktop_capture/linux/x11/shared_x_display.cc +++ b/modules/desktop_capture/linux/x11/shared_x_display.cc @@ -34,7 +34,7 @@ SharedXDisplay::~SharedXDisplay() { scoped_refptr SharedXDisplay::Create( absl::string_view display_name) { Display* display = XOpenDisplay( - display_name.empty() ? NULL : std::string(display_name).c_str()); + display_name.empty() ? nullptr : std::string(display_name).c_str()); if (!display) { RTC_LOG(LS_ERROR) << "Unable to open display"; return nullptr; diff --git a/modules/desktop_capture/linux/x11/x_server_pixel_buffer.cc b/modules/desktop_capture/linux/x11/x_server_pixel_buffer.cc index 7ab01cae04..d62a014ae8 100644 --- a/modules/desktop_capture/linux/x11/x_server_pixel_buffer.cc +++ b/modules/desktop_capture/linux/x11/x_server_pixel_buffer.cc @@ -171,7 +171,7 @@ void XServerPixelBuffer::ReleaseSharedMemorySegment() { if (shm_segment_info_->shmaddr != nullptr) shmdt(shm_segment_info_->shmaddr); if (shm_segment_info_->shmid != -1) - shmctl(shm_segment_info_->shmid, IPC_RMID, 0); + shmctl(shm_segment_info_->shmid, IPC_RMID, nullptr); delete shm_segment_info_; shm_segment_info_ = nullptr; } @@ -220,14 +220,14 @@ void XServerPixelBuffer::InitShm(const XWindowAttributes& attributes) { shm_segment_info_->shmaddr = nullptr; shm_segment_info_->readOnly = False; x_shm_image_ = XShmCreateImage(display_, default_visual, default_depth, - ZPixmap, 0, shm_segment_info_, + ZPixmap, nullptr, shm_segment_info_, window_rect_.width(), window_rect_.height()); if (x_shm_image_) { shm_segment_info_->shmid = shmget(IPC_PRIVATE, x_shm_image_->bytes_per_line * x_shm_image_->height, IPC_CREAT | 0600); if (shm_segment_info_->shmid != -1) { - void* shmat_result = shmat(shm_segment_info_->shmid, 0, 0); + void* shmat_result = shmat(shm_segment_info_->shmid, nullptr, 0); if (shmat_result != reinterpret_cast(-1)) { shm_segment_info_->shmaddr = reinterpret_cast(shmat_result); x_shm_image_->data = shm_segment_info_->shmaddr; @@ -258,7 +258,7 @@ void XServerPixelBuffer::InitShm(const XWindowAttributes& attributes) { if (have_pixmaps) have_pixmaps = InitPixmaps(default_depth); - shmctl(shm_segment_info_->shmid, IPC_RMID, 0); + shmctl(shm_segment_info_->shmid, IPC_RMID, nullptr); shm_segment_info_->shmid = -1; RTC_LOG(LS_VERBOSE) << "Using X shared memory extension v" << major << "." @@ -295,7 +295,7 @@ bool XServerPixelBuffer::InitPixmaps(int depth) { if (error_trap.GetLastErrorAndDisable() != 0) { XFreePixmap(display_, shm_pixmap_); shm_pixmap_ = 0; - shm_gc_ = 0; // See shm_pixmap_ comment above. + shm_gc_ = nullptr; // See shm_pixmap_ comment above. return false; } } diff --git a/modules/desktop_capture/screen_capturer_unittest.cc b/modules/desktop_capture/screen_capturer_unittest.cc index 2cd8184e6b..0176314551 100644 --- a/modules/desktop_capture/screen_capturer_unittest.cc +++ b/modules/desktop_capture/screen_capturer_unittest.cc @@ -138,7 +138,7 @@ TEST_F(ScreenCapturerTest, MAYBE_Capture) { EXPECT_GT(frame->size().height(), 0); EXPECT_GE(frame->stride(), frame->size().width() * DesktopFrame::kBytesPerPixel); - EXPECT_TRUE(frame->shared_memory() == NULL); + EXPECT_TRUE(frame->shared_memory() == nullptr); // Verify that the region contains whole screen. EXPECT_FALSE(frame->updated_region().is_empty()); From 99e953c17732682a93710f6c851a4a296dcb10c4 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Thu, 15 May 2025 11:52:06 +0000 Subject: [PATCH 185/292] [cleanup] Replace uses of NULL with nullptr MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Generated using clang-tidy, tools/clang/scripts/build_clang_tools_extra.py \ --fetch out/Default clang-tidy clang-apply-replacements ninja -C out/Default gn gen out/Default --export-compile-commands cd out/Default tools/clang/third_party/llvm/clang-tools-extra/clang-tidy/tool/run-clang-tidy.py -p . \ -clang-tidy-binary out/Default/tools/clang/third_party/llvm/build/bin/clang-tidy \ -clang-apply-replacements-binary \ out/Default/tools/clang/third_party/llvm/build/bin/clang-apply-replacements \ -checks='-*,modernize-use-nullptr' \ -fix This CL was uploaded by git cl split. Bug: webrtc:42232595 Change-Id: Iff256a5f5d5e5f294441e6295a1d4199a2fd1862 No-Iwyu: LSC Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391844 Auto-Submit: Evan Shrubsole Reviewed-by: Björn Terelius Commit-Queue: Björn Terelius Cr-Commit-Position: refs/heads/main@{#44657} --- .../remote_bitrate_estimator_unittest_helper.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc b/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc index 5adbadd296..2d34ed67ef 100644 --- a/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc +++ b/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc @@ -97,7 +97,7 @@ int64_t RtpStream::next_rtp_time() const { // Generates an RTCP packet. RtpStream::RtcpPacket* RtpStream::Rtcp(int64_t time_now_us) { if (time_now_us < next_rtcp_time_) { - return NULL; + return nullptr; } RtcpPacket* rtcp = new RtcpPacket; int64_t send_time_us = time_now_us + kSendSideOffsetUs; From 94667052f394b431aa743253e9cd18b2b598152c Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Fri, 16 May 2025 01:01:59 -0700 Subject: [PATCH 186/292] Roll chromium_revision a208c76a1b..083b3eb60c (1461082:1461214) Change log: https://chromium.googlesource.com/chromium/src/+log/a208c76a1b..083b3eb60c Full diff: https://chromium.googlesource.com/chromium/src/+/a208c76a1b..083b3eb60c Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/75762d452e..1f29789660 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/524367bd1e..db8ce42b66 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/19995fa38d..7ba47582ec * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/5c9c7bff04..ec61df4b22 * src/third_party/freetype/src: https://chromium.googlesource.com/chromium/src/third_party/freetype2.git/+log/7172bd11ba..8082aba5f2 * src/third_party/googletest/src: https://chromium.googlesource.com/external/github.com/google/googletest.git/+log/9f79a9597a..571930618f * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/23167060ae..fbce37dd3e * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/ecedd4647b..17fd1f3675 DEPS diff: https://chromium.googlesource.com/chromium/src/+/a208c76a1b..083b3eb60c/DEPS No update to Clang. BUG=None Change-Id: Id52d9c193a9595fa76f0d74ba240f69ad6c55f31 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392120 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44658} --- DEPS | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/DEPS b/DEPS index 89084e88bf..54147a936c 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': 'a208c76a1b6362ac1b194c8253ee8b2b845cc6b1', + 'chromium_revision': '083b3eb60c3bf65c15ef2e737a78bb18c7709374', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@75762d452eaa1d43c34921f88dc6698ec08c2f32', + 'https://chromium.googlesource.com/chromium/src/base@1f29789660c727524886a1a662b5e58058fa42f7', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@524367bd1ea6a622c7d304907af0c07476f7774f', + 'https://chromium.googlesource.com/chromium/src/build@db8ce42b66262db8f53a3dc40c6241c23ed97243', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@e9b36e9ffb02b268d194c3f82f89952d09d0c714', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -88,9 +88,9 @@ deps = { 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@19995fa38d7748c46ccc3f90b793bd0926a2089b', + 'https://chromium.googlesource.com/chromium/src/testing@7ba47582ec28f9e542ac3e6186a2d4c5d8775da3', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@5c9c7bff045e1f0bfb26d5471406fc4875584a56', + 'https://chromium.googlesource.com/chromium/src/third_party@ec61df4b2281ada165462e76d55a972be0d58e96', 'src/buildtools/linux64': { 'packages': [ @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@23167060ae192ce2eef0e581018b37d34c2baf5a', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@fbce37dd3e925eef938e9d9ee24345933a07541f', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -511,7 +511,7 @@ deps = { 'condition': 'checkout_linux', }, 'src/third_party/freetype/src': - 'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@7172bd11badd468f6a86dba0b1769d624ead885c', + 'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@8082aba5f2ddc334b6dfe89a10db13a1b120f76e', 'src/third_party/harfbuzz-ng/src': 'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@9f83bbbe64654b45ba5bb06927ff36c2e7588495', 'src/third_party/google_benchmark/src': { @@ -525,7 +525,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/googletest/src': - 'https://chromium.googlesource.com/external/github.com/google/googletest.git@9f79a9597ad9b5394e5c620ebf76824f77ffbde4', + 'https://chromium.googlesource.com/external/github.com/google/googletest.git@571930618fa96eabcd05b573285edbee9fc13bae', 'src/third_party/icu': { 'url': 'https://chromium.googlesource.com/chromium/deps/icu.git@b929596baebf0ab4ac7ec07f38365db4c50a559d', }, @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@ecedd4647b4622ea134d8d3fd08f19f9710433d9', + 'https://chromium.googlesource.com/chromium/src/tools@17fd1f36756107cbdc25ca57d87a56ecde7a1824', 'src/third_party/espresso': { 'packages': [ From f10fb5878ecedbefc36865713ff514f8c537b3cc Mon Sep 17 00:00:00 2001 From: Jeremy Leconte Date: Fri, 16 May 2025 10:15:05 +0200 Subject: [PATCH 187/292] Allow sample rate of 24kHz in the test audio device. Change-Id: Iedd35985290c1c97145f4e6a777d07ca82cba8ee Bug: None No-Iwyu: Avoid unrelated changes Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392101 Commit-Queue: Jeremy Leconte Reviewed-by: Henrik Andreassson Cr-Commit-Position: refs/heads/main@{#44659} --- modules/audio_device/test_audio_device_impl.cc | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/modules/audio_device/test_audio_device_impl.cc b/modules/audio_device/test_audio_device_impl.cc index 2e12be175b..32944d0539 100644 --- a/modules/audio_device/test_audio_device_impl.cc +++ b/modules/audio_device/test_audio_device_impl.cc @@ -41,8 +41,8 @@ TestAudioDevice::TestAudioDevice( rendering_(false), capturing_(false) { auto good_sample_rate = [](int sr) { - return sr == 8000 || sr == 16000 || sr == 32000 || sr == 44100 || - sr == 48000; + return sr == 8000 || sr == 16000 || sr == 24000 || sr == 32000 || + sr == 44100 || sr == 48000; }; if (renderer_) { From 32f71e55488212e731c94f6c12eb09504720e21e Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Thu, 15 May 2025 11:50:59 +0000 Subject: [PATCH 188/292] [cleanup] Replace uses of NULL with nullptr Generated using clang-tidy, tools/clang/scripts/build_clang_tools_extra.py \ --fetch out/Default clang-tidy clang-apply-replacements ninja -C out/Default gn gen out/Default --export-compile-commands cd out/Default tools/clang/third_party/llvm/clang-tools-extra/clang-tidy/tool/run-clang-tidy.py -p . \ -clang-tidy-binary out/Default/tools/clang/third_party/llvm/build/bin/clang-tidy \ -clang-apply-replacements-binary \ out/Default/tools/clang/third_party/llvm/build/bin/clang-apply-replacements \ -checks='-*,modernize-use-nullptr' \ -fix This CL was uploaded by git cl split. Bug: webrtc:42232237 Change-Id: I6cdd8dac9db814286a21a165d2b03dd3dcda46ac No-Iwyu: LSC Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391641 Auto-Submit: Evan Shrubsole Reviewed-by: Ilya Nikolaevskiy Commit-Queue: Evan Shrubsole Cr-Commit-Position: refs/heads/main@{#44660} --- media/base/fake_media_engine.cc | 2 +- media/base/stream_params.cc | 2 +- media/base/stream_params_unittest.cc | 2 +- media/engine/webrtc_video_engine.cc | 16 ++++++++-------- media/engine/webrtc_video_engine_unittest.cc | 2 +- 5 files changed, 12 insertions(+), 12 deletions(-) diff --git a/media/base/fake_media_engine.cc b/media/base/fake_media_engine.cc index 35e8f137d9..9a9b0c1611 100644 --- a/media/base/fake_media_engine.cc +++ b/media/base/fake_media_engine.cc @@ -494,7 +494,7 @@ bool FakeVideoMediaReceiveChannel::AddRecvStream(const StreamParams& sp) { if (!RtpReceiveChannelHelper< VideoMediaReceiveChannelInterface>::AddRecvStream(sp)) return false; - sinks_[sp.first_ssrc()] = NULL; + sinks_[sp.first_ssrc()] = nullptr; output_delays_[sp.first_ssrc()] = 0; return true; } diff --git a/media/base/stream_params.cc b/media/base/stream_params.cc index 8346e4cd47..94e2dea50f 100644 --- a/media/base/stream_params.cc +++ b/media/base/stream_params.cc @@ -178,7 +178,7 @@ void StreamParams::GenerateSsrcs(int num_layers, void StreamParams::GetPrimarySsrcs(std::vector* primary_ssrcs) const { const SsrcGroup* sim_group = get_ssrc_group(kSimSsrcGroupSemantics); - if (sim_group == NULL) { + if (sim_group == nullptr) { primary_ssrcs->push_back(first_ssrc()); } else { primary_ssrcs->insert(primary_ssrcs->end(), sim_group->ssrcs.begin(), diff --git a/media/base/stream_params_unittest.cc b/media/base/stream_params_unittest.cc index efb113a68b..b89c5b9133 100644 --- a/media/base/stream_params_unittest.cc +++ b/media/base/stream_params_unittest.cc @@ -102,7 +102,7 @@ TEST(StreamParams, HasSsrcGroup) { TEST(StreamParams, GetSsrcGroup) { webrtc::StreamParams sp = CreateStreamParamsWithSsrcGroup("XYZ", kSsrcs2, arraysize(kSsrcs2)); - EXPECT_EQ(NULL, sp.get_ssrc_group("xyz")); + EXPECT_EQ(nullptr, sp.get_ssrc_group("xyz")); EXPECT_EQ(&sp.ssrc_groups[0], sp.get_ssrc_group("XYZ")); } diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc index ea3c6a7c34..3a3d197cc6 100644 --- a/media/engine/webrtc_video_engine.cc +++ b/media/engine/webrtc_video_engine.cc @@ -1869,7 +1869,7 @@ WebRtcVideoSendChannel::WebRtcVideoSendStream::WebRtcVideoSendStream( } WebRtcVideoSendChannel::WebRtcVideoSendStream::~WebRtcVideoSendStream() { - if (stream_ != NULL) { + if (stream_ != nullptr) { call_->DestroyVideoSendStream(stream_); } } @@ -2605,7 +2605,7 @@ WebRtcVideoSendChannel::WebRtcVideoSendStream::GetAggregatedVideoSenderInfo( void WebRtcVideoSendChannel::WebRtcVideoSendStream::FillBitrateInfo( BandwidthEstimationInfo* bwe_info) { RTC_DCHECK_RUN_ON(&thread_checker_); - if (stream_ == NULL) { + if (stream_ == nullptr) { return; } VideoSendStream::Stats stats = stream_->GetStats(); @@ -2659,7 +2659,7 @@ void WebRtcVideoSendChannel::WebRtcVideoSendStream::RecreateWebRtcStream() { config.encoder_settings.enable_frame_instrumentation_generator = true; } - if (stream_ != NULL) { + if (stream_ != nullptr) { // TODO: webrtc:40644448 - Make sure the stats are not updated between // GetStats and DestroyVideoSendStream. VideoSendStream::Stats stats = stream_->GetStats(); @@ -2672,7 +2672,7 @@ void WebRtcVideoSendChannel::WebRtcVideoSendStream::RecreateWebRtcStream() { parameters_.encoder_config.Copy()); } - parameters_.encoder_config.encoder_specific_settings = NULL; + parameters_.encoder_config.encoder_specific_settings = nullptr; // Calls stream_->StartPerRtpStream() to start the VideoSendStream // if necessary conditions are met. @@ -2688,7 +2688,7 @@ void WebRtcVideoSendChannel::WebRtcVideoSendStream::RecreateWebRtcStream() { void WebRtcVideoSendChannel::WebRtcVideoSendStream::GenerateKeyFrame( const std::vector& rids) { RTC_DCHECK_RUN_ON(&thread_checker_); - if (stream_ != NULL) { + if (stream_ != nullptr) { stream_->GenerateKeyFrame(rids); } else { RTC_LOG(LS_WARNING) @@ -3416,12 +3416,12 @@ WebRtcVideoReceiveChannel::WebRtcVideoReceiveStream::WebRtcVideoReceiveStream( const FlexfecReceiveStream::Config& flexfec_config) : call_(call), stream_params_(sp), - stream_(NULL), + stream_(nullptr), default_stream_(default_stream), config_(std::move(config)), flexfec_config_(flexfec_config), flexfec_stream_(nullptr), - sink_(NULL), + sink_(nullptr), first_frame_timestamp_(-1), estimated_remote_start_ntp_time_ms_(0), receiving_(false) { @@ -3725,7 +3725,7 @@ void WebRtcVideoReceiveChannel::WebRtcVideoReceiveStream::OnFrame( if (frame.ntp_time_ms() > 0) estimated_remote_start_ntp_time_ms_ = frame.ntp_time_ms() - elapsed_time_ms; - if (sink_ == NULL) { + if (sink_ == nullptr) { RTC_LOG(LS_WARNING) << "VideoReceiveStreamInterface not connected to a VideoSink."; return; diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc index 8d7f803b44..ffedad59b8 100644 --- a/media/engine/webrtc_video_engine_unittest.cc +++ b/media/engine/webrtc_video_engine_unittest.cc @@ -2294,7 +2294,7 @@ TEST_F(WebRtcVideoChannelBaseTest, SetSendSsrcAfterSetCodecs) { TEST_F(WebRtcVideoChannelBaseTest, SetSink) { RtpPacketReceived packet; packet.SetSsrc(kSsrc); - receive_channel_->SetDefaultSink(NULL); + receive_channel_->SetDefaultSink(nullptr); EXPECT_TRUE(SetDefaultCodec()); EXPECT_TRUE(SetSend(true)); EXPECT_EQ(0, renderer_.num_rendered_frames()); From b215bed9f341fbd21fd9428d3b093cf30026e0a4 Mon Sep 17 00:00:00 2001 From: Sam Maier Date: Thu, 15 May 2025 09:40:47 -0400 Subject: [PATCH 189/292] Using renamed android_ndk_api_level This is meant to accompany this corresponding Chromium change: https://chromium-review.googlesource.com/c/chromium/src/+/6547915 Change-Id: I3cbeb416941b27d14fb5aec9384327c164f7a2c2 Bug: None Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391781 Reviewed-by: Jeremy Leconte Auto-Submit: Sam Maier Reviewed-by: Mirko Bonadei Commit-Queue: Sam Maier Cr-Commit-Position: refs/heads/main@{#44661} --- .gn | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/.gn b/.gn index 5f2f67fbef..d29e7e3ced 100644 --- a/.gn +++ b/.gn @@ -55,8 +55,7 @@ default_args = { ios_deployment_target = "14.0" # The SDK API level, in contrast, is set by build/android/AndroidManifest.xml. - android32_ndk_api_level = 21 - android64_ndk_api_level = 21 + android_ndk_api_level = 21 # WebRTC does not provide the gflags dependency. Because libyuv uses it only # for unittests, it can be disabled (see third_party/libyuv/BUILD.gn) From cc46fb91c5f8bd4cba637f6ebfaa5f278efc4458 Mon Sep 17 00:00:00 2001 From: Harald Alvestrand Date: Fri, 16 May 2025 13:00:26 +0000 Subject: [PATCH 190/292] Use std::numbers constants Part of modernizing to C++20. Bug: None No-IWYU: Other cleanup Change-Id: I2c53789d845486436f37c887b216e7d502c915ac Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392104 Commit-Queue: Harald Alvestrand Reviewed-by: Evan Shrubsole Cr-Commit-Position: refs/heads/main@{#44662} --- audio/audio_state_unittest.cc | 3 ++- common_audio/window_generator.cc | 5 +++-- .../neteq/dtmf_tone_generator_unittest.cc | 3 ++- modules/audio_mixer/sine_wave_generator.cc | 4 +++- .../aec3/comfort_noise_generator.cc | 9 +++++++-- modules/audio_processing/aec3/decimator_unittest.cc | 3 ++- .../aec3/render_signal_analyzer_unittest.cc | 3 ++- .../aec3/suppression_filter_unittest.cc | 3 ++- modules/audio_processing/agc/utility.cc | 4 +++- .../audio_processing/agc2/agc2_testing_common.cc | 4 +++- modules/audio_processing/ns/fast_math.cc | 6 ++++-- modules/audio_processing/ns/noise_estimator.cc | 8 +++++++- modules/audio_processing/three_band_filter_bank.cc | 13 +++++++++---- .../event_based_exponential_moving_average.cc | 3 ++- rtc_base/numerics/percentile_filter_unittest.cc | 9 +++++---- rtc_base/random.cc | 4 +++- rtc_base/random_unittest.cc | 3 ++- 17 files changed, 61 insertions(+), 26 deletions(-) diff --git a/audio/audio_state_unittest.cc b/audio/audio_state_unittest.cc index 6a98bdd357..c886a13b86 100644 --- a/audio/audio_state_unittest.cc +++ b/audio/audio_state_unittest.cc @@ -11,6 +11,7 @@ #include "audio/audio_state.h" #include +#include #include #include @@ -150,7 +151,7 @@ std::vector Create10msTestData(int sample_rate_hz, const int samples_per_channel = sample_rate_hz / 100; std::vector audio_data(samples_per_channel * num_channels, 0); // Fill the first channel with a 1kHz sine wave. - const float inc = (2 * 3.14159265f * 1000) / sample_rate_hz; + const float inc = (2 * std::numbers::pi_v * 1000) / sample_rate_hz; float w = 0.f; for (int i = 0; i < samples_per_channel; ++i) { audio_data[i * num_channels] = static_cast(32767.f * std::sin(w)); diff --git a/common_audio/window_generator.cc b/common_audio/window_generator.cc index da5603d9e7..7c33d255f0 100644 --- a/common_audio/window_generator.cc +++ b/common_audio/window_generator.cc @@ -14,6 +14,7 @@ #include #include +#include #include "rtc_base/checks.h" @@ -41,7 +42,7 @@ void WindowGenerator::Hanning(int length, float* window) { RTC_CHECK(window != nullptr); for (int i = 0; i < length; ++i) { window[i] = - 0.5f * (1 - cosf(2 * static_cast(M_PI) * i / (length - 1))); + 0.5f * (1 - cosf(2 * std::numbers::pi_v * i / (length - 1))); } } @@ -56,7 +57,7 @@ void WindowGenerator::KaiserBesselDerived(float alpha, for (size_t i = 0; i <= half; ++i) { complex r = (4.0f * i) / length - 1.0f; - sum += I0(static_cast(M_PI) * alpha * sqrt(1.0f - r * r)).real(); + sum += I0(std::numbers::pi_v * alpha * sqrt(1.0f - r * r)).real(); window[i] = sum; } for (size_t i = length - 1; i >= half; --i) { diff --git a/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc b/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc index 10e4c21309..7a7d65aeff 100644 --- a/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc +++ b/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc @@ -15,6 +15,7 @@ #include #include +#include #include "common_audio/include/audio_util.h" #include "modules/audio_coding/neteq/audio_multi_vector.h" @@ -46,7 +47,7 @@ class DtmfToneGeneratorTest : public ::testing::Test { double f1 = kLowFreqHz[event]; double f2 = kHighFreqHz[event]; - const double pi = 3.14159265358979323846; + const double pi = std::numbers::pi; for (int n = 0; n < kNumSamples; ++n) { double x = k3dbAttenuation * sin(2.0 * pi * f1 / fs_hz * (-n - 1)) + diff --git a/modules/audio_mixer/sine_wave_generator.cc b/modules/audio_mixer/sine_wave_generator.cc index deb88468d6..be3dd0cfa6 100644 --- a/modules/audio_mixer/sine_wave_generator.cc +++ b/modules/audio_mixer/sine_wave_generator.cc @@ -13,12 +13,14 @@ #include #include +#include + #include "rtc_base/numerics/safe_conversions.h" namespace webrtc { namespace { -constexpr float kPi = 3.14159265f; +constexpr float kPi = std::numbers::pi_v; } // namespace void SineWaveGenerator::GenerateNextFrame(AudioFrame* frame) { diff --git a/modules/audio_processing/aec3/comfort_noise_generator.cc b/modules/audio_processing/aec3/comfort_noise_generator.cc index ed643f4400..17a3b70d36 100644 --- a/modules/audio_processing/aec3/comfort_noise_generator.cc +++ b/modules/audio_processing/aec3/comfort_noise_generator.cc @@ -21,6 +21,7 @@ #include #include #include +#include #include #include "common_audio/signal_processing/include/signal_processing_library.h" @@ -31,6 +32,8 @@ namespace webrtc { namespace { +using std::numbers::sqrt2_v; + // Computes the noise floor value that matches a WGN input of noise_floor_dbfs. float GetNoiseFloorFactor(float noise_floor_dbfs) { // kdBfsNormalization = 20.f*log10(32768.f). @@ -39,14 +42,16 @@ float GetNoiseFloorFactor(float noise_floor_dbfs) { } // Table of sqrt(2) * sin(2*pi*i/32). +// clang-format off constexpr float kSqrt2Sin[32] = { +0.0000000f, +0.2758994f, +0.5411961f, +0.7856950f, +1.0000000f, - +1.1758756f, +1.3065630f, +1.3870398f, +1.4142136f, +1.3870398f, + +1.1758756f, +1.3065630f, +1.3870398f, +sqrt2_v, +1.3870398f, +1.3065630f, +1.1758756f, +1.0000000f, +0.7856950f, +0.5411961f, +0.2758994f, +0.0000000f, -0.2758994f, -0.5411961f, -0.7856950f, - -1.0000000f, -1.1758756f, -1.3065630f, -1.3870398f, -1.4142136f, + -1.0000000f, -1.1758756f, -1.3065630f, -1.3870398f, -sqrt2_v, -1.3870398f, -1.3065630f, -1.1758756f, -1.0000000f, -0.7856950f, -0.5411961f, -0.2758994f}; +// clang-format on void GenerateComfortNoise(Aec3Optimization optimization, const std::array& N2, diff --git a/modules/audio_processing/aec3/decimator_unittest.cc b/modules/audio_processing/aec3/decimator_unittest.cc index 3ff4294fbb..2c6d96de6d 100644 --- a/modules/audio_processing/aec3/decimator_unittest.cc +++ b/modules/audio_processing/aec3/decimator_unittest.cc @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include @@ -35,7 +36,7 @@ std::string ProduceDebugText(int sample_rate_hz) { } constexpr size_t kDownSamplingFactors[] = {2, 4, 8}; -constexpr float kPi = 3.141592f; +constexpr float kPi = std::numbers::pi_v; constexpr size_t kNumStartupBlocks = 50; constexpr size_t kNumBlocks = 1000; diff --git a/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc b/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc index e51be2faac..9ef60406a3 100644 --- a/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc +++ b/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc @@ -14,6 +14,7 @@ #include #include +#include #include #include "api/array_view.h" @@ -29,7 +30,7 @@ namespace webrtc { namespace { -constexpr float kPi = 3.141592f; +constexpr float kPi = std::numbers::pi_v; void ProduceSinusoidInNoise(int sample_rate_hz, size_t sinusoid_channel, diff --git a/modules/audio_processing/aec3/suppression_filter_unittest.cc b/modules/audio_processing/aec3/suppression_filter_unittest.cc index 464f5cfed2..8ecc7ecfea 100644 --- a/modules/audio_processing/aec3/suppression_filter_unittest.cc +++ b/modules/audio_processing/aec3/suppression_filter_unittest.cc @@ -14,6 +14,7 @@ #include #include +#include #include #include "test/gtest.h" @@ -21,7 +22,7 @@ namespace webrtc { namespace { -constexpr float kPi = 3.141592f; +constexpr float kPi = std::numbers::pi_v; void ProduceSinusoid(int sample_rate_hz, float sinusoidal_frequency_hz, diff --git a/modules/audio_processing/agc/utility.cc b/modules/audio_processing/agc/utility.cc index 2a87e5ce74..b8fa3605ea 100644 --- a/modules/audio_processing/agc/utility.cc +++ b/modules/audio_processing/agc/utility.cc @@ -12,9 +12,11 @@ #include +#include + namespace webrtc { -static const double kLog10 = 2.30258509299; +static const double kLog10 = std::numbers::ln10; static const double kLinear2DbScale = 20.0 / kLog10; static const double kLinear2LoudnessScale = 13.4 / kLog10; diff --git a/modules/audio_processing/agc2/agc2_testing_common.cc b/modules/audio_processing/agc2/agc2_testing_common.cc index 8f3e9db4c4..b9d0492fca 100644 --- a/modules/audio_processing/agc2/agc2_testing_common.cc +++ b/modules/audio_processing/agc2/agc2_testing_common.cc @@ -12,6 +12,8 @@ #include +#include + #include "rtc_base/checks.h" namespace webrtc { @@ -56,7 +58,7 @@ SineGenerator::SineGenerator(float amplitude, } float SineGenerator::operator()() { - constexpr float kPi = 3.1415926536f; + constexpr float kPi = std::numbers::pi_v; x_radians_ += frequency_hz_ / sample_rate_hz_ * 2 * kPi; if (x_radians_ >= 2 * kPi) { x_radians_ -= 2 * kPi; diff --git a/modules/audio_processing/ns/fast_math.cc b/modules/audio_processing/ns/fast_math.cc index 72f6fb2145..7b8fe1f55b 100644 --- a/modules/audio_processing/ns/fast_math.cc +++ b/modules/audio_processing/ns/fast_math.cc @@ -13,6 +13,8 @@ #include #include +#include + #include "rtc_base/checks.h" namespace webrtc { @@ -53,7 +55,7 @@ float PowApproximation(float x, float p) { } float LogApproximation(float x) { - constexpr float kLogOf2 = 0.69314718056f; + constexpr float kLogOf2 = std::numbers::ln2_v; return FastLog2f(x) * kLogOf2; } @@ -64,7 +66,7 @@ void LogApproximation(ArrayView x, ArrayView y) { } float ExpApproximation(float x) { - constexpr float kLog10Ofe = 0.4342944819f; + constexpr float kLog10Ofe = std::numbers::log10e_v; return PowApproximation(10.f, x * kLog10Ofe); } diff --git a/modules/audio_processing/ns/noise_estimator.cc b/modules/audio_processing/ns/noise_estimator.cc index bf21d38d0b..00b647ca35 100644 --- a/modules/audio_processing/ns/noise_estimator.cc +++ b/modules/audio_processing/ns/noise_estimator.cc @@ -14,6 +14,7 @@ #include #include #include +#include #include "api/array_view.h" #include "modules/audio_processing/ns/fast_math.h" @@ -25,10 +26,14 @@ namespace webrtc { namespace { +using std::numbers::ln10_v; + // Log(i). +// clang-format off constexpr std::array log_table = { 0.f, 0.f, 0.f, 0.f, 0.f, 1.609438f, 1.791759f, - 1.945910f, 2.079442f, 2.197225f, 2.302585f, 2.397895f, 2.484907f, 2.564949f, + 1.945910f, 2.079442f, 2.197225f, ln10_v, 2.397895f, 2.484907f, + 2.564949f, 2.639057f, 2.708050f, 2.772589f, 2.833213f, 2.890372f, 2.944439f, 2.995732f, 3.044522f, 3.091043f, 3.135494f, 3.178054f, 3.218876f, 3.258097f, 3.295837f, 3.332205f, 3.367296f, 3.401197f, 3.433987f, 3.465736f, 3.496507f, 3.526361f, @@ -46,6 +51,7 @@ constexpr std::array log_table = { 4.718499f, 4.727388f, 4.736198f, 4.744932f, 4.753591f, 4.762174f, 4.770685f, 4.779124f, 4.787492f, 4.795791f, 4.804021f, 4.812184f, 4.820282f, 4.828314f, 4.836282f, 4.844187f, 4.852030f}; +// clang-format on } // namespace diff --git a/modules/audio_processing/three_band_filter_bank.cc b/modules/audio_processing/three_band_filter_bank.cc index d69db9bf9a..45b44a7762 100644 --- a/modules/audio_processing/three_band_filter_bank.cc +++ b/modules/audio_processing/three_band_filter_bank.cc @@ -33,6 +33,7 @@ #include "modules/audio_processing/three_band_filter_bank.h" #include +#include #include "rtc_base/checks.h" @@ -88,17 +89,21 @@ const float constexpr int kZeroFilterIndex1 = 3; constexpr int kZeroFilterIndex2 = 9; +constexpr float kSqrt3 = std::numbers::sqrt3_v; + +// clang-format off const float kDctModulation[ThreeBandFilterBank::kNumNonZeroFilters][kDctSize] = {{2.f, 2.f, 2.f}, - {1.73205077f, 0.f, -1.73205077f}, + {kSqrt3, 0.f, -kSqrt3}, {1.f, -2.f, 1.f}, {-1.f, 2.f, -1.f}, - {-1.73205077f, 0.f, 1.73205077f}, + {-kSqrt3, 0.f, kSqrt3}, {-2.f, -2.f, -2.f}, - {-1.73205077f, 0.f, 1.73205077f}, + {-kSqrt3, 0.f, kSqrt3}, {-1.f, 2.f, -1.f}, {1.f, -2.f, 1.f}, - {1.73205077f, 0.f, -1.73205077f}}; + {kSqrt3, 0.f, -kSqrt3}}; +// clang-format on // Filters the input signal `in` with the filter `filter` using a shift by // `in_shift`, taking into account the previous state. diff --git a/rtc_base/numerics/event_based_exponential_moving_average.cc b/rtc_base/numerics/event_based_exponential_moving_average.cc index 27b2066a97..419902dc52 100644 --- a/rtc_base/numerics/event_based_exponential_moving_average.cc +++ b/rtc_base/numerics/event_based_exponential_moving_average.cc @@ -13,6 +13,7 @@ #include #include #include +#include #include "rtc_base/checks.h" @@ -35,7 +36,7 @@ EventBasedExponentialMovingAverage::EventBasedExponentialMovingAverage( } void EventBasedExponentialMovingAverage::SetHalfTime(int half_time) { - tau_ = static_cast(half_time) / log(2); + tau_ = static_cast(half_time) / std::numbers::ln2; Reset(); } diff --git a/rtc_base/numerics/percentile_filter_unittest.cc b/rtc_base/numerics/percentile_filter_unittest.cc index d6baa32001..d6826f42e1 100644 --- a/rtc_base/numerics/percentile_filter_unittest.cc +++ b/rtc_base/numerics/percentile_filter_unittest.cc @@ -15,6 +15,7 @@ #include #include #include +#include #include #include "absl/algorithm/container.h" @@ -58,10 +59,10 @@ TEST(PercentileFilterTest, MaxFilter) { TEST(PercentileFilterTest, MedianFilterDouble) { PercentileFilter filter(0.5f); - filter.Insert(2.71828); - filter.Insert(3.14159); - filter.Insert(1.41421); - EXPECT_EQ(2.71828, filter.GetPercentileValue()); + filter.Insert(std::numbers::e); + filter.Insert(std::numbers::pi); + filter.Insert(std::numbers::sqrt2); + EXPECT_FLOAT_EQ(std::numbers::e, filter.GetPercentileValue()); } TEST(PercentileFilterTest, MedianFilterInt) { diff --git a/rtc_base/random.cc b/rtc_base/random.cc index 55b8749cb7..28e48f0d3d 100644 --- a/rtc_base/random.cc +++ b/rtc_base/random.cc @@ -11,6 +11,8 @@ #include +#include + #include "rtc_base/checks.h" #include "rtc_base/numerics/safe_conversions.h" @@ -71,7 +73,7 @@ double Random::Gaussian(double mean, double standard_deviation) { // interval (0, 1]. Note that we rely on NextOutput to generate integers // in the range [1, 2^64-1]. Normally this behavior is a bit frustrating, // but here it is exactly what we need. - const double kPi = 3.14159265358979323846; + const double kPi = std::numbers::pi; double u1 = static_cast(NextOutput()) / static_cast(0xFFFFFFFFFFFFFFFFull); double u2 = static_cast(NextOutput()) / diff --git a/rtc_base/random_unittest.cc b/rtc_base/random_unittest.cc index 4eb6f754eb..ffcd3ed72a 100644 --- a/rtc_base/random_unittest.cc +++ b/rtc_base/random_unittest.cc @@ -13,6 +13,7 @@ #include #include +#include #include #include "rtc_base/numerics/math_utils.h" // unsigned difference @@ -286,7 +287,7 @@ TEST(RandomNumberGeneratorTest, Gaussian) { } } - const double kPi = 3.14159265358979323846; + const double kPi = std::numbers::pi; const double kScale = 1 / (kStddev * sqrt(2.0 * kPi)); const double kDiv = -2.0 * kStddev * kStddev; for (int n = 0; n < kBuckets; ++n) { From 21043c8d87a44b20581ed94ccadf93d68f79039c Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Fri, 16 May 2025 07:02:01 -0700 Subject: [PATCH 191/292] Roll chromium_revision 083b3eb60c..07a04e1af2 (1461214:1461321) Change log: https://chromium.googlesource.com/chromium/src/+log/083b3eb60c..07a04e1af2 Full diff: https://chromium.googlesource.com/chromium/src/+/083b3eb60c..07a04e1af2 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/1f29789660..25ef6cb7fa * src/build: https://chromium.googlesource.com/chromium/src/build/+log/db8ce42b66..34cf410968 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/a71b3cb45c..7d88136926 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/7ba47582ec..c5062b897d * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/ec61df4b22..f784079038 * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/ffd1c50cbd..41104509db * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/109a2ebe2b..99c9ef8102 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/17fd1f3675..73d3d2d400 DEPS diff: https://chromium.googlesource.com/chromium/src/+/083b3eb60c..07a04e1af2/DEPS No update to Clang. BUG=None Change-Id: I1a7609a37cebdd1e8cb277e1f6f941bfcda10adf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392180 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44663} --- DEPS | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/DEPS b/DEPS index 54147a936c..24c2b4fad0 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '083b3eb60c3bf65c15ef2e737a78bb18c7709374', + 'chromium_revision': '07a04e1af2558471e7de506d6e2d94d7572ef966', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@1f29789660c727524886a1a662b5e58058fa42f7', + 'https://chromium.googlesource.com/chromium/src/base@25ef6cb7fae4eb08709090486e741f403e64c4e8', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@db8ce42b66262db8f53a3dc40c6241c23ed97243', + 'https://chromium.googlesource.com/chromium/src/build@34cf410968a36b915e3ffda3007ff9c291aac27a', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@e9b36e9ffb02b268d194c3f82f89952d09d0c714', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@a71b3cb45c04b1c1baa034882d0157f52425ae3e', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@7d881369269f03a0adbd93ba24a715b30bd3eae8', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@7ba47582ec28f9e542ac3e6186a2d4c5d8775da3', + 'https://chromium.googlesource.com/chromium/src/testing@c5062b897df027d714bb5ede51ec8bc764b7a0da', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@ec61df4b2281ada165462e76d55a972be0d58e96', + 'https://chromium.googlesource.com/chromium/src/third_party@f784079038e18926b1dfb532a606b7fb75a7a20b', 'src/buildtools/linux64': { 'packages': [ @@ -484,7 +484,7 @@ deps = { 'src/third_party/breakpad/breakpad': 'https://chromium.googlesource.com/breakpad/breakpad.git@fad82477312f786994151f01c352813fabf924c0', 'src/third_party/catapult': - 'https://chromium.googlesource.com/catapult.git@ffd1c50cbd19e3f0c5de2fd1d20b43adc04b6e85', + 'https://chromium.googlesource.com/catapult.git@41104509db3116fd5a5e3af4f8242b59de7adc67', 'src/third_party/ced/src': { 'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5', }, @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '109a2ebe2b65215b8ba3ff0d6a3a8361220a0992', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '99c9ef8102d3770e6d5ba91d9ca400b751b0af9b', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@17fd1f36756107cbdc25ca57d87a56ecde7a1824', + 'https://chromium.googlesource.com/chromium/src/tools@73d3d2d4007a1d9255da7d5dde863a6d9c666ee1', 'src/third_party/espresso': { 'packages': [ From 430d5a390d63b002203edc446b24c628f1db2bba Mon Sep 17 00:00:00 2001 From: Jeremy Leconte Date: Fri, 16 May 2025 16:21:41 +0200 Subject: [PATCH 192/292] Stop using update.py and update_rust.py Starting from [1], both clang and Rust toolchains are updated using GCS. For this reason, the previous update method based on python scripts can be removed. [1]- https://webrtc-review.googlesource.com/c/src/+/389541 Bug: None Change-Id: Iebd9b6095deecf4edfa009d867182a6ee0c4f864 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392105 Reviewed-by: Mirko Bonadei Commit-Queue: Jeremy Leconte Cr-Commit-Position: refs/heads/main@{#44664} --- DEPS | 28 -------------------- tools_webrtc/autoroller/roll_deps.py | 38 +--------------------------- 2 files changed, 1 insertion(+), 65 deletions(-) diff --git a/DEPS b/DEPS index 24c2b4fad0..ebccc019a4 100644 --- a/DEPS +++ b/DEPS @@ -2197,34 +2197,6 @@ hooks = [ '{checkout_fuchsia_boot_images}', ], }, - { - # Note: On Win, this should run after win_toolchain, as it may use it. - 'name': 'clang', - 'pattern': '.', - 'action': ['python3', 'src/tools/clang/scripts/update.py'], - }, - { - # This is supposed to support the same set of platforms as 'clang' above. - 'name': 'clang_coverage', - 'pattern': '.', - 'condition': 'checkout_clang_coverage_tools', - 'action': ['python3', 'src/tools/clang/scripts/update.py', - '--package=coverage_tools'], - }, - { - # This is also supposed to support the same set of platforms as 'clang' - # above. LLVM ToT support isn't provided at the moment. - 'name': 'clangd', - 'pattern': '.', - 'condition': 'checkout_clangd', - 'action': ['vpython3', 'src/tools/clang/scripts/update.py', - '--package=clangd'], - }, - { - 'name': 'rust', - 'pattern': '.', - 'action': ['python3', 'src/tools/rust/update_rust.py'], - }, { # Update LASTCHANGE. 'name': 'lastchange', diff --git a/tools_webrtc/autoroller/roll_deps.py b/tools_webrtc/autoroller/roll_deps.py index e0af99a309..7165f8fdd6 100755 --- a/tools_webrtc/autoroller/roll_deps.py +++ b/tools_webrtc/autoroller/roll_deps.py @@ -97,10 +97,6 @@ def FindRootPath(): find_depot_tools.add_depot_tools_to_path() -CLANG_UPDATE_SCRIPT_URL_PATH = 'tools/clang/scripts/update.py' -CLANG_UPDATE_SCRIPT_LOCAL_PATH = os.path.join(CHECKOUT_ROOT_DIR, 'tools', - 'clang', 'scripts', 'update.py') - DepsEntry = collections.namedtuple('DepsEntry', 'path url revision') ChangedDep = collections.namedtuple('ChangedDep', 'path url current_rev new_rev') @@ -523,26 +519,6 @@ def CalculateChangedDeps(webrtc_deps, new_cr_deps): return sorted(result) -def CalculateChangedClang(new_cr_rev): - - def GetClangRev(lines): - for line in lines: - match = CLANG_REVISION_RE.match(line) - if match: - return match.group(1) - raise RollError('Could not parse Clang revision!') - - with open(CLANG_UPDATE_SCRIPT_LOCAL_PATH, 'r') as f: - current_lines = f.readlines() - current_rev = GetClangRev(current_lines) - - new_clang_update_py = ReadRemoteCrFile(CLANG_UPDATE_SCRIPT_URL_PATH, - new_cr_rev).splitlines() - new_rev = GetClangRev(new_clang_update_py) - return ChangedDep(CLANG_UPDATE_SCRIPT_LOCAL_PATH, None, current_rev, - new_rev) - - def GenerateCommitMessage( rev_update, current_commit_pos, @@ -550,7 +526,6 @@ def GenerateCommitMessage( changed_deps_list, added_deps_paths=None, removed_deps_paths=None, - clang_change=None, ): current_cr_rev = rev_update.current_chromium_rev[0:10] new_cr_rev = rev_update.new_chromium_rev[0:10] @@ -600,15 +575,6 @@ def Section(adjective, deps): else: commit_msg.append('No dependencies changed.') - if clang_change and clang_change.current_rev != clang_change.new_rev: - commit_msg.append('Clang version changed %s:%s' % - (clang_change.current_rev, clang_change.new_rev)) - change_url = CHROMIUM_FILE_TEMPLATE % (rev_interval, - CLANG_UPDATE_SCRIPT_URL_PATH) - commit_msg.append('Details: %s\n' % change_url) - else: - commit_msg.append('No update to Clang.\n') - commit_msg.append('BUG=None') return '\n'.join(commit_msg) @@ -857,15 +823,13 @@ def main(): 'Remove them or add them to either ' 'WEBRTC_ONLY_DEPS or DONT_AUTOROLL_THESE.' % other_deps) - clang_change = CalculateChangedClang(rev_update.new_chromium_rev) commit_msg = GenerateCommitMessage( rev_update, current_commit_pos, new_commit_pos, changed_deps, added_deps_paths=new_generated_android_deps, - removed_deps_paths=removed_generated_android_deps, - clang_change=clang_change) + removed_deps_paths=removed_generated_android_deps) logging.debug('Commit message:\n%s', commit_msg) _CreateRollBranch(opts.dry_run) From 1135f82c7ce36f8f0441bc4e9d16b33756ec5d6b Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Thu, 15 May 2025 11:50:28 +0000 Subject: [PATCH 193/292] [cleanup] Replace uses of NULL with nullptr MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Generated using clang-tidy, tools/clang/scripts/build_clang_tools_extra.py \ --fetch out/Default clang-tidy clang-apply-replacements ninja -C out/Default gn gen out/Default --export-compile-commands cd out/Default tools/clang/third_party/llvm/clang-tools-extra/clang-tidy/tool/run-clang-tidy.py -p . \ -clang-tidy-binary out/Default/tools/clang/third_party/llvm/build/bin/clang-tidy \ -clang-apply-replacements-binary \ out/Default/tools/clang/third_party/llvm/build/bin/clang-apply-replacements \ -checks='-*,modernize-use-nullptr' \ -fix This CL was uploaded by git cl split. Bug: webrtc:42232237 Change-Id: I4847ca2b337a4507ae9daf1dbd5f52e7cd15b55d No-Iwyu: LSC Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391843 Auto-Submit: Evan Shrubsole Commit-Queue: Erik Språng Reviewed-by: Erik Språng Cr-Commit-Position: refs/heads/main@{#44665} --- common_video/libyuv/libyuv_unittest.cc | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/common_video/libyuv/libyuv_unittest.cc b/common_video/libyuv/libyuv_unittest.cc index fed19fd2e1..375eb4a3fd 100644 --- a/common_video/libyuv/libyuv_unittest.cc +++ b/common_video/libyuv/libyuv_unittest.cc @@ -95,7 +95,7 @@ class TestLibYuv : public ::testing::Test { }; TestLibYuv::TestLibYuv() - : source_file_(NULL), + : source_file_(nullptr), orig_frame_(), width_(352), height_(288), @@ -106,7 +106,7 @@ TestLibYuv::TestLibYuv() void TestLibYuv::SetUp() { const std::string input_file_name = test::ResourcePath("foreman_cif", "yuv"); source_file_ = fopen(input_file_name.c_str(), "rb"); - ASSERT_TRUE(source_file_ != NULL) + ASSERT_TRUE(source_file_ != nullptr) << "Cannot read file: " << input_file_name << "\n"; scoped_refptr buffer( @@ -120,10 +120,10 @@ void TestLibYuv::SetUp() { } void TestLibYuv::TearDown() { - if (source_file_ != NULL) { + if (source_file_ != nullptr) { ASSERT_EQ(0, fclose(source_file_)); } - source_file_ = NULL; + source_file_ = nullptr; } TEST_F(TestLibYuv, ConvertTest) { @@ -132,7 +132,7 @@ TEST_F(TestLibYuv, ConvertTest) { std::string output_file_name = test::OutputPath() + "LibYuvTest_conversion.yuv"; FILE* output_file = fopen(output_file_name.c_str(), "wb"); - ASSERT_TRUE(output_file != NULL); + ASSERT_TRUE(output_file != nullptr); double psnr = 0.0; @@ -298,7 +298,7 @@ TEST_F(TestLibYuv, ConvertAlignedFrame) { std::string output_file_name = test::OutputPath() + "LibYuvTest_conversion.yuv"; FILE* output_file = fopen(output_file_name.c_str(), "wb"); - ASSERT_TRUE(output_file != NULL); + ASSERT_TRUE(output_file != nullptr); double psnr = 0.0; From bbba66f6c7f9b2be65ec8a70825c5d230b503529 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Thu, 15 May 2025 11:53:56 +0000 Subject: [PATCH 194/292] [cleanup] Replace uses of NULL with nullptr MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Generated using clang-tidy, tools/clang/scripts/build_clang_tools_extra.py \ --fetch out/Default clang-tidy clang-apply-replacements ninja -C out/Default gn gen out/Default --export-compile-commands cd out/Default tools/clang/third_party/llvm/clang-tools-extra/clang-tidy/tool/run-clang-tidy.py -p . \ -clang-tidy-binary out/Default/tools/clang/third_party/llvm/build/bin/clang-tidy \ -clang-apply-replacements-binary \ out/Default/tools/clang/third_party/llvm/build/bin/clang-apply-replacements \ -checks='-*,modernize-use-nullptr' \ -fix This CL was uploaded by git cl split. Bug: webrtc:42232237 Change-Id: I8e519d0facafeef8bdc535057bdc9dfb07b77092 No-Iwyu: LSC Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391805 Auto-Submit: Evan Shrubsole Commit-Queue: Erik Språng Reviewed-by: Erik Språng Cr-Commit-Position: refs/heads/main@{#44666} --- test/configurable_frame_size_encoder.cc | 2 +- test/gl/gl_renderer.cc | 4 ++-- test/linux/glx_renderer.cc | 32 ++++++++++++------------- test/linux/video_renderer_linux.cc | 4 ++-- test/rtp_file_reader_unittest.cc | 4 ++-- test/rtp_file_writer.cc | 12 +++++----- test/rtp_file_writer_unittest.cc | 6 ++--- test/testsupport/file_utils.cc | 2 +- test/testsupport/file_utils_unittest.cc | 4 ++-- test/testsupport/perf_test.cc | 2 +- test/testsupport/y4m_frame_generator.cc | 2 +- test/testsupport/y4m_frame_reader.cc | 4 ++-- test/testsupport/yuv_frame_reader.cc | 2 +- 13 files changed, 40 insertions(+), 40 deletions(-) diff --git a/test/configurable_frame_size_encoder.cc b/test/configurable_frame_size_encoder.cc index e18a87cd7e..51274f49db 100644 --- a/test/configurable_frame_size_encoder.cc +++ b/test/configurable_frame_size_encoder.cc @@ -26,7 +26,7 @@ namespace test { ConfigurableFrameSizeEncoder::ConfigurableFrameSizeEncoder( size_t max_frame_size) - : callback_(NULL), + : callback_(nullptr), current_frame_size_(max_frame_size), codec_type_(kVideoCodecGeneric) {} diff --git a/test/gl/gl_renderer.cc b/test/gl/gl_renderer.cc index 21a11953f7..4a16419b8a 100644 --- a/test/gl/gl_renderer.cc +++ b/test/gl/gl_renderer.cc @@ -19,7 +19,7 @@ namespace webrtc { namespace test { GlRenderer::GlRenderer() - : is_init_(false), buffer_(NULL), width_(0), height_(0) {} + : is_init_(false), buffer_(nullptr), width_(0), height_(0) {} void GlRenderer::Init() { RTC_DCHECK(!is_init_); @@ -36,7 +36,7 @@ void GlRenderer::Destroy() { is_init_ = false; delete[] buffer_; - buffer_ = NULL; + buffer_ = nullptr; glDeleteTextures(1, &texture_); } diff --git a/test/linux/glx_renderer.cc b/test/linux/glx_renderer.cc index 43f2cdafd8..8e66b9272c 100644 --- a/test/linux/glx_renderer.cc +++ b/test/linux/glx_renderer.cc @@ -18,7 +18,7 @@ namespace webrtc { namespace test { GlxRenderer::GlxRenderer(size_t width, size_t height) - : width_(width), height_(height), display_(NULL), context_(NULL) { + : width_(width), height_(height), display_(nullptr), context_(nullptr) { RTC_DCHECK_GT(width, 0); RTC_DCHECK_GT(height, 0); } @@ -28,7 +28,7 @@ GlxRenderer::~GlxRenderer() { } bool GlxRenderer::Init(const char* window_title) { - if ((display_ = XOpenDisplay(NULL)) == NULL) { + if ((display_ = XOpenDisplay(nullptr)) == nullptr) { Destroy(); return false; } @@ -41,13 +41,13 @@ bool GlxRenderer::Init(const char* window_title) { GLX_BLUE_SIZE, 4, GLX_DEPTH_SIZE, 16, None, }; - if ((vi = glXChooseVisual(display_, screen, attr_list)) == NULL) { + if ((vi = glXChooseVisual(display_, screen, attr_list)) == nullptr) { Destroy(); return false; } - context_ = glXCreateContext(display_, vi, 0, true); - if (context_ == NULL) { + context_ = glXCreateContext(display_, vi, nullptr, true); + if (context_ == nullptr) { Destroy(); return false; } @@ -64,7 +64,7 @@ bool GlxRenderer::Init(const char* window_title) { XFree(vi); XSetStandardProperties(display_, window_, window_title, window_title, None, - NULL, 0, NULL); + nullptr, 0, nullptr); Atom wm_delete = XInternAtom(display_, "WM_DELETE_WINDOW", True); if (wm_delete != None) { @@ -78,7 +78,7 @@ bool GlxRenderer::Init(const char* window_title) { return false; } GlRenderer::Init(); - if (!glXMakeCurrent(display_, None, NULL)) { + if (!glXMakeCurrent(display_, None, nullptr)) { Destroy(); return false; } @@ -88,17 +88,17 @@ bool GlxRenderer::Init(const char* window_title) { } void GlxRenderer::Destroy() { - if (context_ != NULL) { + if (context_ != nullptr) { glXMakeCurrent(display_, window_, context_); GlRenderer::Destroy(); - glXMakeCurrent(display_, None, NULL); + glXMakeCurrent(display_, None, nullptr); glXDestroyContext(display_, context_); - context_ = NULL; + context_ = nullptr; } - if (display_ != NULL) { + if (display_ != nullptr) { XCloseDisplay(display_); - display_ = NULL; + display_ = nullptr; } } @@ -109,7 +109,7 @@ GlxRenderer* GlxRenderer::Create(const char* window_title, if (!glx_renderer->Init(window_title)) { // TODO(pbos): Add GLX-failed warning here? delete glx_renderer; - return NULL; + return nullptr; } return glx_renderer; } @@ -121,12 +121,12 @@ void GlxRenderer::Resize(size_t width, size_t height) { abort(); } GlRenderer::ResizeViewport(width_, height_); - if (!glXMakeCurrent(display_, None, NULL)) { + if (!glXMakeCurrent(display_, None, nullptr)) { abort(); } XSizeHints* size_hints = XAllocSizeHints(); - if (size_hints == NULL) { + if (size_hints == nullptr) { abort(); } size_hints->flags = PAspect; @@ -167,7 +167,7 @@ void GlxRenderer::OnFrame(const VideoFrame& frame) { GlRenderer::OnFrame(frame); glXSwapBuffers(display_, window_); - if (!glXMakeCurrent(display_, None, NULL)) { + if (!glXMakeCurrent(display_, None, nullptr)) { abort(); } } diff --git a/test/linux/video_renderer_linux.cc b/test/linux/video_renderer_linux.cc index 74c95235df..3d07bd0326 100644 --- a/test/linux/video_renderer_linux.cc +++ b/test/linux/video_renderer_linux.cc @@ -19,10 +19,10 @@ VideoRenderer* VideoRenderer::CreatePlatformRenderer(const char* window_title, size_t width, size_t height) { GlxRenderer* glx_renderer = GlxRenderer::Create(window_title, width, height); - if (glx_renderer != NULL) { + if (glx_renderer != nullptr) { return glx_renderer; } - return NULL; + return nullptr; } } // namespace test } // namespace webrtc diff --git a/test/rtp_file_reader_unittest.cc b/test/rtp_file_reader_unittest.cc index eea88c9d97..eb15d873f0 100644 --- a/test/rtp_file_reader_unittest.cc +++ b/test/rtp_file_reader_unittest.cc @@ -27,7 +27,7 @@ class TestRtpFileReader : public ::testing::Test { test::ResourcePath("video_coding/" + filename, "rtp"); rtp_packet_source_.reset( test::RtpFileReader::Create(test::RtpFileReader::kRtpDump, filepath)); - ASSERT_TRUE(rtp_packet_source_.get() != NULL); + ASSERT_TRUE(rtp_packet_source_.get() != nullptr); headers_only_file_ = headers_only_file; } @@ -68,7 +68,7 @@ class TestPcapFileReader : public ::testing::Test { test::ResourcePath("video_coding/" + filename, "pcap"); rtp_packet_source_.reset( test::RtpFileReader::Create(test::RtpFileReader::kPcap, filepath)); - ASSERT_TRUE(rtp_packet_source_.get() != NULL); + ASSERT_TRUE(rtp_packet_source_.get() != nullptr); } int CountRtpPackets() { diff --git a/test/rtp_file_writer.cc b/test/rtp_file_writer.cc index f992756aab..118d8cdd29 100644 --- a/test/rtp_file_writer.cc +++ b/test/rtp_file_writer.cc @@ -29,13 +29,13 @@ static const char kFirstLine[] = "#!rtpplay1.0 0.0.0.0/0\n"; class RtpDumpWriter : public RtpFileWriter { public: explicit RtpDumpWriter(FILE* file) : file_(file) { - RTC_CHECK(file_ != NULL); + RTC_CHECK(file_ != nullptr); Init(); } ~RtpDumpWriter() override { - if (file_ != NULL) { + if (file_ != nullptr) { fclose(file_); - file_ = NULL; + file_ = nullptr; } } @@ -98,16 +98,16 @@ class RtpDumpWriter : public RtpFileWriter { RtpFileWriter* RtpFileWriter::Create(FileFormat format, const std::string& filename) { FILE* file = fopen(filename.c_str(), "wb"); - if (file == NULL) { + if (file == nullptr) { printf("ERROR: Can't open file: %s\n", filename.c_str()); - return NULL; + return nullptr; } switch (format) { case kRtpDump: return new RtpDumpWriter(file); } fclose(file); - return NULL; + return nullptr; } } // namespace test diff --git a/test/rtp_file_writer_unittest.cc b/test/rtp_file_writer_unittest.cc index 2396d7c346..5d51e46ef4 100644 --- a/test/rtp_file_writer_unittest.cc +++ b/test/rtp_file_writer_unittest.cc @@ -30,7 +30,7 @@ class RtpFileWriterTest : public ::testing::Test { } void WriteRtpPackets(int num_packets, int time_ms_offset = 0) { - ASSERT_TRUE(rtp_writer_.get() != NULL); + ASSERT_TRUE(rtp_writer_.get() != nullptr); test::RtpPacket packet; for (int i = 1; i <= num_packets; ++i) { packet.length = i; @@ -44,11 +44,11 @@ class RtpFileWriterTest : public ::testing::Test { void CloseOutputFile() { rtp_writer_.reset(); } void VerifyFileContents(int expected_packets) { - ASSERT_TRUE(rtp_writer_.get() == NULL) + ASSERT_TRUE(rtp_writer_.get() == nullptr) << "Must call CloseOutputFile before VerifyFileContents"; std::unique_ptr rtp_reader( test::RtpFileReader::Create(test::RtpFileReader::kRtpDump, filename_)); - ASSERT_TRUE(rtp_reader.get() != NULL); + ASSERT_TRUE(rtp_reader.get() != nullptr); test::RtpPacket packet; int i = 0; while (rtp_reader->NextPacket(&packet)) { diff --git a/test/testsupport/file_utils.cc b/test/testsupport/file_utils.cc index 3148863536..27bb59cbaa 100644 --- a/test/testsupport/file_utils.cc +++ b/test/testsupport/file_utils.cc @@ -277,7 +277,7 @@ std::string JoinFilename(absl::string_view dir, absl::string_view name) { size_t GetFileSize(absl::string_view filename) { FILE* f = fopen(std::string(filename).c_str(), "rb"); size_t size = 0; - if (f != NULL) { + if (f != nullptr) { if (fseek(f, 0, SEEK_END) == 0) { size = ftell(f); } diff --git a/test/testsupport/file_utils_unittest.cc b/test/testsupport/file_utils_unittest.cc index 104db59221..62745cb0fa 100644 --- a/test/testsupport/file_utils_unittest.cc +++ b/test/testsupport/file_utils_unittest.cc @@ -154,7 +154,7 @@ TEST_F(FileUtilsTest, GenerateTempFilename) { ASSERT_FALSE(test::FileExists(temp_filename)) << "File exists: " << temp_filename; FILE* file = fopen(temp_filename.c_str(), "wb"); - ASSERT_TRUE(file != NULL) << "Failed to open file: " << temp_filename; + ASSERT_TRUE(file != nullptr) << "Failed to open file: " << temp_filename; ASSERT_GT(fprintf(file, "%s", "Dummy data"), 0) << "Failed to write to file: " << temp_filename; fclose(file); @@ -216,7 +216,7 @@ TEST_F(FileUtilsTest, GetFileSizeExistingFile) { std::string temp_filename = test::TempFilename(test::OutputPath(), "fileutils_unittest"); FILE* file = fopen(temp_filename.c_str(), "wb"); - ASSERT_TRUE(file != NULL) << "Failed to open file: " << temp_filename; + ASSERT_TRUE(file != nullptr) << "Failed to open file: " << temp_filename; ASSERT_GT(fprintf(file, "%s", "Dummy data"), 0) << "Failed to write to file: " << temp_filename; fclose(file); diff --git a/test/testsupport/perf_test.cc b/test/testsupport/perf_test.cc index cec361bd8c..4c5e0f447e 100644 --- a/test/testsupport/perf_test.cc +++ b/test/testsupport/perf_test.cc @@ -243,7 +243,7 @@ bool WritePerfResults(const std::string& output_path) { std::string results = GetPerfResults(); CreateDir(DirName(output_path)); FILE* output = fopen(output_path.c_str(), "wb"); - if (output == NULL) { + if (output == nullptr) { printf("Failed to write to %s.\n", output_path.c_str()); return false; } diff --git a/test/testsupport/y4m_frame_generator.cc b/test/testsupport/y4m_frame_generator.cc index 1c5e0c8a58..f7b0eb0f88 100644 --- a/test/testsupport/y4m_frame_generator.cc +++ b/test/testsupport/y4m_frame_generator.cc @@ -36,7 +36,7 @@ Y4mFrameGenerator::Y4mFrameGenerator(absl::string_view filename, : filename_(filename), repeat_mode_(repeat_mode) { // Read resolution from the Y4M header. FILE* file = fopen(filename_.c_str(), "r"); - RTC_CHECK(file != NULL) << "Cannot open " << filename_; + RTC_CHECK(file != nullptr) << "Cannot open " << filename_; char header[kHeaderBytesToRead]; RTC_CHECK(fgets(header, sizeof(header), file) != nullptr) << "File " << filename_ << " is too small"; diff --git a/test/testsupport/y4m_frame_reader.cc b/test/testsupport/y4m_frame_reader.cc index 4501ddba41..6d9649d707 100644 --- a/test/testsupport/y4m_frame_reader.cc +++ b/test/testsupport/y4m_frame_reader.cc @@ -32,11 +32,11 @@ void ParseY4mHeader(std::string filepath, Resolution* resolution, int* header_size) { FILE* file = fopen(filepath.c_str(), "r"); - RTC_CHECK(file != NULL) << "Cannot open " << filepath; + RTC_CHECK(file != nullptr) << "Cannot open " << filepath; // Length of Y4M header is technically unlimited due to the comment tag 'X'. char h[1024]; - RTC_CHECK(fgets(h, sizeof(h), file) != NULL) + RTC_CHECK(fgets(h, sizeof(h), file) != nullptr) << "File " << filepath << " is too small"; fclose(file); diff --git a/test/testsupport/yuv_frame_reader.cc b/test/testsupport/yuv_frame_reader.cc index ea1a586e9f..4a7273520d 100644 --- a/test/testsupport/yuv_frame_reader.cc +++ b/test/testsupport/yuv_frame_reader.cc @@ -94,7 +94,7 @@ void YuvFrameReaderImpl::Init() { CalcBufferSize(VideoType::kI420, resolution_.width, resolution_.height); file_ = fopen(filepath_.c_str(), "rb"); - RTC_CHECK(file_ != NULL) << "Cannot open " << filepath_; + RTC_CHECK(file_ != nullptr) << "Cannot open " << filepath_; size_t file_size_bytes = GetFileSize(filepath_); RTC_CHECK_GT(file_size_bytes, 0u) << "File " << filepath_ << " is empty"; From 0c5a27bf2f7956a50b392be229b5be3dce4f010b Mon Sep 17 00:00:00 2001 From: Fredrik Solenberg Date: Thu, 15 May 2025 23:44:11 +0200 Subject: [PATCH 195/292] Move denormal_disabler to rtc_base/ Bug: webrtc:42228262 Change-Id: Iae81e9eb4499c7527293c6f23090cb86f8050c49 No-Iwyu: Avoid unrelated changes Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391461 Auto-Submit: Fredrik Solenberg Reviewed-by: Tomas Gunnarsson Commit-Queue: Tomas Gunnarsson Cr-Commit-Position: refs/heads/main@{#44667} --- modules/audio_processing/BUILD.gn | 4 ++-- modules/audio_processing/audio_processing_impl.cc | 2 +- rtc_base/BUILD.gn | 15 +++++++++++++++ .../source => rtc_base}/denormal_disabler.cc | 3 ++- .../include => rtc_base}/denormal_disabler.h | 8 +++----- .../denormal_disabler_unittest.cc | 3 +-- system_wrappers/BUILD.gn | 15 --------------- 7 files changed, 24 insertions(+), 26 deletions(-) rename {system_wrappers/source => rtc_base}/denormal_disabler.cc (98%) rename {system_wrappers/include => rtc_base}/denormal_disabler.h (90%) rename {system_wrappers/source => rtc_base}/denormal_disabler_unittest.cc (99%) diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn index 66979a7c99..2a7e72d5f1 100644 --- a/modules/audio_processing/BUILD.gn +++ b/modules/audio_processing/BUILD.gn @@ -170,6 +170,7 @@ rtc_library("audio_processing") { "../../common_audio:fir_filter_factory", "../../common_audio/third_party/ooura:fft_size_256", "../../rtc_base:checks", + "../../rtc_base:denormal_disabler", "../../rtc_base:event_tracer", "../../rtc_base:gtest_prod", "../../rtc_base:logging", @@ -181,7 +182,6 @@ rtc_library("audio_processing") { "../../rtc_base/synchronization:mutex", "../../rtc_base/system:rtc_export", "../../system_wrappers", - "../../system_wrappers:denormal_disabler", "../../system_wrappers:metrics", "aec3", "aec_dump", @@ -348,6 +348,7 @@ if (rtc_include_tests) { "../../common_audio", "../../common_audio:common_audio_c", "../../rtc_base:checks", + "../../rtc_base:denormal_disabler", "../../rtc_base:gtest_prod", "../../rtc_base:macromagic", "../../rtc_base:platform_thread", @@ -365,7 +366,6 @@ if (rtc_include_tests) { "../../rtc_base/system:arch", "../../rtc_base/system:file_wrapper", "../../system_wrappers", - "../../system_wrappers:denormal_disabler", "../../test:field_trial", "../../test:fileutils", "../../test:rtc_expect_death", diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc index 56a9149551..dfca465b63 100644 --- a/modules/audio_processing/audio_processing_impl.cc +++ b/modules/audio_processing/audio_processing_impl.cc @@ -33,10 +33,10 @@ #include "modules/audio_processing/include/audio_frame_view.h" #include "modules/audio_processing/logging/apm_data_dumper.h" #include "rtc_base/checks.h" +#include "rtc_base/denormal_disabler.h" #include "rtc_base/logging.h" #include "rtc_base/time_utils.h" #include "rtc_base/trace_event.h" -#include "system_wrappers/include/denormal_disabler.h" #include "system_wrappers/include/metrics.h" #define RETURN_ON_ERR(expr) \ diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn index 4af30800ed..bf217ee16a 100644 --- a/rtc_base/BUILD.gn +++ b/rtc_base/BUILD.gn @@ -182,6 +182,19 @@ rtc_library("copy_on_write_buffer") { ] } +rtc_library("denormal_disabler") { + visibility = [ "*" ] + public = [ "denormal_disabler.h" ] + sources = [ "denormal_disabler.cc" ] + deps = [ + "../rtc_base:checks", + "../rtc_base/system:arch", + ] + if (is_clang) { + cflags_cc = [ "-Wno-unused-private-field" ] + } +} + rtc_library("event_tracer") { visibility = [ "*" ] sources = [ @@ -2195,6 +2208,7 @@ if (rtc_include_tests) { "crc32_unittest.cc", "crypto_random_unittest.cc", "data_rate_limiter_unittest.cc", + "denormal_disabler_unittest.cc", "fake_clock_unittest.cc", "ip_address_unittest.cc", "memory_usage_unittest.cc", @@ -2220,6 +2234,7 @@ if (rtc_include_tests) { ":crc32", ":crypto_random", ":data_rate_limiter", + ":denormal_disabler", ":digest", ":gunit_helpers", ":ifaddrs_converter", diff --git a/system_wrappers/source/denormal_disabler.cc b/rtc_base/denormal_disabler.cc similarity index 98% rename from system_wrappers/source/denormal_disabler.cc rename to rtc_base/denormal_disabler.cc index bb9c05643c..6f2ebb2857 100644 --- a/system_wrappers/source/denormal_disabler.cc +++ b/rtc_base/denormal_disabler.cc @@ -8,9 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "system_wrappers/include/denormal_disabler.h" +#include "rtc_base/denormal_disabler.h" #include "rtc_base/checks.h" +#include "rtc_base/system/arch.h" namespace webrtc { namespace { diff --git a/system_wrappers/include/denormal_disabler.h b/rtc_base/denormal_disabler.h similarity index 90% rename from system_wrappers/include/denormal_disabler.h rename to rtc_base/denormal_disabler.h index bd3d401327..8c6363acf5 100644 --- a/system_wrappers/include/denormal_disabler.h +++ b/rtc_base/denormal_disabler.h @@ -8,10 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef SYSTEM_WRAPPERS_INCLUDE_DENORMAL_DISABLER_H_ -#define SYSTEM_WRAPPERS_INCLUDE_DENORMAL_DISABLER_H_ - -#include "rtc_base/system/arch.h" +#ifndef RTC_BASE_DENORMAL_DISABLER_H_ +#define RTC_BASE_DENORMAL_DISABLER_H_ namespace webrtc { @@ -53,4 +51,4 @@ class DenormalDisabler { } // namespace webrtc -#endif // SYSTEM_WRAPPERS_INCLUDE_DENORMAL_DISABLER_H_ +#endif // RTC_BASE_DENORMAL_DISABLER_H_ diff --git a/system_wrappers/source/denormal_disabler_unittest.cc b/rtc_base/denormal_disabler_unittest.cc similarity index 99% rename from system_wrappers/source/denormal_disabler_unittest.cc rename to rtc_base/denormal_disabler_unittest.cc index a2849f853f..5d5f249b7e 100644 --- a/system_wrappers/source/denormal_disabler_unittest.cc +++ b/rtc_base/denormal_disabler_unittest.cc @@ -8,11 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "system_wrappers/include/denormal_disabler.h" +#include "rtc_base/denormal_disabler.h" #include #include -#include #include "rtc_base/checks.h" #include "test/gtest.h" diff --git a/system_wrappers/BUILD.gn b/system_wrappers/BUILD.gn index 3b23716960..318bbaff90 100644 --- a/system_wrappers/BUILD.gn +++ b/system_wrappers/BUILD.gn @@ -111,25 +111,11 @@ rtc_library("metrics") { ] } -rtc_library("denormal_disabler") { - visibility = [ "*" ] - public = [ "include/denormal_disabler.h" ] - sources = [ "source/denormal_disabler.cc" ] - deps = [ - "../rtc_base:checks", - "../rtc_base/system:arch", - ] - if (is_clang) { - cflags_cc = [ "-Wno-unused-private-field" ] - } -} - if (rtc_include_tests && !build_with_chromium) { rtc_test("system_wrappers_unittests") { testonly = true sources = [ "source/clock_unittest.cc", - "source/denormal_disabler_unittest.cc", "source/field_trial_unittest.cc", "source/metrics_default_unittest.cc", "source/metrics_unittest.cc", @@ -137,7 +123,6 @@ if (rtc_include_tests && !build_with_chromium) { ] deps = [ - ":denormal_disabler", ":field_trial", ":metrics", ":system_wrappers", From bf70c05a1a2bd2204c6fee838ea1b6cc56a1b2c2 Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Fri, 16 May 2025 16:53:37 +0200 Subject: [PATCH 196/292] Cleanup usage of arraysize macro in api/ Bug: webrtc:407433458 Change-Id: Ica0448329fc3cc734cb6a9cbab128d6cca19f98a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392142 Reviewed-by: Harald Alvestrand Commit-Queue: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#44668} --- api/BUILD.gn | 3 --- api/DEPS | 1 - api/array_view.h | 2 +- api/audio/BUILD.gn | 2 -- api/audio/audio_processing.cc | 2 -- api/audio/audio_processing.h | 13 +++---------- api/audio/channel_layout.cc | 9 +++++---- api/transport/BUILD.gn | 1 - api/transport/stun_unittest.cc | 3 +-- api/video_codecs/BUILD.gn | 1 - api/video_codecs/h264_profile_level_id.cc | 3 +-- api/video_codecs/h265_profile_tier_level.cc | 3 +-- 12 files changed, 12 insertions(+), 31 deletions(-) diff --git a/api/BUILD.gn b/api/BUILD.gn index 6e32258531..d0e63949c6 100644 --- a/api/BUILD.gn +++ b/api/BUILD.gn @@ -430,7 +430,6 @@ rtc_library("rtc_error") { deps = [ "../rtc_base:checks", "../rtc_base:logging", - "../rtc_base:macromagic", "../rtc_base/system:rtc_export", "//third_party/abseil-cpp/absl/meta:type_traits", "//third_party/abseil-cpp/absl/strings", @@ -888,7 +887,6 @@ rtc_source_set("simulated_network_api") { visibility = [ "*" ] sources = [ "test/simulated_network.h" ] deps = [ - "../rtc_base:macromagic", "../rtc_base:random", "transport:ecn_marking", "units:data_rate", @@ -976,7 +974,6 @@ rtc_source_set("refcountedbase") { sources = [ "ref_counted_base.h" ] deps = [ ":ref_count", - "../rtc_base:macromagic", "../rtc_base:refcount", ] } diff --git a/api/DEPS b/api/DEPS index 0efc076c3e..7f168075d9 100644 --- a/api/DEPS +++ b/api/DEPS @@ -49,7 +49,6 @@ specific_include_rules = { ], ".*\.h": [ - "+rtc_base/arraysize.h", "+rtc_base/checks.h", "+rtc_base/system/rtc_export.h", "+rtc_base/system/rtc_export_template.h", diff --git a/api/array_view.h b/api/array_view.h index db85b47991..577d6cb46c 100644 --- a/api/array_view.h +++ b/api/array_view.h @@ -40,7 +40,7 @@ namespace webrtc { // std::vector, webrtc::Buffer, ...), but it's error-prone because the caller // has to correctly specify the array length: // -// Contains17(arr, arraysize(arr)); // C array +// Contains17(arr, std::size(arr)); // C array // Contains17(arr.data(), arr.size()); // std::vector // Contains17(arr, size); // pointer + size // ... diff --git a/api/audio/BUILD.gn b/api/audio/BUILD.gn index 09662a4efd..2e3c44a3ad 100644 --- a/api/audio/BUILD.gn +++ b/api/audio/BUILD.gn @@ -52,7 +52,6 @@ rtc_library("audio_frame_api") { "..:rtp_packet_info", "../../rtc_base:checks", "../../rtc_base:logging", - "../../rtc_base:macromagic", "../../rtc_base:timeutils", ] } @@ -88,7 +87,6 @@ rtc_library("audio_processing") { "..:ref_count", "..:scoped_refptr", "../../rtc_base:checks", - "../../rtc_base:macromagic", "../../rtc_base:stringutils", "../../rtc_base/system:arch", "../../rtc_base/system:file_wrapper", diff --git a/api/audio/audio_processing.cc b/api/audio/audio_processing.cc index 2941001c1d..76f0d71ddc 100644 --- a/api/audio/audio_processing.cc +++ b/api/audio/audio_processing.cc @@ -55,8 +55,6 @@ std::string GainController1ModeToString(const Agc1Config::Mode& mode) { } // namespace -constexpr int AudioProcessing::kNativeSampleRatesHz[]; - void CustomProcessing::SetRuntimeSetting( AudioProcessing::RuntimeSetting /* setting */) {} diff --git a/api/audio/audio_processing.h b/api/audio/audio_processing.h index 6a56874fe7..b7882133ff 100644 --- a/api/audio/audio_processing.h +++ b/api/audio/audio_processing.h @@ -36,7 +36,6 @@ #include "api/ref_count.h" #include "api/scoped_refptr.h" #include "api/task_queue/task_queue_base.h" -#include "rtc_base/arraysize.h" #include "rtc_base/checks.h" #include "rtc_base/system/rtc_export.h" @@ -691,22 +690,16 @@ class RTC_EXPORT AudioProcessing : public RefCountInterface { }; // Native rates supported by the integer interfaces. - enum NativeRate { + enum NativeRate : int { kSampleRate8kHz = 8000, kSampleRate16kHz = 16000, kSampleRate32kHz = 32000, kSampleRate48kHz = 48000 }; - // TODO(kwiberg): We currently need to support a compiler (Visual C++) that - // complains if we don't explicitly state the size of the array here. Remove - // the size when that's no longer the case. - static constexpr int kNativeSampleRatesHz[4] = { + static constexpr std::array kNativeSampleRatesHz = { kSampleRate8kHz, kSampleRate16kHz, kSampleRate32kHz, kSampleRate48kHz}; - static constexpr size_t kNumNativeSampleRates = - arraysize(kNativeSampleRatesHz); - static constexpr int kMaxNativeSampleRateHz = - kNativeSampleRatesHz[kNumNativeSampleRates - 1]; + static constexpr int kMaxNativeSampleRateHz = kNativeSampleRatesHz.back(); // APM processes audio in chunks of about 10 ms. See GetFrameSize() for // details. diff --git a/api/audio/channel_layout.cc b/api/audio/channel_layout.cc index e4ae356fab..596db90cfd 100644 --- a/api/audio/channel_layout.cc +++ b/api/audio/channel_layout.cc @@ -12,7 +12,8 @@ #include -#include "rtc_base/arraysize.h" +#include + #include "rtc_base/checks.h" #include "rtc_base/logging.h" @@ -170,7 +171,7 @@ static const int kChannelOrderings[CHANNEL_LAYOUT_MAX + 1][CHANNELS_MAX + 1] = { }; int ChannelLayoutToChannelCount(ChannelLayout layout) { - RTC_DCHECK_LT(static_cast(layout), arraysize(kLayoutToChannels)); + RTC_DCHECK_LT(static_cast(layout), std::size(kLayoutToChannels)); RTC_DCHECK_LE(kLayoutToChannels[layout], kMaxConcurrentChannels); return kLayoutToChannels[layout]; } @@ -201,8 +202,8 @@ ChannelLayout GuessChannelLayout(int channels) { } int ChannelOrder(ChannelLayout layout, Channels channel) { - RTC_DCHECK_LT(static_cast(layout), arraysize(kChannelOrderings)); - RTC_DCHECK_LT(static_cast(channel), arraysize(kChannelOrderings[0])); + RTC_DCHECK_LT(static_cast(layout), std::size(kChannelOrderings)); + RTC_DCHECK_LT(static_cast(channel), std::size(kChannelOrderings[0])); return kChannelOrderings[layout][channel]; } diff --git a/api/transport/BUILD.gn b/api/transport/BUILD.gn index 624788b855..11b03496be 100644 --- a/api/transport/BUILD.gn +++ b/api/transport/BUILD.gn @@ -175,7 +175,6 @@ if (rtc_include_tests) { "../../rtc_base:byte_buffer", "../../rtc_base:byte_order", "../../rtc_base:ip_address", - "../../rtc_base:macromagic", "../../rtc_base:socket_address", "../../system_wrappers:metrics", "../../test:test_support", diff --git a/api/transport/stun_unittest.cc b/api/transport/stun_unittest.cc index a49f1ecd66..0aeff066d4 100644 --- a/api/transport/stun_unittest.cc +++ b/api/transport/stun_unittest.cc @@ -18,7 +18,6 @@ #include #include "api/array_view.h" -#include "rtc_base/arraysize.h" #include "rtc_base/byte_buffer.h" #include "rtc_base/byte_order.h" #include "rtc_base/ip_address.h" @@ -566,7 +565,7 @@ TEST_F(StunTest, MessageTypes) { int types[] = {STUN_BINDING_REQUEST, STUN_BINDING_INDICATION, STUN_BINDING_RESPONSE, STUN_BINDING_ERROR_RESPONSE}; - for (size_t i = 0; i < arraysize(types); ++i) { + for (size_t i = 0; i < std::size(types); ++i) { EXPECT_EQ(i == 0U, IsStunRequestType(types[i])); EXPECT_EQ(i == 1U, IsStunIndicationType(types[i])); EXPECT_EQ(i == 2U, IsStunSuccessResponseType(types[i])); diff --git a/api/video_codecs/BUILD.gn b/api/video_codecs/BUILD.gn index 3732e6021f..802f8cc436 100644 --- a/api/video_codecs/BUILD.gn +++ b/api/video_codecs/BUILD.gn @@ -86,7 +86,6 @@ rtc_library("video_codecs_api") { "../../modules/video_coding:codec_globals_headers", "../../rtc_base:checks", "../../rtc_base:logging", - "../../rtc_base:macromagic", "../../rtc_base:refcount", "../../rtc_base:stringutils", "../../rtc_base/system:rtc_export", diff --git a/api/video_codecs/h264_profile_level_id.cc b/api/video_codecs/h264_profile_level_id.cc index 0bdaed7fd3..4dafda078f 100644 --- a/api/video_codecs/h264_profile_level_id.cc +++ b/api/video_codecs/h264_profile_level_id.cc @@ -18,7 +18,6 @@ #include #include "api/rtp_parameters.h" -#include "rtc_base/arraysize.h" namespace webrtc { @@ -166,7 +165,7 @@ std::optional H264SupportedLevel(int max_frame_pixel_count, float max_fps) { static const int kPixelsPerMacroblock = 16 * 16; - for (int i = arraysize(kLevelConstraints) - 1; i >= 0; --i) { + for (int i = std::ssize(kLevelConstraints) - 1; i >= 0; --i) { const LevelConstraint& level_constraint = kLevelConstraints[i]; if (level_constraint.max_macroblock_frame_size * kPixelsPerMacroblock <= max_frame_pixel_count && diff --git a/api/video_codecs/h265_profile_tier_level.cc b/api/video_codecs/h265_profile_tier_level.cc index f63321a293..306effa96a 100644 --- a/api/video_codecs/h265_profile_tier_level.cc +++ b/api/video_codecs/h265_profile_tier_level.cc @@ -15,7 +15,6 @@ #include "api/rtp_parameters.h" #include "api/video/resolution.h" -#include "rtc_base/arraysize.h" #include "rtc_base/string_to_number.h" namespace webrtc { @@ -310,7 +309,7 @@ std::optional GetSupportedH265Level(const Resolution& resolution, int aligned_height = (resolution.height + kMinCbSizeYMax - 1) & ~(kMinCbSizeYMax - 1); - for (int i = arraysize(kLevelConstraints) - 1; i >= 0; --i) { + for (int i = std::ssize(kLevelConstraints) - 1; i >= 0; --i) { const LevelConstraint& level_constraint = kLevelConstraints[i]; if (level_constraint.max_luma_picture_size <= aligned_width * aligned_height && From 285d5bf66b3ca780092aaa32e3787c0ad38a076e Mon Sep 17 00:00:00 2001 From: Tommi Date: Tue, 13 May 2025 21:31:48 +0200 Subject: [PATCH 197/292] Make Resample10Msec with views return void and update PushResampler. MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Since we're using views, there's liberal checking that the dimensions are correct when calling methods such as CopySamples(). This allows us to remove error checks when calling Resample methods. Bug: chromium:335805780 Change-Id: Ib7af879d3e7e7a84e4f10148ccc264b95134ef19 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390922 Reviewed-by: Per Åhgren Commit-Queue: Tomas Gunnarsson Cr-Commit-Position: refs/heads/main@{#44669} --- audio/audio_transport_impl.cc | 36 +++++++++++-------- audio/remix_resample.cc | 12 +++---- .../resampler/include/push_resampler.h | 11 +++--- common_audio/resampler/push_resampler.cc | 16 ++++----- modules/audio_coding/acm2/acm_resampler.cc | 30 +++++----------- modules/audio_coding/acm2/acm_resampler.h | 8 ++--- .../audio_coding/acm2/audio_coding_module.cc | 33 +++-------------- modules/audio_coding/test/opus_test.cc | 5 ++- modules/audio_processing/BUILD.gn | 1 + .../audio_processing_unittest.cc | 27 +++++++++----- 10 files changed, 76 insertions(+), 103 deletions(-) diff --git a/audio/audio_transport_impl.cc b/audio/audio_transport_impl.cc index ef8d3baf8d..cd83fd9e3b 100644 --- a/audio/audio_transport_impl.cc +++ b/audio/audio_transport_impl.cc @@ -11,15 +11,24 @@ #include "audio/audio_transport_impl.h" #include +#include +#include #include +#include #include +#include +#include "api/audio/audio_frame.h" +#include "api/audio/audio_mixer.h" +#include "api/audio/audio_view.h" #include "audio/remix_resample.h" #include "audio/utility/audio_frame_operations.h" #include "call/audio_sender.h" +#include "common_audio/resampler/include/push_resampler.h" #include "modules/async_audio_processing/async_audio_processing.h" #include "modules/audio_processing/include/audio_frame_proxies.h" #include "rtc_base/checks.h" +#include "rtc_base/synchronization/mutex.h" #include "rtc_base/trace_event.h" namespace webrtc { @@ -67,10 +76,10 @@ void ProcessCaptureFrame(uint32_t delay_ms, // Resample audio in `frame` to given sample rate preserving the // channel count and place the result in `destination`. -int Resample(const AudioFrame& frame, - const int destination_sample_rate, - PushResampler* resampler, - InterleavedView destination) { +void Resample(const AudioFrame& frame, + const int destination_sample_rate, + PushResampler* resampler, + InterleavedView destination) { TRACE_EVENT2("webrtc", "Resample", "frame sample rate", frame.sample_rate_hz_, "destination_sample_rate", destination_sample_rate); const size_t target_number_of_samples_per_channel = @@ -82,7 +91,7 @@ int Resample(const AudioFrame& frame, frame.num_channels_ * target_number_of_samples_per_channel); // TODO(yujo): Add special case handling of muted frames. - return resampler->Resample(frame.data_view(), destination); + resampler->Resample(frame.data_view(), destination); } } // namespace @@ -233,11 +242,10 @@ int32_t AudioTransportImpl::NeedMorePlayData(const size_t nSamples, RTC_DCHECK_EQ(error, AudioProcessing::kNoError); } - nSamplesOut = - Resample(mixed_frame_, samplesPerSec, &render_resampler_, - InterleavedView(static_cast(audioSamples), - nSamples, nChannels)); - RTC_DCHECK_EQ(nSamplesOut, nChannels * nSamples); + InterleavedView resampled(static_cast(audioSamples), + nSamples, nChannels); + Resample(mixed_frame_, samplesPerSec, &render_resampler_, resampled); + nSamplesOut = resampled.size(); return 0; } @@ -266,11 +274,9 @@ void AudioTransportImpl::PullRenderData(int bits_per_sample, *elapsed_time_ms = mixed_frame_.elapsed_time_ms_; *ntp_time_ms = mixed_frame_.ntp_time_ms_; - int output_samples = - Resample(mixed_frame_, sample_rate, &render_resampler_, - InterleavedView(static_cast(audio_data), - number_of_frames, number_of_channels)); - RTC_DCHECK_EQ(output_samples, number_of_channels * number_of_frames); + Resample(mixed_frame_, sample_rate, &render_resampler_, + InterleavedView(static_cast(audio_data), + number_of_frames, number_of_channels)); } void AudioTransportImpl::UpdateAudioSenders(std::vector senders, diff --git a/audio/remix_resample.cc b/audio/remix_resample.cc index 93e86517ae..0f87803dc0 100644 --- a/audio/remix_resample.cc +++ b/audio/remix_resample.cc @@ -11,12 +11,13 @@ #include "audio/remix_resample.h" #include +#include #include "api/audio/audio_frame.h" +#include "api/audio/audio_view.h" #include "audio/utility/audio_frame_operations.h" #include "common_audio/resampler/include/push_resampler.h" #include "rtc_base/checks.h" -#include "rtc_base/logging.h" namespace webrtc { namespace voe { @@ -70,12 +71,9 @@ void RemixAndResample(InterleavedView src_data, // `dst_frame` as a target buffer with the same number of channels as the // source. auto original_dst_number_of_channels = dst_frame->num_channels_; - int out_length = resampler->Resample( - src_data, dst_frame->mutable_data(dst_frame->samples_per_channel_, - src_data.num_channels())); - RTC_CHECK_NE(out_length, -1) << "src_data.size=" << src_data.size(); - RTC_DCHECK_EQ(dst_frame->samples_per_channel(), - out_length / src_data.num_channels()); + resampler->Resample(src_data, + dst_frame->mutable_data(dst_frame->samples_per_channel_, + src_data.num_channels())); // Upmix after resampling. if (src_data.num_channels() == 1 && original_dst_number_of_channels == 2) { diff --git a/common_audio/resampler/include/push_resampler.h b/common_audio/resampler/include/push_resampler.h index 20d40423d1..fe33d56c35 100644 --- a/common_audio/resampler/include/push_resampler.h +++ b/common_audio/resampler/include/push_resampler.h @@ -32,13 +32,10 @@ class PushResampler final { size_t num_channels); ~PushResampler(); - // Returns the total number of samples provided in destination (e.g. 32 kHz, - // 2 channel audio gives 640 samples) - never a negative value. - int Resample(InterleavedView src, InterleavedView dst); - // For when a deinterleaved/mono channel already exists and we can skip the - // deinterleaved operation. The return value is the number of resampled - // samples. - int Resample(MonoView src, MonoView dst); + void Resample(InterleavedView src, InterleavedView dst); + // For when a deinterleaved (or mono) channel already exists and we can skip + // the deinterleaved operation. + void Resample(MonoView src, MonoView dst); private: // Ensures that source and destination buffers for deinterleaving are diff --git a/common_audio/resampler/push_resampler.cc b/common_audio/resampler/push_resampler.cc index 2e75679c82..d35b9af605 100644 --- a/common_audio/resampler/push_resampler.cc +++ b/common_audio/resampler/push_resampler.cc @@ -15,7 +15,7 @@ #include -#include "api/audio/audio_frame.h" +#include "api/audio/audio_view.h" #include "common_audio/include/audio_util.h" #include "common_audio/resampler/push_sinc_resampler.h" #include "rtc_base/checks.h" @@ -77,8 +77,8 @@ void PushResampler::EnsureInitialized(size_t src_samples_per_channel, } template -int PushResampler::Resample(InterleavedView src, - InterleavedView dst) { +void PushResampler::Resample(InterleavedView src, + InterleavedView dst) { EnsureInitialized(SamplesPerChannel(src), SamplesPerChannel(dst), NumChannels(src)); @@ -91,7 +91,7 @@ int PushResampler::Resample(InterleavedView src, // The old resampler provides this memcpy facility in the case of matching // sample rates, so reproduce it here for the sinc resampler. CopySamples(dst, src); - return static_cast(src.data().size()); + return; } Deinterleave(src, source_view_); @@ -103,21 +103,19 @@ int PushResampler::Resample(InterleavedView src, } Interleave(destination_view_, dst); - return static_cast(dst.size()); } template -int PushResampler::Resample(MonoView src, MonoView dst) { +void PushResampler::Resample(MonoView src, MonoView dst) { RTC_DCHECK_EQ(resamplers_.size(), 1); RTC_DCHECK_EQ(SamplesPerChannel(src), SamplesPerChannel(source_view_)); RTC_DCHECK_EQ(SamplesPerChannel(dst), SamplesPerChannel(destination_view_)); if (SamplesPerChannel(src) == SamplesPerChannel(dst)) { CopySamples(dst, src); - return static_cast(src.size()); + } else { + resamplers_[0]->Resample(src, dst); } - - return resamplers_[0]->Resample(src, dst); } // Explictly generate required instantiations. diff --git a/modules/audio_coding/acm2/acm_resampler.cc b/modules/audio_coding/acm2/acm_resampler.cc index 346ec68377..1587eccfbb 100644 --- a/modules/audio_coding/acm2/acm_resampler.cc +++ b/modules/audio_coding/acm2/acm_resampler.cc @@ -39,34 +39,20 @@ int ACMResampler::Resample10Msec(const int16_t* in_audio, SampleRateToDefaultChannelSize(out_freq_hz), num_audio_channels); RTC_DCHECK_GE(out_capacity_samples, dst.size()); - return Resample10Msec(src, in_freq_hz, dst, out_freq_hz); + Resample10Msec(src, in_freq_hz, dst, out_freq_hz); + return dst.samples_per_channel(); } -int ACMResampler::Resample10Msec(InterleavedView src, - int in_freq_hz, - InterleavedView dst, - int out_freq_hz) { +void ACMResampler::Resample10Msec(InterleavedView src, + int in_freq_hz, + InterleavedView dst, + int out_freq_hz) { RTC_DCHECK_EQ(src.num_channels(), dst.num_channels()); if (in_freq_hz == out_freq_hz) { - // TODO(tommi): Could this `if()` be replaced with a stricter check? - // RTC_CHECK_EQ(dst.size(), src.size()); - // Eventually change the method to never return -1 and not have to - // expect that at higher level call sites. The Resample() method below - // can never return -1 - if (dst.size() < src.size()) { - RTC_DCHECK_NOTREACHED(); - return -1; - } CopySamples(dst, src); - RTC_DCHECK_EQ(dst.samples_per_channel(), src.samples_per_channel()); - return static_cast(dst.samples_per_channel()); + } else { + resampler_.Resample(src, dst); } - - int out_length = resampler_.Resample(src, dst); - RTC_CHECK_GE(out_length, 0); - RTC_DCHECK_EQ(out_length, dst.size()); - RTC_DCHECK_EQ(out_length / dst.num_channels(), dst.samples_per_channel()); - return static_cast(dst.samples_per_channel()); } ResamplerHelper::ResamplerHelper() { diff --git a/modules/audio_coding/acm2/acm_resampler.h b/modules/audio_coding/acm2/acm_resampler.h index abf08f063c..256884bfbf 100644 --- a/modules/audio_coding/acm2/acm_resampler.h +++ b/modules/audio_coding/acm2/acm_resampler.h @@ -36,10 +36,10 @@ class ACMResampler { size_t out_capacity_samples, int16_t* out_audio); - int Resample10Msec(InterleavedView src, - int src_freq_hz, - InterleavedView dst, - int dst_freq_hz); + void Resample10Msec(InterleavedView src, + int src_freq_hz, + InterleavedView dst, + int dst_freq_hz); private: PushResampler resampler_; diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc index 373e3a34c1..32a36a7e6b 100644 --- a/modules/audio_coding/acm2/audio_coding_module.cc +++ b/modules/audio_coding/acm2/audio_coding_module.cc @@ -153,12 +153,6 @@ class AudioCodingModuleImpl final : public AudioCodingModule { const AudioFrame* AddDataNoPreProcess(const AudioFrame& in_frame) RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_); - // Resamples `source` into the `preprocess_frame_` buffer. - // `sample_rate` refers to the sample rate of `source`. - bool ResampleToPreProcessFrame(InterleavedView source, - int sample_rate) - RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_); - // Change required states after starting to receive the codec corresponding // to `index`. int UpdateUponReceivingCodec(int index); @@ -512,10 +506,11 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame, preprocess_frame_.SetSampleRateAndChannelSize(encoder_stack_->SampleRateHz()); if (resample) { - if (!ResampleToPreProcessFrame(resample_src_audio, - in_frame.sample_rate_hz())) { - return -1; - } + resampler_.Resample10Msec( + resample_src_audio, in_frame.sample_rate_hz(), + preprocess_frame_.mutable_data(preprocess_frame_.samples_per_channel(), + preprocess_frame_.num_channels()), + preprocess_frame_.sample_rate_hz()); } expected_codec_ts_ += @@ -546,24 +541,6 @@ const AudioFrame* AudioCodingModuleImpl::AddDataNoPreProcess( return ret; } -bool AudioCodingModuleImpl::ResampleToPreProcessFrame( - InterleavedView source, - int sample_rate) { - int samples_per_channel = resampler_.Resample10Msec( - source, sample_rate, - preprocess_frame_.mutable_data(preprocess_frame_.samples_per_channel(), - preprocess_frame_.num_channels()), - preprocess_frame_.sample_rate_hz()); - - // TODO(tommi): Resample10Msec() should never return an error. Remove this - // check and change this method to return void. - if (samples_per_channel < 0) { - RTC_LOG(LS_ERROR) << "Cannot add 10 ms audio, resampling failed"; - } - - return samples_per_channel >= 0; -} - ///////////////////////////////////////// // (FEC) Forward Error Correction (codec internal) // diff --git a/modules/audio_coding/test/opus_test.cc b/modules/audio_coding/test/opus_test.cc index 1f31e3c2a9..da44f26e22 100644 --- a/modules/audio_coding/test/opus_test.cc +++ b/modules/audio_coding/test/opus_test.cc @@ -267,9 +267,8 @@ void OpusTest::Run(TestPackStereo* channel, // If input audio is sampled at 32 kHz, resampling to 48 kHz is required. InterleavedView dst(&audio[written_samples], 480, channels); - EXPECT_EQ(480, resampler_.Resample10Msec(audio_frame.data_view(), - audio_frame.sample_rate_hz_, dst, - 48000)); + resampler_.Resample10Msec(audio_frame.data_view(), + audio_frame.sample_rate_hz_, dst, 48000); written_samples += 480 * channels; // Sometimes we need to loop over the audio vector to produce the right diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn index 2a7e72d5f1..b42ded6f23 100644 --- a/modules/audio_processing/BUILD.gn +++ b/modules/audio_processing/BUILD.gn @@ -341,6 +341,7 @@ if (rtc_include_tests) { "../../api/audio:aec3_factory", "../../api/audio:audio_frame_api", "../../api/audio:audio_processing", + "../../api/audio:audio_processing_statistics", "../../api/audio:builtin_audio_processing_builder", "../../api/audio:echo_detector_creator", "../../api/environment", diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc index 373b574d7a..0961036ec4 100644 --- a/modules/audio_processing/audio_processing_unittest.cc +++ b/modules/audio_processing/audio_processing_unittest.cc @@ -13,41 +13,54 @@ #include #include +#include #include +#include +#include +#include #include +#include #include #include +#include #include #include +#include +#include +#include #include "absl/flags/flag.h" #include "absl/strings/string_view.h" +#include "api/array_view.h" +#include "api/audio/audio_processing_statistics.h" +#include "api/audio/audio_view.h" #include "api/audio/builtin_audio_processing_builder.h" +#include "api/audio/echo_control.h" #include "api/audio/echo_detector_creator.h" +#include "api/environment/environment.h" #include "api/environment/environment_factory.h" #include "api/make_ref_counted.h" +#include "api/scoped_refptr.h" +#include "common_audio/channel_buffer.h" #include "common_audio/include/audio_util.h" #include "common_audio/resampler/include/push_resampler.h" #include "common_audio/resampler/push_sinc_resampler.h" -#include "common_audio/signal_processing/include/signal_processing_library.h" #include "modules/audio_processing/aec_dump/aec_dump_factory.h" -#include "modules/audio_processing/audio_processing_impl.h" #include "modules/audio_processing/include/mock_audio_processing.h" #include "modules/audio_processing/test/protobuf_utils.h" #include "modules/audio_processing/test/test_utils.h" #include "rtc_base/arraysize.h" #include "rtc_base/checks.h" #include "rtc_base/fake_clock.h" -#include "rtc_base/gtest_prod_util.h" #include "rtc_base/numerics/safe_conversions.h" #include "rtc_base/numerics/safe_minmax.h" #include "rtc_base/protobuf_utils.h" #include "rtc_base/strings/string_builder.h" #include "rtc_base/swap_queue.h" -#include "rtc_base/system/arch.h" +#include "rtc_base/system/file_wrapper.h" #include "rtc_base/task_queue_for_test.h" -#include "rtc_base/thread.h" #include "system_wrappers/include/cpu_features_wrapper.h" +#include "test/gmock.h" #include "test/gtest.h" #include "test/testsupport/file_utils.h" @@ -1669,7 +1682,6 @@ TEST_F(ApmTest, DebugDumpFromFileHandle) { // of enabled components. TEST_F(ApmTest, Process) { - GOOGLE_PROTOBUF_VERIFY_VERSION; audioproc::OutputData ref_data; if (!absl::GetFlag(FLAGS_write_apm_ref_data)) { @@ -2166,8 +2178,7 @@ TEST_P(AudioProcessingTest, Formats) { out_num); InterleavedView dst(cmp_data.get(), ref_samples_per_channel, out_num); - ASSERT_EQ(ref_length, - static_cast(resampler.Resample(src, dst))); + resampler.Resample(src, dst); out_ptr = cmp_data.get(); } From 132a436aa128f6154c2e20988a575850614df2cf Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Fri, 16 May 2025 11:03:19 -0700 Subject: [PATCH 198/292] Roll chromium_revision 07a04e1af2..fd85fb08c2 (1461321:1461479) Change log: https://chromium.googlesource.com/chromium/src/+log/07a04e1af2..fd85fb08c2 Full diff: https://chromium.googlesource.com/chromium/src/+/07a04e1af2..fd85fb08c2 Changed dependencies * src/build: https://chromium.googlesource.com/chromium/src/build/+log/34cf410968..f2a9b5fb04 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/7d88136926..9ee52f4901 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/c5062b897d..978e19de47 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/f784079038..f8dbce93f1 * src/third_party/fontconfig/src: https://chromium.googlesource.com/external/fontconfig.git/+log/14d466b30a..24ed9b4d73 * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/fbce37dd3e..9c99dc2f9c * src/third_party/libc++abi/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git/+log/069652e9cb..9810fb23f6 * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/99c9ef8102..db280603c2 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/73d3d2d400..35f247aa02 DEPS diff: https://chromium.googlesource.com/chromium/src/+/07a04e1af2..fd85fb08c2/DEPS BUG=None Change-Id: If3d09812f22991cfe4be17454d61df13a6aa9746 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392220 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44670} --- DEPS | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/DEPS b/DEPS index ebccc019a4..94b60d0123 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '07a04e1af2558471e7de506d6e2d94d7572ef966', + 'chromium_revision': 'fd85fb08c2761a7d1ed8cc4a033fcbd74ed4da82', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -75,7 +75,7 @@ deps = { 'src/base': 'https://chromium.googlesource.com/chromium/src/base@25ef6cb7fae4eb08709090486e741f403e64c4e8', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@34cf410968a36b915e3ffda3007ff9c291aac27a', + 'https://chromium.googlesource.com/chromium/src/build@f2a9b5fb04dcea4060c73f488e2820d89c045554', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@e9b36e9ffb02b268d194c3f82f89952d09d0c714', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@7d881369269f03a0adbd93ba24a715b30bd3eae8', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@9ee52f4901b5bb09ce0735b4812ff80bd87540ec', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@c5062b897df027d714bb5ede51ec8bc764b7a0da', + 'https://chromium.googlesource.com/chromium/src/testing@978e19de472164a161976a8b44c6daba6f9ef935', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@f784079038e18926b1dfb532a606b7fb75a7a20b', + 'https://chromium.googlesource.com/chromium/src/third_party@f8dbce93f1672dd92410b40ab22c9b074e9bf4cf', 'src/buildtools/linux64': { 'packages': [ @@ -327,7 +327,7 @@ deps = { 'src/third_party/libc++/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@a01c02c9d4acbdae3b7e8a2f3ee58579a9c29f96', 'src/third_party/libc++abi/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@069652e9cbca8e3b5e2371057a206f91f2f710f6', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@9810fb23f6ba666f017c2b67c67de2bcac2b44bd', 'src/third_party/llvm-libc/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@9aefa1cede964571c0b0ef41460cc5c3bbc6b4b8', 'src/third_party/libunwind/src': @@ -503,11 +503,11 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@fbce37dd3e925eef938e9d9ee24345933a07541f', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@9c99dc2f9ca03c45c5fa4f48867ca55dcd3e8840', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { - 'url': 'https://chromium.googlesource.com/external/fontconfig.git@14d466b30a8ab4a9d789977ed94f2c30e7209267', + 'url': 'https://chromium.googlesource.com/external/fontconfig.git@24ed9b4d73dc068ed2baeb0da583c51879d5cabc', 'condition': 'checkout_linux', }, 'src/third_party/freetype/src': @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '99c9ef8102d3770e6d5ba91d9ca400b751b0af9b', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'db280603c217b4abb469b09f543f0713f906da5c', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@73d3d2d4007a1d9255da7d5dde863a6d9c666ee1', + 'https://chromium.googlesource.com/chromium/src/tools@35f247aa02ac44afc85e79b75e419aab9c80da42', 'src/third_party/espresso': { 'packages': [ From 20594492895dc66464506cbd1a349423dce6bf13 Mon Sep 17 00:00:00 2001 From: webrtc-version-updater Date: Fri, 16 May 2025 21:03:42 -0700 Subject: [PATCH 199/292] Update WebRTC code version (2025-05-17T04:03:23). Bug: None Change-Id: Iddf2013019b88bf9cf760df9a004f47bc0eb710e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392183 Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Cr-Commit-Position: refs/heads/main@{#44671} --- call/version.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/call/version.cc b/call/version.cc index 3758d817ce..99522962d7 100644 --- a/call/version.cc +++ b/call/version.cc @@ -13,7 +13,7 @@ namespace webrtc { // The timestamp is always in UTC. -const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-16T04:10:38"; +const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-17T04:03:23"; void LoadWebRTCVersionInRegister() { // Using volatile to instruct the compiler to not optimize `p` away even From 96e7cd275de82af837ff4c13502d52d1bf34f86d Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Sun, 18 May 2025 03:05:11 -0700 Subject: [PATCH 200/292] Roll chromium_revision fd85fb08c2..6e49907b7a (1461479:1461861) Change log: https://chromium.googlesource.com/chromium/src/+log/fd85fb08c2..6e49907b7a Full diff: https://chromium.googlesource.com/chromium/src/+/fd85fb08c2..6e49907b7a Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/25ef6cb7fa..3ca2b4045a * src/build: https://chromium.googlesource.com/chromium/src/build/+log/f2a9b5fb04..a2bd4bf231 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/9ee52f4901..9ec1fa7715 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/978e19de47..f9e21d5658 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/f8dbce93f1..bd63626edb * src/third_party/android_deps/autorolled/cipd: PY5tSZDTL7W8p6Yi4BuTUezpSGjWKlBrmJKhcuwdcx0C..5jhxywV7wAc6P8KUwq9kM4caswnIJ4Y0n8PEaEAIpNMC * src/third_party/androidx/cipd: TIfHUjgo7_UN847vl82ZF4Pku0QVun5z2gYKRVxOL4EC..dKxftVsUeWTHR64K1ywUo9zWElvcUVqJnGPnDVQbVGcC * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/99eba5614c..a934ee9e1f * src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/647003beca..1d27130259 * src/third_party/fontconfig/src: https://chromium.googlesource.com/external/fontconfig.git/+log/24ed9b4d73..14d466b30a * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/9c99dc2f9c..0c155e4beb * src/third_party/kotlin_stdlib/cipd: eYKSu9agsqusPORDTTSPHcg632V_j9kIvvufVJiULXIC..8JF4DFDjaasGcuqxNRqw0DePRizH5yR1-Al191P4K9wC * src/third_party/kotlinc/current: DMzAXJeHsmdOETLSEAYTfnPUX4VvhvmX2__q71V_Ia8C..XmaM7JA4hB75AuMdzCegF-XYzXtoHKOA1anrWqAJL3QC * src/third_party/llvm-libc/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git/+log/9aefa1cede..9c3ae3120f * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/db280603c2..5acde97bf6 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/35f247aa02..2187342a4c DEPS diff: https://chromium.googlesource.com/chromium/src/+/fd85fb08c2..6e49907b7a/DEPS BUG=None Change-Id: I627be75e74fbfbaf64c1a12b00c5f58b148cb266 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392196 Commit-Queue: Autoroller Bot-Commit: Autoroller Cr-Commit-Position: refs/heads/main@{#44672} --- DEPS | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) diff --git a/DEPS b/DEPS index 94b60d0123..8384405ed6 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': 'fd85fb08c2761a7d1ed8cc4a033fcbd74ed4da82', + 'chromium_revision': '6e49907b7a85360123f20af8910539f0a91d97d9', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@25ef6cb7fae4eb08709090486e741f403e64c4e8', + 'https://chromium.googlesource.com/chromium/src/base@3ca2b4045a996236ad96e5513aaf6bd07f20fe28', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@f2a9b5fb04dcea4060c73f488e2820d89c045554', + 'https://chromium.googlesource.com/chromium/src/build@a2bd4bf2317610bf5d7f113828f94b16dca4f669', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@e9b36e9ffb02b268d194c3f82f89952d09d0c714', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@9ee52f4901b5bb09ce0735b4812ff80bd87540ec', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@9ec1fa771571a7eae57cc2f765d9950272cd943f', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@978e19de472164a161976a8b44c6daba6f9ef935', + 'https://chromium.googlesource.com/chromium/src/testing@f9e21d5658bc669e61f53ca2f5bfa6045bc8defe', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@f8dbce93f1672dd92410b40ab22c9b074e9bf4cf', + 'https://chromium.googlesource.com/chromium/src/third_party@bd63626edb264c364b67e580191110da71641a30', 'src/buildtools/linux64': { 'packages': [ @@ -323,13 +323,13 @@ deps = { 'src/third_party/clang-format/script': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@37f6e68a107df43b7d7e044fd36a13cbae3413f2', 'src/third_party/compiler-rt/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@647003becabf611bc63c0e959e84060376d371be', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@1d27130259209a2b5c8ac992b832fc5522f38a8d', 'src/third_party/libc++/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@a01c02c9d4acbdae3b7e8a2f3ee58579a9c29f96', 'src/third_party/libc++abi/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@9810fb23f6ba666f017c2b67c67de2bcac2b44bd', 'src/third_party/llvm-libc/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@9aefa1cede964571c0b0ef41460cc5c3bbc6b4b8', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@9c3ae3120fe83b998d0498dcc9ad3a56c29fad0c', 'src/third_party/libunwind/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@8575f4ae4fcf8892938bd9766cf1a5c90a0ed04e', @@ -480,7 +480,7 @@ deps = { }, 'src/third_party/boringssl/src': - 'https://boringssl.googlesource.com/boringssl.git@99eba5614c519092e3d797a95dade2ac2660d5d9', + 'https://boringssl.googlesource.com/boringssl.git@a934ee9e1fe4397e682f9f18b1f4f061d7400f9d', 'src/third_party/breakpad/breakpad': 'https://chromium.googlesource.com/breakpad/breakpad.git@fad82477312f786994151f01c352813fabf924c0', 'src/third_party/catapult': @@ -503,11 +503,11 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@9c99dc2f9ca03c45c5fa4f48867ca55dcd3e8840', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@0c155e4beb9c1706fdafce2326b618a5347c4510', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { - 'url': 'https://chromium.googlesource.com/external/fontconfig.git@24ed9b4d73dc068ed2baeb0da583c51879d5cabc', + 'url': 'https://chromium.googlesource.com/external/fontconfig.git@14d466b30a8ab4a9d789977ed94f2c30e7209267', 'condition': 'checkout_linux', }, 'src/third_party/freetype/src': @@ -563,7 +563,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/kotlin_stdlib', - 'version': 'eYKSu9agsqusPORDTTSPHcg632V_j9kIvvufVJiULXIC', + 'version': '8JF4DFDjaasGcuqxNRqw0DePRizH5yR1-Al191P4K9wC', }, ], 'condition': 'checkout_android', @@ -574,7 +574,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/kotlinc', - 'version': 'DMzAXJeHsmdOETLSEAYTfnPUX4VvhvmX2__q71V_Ia8C', + 'version': 'XmaM7JA4hB75AuMdzCegF-XYzXtoHKOA1anrWqAJL3QC', }, ], 'condition': 'checkout_android', @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'db280603c217b4abb469b09f543f0713f906da5c', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '5acde97bf62e479089bd6466a8d84cd80846f502', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@35f247aa02ac44afc85e79b75e419aab9c80da42', + 'https://chromium.googlesource.com/chromium/src/tools@2187342a4c9b1ddc5ab4b4df5784e9f5d76a6e2d', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'TIfHUjgo7_UN847vl82ZF4Pku0QVun5z2gYKRVxOL4EC', + 'version': 'dKxftVsUeWTHR64K1ywUo9zWElvcUVqJnGPnDVQbVGcC', }, ], 'condition': 'checkout_android and non_git_source', @@ -827,7 +827,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_deps/autorolled', - 'version': 'PY5tSZDTL7W8p6Yi4BuTUezpSGjWKlBrmJKhcuwdcx0C', + 'version': '5jhxywV7wAc6P8KUwq9kM4caswnIJ4Y0n8PEaEAIpNMC', }, ], 'condition': 'checkout_android and non_git_source', From 658d2f57f201457e9f8fb7c76ab7ce25505b580a Mon Sep 17 00:00:00 2001 From: Jeremy Leconte Date: Fri, 16 May 2025 13:58:43 +0200 Subject: [PATCH 201/292] [Doc] Fix broken buganizer template links. Change-Id: Ie629097bd78864afe97d1fc71845a35fb82cd9e9 Bug: None Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392140 Commit-Queue: Jeremy Leconte Reviewed-by: Artem Titov Cr-Commit-Position: refs/heads/main@{#44673} --- g3doc/become_a_committer.md | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/g3doc/become_a_committer.md b/g3doc/become_a_committer.md index fcd1849b6e..725b9faa7a 100644 --- a/g3doc/become_a_committer.md +++ b/g3doc/become_a_committer.md @@ -88,5 +88,5 @@ demonstrate activity and maintain their WebRTC Committer status. [3]: https://webrtc.googlesource.com/src/+/refs/heads/main/g3doc/style-guide.md [4]: https://developers.google.com/open-source/cla/individual?hl=en [5]: https://google.github.io/styleguide/cppguide.html -[6]: https://bugs.chromium.org/p/webrtc/issues/entry?template=Become+WebRTC+committer -[7]: https://bugs.chromium.org/p/webrtc/issues/entry?template=Get+tryjob+access +[6]: https://issues.webrtc.org/issues/new?component=1565136&template=1986141 +[7]: https://issues.webrtc.org/issues/new?component=1565136&template=1986372 From 21edb77f3afd9e2e70338c616bbe19b146cfb771 Mon Sep 17 00:00:00 2001 From: webrtc-version-updater Date: Sun, 18 May 2025 21:06:38 -0700 Subject: [PATCH 202/292] Update WebRTC code version (2025-05-19T04:06:36). Bug: None Change-Id: Id80a38b09dea9257eb93b282d261e372070067d7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392405 Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Cr-Commit-Position: refs/heads/main@{#44674} --- call/version.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/call/version.cc b/call/version.cc index 99522962d7..e8cec37240 100644 --- a/call/version.cc +++ b/call/version.cc @@ -13,7 +13,7 @@ namespace webrtc { // The timestamp is always in UTC. -const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-17T04:03:23"; +const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-19T04:06:36"; void LoadWebRTCVersionInRegister() { // Using volatile to instruct the compiler to not optimize `p` away even From 3bf535164c7c13ad6fc5b5dab313dc6acc7c2552 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Mon, 19 May 2025 01:02:49 -0700 Subject: [PATCH 203/292] Roll chromium_revision 6e49907b7a..eaafcde8f7 (1461861:1461988) Change log: https://chromium.googlesource.com/chromium/src/+log/6e49907b7a..eaafcde8f7 Full diff: https://chromium.googlesource.com/chromium/src/+/6e49907b7a..eaafcde8f7 Changed dependencies * siso_version: git_revision:30ea210712239b874febcadf661f118fc2c0f6d7..git_revision:3753a55d0911b4ea3dd000263088f60edb8d947b * src/base: https://chromium.googlesource.com/chromium/src/base/+log/3ca2b4045a..70882196f1 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/a2bd4bf231..a5b6452978 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/9ec1fa7715..459d90be1b * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/f9e21d5658..26a050e9cd * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/bd63626edb..ef835c97f8 * src/third_party/freetype/src: https://chromium.googlesource.com/chromium/src/third_party/freetype2.git/+log/8082aba5f2..46edeb321c * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/0c155e4beb..0483fbe634 * src/third_party/siso/cipd: git_revision:30ea210712239b874febcadf661f118fc2c0f6d7..git_revision:3753a55d0911b4ea3dd000263088f60edb8d947b * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/2187342a4c..312386a772 DEPS diff: https://chromium.googlesource.com/chromium/src/+/6e49907b7a..eaafcde8f7/DEPS BUG=None Change-Id: Iac5ea1489f63adeb47373effb7fb2a8b54a99d66 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392422 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44675} --- DEPS | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/DEPS b/DEPS index 8384405ed6..502a944f01 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '6e49907b7a85360123f20af8910539f0a91d97d9', + 'chromium_revision': 'eaafcde8f79c4b1af5adfe758d16e3c7edc72f18', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -53,7 +53,7 @@ vars = { # reclient CIPD package version 'reclient_version': 're_client_version:0.178.0.5ee9d3e8-gomaip', # siso CIPD package version. - 'siso_version': 'git_revision:30ea210712239b874febcadf661f118fc2c0f6d7', + 'siso_version': 'git_revision:3753a55d0911b4ea3dd000263088f60edb8d947b', # ninja CIPD package. 'ninja_package': 'infra/3pp/tools/ninja/', @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@3ca2b4045a996236ad96e5513aaf6bd07f20fe28', + 'https://chromium.googlesource.com/chromium/src/base@70882196f16e1546cd859093b5c5a18060bc5a4e', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@a2bd4bf2317610bf5d7f113828f94b16dca4f669', + 'https://chromium.googlesource.com/chromium/src/build@a5b64529780d6282198069381f403a2baf28e1ef', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@e9b36e9ffb02b268d194c3f82f89952d09d0c714', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@9ec1fa771571a7eae57cc2f765d9950272cd943f', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@459d90be1bb4f23bd33c528de5e8e7a5b9be748c', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@f9e21d5658bc669e61f53ca2f5bfa6045bc8defe', + 'https://chromium.googlesource.com/chromium/src/testing@26a050e9cd2c2c69d85e40345343397e5e7ea0e0', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@bd63626edb264c364b67e580191110da71641a30', + 'https://chromium.googlesource.com/chromium/src/third_party@ef835c97f8c49a4587a5493d599a95231c4c3252', 'src/buildtools/linux64': { 'packages': [ @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@0c155e4beb9c1706fdafce2326b618a5347c4510', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@0483fbe63478b4868be671e91f61b7b855df6a3f', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -511,7 +511,7 @@ deps = { 'condition': 'checkout_linux', }, 'src/third_party/freetype/src': - 'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@8082aba5f2ddc334b6dfe89a10db13a1b120f76e', + 'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@46edeb321ca346f98b9b744b49aaf2f8e95bd17a', 'src/third_party/harfbuzz-ng/src': 'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@9f83bbbe64654b45ba5bb06927ff36c2e7588495', 'src/third_party/google_benchmark/src': { @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@2187342a4c9b1ddc5ab4b4df5784e9f5d76a6e2d', + 'https://chromium.googlesource.com/chromium/src/tools@312386a772d0a6a987078facd43d2ca4d17cf7f3', 'src/third_party/espresso': { 'packages': [ From b3ceadf8873160839b1c3f7c6899f945aae5e44c Mon Sep 17 00:00:00 2001 From: Junji Watanabe Date: Thu, 15 May 2025 14:30:42 +0900 Subject: [PATCH 204/292] Change build directory layout of build_ios_libs.py This is a preparation for this script to use Siso. Siso's remote execution config assumes that the build directory depth to be 2. e.g. out/foo This CL changes the build directory layout of build_ios_libs.py from `out_ios_libs/{environment}/{cpu_arch}_libs` to `out_ios_libs/{environment}_{cpu_arch}_libs`. This is also aligned with the Chromium's build guide. https://chromium.googlesource.com/chromium/src/+/main/docs/linux/build_instructions.md#setting-up-the-build Test: ios_api_framework in CQ. Bug: b/412968361 Change-Id: I96ceb316fdbf4d16fda7463219d99abc7f65ff33 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391681 Reviewed-by: Mirko Bonadei Reviewed-by: Christoffer Dewerin Commit-Queue: Christoffer Dewerin Auto-Submit: Junji Watanabe Cr-Commit-Position: refs/heads/main@{#44676} --- tools_webrtc/ios/build_ios_libs.py | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/tools_webrtc/ios/build_ios_libs.py b/tools_webrtc/ios/build_ios_libs.py index 50c9398e17..816cee5861 100755 --- a/tools_webrtc/ios/build_ios_libs.py +++ b/tools_webrtc/ios/build_ios_libs.py @@ -126,10 +126,11 @@ def _CleanArtifacts(output_dir): def _CleanTemporary(output_dir, architectures): if os.path.isdir(output_dir): logging.info('Removing temporary build files.') - for arch in architectures: - arch_lib_path = os.path.join(output_dir, arch) - if os.path.isdir(arch_lib_path): - shutil.rmtree(arch_lib_path) + for (env, archs) in architectures.items(): + for arch in archs: + arch_lib_path = '%s_%s_libs' % (env, arch) + if os.path.isdir(arch_lib_path): + shutil.rmtree(arch_lib_path) def _ParseArchitecture(architectures): @@ -232,7 +233,7 @@ def main(): gn_args = args.extra_gn_args if args.purify: - _CleanTemporary(args.output_dir, list(architectures.keys())) + _CleanTemporary(args.output_dir, architectures) return 0 gn_target_name = 'framework_objc' @@ -249,7 +250,7 @@ def main(): framework_paths.append(framework_path) lib_paths = [] for arch in archs: - lib_path = os.path.join(framework_path, arch + '_libs') + lib_path = '%s_%s_libs' % (framework_path, arch) lib_paths.append(lib_path) BuildWebRTC(lib_path, environment, arch, args.build_config, gn_target_name, ios_deployment_target, From ad180cf625fdc7161b426ded63a4e0af17d9c666 Mon Sep 17 00:00:00 2001 From: Tommi Date: Mon, 19 May 2025 11:00:28 +0200 Subject: [PATCH 205/292] Make AudioVector classes and SyncBuffer final, non virtual unless needed MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Bug: chromium:335805780 Change-Id: I28bd4451c7e94e044174e751eacbca49df9898d5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392321 Commit-Queue: Tomas Gunnarsson Reviewed-by: Per Åhgren Cr-Commit-Position: refs/heads/main@{#44677} --- .../audio_coding/neteq/audio_multi_vector.cc | 97 ++++++++----------- .../audio_coding/neteq/audio_multi_vector.h | 47 +++++---- modules/audio_coding/neteq/audio_vector.h | 50 +++++----- modules/audio_coding/neteq/sync_buffer.cc | 2 +- modules/audio_coding/neteq/sync_buffer.h | 15 +-- 5 files changed, 96 insertions(+), 115 deletions(-) diff --git a/modules/audio_coding/neteq/audio_multi_vector.cc b/modules/audio_coding/neteq/audio_multi_vector.cc index aa452808e1..0b68d6b9b3 100644 --- a/modules/audio_coding/neteq/audio_multi_vector.cc +++ b/modules/audio_coding/neteq/audio_multi_vector.cc @@ -21,50 +21,39 @@ namespace webrtc { -AudioMultiVector::AudioMultiVector(size_t N) { +AudioMultiVector::AudioMultiVector(size_t N) : channels_(N) { RTC_DCHECK_GT(N, 0); - if (N < 1) - N = 1; - for (size_t n = 0; n < N; ++n) { - channels_.push_back(new AudioVector); + for (auto& c : channels_) { + c.reset(new AudioVector()); } - num_channels_ = N; } -AudioMultiVector::AudioMultiVector(size_t N, size_t initial_size) { +AudioMultiVector::AudioMultiVector(size_t N, size_t initial_size) + : channels_(N) { RTC_DCHECK_GT(N, 0); - if (N < 1) - N = 1; - for (size_t n = 0; n < N; ++n) { - channels_.push_back(new AudioVector(initial_size)); + for (auto& c : channels_) { + c.reset(new AudioVector(initial_size)); } - num_channels_ = N; } -AudioMultiVector::~AudioMultiVector() { - std::vector::iterator it = channels_.begin(); - while (it != channels_.end()) { - delete (*it); - ++it; - } -} +AudioMultiVector::~AudioMultiVector() = default; void AudioMultiVector::Clear() { - for (size_t i = 0; i < num_channels_; ++i) { - channels_[i]->Clear(); + for (auto& c : channels_) { + c->Clear(); } } void AudioMultiVector::Zeros(size_t length) { - for (size_t i = 0; i < num_channels_; ++i) { - channels_[i]->Clear(); - channels_[i]->Extend(length); + for (auto& c : channels_) { + c->Clear(); + c->Extend(length); } } void AudioMultiVector::CopyTo(AudioMultiVector* copy_to) const { if (copy_to) { - for (size_t i = 0; i < num_channels_; ++i) { + for (size_t i = 0; i < Channels(); ++i) { channels_[i]->CopyTo(&(*copy_to)[i]); } } @@ -72,21 +61,21 @@ void AudioMultiVector::CopyTo(AudioMultiVector* copy_to) const { void AudioMultiVector::PushBackInterleaved( ArrayView append_this) { - RTC_DCHECK_EQ(append_this.size() % num_channels_, 0); + RTC_DCHECK_EQ(append_this.size() % Channels(), 0); if (append_this.empty()) { return; } - if (num_channels_ == 1) { + if (Channels() == 1) { // Special case to avoid extra allocation and data shuffling. channels_[0]->PushBack(append_this.data(), append_this.size()); return; } - size_t length_per_channel = append_this.size() / num_channels_; + size_t length_per_channel = append_this.size() / Channels(); int16_t* temp_array = new int16_t[length_per_channel]; // Temporary storage. - for (size_t channel = 0; channel < num_channels_; ++channel) { + for (size_t channel = 0; channel < Channels(); ++channel) { // Copy elements to `temp_array`. for (size_t i = 0; i < length_per_channel; ++i) { - temp_array[i] = append_this[channel + i * num_channels_]; + temp_array[i] = append_this[channel + i * Channels()]; } channels_[channel]->PushBack(temp_array, length_per_channel); } @@ -94,9 +83,9 @@ void AudioMultiVector::PushBackInterleaved( } void AudioMultiVector::PushBack(const AudioMultiVector& append_this) { - RTC_DCHECK_EQ(num_channels_, append_this.num_channels_); - if (num_channels_ == append_this.num_channels_) { - for (size_t i = 0; i < num_channels_; ++i) { + RTC_DCHECK_EQ(Channels(), append_this.Channels()); + if (Channels() == append_this.Channels()) { + for (size_t i = 0; i < Channels(); ++i) { channels_[i]->PushBack(append_this[i]); } } @@ -107,23 +96,23 @@ void AudioMultiVector::PushBackFromIndex(const AudioMultiVector& append_this, RTC_DCHECK_LT(index, append_this.Size()); index = std::min(index, append_this.Size() - 1); size_t length = append_this.Size() - index; - RTC_DCHECK_EQ(num_channels_, append_this.num_channels_); - if (num_channels_ == append_this.num_channels_) { - for (size_t i = 0; i < num_channels_; ++i) { + RTC_DCHECK_EQ(Channels(), append_this.Channels()); + if (Channels() == append_this.Channels()) { + for (size_t i = 0; i < Channels(); ++i) { channels_[i]->PushBack(append_this[i], length, index); } } } void AudioMultiVector::PopFront(size_t length) { - for (size_t i = 0; i < num_channels_; ++i) { - channels_[i]->PopFront(length); + for (auto& c : channels_) { + c->PopFront(length); } } void AudioMultiVector::PopBack(size_t length) { - for (size_t i = 0; i < num_channels_; ++i) { - channels_[i]->PopBack(length); + for (auto& c : channels_) { + c->PopBack(length); } } @@ -142,13 +131,13 @@ size_t AudioMultiVector::ReadInterleavedFromIndex(size_t start_index, if (length + start_index > Size()) { length = Size() - start_index; } - if (num_channels_ == 1) { + if (Channels() == 1) { // Special case to avoid the nested for loop below. (*this)[0].CopyTo(length, start_index, destination); return length; } for (size_t i = 0; i < length; ++i) { - for (size_t channel = 0; channel < num_channels_; ++channel) { + for (size_t channel = 0; channel < Channels(); ++channel) { destination[index] = (*this)[channel][i + start_index]; ++index; } @@ -165,12 +154,12 @@ size_t AudioMultiVector::ReadInterleavedFromEnd(size_t length, void AudioMultiVector::OverwriteAt(const AudioMultiVector& insert_this, size_t length, size_t position) { - RTC_DCHECK_EQ(num_channels_, insert_this.num_channels_); + RTC_DCHECK_EQ(Channels(), insert_this.Channels()); // Cap `length` at the length of `insert_this`. RTC_DCHECK_LE(length, insert_this.Size()); length = std::min(length, insert_this.Size()); - if (num_channels_ == insert_this.num_channels_) { - for (size_t i = 0; i < num_channels_; ++i) { + if (Channels() == insert_this.Channels()) { + for (size_t i = 0; i < Channels(); ++i) { channels_[i]->OverwriteAt(insert_this[i], length, position); } } @@ -178,16 +167,16 @@ void AudioMultiVector::OverwriteAt(const AudioMultiVector& insert_this, void AudioMultiVector::CrossFade(const AudioMultiVector& append_this, size_t fade_length) { - RTC_DCHECK_EQ(num_channels_, append_this.num_channels_); - if (num_channels_ == append_this.num_channels_) { - for (size_t i = 0; i < num_channels_; ++i) { + RTC_DCHECK_EQ(Channels(), append_this.Channels()); + if (Channels() == append_this.Channels()) { + for (size_t i = 0; i < Channels(); ++i) { channels_[i]->CrossFade(append_this[i], fade_length); } } } size_t AudioMultiVector::Channels() const { - return num_channels_; + return channels_.size(); } size_t AudioMultiVector::Size() const { @@ -198,8 +187,8 @@ size_t AudioMultiVector::Size() const { void AudioMultiVector::AssertSize(size_t required_size) { if (Size() < required_size) { size_t extend_length = required_size - Size(); - for (size_t channel = 0; channel < num_channels_; ++channel) { - channels_[channel]->Extend(extend_length); + for (auto& c : channels_) { + c->Extend(extend_length); } } } @@ -210,9 +199,9 @@ bool AudioMultiVector::Empty() const { } void AudioMultiVector::CopyChannel(size_t from_channel, size_t to_channel) { - RTC_DCHECK_LT(from_channel, num_channels_); - RTC_DCHECK_LT(to_channel, num_channels_); - channels_[from_channel]->CopyTo(channels_[to_channel]); + RTC_DCHECK_LT(from_channel, Channels()); + RTC_DCHECK_LT(to_channel, Channels()); + channels_[from_channel]->CopyTo(channels_[to_channel].get()); } const AudioVector& AudioMultiVector::operator[](size_t index) const { diff --git a/modules/audio_coding/neteq/audio_multi_vector.h b/modules/audio_coding/neteq/audio_multi_vector.h index 6cb4def994..b87a4432b8 100644 --- a/modules/audio_coding/neteq/audio_multi_vector.h +++ b/modules/audio_coding/neteq/audio_multi_vector.h @@ -14,6 +14,7 @@ #include #include +#include #include #include "api/array_view.h" @@ -37,16 +38,16 @@ class AudioMultiVector { AudioMultiVector& operator=(const AudioMultiVector&) = delete; // Deletes all values and make the vector empty. - virtual void Clear(); + void Clear(); // Clears the vector and inserts `length` zeros into each channel. - virtual void Zeros(size_t length); + void Zeros(size_t length); // Copies all values from this vector to `copy_to`. Any contents in `copy_to` // are deleted. After the operation is done, `copy_to` will be an exact // replica of this object. The source and the destination must have the same // number of channels. - virtual void CopyTo(AudioMultiVector* copy_to) const; + void CopyTo(AudioMultiVector* copy_to) const; // Appends the contents of `append_this` to the end of this object. The array // is assumed to be channel-interleaved. The length must be an even multiple @@ -61,34 +62,32 @@ class AudioMultiVector { // Appends the contents of AudioMultiVector `append_this` to this object, // taken from `index` up until the end of `append_this`. The length of this // object is increased. - virtual void PushBackFromIndex(const AudioMultiVector& append_this, - size_t index); + void PushBackFromIndex(const AudioMultiVector& append_this, size_t index); // Removes `length` elements from the beginning of this object, from each // channel. - virtual void PopFront(size_t length); + void PopFront(size_t length); // Removes `length` elements from the end of this object, from each // channel. - virtual void PopBack(size_t length); + void PopBack(size_t length); // Reads `length` samples from each channel and writes them interleaved to // `destination`. The total number of elements written to `destination` is // returned, i.e., `length` * number of channels. If the AudioMultiVector // contains less than `length` samples per channel, this is reflected in the // return value. - virtual size_t ReadInterleaved(size_t length, int16_t* destination) const; + size_t ReadInterleaved(size_t length, int16_t* destination) const; // Like ReadInterleaved() above, but reads from `start_index` instead of from // the beginning. - virtual size_t ReadInterleavedFromIndex(size_t start_index, - size_t length, - int16_t* destination) const; + size_t ReadInterleavedFromIndex(size_t start_index, + size_t length, + int16_t* destination) const; // Like ReadInterleaved() above, but reads from the end instead of from // the beginning. - virtual size_t ReadInterleavedFromEnd(size_t length, - int16_t* destination) const; + size_t ReadInterleavedFromEnd(size_t length, int16_t* destination) const; // Overwrites each channel in this AudioMultiVector with values taken from // `insert_this`. The values are taken from the beginning of `insert_this` and @@ -97,32 +96,31 @@ class AudioMultiVector { // extends beyond the end of the current AudioVector, the vector is extended // to accommodate the new data. `length` is limited to the length of // `insert_this`. - virtual void OverwriteAt(const AudioMultiVector& insert_this, - size_t length, - size_t position); + void OverwriteAt(const AudioMultiVector& insert_this, + size_t length, + size_t position); // Appends `append_this` to the end of the current vector. Lets the two // vectors overlap by `fade_length` samples (per channel), and cross-fade // linearly in this region. - virtual void CrossFade(const AudioMultiVector& append_this, - size_t fade_length); + void CrossFade(const AudioMultiVector& append_this, size_t fade_length); // Returns the number of channels. - virtual size_t Channels() const; + size_t Channels() const; // Returns the number of elements per channel in this AudioMultiVector. - virtual size_t Size() const; + size_t Size() const; // Verify that each channel can hold at least `required_size` elements. If // not, extend accordingly. - virtual void AssertSize(size_t required_size); + void AssertSize(size_t required_size); - virtual bool Empty() const; + bool Empty() const; // Copies the data between two channels in the AudioMultiVector. The method // does not add any new channel. Thus, `from_channel` and `to_channel` must // both be valid channel numbers. - virtual void CopyChannel(size_t from_channel, size_t to_channel); + void CopyChannel(size_t from_channel, size_t to_channel); // Accesses and modifies a channel (i.e., an AudioVector object) of this // AudioMultiVector. @@ -130,8 +128,7 @@ class AudioMultiVector { AudioVector& operator[](size_t index); protected: - std::vector channels_; - size_t num_channels_; + std::vector> channels_; }; } // namespace webrtc diff --git a/modules/audio_coding/neteq/audio_vector.h b/modules/audio_coding/neteq/audio_vector.h index d68f3ec6be..6b37aedf5f 100644 --- a/modules/audio_coding/neteq/audio_vector.h +++ b/modules/audio_coding/neteq/audio_vector.h @@ -20,7 +20,7 @@ namespace webrtc { -class AudioVector { +class AudioVector final { public: // Creates an empty AudioVector. AudioVector(); @@ -28,91 +28,85 @@ class AudioVector { // Creates an AudioVector with an initial size. explicit AudioVector(size_t initial_size); - virtual ~AudioVector(); + ~AudioVector(); AudioVector(const AudioVector&) = delete; AudioVector& operator=(const AudioVector&) = delete; // Deletes all values and make the vector empty. - virtual void Clear(); + void Clear(); // Copies all values from this vector to `copy_to`. Any contents in `copy_to` // are deleted before the copy operation. After the operation is done, // `copy_to` will be an exact replica of this object. - virtual void CopyTo(AudioVector* copy_to) const; + void CopyTo(AudioVector* copy_to) const; // Copies `length` values from `position` in this vector to `copy_to`. - virtual void CopyTo(size_t length, size_t position, int16_t* copy_to) const; + void CopyTo(size_t length, size_t position, int16_t* copy_to) const; // Prepends the contents of AudioVector `prepend_this` to this object. The // length of this object is increased with the length of `prepend_this`. - virtual void PushFront(const AudioVector& prepend_this); + void PushFront(const AudioVector& prepend_this); // Same as above, but with an array `prepend_this` with `length` elements as // source. - virtual void PushFront(const int16_t* prepend_this, size_t length); + void PushFront(const int16_t* prepend_this, size_t length); // Same as PushFront but will append to the end of this object. - virtual void PushBack(const AudioVector& append_this); + void PushBack(const AudioVector& append_this); // Appends a segment of `append_this` to the end of this object. The segment // starts from `position` and has `length` samples. - virtual void PushBack(const AudioVector& append_this, - size_t length, - size_t position); + void PushBack(const AudioVector& append_this, size_t length, size_t position); // Same as PushFront but will append to the end of this object. - virtual void PushBack(const int16_t* append_this, size_t length); + void PushBack(const int16_t* append_this, size_t length); // Removes `length` elements from the beginning of this object. - virtual void PopFront(size_t length); + void PopFront(size_t length); // Removes `length` elements from the end of this object. - virtual void PopBack(size_t length); + void PopBack(size_t length); // Extends this object with `extra_length` elements at the end. The new // elements are initialized to zero. - virtual void Extend(size_t extra_length); + void Extend(size_t extra_length); // Inserts `length` elements taken from the array `insert_this` and insert // them at `position`. The length of the AudioVector is increased by `length`. // `position` = 0 means that the new values are prepended to the vector. // `position` = Size() means that the new values are appended to the vector. - virtual void InsertAt(const int16_t* insert_this, - size_t length, - size_t position); + void InsertAt(const int16_t* insert_this, size_t length, size_t position); // Like InsertAt, but inserts `length` zero elements at `position`. - virtual void InsertZerosAt(size_t length, size_t position); + void InsertZerosAt(size_t length, size_t position); // Overwrites `length` elements of this AudioVector starting from `position` // with first values in `AudioVector`. The definition of `position` // is the same as for InsertAt(). If `length` and `position` are selected // such that the new data extends beyond the end of the current AudioVector, // the vector is extended to accommodate the new data. - virtual void OverwriteAt(const AudioVector& insert_this, - size_t length, - size_t position); + void OverwriteAt(const AudioVector& insert_this, + size_t length, + size_t position); // Overwrites `length` elements of this AudioVector with values taken from the // array `insert_this`, starting at `position`. The definition of `position` // is the same as for InsertAt(). If `length` and `position` are selected // such that the new data extends beyond the end of the current AudioVector, // the vector is extended to accommodate the new data. - virtual void OverwriteAt(const int16_t* insert_this, - size_t length, - size_t position); + void OverwriteAt(const int16_t* insert_this, size_t length, size_t position); // Appends `append_this` to the end of the current vector. Lets the two // vectors overlap by `fade_length` samples, and cross-fade linearly in this // region. - virtual void CrossFade(const AudioVector& append_this, size_t fade_length); + void CrossFade(const AudioVector& append_this, size_t fade_length); // Returns the number of elements in this AudioVector. - virtual size_t Size() const; + size_t Size() const; // Returns true if this AudioVector is empty. - virtual bool Empty() const; + bool Empty() const; // Accesses and modifies an element of AudioVector. inline const int16_t& operator[](size_t index) const { diff --git a/modules/audio_coding/neteq/sync_buffer.cc b/modules/audio_coding/neteq/sync_buffer.cc index 87d56baf21..26fa54f2a6 100644 --- a/modules/audio_coding/neteq/sync_buffer.cc +++ b/modules/audio_coding/neteq/sync_buffer.cc @@ -78,7 +78,7 @@ void SyncBuffer::ReplaceAtIndex(const AudioMultiVector& insert_this, size_t position) { position = std::min(position, Size()); // Cap `position` in the valid range. length = std::min(length, Size() - position); - AudioMultiVector::OverwriteAt(insert_this, length, position); + OverwriteAt(insert_this, length, position); } void SyncBuffer::ReplaceAtIndex(const AudioMultiVector& insert_this, diff --git a/modules/audio_coding/neteq/sync_buffer.h b/modules/audio_coding/neteq/sync_buffer.h index ea81babca6..7374fdd685 100644 --- a/modules/audio_coding/neteq/sync_buffer.h +++ b/modules/audio_coding/neteq/sync_buffer.h @@ -23,7 +23,7 @@ namespace webrtc { -class SyncBuffer : public AudioMultiVector { +class SyncBuffer final : public AudioMultiVector { public: SyncBuffer(size_t channels, size_t length) : AudioMultiVector(channels, length), @@ -34,6 +34,8 @@ class SyncBuffer : public AudioMultiVector { SyncBuffer(const SyncBuffer&) = delete; SyncBuffer& operator=(const SyncBuffer&) = delete; + ~SyncBuffer() override = default; + // Returns the number of samples yet to play out from the buffer. size_t FutureLength() const; @@ -57,7 +59,7 @@ class SyncBuffer : public AudioMultiVector { // Inserts `length` zeros into each channel at index `position`. The size of // the SyncBuffer is kept constant, which means that the last `length` // elements in each channel will be purged. - virtual void InsertZerosAtIndex(size_t length, size_t position); + void InsertZerosAtIndex(size_t length, size_t position); // Overwrites each channel in this SyncBuffer with values taken from // `insert_this`. The values are taken from the beginning of `insert_this` and @@ -66,14 +68,13 @@ class SyncBuffer : public AudioMultiVector { // and `position` are selected such that the new data would extend beyond the // end of the current SyncBuffer, the buffer is not extended. // The `next_index_` is not updated. - virtual void ReplaceAtIndex(const AudioMultiVector& insert_this, - size_t length, - size_t position); + void ReplaceAtIndex(const AudioMultiVector& insert_this, + size_t length, + size_t position); // Same as the above method, but where all of `insert_this` is written (with // the same constraints as above, that the SyncBuffer is not extended). - virtual void ReplaceAtIndex(const AudioMultiVector& insert_this, - size_t position); + void ReplaceAtIndex(const AudioMultiVector& insert_this, size_t position); // Reads `requested_len` samples from each channel and writes them interleaved // into `output`. The `next_index_` is updated to point to the sample to read From 0acb637a0c053798742bf3e802554d28156075a5 Mon Sep 17 00:00:00 2001 From: Jeremy Leconte Date: Mon, 19 May 2025 14:33:47 +0200 Subject: [PATCH 206/292] Disable Android FYI from LKGR Change-Id: Iab77d59d4e08b1f7fc5a468d9870338aae9efd77 Bug: b/418717425 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392480 Reviewed-by: Ilya Nikolaevskiy Commit-Queue: Jeremy Leconte Cr-Commit-Position: refs/heads/main@{#44678} --- infra/config/config.star | 3 ++- infra/config/cr-buildbucket.cfg | 1 - 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/infra/config/config.star b/infra/config/config.star index c31054c7b6..ba5be58866 100755 --- a/infra/config/config.star +++ b/infra/config/config.star @@ -905,7 +905,8 @@ lkgr_config = { "WebRTC Chromium FYI Android Builder ARM64 (dbg)", "WebRTC Chromium FYI Android Builder", "WebRTC Chromium FYI Android Tests (dbg)", - "WebRTC Chromium FYI Android Tests ARM64 (dbg)", + # TODO: b/418717425 - Reenable when flakiness is fixed. + # "WebRTC Chromium FYI Android Tests ARM64 (dbg)", "WebRTC Chromium FYI Linux Builder (dbg)", "WebRTC Chromium FYI Linux Builder", "WebRTC Chromium FYI Linux Tester", diff --git a/infra/config/cr-buildbucket.cfg b/infra/config/cr-buildbucket.cfg index 1d79a2971e..cc571f3a7a 100644 --- a/infra/config/cr-buildbucket.cfg +++ b/infra/config/cr-buildbucket.cfg @@ -2190,7 +2190,6 @@ buckets { ' "WebRTC Chromium FYI Android Builder ARM64 (dbg)",' ' "WebRTC Chromium FYI Android Builder",' ' "WebRTC Chromium FYI Android Tests (dbg)",' - ' "WebRTC Chromium FYI Android Tests ARM64 (dbg)",' ' "WebRTC Chromium FYI Linux Builder (dbg)",' ' "WebRTC Chromium FYI Linux Builder",' ' "WebRTC Chromium FYI Linux Tester",' From 8f88273f1c396fe333f865fcaf078aefdfd8927f Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Mon, 19 May 2025 07:02:11 -0700 Subject: [PATCH 207/292] Roll chromium_revision eaafcde8f7..ef35462da9 (1461988:1462122) Change log: https://chromium.googlesource.com/chromium/src/+log/eaafcde8f7..ef35462da9 Full diff: https://chromium.googlesource.com/chromium/src/+/eaafcde8f7..ef35462da9 Changed dependencies * src/buildtools: https://chromium.googlesource.com/chromium/src/buildtools/+log/e9b36e9ffb..cd169314af * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/459d90be1b..58194b05f7 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/26a050e9cd..0edef62e15 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/ef835c97f8..dacddfcd6f * src/third_party/androidx/cipd: dKxftVsUeWTHR64K1ywUo9zWElvcUVqJnGPnDVQbVGcC..T6iAJzbVHlw3lE6kxx7uAYnuanajCjzpRUt8G7WIrgEC * src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/1d27130259..21a487e9fa * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/0483fbe634..60436753f7 * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/5acde97bf6..998e664be7 * src/third_party/turbine/cipd: PbV7UFdzFIl6b_4lNwsj4VnlvnoULNAZRDwsndTGXTsC..VGtOG2ivl1SJR7Lai5FQddIu15mWCYDnp47QtozMQeoC * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/312386a772..fc1aaf91a5 DEPS diff: https://chromium.googlesource.com/chromium/src/+/eaafcde8f7..ef35462da9/DEPS BUG=None Change-Id: Icd2d60ffc79dc2fc257b3e2717be48f956ee7a1b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392425 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44679} --- DEPS | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/DEPS b/DEPS index 502a944f01..8745bff0d6 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': 'eaafcde8f79c4b1af5adfe758d16e3c7edc72f18', + 'chromium_revision': 'ef35462da9b45979619e59c9542a0b7eef4036b5', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -77,20 +77,20 @@ deps = { 'src/build': 'https://chromium.googlesource.com/chromium/src/build@a5b64529780d6282198069381f403a2baf28e1ef', 'src/buildtools': - 'https://chromium.googlesource.com/chromium/src/buildtools@e9b36e9ffb02b268d194c3f82f89952d09d0c714', + 'https://chromium.googlesource.com/chromium/src/buildtools@cd169314af80e77a75eb25e2c0df8798f35bc04a', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. 'src/examples/androidtests/third_party/gradle': { 'url': 'https://chromium.googlesource.com/external/github.com/gradle/gradle.git@f2d1fb54a951d8b11d25748e4711bec8d128d7e3', 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@459d90be1bb4f23bd33c528de5e8e7a5b9be748c', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@58194b05f7ca51bb6b30367cc621f4914c4394b1', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@26a050e9cd2c2c69d85e40345343397e5e7ea0e0', + 'https://chromium.googlesource.com/chromium/src/testing@0edef62e15b448d4632e6df629b8ed06c7694ec4', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@ef835c97f8c49a4587a5493d599a95231c4c3252', + 'https://chromium.googlesource.com/chromium/src/third_party@dacddfcd6fd4d8275e3b7a20c0765dd1dd1ba964', 'src/buildtools/linux64': { 'packages': [ @@ -323,7 +323,7 @@ deps = { 'src/third_party/clang-format/script': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@37f6e68a107df43b7d7e044fd36a13cbae3413f2', 'src/third_party/compiler-rt/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@1d27130259209a2b5c8ac992b832fc5522f38a8d', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@21a487e9fae57351be896c5c34dce665e0990c76', 'src/third_party/libc++/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@a01c02c9d4acbdae3b7e8a2f3ee58579a9c29f96', 'src/third_party/libc++abi/src': @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@0483fbe63478b4868be671e91f61b7b855df6a3f', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@60436753f7fea32a0ae4a168d08803db52e83b77', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '5acde97bf62e479089bd6466a8d84cd80846f502', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '998e664be79bba4703c0674ff62a29e37cbeccdb', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@312386a772d0a6a987078facd43d2ca4d17cf7f3', + 'https://chromium.googlesource.com/chromium/src/tools@fc1aaf91a5b9264951422799a02b1c96d6dd8b22', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'dKxftVsUeWTHR64K1ywUo9zWElvcUVqJnGPnDVQbVGcC', + 'version': 'T6iAJzbVHlw3lE6kxx7uAYnuanajCjzpRUt8G7WIrgEC', }, ], 'condition': 'checkout_android and non_git_source', @@ -777,7 +777,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/turbine', - 'version': 'PbV7UFdzFIl6b_4lNwsj4VnlvnoULNAZRDwsndTGXTsC', + 'version': 'VGtOG2ivl1SJR7Lai5FQddIu15mWCYDnp47QtozMQeoC', }, ], 'condition': 'checkout_android', From e08286d180ea3c2db399e758cf873ec33b591c82 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Mon, 19 May 2025 11:02:02 -0700 Subject: [PATCH 208/292] Roll chromium_revision ef35462da9..501a302fb1 (1462122:1462324) Change log: https://chromium.googlesource.com/chromium/src/+log/ef35462da9..501a302fb1 Full diff: https://chromium.googlesource.com/chromium/src/+/ef35462da9..501a302fb1 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/70882196f1..ada68cc8fe * src/build: https://chromium.googlesource.com/chromium/src/build/+log/a5b6452978..34988eff97 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/58194b05f7..9670766b89 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/0edef62e15..ac075709d0 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/dacddfcd6f..51693a8265 * src/third_party/androidx/cipd: T6iAJzbVHlw3lE6kxx7uAYnuanajCjzpRUt8G7WIrgEC..Lnwi6S4AHbruhZPbul63nDrhnbshtefsUCrilju6ptEC * src/third_party/freetype/src: https://chromium.googlesource.com/chromium/src/third_party/freetype2.git/+log/46edeb321c..b87f48a73e * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/60436753f7..34e0eb6ead * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/998e664be7..15701312cf * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/fc1aaf91a5..079aa8f4be DEPS diff: https://chromium.googlesource.com/chromium/src/+/ef35462da9..501a302fb1/DEPS BUG=None Change-Id: Ifddc66bc11e6eb324f58a3a05faf0656fdc151a6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392407 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44680} --- DEPS | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/DEPS b/DEPS index 8745bff0d6..6b2d026a42 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': 'ef35462da9b45979619e59c9542a0b7eef4036b5', + 'chromium_revision': '501a302fb1562cfbea0947f03615857e5bcc1d88', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@70882196f16e1546cd859093b5c5a18060bc5a4e', + 'https://chromium.googlesource.com/chromium/src/base@ada68cc8fe3778e1bb1fe61701a200bd45d6e4a3', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@a5b64529780d6282198069381f403a2baf28e1ef', + 'https://chromium.googlesource.com/chromium/src/build@34988eff976770c637669a3724813b949db6e6e9', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@cd169314af80e77a75eb25e2c0df8798f35bc04a', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@58194b05f7ca51bb6b30367cc621f4914c4394b1', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@9670766b89d2f3d67fcc15de79ffba09516964b8', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@0edef62e15b448d4632e6df629b8ed06c7694ec4', + 'https://chromium.googlesource.com/chromium/src/testing@ac075709d047e628249f73bf40ee26e7a75c99a2', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@dacddfcd6fd4d8275e3b7a20c0765dd1dd1ba964', + 'https://chromium.googlesource.com/chromium/src/third_party@51693a8265c9de0283806b965b82197e51d55ba4', 'src/buildtools/linux64': { 'packages': [ @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@60436753f7fea32a0ae4a168d08803db52e83b77', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@34e0eb6eadf97d95ea90da963d1dc2cf2b41dbe5', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -511,7 +511,7 @@ deps = { 'condition': 'checkout_linux', }, 'src/third_party/freetype/src': - 'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@46edeb321ca346f98b9b744b49aaf2f8e95bd17a', + 'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@b87f48a73e3ac66c0a2b4155209a62cdcbea61b2', 'src/third_party/harfbuzz-ng/src': 'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@9f83bbbe64654b45ba5bb06927ff36c2e7588495', 'src/third_party/google_benchmark/src': { @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '998e664be79bba4703c0674ff62a29e37cbeccdb', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '15701312cfc4a5c2c3eef159baa0df80dbb917aa', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@fc1aaf91a5b9264951422799a02b1c96d6dd8b22', + 'https://chromium.googlesource.com/chromium/src/tools@079aa8f4beb3ca18c6936e7c6a1c7f4b67d37581', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'T6iAJzbVHlw3lE6kxx7uAYnuanajCjzpRUt8G7WIrgEC', + 'version': 'Lnwi6S4AHbruhZPbul63nDrhnbshtefsUCrilju6ptEC', }, ], 'condition': 'checkout_android and non_git_source', From 2e8ca304bcc622bab8c8d487f9eb68a50acd1af6 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Mon, 19 May 2025 15:10:52 -0700 Subject: [PATCH 209/292] Roll chromium_revision 501a302fb1..fcfd887fc2 (1462324:1462461) Change log: https://chromium.googlesource.com/chromium/src/+log/501a302fb1..fcfd887fc2 Full diff: https://chromium.googlesource.com/chromium/src/+/501a302fb1..fcfd887fc2 Changed dependencies * src/build: https://chromium.googlesource.com/chromium/src/build/+log/34988eff97..acc673b173 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/9670766b89..0c74a70e1f * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/ac075709d0..5f6f6f9f72 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/51693a8265..c7a59c18eb * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/f810dae272..196d941610 * src/third_party/googletest/src: https://chromium.googlesource.com/external/github.com/google/googletest.git/+log/571930618f..fa8438ae6b * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/34e0eb6ead..bf41d8f75b * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/15701312cf..5a66f76326 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/079aa8f4be..08504d5c50 DEPS diff: https://chromium.googlesource.com/chromium/src/+/501a302fb1..fcfd887fc2/DEPS BUG=None Change-Id: I77ca2b49c54b1e25c14a1b2f1e689eb9ce86fb62 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392427 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44681} --- DEPS | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/DEPS b/DEPS index 6b2d026a42..711d9f085e 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '501a302fb1562cfbea0947f03615857e5bcc1d88', + 'chromium_revision': 'fcfd887fc260846cf136efee02e92a733cd4383b', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -75,7 +75,7 @@ deps = { 'src/base': 'https://chromium.googlesource.com/chromium/src/base@ada68cc8fe3778e1bb1fe61701a200bd45d6e4a3', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@34988eff976770c637669a3724813b949db6e6e9', + 'https://chromium.googlesource.com/chromium/src/build@acc673b173b1ace9099d2dc56c7da078f378528e', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@cd169314af80e77a75eb25e2c0df8798f35bc04a', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@9670766b89d2f3d67fcc15de79ffba09516964b8', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@0c74a70e1f0e73fd0f5bd75470faf83981e43b6b', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@ac075709d047e628249f73bf40ee26e7a75c99a2', + 'https://chromium.googlesource.com/chromium/src/testing@5f6f6f9f720f0495513978181fdf862eaf61dfdc', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@51693a8265c9de0283806b965b82197e51d55ba4', + 'https://chromium.googlesource.com/chromium/src/third_party@c7a59c18eb200d55a1283963c0456e0d3e8ea15f', 'src/buildtools/linux64': { 'packages': [ @@ -497,13 +497,13 @@ deps = { 'src/third_party/crc32c/src': 'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558', 'src/third_party/depot_tools': - 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@f810dae27200e4209bd0cba85bd2266ec0d55312', + 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@196d9416106a6ace0729b09acda46546c4a12ce1', 'src/third_party/ffmpeg': 'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@dcdd0fa51b65a0b1688ff6b8f0cc81908f09ded2', 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@34e0eb6eadf97d95ea90da963d1dc2cf2b41dbe5', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@bf41d8f75bcdbf5ef6d8b721e8cee1e8bf547dc5', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -525,7 +525,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/googletest/src': - 'https://chromium.googlesource.com/external/github.com/google/googletest.git@571930618fa96eabcd05b573285edbee9fc13bae', + 'https://chromium.googlesource.com/external/github.com/google/googletest.git@fa8438ae6b70c57010177de47a9f13d7041a6328', 'src/third_party/icu': { 'url': 'https://chromium.googlesource.com/chromium/deps/icu.git@b929596baebf0ab4ac7ec07f38365db4c50a559d', }, @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '15701312cfc4a5c2c3eef159baa0df80dbb917aa', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '5a66f76326a04bf33f05433665cd77b7eb288fcc', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@079aa8f4beb3ca18c6936e7c6a1c7f4b67d37581', + 'https://chromium.googlesource.com/chromium/src/tools@08504d5c5013a559de309c86329dae0c893500ad', 'src/third_party/espresso': { 'packages': [ From c03d7d6382f1d21f785aef91c7f23022e3fe7a77 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Mon, 19 May 2025 21:02:35 -0700 Subject: [PATCH 210/292] Roll chromium_revision fcfd887fc2..c89ca3267f (1462461:1462567) Change log: https://chromium.googlesource.com/chromium/src/+log/fcfd887fc2..c89ca3267f Full diff: https://chromium.googlesource.com/chromium/src/+/fcfd887fc2..c89ca3267f Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/ada68cc8fe..cf27d977ab * src/build: https://chromium.googlesource.com/chromium/src/build/+log/acc673b173..79fe614714 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/5f6f6f9f72..c25e306ec3 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/c7a59c18eb..74eaeae48c * src/third_party/androidx/cipd: Lnwi6S4AHbruhZPbul63nDrhnbshtefsUCrilju6ptEC..9gK5fctsrqSL7ITHbaKR8oEVCBo_gw0XIMzM9kvuSI0C * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/bf41d8f75b..198e465646 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/08504d5c50..e9b9249726 DEPS diff: https://chromium.googlesource.com/chromium/src/+/fcfd887fc2..c89ca3267f/DEPS BUG=None Change-Id: Ia861c21938229819b67d7bea8d285b31effd2329 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392429 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44682} --- DEPS | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/DEPS b/DEPS index 711d9f085e..3fd5d840c2 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': 'fcfd887fc260846cf136efee02e92a733cd4383b', + 'chromium_revision': 'c89ca3267f93966b53d212ada6baf989ca17ed09', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@ada68cc8fe3778e1bb1fe61701a200bd45d6e4a3', + 'https://chromium.googlesource.com/chromium/src/base@cf27d977ab48cbc6eaf94ff7499cdf5e7978993f', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@acc673b173b1ace9099d2dc56c7da078f378528e', + 'https://chromium.googlesource.com/chromium/src/build@79fe6147149c124bf7b4370aa623db917609486b', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@cd169314af80e77a75eb25e2c0df8798f35bc04a', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -88,9 +88,9 @@ deps = { 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@5f6f6f9f720f0495513978181fdf862eaf61dfdc', + 'https://chromium.googlesource.com/chromium/src/testing@c25e306ec3d55d5b3189908e3a35cdfafd8a474e', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@c7a59c18eb200d55a1283963c0456e0d3e8ea15f', + 'https://chromium.googlesource.com/chromium/src/third_party@74eaeae48c7c505a8dfe393eb0e16f3d82e0b881', 'src/buildtools/linux64': { 'packages': [ @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@bf41d8f75bcdbf5ef6d8b721e8cee1e8bf547dc5', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@198e465646b36a6106bbc93d26c80b3850791966', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@08504d5c5013a559de309c86329dae0c893500ad', + 'https://chromium.googlesource.com/chromium/src/tools@e9b9249726464abbfc9525625dfeca2ad0064e59', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'Lnwi6S4AHbruhZPbul63nDrhnbshtefsUCrilju6ptEC', + 'version': '9gK5fctsrqSL7ITHbaKR8oEVCBo_gw0XIMzM9kvuSI0C', }, ], 'condition': 'checkout_android and non_git_source', From 388d74528ec7e6a1473fa632943b2ecca98a39e9 Mon Sep 17 00:00:00 2001 From: webrtc-version-updater Date: Mon, 19 May 2025 21:06:22 -0700 Subject: [PATCH 211/292] Update WebRTC code version (2025-05-20T04:05:45). Bug: None Change-Id: I1223d56786ed1381084887756043eab4db55dda0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392430 Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Cr-Commit-Position: refs/heads/main@{#44683} --- call/version.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/call/version.cc b/call/version.cc index e8cec37240..8c9ae5dc4c 100644 --- a/call/version.cc +++ b/call/version.cc @@ -13,7 +13,7 @@ namespace webrtc { // The timestamp is always in UTC. -const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-19T04:06:36"; +const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-20T04:05:45"; void LoadWebRTCVersionInRegister() { // Using volatile to instruct the compiler to not optimize `p` away even From 5d165b1171f2a13ac7cf7e8c979050140959eb65 Mon Sep 17 00:00:00 2001 From: Junji Watanabe Date: Thu, 15 May 2025 14:34:41 +0900 Subject: [PATCH 212/292] Use Siso in iOS/Android build scripts build_aar.py and build_ios_libs.py are still using Ninja to build. This CL switches them to use Siso. The CL needs to be submitted with the corresponding recipe change https://crrev.com/c/6546855. Bug: b/412968361 Change-Id: I46042aaaac833f7840bc37510b30d2c45d2d0a31 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391480 Reviewed-by: Mirko Bonadei Commit-Queue: Junji Watanabe Reviewed-by: Christoffer Dewerin Cr-Commit-Position: refs/heads/main@{#44684} --- tools_webrtc/android/build_aar.py | 15 +++++++++++---- tools_webrtc/ios/build_ios_libs.py | 12 +++++++++--- 2 files changed, 20 insertions(+), 7 deletions(-) diff --git a/tools_webrtc/android/build_aar.py b/tools_webrtc/android/build_aar.py index d3da72694c..e394f75883 100755 --- a/tools_webrtc/android/build_aar.py +++ b/tools_webrtc/android/build_aar.py @@ -118,8 +118,10 @@ def _RunGN(args): def _RunNinja(output_directory, args): cmd = [ - os.path.join(SRC_DIR, 'third_party', 'ninja', 'ninja'), '-C', - output_directory + os.path.join(SRC_DIR, 'third_party', 'siso', 'cipd', 'siso'), + 'ninja', + '-C', + output_directory, ] cmd.extend(args) logging.debug('Running: %r', cmd) @@ -175,9 +177,14 @@ def Build(build_dir, arch, use_remoteexec, extra_gn_args, extra_gn_switches, 'is_component_build': False, 'rtc_include_tests': False, 'target_cpu': _GetTargetCpu(arch), - 'use_remoteexec': use_remoteexec, 'android_static_analysis': "off", + 'use_siso': True, } + if use_remoteexec: + gn_args.update({ + 'use_remoteexec': True, + 'use_reclient': False, + }) arm_version = _GetArmVersion(arch) if arm_version: gn_args['arm_version'] = arm_version @@ -191,7 +198,7 @@ def Build(build_dir, arch, use_remoteexec, extra_gn_args, extra_gn_switches, ninja_args = TARGETS[:] if use_remoteexec: - ninja_args.extend(['-j', '200']) + ninja_args.extend(['-remote_jobs', '200']) ninja_args.extend(extra_ninja_switches) _RunNinja(output_directory, ninja_args) diff --git a/tools_webrtc/ios/build_ios_libs.py b/tools_webrtc/ios/build_ios_libs.py index 816cee5861..08ae2f77ec 100755 --- a/tools_webrtc/ios/build_ios_libs.py +++ b/tools_webrtc/ios/build_ios_libs.py @@ -190,8 +190,13 @@ def BuildWebRTC(output_dir, target_environment, target_arch, flavor, ('true' if libvpx_build_vp9 else 'false')) gn_args.append('use_lld=true') - gn_args.append('use_remoteexec=' + ('true' if use_remoteexec else 'false')) gn_args.append('rtc_enable_objc_symbol_export=true') + gn_args.append('use_siso=true') + if use_remoteexec: + gn_args.extend([ + 'use_remoteexec=true', + 'use_reclient=false', + ]) args_string = ' '.join(gn_args + extra_gn_args) logging.info('Building WebRTC with args: %s', args_string) @@ -207,13 +212,14 @@ def BuildWebRTC(output_dir, target_environment, target_arch, flavor, logging.info('Building target: %s', gn_target_name) cmd = [ - os.path.join(SRC_DIR, 'third_party', 'ninja', 'ninja'), + os.path.join(SRC_DIR, 'third_party', 'siso', 'cipd', 'siso'), + 'ninja', '-C', output_dir, gn_target_name, ] if use_remoteexec: - cmd.extend(['-j', '200']) + cmd.extend(['-remote_jobs', '200']) _RunCommand(cmd) From ee93f2d1fa2e3c470feed8a9a6169077fcb1f6e8 Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Fri, 16 May 2025 11:45:49 +0200 Subject: [PATCH 213/292] Propagate Environment through the JavaAudioDeviceModule MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Bug: webrtc:413413572 Change-Id: I1f26b3ac48dd856cdc4a6c5d4161ddc3707158a0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391760 Auto-Submit: Danil Chapovalov Reviewed-by: Zoé Lepaul Commit-Queue: Zoé Lepaul Cr-Commit-Position: refs/heads/main@{#44685} --- sdk/android/BUILD.gn | 13 +++++---- .../webrtc/audio/JavaAudioDeviceModule.java | 8 ++++-- .../audio_device_android.cc | 2 +- .../src/jni/audio_device/audio_track_jni.cc | 28 +++++++++++++------ .../src/jni/audio_device/audio_track_jni.h | 9 ++++-- .../audio_device/java_audio_device_module.cc | 17 +++++++---- 6 files changed, 52 insertions(+), 25 deletions(-) diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn index f1e75262b3..06d81a9719 100644 --- a/sdk/android/BUILD.gn +++ b/sdk/android/BUILD.gn @@ -867,16 +867,17 @@ if (current_os == "linux" || is_android) { visibility = [ "*" ] sources = [ "src/jni/audio_device/java_audio_device_module.cc" ] - # TODO: bugs.webrtc.org/413413572 - Remove poison when Environment is - # propagated + # TODO: bugs.webrtc.org/413413572 - Remove poison when + # 'audio_device_module_base' is no longer poisonous. allow_poison = [ "environment_construction" ] deps = [ - ":base_jni", + ":audio_device_module_base", ":generated_java_audio_jni", ":java_audio_device_module", + ":native_api_jni", + "../../api/audio:audio_device", "../../api/environment", - "../../api/environment:environment_factory", "//third_party/jni_zero", ] } @@ -1274,15 +1275,17 @@ if (current_os == "linux" || is_android) { ":audio_device_module_base", ":base_jni", ":generated_java_audio_device_module_native_jni", + ":internal_jni", + ":native_api_jni", "../../api:sequence_checker", "../../api/audio:audio_device", + "../../api/environment", "../../modules/audio_device:audio_device_buffer", "../../rtc_base:checks", "../../rtc_base:logging", "../../rtc_base:macromagic", "../../rtc_base:platform_thread", "../../rtc_base:timeutils", - "../../system_wrappers:field_trial", "../../system_wrappers:metrics", "//third_party/jni_zero", ] diff --git a/sdk/android/api/org/webrtc/audio/JavaAudioDeviceModule.java b/sdk/android/api/org/webrtc/audio/JavaAudioDeviceModule.java index 8e38c1f34f..bbf1c3a753 100644 --- a/sdk/android/api/org/webrtc/audio/JavaAudioDeviceModule.java +++ b/sdk/android/api/org/webrtc/audio/JavaAudioDeviceModule.java @@ -386,11 +386,12 @@ private JavaAudioDeviceModule(Context context, AudioManager audioManager, } @Override - public long getNativeAudioDeviceModulePointer() { + public long getNative(long webrtcEnvRef) { synchronized (nativeLock) { if (nativeAudioDeviceModule == 0) { nativeAudioDeviceModule = nativeCreateAudioDeviceModule(context, audioManager, audioInput, - audioOutput, inputSampleRate, outputSampleRate, useStereoInput, useStereoOutput); + audioOutput, webrtcEnvRef, inputSampleRate, outputSampleRate, useStereoInput, + useStereoOutput); } return nativeAudioDeviceModule; } @@ -438,5 +439,6 @@ public void setPreferredInputDevice(AudioDeviceInfo preferredInputDevice) { private static native long nativeCreateAudioDeviceModule(Context context, AudioManager audioManager, WebRtcAudioRecord audioInput, WebRtcAudioTrack audioOutput, - int inputSampleRate, int outputSampleRate, boolean useStereoInput, boolean useStereoOutput); + long webrtcEnvRef, int inputSampleRate, int outputSampleRate, boolean useStereoInput, + boolean useStereoOutput); } diff --git a/sdk/android/native_api/audio_device_module/audio_device_android.cc b/sdk/android/native_api/audio_device_module/audio_device_android.cc index eee261682c..200c37aa8b 100644 --- a/sdk/android/native_api/audio_device_module/audio_device_android.cc +++ b/sdk/android/native_api/audio_device_module/audio_device_android.cc @@ -125,7 +125,7 @@ scoped_refptr CreateJavaAudioDeviceModule( jni::AudioRecordJni::CreateJavaWebRtcAudioRecord(env, j_context, j_audio_manager)); auto audio_output = std::make_unique( - env, output_parameters, + env, webrtc_env, output_parameters, jni::AudioTrackJni::CreateJavaWebRtcAudioTrack(env, j_context, j_audio_manager)); return CreateAudioDeviceModuleFromInputAndOutput( diff --git a/sdk/android/src/jni/audio_device/audio_track_jni.cc b/sdk/android/src/jni/audio_device/audio_track_jni.cc index 097c07c5d5..59d6d75f36 100644 --- a/sdk/android/src/jni/audio_device/audio_track_jni.cc +++ b/sdk/android/src/jni/audio_device/audio_track_jni.cc @@ -10,15 +10,22 @@ #include "sdk/android/src/jni/audio_device/audio_track_jni.h" -#include +#include -#include "rtc_base/arraysize.h" +#include +#include +#include +#include + +#include "api/audio/audio_device_defines.h" +#include "api/environment/environment.h" +#include "modules/audio_device/audio_device_buffer.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" -#include "rtc_base/platform_thread.h" #include "sdk/android/generated_java_audio_device_module_native_jni/WebRtcAudioTrack_jni.h" +#include "sdk/android/native_api/jni/scoped_java_ref.h" #include "sdk/android/src/jni/jni_helpers.h" -#include "system_wrappers/include/field_trial.h" +#include "sdk/android/src/jni/jvm.h" #include "system_wrappers/include/metrics.h" #include "third_party/jni_zero/jni_zero.h" @@ -35,9 +42,11 @@ ScopedJavaLocalRef AudioTrackJni::CreateJavaWebRtcAudioTrack( AudioTrackJni::AudioTrackJni( JNIEnv* env, + const Environment& webrtc_env, const AudioParameters& audio_parameters, const jni_zero::JavaRef& j_webrtc_audio_track) - : j_audio_track_(env, j_webrtc_audio_track), + : webrtc_env_(webrtc_env), + j_audio_track_(env, j_webrtc_audio_track), audio_parameters_(audio_parameters), direct_buffer_address_(nullptr), direct_buffer_capacity_in_bytes_(0), @@ -84,10 +93,11 @@ int32_t AudioTrackJni::InitPlayout() { return 0; } RTC_DCHECK(!playing_); - double buffer_size_factor = strtod( - field_trial::FindFullName("WebRTC-AudioDevicePlayoutBufferSizeFactor") - .c_str(), - nullptr); + double buffer_size_factor = + strtod(webrtc_env_.field_trials() + .Lookup("WebRTC-AudioDevicePlayoutBufferSizeFactor") + .c_str(), + nullptr); if (buffer_size_factor == 0) buffer_size_factor = 1.0; int requested_buffer_size_bytes = Java_WebRtcAudioTrack_initPlayout( diff --git a/sdk/android/src/jni/audio_device/audio_track_jni.h b/sdk/android/src/jni/audio_device/audio_track_jni.h index fee50ef77f..8044182ef1 100644 --- a/sdk/android/src/jni/audio_device/audio_track_jni.h +++ b/sdk/android/src/jni/audio_device/audio_track_jni.h @@ -13,14 +13,16 @@ #include -#include +#include +#include #include #include "api/audio/audio_device_defines.h" +#include "api/environment/environment.h" #include "api/sequence_checker.h" #include "modules/audio_device/audio_device_buffer.h" -#include "sdk/android/src/jni/audio_device/audio_common.h" #include "sdk/android/src/jni/audio_device/audio_device_module.h" +#include "third_party/jni_zero/jni_zero.h" namespace webrtc { @@ -47,6 +49,7 @@ class AudioTrackJni : public AudioOutput { const jni_zero::JavaRef& j_audio_manager); AudioTrackJni(JNIEnv* env, + const Environment& webrtc_env, const AudioParameters& audio_parameters, const jni_zero::JavaRef& j_webrtc_audio_track); ~AudioTrackJni() override; @@ -85,6 +88,8 @@ class AudioTrackJni : public AudioOutput { void GetPlayoutData(JNIEnv* env, size_t length); private: + const Environment webrtc_env_; + // Stores thread ID in constructor. SequenceChecker thread_checker_; diff --git a/sdk/android/src/jni/audio_device/java_audio_device_module.cc b/sdk/android/src/jni/audio_device/java_audio_device_module.cc index 7f58163a7f..a8152a39c4 100644 --- a/sdk/android/src/jni/audio_device/java_audio_device_module.cc +++ b/sdk/android/src/jni/audio_device/java_audio_device_module.cc @@ -8,14 +8,20 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include + #include +#include +#include "api/audio/audio_device.h" +#include "api/audio/audio_device_defines.h" #include "api/environment/environment.h" -#include "api/environment/environment_factory.h" #include "sdk/android/generated_java_audio_jni/JavaAudioDeviceModule_jni.h" +#include "sdk/android/native_api/jni/java_types.h" +#include "sdk/android/src/jni/audio_device/audio_common.h" +#include "sdk/android/src/jni/audio_device/audio_device_module.h" #include "sdk/android/src/jni/audio_device/audio_record_jni.h" #include "sdk/android/src/jni/audio_device/audio_track_jni.h" -#include "sdk/android/src/jni/jni_helpers.h" #include "third_party/jni_zero/jni_zero.h" namespace webrtc { @@ -27,11 +33,12 @@ static jlong JNI_JavaAudioDeviceModule_CreateAudioDeviceModule( const jni_zero::JavaParamRef& j_audio_manager, const jni_zero::JavaParamRef& j_webrtc_audio_record, const jni_zero::JavaParamRef& j_webrtc_audio_track, + jlong webrtcEnvRef, int input_sample_rate, int output_sample_rate, jboolean j_use_stereo_input, jboolean j_use_stereo_output) { - const Environment webrtc_env = CreateEnvironment(); + const Environment& webrtc_env = *reinterpret_cast(webrtcEnvRef); AudioParameters input_parameters; AudioParameters output_parameters; GetAudioParameters(env, j_context, j_audio_manager, input_sample_rate, @@ -41,8 +48,8 @@ static jlong JNI_JavaAudioDeviceModule_CreateAudioDeviceModule( auto audio_input = std::make_unique( env, input_parameters, kHighLatencyModeDelayEstimateInMilliseconds, j_webrtc_audio_record); - auto audio_output = std::make_unique(env, output_parameters, - j_webrtc_audio_track); + auto audio_output = std::make_unique( + env, webrtc_env, output_parameters, j_webrtc_audio_track); return NativeToJavaPointer( CreateAudioDeviceModuleFromInputAndOutput( webrtc_env, AudioDeviceModule::kAndroidJavaAudio, j_use_stereo_input, From 4f2ffb40461e5a4ff612864f2f370c72315ad586 Mon Sep 17 00:00:00 2001 From: Jonas Oreland Date: Tue, 20 May 2025 08:31:21 +0200 Subject: [PATCH 214/292] WebRTC/PQC - Fix copy of CryptoOptions Remove home-grown copy constructor of CryptoOptions (that I failed to update!) and use compiler generated instead. BUG=webrtc:404763475 No-Iwyu: Doesn't seem to understand the #ifdef surrounding open/boring ssl Change-Id: Ia3e8e251760ab46c585076dfe9e74fbda98ea420 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392560 Reviewed-by: Tomas Gunnarsson Reviewed-by: Per Kjellander Commit-Queue: Jonas Oreland Cr-Commit-Position: refs/heads/main@{#44686} --- api/crypto/crypto_options.cc | 7 ------- api/crypto/crypto_options.h | 2 -- api/crypto/crypto_options_unittest.cc | 11 +++++++++++ 3 files changed, 11 insertions(+), 9 deletions(-) diff --git a/api/crypto/crypto_options.cc b/api/crypto/crypto_options.cc index 00c86d57ff..249bc34105 100644 --- a/api/crypto/crypto_options.cc +++ b/api/crypto/crypto_options.cc @@ -23,13 +23,6 @@ namespace webrtc { CryptoOptions::CryptoOptions() {} -CryptoOptions::CryptoOptions(const CryptoOptions& other) { - srtp = other.srtp; - sframe = other.sframe; -} - -CryptoOptions::~CryptoOptions() {} - // static CryptoOptions CryptoOptions::NoGcm() { CryptoOptions options; diff --git a/api/crypto/crypto_options.h b/api/crypto/crypto_options.h index 8b8dc08f59..8adfe4eb36 100644 --- a/api/crypto/crypto_options.h +++ b/api/crypto/crypto_options.h @@ -27,8 +27,6 @@ namespace webrtc { // and are only applicable to native use cases of WebRTC. struct RTC_EXPORT CryptoOptions { CryptoOptions(); - CryptoOptions(const CryptoOptions& other); - ~CryptoOptions(); // Helper method to return an instance of the CryptoOptions with GCM crypto // suites disabled. This method should be used instead of depending on current diff --git a/api/crypto/crypto_options_unittest.cc b/api/crypto/crypto_options_unittest.cc index d7d59ca7b4..6abc806c3a 100644 --- a/api/crypto/crypto_options_unittest.cc +++ b/api/crypto/crypto_options_unittest.cc @@ -128,3 +128,14 @@ TEST(EphemeralKeyExchangeCipherGroupsTest, Update) { groups.Update(&field_trials, &disable); EXPECT_EQ(groups.GetEnabled(), expected); } + +TEST(EphemeralKeyExchangeCipherGroupsTest, CopyCryptoOptions) { + webrtc::CryptoOptions options; + options.ephemeral_key_exchange_cipher_groups.SetEnabled({ + webrtc::CryptoOptions::EphemeralKeyExchangeCipherGroups::kX25519_MLKEM768, + }); + webrtc::CryptoOptions copy1 = options; + webrtc::CryptoOptions copy2(options); + EXPECT_EQ(options, copy1); + EXPECT_EQ(options, copy2); +} From accbf4d938ac44f4797012c020382463d8587d13 Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Mon, 5 May 2025 16:00:00 +0200 Subject: [PATCH 215/292] Delete java PeerConnectionFactory.fieldTrialsFindFullName MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Quering global field trials is deprecated. Bug: webrtc:42220378 Change-Id: I34f374f2fa0bfb46747b131ce22a1b1118dab593 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389601 Reviewed-by: Zoé Lepaul Commit-Queue: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#44687} --- sdk/android/api/org/webrtc/PeerConnectionFactory.java | 11 ----------- sdk/android/src/jni/pc/peer_connection_factory.cc | 8 -------- 2 files changed, 19 deletions(-) diff --git a/sdk/android/api/org/webrtc/PeerConnectionFactory.java b/sdk/android/api/org/webrtc/PeerConnectionFactory.java index eac3bb02a0..7523dcadfb 100644 --- a/sdk/android/api/org/webrtc/PeerConnectionFactory.java +++ b/sdk/android/api/org/webrtc/PeerConnectionFactory.java @@ -347,16 +347,6 @@ public static void initializeFieldTrials(String fieldTrialsInitString) { nativeInitializeFieldTrials(fieldTrialsInitString); } - // Wrapper of webrtc::field_trial::FindFullName. Develop the feature with default behaviour off. - // Example usage: - // if (PeerConnectionFactory.fieldTrialsFindFullName("WebRTCExperiment").equals("Enabled")) { - // method1(); - // } else { - // method2(); - // } - public static String fieldTrialsFindFullName(String name) { - return NativeLibrary.isLoaded() ? nativeFindFieldTrialsFullName(name) : ""; - } // Start/stop internal capturing of internal tracing. public static boolean startInternalTracingCapture(String tracingFilename) { return nativeStartInternalTracingCapture(tracingFilename); @@ -604,7 +594,6 @@ private void onSignalingThreadReady() { // (for example, at application startup time). private static native void nativeInitializeAndroidGlobals(); private static native void nativeInitializeFieldTrials(String fieldTrialsInitString); - private static native String nativeFindFieldTrialsFullName(String name); private static native void nativeInitializeInternalTracer(); // Internal tracing shutdown, called to prevent resource leaks. Must be called after // PeerConnectionFactory is gone to prevent races with code performing tracing. diff --git a/sdk/android/src/jni/pc/peer_connection_factory.cc b/sdk/android/src/jni/pc/peer_connection_factory.cc index fbab65f66e..d34f8c0d4d 100644 --- a/sdk/android/src/jni/pc/peer_connection_factory.cc +++ b/sdk/android/src/jni/pc/peer_connection_factory.cc @@ -219,14 +219,6 @@ static void JNI_PeerConnectionFactory_InitializeInternalTracer(JNIEnv* jni) { tracing::SetupInternalTracer(); } -static jni_zero::ScopedJavaLocalRef -JNI_PeerConnectionFactory_FindFieldTrialsFullName( - JNIEnv* jni, - const jni_zero::JavaParamRef& j_name) { - return NativeToJavaString( - jni, field_trial::FindFullName(JavaToStdString(jni, j_name))); -} - static jboolean JNI_PeerConnectionFactory_StartInternalTracingCapture( JNIEnv* jni, const jni_zero::JavaParamRef& j_event_tracing_filename) { From 9368fa72980cc376e41835691b398fd5616f71be Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Mon, 19 May 2025 14:04:43 +0000 Subject: [PATCH 216/292] [RtcEventLogVisuzlizer] Propagate field trials without global string MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Bug: webrtc:42225026 Change-Id: I582e0a6c82a8dd2fe58e88947a173f0d355d41de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392500 Reviewed-by: Björn Terelius Commit-Queue: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#44688} --- rtc_tools/BUILD.gn | 9 +++--- .../rtc_event_log_visualizer/analyze_audio.cc | 18 ++++++----- .../rtc_event_log_visualizer/analyze_audio.h | 6 ++-- .../rtc_event_log_visualizer/analyzer.cc | 24 +++++++------- rtc_tools/rtc_event_log_visualizer/analyzer.h | 7 ++-- .../analyzer_bindings.cc | 4 ++- .../log_simulation.cc | 10 ++++-- .../rtc_event_log_visualizer/log_simulation.h | 5 +-- rtc_tools/rtc_event_log_visualizer/main.cc | 32 ++++++++++++------- 9 files changed, 70 insertions(+), 45 deletions(-) diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn index 052ffd9e39..c853f3408b 100644 --- a/rtc_tools/BUILD.gn +++ b/rtc_tools/BUILD.gn @@ -370,6 +370,7 @@ if (!build_with_chromium) { ":chart_proto", "../api:candidate", "../api:dtls_transport_interface", + "../api:field_trials", "../api:function_view", "../api:make_ref_counted", "../api:rtp_headers", @@ -383,7 +384,6 @@ if (!build_with_chromium) { "../api/rtc_event_log:rtc_event_log", "../api/transport:bandwidth_usage", "../api/transport:ecn_marking", - "../api/transport:field_trial_based_config", "../api/transport:goog_cc", "../api/transport:network_control", "../api/units:data_rate", @@ -421,8 +421,6 @@ if (!build_with_chromium) { "../rtc_base:stringutils", "../rtc_base/network:sent_packet", "../system_wrappers", - "../system_wrappers:field_trial", - "../test:explicit_key_value_config", "//third_party/abseil-cpp/absl/algorithm:container", "//third_party/abseil-cpp/absl/base:core_headers", "//third_party/abseil-cpp/absl/functional:bind_front", @@ -440,6 +438,7 @@ if (!build_with_chromium) { deps = [ ":chart_proto", ":event_log_visualizer_utils", + "../api/environment:environment_factory", "//api/units:time_delta", "//logging:rtc_event_log_parser", "//rtc_base:protobuf_utils", @@ -518,6 +517,9 @@ if (rtc_include_tests) { deps = [ ":chart_proto", ":event_log_visualizer_utils", + "../api:field_trials", + "../api/environment", + "../api/environment:environment_factory", "../api/neteq:neteq_api", "../api/rtc_event_log", "../api/units:time_delta", @@ -527,7 +529,6 @@ if (rtc_include_tests) { "../rtc_base:checks", "../rtc_base:logging", "../rtc_base:protobuf_utils", - "../system_wrappers:field_trial", "//third_party/abseil-cpp/absl/algorithm:container", "//third_party/abseil-cpp/absl/flags:config", "//third_party/abseil-cpp/absl/flags:flag", diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc b/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc index 7a34309d53..ed3629a787 100644 --- a/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc +++ b/rtc_tools/rtc_event_log_visualizer/analyze_audio.cc @@ -44,7 +44,6 @@ #include "rtc_base/checks.h" #include "rtc_tools/rtc_event_log_visualizer/analyzer_common.h" #include "rtc_tools/rtc_event_log_visualizer/plot_base.h" -#include "system_wrappers/include/field_trial.h" namespace webrtc { @@ -242,8 +241,9 @@ class ReplacementAudioDecoderFactory : public AudioDecoderFactory { std::unique_ptr CreateNetEqTestAndRun( const ParsedRtcEventLog& parsed_log, uint32_t ssrc, - const std::string& replacement_file_name, - int file_sample_rate_hz) { + absl::string_view replacement_file_name, + int file_sample_rate_hz, + absl::string_view field_trials) { std::unique_ptr input = test::CreateNetEqEventLogInput(parsed_log, ssrc); if (!input) { @@ -278,7 +278,7 @@ std::unique_ptr CreateNetEqTestAndRun( NetEq::Config config; test::NetEqTest test(config, decoder_factory, codecs, /*text_log=*/nullptr, /*factory=*/nullptr, std::move(input), std::move(output), - callbacks, field_trial::GetFieldTrialString()); + callbacks, field_trials); test.Run(); return neteq_stats_getter; } @@ -286,12 +286,14 @@ std::unique_ptr CreateNetEqTestAndRun( NetEqStatsGetterMap SimulateNetEq(const ParsedRtcEventLog& parsed_log, const AnalyzerConfig& config, - const std::string& replacement_file_name, - int file_sample_rate_hz) { + absl::string_view replacement_file_name, + int file_sample_rate_hz, + absl::string_view field_trials) { NetEqStatsGetterMap neteq_stats; for (uint32_t ssrc : parsed_log.incoming_audio_ssrcs()) { - std::unique_ptr stats = CreateNetEqTestAndRun( - parsed_log, ssrc, replacement_file_name, file_sample_rate_hz); + std::unique_ptr stats = + CreateNetEqTestAndRun(parsed_log, ssrc, replacement_file_name, + file_sample_rate_hz, field_trials); if (stats) { neteq_stats[ssrc] = std::move(stats); } diff --git a/rtc_tools/rtc_event_log_visualizer/analyze_audio.h b/rtc_tools/rtc_event_log_visualizer/analyze_audio.h index c5132f25a8..094f9d91b6 100644 --- a/rtc_tools/rtc_event_log_visualizer/analyze_audio.h +++ b/rtc_tools/rtc_event_log_visualizer/analyze_audio.h @@ -16,6 +16,7 @@ #include #include +#include "absl/strings/string_view.h" #include "api/function_view.h" #include "api/neteq/neteq.h" #include "logging/rtc_event_log/rtc_event_log_parser.h" @@ -48,8 +49,9 @@ using NetEqStatsGetterMap = std::map>; NetEqStatsGetterMap SimulateNetEq(const ParsedRtcEventLog& parsed_log, const AnalyzerConfig& config, - const std::string& replacement_file_name, - int file_sample_rate_hz); + absl::string_view replacement_file_name, + int file_sample_rate_hz, + absl::string_view field_trials); void CreateAudioJitterBufferGraph(const ParsedRtcEventLog& parsed_log, const AnalyzerConfig& config, diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer.cc b/rtc_tools/rtc_event_log_visualizer/analyzer.cc index 544f144cc1..f1afcc4b90 100644 --- a/rtc_tools/rtc_event_log_visualizer/analyzer.cc +++ b/rtc_tools/rtc_event_log_visualizer/analyzer.cc @@ -30,10 +30,11 @@ #include "absl/strings/string_view.h" #include "api/candidate.h" #include "api/dtls_transport_interface.h" +#include "api/environment/environment.h" #include "api/environment/environment_factory.h" +#include "api/field_trials.h" #include "api/function_view.h" #include "api/media_types.h" -#include "api/rtc_event_log/rtc_event_log.h" #include "api/rtp_headers.h" #include "api/transport/bandwidth_usage.h" #include "api/transport/ecn_marking.h" @@ -77,7 +78,6 @@ #include "rtc_tools/rtc_event_log_visualizer/log_simulation.h" #include "rtc_tools/rtc_event_log_visualizer/plot_base.h" #include "system_wrappers/include/clock.h" -#include "test/explicit_key_value_config.h" namespace webrtc { @@ -471,7 +471,7 @@ std::map BuildCandidateIdLogDescriptionMap( EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log, bool normalize_time) - : parsed_log_(log) { + : env_(CreateEnvironment()), parsed_log_(log) { config_.window_duration_ = TimeDelta::Millis(250); config_.step_ = TimeDelta::Millis(10); if (!log.start_log_events().empty()) { @@ -493,9 +493,10 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log, << " seconds long."; } -EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log, +EventLogAnalyzer::EventLogAnalyzer(const Environment& env, + const ParsedRtcEventLog& log, const AnalyzerConfig& config) - : parsed_log_(log), config_(config) { + : env_(env), parsed_log_(log), config_(config) { RTC_LOG(LS_INFO) << "Log is " << (parsed_log_.last_timestamp().ms() - parsed_log_.first_timestamp().ms()) / @@ -1533,7 +1534,7 @@ void EventLogAnalyzer::CreateGoogCcSimulationGraph(Plot* plot) const { PointStyle::kHighlight); LogBasedNetworkControllerSimulation simulation( - std::make_unique(), + env_, std::make_unique(), [&](const NetworkControlUpdate& update, Timestamp at_time) { if (update.target_rate) { target_rates.points.emplace_back( @@ -1762,13 +1763,12 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) const { SimulatedClock clock(0); BitrateObserver observer; - RtcEventLogNull null_event_log; TransportFeedbackAdapter transport_feedback; auto factory = GoogCcNetworkControllerFactory(); TimeDelta process_interval = factory.GetProcessInterval(); // TODO(holmer): Log the call config and use that here instead. static const uint32_t kDefaultStartBitrateBps = 300000; - NetworkControllerConfig cc_config(CreateEnvironment(&null_event_log)); + NetworkControllerConfig cc_config(env_); cc_config.constraints.at_time = Timestamp::Micros(clock.TimeInMicroseconds()); cc_config.constraints.starting_rate = DataRate::BitsPerSec(kDefaultStartBitrateBps); @@ -1809,12 +1809,12 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) const { }; RateStatistics raw_acked_bitrate(750, 8000); - test::ExplicitKeyValueConfig throughput_config( + FieldTrials throughput_config( "WebRTC-Bwe-RobustThroughputEstimatorSettings/enabled:true/"); std::unique_ptr robust_throughput_estimator( AcknowledgedBitrateEstimatorInterface::Create(&throughput_config)); - test::ExplicitKeyValueConfig acked_bitrate_config( + FieldTrials acked_bitrate_config( "WebRTC-Bwe-RobustThroughputEstimatorSettings/enabled:false/"); std::unique_ptr acknowledged_bitrate_estimator( @@ -1968,9 +1968,11 @@ void EventLogAnalyzer::CreateReceiveSideBweSimulationGraph(Plot* plot) const { } SimulatedClock clock(0); + EnvironmentFactory env_factory(env_); + env_factory.Set(&clock); RembInterceptor remb_interceptor; ReceiveSideCongestionController rscc( - CreateEnvironment(&clock), [](auto...) {}, + env_factory.Create(), [](auto...) {}, absl::bind_front(&RembInterceptor::SendRemb, &remb_interceptor)); // TODO(holmer): Log the call config and use that here instead. // static const uint32_t kDefaultStartBitrateBps = 300000; diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer.h b/rtc_tools/rtc_event_log_visualizer/analyzer.h index 5bb0840728..134a45fa63 100644 --- a/rtc_tools/rtc_event_log_visualizer/analyzer.h +++ b/rtc_tools/rtc_event_log_visualizer/analyzer.h @@ -14,10 +14,10 @@ #include #include #include -#include #include #include +#include "api/environment/environment.h" #include "api/function_view.h" #include "logging/rtc_event_log/rtc_event_log_parser.h" #include "modules/rtp_rtcp/source/rtcp_packet/report_block.h" @@ -62,7 +62,9 @@ class EventLogAnalyzer { // duration of its lifetime. The ParsedRtcEventLogNew must not be destroyed or // modified while the EventLogAnalyzer is being used. EventLogAnalyzer(const ParsedRtcEventLog& log, bool normalize_time); - EventLogAnalyzer(const ParsedRtcEventLog& log, const AnalyzerConfig& config); + EventLogAnalyzer(const Environment& env, + const ParsedRtcEventLog& log, + const AnalyzerConfig& config); void CreateGraphsByName(const std::vector& names, PlotCollection* collection) const; @@ -148,6 +150,7 @@ class EventLogAnalyzer { const std::string& label) const; void CreateEcnFeedbackGraph(Plot* plot, PacketDirection direction) const; + const Environment env_; const ParsedRtcEventLog& parsed_log_; // A list of SSRCs we are interested in analysing. diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer_bindings.cc b/rtc_tools/rtc_event_log_visualizer/analyzer_bindings.cc index 254a76387d..a50e0fbf53 100644 --- a/rtc_tools/rtc_event_log_visualizer/analyzer_bindings.cc +++ b/rtc_tools/rtc_event_log_visualizer/analyzer_bindings.cc @@ -18,6 +18,7 @@ #include #include "absl/strings/string_view.h" +#include "api/environment/environment_factory.h" #include "api/units/time_delta.h" #include "logging/rtc_event_log/rtc_event_log_parser.h" #include "rtc_base/numerics/safe_conversions.h" @@ -82,7 +83,8 @@ void analyze_rtc_event_log(const char* log_contents, return; } - webrtc::EventLogAnalyzer analyzer(parsed_log, config); + webrtc::EventLogAnalyzer analyzer(webrtc::CreateEnvironment(), parsed_log, + config); analyzer.InitializeMapOfNamedGraphs(/*show_detector_state=*/false, /*show_alr_state=*/false, /*show_link_capacity=*/false); diff --git a/rtc_tools/rtc_event_log_visualizer/log_simulation.cc b/rtc_tools/rtc_event_log_visualizer/log_simulation.cc index 47bdb76bbb..82f340632e 100644 --- a/rtc_tools/rtc_event_log_visualizer/log_simulation.cc +++ b/rtc_tools/rtc_event_log_visualizer/log_simulation.cc @@ -16,7 +16,7 @@ #include #include -#include "api/environment/environment_factory.h" +#include "api/environment/environment.h" #include "api/transport/network_control.h" #include "api/transport/network_types.h" #include "api/units/data_rate.h" @@ -30,15 +30,19 @@ #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/ntp_time_util.h" #include "modules/rtp_rtcp/source/rtp_packet_to_send.h" +#include "rtc_base/checks.h" #include "rtc_base/network/sent_packet.h" #include "system_wrappers/include/clock.h" namespace webrtc { LogBasedNetworkControllerSimulation::LogBasedNetworkControllerSimulation( + const Environment& env, std::unique_ptr factory, std::function update_handler) - : update_handler_(update_handler), factory_(std::move(factory)) {} + : env_(env), + update_handler_(update_handler), + factory_(std::move(factory)) {} LogBasedNetworkControllerSimulation::~LogBasedNetworkControllerSimulation() {} @@ -49,7 +53,7 @@ void LogBasedNetworkControllerSimulation::HandleStateUpdate( void LogBasedNetworkControllerSimulation::ProcessUntil(Timestamp to_time) { if (last_process_.IsInfinite()) { - NetworkControllerConfig config(CreateEnvironment(&null_event_log_)); + NetworkControllerConfig config(env_); config.constraints.at_time = to_time; config.constraints.min_data_rate = DataRate::KilobitsPerSec(30); config.constraints.starting_rate = DataRate::KilobitsPerSec(300); diff --git a/rtc_tools/rtc_event_log_visualizer/log_simulation.h b/rtc_tools/rtc_event_log_visualizer/log_simulation.h index dbb0a1c798..0c0f8959b6 100644 --- a/rtc_tools/rtc_event_log_visualizer/log_simulation.h +++ b/rtc_tools/rtc_event_log_visualizer/log_simulation.h @@ -17,7 +17,7 @@ #include #include -#include "api/rtc_event_log/rtc_event_log.h" +#include "api/environment/environment.h" #include "api/transport/network_control.h" #include "api/transport/network_types.h" #include "api/units/timestamp.h" @@ -33,6 +33,7 @@ namespace webrtc { class LogBasedNetworkControllerSimulation { public: explicit LogBasedNetworkControllerSimulation( + const Environment& env, std::unique_ptr factory, std::function update_handler); @@ -53,8 +54,8 @@ class LogBasedNetworkControllerSimulation { void OnFeedback(const LoggedRtcpPacketTransportFeedback& feedback); void OnReceiverReport(const LoggedRtcpPacketReceiverReport& report); void OnIceConfig(const LoggedIceCandidatePairConfig& candidate); - RtcEventLogNull null_event_log_; + const Environment env_; const std::function update_handler_; std::unique_ptr factory_; diff --git a/rtc_tools/rtc_event_log_visualizer/main.cc b/rtc_tools/rtc_event_log_visualizer/main.cc index 466203baf2..a410985040 100644 --- a/rtc_tools/rtc_event_log_visualizer/main.cc +++ b/rtc_tools/rtc_event_log_visualizer/main.cc @@ -28,6 +28,9 @@ #include "absl/flags/usage_config.h" #include "absl/strings/match.h" #include "absl/strings/string_view.h" +#include "api/environment/environment.h" +#include "api/environment/environment_factory.h" +#include "api/field_trials.h" #include "api/neteq/neteq.h" #include "api/units/time_delta.h" #include "logging/rtc_event_log/rtc_event_log_parser.h" @@ -40,7 +43,6 @@ #include "rtc_tools/rtc_event_log_visualizer/conversational_speech_en.h" #include "rtc_tools/rtc_event_log_visualizer/plot_base.h" #include "rtc_tools/rtc_event_log_visualizer/proto/chart.pb.h" -#include "system_wrappers/include/field_trial.h" ABSL_FLAG(std::string, plot, @@ -157,10 +159,9 @@ int main(int argc, char* argv[]) { } webrtc::LogMessage::SetLogToStderr(true); - // InitFieldTrialsFromString stores the char*, so the char array must outlive - // the application. const std::string field_trials = absl::GetFlag(FLAGS_force_fieldtrials); - webrtc::field_trial::InitFieldTrialsFromString(field_trials.c_str()); + webrtc::Environment env = webrtc::CreateEnvironment( + std::make_unique(field_trials)); webrtc::ParsedRtcEventLog::UnconfiguredHeaderExtensions header_extensions = webrtc::ParsedRtcEventLog::UnconfiguredHeaderExtensions::kDontParse; @@ -213,7 +214,7 @@ int main(int argc, char* argv[]) { has_generated_wav_file = true; } - webrtc::EventLogAnalyzer analyzer(parsed_log, config); + webrtc::EventLogAnalyzer analyzer(env, parsed_log, config); analyzer.InitializeMapOfNamedGraphs(absl::GetFlag(FLAGS_show_detector_state), absl::GetFlag(FLAGS_show_alr_state), absl::GetFlag(FLAGS_show_link_capacity)); @@ -329,7 +330,8 @@ int main(int argc, char* argv[]) { if (absl::c_find(plot_names, "simulated_neteq_expand_rate") != plot_names.end()) { if (!neteq_stats) { - neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000); + neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000, + field_trials); } webrtc::CreateNetEqNetworkStatsGraph( parsed_log, config, *neteq_stats, @@ -341,7 +343,8 @@ int main(int argc, char* argv[]) { if (absl::c_find(plot_names, "simulated_neteq_speech_expand_rate") != plot_names.end()) { if (!neteq_stats) { - neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000); + neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000, + field_trials); } webrtc::CreateNetEqNetworkStatsGraph( parsed_log, config, *neteq_stats, @@ -354,7 +357,8 @@ int main(int argc, char* argv[]) { if (absl::c_find(plot_names, "simulated_neteq_accelerate_rate") != plot_names.end()) { if (!neteq_stats) { - neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000); + neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000, + field_trials); } webrtc::CreateNetEqNetworkStatsGraph( parsed_log, config, *neteq_stats, @@ -367,7 +371,8 @@ int main(int argc, char* argv[]) { if (absl::c_find(plot_names, "simulated_neteq_preemptive_rate") != plot_names.end()) { if (!neteq_stats) { - neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000); + neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000, + field_trials); } webrtc::CreateNetEqNetworkStatsGraph( parsed_log, config, *neteq_stats, @@ -380,7 +385,8 @@ int main(int argc, char* argv[]) { if (absl::c_find(plot_names, "simulated_neteq_concealment_events") != plot_names.end()) { if (!neteq_stats) { - neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000); + neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000, + field_trials); } webrtc::CreateNetEqLifetimeStatsGraph( parsed_log, config, *neteq_stats, @@ -393,7 +399,8 @@ int main(int argc, char* argv[]) { if (absl::c_find(plot_names, "simulated_neteq_preferred_buffer_size") != plot_names.end()) { if (!neteq_stats) { - neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000); + neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000, + field_trials); } webrtc::CreateNetEqNetworkStatsGraph( parsed_log, config, *neteq_stats, @@ -412,7 +419,8 @@ int main(int argc, char* argv[]) { if (absl::c_find(plot_names, "simulated_neteq_jitter_buffer_delay") != plot_names.end()) { if (!neteq_stats) { - neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000); + neteq_stats = webrtc::SimulateNetEq(parsed_log, config, wav_path, 48000, + field_trials); } for (auto it = neteq_stats->cbegin(); it != neteq_stats->cend(); ++it) { webrtc::CreateAudioJitterBufferGraph( From bd6bade49db8c1071ca7a2e26dbf5a3ededa7769 Mon Sep 17 00:00:00 2001 From: Jeremy Leconte Date: Tue, 20 May 2025 10:13:20 +0200 Subject: [PATCH 217/292] Temporarly disable fyi android tests bot from LKGR. It seems that increasing the number of shards has increase the flakiness rate. Change-Id: I9eb69455e1d954d54565a9b779af5ae354407281 Bug: chromium:418717425 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392561 Commit-Queue: Jeremy Leconte Reviewed-by: Ilya Nikolaevskiy Cr-Commit-Position: refs/heads/main@{#44689} --- infra/config/config.star | 2 +- infra/config/cr-buildbucket.cfg | 1 - 2 files changed, 1 insertion(+), 2 deletions(-) diff --git a/infra/config/config.star b/infra/config/config.star index ba5be58866..2ee74ee097 100755 --- a/infra/config/config.star +++ b/infra/config/config.star @@ -904,8 +904,8 @@ lkgr_config = { "WebRTC Chromium FYI Android Builder (dbg)", "WebRTC Chromium FYI Android Builder ARM64 (dbg)", "WebRTC Chromium FYI Android Builder", - "WebRTC Chromium FYI Android Tests (dbg)", # TODO: b/418717425 - Reenable when flakiness is fixed. + # "WebRTC Chromium FYI Android Tests (dbg)", # "WebRTC Chromium FYI Android Tests ARM64 (dbg)", "WebRTC Chromium FYI Linux Builder (dbg)", "WebRTC Chromium FYI Linux Builder", diff --git a/infra/config/cr-buildbucket.cfg b/infra/config/cr-buildbucket.cfg index cc571f3a7a..8866e25b77 100644 --- a/infra/config/cr-buildbucket.cfg +++ b/infra/config/cr-buildbucket.cfg @@ -2189,7 +2189,6 @@ buckets { ' "WebRTC Chromium FYI Android Builder (dbg)",' ' "WebRTC Chromium FYI Android Builder ARM64 (dbg)",' ' "WebRTC Chromium FYI Android Builder",' - ' "WebRTC Chromium FYI Android Tests (dbg)",' ' "WebRTC Chromium FYI Linux Builder (dbg)",' ' "WebRTC Chromium FYI Linux Builder",' ' "WebRTC Chromium FYI Linux Tester",' From 41eb9cd3c7ec1ca38cc9e4037cc2256125d560bd Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Tue, 20 May 2025 03:03:39 -0700 Subject: [PATCH 218/292] Roll chromium_revision c89ca3267f..cbbfcc5fee (1462567:1462671) Change log: https://chromium.googlesource.com/chromium/src/+log/c89ca3267f..cbbfcc5fee Full diff: https://chromium.googlesource.com/chromium/src/+/c89ca3267f..cbbfcc5fee Changed dependencies * src/build: https://chromium.googlesource.com/chromium/src/build/+log/79fe614714..2263856e4c * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/0c74a70e1f..d4dbffa976 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/c25e306ec3..d971856aa6 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/74eaeae48c..8dbab0fa74 * src/third_party/androidx/cipd: 9gK5fctsrqSL7ITHbaKR8oEVCBo_gw0XIMzM9kvuSI0C..eaknFYxsT8k1aKnroa-MOOT-nxE41krXhpS9QxURsp0C * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/198e465646..17ab5def84 * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/5a66f76326..a9b5b095da * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/e9b9249726..043e67c9de DEPS diff: https://chromium.googlesource.com/chromium/src/+/c89ca3267f..cbbfcc5fee/DEPS BUG=None Change-Id: I122196c9a2b450e1e420d518ea17f41e4984f183 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392433 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44690} --- DEPS | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/DEPS b/DEPS index 3fd5d840c2..e41ca9a5c9 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': 'c89ca3267f93966b53d212ada6baf989ca17ed09', + 'chromium_revision': 'cbbfcc5feef0830b8fe17ad5ce545a647e8557a1', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -75,7 +75,7 @@ deps = { 'src/base': 'https://chromium.googlesource.com/chromium/src/base@cf27d977ab48cbc6eaf94ff7499cdf5e7978993f', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@79fe6147149c124bf7b4370aa623db917609486b', + 'https://chromium.googlesource.com/chromium/src/build@2263856e4c6de03a8d6cfddebb4e5ef573af2d2e', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@cd169314af80e77a75eb25e2c0df8798f35bc04a', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@0c74a70e1f0e73fd0f5bd75470faf83981e43b6b', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@d4dbffa976fbad72c3e3253982f4757e693652af', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@c25e306ec3d55d5b3189908e3a35cdfafd8a474e', + 'https://chromium.googlesource.com/chromium/src/testing@d971856aa6d7475b3a39e76d96d7d3afc8abf133', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@74eaeae48c7c505a8dfe393eb0e16f3d82e0b881', + 'https://chromium.googlesource.com/chromium/src/third_party@8dbab0fa7419236ec3c45bad9677ce5ca734b907', 'src/buildtools/linux64': { 'packages': [ @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@198e465646b36a6106bbc93d26c80b3850791966', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@17ab5def84090ca0d2da4c1cdd1286283be1404e', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '5a66f76326a04bf33f05433665cd77b7eb288fcc', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'a9b5b095dac6ee69731e25683949fb181a57fa24', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@e9b9249726464abbfc9525625dfeca2ad0064e59', + 'https://chromium.googlesource.com/chromium/src/tools@043e67c9de12e6bceffae48a3e3cdb56245bd726', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': '9gK5fctsrqSL7ITHbaKR8oEVCBo_gw0XIMzM9kvuSI0C', + 'version': 'eaknFYxsT8k1aKnroa-MOOT-nxE41krXhpS9QxURsp0C', }, ], 'condition': 'checkout_android and non_git_source', From 2a872603a2031842d676cd00868993fdcfe1ceec Mon Sep 17 00:00:00 2001 From: Gustaf Ullberg Date: Mon, 19 May 2025 08:56:29 +0200 Subject: [PATCH 219/292] Use sos notation over zpk notation for decimator anti-aliasing filters MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This CL also removes the unused downsampling factor 2 from the decimator and changes the formatting of filter coefficients. Bug: webrtc:417439718 Change-Id: If62a71bd4975b091a5bdb759aba0b066838a4d94 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392100 Auto-Submit: Gustaf Ullberg Reviewed-by: Per Åhgren Commit-Queue: Per Åhgren Cr-Commit-Position: refs/heads/main@{#44691} --- .../aec3/adaptive_fir_filter_unittest.cc | 5 +- modules/audio_processing/aec3/decimator.cc | 86 +++++++++---------- .../aec3/decimator_unittest.cc | 4 +- .../echo_path_delay_estimator_unittest.cc | 9 +- .../aec3/matched_filter_unittest.cc | 2 +- .../aec3/render_delay_controller_unittest.cc | 11 ++- .../aec3/subtractor_unittest.cc | 5 +- modules/audio_processing/high_pass_filter.cc | 21 +++-- .../utility/cascaded_biquad_filter.cc | 46 ---------- .../utility/cascaded_biquad_filter.h | 16 ---- .../cascaded_biquad_filter_unittest.cc | 62 +++---------- 11 files changed, 92 insertions(+), 175 deletions(-) diff --git a/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc b/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc index 387a0744c1..d229abea32 100644 --- a/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc +++ b/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc @@ -513,8 +513,9 @@ TEST_P(AdaptiveFirFilterMultiChannel, FilterAndAdapt) { std::array E2_coarse; // [B,A] = butter(2,100/8000,'high') constexpr std::array - kHighPassFilterCoefficients = { - {{{0.97261f, -1.94523f, 0.97261f}, {-1.94448f, 0.94598f}}}}; + kHighPassFilterCoefficients = {{ + {{0.97261f, -1.94523f, 0.97261f}, {-1.94448f, 0.94598f}}, + }}; for (auto& Y2_ch : Y2) { Y2_ch.fill(0.f); } diff --git a/modules/audio_processing/aec3/decimator.cc b/modules/audio_processing/aec3/decimator.cc index beac73e22e..0ac9be7b5c 100644 --- a/modules/audio_processing/aec3/decimator.cc +++ b/modules/audio_processing/aec3/decimator.cc @@ -10,63 +10,61 @@ #include "modules/audio_processing/aec3/decimator.h" #include -#include +#include +#include "api/array_view.h" #include "modules/audio_processing/aec3/aec3_common.h" +#include "modules/audio_processing/utility/cascaded_biquad_filter.h" #include "rtc_base/checks.h" namespace webrtc { namespace { +// signal.ellip(6, 1, 40, 1800/8000, 'lowpass', output='sos') +constexpr std::array + kLowPassFilterDs4 = {{ + {{0.0180919877f, 0.00320961363f, 0.0180919877f}, + {-1.5183195f, 0.633165865f}}, + {{1.0f, -1.24550459f, 1.0f}, {-1.49784254f, 0.853586692f}}, + {{1.0f, -1.4221681f, 1.0f}, {-1.49791282f, 0.969572384f}}, + }}; -// signal.butter(2, 3400/8000.0, 'lowpass', analog=False) -const std::vector GetLowPassFilterDS2() { - return std::vector{ - {{-1.f, 0.f}, {0.13833231f, 0.40743176f}, 0.22711796393486466f}, - {{-1.f, 0.f}, {0.13833231f, 0.40743176f}, 0.22711796393486466f}, - {{-1.f, 0.f}, {0.13833231f, 0.40743176f}, 0.22711796393486466f}}; -} - -// signal.ellip(6, 1, 40, 1800/8000, btype='lowpass', analog=False) -const std::vector GetLowPassFilterDS4() { - return std::vector{ - {{-0.08873842f, 0.99605496f}, {0.75916227f, 0.23841065f}, 0.26250696827f}, - {{0.62273832f, 0.78243018f}, {0.74892112f, 0.5410152f}, 0.26250696827f}, - {{0.71107693f, 0.70311421f}, {0.74895534f, 0.63924616f}, 0.26250696827f}}; -} +// signal.cheby1(1, 6, [1000/8000, 2000/8000], 'bandpass', output='sos') +// repeated 5 times. +constexpr std::array + kBandPassFilterDs8 = {{ + {{0.103304783f, 0.0f, -0.103304783f}, {-1.520363f, 0.793390435f}}, + {{0.103304783f, 0.0f, -0.103304783f}, {-1.520363f, 0.793390435f}}, + {{0.103304783f, 0.0f, -0.103304783f}, {-1.520363f, 0.793390435f}}, + {{0.103304783f, 0.0f, -0.103304783f}, {-1.520363f, 0.793390435f}}, + {{0.103304783f, 0.0f, -0.103304783f}, {-1.520363f, 0.793390435f}}, + }}; -// signal.cheby1(1, 6, [1000/8000, 2000/8000], btype='bandpass', analog=False) -const std::vector GetBandPassFilterDS8() { - return std::vector{ - {{1.f, 0.f}, {0.7601815f, 0.46423542f}, 0.10330478266505948f, true}, - {{1.f, 0.f}, {0.7601815f, 0.46423542f}, 0.10330478266505948f, true}, - {{1.f, 0.f}, {0.7601815f, 0.46423542f}, 0.10330478266505948f, true}, - {{1.f, 0.f}, {0.7601815f, 0.46423542f}, 0.10330478266505948f, true}, - {{1.f, 0.f}, {0.7601815f, 0.46423542f}, 0.10330478266505948f, true}}; -} +// signal.butter(2, 1000/8000.0, 'highpass', output='sos') +constexpr std::array + kHighPassFilter = {{ + {{0.757076375f, -1.51415275f, 0.757076375f}, + {-1.45424359f, 0.574061915f}}, + }}; -// signal.butter(2, 1000/8000.0, 'highpass', analog=False) -const std::vector GetHighPassFilter() { - return std::vector{ - {{1.f, 0.f}, {0.72712179f, 0.21296904f}, 0.7570763753338849f}}; -} - -const std::vector GetPassThroughFilter() { - return std::vector{}; -} +constexpr std::array + kPassThroughFilter = {{}}; } // namespace Decimator::Decimator(size_t down_sampling_factor) : down_sampling_factor_(down_sampling_factor), - anti_aliasing_filter_(down_sampling_factor_ == 4 - ? GetLowPassFilterDS4() - : (down_sampling_factor_ == 8 - ? GetBandPassFilterDS8() - : GetLowPassFilterDS2())), - noise_reduction_filter_(down_sampling_factor_ == 8 - ? GetPassThroughFilter() - : GetHighPassFilter()) { - RTC_DCHECK(down_sampling_factor_ == 2 || down_sampling_factor_ == 4 || - down_sampling_factor_ == 8); + anti_aliasing_filter_( + down_sampling_factor_ == 4 + ? ArrayView( + kLowPassFilterDs4) + : ArrayView( + kBandPassFilterDs8)), + noise_reduction_filter_( + down_sampling_factor_ == 8 + ? (ArrayView( + kPassThroughFilter)) + : (ArrayView( + kHighPassFilter))) { + RTC_DCHECK(down_sampling_factor_ == 4 || down_sampling_factor_ == 8); } void Decimator::Decimate(ArrayView in, ArrayView out) { diff --git a/modules/audio_processing/aec3/decimator_unittest.cc b/modules/audio_processing/aec3/decimator_unittest.cc index 2c6d96de6d..69785c12b0 100644 --- a/modules/audio_processing/aec3/decimator_unittest.cc +++ b/modules/audio_processing/aec3/decimator_unittest.cc @@ -21,7 +21,9 @@ #include #include +#include "api/array_view.h" #include "modules/audio_processing/aec3/aec3_common.h" +#include "rtc_base/checks.h" #include "rtc_base/strings/string_builder.h" #include "test/gtest.h" @@ -35,7 +37,7 @@ std::string ProduceDebugText(int sample_rate_hz) { return ss.Release(); } -constexpr size_t kDownSamplingFactors[] = {2, 4, 8}; +constexpr size_t kDownSamplingFactors[] = {4, 8}; constexpr float kPi = std::numbers::pi_v; constexpr size_t kNumStartupBlocks = 50; constexpr size_t kNumBlocks = 1000; diff --git a/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc b/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc index 8a105041e0..1fbac7642a 100644 --- a/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc +++ b/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc @@ -11,13 +11,20 @@ #include "modules/audio_processing/aec3/echo_path_delay_estimator.h" #include +#include +#include +#include #include +#include #include "api/audio/echo_canceller3_config.h" #include "modules/audio_processing/aec3/aec3_common.h" +#include "modules/audio_processing/aec3/block.h" +#include "modules/audio_processing/aec3/delay_estimate.h" #include "modules/audio_processing/aec3/render_delay_buffer.h" #include "modules/audio_processing/logging/apm_data_dumper.h" #include "modules/audio_processing/test/echo_canceller_test_tools.h" +#include "rtc_base/checks.h" #include "rtc_base/random.h" #include "rtc_base/strings/string_builder.h" #include "test/gtest.h" @@ -74,7 +81,7 @@ TEST(EchoPathDelayEstimator, DelayEstimation) { Block render(kNumBands, kNumRenderChannels); Block capture(/*num_bands=*/1, kNumCaptureChannels); ApmDataDumper data_dumper(0); - constexpr size_t kDownSamplingFactors[] = {2, 4, 8}; + constexpr size_t kDownSamplingFactors[] = {4, 8}; for (auto down_sampling_factor : kDownSamplingFactors) { EchoCanceller3Config config; config.delay.delay_headroom_samples = 0; diff --git a/modules/audio_processing/aec3/matched_filter_unittest.cc b/modules/audio_processing/aec3/matched_filter_unittest.cc index 332f877aad..691bca4b48 100644 --- a/modules/audio_processing/aec3/matched_filter_unittest.cc +++ b/modules/audio_processing/aec3/matched_filter_unittest.cc @@ -52,7 +52,7 @@ std::string ProduceDebugText(size_t delay, size_t down_sampling_factor) { } constexpr size_t kNumMatchedFilters = 10; -constexpr size_t kDownSamplingFactors[] = {2, 4, 8}; +constexpr size_t kDownSamplingFactors[] = {4, 8}; constexpr size_t kWindowSizeSubBlocks = 32; constexpr size_t kAlignmentShiftSubBlocks = kWindowSizeSubBlocks * 3 / 4; diff --git a/modules/audio_processing/aec3/render_delay_controller_unittest.cc b/modules/audio_processing/aec3/render_delay_controller_unittest.cc index a3c308e39c..9caa2c0a25 100644 --- a/modules/audio_processing/aec3/render_delay_controller_unittest.cc +++ b/modules/audio_processing/aec3/render_delay_controller_unittest.cc @@ -11,16 +11,19 @@ #include "modules/audio_processing/aec3/render_delay_controller.h" #include +#include #include +#include #include #include +#include "api/audio/echo_canceller3_config.h" #include "modules/audio_processing/aec3/aec3_common.h" -#include "modules/audio_processing/aec3/block_processor.h" -#include "modules/audio_processing/aec3/decimator.h" +#include "modules/audio_processing/aec3/block.h" +#include "modules/audio_processing/aec3/delay_estimate.h" #include "modules/audio_processing/aec3/render_delay_buffer.h" -#include "modules/audio_processing/logging/apm_data_dumper.h" #include "modules/audio_processing/test/echo_canceller_test_tools.h" +#include "rtc_base/checks.h" #include "rtc_base/random.h" #include "rtc_base/strings/string_builder.h" #include "test/gtest.h" @@ -45,7 +48,7 @@ std::string ProduceDebugText(int sample_rate_hz, return ss.Release(); } -constexpr size_t kDownSamplingFactors[] = {2, 4, 8}; +constexpr size_t kDownSamplingFactors[] = {4, 8}; } // namespace diff --git a/modules/audio_processing/aec3/subtractor_unittest.cc b/modules/audio_processing/aec3/subtractor_unittest.cc index d940df47ef..48422895e0 100644 --- a/modules/audio_processing/aec3/subtractor_unittest.cc +++ b/modules/audio_processing/aec3/subtractor_unittest.cc @@ -99,8 +99,9 @@ std::vector RunSubtractorTest( // [B,A] = butter(2,100/8000,'high') constexpr std::array - kHighPassFilterCoefficients = { - {{{0.97261f, -1.94523f, 0.97261f}, {-1.94448f, 0.94598f}}}}; + kHighPassFilterCoefficients = {{ + {{0.97261f, -1.94523f, 0.97261f}, {-1.94448f, 0.94598f}}, + }}; std::vector> x_hp_filter( num_render_channels); for (size_t ch = 0; ch < num_render_channels; ++ch) { diff --git a/modules/audio_processing/high_pass_filter.cc b/modules/audio_processing/high_pass_filter.cc index 276b5a2c53..45116bc7eb 100644 --- a/modules/audio_processing/high_pass_filter.cc +++ b/modules/audio_processing/high_pass_filter.cc @@ -24,21 +24,24 @@ namespace webrtc { namespace { // [B,A] = butter(2,100/8000,'high') constexpr std::array - kHighPassFilterCoefficients16kHz = { - {{{0.972613898f, -1.945227797f, 0.972613898f}, - {-1.944477658f, 0.945977936f}}}}; + kHighPassFilterCoefficients16kHz = {{ + {{0.972613898f, -1.945227797f, 0.972613898f}, + {-1.944477658f, 0.945977936f}}, + }}; // [B,A] = butter(2,100/16000,'high') constexpr std::array - kHighPassFilterCoefficients32kHz = { - {{{0.986211925f, -1.972423849f, 0.986211925f}, - {-1.972233729f, 0.972613969f}}}}; + kHighPassFilterCoefficients32kHz = {{ + {{0.986211925f, -1.972423849f, 0.986211925f}, + {-1.972233729f, 0.972613969f}}, + }}; // [B,A] = butter(2,100/24000,'high') constexpr std::array - kHighPassFilterCoefficients48kHz = { - {{{0.990786698f, -1.981573396f, 0.990786698f}, - {-1.981488509f, 0.981658283f}}}}; + kHighPassFilterCoefficients48kHz = {{ + {{0.990786698f, -1.981573396f, 0.990786698f}, + {-1.981488509f, 0.981658283f}}, + }}; ArrayView ChooseCoefficients( int sample_rate_hz) { diff --git a/modules/audio_processing/utility/cascaded_biquad_filter.cc b/modules/audio_processing/utility/cascaded_biquad_filter.cc index d56b62f26f..d2f4952aa8 100644 --- a/modules/audio_processing/utility/cascaded_biquad_filter.cc +++ b/modules/audio_processing/utility/cascaded_biquad_filter.cc @@ -10,7 +10,6 @@ #include "modules/audio_processing/utility/cascaded_biquad_filter.h" #include -#include #include #include @@ -19,44 +18,6 @@ namespace webrtc { -CascadedBiQuadFilter::BiQuadParam::BiQuadParam(std::complex zero, - std::complex pole, - float gain, - bool mirror_zero_along_i_axis) - : zero(zero), - pole(pole), - gain(gain), - mirror_zero_along_i_axis(mirror_zero_along_i_axis) {} - -CascadedBiQuadFilter::BiQuadParam::BiQuadParam(const BiQuadParam&) = default; - -CascadedBiQuadFilter::BiQuad::BiQuad( - const CascadedBiQuadFilter::BiQuadParam& param) - : x(), y() { - float z_r = std::real(param.zero); - float z_i = std::imag(param.zero); - float p_r = std::real(param.pole); - float p_i = std::imag(param.pole); - float gain = param.gain; - - if (param.mirror_zero_along_i_axis) { - // Assuming zeroes at z_r and -z_r. - RTC_DCHECK(z_i == 0.f); - coefficients.b[0] = gain * 1.f; - coefficients.b[1] = 0.f; - coefficients.b[2] = gain * -(z_r * z_r); - } else { - // Assuming zeros at (z_r + z_i*i) and (z_r - z_i*i). - coefficients.b[0] = gain * 1.f; - coefficients.b[1] = gain * -2.f * z_r; - coefficients.b[2] = gain * (z_r * z_r + z_i * z_i); - } - - // Assuming poles at (p_r + p_i*i) and (p_r - p_i*i). - coefficients.a[0] = -2.f * p_r; - coefficients.a[1] = p_r * p_r + p_i * p_i; -} - void CascadedBiQuadFilter::BiQuad::BiQuad::Reset() { x[0] = x[1] = y[0] = y[1] = 0.f; } @@ -68,13 +29,6 @@ CascadedBiQuadFilter::CascadedBiQuadFilter( } } -CascadedBiQuadFilter::CascadedBiQuadFilter( - const std::vector& biquad_params) { - for (const auto& param : biquad_params) { - biquads_.push_back(BiQuad(param)); - } -} - CascadedBiQuadFilter::~CascadedBiQuadFilter() = default; void CascadedBiQuadFilter::Process(ArrayView x, diff --git a/modules/audio_processing/utility/cascaded_biquad_filter.h b/modules/audio_processing/utility/cascaded_biquad_filter.h index 7050f98105..25d8d504ec 100644 --- a/modules/audio_processing/utility/cascaded_biquad_filter.h +++ b/modules/audio_processing/utility/cascaded_biquad_filter.h @@ -13,7 +13,6 @@ #include -#include #include #include "api/array_view.h" @@ -24,18 +23,6 @@ namespace webrtc { // is direct form 1. class CascadedBiQuadFilter { public: - struct BiQuadParam { - BiQuadParam(std::complex zero, - std::complex pole, - float gain, - bool mirror_zero_along_i_axis = false); - explicit BiQuadParam(const BiQuadParam&); - std::complex zero; - std::complex pole; - float gain; - bool mirror_zero_along_i_axis; - }; - struct BiQuadCoefficients { float b[3]; float a[2]; @@ -44,7 +31,6 @@ class CascadedBiQuadFilter { struct BiQuad { explicit BiQuad(const BiQuadCoefficients& coefficients) : coefficients(coefficients), x(), y() {} - explicit BiQuad(const CascadedBiQuadFilter::BiQuadParam& param); void Reset(); BiQuadCoefficients coefficients; float x[2]; @@ -53,8 +39,6 @@ class CascadedBiQuadFilter { CascadedBiQuadFilter( ArrayView coefficients); - explicit CascadedBiQuadFilter( - const std::vector& biquad_params); ~CascadedBiQuadFilter(); CascadedBiQuadFilter(const CascadedBiQuadFilter&) = delete; CascadedBiQuadFilter& operator=(const CascadedBiQuadFilter&) = delete; diff --git a/modules/audio_processing/utility/cascaded_biquad_filter_unittest.cc b/modules/audio_processing/utility/cascaded_biquad_filter_unittest.cc index cf213787fc..d9ab839a93 100644 --- a/modules/audio_processing/utility/cascaded_biquad_filter_unittest.cc +++ b/modules/audio_processing/utility/cascaded_biquad_filter_unittest.cc @@ -25,18 +25,25 @@ namespace { // Coefficients for a second order Butterworth high-pass filter with cutoff // frequency 100 Hz. constexpr std::array - kHighPassFilterCoefficients = { - {{{0.97261f, -1.94523f, 0.97261f}, {-1.94448f, 0.94598f}}}}; + kHighPassFilterCoefficients = {{ + {{0.97261f, -1.94523f, 0.97261f}, {-1.94448f, 0.94598f}}, + }}; constexpr std::array - kTransparentCoefficients = {{{{1.f, 0.f, 0.f}, {0.f, 0.f}}}}; + kTransparentCoefficients = {{ + {{1.f, 0.f, 0.f}, {0.f, 0.f}}, + }}; constexpr std::array - kBlockingCoefficients = {{{{0.f, 0.f, 0.f}, {0.f, 0.f}}}}; + kBlockingCoefficients = {{ + {{0.f, 0.f, 0.f}, {0.f, 0.f}}, + }}; constexpr std::array - kCascadedCoefficients = { - {{{-1.f, 0.f, 0.f}, {0.f, 0.f}}, {{0.f, 0.f, -1.f}, {0.f, 0.f}}}}; + kCascadedCoefficients = {{ + {{-1.f, 0.f, 0.f}, {0.f, 0.f}}, + {{0.f, 0.f, -1.f}, {0.f, 0.f}}, + }}; std::vector CreateInputWithIncreasingValues(size_t vector_length) { std::vector v(vector_length); @@ -143,47 +150,4 @@ TEST(CascadedBiquadFilterDeathTest, InputSizeCheckVerification) { EXPECT_DEATH(filter.Process(input, output), ""); } #endif - -// Verifies the conversion from zero, pole, gain to filter coefficients for -// lowpass filter. -TEST(CascadedBiquadFilter, BiQuadParamLowPass) { - CascadedBiQuadFilter::BiQuadParam param( - {-1.0f, 0.0f}, {0.23146901f, 0.39514232f}, 0.1866943331163784f); - CascadedBiQuadFilter::BiQuad filter(param); - const float epsilon = 1e-6f; - EXPECT_NEAR(filter.coefficients.b[0], 0.18669433f, epsilon); - EXPECT_NEAR(filter.coefficients.b[1], 0.37338867f, epsilon); - EXPECT_NEAR(filter.coefficients.b[2], 0.18669433f, epsilon); - EXPECT_NEAR(filter.coefficients.a[0], -0.46293803f, epsilon); - EXPECT_NEAR(filter.coefficients.a[1], 0.20971536f, epsilon); -} - -// Verifies the conversion from zero, pole, gain to filter coefficients for -// highpass filter. -TEST(CascadedBiquadFilter, BiQuadParamHighPass) { - CascadedBiQuadFilter::BiQuadParam param( - {1.0f, 0.0f}, {0.72712179f, 0.21296904f}, 0.75707637533388494f); - CascadedBiQuadFilter::BiQuad filter(param); - const float epsilon = 1e-6f; - EXPECT_NEAR(filter.coefficients.b[0], 0.75707638f, epsilon); - EXPECT_NEAR(filter.coefficients.b[1], -1.51415275f, epsilon); - EXPECT_NEAR(filter.coefficients.b[2], 0.75707638f, epsilon); - EXPECT_NEAR(filter.coefficients.a[0], -1.45424359f, epsilon); - EXPECT_NEAR(filter.coefficients.a[1], 0.57406192f, epsilon); -} - -// Verifies the conversion from zero, pole, gain to filter coefficients for -// bandpass filter. -TEST(CascadedBiquadFilter, BiQuadParamBandPass) { - CascadedBiQuadFilter::BiQuadParam param( - {1.0f, 0.0f}, {1.11022302e-16f, 0.71381051f}, 0.2452372752527856f, true); - CascadedBiQuadFilter::BiQuad filter(param); - const float epsilon = 1e-6f; - EXPECT_NEAR(filter.coefficients.b[0], 0.24523728f, epsilon); - EXPECT_NEAR(filter.coefficients.b[1], 0.f, epsilon); - EXPECT_NEAR(filter.coefficients.b[2], -0.24523728f, epsilon); - EXPECT_NEAR(filter.coefficients.a[0], -2.22044605e-16f, epsilon); - EXPECT_NEAR(filter.coefficients.a[1], 5.09525449e-01f, epsilon); -} - } // namespace webrtc From c0fdc1b2873a76a860692d664b0c5e80ee192333 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Thu, 15 May 2025 11:51:27 +0000 Subject: [PATCH 220/292] [cleanup] Replace uses of NULL with nullptr MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Generated using clang-tidy, tools/clang/scripts/build_clang_tools_extra.py \ --fetch out/Default clang-tidy clang-apply-replacements ninja -C out/Default gn gen out/Default --export-compile-commands cd out/Default tools/clang/third_party/llvm/clang-tools-extra/clang-tidy/tool/run-clang-tidy.py -p . \ -clang-tidy-binary out/Default/tools/clang/third_party/llvm/build/bin/clang-tidy \ -clang-apply-replacements-binary \ out/Default/tools/clang/third_party/llvm/build/bin/clang-apply-replacements \ -checks='-*,modernize-use-nullptr' \ -fix This CL was uploaded by git cl split. Bug: webrtc:42232237 Change-Id: Iba6258b9c83bf8623fc54f483ab72824ead462ef No-Iwyu: LSC Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391642 Commit-Queue: Per Åhgren Reviewed-by: Per Åhgren Auto-Submit: Evan Shrubsole Cr-Commit-Position: refs/heads/main@{#44692} --- .../aec_dump/aec_dump_unittest.cc | 2 +- modules/audio_processing/aecm/aecm_core.cc | 36 +++---- modules/audio_processing/aecm/aecm_core_c.cc | 10 +- .../aecm/echo_control_mobile.cc | 34 +++--- .../audio_processing/agc/legacy/analog_agc.cc | 10 +- .../agc/loudness_histogram.cc | 2 +- .../agc/loudness_histogram_unittest.cc | 2 +- .../audio_processing_unittest.cc | 54 +++++----- .../echo_control_mobile_impl.cc | 4 +- .../utility/delay_estimator.cc | 55 +++++----- .../utility/delay_estimator_unittest.cc | 100 +++++++++--------- .../utility/delay_estimator_wrapper.cc | 68 ++++++------ .../vad/pitch_based_vad_unittest.cc | 8 +- .../audio_processing/vad/pole_zero_filter.cc | 6 +- .../vad/standalone_vad_unittest.cc | 4 +- .../vad/vad_audio_proc_unittest.cc | 4 +- .../vad/vad_circular_buffer.cc | 2 +- 17 files changed, 202 insertions(+), 199 deletions(-) diff --git a/modules/audio_processing/aec_dump/aec_dump_unittest.cc b/modules/audio_processing/aec_dump/aec_dump_unittest.cc index f928ae8faa..935402b31d 100644 --- a/modules/audio_processing/aec_dump/aec_dump_unittest.cc +++ b/modules/audio_processing/aec_dump/aec_dump_unittest.cc @@ -77,7 +77,7 @@ TEST(AecDumper, WriteToFile) { // Verify the file has been written after the AecDump d-tor has // finished. FILE* fid = fopen(filename.c_str(), "r"); - ASSERT_TRUE(fid != NULL); + ASSERT_TRUE(fid != nullptr); // Clean it up. ASSERT_EQ(0, fclose(fid)); diff --git a/modules/audio_processing/aecm/aecm_core.cc b/modules/audio_processing/aecm/aecm_core.cc index 7d3cd9df55..8b1e12794c 100644 --- a/modules/audio_processing/aecm/aecm_core.cc +++ b/modules/audio_processing/aecm/aecm_core.cc @@ -193,50 +193,50 @@ AecmCore* WebRtcAecm_CreateCore() { WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(int16_t)); if (!aecm->farFrameBuf) { WebRtcAecm_FreeCore(aecm); - return NULL; + return nullptr; } aecm->nearNoisyFrameBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(int16_t)); if (!aecm->nearNoisyFrameBuf) { WebRtcAecm_FreeCore(aecm); - return NULL; + return nullptr; } aecm->nearCleanFrameBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(int16_t)); if (!aecm->nearCleanFrameBuf) { WebRtcAecm_FreeCore(aecm); - return NULL; + return nullptr; } aecm->outFrameBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(int16_t)); if (!aecm->outFrameBuf) { WebRtcAecm_FreeCore(aecm); - return NULL; + return nullptr; } aecm->delay_estimator_farend = WebRtc_CreateDelayEstimatorFarend(PART_LEN1, MAX_DELAY); - if (aecm->delay_estimator_farend == NULL) { + if (aecm->delay_estimator_farend == nullptr) { WebRtcAecm_FreeCore(aecm); - return NULL; + return nullptr; } aecm->delay_estimator = WebRtc_CreateDelayEstimator(aecm->delay_estimator_farend, 0); - if (aecm->delay_estimator == NULL) { + if (aecm->delay_estimator == nullptr) { WebRtcAecm_FreeCore(aecm); - return NULL; + return nullptr; } // TODO(bjornv): Explicitly disable robust delay validation until no // performance regression has been established. Then remove the line. WebRtc_enable_robust_validation(aecm->delay_estimator, 0); aecm->real_fft = WebRtcSpl_CreateRealFFT(PART_LEN_SHIFT); - if (aecm->real_fft == NULL) { + if (aecm->real_fft == nullptr) { WebRtcAecm_FreeCore(aecm); - return NULL; + return nullptr; } // Init some aecm pointers. 16 and 32 byte alignment is only necessary @@ -491,7 +491,7 @@ int WebRtcAecm_Control(AecmCore* aecm, int delay, int nlpFlag) { } void WebRtcAecm_FreeCore(AecmCore* aecm) { - if (aecm == NULL) { + if (aecm == nullptr) { return; } @@ -516,7 +516,7 @@ int WebRtcAecm_ProcessFrame(AecmCore* aecm, int16_t* outBlock = (int16_t*)(((uintptr_t)outBlock_buf + 15) & ~15); int16_t farFrame[FRAME_LEN]; - const int16_t* out_ptr = NULL; + const int16_t* out_ptr = nullptr; int size = 0; // Buffer the current frame. @@ -528,24 +528,24 @@ int WebRtcAecm_ProcessFrame(AecmCore* aecm, // to pass the smaller blocks individually. WebRtc_WriteBuffer(aecm->farFrameBuf, farFrame, FRAME_LEN); WebRtc_WriteBuffer(aecm->nearNoisyFrameBuf, nearendNoisy, FRAME_LEN); - if (nearendClean != NULL) { + if (nearendClean != nullptr) { WebRtc_WriteBuffer(aecm->nearCleanFrameBuf, nearendClean, FRAME_LEN); } // Process as many blocks as possible. while (WebRtc_available_read(aecm->farFrameBuf) >= PART_LEN) { int16_t far_block[PART_LEN]; - const int16_t* far_block_ptr = NULL; + const int16_t* far_block_ptr = nullptr; int16_t near_noisy_block[PART_LEN]; - const int16_t* near_noisy_block_ptr = NULL; + const int16_t* near_noisy_block_ptr = nullptr; WebRtc_ReadBuffer(aecm->farFrameBuf, (void**)&far_block_ptr, far_block, PART_LEN); WebRtc_ReadBuffer(aecm->nearNoisyFrameBuf, (void**)&near_noisy_block_ptr, near_noisy_block, PART_LEN); - if (nearendClean != NULL) { + if (nearendClean != nullptr) { int16_t near_clean_block[PART_LEN]; - const int16_t* near_clean_block_ptr = NULL; + const int16_t* near_clean_block_ptr = nullptr; WebRtc_ReadBuffer(aecm->nearCleanFrameBuf, (void**)&near_clean_block_ptr, near_clean_block, PART_LEN); @@ -555,7 +555,7 @@ int WebRtcAecm_ProcessFrame(AecmCore* aecm, } } else { if (WebRtcAecm_ProcessBlock(aecm, far_block_ptr, near_noisy_block_ptr, - NULL, outBlock) == -1) { + nullptr, outBlock) == -1) { return -1; } } diff --git a/modules/audio_processing/aecm/aecm_core_c.cc b/modules/audio_processing/aecm/aecm_core_c.cc index 1d750da3fe..1a975d2adc 100644 --- a/modules/audio_processing/aecm/aecm_core_c.cc +++ b/modules/audio_processing/aecm/aecm_core_c.cc @@ -243,7 +243,7 @@ static void InverseFFTAndWindow(AecmCore* aecm, memcpy(aecm->xBuf, aecm->xBuf + PART_LEN, sizeof(int16_t) * PART_LEN); memcpy(aecm->dBufNoisy, aecm->dBufNoisy + PART_LEN, sizeof(int16_t) * PART_LEN); - if (nearendClean != NULL) { + if (nearendClean != nullptr) { memcpy(aecm->dBufClean, aecm->dBufClean + PART_LEN, sizeof(int16_t) * PART_LEN); } @@ -390,7 +390,7 @@ int RTC_NO_SANITIZE("signed-integer-overflow") // bugs.webrtc.org/8200 uint16_t dfaNoisy[PART_LEN1]; uint16_t dfaClean[PART_LEN1]; uint16_t* ptrDfaClean = dfaClean; - const uint16_t* far_spectrum_ptr = NULL; + const uint16_t* far_spectrum_ptr = nullptr; // 32 byte aligned buffers (with +8 or +16). // TODO(kma): define fft with ComplexInt16. @@ -435,7 +435,7 @@ int RTC_NO_SANITIZE("signed-integer-overflow") // bugs.webrtc.org/8200 // Buffer near and far end signals memcpy(aecm->xBuf + PART_LEN, farend, sizeof(int16_t) * PART_LEN); memcpy(aecm->dBufNoisy + PART_LEN, nearendNoisy, sizeof(int16_t) * PART_LEN); - if (nearendClean != NULL) { + if (nearendClean != nullptr) { memcpy(aecm->dBufClean + PART_LEN, nearendClean, sizeof(int16_t) * PART_LEN); } @@ -449,7 +449,7 @@ int RTC_NO_SANITIZE("signed-integer-overflow") // bugs.webrtc.org/8200 aecm->dfaNoisyQDomainOld = aecm->dfaNoisyQDomain; aecm->dfaNoisyQDomain = (int16_t)zerosDBufNoisy; - if (nearendClean == NULL) { + if (nearendClean == nullptr) { ptrDfaClean = dfaNoisy; aecm->dfaCleanQDomainOld = aecm->dfaNoisyQDomainOld; aecm->dfaCleanQDomain = aecm->dfaNoisyQDomain; @@ -487,7 +487,7 @@ int RTC_NO_SANITIZE("signed-integer-overflow") // bugs.webrtc.org/8200 // Get aligned far end spectrum far_spectrum_ptr = WebRtcAecm_AlignedFarend(aecm, &far_q, delay); zerosXBuf = (int16_t)far_q; - if (far_spectrum_ptr == NULL) { + if (far_spectrum_ptr == nullptr) { return -1; } diff --git a/modules/audio_processing/aecm/echo_control_mobile.cc b/modules/audio_processing/aecm/echo_control_mobile.cc index 14522c0f1d..1f58984441 100644 --- a/modules/audio_processing/aecm/echo_control_mobile.cc +++ b/modules/audio_processing/aecm/echo_control_mobile.cc @@ -95,13 +95,13 @@ void* WebRtcAecm_Create() { aecm->aecmCore = WebRtcAecm_CreateCore(); if (!aecm->aecmCore) { WebRtcAecm_Free(aecm); - return NULL; + return nullptr; } aecm->farendBuf = WebRtc_CreateBuffer(kBufSizeSamp, sizeof(int16_t)); if (!aecm->farendBuf) { WebRtcAecm_Free(aecm); - return NULL; + return nullptr; } #ifdef AEC_DEBUG @@ -121,7 +121,7 @@ void* WebRtcAecm_Create() { void WebRtcAecm_Free(void* aecmInst) { AecMobile* aecm = static_cast(aecmInst); - if (aecm == NULL) { + if (aecm == nullptr) { return; } @@ -145,7 +145,7 @@ int32_t WebRtcAecm_Init(void* aecmInst, int32_t sampFreq) { AecMobile* aecm = static_cast(aecmInst); AecmConfig aecConfig; - if (aecm == NULL) { + if (aecm == nullptr) { return -1; } @@ -199,10 +199,10 @@ int32_t WebRtcAecm_GetBufferFarendError(void* aecmInst, size_t nrOfSamples) { AecMobile* aecm = static_cast(aecmInst); - if (aecm == NULL) + if (aecm == nullptr) return -1; - if (farend == NULL) + if (farend == nullptr) return AECM_NULL_POINTER_ERROR; if (aecm->initFlag != kInitCheck) @@ -251,15 +251,15 @@ int32_t WebRtcAecm_Process(void* aecmInst, short msInAECBuf; #endif - if (aecm == NULL) { + if (aecm == nullptr) { return -1; } - if (nearendNoisy == NULL) { + if (nearendNoisy == nullptr) { return AECM_NULL_POINTER_ERROR; } - if (out == NULL) { + if (out == nullptr) { return AECM_NULL_POINTER_ERROR; } @@ -285,7 +285,7 @@ int32_t WebRtcAecm_Process(void* aecmInst, nBlocks10ms = nFrames / aecm->aecmCore->mult; if (aecm->ECstartup) { - if (nearendClean == NULL) { + if (nearendClean == nullptr) { if (out != nearendNoisy) { memcpy(out, nearendNoisy, sizeof(short) * nrOfSamples); } @@ -362,7 +362,7 @@ int32_t WebRtcAecm_Process(void* aecmInst, // Note only 1 block supported for nb and 2 blocks for wb for (i = 0; i < nFrames; i++) { int16_t farend[FRAME_LEN]; - const int16_t* farend_ptr = NULL; + const int16_t* farend_ptr = nullptr; nmbrOfFilledBuffers = (short)WebRtc_available_read(aecm->farendBuf) / FRAME_LEN; @@ -393,7 +393,7 @@ int32_t WebRtcAecm_Process(void* aecmInst, &out[FRAME_LEN * i], aecm->knownDelay);*/ if (WebRtcAecm_ProcessFrame( aecm->aecmCore, farend_ptr, &nearendNoisy[FRAME_LEN * i], - (nearendClean ? &nearendClean[FRAME_LEN * i] : NULL), + (nearendClean ? &nearendClean[FRAME_LEN * i] : nullptr), &out[FRAME_LEN * i]) == -1) return -1; } @@ -412,7 +412,7 @@ int32_t WebRtcAecm_Process(void* aecmInst, int32_t WebRtcAecm_set_config(void* aecmInst, AecmConfig config) { AecMobile* aecm = static_cast(aecmInst); - if (aecm == NULL) { + if (aecm == nullptr) { return -1; } @@ -486,10 +486,10 @@ int32_t WebRtcAecm_InitEchoPath(void* aecmInst, AecMobile* aecm = static_cast(aecmInst); const int16_t* echo_path_ptr = static_cast(echo_path); - if (aecmInst == NULL) { + if (aecmInst == nullptr) { return -1; } - if (echo_path == NULL) { + if (echo_path == nullptr) { return AECM_NULL_POINTER_ERROR; } if (size_bytes != WebRtcAecm_echo_path_size_bytes()) { @@ -511,10 +511,10 @@ int32_t WebRtcAecm_GetEchoPath(void* aecmInst, AecMobile* aecm = static_cast(aecmInst); int16_t* echo_path_ptr = static_cast(echo_path); - if (aecmInst == NULL) { + if (aecmInst == nullptr) { return -1; } - if (echo_path == NULL) { + if (echo_path == nullptr) { return AECM_NULL_POINTER_ERROR; } if (size_bytes != WebRtcAecm_echo_path_size_bytes()) { diff --git a/modules/audio_processing/agc/legacy/analog_agc.cc b/modules/audio_processing/agc/legacy/analog_agc.cc index e40a3f1629..d946060e85 100644 --- a/modules/audio_processing/agc/legacy/analog_agc.cc +++ b/modules/audio_processing/agc/legacy/analog_agc.cc @@ -265,7 +265,7 @@ int WebRtcAgc_GetAddFarendError(void* state, size_t samples) { LegacyAgc* stt; stt = reinterpret_cast(state); - if (stt == NULL) + if (stt == nullptr) return -1; if (stt->fs == 8000) { @@ -943,7 +943,7 @@ int WebRtcAgc_Analyze(void* agcInst, int32_t gains[11]) { LegacyAgc* stt = reinterpret_cast(agcInst); - if (stt == NULL) { + if (stt == nullptr) { return -1; } @@ -1005,7 +1005,7 @@ int WebRtcAgc_set_config(void* agcInst, WebRtcAgcConfig agcConfig) { LegacyAgc* stt; stt = reinterpret_cast(agcInst); - if (stt == NULL) { + if (stt == nullptr) { return -1; } @@ -1053,11 +1053,11 @@ int WebRtcAgc_get_config(void* agcInst, WebRtcAgcConfig* config) { LegacyAgc* stt; stt = reinterpret_cast(agcInst); - if (stt == NULL) { + if (stt == nullptr) { return -1; } - if (config == NULL) { + if (config == nullptr) { stt->lastError = AGC_NULL_POINTER_ERROR; return -1; } diff --git a/modules/audio_processing/agc/loudness_histogram.cc b/modules/audio_processing/agc/loudness_histogram.cc index b0a1f53b97..492d00b390 100644 --- a/modules/audio_processing/agc/loudness_histogram.cc +++ b/modules/audio_processing/agc/loudness_histogram.cc @@ -174,7 +174,7 @@ LoudnessHistogram* LoudnessHistogram::Create() { LoudnessHistogram* LoudnessHistogram::Create(int window_size) { if (window_size < 0) - return NULL; + return nullptr; return new LoudnessHistogram(window_size); } diff --git a/modules/audio_processing/agc/loudness_histogram_unittest.cc b/modules/audio_processing/agc/loudness_histogram_unittest.cc index bbc0a7ee92..a4f1020ad3 100644 --- a/modules/audio_processing/agc/loudness_histogram_unittest.cc +++ b/modules/audio_processing/agc/loudness_histogram_unittest.cc @@ -53,7 +53,7 @@ void LoudnessHistogramTest::TestClean() { void LoudnessHistogramTest::RunTest(bool enable_circular_buff, absl::string_view filename) { FILE* in_file = fopen(std::string(filename).c_str(), "rb"); - ASSERT_TRUE(in_file != NULL); + ASSERT_TRUE(in_file != nullptr); if (enable_circular_buff) { int buffer_size; EXPECT_EQ(fread(&buffer_size, sizeof(buffer_size), 1, in_file), 1u); diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc index 0961036ec4..86d049ee63 100644 --- a/modules/audio_processing/audio_processing_unittest.cc +++ b/modules/audio_processing/audio_processing_unittest.cc @@ -171,7 +171,7 @@ int16_t MaxAudioFrame(const Int16FrameData& frame) { void OpenFileAndWriteMessage(absl::string_view filename, const MessageLite& msg) { FILE* file = fopen(std::string(filename).c_str(), "wb"); - ASSERT_TRUE(file != NULL); + ASSERT_TRUE(file != nullptr); int32_t size = checked_cast(msg.ByteSizeLong()); ASSERT_GT(size, 0); @@ -253,7 +253,7 @@ void ClearTempOutFiles() { void OpenFileAndReadMessage(absl::string_view filename, MessageLite* msg) { FILE* file = fopen(std::string(filename).c_str(), "rb"); - ASSERT_TRUE(file != NULL); + ASSERT_TRUE(file != nullptr); ReadMessageFromFile(file, msg); fclose(file); } @@ -438,9 +438,9 @@ ApmTest::ApmTest() ref_filename_(GetReferenceFilename()), output_sample_rate_hz_(0), num_output_channels_(0), - far_file_(NULL), - near_file_(NULL), - out_file_(NULL) { + far_file_(nullptr), + near_file_(nullptr), + out_file_(nullptr) { apm_ = BuiltinAudioProcessingBuilder().Build(CreateEnvironment()); AudioProcessing::Config apm_config = apm_->GetConfig(); apm_config.gain_controller1.analog_gain_controller.enabled = false; @@ -449,7 +449,7 @@ ApmTest::ApmTest() } void ApmTest::SetUp() { - ASSERT_TRUE(apm_.get() != NULL); + ASSERT_TRUE(apm_.get() != nullptr); Init(32000, 32000, 32000, 2, 2, 2, false); } @@ -458,17 +458,17 @@ void ApmTest::TearDown() { if (far_file_) { ASSERT_EQ(0, fclose(far_file_)); } - far_file_ = NULL; + far_file_ = nullptr; if (near_file_) { ASSERT_EQ(0, fclose(near_file_)); } - near_file_ = NULL; + near_file_ = nullptr; if (out_file_) { ASSERT_EQ(0, fclose(out_file_)); } - out_file_ = NULL; + out_file_ = nullptr; } void ApmTest::Init(AudioProcessing* ap) { @@ -500,14 +500,16 @@ void ApmTest::Init(int sample_rate_hz, } std::string filename = ResourceFilePath("far", sample_rate_hz); far_file_ = fopen(filename.c_str(), "rb"); - ASSERT_TRUE(far_file_ != NULL) << "Could not open file " << filename << "\n"; + ASSERT_TRUE(far_file_ != nullptr) + << "Could not open file " << filename << "\n"; if (near_file_) { ASSERT_EQ(0, fclose(near_file_)); } filename = ResourceFilePath("near", sample_rate_hz); near_file_ = fopen(filename.c_str(), "rb"); - ASSERT_TRUE(near_file_ != NULL) << "Could not open file " << filename << "\n"; + ASSERT_TRUE(near_file_ != nullptr) + << "Could not open file " << filename << "\n"; if (open_output_file) { if (out_file_) { @@ -518,7 +520,7 @@ void ApmTest::Init(int sample_rate_hz, reverse_sample_rate_hz, num_input_channels, num_output_channels, num_reverse_channels, num_reverse_channels, kForward); out_file_ = fopen(filename.c_str(), "wb"); - ASSERT_TRUE(out_file_ != NULL) + ASSERT_TRUE(out_file_ != nullptr) << "Could not open file " << filename << "\n"; } } @@ -552,7 +554,7 @@ bool ApmTest::ReadFrame(FILE* file, } bool ApmTest::ReadFrame(FILE* file, Int16FrameData* frame) { - return ReadFrame(file, frame, NULL); + return ReadFrame(file, frame, nullptr); } // If the end of the file has been reached, rewind it and attempt to read the @@ -567,7 +569,7 @@ void ApmTest::ReadFrameWithRewind(FILE* /* file */, } void ApmTest::ReadFrameWithRewind(FILE* file, Int16FrameData* frame) { - ReadFrameWithRewind(file, frame, NULL); + ReadFrameWithRewind(file, frame, nullptr); } int ApmTest::ProcessStreamChooser(Format format) { @@ -1429,7 +1431,7 @@ void ApmTest::ProcessDebugDump(absl::string_view in_filename, int max_size_bytes) { TaskQueueForTest worker_queue("ApmTest_worker_queue"); FILE* in_file = fopen(std::string(in_filename).c_str(), "rb"); - ASSERT_TRUE(in_file != NULL); + ASSERT_TRUE(in_file != nullptr); audioproc::Event event_msg; bool first_init = true; @@ -1541,9 +1543,9 @@ void ApmTest::VerifyDebugDumpTest(Format format) { FILE* ref_file = fopen(ref_filename.c_str(), "rb"); FILE* out_file = fopen(out_filename.c_str(), "rb"); FILE* limited_file = fopen(limited_filename.c_str(), "rb"); - ASSERT_TRUE(ref_file != NULL); - ASSERT_TRUE(out_file != NULL); - ASSERT_TRUE(limited_file != NULL); + ASSERT_TRUE(ref_file != nullptr); + ASSERT_TRUE(out_file != nullptr); + ASSERT_TRUE(limited_file != nullptr); std::unique_ptr ref_bytes; std::unique_ptr out_bytes; std::unique_ptr limited_bytes; @@ -1626,7 +1628,7 @@ TEST_F(ApmTest, DebugDump) { // Verify the file has been written. FILE* fid = fopen(filename.c_str(), "r"); - ASSERT_TRUE(fid != NULL); + ASSERT_TRUE(fid != nullptr); // Clean it up. ASSERT_EQ(0, fclose(fid)); @@ -1670,7 +1672,7 @@ TEST_F(ApmTest, DebugDumpFromFileHandle) { // Verify the file has been written. FILE* fid = fopen(filename.c_str(), "r"); - ASSERT_TRUE(fid != NULL); + ASSERT_TRUE(fid != nullptr); // Clean it up. ASSERT_EQ(0, fclose(fid)); @@ -1989,10 +1991,10 @@ class AudioProcessingTest num_reverse_input_channels, num_reverse_output_channels, kReverse) .c_str(), "wb"); - ASSERT_TRUE(far_file != NULL); - ASSERT_TRUE(near_file != NULL); - ASSERT_TRUE(out_file != NULL); - ASSERT_TRUE(rev_out_file != NULL); + ASSERT_TRUE(far_file != nullptr); + ASSERT_TRUE(near_file != nullptr); + ASSERT_TRUE(out_file != nullptr); + ASSERT_TRUE(rev_out_file != nullptr); ChannelBuffer fwd_cb(AudioProcessing::GetFrameSize(input_rate), num_input_channels); @@ -2123,8 +2125,8 @@ TEST_P(AudioProcessingTest, Formats) { cf[i].num_reverse_output, file_direction) .c_str(), "rb"); - ASSERT_TRUE(out_file != NULL); - ASSERT_TRUE(ref_file != NULL); + ASSERT_TRUE(out_file != nullptr); + ASSERT_TRUE(ref_file != nullptr); const size_t ref_samples_per_channel = AudioProcessing::GetFrameSize(ref_rate); diff --git a/modules/audio_processing/echo_control_mobile_impl.cc b/modules/audio_processing/echo_control_mobile_impl.cc index aff9d4e722..e1e0d3fcfc 100644 --- a/modules/audio_processing/echo_control_mobile_impl.cc +++ b/modules/audio_processing/echo_control_mobile_impl.cc @@ -191,9 +191,9 @@ int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio, split_bands = nullptr; } - if (noisy == NULL) { + if (noisy == nullptr) { noisy = clean; - clean = NULL; + clean = nullptr; } for (size_t render = 0; render < stream_properties_->num_reverse_channels; ++render) { diff --git a/modules/audio_processing/utility/delay_estimator.cc b/modules/audio_processing/utility/delay_estimator.cc index 6868392f6f..6f97b28a77 100644 --- a/modules/audio_processing/utility/delay_estimator.cc +++ b/modules/audio_processing/utility/delay_estimator.cc @@ -270,38 +270,38 @@ static int RobustValidation(const BinaryDelayEstimator* self, } void WebRtc_FreeBinaryDelayEstimatorFarend(BinaryDelayEstimatorFarend* self) { - if (self == NULL) { + if (self == nullptr) { return; } free(self->binary_far_history); - self->binary_far_history = NULL; + self->binary_far_history = nullptr; free(self->far_bit_counts); - self->far_bit_counts = NULL; + self->far_bit_counts = nullptr; free(self); } BinaryDelayEstimatorFarend* WebRtc_CreateBinaryDelayEstimatorFarend( int history_size) { - BinaryDelayEstimatorFarend* self = NULL; + BinaryDelayEstimatorFarend* self = nullptr; if (history_size > 1) { // Sanity conditions fulfilled. self = static_cast( malloc(sizeof(BinaryDelayEstimatorFarend))); } - if (self == NULL) { - return NULL; + if (self == nullptr) { + return nullptr; } self->history_size = 0; - self->binary_far_history = NULL; - self->far_bit_counts = NULL; + self->binary_far_history = nullptr; + self->far_bit_counts = nullptr; if (WebRtc_AllocateFarendBufferMemory(self, history_size) == 0) { WebRtc_FreeBinaryDelayEstimatorFarend(self); - self = NULL; + self = nullptr; } return self; } @@ -315,7 +315,8 @@ int WebRtc_AllocateFarendBufferMemory(BinaryDelayEstimatorFarend* self, history_size * sizeof(*self->binary_far_history))); self->far_bit_counts = static_cast(realloc( self->far_bit_counts, history_size * sizeof(*self->far_bit_counts))); - if ((self->binary_far_history == NULL) || (self->far_bit_counts == NULL)) { + if ((self->binary_far_history == nullptr) || + (self->far_bit_counts == nullptr)) { history_size = 0; } // Fill with zeros if we have expanded the buffers. @@ -386,25 +387,25 @@ void WebRtc_AddBinaryFarSpectrum(BinaryDelayEstimatorFarend* handle, } void WebRtc_FreeBinaryDelayEstimator(BinaryDelayEstimator* self) { - if (self == NULL) { + if (self == nullptr) { return; } free(self->mean_bit_counts); - self->mean_bit_counts = NULL; + self->mean_bit_counts = nullptr; free(self->bit_counts); - self->bit_counts = NULL; + self->bit_counts = nullptr; free(self->binary_near_history); - self->binary_near_history = NULL; + self->binary_near_history = nullptr; free(self->histogram); - self->histogram = NULL; + self->histogram = nullptr; // BinaryDelayEstimator does not have ownership of `farend`, hence we do not // free the memory here. That should be handled separately by the user. - self->farend = NULL; + self->farend = nullptr; free(self); } @@ -412,15 +413,15 @@ void WebRtc_FreeBinaryDelayEstimator(BinaryDelayEstimator* self) { BinaryDelayEstimator* WebRtc_CreateBinaryDelayEstimator( BinaryDelayEstimatorFarend* farend, int max_lookahead) { - BinaryDelayEstimator* self = NULL; + BinaryDelayEstimator* self = nullptr; - if ((farend != NULL) && (max_lookahead >= 0)) { + if ((farend != nullptr) && (max_lookahead >= 0)) { // Sanity conditions fulfilled. self = static_cast( malloc(sizeof(BinaryDelayEstimator))); } - if (self == NULL) { - return NULL; + if (self == nullptr) { + return nullptr; } self->farend = farend; @@ -432,15 +433,15 @@ BinaryDelayEstimator* WebRtc_CreateBinaryDelayEstimator( self->lookahead = max_lookahead; // Allocate memory for spectrum and history buffers. - self->mean_bit_counts = NULL; - self->bit_counts = NULL; - self->histogram = NULL; + self->mean_bit_counts = nullptr; + self->bit_counts = nullptr; + self->histogram = nullptr; self->binary_near_history = static_cast( malloc((max_lookahead + 1) * sizeof(*self->binary_near_history))); - if (self->binary_near_history == NULL || + if (self->binary_near_history == nullptr || WebRtc_AllocateHistoryBufferMemory(self, farend->history_size) == 0) { WebRtc_FreeBinaryDelayEstimator(self); - self = NULL; + self = nullptr; } return self; @@ -465,8 +466,8 @@ int WebRtc_AllocateHistoryBufferMemory(BinaryDelayEstimator* self, self->histogram = static_cast( realloc(self->histogram, (history_size + 1) * sizeof(*self->histogram))); - if ((self->mean_bit_counts == NULL) || (self->bit_counts == NULL) || - (self->histogram == NULL)) { + if ((self->mean_bit_counts == nullptr) || (self->bit_counts == nullptr) || + (self->histogram == nullptr)) { history_size = 0; } // Fill with zeros if we have expanded the buffers. diff --git a/modules/audio_processing/utility/delay_estimator_unittest.cc b/modules/audio_processing/utility/delay_estimator_unittest.cc index 6052612ef3..b026088a90 100644 --- a/modules/audio_processing/utility/delay_estimator_unittest.cc +++ b/modules/audio_processing/utility/delay_estimator_unittest.cc @@ -67,12 +67,12 @@ class DelayEstimatorTest : public ::testing::Test { }; DelayEstimatorTest::DelayEstimatorTest() - : handle_(NULL), - self_(NULL), - farend_handle_(NULL), - farend_self_(NULL), - binary_(NULL), - binary_farend_(NULL), + : handle_(nullptr), + self_(nullptr), + farend_handle_(nullptr), + farend_self_(nullptr), + binary_(nullptr), + binary_farend_(nullptr), spectrum_size_(kSpectrumSize) { // Dummy input data are set with more or less arbitrary non-zero values. memset(far_f_, 1, sizeof(far_f_)); @@ -91,28 +91,28 @@ DelayEstimatorTest::DelayEstimatorTest() void DelayEstimatorTest::SetUp() { farend_handle_ = WebRtc_CreateDelayEstimatorFarend(kSpectrumSize, kHistorySize); - ASSERT_TRUE(farend_handle_ != NULL); + ASSERT_TRUE(farend_handle_ != nullptr); farend_self_ = reinterpret_cast(farend_handle_); handle_ = WebRtc_CreateDelayEstimator(farend_handle_, kLookahead); - ASSERT_TRUE(handle_ != NULL); + ASSERT_TRUE(handle_ != nullptr); self_ = reinterpret_cast(handle_); binary_farend_ = WebRtc_CreateBinaryDelayEstimatorFarend(kHistorySize); - ASSERT_TRUE(binary_farend_ != NULL); + ASSERT_TRUE(binary_farend_ != nullptr); binary_ = WebRtc_CreateBinaryDelayEstimator(binary_farend_, kLookahead); - ASSERT_TRUE(binary_ != NULL); + ASSERT_TRUE(binary_ != nullptr); } void DelayEstimatorTest::TearDown() { WebRtc_FreeDelayEstimator(handle_); - handle_ = NULL; - self_ = NULL; + handle_ = nullptr; + self_ = nullptr; WebRtc_FreeDelayEstimatorFarend(farend_handle_); - farend_handle_ = NULL; - farend_self_ = NULL; + farend_handle_ = nullptr; + farend_self_ = nullptr; WebRtc_FreeBinaryDelayEstimator(binary_); - binary_ = NULL; + binary_ = nullptr; WebRtc_FreeBinaryDelayEstimatorFarend(binary_farend_); - binary_farend_ = NULL; + binary_farend_ = nullptr; } void DelayEstimatorTest::Init() { @@ -218,7 +218,7 @@ void DelayEstimatorTest::RunBinarySpectraTest(int near_offset, RunBinarySpectra(binary_, binary2, near_offset, lookahead_offset, offset); } WebRtc_FreeBinaryDelayEstimator(binary2); - binary2 = NULL; + binary2 = nullptr; binary_->robust_validation_enabled = 0; // Reset reference. } @@ -231,29 +231,29 @@ TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfWrapper) { // create failure. void* handle = farend_handle_; handle = WebRtc_CreateDelayEstimatorFarend(33, kHistorySize); - EXPECT_TRUE(handle == NULL); + EXPECT_TRUE(handle == nullptr); handle = WebRtc_CreateDelayEstimatorFarend(kSpectrumSize, 1); - EXPECT_TRUE(handle == NULL); + EXPECT_TRUE(handle == nullptr); handle = handle_; - handle = WebRtc_CreateDelayEstimator(NULL, kLookahead); - EXPECT_TRUE(handle == NULL); + handle = WebRtc_CreateDelayEstimator(nullptr, kLookahead); + EXPECT_TRUE(handle == nullptr); handle = WebRtc_CreateDelayEstimator(farend_handle_, -1); - EXPECT_TRUE(handle == NULL); + EXPECT_TRUE(handle == nullptr); // WebRtc_InitDelayEstimatorFarend() and WebRtc_InitDelayEstimator() should // return -1 if we have a NULL pointer as `handle`. - EXPECT_EQ(-1, WebRtc_InitDelayEstimatorFarend(NULL)); - EXPECT_EQ(-1, WebRtc_InitDelayEstimator(NULL)); + EXPECT_EQ(-1, WebRtc_InitDelayEstimatorFarend(nullptr)); + EXPECT_EQ(-1, WebRtc_InitDelayEstimator(nullptr)); // WebRtc_AddFarSpectrumFloat() should return -1 if we have: // 1) NULL pointer as `handle`. // 2) NULL pointer as far-end spectrum. // 3) Incorrect spectrum size. - EXPECT_EQ(-1, WebRtc_AddFarSpectrumFloat(NULL, far_f_, spectrum_size_)); + EXPECT_EQ(-1, WebRtc_AddFarSpectrumFloat(nullptr, far_f_, spectrum_size_)); // Use `farend_handle_` which is properly created at SetUp(). - EXPECT_EQ(-1, - WebRtc_AddFarSpectrumFloat(farend_handle_, NULL, spectrum_size_)); + EXPECT_EQ( + -1, WebRtc_AddFarSpectrumFloat(farend_handle_, nullptr, spectrum_size_)); EXPECT_EQ(-1, WebRtc_AddFarSpectrumFloat(farend_handle_, far_f_, spectrum_size_ + 1)); @@ -262,9 +262,9 @@ TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfWrapper) { // 2) NULL pointer as far-end spectrum. // 3) Incorrect spectrum size. // 4) Too high precision in far-end spectrum (Q-domain > 15). - EXPECT_EQ(-1, WebRtc_AddFarSpectrumFix(NULL, far_u16_, spectrum_size_, 0)); - EXPECT_EQ(-1, - WebRtc_AddFarSpectrumFix(farend_handle_, NULL, spectrum_size_, 0)); + EXPECT_EQ(-1, WebRtc_AddFarSpectrumFix(nullptr, far_u16_, spectrum_size_, 0)); + EXPECT_EQ( + -1, WebRtc_AddFarSpectrumFix(farend_handle_, nullptr, spectrum_size_, 0)); EXPECT_EQ(-1, WebRtc_AddFarSpectrumFix(farend_handle_, far_u16_, spectrum_size_ + 1, 0)); EXPECT_EQ(-1, WebRtc_AddFarSpectrumFix(farend_handle_, far_u16_, @@ -273,11 +273,11 @@ TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfWrapper) { // WebRtc_set_history_size() should return -1 if: // 1) `handle` is a NULL. // 2) `history_size` <= 1. - EXPECT_EQ(-1, WebRtc_set_history_size(NULL, 1)); + EXPECT_EQ(-1, WebRtc_set_history_size(nullptr, 1)); EXPECT_EQ(-1, WebRtc_set_history_size(handle_, 1)); // WebRtc_history_size() should return -1 if: // 1) NULL pointer input. - EXPECT_EQ(-1, WebRtc_history_size(NULL)); + EXPECT_EQ(-1, WebRtc_history_size(nullptr)); // 2) there is a mismatch between history size. void* tmp_handle = WebRtc_CreateDelayEstimator(farend_handle_, kHistorySize); EXPECT_EQ(0, WebRtc_InitDelayEstimator(tmp_handle)); @@ -295,21 +295,21 @@ TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfWrapper) { // WebRtc_set_allowed_offset() should return -1 if we have: // 1) NULL pointer as `handle`. // 2) `allowed_offset` < 0. - EXPECT_EQ(-1, WebRtc_set_allowed_offset(NULL, 0)); + EXPECT_EQ(-1, WebRtc_set_allowed_offset(nullptr, 0)); EXPECT_EQ(-1, WebRtc_set_allowed_offset(handle_, -1)); - EXPECT_EQ(-1, WebRtc_get_allowed_offset(NULL)); + EXPECT_EQ(-1, WebRtc_get_allowed_offset(nullptr)); // WebRtc_enable_robust_validation() should return -1 if we have: // 1) NULL pointer as `handle`. // 2) Incorrect `enable` value (not 0 or 1). - EXPECT_EQ(-1, WebRtc_enable_robust_validation(NULL, kEnable[0])); + EXPECT_EQ(-1, WebRtc_enable_robust_validation(nullptr, kEnable[0])); EXPECT_EQ(-1, WebRtc_enable_robust_validation(handle_, -1)); EXPECT_EQ(-1, WebRtc_enable_robust_validation(handle_, 2)); // WebRtc_is_robust_validation_enabled() should return -1 if we have NULL // pointer as `handle`. - EXPECT_EQ(-1, WebRtc_is_robust_validation_enabled(NULL)); + EXPECT_EQ(-1, WebRtc_is_robust_validation_enabled(nullptr)); // WebRtc_DelayEstimatorProcessFloat() should return -1 if we have: // 1) NULL pointer as `handle`. @@ -317,11 +317,11 @@ TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfWrapper) { // 3) Incorrect spectrum size. // 4) Non matching history sizes if multiple delay estimators using the same // far-end reference. - EXPECT_EQ(-1, - WebRtc_DelayEstimatorProcessFloat(NULL, near_f_, spectrum_size_)); + EXPECT_EQ( + -1, WebRtc_DelayEstimatorProcessFloat(nullptr, near_f_, spectrum_size_)); // Use `handle_` which is properly created at SetUp(). - EXPECT_EQ(-1, - WebRtc_DelayEstimatorProcessFloat(handle_, NULL, spectrum_size_)); + EXPECT_EQ( + -1, WebRtc_DelayEstimatorProcessFloat(handle_, nullptr, spectrum_size_)); EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFloat(handle_, near_f_, spectrum_size_ + 1)); // `tmp_handle` is already in a non-matching state. @@ -335,10 +335,10 @@ TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfWrapper) { // 4) Too high precision in near-end spectrum (Q-domain > 15). // 5) Non matching history sizes if multiple delay estimators using the same // far-end reference. + EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFix(nullptr, near_u16_, + spectrum_size_, 0)); EXPECT_EQ( - -1, WebRtc_DelayEstimatorProcessFix(NULL, near_u16_, spectrum_size_, 0)); - EXPECT_EQ(-1, - WebRtc_DelayEstimatorProcessFix(handle_, NULL, spectrum_size_, 0)); + -1, WebRtc_DelayEstimatorProcessFix(handle_, nullptr, spectrum_size_, 0)); EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFix(handle_, near_u16_, spectrum_size_ + 1, 0)); EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFix(handle_, near_u16_, @@ -349,7 +349,7 @@ TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfWrapper) { WebRtc_FreeDelayEstimator(tmp_handle); // WebRtc_last_delay() should return -1 if we have a NULL pointer as `handle`. - EXPECT_EQ(-1, WebRtc_last_delay(NULL)); + EXPECT_EQ(-1, WebRtc_last_delay(nullptr)); // Free any local memory if needed. WebRtc_FreeDelayEstimator(handle); @@ -479,7 +479,7 @@ TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfBinaryEstimatorFarend) { // Make sure we have a non-NULL value at start, so we can detect NULL after // create failure. binary = WebRtc_CreateBinaryDelayEstimatorFarend(1); - EXPECT_TRUE(binary == NULL); + EXPECT_TRUE(binary == nullptr); } TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfBinaryEstimator) { @@ -492,10 +492,10 @@ TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfBinaryEstimator) { // `binary_handle` should be NULL. // Make sure we have a non-NULL value at start, so we can detect NULL after // create failure. - binary_handle = WebRtc_CreateBinaryDelayEstimator(NULL, kLookahead); - EXPECT_TRUE(binary_handle == NULL); + binary_handle = WebRtc_CreateBinaryDelayEstimator(nullptr, kLookahead); + EXPECT_TRUE(binary_handle == nullptr); binary_handle = WebRtc_CreateBinaryDelayEstimator(binary_farend_, -1); - EXPECT_TRUE(binary_handle == NULL); + EXPECT_TRUE(binary_handle == nullptr); } TEST_F(DelayEstimatorTest, MeanEstimatorFix) { @@ -580,9 +580,9 @@ TEST_F(DelayEstimatorTest, AllowedOffsetNoImpactWhenRobustValidationDisabled) { TEST_F(DelayEstimatorTest, VerifyLookaheadAtCreate) { void* farend_handle = WebRtc_CreateDelayEstimatorFarend(kSpectrumSize, kMaxDelay); - ASSERT_TRUE(farend_handle != NULL); + ASSERT_TRUE(farend_handle != nullptr); void* handle = WebRtc_CreateDelayEstimator(farend_handle, kLookahead); - ASSERT_TRUE(handle != NULL); + ASSERT_TRUE(handle != nullptr); EXPECT_EQ(kLookahead, WebRtc_lookahead(handle)); WebRtc_FreeDelayEstimator(handle); WebRtc_FreeDelayEstimatorFarend(farend_handle); diff --git a/modules/audio_processing/utility/delay_estimator_wrapper.cc b/modules/audio_processing/utility/delay_estimator_wrapper.cc index 3b1409cc0b..216066afe0 100644 --- a/modules/audio_processing/utility/delay_estimator_wrapper.cc +++ b/modules/audio_processing/utility/delay_estimator_wrapper.cc @@ -127,21 +127,21 @@ static uint32_t BinarySpectrumFloat(const float* spectrum, void WebRtc_FreeDelayEstimatorFarend(void* handle) { DelayEstimatorFarend* self = (DelayEstimatorFarend*)handle; - if (handle == NULL) { + if (handle == nullptr) { return; } free(self->mean_far_spectrum); - self->mean_far_spectrum = NULL; + self->mean_far_spectrum = nullptr; WebRtc_FreeBinaryDelayEstimatorFarend(self->binary_farend); - self->binary_farend = NULL; + self->binary_farend = nullptr; free(self); } void* WebRtc_CreateDelayEstimatorFarend(int spectrum_size, int history_size) { - DelayEstimatorFarend* self = NULL; + DelayEstimatorFarend* self = nullptr; // Check if the sub band used in the delay estimation is small enough to fit // the binary spectra in a uint32_t. @@ -152,23 +152,23 @@ void* WebRtc_CreateDelayEstimatorFarend(int spectrum_size, int history_size) { malloc(sizeof(DelayEstimatorFarend))); } - if (self != NULL) { + if (self != nullptr) { int memory_fail = 0; // Allocate memory for the binary far-end spectrum handling. self->binary_farend = WebRtc_CreateBinaryDelayEstimatorFarend(history_size); - memory_fail |= (self->binary_farend == NULL); + memory_fail |= (self->binary_farend == nullptr); // Allocate memory for spectrum buffers. self->mean_far_spectrum = static_cast( malloc(spectrum_size * sizeof(SpectrumType))); - memory_fail |= (self->mean_far_spectrum == NULL); + memory_fail |= (self->mean_far_spectrum == nullptr); self->spectrum_size = spectrum_size; if (memory_fail) { WebRtc_FreeDelayEstimatorFarend(self); - self = NULL; + self = nullptr; } } @@ -178,7 +178,7 @@ void* WebRtc_CreateDelayEstimatorFarend(int spectrum_size, int history_size) { int WebRtc_InitDelayEstimatorFarend(void* handle) { DelayEstimatorFarend* self = (DelayEstimatorFarend*)handle; - if (self == NULL) { + if (self == nullptr) { return -1; } @@ -207,10 +207,10 @@ int WebRtc_AddFarSpectrumFix(void* handle, DelayEstimatorFarend* self = (DelayEstimatorFarend*)handle; uint32_t binary_spectrum = 0; - if (self == NULL) { + if (self == nullptr) { return -1; } - if (far_spectrum == NULL) { + if (far_spectrum == nullptr) { // Empty far end spectrum. return -1; } @@ -237,10 +237,10 @@ int WebRtc_AddFarSpectrumFloat(void* handle, DelayEstimatorFarend* self = (DelayEstimatorFarend*)handle; uint32_t binary_spectrum = 0; - if (self == NULL) { + if (self == nullptr) { return -1; } - if (far_spectrum == NULL) { + if (far_spectrum == nullptr) { // Empty far end spectrum. return -1; } @@ -260,45 +260,45 @@ int WebRtc_AddFarSpectrumFloat(void* handle, void WebRtc_FreeDelayEstimator(void* handle) { DelayEstimator* self = (DelayEstimator*)handle; - if (handle == NULL) { + if (handle == nullptr) { return; } free(self->mean_near_spectrum); - self->mean_near_spectrum = NULL; + self->mean_near_spectrum = nullptr; WebRtc_FreeBinaryDelayEstimator(self->binary_handle); - self->binary_handle = NULL; + self->binary_handle = nullptr; free(self); } void* WebRtc_CreateDelayEstimator(void* farend_handle, int max_lookahead) { - DelayEstimator* self = NULL; + DelayEstimator* self = nullptr; DelayEstimatorFarend* farend = (DelayEstimatorFarend*)farend_handle; - if (farend_handle != NULL) { + if (farend_handle != nullptr) { self = static_cast(malloc(sizeof(DelayEstimator))); } - if (self != NULL) { + if (self != nullptr) { int memory_fail = 0; // Allocate memory for the farend spectrum handling. self->binary_handle = WebRtc_CreateBinaryDelayEstimator(farend->binary_farend, max_lookahead); - memory_fail |= (self->binary_handle == NULL); + memory_fail |= (self->binary_handle == nullptr); // Allocate memory for spectrum buffers. self->mean_near_spectrum = static_cast( malloc(farend->spectrum_size * sizeof(SpectrumType))); - memory_fail |= (self->mean_near_spectrum == NULL); + memory_fail |= (self->mean_near_spectrum == nullptr); self->spectrum_size = farend->spectrum_size; if (memory_fail) { WebRtc_FreeDelayEstimator(self); - self = NULL; + self = nullptr; } } @@ -308,7 +308,7 @@ void* WebRtc_CreateDelayEstimator(void* farend_handle, int max_lookahead) { int WebRtc_InitDelayEstimator(void* handle) { DelayEstimator* self = (DelayEstimator*)handle; - if (self == NULL) { + if (self == nullptr) { return -1; } @@ -333,7 +333,7 @@ int WebRtc_SoftResetDelayEstimator(void* handle, int delay_shift) { int WebRtc_set_history_size(void* handle, int history_size) { DelayEstimator* self = static_cast(handle); - if ((self == NULL) || (history_size <= 1)) { + if ((self == nullptr) || (history_size <= 1)) { return -1; } return WebRtc_AllocateHistoryBufferMemory(self->binary_handle, history_size); @@ -342,7 +342,7 @@ int WebRtc_set_history_size(void* handle, int history_size) { int WebRtc_history_size(const void* handle) { const DelayEstimator* self = static_cast(handle); - if (self == NULL) { + if (self == nullptr) { return -1; } if (self->binary_handle->farend->history_size != @@ -375,7 +375,7 @@ int WebRtc_lookahead(void* handle) { int WebRtc_set_allowed_offset(void* handle, int allowed_offset) { DelayEstimator* self = (DelayEstimator*)handle; - if ((self == NULL) || (allowed_offset < 0)) { + if ((self == nullptr) || (allowed_offset < 0)) { return -1; } self->binary_handle->allowed_offset = allowed_offset; @@ -385,7 +385,7 @@ int WebRtc_set_allowed_offset(void* handle, int allowed_offset) { int WebRtc_get_allowed_offset(const void* handle) { const DelayEstimator* self = (const DelayEstimator*)handle; - if (self == NULL) { + if (self == nullptr) { return -1; } return self->binary_handle->allowed_offset; @@ -394,7 +394,7 @@ int WebRtc_get_allowed_offset(const void* handle) { int WebRtc_enable_robust_validation(void* handle, int enable) { DelayEstimator* self = (DelayEstimator*)handle; - if (self == NULL) { + if (self == nullptr) { return -1; } if ((enable < 0) || (enable > 1)) { @@ -408,7 +408,7 @@ int WebRtc_enable_robust_validation(void* handle, int enable) { int WebRtc_is_robust_validation_enabled(const void* handle) { const DelayEstimator* self = (const DelayEstimator*)handle; - if (self == NULL) { + if (self == nullptr) { return -1; } return self->binary_handle->robust_validation_enabled; @@ -421,10 +421,10 @@ int WebRtc_DelayEstimatorProcessFix(void* handle, DelayEstimator* self = (DelayEstimator*)handle; uint32_t binary_spectrum = 0; - if (self == NULL) { + if (self == nullptr) { return -1; } - if (near_spectrum == NULL) { + if (near_spectrum == nullptr) { // Empty near end spectrum. return -1; } @@ -451,10 +451,10 @@ int WebRtc_DelayEstimatorProcessFloat(void* handle, DelayEstimator* self = (DelayEstimator*)handle; uint32_t binary_spectrum = 0; - if (self == NULL) { + if (self == nullptr) { return -1; } - if (near_spectrum == NULL) { + if (near_spectrum == nullptr) { // Empty near end spectrum. return -1; } @@ -473,7 +473,7 @@ int WebRtc_DelayEstimatorProcessFloat(void* handle, int WebRtc_last_delay(void* handle) { DelayEstimator* self = (DelayEstimator*)handle; - if (self == NULL) { + if (self == nullptr) { return -1; } diff --git a/modules/audio_processing/vad/pitch_based_vad_unittest.cc b/modules/audio_processing/vad/pitch_based_vad_unittest.cc index 4a8331a769..3b7b956fa1 100644 --- a/modules/audio_processing/vad/pitch_based_vad_unittest.cc +++ b/modules/audio_processing/vad/pitch_based_vad_unittest.cc @@ -24,22 +24,22 @@ TEST(PitchBasedVadTest, VoicingProbabilityTest) { std::string spectral_peak_file_name = test::ResourcePath("audio_processing/agc/agc_spectral_peak", "dat"); FILE* spectral_peak_file = fopen(spectral_peak_file_name.c_str(), "rb"); - ASSERT_TRUE(spectral_peak_file != NULL); + ASSERT_TRUE(spectral_peak_file != nullptr); std::string pitch_gain_file_name = test::ResourcePath("audio_processing/agc/agc_pitch_gain", "dat"); FILE* pitch_gain_file = fopen(pitch_gain_file_name.c_str(), "rb"); - ASSERT_TRUE(pitch_gain_file != NULL); + ASSERT_TRUE(pitch_gain_file != nullptr); std::string pitch_lag_file_name = test::ResourcePath("audio_processing/agc/agc_pitch_lag", "dat"); FILE* pitch_lag_file = fopen(pitch_lag_file_name.c_str(), "rb"); - ASSERT_TRUE(pitch_lag_file != NULL); + ASSERT_TRUE(pitch_lag_file != nullptr); std::string voicing_prob_file_name = test::ResourcePath("audio_processing/agc/agc_voicing_prob", "dat"); FILE* voicing_prob_file = fopen(voicing_prob_file_name.c_str(), "rb"); - ASSERT_TRUE(voicing_prob_file != NULL); + ASSERT_TRUE(voicing_prob_file != nullptr); PitchBasedVad vad_; diff --git a/modules/audio_processing/vad/pole_zero_filter.cc b/modules/audio_processing/vad/pole_zero_filter.cc index e7a611309c..97688a44cb 100644 --- a/modules/audio_processing/vad/pole_zero_filter.cc +++ b/modules/audio_processing/vad/pole_zero_filter.cc @@ -22,8 +22,8 @@ PoleZeroFilter* PoleZeroFilter::Create(const float* numerator_coefficients, size_t order_denominator) { if (order_numerator > kMaxFilterOrder || order_denominator > kMaxFilterOrder || denominator_coefficients[0] == 0 || - numerator_coefficients == NULL || denominator_coefficients == NULL) - return NULL; + numerator_coefficients == nullptr || denominator_coefficients == nullptr) + return nullptr; return new PoleZeroFilter(numerator_coefficients, order_numerator, denominator_coefficients, order_denominator); } @@ -66,7 +66,7 @@ static float FilterArPast(const T* past, int PoleZeroFilter::Filter(const int16_t* in, size_t num_input_samples, float* output) { - if (in == NULL || output == NULL) + if (in == nullptr || output == nullptr) return -1; // This is the typical case, just a memcpy. const size_t k = std::min(num_input_samples, highest_order_); diff --git a/modules/audio_processing/vad/standalone_vad_unittest.cc b/modules/audio_processing/vad/standalone_vad_unittest.cc index 0fa2ed78b1..139f49a235 100644 --- a/modules/audio_processing/vad/standalone_vad_unittest.cc +++ b/modules/audio_processing/vad/standalone_vad_unittest.cc @@ -66,11 +66,11 @@ TEST(StandaloneVadTest, ActivityDetection) { FILE* pcm_file = fopen(test::ResourcePath("audio_processing/agc/agc_audio", "pcm").c_str(), "rb"); - ASSERT_TRUE(pcm_file != NULL); + ASSERT_TRUE(pcm_file != nullptr); FILE* reference_file = fopen( test::ResourcePath("audio_processing/agc/agc_vad", "dat").c_str(), "rb"); - ASSERT_TRUE(reference_file != NULL); + ASSERT_TRUE(reference_file != nullptr); // Reference activities are prepared with 0 aggressiveness. ASSERT_EQ(0, vad->set_mode(0)); diff --git a/modules/audio_processing/vad/vad_audio_proc_unittest.cc b/modules/audio_processing/vad/vad_audio_proc_unittest.cc index 0afed84c35..9fe3dc6c03 100644 --- a/modules/audio_processing/vad/vad_audio_proc_unittest.cc +++ b/modules/audio_processing/vad/vad_audio_proc_unittest.cc @@ -31,12 +31,12 @@ TEST(AudioProcessingTest, DISABLED_ComputingFirstSpectralPeak) { std::string peak_file_name = test::ResourcePath("audio_processing/agc/agc_spectral_peak", "dat"); FILE* peak_file = fopen(peak_file_name.c_str(), "rb"); - ASSERT_TRUE(peak_file != NULL); + ASSERT_TRUE(peak_file != nullptr); std::string pcm_file_name = test::ResourcePath("audio_processing/agc/agc_audio", "pcm"); FILE* pcm_file = fopen(pcm_file_name.c_str(), "rb"); - ASSERT_TRUE(pcm_file != NULL); + ASSERT_TRUE(pcm_file != nullptr); // Read 10 ms audio in each iteration. const size_t kDataLength = kLength10Ms; diff --git a/modules/audio_processing/vad/vad_circular_buffer.cc b/modules/audio_processing/vad/vad_circular_buffer.cc index 31f14d7f64..e05b0a584f 100644 --- a/modules/audio_processing/vad/vad_circular_buffer.cc +++ b/modules/audio_processing/vad/vad_circular_buffer.cc @@ -31,7 +31,7 @@ void VadCircularBuffer::Reset() { VadCircularBuffer* VadCircularBuffer::Create(int buffer_size) { if (buffer_size <= 0) - return NULL; + return nullptr; return new VadCircularBuffer(buffer_size); } From a1ccbe3441c7f0ce7222ae05d7d4ef7737368329 Mon Sep 17 00:00:00 2001 From: Mirko Bonadei Date: Tue, 20 May 2025 12:43:44 +0000 Subject: [PATCH 221/292] Remove tools_webrtc/matlab. This code has not been used for almost 10 years. Bug: webrtc:418969535 Change-Id: If8660f65e8a0e069facb0045e960da531583afb3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392501 Commit-Queue: Mirko Bonadei Reviewed-by: Henrik Lundin Commit-Queue: Henrik Lundin Auto-Submit: Mirko Bonadei Cr-Commit-Position: refs/heads/main@{#44693} --- tools_webrtc/matlab/maxUnwrap.m | 26 ---- tools_webrtc/matlab/parseLog.m | 55 ------- tools_webrtc/matlab/rtpAnalyze.m | 252 ------------------------------- 3 files changed, 333 deletions(-) delete mode 100644 tools_webrtc/matlab/maxUnwrap.m delete mode 100644 tools_webrtc/matlab/parseLog.m delete mode 100644 tools_webrtc/matlab/rtpAnalyze.m diff --git a/tools_webrtc/matlab/maxUnwrap.m b/tools_webrtc/matlab/maxUnwrap.m deleted file mode 100644 index cda48278b8..0000000000 --- a/tools_webrtc/matlab/maxUnwrap.m +++ /dev/null @@ -1,26 +0,0 @@ -% // clang-format off -function sequence = maxUnwrap(sequence, max) -% -% sequence = maxUnwrap(sequence, max) -% Unwraps when a wrap around is detected. -% -% Arguments -% -% sequence: The vector to unwrap. -% max: The maximum value that the sequence can take, -% and after which it will wrap over to 0. -% -% Return value -% -% sequence: The unwrapped vector. -% - -% Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. -% -% Use of this source code is governed by a BSD-style license -% that can be found in the LICENSE file in the root of the source -% tree. An additional intellectual property rights grant can be found -% in the file PATENTS. All contributing project authors may -% be found in the AUTHORS file in the root of the source tree. - -sequence = round((unwrap(2 * pi * sequence / max) * max) / (2 * pi)); diff --git a/tools_webrtc/matlab/parseLog.m b/tools_webrtc/matlab/parseLog.m deleted file mode 100644 index 62c9388621..0000000000 --- a/tools_webrtc/matlab/parseLog.m +++ /dev/null @@ -1,55 +0,0 @@ -% // clang-format off -function parsed = parseLog(filename) -% -% parsed = parseLog(filename) -% Parses a DataLog text file, with the filename specified in the string -% filename, into a struct with each column name as a field, and with the -% column data stored as a vector in that field. -% -% Arguments -% -% filename: A string with the name of the file to parse. -% -% Return value -% -% parsed: A struct containing each column parsed from the input file -% as a field and with the column data stored as a vector in that -% field. -% - -% Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. -% -% Use of this source code is governed by a BSD-style license -% that can be found in the LICENSE file in the root of the source -% tree. An additional intellectual property rights grant can be found -% in the file PATENTS. All contributing project authors may -% be found in the AUTHORS file in the root of the source tree. - -table = importdata(filename, ',', 1); -if ~isstruct(table) - error('Malformed file, possibly empty or lacking data entries') -end - -colheaders = table.textdata; -if length(colheaders) == 1 - colheaders = regexp(table.textdata{1}, ',', 'split'); -end - -parsed = struct; -i = 1; -while i <= length(colheaders) - % Checking for a multi-value column. - m = regexp(colheaders{i}, '([\w\t]+)\[(\d+)\]', 'tokens'); - if ~isempty(m) - % Parse a multi-value column - n = str2double(m{1}{2}) - 1; - parsed.(strrep(m{1}{1}, ' ', '_')) = table.data(:, i:i+n); - i = i + n + 1; - elseif ~isempty(colheaders{i}) - % Parse a single-value column - parsed.(strrep(colheaders{i}, ' ', '_')) = table.data(:, i); - i = i + 1; - else - error('Empty column'); - end -end diff --git a/tools_webrtc/matlab/rtpAnalyze.m b/tools_webrtc/matlab/rtpAnalyze.m deleted file mode 100644 index c64756579e..0000000000 --- a/tools_webrtc/matlab/rtpAnalyze.m +++ /dev/null @@ -1,252 +0,0 @@ -% // clang-format off -function rtpAnalyze( input_file ) -%RTP_ANALYZE Analyze RTP stream(s) from a txt file -% The function takes the output from the command line tool rtp_analyze -% and analyzes the stream(s) therein. First, process your rtpdump file -% through rtp_analyze (from command line): -% $ out/Debug/rtp_analyze my_file.rtp my_file.txt -% Then load it with this function (in Matlab): -% >> rtpAnalyze('my_file.txt') - -% Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. -% -% Use of this source code is governed by a BSD-style license -% that can be found in the LICENSE file in the root of the source -% tree. An additional intellectual property rights grant can be found -% in the file PATENTS. All contributing project authors may -% be found in the AUTHORS file in the root of the source tree. - -[SeqNo,TimeStamp,ArrTime,Size,PT,M,SSRC] = importfile(input_file); - -%% Filter out RTCP packets. -% These appear as RTP packets having payload types 72 through 76. -ix = not(ismember(PT, 72:76)); -fprintf('Removing %i RTCP packets\n', length(SeqNo) - sum(ix)); -SeqNo = SeqNo(ix); -TimeStamp = TimeStamp(ix); -ArrTime = ArrTime(ix); -Size = Size(ix); -PT = PT(ix); -M = M(ix); -SSRC = SSRC(ix); - -%% Find streams. -[uSSRC, ~, uix] = unique(SSRC); - -% If there are multiple streams, select one and purge the other -% streams from the data vectors. If there is only one stream, the -% vectors are good to use as they are. -if length(uSSRC) > 1 - for i=1:length(uSSRC) - uPT = unique(PT(uix == i)); - fprintf('%i: %s (%d packets, pt: %i', i, uSSRC{i}, ... - length(find(uix==i)), uPT(1)); - if length(uPT) > 1 - fprintf(', %i', uPT(2:end)); - end - fprintf(')\n'); - end - sel = input('Select stream number: '); - if sel < 1 || sel > length(uSSRC) - error('Out of range'); - end - ix = find(uix == sel); - % This is where the data vectors are trimmed. - SeqNo = SeqNo(ix); - TimeStamp = TimeStamp(ix); - ArrTime = ArrTime(ix); - Size = Size(ix); - PT = PT(ix); - M = M(ix); - SSRC = SSRC(ix); -end - -%% Unwrap SeqNo and TimeStamp. -SeqNoUW = maxUnwrap(SeqNo, 65535); -TimeStampUW = maxUnwrap(TimeStamp, 4294967295); - -%% Generate some stats for the stream. -fprintf('Statistics:\n'); -fprintf('SSRC: %s\n', SSRC{1}); -uPT = unique(PT); -if length(uPT) > 1 - warning('This tool cannot yet handle changes in codec sample rate'); -end -fprintf('Payload type(s): %i', uPT(1)); -if length(uPT) > 1 - fprintf(', %i', uPT(2:end)); -end -fprintf('\n'); -fprintf('Packets: %i\n', length(SeqNo)); -SortSeqNo = sort(SeqNoUW); -fprintf('Missing sequence numbers: %i\n', ... - length(find(diff(SortSeqNo) > 1))); -fprintf('Duplicated packets: %i\n', length(find(diff(SortSeqNo) == 0))); -reorderIx = findReorderedPackets(SeqNoUW); -fprintf('Reordered packets: %i\n', length(reorderIx)); -tsdiff = diff(TimeStampUW); -tsdiff = tsdiff(diff(SeqNoUW) == 1); -[utsdiff, ~, ixtsdiff] = unique(tsdiff); -fprintf('Common packet sizes:\n'); -for i = 1:length(utsdiff) - fprintf(' %i samples (%i%%)\n', ... - utsdiff(i), ... - round(100 * length(find(ixtsdiff == i))/length(ixtsdiff))); -end - -%% Trying to figure out sample rate. -fs_est = (TimeStampUW(end) - TimeStampUW(1)) / (ArrTime(end) - ArrTime(1)); -fs_vec = [8, 16, 32, 48]; -fs = 0; -for f = fs_vec - if abs((fs_est-f)/f) < 0.05 % 5% margin - fs = f; - break; - end -end -if fs == 0 - fprintf('Cannot determine sample rate. I get it to %.2f kHz\n', ... - fs_est); - fs = input('Please, input a sample rate (in kHz): '); -else - fprintf('Sample rate estimated to %i kHz\n', fs); -end - -SendTimeMs = (TimeStampUW - TimeStampUW(1)) / fs; - -fprintf('Stream duration at sender: %.1f seconds\n', ... - (SendTimeMs(end) - SendTimeMs(1)) / 1000); - -fprintf('Stream duration at receiver: %.1f seconds\n', ... - (ArrTime(end) - ArrTime(1)) / 1000); - -fprintf('Clock drift: %.2f%%\n', ... - 100 * ((ArrTime(end) - ArrTime(1)) / ... - (SendTimeMs(end) - SendTimeMs(1)) - 1)); - -fprintf('Sent average bitrate: %i kbps\n', ... - round(sum(Size) * 8 / (SendTimeMs(end)-SendTimeMs(1)))); - -fprintf('Received average bitrate: %i kbps\n', ... - round(sum(Size) * 8 / (ArrTime(end)-ArrTime(1)))); - -%% Plots. -delay = ArrTime - SendTimeMs; -delay = delay - min(delay); -delayOrdered = delay; -delayOrdered(reorderIx) = nan; % Set reordered packets to NaN. -delayReordered = delay(reorderIx); % Pick the reordered packets. -sendTimeMsReordered = SendTimeMs(reorderIx); - -% Sort time arrays in packet send order. -[~, sortix] = sort(SeqNoUW); -SendTimeMs = SendTimeMs(sortix); -Size = Size(sortix); -delayOrdered = delayOrdered(sortix); - -figure -plot(SendTimeMs / 1000, delayOrdered, ... - sendTimeMsReordered / 1000, delayReordered, 'r.'); -xlabel('Send time [s]'); -ylabel('Relative transport delay [ms]'); -title(sprintf('SSRC: %s', SSRC{1})); - -SendBitrateKbps = 8 * Size(1:end-1) ./ diff(SendTimeMs); -figure -plot(SendTimeMs(1:end-1)/1000, SendBitrateKbps); -xlabel('Send time [s]'); -ylabel('Send bitrate [kbps]'); -end - -%% Subfunctions. - -% findReorderedPackets returns the index to all packets that are considered -% old compared with the largest seen sequence number. The input seqNo must -% be unwrapped for this to work. -function reorderIx = findReorderedPackets(seqNo) -largestSeqNo = seqNo(1); -reorderIx = []; -for i = 2:length(seqNo) - if seqNo(i) < largestSeqNo - reorderIx = [reorderIx; i]; %#ok - else - largestSeqNo = seqNo(i); - end -end -end - -%% Auto-generated subfunction. -function [SeqNo,TimeStamp,SendTime,Size,PT,M,SSRC] = ... - importfile(filename, startRow, endRow) -%IMPORTFILE Import numeric data from a text file as column vectors. -% [SEQNO,TIMESTAMP,SENDTIME,SIZE,PT,M,SSRC] = IMPORTFILE(FILENAME) Reads -% data from text file FILENAME for the default selection. -% -% [SEQNO,TIMESTAMP,SENDTIME,SIZE,PT,M,SSRC] = IMPORTFILE(FILENAME, -% STARTROW, ENDROW) Reads data from rows STARTROW through ENDROW of text -% file FILENAME. -% -% Example: -% [SeqNo,TimeStamp,SendTime,Size,PT,M,SSRC] = -% importfile('rtpdump_recv.txt',2, 123); -% -% See also TEXTSCAN. - -% Auto-generated by MATLAB on 2015/05/28 09:55:50 - -%% Initialize variables. -if nargin<=2 - startRow = 2; - endRow = inf; -end - -%% Format string for each line of text: -% column1: double (%f) -% column2: double (%f) -% column3: double (%f) -% column4: double (%f) -% column5: double (%f) -% column6: double (%f) -% column7: text (%s) -% For more information, see the TEXTSCAN documentation. -formatSpec = '%5f%11f%11f%6f%6f%3f%s%[^\n\r]'; - -%% Open the text file. -fileID = fopen(filename,'r'); - -%% Read columns of data according to format string. -% This call is based on the structure of the file used to generate this -% code. If an error occurs for a different file, try regenerating the code -% from the Import Tool. -dataArray = textscan(fileID, formatSpec, endRow(1)-startRow(1)+1, ... - 'Delimiter', '', 'WhiteSpace', '', 'HeaderLines', startRow(1)-1, ... - 'ReturnOnError', false); -for block=2:length(startRow) - frewind(fileID); - dataArrayBlock = textscan(fileID, formatSpec, ... - endRow(block)-startRow(block)+1, 'Delimiter', '', 'WhiteSpace', ... - '', 'HeaderLines', startRow(block)-1, 'ReturnOnError', false); - for col=1:length(dataArray) - dataArray{col} = [dataArray{col};dataArrayBlock{col}]; - end -end - -%% Close the text file. -fclose(fileID); - -%% Post processing for unimportable data. -% No unimportable data rules were applied during the import, so no post -% processing code is included. To generate code which works for -% unimportable data, select unimportable cells in a file and regenerate the -% script. - -%% Allocate imported array to column variable names -SeqNo = dataArray{:, 1}; -TimeStamp = dataArray{:, 2}; -SendTime = dataArray{:, 3}; -Size = dataArray{:, 4}; -PT = dataArray{:, 5}; -M = dataArray{:, 6}; -SSRC = dataArray{:, 7}; -end - From bdfa3ab007bc4dc8b1c265f67f55573b7c13ebd7 Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Mon, 19 May 2025 12:00:16 +0000 Subject: [PATCH 222/292] Cleanup arraysize usage in media/ Bug: webrtc:407433458 Change-Id: I3ded72cdc8d84dddae676f5ba769029382b65d11 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392460 Auto-Submit: Danil Chapovalov Commit-Queue: Danil Chapovalov Commit-Queue: Rasmus Brandt Reviewed-by: Rasmus Brandt Cr-Commit-Position: refs/heads/main@{#44694} --- media/BUILD.gn | 2 -- media/base/stream_params_unittest.cc | 36 ++++++++------------ media/base/test_utils.h | 3 +- media/base/video_adapter_unittest.cc | 7 ++-- media/base/video_common.cc | 7 ++-- media/engine/webrtc_video_engine_unittest.cc | 19 +++++------ media/engine/webrtc_voice_engine_unittest.cc | 14 +++----- 7 files changed, 35 insertions(+), 53 deletions(-) diff --git a/media/BUILD.gn b/media/BUILD.gn index 0ba26c7e82..a11de124ce 100644 --- a/media/BUILD.gn +++ b/media/BUILD.gn @@ -221,7 +221,6 @@ rtc_library("video_common") { deps = [ "../api:array_view", "../rtc_base:checks", - "../rtc_base:macromagic", "../rtc_base:stringutils", "../rtc_base:timeutils", "../rtc_base/system:rtc_export", @@ -1032,7 +1031,6 @@ if (rtc_include_tests) { "../rtc_base:checks", "../rtc_base:copy_on_write_buffer", "../rtc_base:dscp", - "../rtc_base:macromagic", "../rtc_base:safe_conversions", "../rtc_base:socket", "../rtc_base:threading", diff --git a/media/base/stream_params_unittest.cc b/media/base/stream_params_unittest.cc index b89c5b9133..2b9a0f3243 100644 --- a/media/base/stream_params_unittest.cc +++ b/media/base/stream_params_unittest.cc @@ -16,8 +16,8 @@ #include #include +#include "api/array_view.h" #include "media/base/test_utils.h" -#include "rtc_base/arraysize.h" #include "rtc_base/unique_id_generator.h" #include "test/gmock.h" #include "test/gtest.h" @@ -30,10 +30,9 @@ static const uint32_t kSsrcs2[] = {1, 2}; static webrtc::StreamParams CreateStreamParamsWithSsrcGroup( const std::string& semantics, - const uint32_t ssrcs_in[], - size_t len) { + webrtc::ArrayView ssrcs_in) { webrtc::StreamParams stream; - std::vector ssrcs(ssrcs_in, ssrcs_in + len); + std::vector ssrcs(ssrcs_in.begin(), ssrcs_in.end()); webrtc::SsrcGroup sg(semantics, ssrcs); stream.ssrcs = ssrcs; stream.ssrc_groups.push_back(sg); @@ -48,8 +47,8 @@ TEST(SsrcGroup, EqualNotEqual) { webrtc::SsrcGroup("abc", MAKE_VECTOR(kSsrcs2)), }; - for (size_t i = 0; i < arraysize(ssrc_groups); ++i) { - for (size_t j = 0; j < arraysize(ssrc_groups); ++j) { + for (size_t i = 0; i < std::size(ssrc_groups); ++i) { + for (size_t j = 0; j < std::size(ssrc_groups); ++j) { EXPECT_EQ((ssrc_groups[i] == ssrc_groups[j]), (i == j)); EXPECT_EQ((ssrc_groups[i] != ssrc_groups[j]), (i != j)); } @@ -85,8 +84,7 @@ TEST(StreamParams, CreateLegacy) { } TEST(StreamParams, HasSsrcGroup) { - webrtc::StreamParams sp = - CreateStreamParamsWithSsrcGroup("XYZ", kSsrcs2, arraysize(kSsrcs2)); + webrtc::StreamParams sp = CreateStreamParamsWithSsrcGroup("XYZ", kSsrcs2); EXPECT_EQ(2U, sp.ssrcs.size()); EXPECT_EQ(kSsrcs2[0], sp.first_ssrc()); EXPECT_TRUE(sp.has_ssrcs()); @@ -100,8 +98,7 @@ TEST(StreamParams, HasSsrcGroup) { } TEST(StreamParams, GetSsrcGroup) { - webrtc::StreamParams sp = - CreateStreamParamsWithSsrcGroup("XYZ", kSsrcs2, arraysize(kSsrcs2)); + webrtc::StreamParams sp = CreateStreamParamsWithSsrcGroup("XYZ", kSsrcs2); EXPECT_EQ(nullptr, sp.get_ssrc_group("xyz")); EXPECT_EQ(&sp.ssrc_groups[0], sp.get_ssrc_group("XYZ")); } @@ -120,18 +117,14 @@ TEST(StreamParams, HasStreamWithNoSsrcs) { TEST(StreamParams, EqualNotEqual) { webrtc::StreamParams l1 = webrtc::StreamParams::CreateLegacy(1); webrtc::StreamParams l2 = webrtc::StreamParams::CreateLegacy(2); - webrtc::StreamParams sg1 = - CreateStreamParamsWithSsrcGroup("ABC", kSsrcs1, arraysize(kSsrcs1)); - webrtc::StreamParams sg2 = - CreateStreamParamsWithSsrcGroup("ABC", kSsrcs2, arraysize(kSsrcs2)); - webrtc::StreamParams sg3 = - CreateStreamParamsWithSsrcGroup("Abc", kSsrcs2, arraysize(kSsrcs2)); - webrtc::StreamParams sg4 = - CreateStreamParamsWithSsrcGroup("abc", kSsrcs2, arraysize(kSsrcs2)); + webrtc::StreamParams sg1 = CreateStreamParamsWithSsrcGroup("ABC", kSsrcs1); + webrtc::StreamParams sg2 = CreateStreamParamsWithSsrcGroup("ABC", kSsrcs2); + webrtc::StreamParams sg3 = CreateStreamParamsWithSsrcGroup("Abc", kSsrcs2); + webrtc::StreamParams sg4 = CreateStreamParamsWithSsrcGroup("abc", kSsrcs2); webrtc::StreamParams sps[] = {l1, l2, sg1, sg2, sg3, sg4}; - for (size_t i = 0; i < arraysize(sps); ++i) { - for (size_t j = 0; j < arraysize(sps); ++j) { + for (size_t i = 0; i < std::size(sps); ++i) { + for (size_t j = 0; j < std::size(sps); ++j) { EXPECT_EQ((sps[i] == sps[j]), (i == j)); EXPECT_EQ((sps[i] != sps[j]), (i != j)); } @@ -227,8 +220,7 @@ TEST(StreamParams, FecFrFunctions) { } TEST(StreamParams, ToString) { - webrtc::StreamParams sp = - CreateStreamParamsWithSsrcGroup("XYZ", kSsrcs2, arraysize(kSsrcs2)); + webrtc::StreamParams sp = CreateStreamParamsWithSsrcGroup("XYZ", kSsrcs2); sp.set_stream_ids({"stream_id"}); EXPECT_STREQ( "{ssrcs:[1,2];ssrc_groups:{semantics:XYZ;ssrcs:[1,2]};stream_ids:stream_" diff --git a/media/base/test_utils.h b/media/base/test_utils.h index 94f4492752..403b4922b0 100644 --- a/media/base/test_utils.h +++ b/media/base/test_utils.h @@ -17,7 +17,6 @@ #include #include "media/base/stream_params.h" -#include "rtc_base/arraysize.h" namespace webrtc { class VideoFrame; @@ -34,7 +33,7 @@ template inline std::vector MakeVector(const T a[], size_t s) { return std::vector(a, a + s); } -#define MAKE_VECTOR(a) webrtc::MakeVector(a, arraysize(a)) +#define MAKE_VECTOR(a) webrtc::MakeVector(a, std::size(a)) // Create Simulcast StreamParams with given `ssrcs` and `cname`. StreamParams CreateSimStreamParams(const std::string& cname, diff --git a/media/base/video_adapter_unittest.cc b/media/base/video_adapter_unittest.cc index 98b699657e..8b4a158095 100644 --- a/media/base/video_adapter_unittest.cc +++ b/media/base/video_adapter_unittest.cc @@ -23,7 +23,6 @@ #include "api/video/video_source_interface.h" #include "media/base/fake_frame_source.h" #include "media/base/video_common.h" -#include "rtc_base/arraysize.h" #include "rtc_base/time_utils.h" #include "test/field_trial.h" #include "test/gmock.h" @@ -1104,7 +1103,7 @@ TEST_P(VideoAdapterTest, AdaptResolutionInStepsFirst3_4) { int request_width = kWidth; int request_height = kHeight; - for (size_t i = 0; i < arraysize(kExpectedWidths); ++i) { + for (size_t i = 0; i < std::size(kExpectedWidths); ++i) { // Adapt down one step. adapter_.OnSinkWants(BuildSinkWants(std::nullopt, request_width * request_height - 1, @@ -1133,7 +1132,7 @@ TEST_P(VideoAdapterTest, AdaptResolutionInStepsFirst2_3) { int request_width = kWidth1080p; int request_height = kHeight1080p; - for (size_t i = 0; i < arraysize(kExpectedWidths); ++i) { + for (size_t i = 0; i < std::size(kExpectedWidths); ++i) { // Adapt down one step. adapter_.OnSinkWants(BuildSinkWants(std::nullopt, request_width * request_height - 1, @@ -1162,7 +1161,7 @@ TEST_P(VideoAdapterTest, AdaptResolutionInStepsFirst2x2_3) { int request_width = kWidth1080p4to3; int request_height = kHeight1080p4to3; - for (size_t i = 0; i < arraysize(kExpectedWidths); ++i) { + for (size_t i = 0; i < std::size(kExpectedWidths); ++i) { // Adapt down one step. adapter_.OnSinkWants(BuildSinkWants(std::nullopt, request_width * request_height - 1, diff --git a/media/base/video_common.cc b/media/base/video_common.cc index bc18d85b70..c5abb9e6e5 100644 --- a/media/base/video_common.cc +++ b/media/base/video_common.cc @@ -15,7 +15,6 @@ #include #include "api/array_view.h" -#include "rtc_base/arraysize.h" #include "rtc_base/checks.h" #include "rtc_base/strings/string_builder.h" @@ -44,9 +43,9 @@ static const FourCCAliasEntry kFourCCAliases[] = { }; uint32_t CanonicalFourCC(uint32_t fourcc) { - for (uint32_t i = 0; i < arraysize(kFourCCAliases); ++i) { - if (kFourCCAliases[i].alias == fourcc) { - return kFourCCAliases[i].canonical; + for (const FourCCAliasEntry& entry : kFourCCAliases) { + if (entry.alias == fourcc) { + return entry.canonical; } } // Not an alias, so return it as-is. diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc index ffedad59b8..31b9b4f455 100644 --- a/media/engine/webrtc_video_engine_unittest.cc +++ b/media/engine/webrtc_video_engine_unittest.cc @@ -107,7 +107,6 @@ #include "modules/rtp_rtcp/source/rtp_packet.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "modules/video_coding/svc/scalability_mode_util.h" -#include "rtc_base/arraysize.h" #include "rtc_base/async_packet_socket.h" #include "rtc_base/checks.h" #include "rtc_base/copy_on_write_buffer.h" @@ -3325,21 +3324,21 @@ TEST_F(WebRtcVideoChannelTest, TEST_F(WebRtcVideoChannelTest, SetSendRtpHeaderExtensionsRejectsIncorrectIds) { const int kIncorrectIds[] = {-2, -1, 0, 15, 16}; - for (size_t i = 0; i < arraysize(kIncorrectIds); ++i) { + for (int incorrect_id : kIncorrectIds) { send_parameters_.extensions.push_back( - RtpExtension(RtpExtension::kTimestampOffsetUri, kIncorrectIds[i])); + RtpExtension(RtpExtension::kTimestampOffsetUri, incorrect_id)); EXPECT_FALSE(send_channel_->SetSenderParameters(send_parameters_)) - << "Bad extension id '" << kIncorrectIds[i] << "' accepted."; + << "Bad extension id '" << incorrect_id << "' accepted."; } } TEST_F(WebRtcVideoChannelTest, SetRecvRtpHeaderExtensionsRejectsIncorrectIds) { const int kIncorrectIds[] = {-2, -1, 0, 15, 16}; - for (size_t i = 0; i < arraysize(kIncorrectIds); ++i) { + for (int incorrect_id : kIncorrectIds) { recv_parameters_.extensions.push_back( - RtpExtension(RtpExtension::kTimestampOffsetUri, kIncorrectIds[i])); + RtpExtension(RtpExtension::kTimestampOffsetUri, incorrect_id)); EXPECT_FALSE(receive_channel_->SetReceiverParameters(recv_parameters_)) - << "Bad extension id '" << kIncorrectIds[i] << "' accepted."; + << "Bad extension id '" << incorrect_id << "' accepted."; } } @@ -5242,10 +5241,10 @@ TEST_F(WebRtcVideoChannelTest, SetSendCodecsRejectBadPayloadTypes) { static const int kIncorrectPayloads[] = {-2, -1, 128, 129}; VideoSenderParameters parameters; parameters.codecs.push_back(GetEngineCodec("VP8")); - for (size_t i = 0; i < arraysize(kIncorrectPayloads); ++i) { - parameters.codecs[0].id = kIncorrectPayloads[i]; + for (int incorrect_id : kIncorrectPayloads) { + parameters.codecs[0].id = incorrect_id; EXPECT_FALSE(send_channel_->SetSenderParameters(parameters)) - << "Bad payload type '" << kIncorrectPayloads[i] << "' accepted."; + << "Bad payload type '" << incorrect_id << "' accepted."; } } diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc index c6c44448e9..3bdd0f7250 100644 --- a/media/engine/webrtc_voice_engine_unittest.cc +++ b/media/engine/webrtc_voice_engine_unittest.cc @@ -66,7 +66,6 @@ #include "modules/rtp_rtcp/include/rtp_header_extension_map.h" #include "modules/rtp_rtcp/source/rtp_header_extensions.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" -#include "rtc_base/arraysize.h" #include "rtc_base/byte_order.h" #include "rtc_base/checks.h" #include "rtc_base/copy_on_write_buffer.h" @@ -2458,7 +2457,7 @@ TEST_P(WebRtcVoiceEngineTestFake, CreateAndDeleteMultipleSendStreams) { // Verify that we are in a sending state for all the created streams. EXPECT_TRUE(GetSendStream(ssrc).IsSending()); } - EXPECT_EQ(arraysize(kSsrcs4), call_.GetAudioSendStreams().size()); + EXPECT_EQ(std::size(kSsrcs4), call_.GetAudioSendStreams().size()); // Delete the send streams. for (uint32_t ssrc : kSsrcs4) { @@ -2566,8 +2565,7 @@ TEST_P(WebRtcVoiceEngineTestFake, GetStatsWithMultipleSendStreams) { &receive_info, /*get_and_clear_legacy_stats=*/true)); // We have added 4 send streams. We should see empty stats for all. - EXPECT_EQ(static_cast(arraysize(kSsrcs4)), - send_info.senders.size()); + EXPECT_EQ(std::size(kSsrcs4), send_info.senders.size()); for (const auto& sender : send_info.senders) { VerifyVoiceSenderInfo(sender, false); } @@ -2587,8 +2585,7 @@ TEST_P(WebRtcVoiceEngineTestFake, GetStatsWithMultipleSendStreams) { EXPECT_EQ(true, send_channel_->GetStats(&send_info)); EXPECT_EQ(true, receive_channel_->GetStats( &receive_info, /*get_and_clear_legacy_stats=*/true)); - EXPECT_EQ(static_cast(arraysize(kSsrcs4)), - send_info.senders.size()); + EXPECT_EQ(std::size(kSsrcs4), send_info.senders.size()); EXPECT_EQ(0u, receive_info.receivers.size()); } @@ -2603,8 +2600,7 @@ TEST_P(WebRtcVoiceEngineTestFake, GetStatsWithMultipleSendStreams) { EXPECT_EQ(true, send_channel_->GetStats(&send_info)); EXPECT_EQ(true, receive_channel_->GetStats( &receive_info, /*get_and_clear_legacy_stats=*/true)); - EXPECT_EQ(static_cast(arraysize(kSsrcs4)), - send_info.senders.size()); + EXPECT_EQ(std::size(kSsrcs4), send_info.senders.size()); EXPECT_EQ(1u, receive_info.receivers.size()); VerifyVoiceReceiverInfo(receive_info.receivers[0]); VerifyVoiceSendRecvCodecs(send_info, receive_info); @@ -2813,7 +2809,7 @@ TEST_P(WebRtcVoiceEngineTestFake, RecvWithMultipleStreams) { EXPECT_TRUE(AddRecvStream(ssrc3)); // Create packets with the right SSRCs. unsigned char packets[4][sizeof(kPcmuFrame)]; - for (size_t i = 0; i < arraysize(packets); ++i) { + for (size_t i = 0; i < std::size(packets); ++i) { memcpy(packets[i], kPcmuFrame, sizeof(kPcmuFrame)); webrtc::SetBE32(packets[i] + 8, static_cast(i)); } From 73e837b140a39a612fc8aeda198e201735eb4ead Mon Sep 17 00:00:00 2001 From: Guido Urdaneta Date: Tue, 20 May 2025 15:25:10 +0200 Subject: [PATCH 223/292] Add setter methods to CaptureTime and PayloadType and AudioLevel in FrameTransformerInterface/TransformableAudioFrameInterface. This CL adds setter methods and "CanSet" methods. The CanSet methods are necessary to properly support validation at the call sites (i.e., Chromium). They make it easier to validate that a set of changes can be supported in advance, rather using error codes after individual properties have been set. This CL adds support for these setters for sender audio frames. Bug: chromium:402391734 No-Iwyu: No output from iwyu/apply-include-cleaner Change-Id: Ifaf2e91630c786c716b31cdab320c3d6032ed609 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/388100 Reviewed-by: Tomas Gunnarsson Commit-Queue: Guido Urdaneta Cr-Commit-Position: refs/heads/main@{#44695} --- api/frame_transformer_interface.h | 18 +++++++++-- api/test/mock_transformable_audio_frame.h | 6 ++++ api/test/mock_transformable_frame.h | 4 +++ api/test/mock_transformable_video_frame.h | 4 +++ ...channel_send_frame_transformer_delegate.cc | 25 ++++++++++++++- ...end_frame_transformer_delegate_unittest.cc | 32 +++++++++++++++++++ 6 files changed, 85 insertions(+), 4 deletions(-) diff --git a/api/frame_transformer_interface.h b/api/frame_transformer_interface.h index 873bf6dce3..db3a8cdac3 100644 --- a/api/frame_transformer_interface.h +++ b/api/frame_transformer_interface.h @@ -51,6 +51,8 @@ class TransformableFrameInterface { virtual void SetData(ArrayView data) = 0; virtual uint8_t GetPayloadType() const = 0; + virtual bool CanSetPayloadType() const { return false; } + virtual void SetPayloadType(uint8_t payload_type) { RTC_DCHECK_NOTREACHED(); } virtual uint32_t GetSsrc() const = 0; virtual uint32_t GetTimestamp() const = 0; virtual void SetRTPTimestamp(uint32_t timestamp) = 0; @@ -85,11 +87,17 @@ class TransformableFrameInterface { virtual std::optional ReceiveTime() const = 0; // Timestamp at which the frame was captured in the capturer system. - // The timestamp is expressed in the capturer system's clock relative to the - // NTP epoch (January 1st 1970 00:00 UTC) - // Accessible only if the absolute capture timestamp header extension is + // For receiver frames, the timestamp is expressed in the capturer system's + // clock relative to the NTP epoch (January 1st 1970 00:00 UTC) and is + // available only if the absolute capture timestamp header extension is // enabled. + // For sender frames, the timestamp is expressed relative to the local + // system clock's default epoch. virtual std::optional CaptureTime() const = 0; + virtual bool CanSetCaptureTime() const { return false; } + virtual void SetCaptureTime(std::optional capture_time) { + RTC_DCHECK_NOTREACHED(); + } // Offset between the sender system's clock and the capturer system's clock. // Can be used to express the capture time in the local system's clock as @@ -134,6 +142,10 @@ class TransformableAudioFrameInterface : public TransformableFrameInterface { // dBov. 127 represents digital silence. Only present on remote frames if // the audio level header extension was included. virtual std::optional AudioLevel() const = 0; + virtual bool CanSetAudioLevel() const { return false; } + virtual void SetAudioLevel(std::optional audio_level_dbov) { + RTC_DCHECK_NOTREACHED(); + } }; // Objects implement this interface to be notified with the transformed frame. diff --git a/api/test/mock_transformable_audio_frame.h b/api/test/mock_transformable_audio_frame.h index 552d109f24..c92624a0ad 100644 --- a/api/test/mock_transformable_audio_frame.h +++ b/api/test/mock_transformable_audio_frame.h @@ -31,6 +31,8 @@ class MockTransformableAudioFrame : public TransformableAudioFrameInterface { MOCK_METHOD(void, SetData, (webrtc::ArrayView), (override)); MOCK_METHOD(void, SetRTPTimestamp, (uint32_t), (override)); MOCK_METHOD(uint8_t, GetPayloadType, (), (const, override)); + MOCK_METHOD(bool, CanSetPayloadType, (), (const, override)); + MOCK_METHOD(void, SetPayloadType, (uint8_t), (override)); MOCK_METHOD(uint32_t, GetSsrc, (), (const, override)); MOCK_METHOD(uint32_t, GetTimestamp, (), (const, override)); MOCK_METHOD(std::string, GetMimeType, (), (const, override)); @@ -55,9 +57,13 @@ class MockTransformableAudioFrame : public TransformableAudioFrameInterface { (), (const, override)); MOCK_METHOD(std::optional, AudioLevel, (), (const, override)); + MOCK_METHOD(bool, CanSetAudioLevel, (), (const, override)); + MOCK_METHOD(void, SetAudioLevel, (std::optional), (override)); MOCK_METHOD(std::optional, ReceiveTime, (), (const, override)); MOCK_METHOD(std::optional, CaptureTime, (), (const, override)); + MOCK_METHOD(bool, CanSetCaptureTime, (), (const, override)); + MOCK_METHOD(void, SetCaptureTime, (std::optional), (override)); MOCK_METHOD(std::optional, SenderCaptureTimeOffset, (), diff --git a/api/test/mock_transformable_frame.h b/api/test/mock_transformable_frame.h index 9a1456d6fa..e79834068d 100644 --- a/api/test/mock_transformable_frame.h +++ b/api/test/mock_transformable_frame.h @@ -32,6 +32,8 @@ class MockTransformableFrame : public TransformableFrameInterface { MOCK_METHOD(ArrayView, GetData, (), (const, override)); MOCK_METHOD(void, SetData, (webrtc::ArrayView), (override)); MOCK_METHOD(uint8_t, GetPayloadType, (), (const, override)); + MOCK_METHOD(bool, CanSetPayloadType, (), (const, override)); + MOCK_METHOD(void, SetPayloadType, (uint8_t), (override)); MOCK_METHOD(uint32_t, GetSsrc, (), (const, override)); MOCK_METHOD(uint32_t, GetTimestamp, (), (const, override)); MOCK_METHOD(void, SetRTPTimestamp, (uint32_t), (override)); @@ -42,6 +44,8 @@ class MockTransformableFrame : public TransformableFrameInterface { MOCK_METHOD(std::string, GetMimeType, (), (const, override)); MOCK_METHOD(std::optional, ReceiveTime, (), (const, override)); MOCK_METHOD(std::optional, CaptureTime, (), (const, override)); + MOCK_METHOD(bool, CanSetCaptureTime, (), (const, override)); + MOCK_METHOD(void, SetCaptureTime, (std::optional), (override)); MOCK_METHOD(std::optional, SenderCaptureTimeOffset, (), diff --git a/api/test/mock_transformable_video_frame.h b/api/test/mock_transformable_video_frame.h index 8b13e16e95..61207c2ee9 100644 --- a/api/test/mock_transformable_video_frame.h +++ b/api/test/mock_transformable_video_frame.h @@ -43,6 +43,8 @@ class MockTransformableVideoFrame (const webrtc::VideoFrameMetadata&), (override)); MOCK_METHOD(uint8_t, GetPayloadType, (), (const, override)); + MOCK_METHOD(bool, CanSetPayloadType, (), (const, override)); + MOCK_METHOD(void, SetPayloadType, (uint8_t), (override)); MOCK_METHOD(TransformableFrameInterface::Direction, GetDirection, (), @@ -55,6 +57,8 @@ class MockTransformableVideoFrame (const, override)); MOCK_METHOD(std::optional, ReceiveTime, (), (const, override)); MOCK_METHOD(std::optional, CaptureTime, (), (const, override)); + MOCK_METHOD(bool, CanSetCaptureTime, (), (const, override)); + MOCK_METHOD(void, SetCaptureTime, (std::optional), (override)); MOCK_METHOD(std::optional, SenderCaptureTimeOffset, (), diff --git a/audio/channel_send_frame_transformer_delegate.cc b/audio/channel_send_frame_transformer_delegate.cc index 8a94ce4c7c..f1a8b4e37d 100644 --- a/audio/channel_send_frame_transformer_delegate.cc +++ b/audio/channel_send_frame_transformer_delegate.cc @@ -89,6 +89,10 @@ class TransformableOutgoingAudioFrame } uint8_t GetPayloadType() const override { return payload_type_; } + bool CanSetPayloadType() const override { return true; } + void SetPayloadType(uint8_t payload_type) override { + payload_type_ = payload_type; + } Direction GetDirection() const override { return Direction::kSender; } std::string GetMimeType() const override { return codec_mime_type_; } @@ -112,8 +116,27 @@ class TransformableOutgoingAudioFrame return audio_level_dbov_; } + bool CanSetAudioLevel() const override { return true; } + void SetAudioLevel(std::optional audio_level_dbov) override { + if (audio_level_dbov.has_value() && audio_level_dbov > 127u) { + audio_level_dbov = 127u; + } + audio_level_dbov_ = audio_level_dbov; + } + std::optional ReceiveTime() const override { return std::nullopt; } - std::optional CaptureTime() const override { return std::nullopt; } + + std::optional CaptureTime() const override { + return absolute_capture_timestamp_ms_ + ? std::make_optional( + Timestamp::Millis(*absolute_capture_timestamp_ms_)) + : std::nullopt; + } + bool CanSetCaptureTime() const override { return true; } + void SetCaptureTime(std::optional capture_time) override { + absolute_capture_timestamp_ms_ = + capture_time ? std::make_optional(capture_time->ms()) : std::nullopt; + } std::optional SenderCaptureTimeOffset() const override { return std::nullopt; } diff --git a/audio/channel_send_frame_transformer_delegate_unittest.cc b/audio/channel_send_frame_transformer_delegate_unittest.cc index e9575c1857..2eac45ca4f 100644 --- a/audio/channel_send_frame_transformer_delegate_unittest.cc +++ b/audio/channel_send_frame_transformer_delegate_unittest.cc @@ -278,5 +278,37 @@ TEST(ChannelSendFrameTransformerDelegateTest, CloningReceiverFrameWithCsrcs) { EXPECT_EQ(cloned_frame->AudioLevel(), frame->AudioLevel()); } +TEST(ChannelSendFrameTransformerDelegateTest, SetCaptureTime) { + std::unique_ptr frame = CreateFrame(); + EXPECT_TRUE(frame->CanSetCaptureTime()); + frame->SetCaptureTime(webrtc::Timestamp::Millis(100)); + EXPECT_EQ(frame->CaptureTime(), webrtc::Timestamp::Millis(100)); + frame->SetCaptureTime(std::nullopt); + EXPECT_FALSE(frame->CaptureTime().has_value()); +} + +TEST(ChannelSendFrameTransformerDelegateTest, SetPayloadType) { + std::unique_ptr frame = CreateFrame(); + EXPECT_TRUE(frame->CanSetPayloadType()); + frame->SetPayloadType(45); + EXPECT_EQ(frame->GetPayloadType(), 45); +} + +TEST(ChannelSendFrameTransformerDelegateTest, SetAudioLevel) { + std::unique_ptr frame = CreateFrame(); + EXPECT_TRUE(frame->CanSetAudioLevel()); + frame->SetAudioLevel(45u); + EXPECT_EQ(frame->AudioLevel(), 45u); + frame->SetAudioLevel(std::nullopt); + EXPECT_FALSE(frame->AudioLevel().has_value()); +} + +TEST(ChannelSendFrameTransformerDelegateTest, SetAudioLevelIsClamped) { + std::unique_ptr frame = CreateFrame(); + EXPECT_TRUE(frame->CanSetAudioLevel()); + frame->SetAudioLevel(128u); + EXPECT_EQ(frame->AudioLevel(), 127u); +} + } // namespace } // namespace webrtc From fce732b705bcf6c53af1dc027232141cb5d8d1dd Mon Sep 17 00:00:00 2001 From: Mirko Bonadei Date: Tue, 20 May 2025 14:11:42 +0000 Subject: [PATCH 224/292] Remove tools_webrtc/cpu. This code has not been used for almost 10 years. Bug: webrtc:418969535 Change-Id: I3aef5d9484a6102d7fbeea24680fa4fa6ae4126b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392640 Reviewed-by: Tomas Gunnarsson Commit-Queue: Tomas Gunnarsson Auto-Submit: Mirko Bonadei Cr-Commit-Position: refs/heads/main@{#44696} --- tools_webrtc/cpu/OWNERS | 1 - tools_webrtc/cpu/README | 12 ------ tools_webrtc/cpu/cpu_mon.py | 83 ------------------------------------- 3 files changed, 96 deletions(-) delete mode 100644 tools_webrtc/cpu/OWNERS delete mode 100644 tools_webrtc/cpu/README delete mode 100644 tools_webrtc/cpu/cpu_mon.py diff --git a/tools_webrtc/cpu/OWNERS b/tools_webrtc/cpu/OWNERS deleted file mode 100644 index 0fba125734..0000000000 --- a/tools_webrtc/cpu/OWNERS +++ /dev/null @@ -1 +0,0 @@ -tommi@webrtc.org diff --git a/tools_webrtc/cpu/README b/tools_webrtc/cpu/README deleted file mode 100644 index a0cee30a82..0000000000 --- a/tools_webrtc/cpu/README +++ /dev/null @@ -1,12 +0,0 @@ -This directory contains a little utility for doing CPU measurements. -It requires a Python package, psutil, to be installed. -See: https://pypi.python.org/pypi/psutil - -On mac you can install this package like so: -sudo ARCHFLAGS='-Wno-error=unused-command-line-argument-hard-error-in-future' easy_install psutil - -On Windows: -- TBD (see link above) - -On Linux: -- TBD (see link above) diff --git a/tools_webrtc/cpu/cpu_mon.py b/tools_webrtc/cpu/cpu_mon.py deleted file mode 100644 index 9c25fbd088..0000000000 --- a/tools_webrtc/cpu/cpu_mon.py +++ /dev/null @@ -1,83 +0,0 @@ -#!/usr/bin/env vpython3 -# -# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. -# -# Use of this source code is governed by a BSD-style license -# that can be found in the LICENSE file in the root of the source -# tree. An additional intellectual property rights grant can be found -# in the file PATENTS. All contributing project authors may -# be found in the AUTHORS file in the root of the source tree. - -import sys - -import psutil -import numpy -from matplotlib import pyplot - - -class CpuSnapshot: - def __init__(self, label): - self.label = label - self.samples = [] - - def Capture(self, sample_count): - print(('Capturing %d CPU samples for %s...' % - ((sample_count - len(self.samples)), self.label))) - while len(self.samples) < sample_count: - self.samples.append(psutil.cpu_percent(1.0, False)) - - def Text(self): - return ( - '%s: avg=%s, median=%s, min=%s, max=%s' % - (self.label, numpy.average(self.samples), numpy.median( - self.samples), numpy.min(self.samples), numpy.max(self.samples))) - - def Max(self): - return numpy.max(self.samples) - - -def GrabCpuSamples(sample_count): - print('Label for snapshot (enter to quit): ') - label = eval(input().strip()) - if len(label) == 0: - return None - - snapshot = CpuSnapshot(label) - snapshot.Capture(sample_count) - - return snapshot - - -def main(): - print('How many seconds to capture per snapshot (enter for 60)?') - sample_count = eval(input().strip()) - if len(sample_count) > 0 and int(sample_count) > 0: - sample_count = int(sample_count) - else: - print('Defaulting to 60 samples.') - sample_count = 60 - - snapshots = [] - while True: - snapshot = GrabCpuSamples(sample_count) - if snapshot is None: - break - snapshots.append(snapshot) - - if len(snapshots) == 0: - print('no samples captured') - return -1 - - pyplot.title('CPU usage') - - for s in snapshots: - pyplot.plot(s.samples, label=s.Text(), linewidth=2) - - pyplot.legend() - - pyplot.show() - return 0 - - -if __name__ == '__main__': - sys.exit(main()) From dbe37ebaeb6668f71a1072606af434e009780213 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Tue, 20 May 2025 09:04:05 -0700 Subject: [PATCH 225/292] Roll chromium_revision cbbfcc5fee..6924e57e1f (1462671:1462834) Change log: https://chromium.googlesource.com/chromium/src/+log/cbbfcc5fee..6924e57e1f Full diff: https://chromium.googlesource.com/chromium/src/+/cbbfcc5fee..6924e57e1f Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/cf27d977ab..4078802f3b * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/d4dbffa976..8e8a804b47 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/d971856aa6..583ee76993 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/8dbab0fa74..316ff32605 * src/third_party/androidx/cipd: eaknFYxsT8k1aKnroa-MOOT-nxE41krXhpS9QxURsp0C..soQYlPZv5PZNVpLURHERMQJ9xpNLVvNZ1uzgNs2ZoEIC * src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/21a487e9fa..2a15f4c283 * src/third_party/fontconfig/src: https://chromium.googlesource.com/external/fontconfig.git/+log/14d466b30a..8cf0ce700a * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/17ab5def84..fd420b420d * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/a9b5b095da..5a75e4faa9 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/043e67c9de..f253b1ac5d DEPS diff: https://chromium.googlesource.com/chromium/src/+/cbbfcc5fee..6924e57e1f/DEPS BUG=None Change-Id: Ibb9c1f8c90b7821b32255fbfc064eb2b7818092c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392700 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44697} --- DEPS | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/DEPS b/DEPS index e41ca9a5c9..f7b7b9a296 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': 'cbbfcc5feef0830b8fe17ad5ce545a647e8557a1', + 'chromium_revision': '6924e57e1fd64a1b3e2d24a61ea19f377663e2c3', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,7 +73,7 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@cf27d977ab48cbc6eaf94ff7499cdf5e7978993f', + 'https://chromium.googlesource.com/chromium/src/base@4078802f3b1e87162d796280163fae164d0e2d16', 'src/build': 'https://chromium.googlesource.com/chromium/src/build@2263856e4c6de03a8d6cfddebb4e5ef573af2d2e', 'src/buildtools': @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@d4dbffa976fbad72c3e3253982f4757e693652af', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@8e8a804b47b15254bad16a8789beda5a21d277f0', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@d971856aa6d7475b3a39e76d96d7d3afc8abf133', + 'https://chromium.googlesource.com/chromium/src/testing@583ee76993914517f7d7dc700b55a5ea771ff72c', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@8dbab0fa7419236ec3c45bad9677ce5ca734b907', + 'https://chromium.googlesource.com/chromium/src/third_party@316ff32605d5eb94f2c0bd4b06f375ab0c765edc', 'src/buildtools/linux64': { 'packages': [ @@ -323,7 +323,7 @@ deps = { 'src/third_party/clang-format/script': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@37f6e68a107df43b7d7e044fd36a13cbae3413f2', 'src/third_party/compiler-rt/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@21a487e9fae57351be896c5c34dce665e0990c76', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@2a15f4c28336302d95a664d99c18d3d0a011bd09', 'src/third_party/libc++/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@a01c02c9d4acbdae3b7e8a2f3ee58579a9c29f96', 'src/third_party/libc++abi/src': @@ -503,11 +503,11 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@17ab5def84090ca0d2da4c1cdd1286283be1404e', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@fd420b420d04ec3b274af33a7089ea44fa5de9e1', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { - 'url': 'https://chromium.googlesource.com/external/fontconfig.git@14d466b30a8ab4a9d789977ed94f2c30e7209267', + 'url': 'https://chromium.googlesource.com/external/fontconfig.git@8cf0ce700a8abe0d97ace4bf7efc7f9534b729ba', 'condition': 'checkout_linux', }, 'src/third_party/freetype/src': @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'a9b5b095dac6ee69731e25683949fb181a57fa24', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '5a75e4faa948b13d3bb6567ea2dd6347cabac610', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@043e67c9de12e6bceffae48a3e3cdb56245bd726', + 'https://chromium.googlesource.com/chromium/src/tools@f253b1ac5dfc2a163e271132a45064eacd4d4179', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'eaknFYxsT8k1aKnroa-MOOT-nxE41krXhpS9QxURsp0C', + 'version': 'soQYlPZv5PZNVpLURHERMQJ9xpNLVvNZ1uzgNs2ZoEIC', }, ], 'condition': 'checkout_android and non_git_source', From ccdc929a5f54b19cc4bd6e63a7fd9d8fa2a1a3de Mon Sep 17 00:00:00 2001 From: Tommi Date: Tue, 20 May 2025 17:31:30 +0200 Subject: [PATCH 226/292] Remove the ACMResampler class, use PushResampler directly instead. MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Bug: chromium:335805780 Change-Id: Ieaf127046fe79a53c2bbde71d6e08f3ce07cbf09 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391280 Commit-Queue: Tomas Gunnarsson Reviewed-by: Jakob Ivarsson‎ Cr-Commit-Position: refs/heads/main@{#44698} --- modules/audio_coding/BUILD.gn | 1 + modules/audio_coding/acm2/acm_resampler.cc | 89 +++++-------------- modules/audio_coding/acm2/acm_resampler.h | 25 +----- .../audio_coding/acm2/audio_coding_module.cc | 11 ++- modules/audio_coding/test/opus_test.cc | 5 +- modules/audio_coding/test/opus_test.h | 3 +- 6 files changed, 35 insertions(+), 99 deletions(-) diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn index d27943394a..dd4500f744 100644 --- a/modules/audio_coding/BUILD.gn +++ b/modules/audio_coding/BUILD.gn @@ -760,6 +760,7 @@ rtc_library("audio_coding_modules_tests_shared") { "../../api/neteq:default_neteq_factory", "../../api/neteq:neteq_api", "../../api/units:timestamp", + "../../common_audio", "../../rtc_base:checks", "../../rtc_base:digest", "../../rtc_base:stringutils", diff --git a/modules/audio_coding/acm2/acm_resampler.cc b/modules/audio_coding/acm2/acm_resampler.cc index 1587eccfbb..9daa04d14b 100644 --- a/modules/audio_coding/acm2/acm_resampler.cc +++ b/modules/audio_coding/acm2/acm_resampler.cc @@ -10,51 +10,16 @@ #include "modules/audio_coding/acm2/acm_resampler.h" -#include - +#include #include #include "api/audio/audio_frame.h" #include "api/audio/audio_view.h" #include "rtc_base/checks.h" -#include "rtc_base/logging.h" -#include "rtc_base/numerics/safe_conversions.h" namespace webrtc { namespace acm2 { -ACMResampler::ACMResampler() {} - -ACMResampler::~ACMResampler() {} - -int ACMResampler::Resample10Msec(const int16_t* in_audio, - int in_freq_hz, - int out_freq_hz, - size_t num_audio_channels, - size_t out_capacity_samples, - int16_t* out_audio) { - InterleavedView src( - in_audio, SampleRateToDefaultChannelSize(in_freq_hz), num_audio_channels); - InterleavedView dst(out_audio, - SampleRateToDefaultChannelSize(out_freq_hz), - num_audio_channels); - RTC_DCHECK_GE(out_capacity_samples, dst.size()); - Resample10Msec(src, in_freq_hz, dst, out_freq_hz); - return dst.samples_per_channel(); -} - -void ACMResampler::Resample10Msec(InterleavedView src, - int in_freq_hz, - InterleavedView dst, - int out_freq_hz) { - RTC_DCHECK_EQ(src.num_channels(), dst.num_channels()); - if (in_freq_hz == out_freq_hz) { - CopySamples(dst, src); - } else { - resampler_.Resample(src, dst); - } -} - ResamplerHelper::ResamplerHelper() { ClearSamples(last_audio_buffer_); } @@ -63,44 +28,38 @@ bool ResamplerHelper::MaybeResample(int desired_sample_rate_hz, AudioFrame* audio_frame) { const int current_sample_rate_hz = audio_frame->sample_rate_hz_; RTC_DCHECK_NE(current_sample_rate_hz, 0); + RTC_DCHECK_GT(desired_sample_rate_hz, 0); // Update if resampling is required. + // TODO(tommi): `desired_sample_rate_hz` should never be -1. + // Remove the first check. const bool need_resampling = (desired_sample_rate_hz != -1) && (current_sample_rate_hz != desired_sample_rate_hz); if (need_resampling && !resampled_last_output_frame_) { // Prime the resampler with the last frame. - int16_t temp_output[AudioFrame::kMaxDataSizeSamples]; - int samples_per_channel_int = resampler_.Resample10Msec( - last_audio_buffer_.data(), current_sample_rate_hz, - desired_sample_rate_hz, audio_frame->num_channels_, - AudioFrame::kMaxDataSizeSamples, temp_output); - if (samples_per_channel_int < 0) { - RTC_LOG(LS_ERROR) << "AcmReceiver::GetAudio - " - "Resampling last_audio_buffer_ failed."; - return false; - } + InterleavedView src(last_audio_buffer_.data(), + audio_frame->samples_per_channel(), + audio_frame->num_channels()); + std::array temp_output; + InterleavedView dst( + temp_output.data(), + SampleRateToDefaultChannelSize(desired_sample_rate_hz), + audio_frame->num_channels_); + resampler_.Resample(src, dst); } // TODO(bugs.webrtc.org/3923) Glitches in the output may appear if the output // rate from NetEq changes. if (need_resampling) { - // TODO(yujo): handle this more efficiently for muted frames. - int samples_per_channel_int = resampler_.Resample10Msec( - audio_frame->data(), current_sample_rate_hz, desired_sample_rate_hz, - audio_frame->num_channels_, AudioFrame::kMaxDataSizeSamples, - audio_frame->mutable_data()); - if (samples_per_channel_int < 0) { - RTC_LOG(LS_ERROR) - << "AcmReceiver::GetAudio - Resampling audio_buffer_ failed."; - return false; - } - audio_frame->samples_per_channel_ = - static_cast(samples_per_channel_int); - audio_frame->sample_rate_hz_ = desired_sample_rate_hz; - RTC_DCHECK_EQ(audio_frame->sample_rate_hz_, - dchecked_cast(audio_frame->samples_per_channel_ * 100)); + // Grab the source view of the current layout before changing properties. + InterleavedView src = audio_frame->data_view(); + audio_frame->SetSampleRateAndChannelSize(desired_sample_rate_hz); + InterleavedView dst = audio_frame->mutable_data( + audio_frame->samples_per_channel(), audio_frame->num_channels()); + // TODO(tommi): Don't resample muted audio frames. + resampler_.Resample(src, dst); resampled_last_output_frame_ = true; } else { resampled_last_output_frame_ = false; @@ -108,10 +67,10 @@ bool ResamplerHelper::MaybeResample(int desired_sample_rate_hz, } // Store current audio in `last_audio_buffer_` for next time. - // TODO: b/335805780 - Use CopySamples(). - memcpy(last_audio_buffer_.data(), audio_frame->data(), - sizeof(int16_t) * audio_frame->samples_per_channel_ * - audio_frame->num_channels_); + InterleavedView dst(last_audio_buffer_.data(), + audio_frame->samples_per_channel(), + audio_frame->num_channels()); + CopySamples(dst, audio_frame->data_view()); return true; } diff --git a/modules/audio_coding/acm2/acm_resampler.h b/modules/audio_coding/acm2/acm_resampler.h index 256884bfbf..8b6cee04ce 100644 --- a/modules/audio_coding/acm2/acm_resampler.h +++ b/modules/audio_coding/acm2/acm_resampler.h @@ -17,34 +17,11 @@ #include #include "api/audio/audio_frame.h" -#include "api/audio/audio_view.h" #include "common_audio/resampler/include/push_resampler.h" namespace webrtc { namespace acm2 { -class ACMResampler { - public: - ACMResampler(); - ~ACMResampler(); - - // TODO: b/335805780 - Remove and use only the InterleavedView<> version - int Resample10Msec(const int16_t* in_audio, - int in_freq_hz, - int out_freq_hz, - size_t num_audio_channels, - size_t out_capacity_samples, - int16_t* out_audio); - - void Resample10Msec(InterleavedView src, - int src_freq_hz, - InterleavedView dst, - int dst_freq_hz); - - private: - PushResampler resampler_; -}; - // Helper class to perform resampling if needed, meant to be used after // receiving the audio_frame from NetEq. Provides reasonably glitch free // transitions between different output sample rates from NetEq. @@ -56,7 +33,7 @@ class ResamplerHelper { bool MaybeResample(int desired_sample_rate_hz, AudioFrame* audio_frame); private: - ACMResampler resampler_; + PushResampler resampler_; bool resampled_last_output_frame_ = true; std::array last_audio_buffer_; }; diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc index 32a36a7e6b..41d022ecf3 100644 --- a/modules/audio_coding/acm2/audio_coding_module.cc +++ b/modules/audio_coding/acm2/audio_coding_module.cc @@ -23,8 +23,8 @@ #include "api/audio/audio_view.h" #include "api/audio_codecs/audio_encoder.h" #include "api/function_view.h" +#include "common_audio/resampler/include/push_resampler.h" #include "modules/audio_coding/acm2/acm_remixing.h" -#include "modules/audio_coding/acm2/acm_resampler.h" #include "modules/audio_coding/include/audio_coding_module_typedefs.h" #include "modules/include/module_common_types_public.h" #include "rtc_base/buffer.h" @@ -161,7 +161,7 @@ class AudioCodingModuleImpl final : public AudioCodingModule { Buffer encode_buffer_ RTC_GUARDED_BY(acm_mutex_); uint32_t expected_codec_ts_ RTC_GUARDED_BY(acm_mutex_); uint32_t expected_in_ts_ RTC_GUARDED_BY(acm_mutex_); - acm2::ACMResampler resampler_ RTC_GUARDED_BY(acm_mutex_); + PushResampler resampler_ RTC_GUARDED_BY(acm_mutex_); ChangeLogger bitrate_logger_ RTC_GUARDED_BY(acm_mutex_); // Current encoder stack, provided by a call to RegisterEncoder. @@ -506,11 +506,10 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame, preprocess_frame_.SetSampleRateAndChannelSize(encoder_stack_->SampleRateHz()); if (resample) { - resampler_.Resample10Msec( - resample_src_audio, in_frame.sample_rate_hz(), + resampler_.Resample( + resample_src_audio, preprocess_frame_.mutable_data(preprocess_frame_.samples_per_channel(), - preprocess_frame_.num_channels()), - preprocess_frame_.sample_rate_hz()); + preprocess_frame_.num_channels())); } expected_codec_ts_ += diff --git a/modules/audio_coding/test/opus_test.cc b/modules/audio_coding/test/opus_test.cc index da44f26e22..6b511c25f5 100644 --- a/modules/audio_coding/test/opus_test.cc +++ b/modules/audio_coding/test/opus_test.cc @@ -267,9 +267,8 @@ void OpusTest::Run(TestPackStereo* channel, // If input audio is sampled at 32 kHz, resampling to 48 kHz is required. InterleavedView dst(&audio[written_samples], 480, channels); - resampler_.Resample10Msec(audio_frame.data_view(), - audio_frame.sample_rate_hz_, dst, 48000); - written_samples += 480 * channels; + resampler_.Resample(audio_frame.data_view(), dst); + written_samples += dst.size(); // Sometimes we need to loop over the audio vector to produce the right // number of packets. diff --git a/modules/audio_coding/test/opus_test.h b/modules/audio_coding/test/opus_test.h index d6f0e7c31a..4d2586ee4b 100644 --- a/modules/audio_coding/test/opus_test.h +++ b/modules/audio_coding/test/opus_test.h @@ -18,6 +18,7 @@ #include #include "api/neteq/neteq.h" +#include "common_audio/resampler/include/push_resampler.h" #include "modules/audio_coding/acm2/acm_resampler.h" #include "modules/audio_coding/codecs/opus/opus_inst.h" #include "modules/audio_coding/test/PCMFile.h" @@ -51,7 +52,7 @@ class OpusTest { int counter_; uint8_t payload_type_; uint32_t rtp_timestamp_; - acm2::ACMResampler resampler_; + PushResampler resampler_; WebRtcOpusEncInst* opus_mono_encoder_; WebRtcOpusEncInst* opus_stereo_encoder_; WebRtcOpusDecInst* opus_mono_decoder_; From 3d059dd3ca2653d372ea4463dadc53a40baf7fa8 Mon Sep 17 00:00:00 2001 From: Mirko Bonadei Date: Tue, 20 May 2025 15:05:41 +0000 Subject: [PATCH 227/292] Remove tools_webrtc/clang_tidy.py This code is unused. Bug: webrtc:418969535 Change-Id: Ia5d4c0c68bdd2f59a5fd49578a5327b597f36ec0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392661 Reviewed-by: Jeremy Leconte Auto-Submit: Mirko Bonadei Cr-Commit-Position: refs/heads/main@{#44699} --- tools_webrtc/clang_tidy.py | 97 -------------------------------------- 1 file changed, 97 deletions(-) delete mode 100755 tools_webrtc/clang_tidy.py diff --git a/tools_webrtc/clang_tidy.py b/tools_webrtc/clang_tidy.py deleted file mode 100755 index b3f738eab3..0000000000 --- a/tools_webrtc/clang_tidy.py +++ /dev/null @@ -1,97 +0,0 @@ -#!/usr/bin/env vpython3 - -# Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. -# -# Use of this source code is governed by a BSD-style license -# that can be found in the LICENSE file in the root of the source -# tree. An additional intellectual property rights grant can be found -# in the file PATENTS. All contributing project authors may -# be found in the AUTHORS file in the root of the source tree. -"""Invoke clang-tidy tool. - -Usage: clang_tidy.py file.cc [clang-tidy-args...] - -Just a proof of concept! -We use an embedded clang-tidy whose version doesn't match clang++. -""" - -import argparse -import os -import shutil -import subprocess -import sys -import tempfile -from presubmit_checks_lib.build_helpers import (GetClangTidyPath, - GetCompilationCommand) - -# We enable all checkers by default for investigation purpose. -# This includes clang-analyzer-* checks. -# Individual checkers can be disabled via command line options. -# TODO(bugs.webrtc.com/10258): Select checkers relevant to webrtc guidelines. -CHECKER_OPTION = '-checks=*' - - -def Process(filepath, args): - # Build directory is needed to gather compilation flags. - # Create a temporary one (instead of reusing an existing one) - # to keep the CLI simple and unencumbered. - out_dir = tempfile.mkdtemp('clang_tidy') - - try: - gn_args = [] # Use default build. - command = GetCompilationCommand(filepath, gn_args, out_dir) - - # Remove warning flags. They aren't needed and they cause trouble - # when clang-tidy doesn't match most recent clang. - # Same battle for -f (e.g. -fcomplete-member-pointers). - command = [ - arg for arg in command - if not (arg.startswith('-W') or arg.startswith('-f')) - ] - - # Path from build dir. - rel_path = os.path.relpath(os.path.abspath(filepath), out_dir) - - # Replace clang++ by clang-tidy - command[0:1] = [GetClangTidyPath(), CHECKER_OPTION, rel_path - ] + args + ['--'] # Separator for clang flags. - print("Running: %s" % ' '.join(command)) - # Run from build dir so that relative paths are correct. - p = subprocess.Popen(command, - cwd=out_dir, - stdout=sys.stdout, - stderr=sys.stderr) - p.communicate() - return p.returncode - finally: - shutil.rmtree(out_dir, ignore_errors=True) - - -def ValidateCC(filepath): - """We can only analyze .cc files. Provide explicit message about that.""" - if filepath.endswith('.cc'): - return filepath - msg = ('%s not supported.\n' - 'For now, we can only analyze translation units (.cc files).' % - filepath) - raise argparse.ArgumentTypeError(msg) - - -def Main(): - description = ( - "Run clang-tidy on single cc file.\n" - "Use flags, defines and include paths as in default debug build.\n" - "WARNING, this is a POC version with rough edges.") - parser = argparse.ArgumentParser(description=description) - parser.add_argument('filepath', - help='Specifies the path of the .cc file to analyze.', - type=ValidateCC) - parser.add_argument('args', - nargs=argparse.REMAINDER, - help='Arguments passed to clang-tidy') - parsed_args = parser.parse_args() - return Process(parsed_args.filepath, parsed_args.args) - - -if __name__ == '__main__': - sys.exit(Main()) From 58f9e566f93b0d42b8ff6821f2471163745e5ea3 Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Tue, 20 May 2025 15:17:47 +0000 Subject: [PATCH 228/292] Extend FieldTrials api Intent of new api is to replace field_trial::FieldTrialsStringIsValid with FieldTrials::Create field_trial::MergeFieldTrialsStrings with FieldTrials::Merge and FieldTrials::Set Thus allow to assemble field trials without relying on functions designed for creating global field trials string. Bug: webrtc:42220378 Change-Id: Icc10b202f4d9c8d540fe579bd213d3c83c415263 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392461 Reviewed-by: Tomas Gunnarsson Commit-Queue: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#44700} --- api/BUILD.gn | 2 + api/field_trials.cc | 82 ++++++++++++++++++++------- api/field_trials.h | 51 ++++++++++++++++- api/field_trials_unittest.cc | 106 ++++++++++++++++++++++++++++++++++- 4 files changed, 219 insertions(+), 22 deletions(-) diff --git a/api/BUILD.gn b/api/BUILD.gn index d0e63949c6..d7d103de5b 100644 --- a/api/BUILD.gn +++ b/api/BUILD.gn @@ -1712,6 +1712,8 @@ rtc_library("field_trials") { ":field_trials_registry", "../rtc_base:checks", "../rtc_base/containers:flat_map", + "//third_party/abseil-cpp/absl/base:nullability", + "//third_party/abseil-cpp/absl/memory", "//third_party/abseil-cpp/absl/strings:string_view", ] } diff --git a/api/field_trials.cc b/api/field_trials.cc index c39d04cf17..fc896e1258 100644 --- a/api/field_trials.cc +++ b/api/field_trials.cc @@ -10,36 +10,78 @@ #include "api/field_trials.h" +#include #include #include +#include "absl/base/nullability.h" +#include "absl/memory/memory.h" #include "absl/strings/string_view.h" #include "rtc_base/checks.h" #include "rtc_base/containers/flat_map.h" namespace webrtc { +namespace { -FieldTrials::FieldTrials(absl::string_view s) { +absl::string_view NextKeyOrValue(absl::string_view& s) { + absl::string_view::size_type separator_pos = s.find('/'); + if (separator_pos == absl::string_view::npos) { + // Missing separator '/' after field trial key or value. + return ""; + } + absl::string_view result = s.substr(0, separator_pos); + s.remove_prefix(separator_pos + 1); + return result; +} + +bool Parse(absl::string_view s, + flat_map& key_value_map) { while (!s.empty()) { - absl::string_view::size_type separator_pos = s.find('/'); - RTC_CHECK_NE(separator_pos, absl::string_view::npos) - << "Missing separator '/' after field trial key."; - RTC_CHECK_GT(separator_pos, 0) << "Field trial key cannot be empty."; - absl::string_view key = s.substr(0, separator_pos); - s.remove_prefix(separator_pos + 1); - - RTC_CHECK(!s.empty()) - << "Missing value after field trial key. String ended."; - separator_pos = s.find('/'); - RTC_CHECK_NE(separator_pos, absl::string_view::npos) - << "Missing terminating '/' in field trial string."; - RTC_CHECK_GT(separator_pos, 0) << "Field trial value cannot be empty."; - absl::string_view value = s.substr(0, separator_pos); - s.remove_prefix(separator_pos + 1); - - // If a key is specified multiple times, only the value linked to the first - // key is stored. - key_value_map_.emplace(key, value); + absl::string_view key = NextKeyOrValue(s); + absl::string_view value = NextKeyOrValue(s); + if (key.empty() || value.empty()) { + return false; + } + + auto it = key_value_map.emplace(key, value).first; + if (it->second != value) { + // Duplicate trials with different values is not fine. + return false; + } + } + return true; +} + +} // namespace + +absl_nullable std::unique_ptr FieldTrials::Create( + absl::string_view s) { + flat_map key_value_map; + if (!Parse(s, key_value_map)) { + return nullptr; + } + // Using `new` to access a private constructor. + return absl::WrapUnique(new FieldTrials(std::move(key_value_map))); +} + +FieldTrials::FieldTrials(absl::string_view s) { + RTC_CHECK(Parse(s, key_value_map_)); +} + +void FieldTrials::Merge(const FieldTrials& other) { + for (const auto& [trial, group] : other.key_value_map_) { + key_value_map_.insert_or_assign(trial, group); + } +} + +void FieldTrials::Set(absl::string_view trial, absl::string_view group) { + RTC_CHECK(!trial.empty()); + RTC_CHECK_EQ(trial.find('/'), absl::string_view::npos); + RTC_CHECK_EQ(group.find('/'), absl::string_view::npos); + if (group.empty()) { + key_value_map_.erase(trial); + } else { + key_value_map_.insert_or_assign(trial, group); } } diff --git a/api/field_trials.h b/api/field_trials.h index ff42c32078..4937b506be 100644 --- a/api/field_trials.h +++ b/api/field_trials.h @@ -13,7 +13,9 @@ #include #include +#include +#include "absl/base/nullability.h" #include "absl/strings/string_view.h" #include "api/field_trials_registry.h" #include "rtc_base/containers/flat_map.h" @@ -32,10 +34,41 @@ namespace webrtc { // The field trials are injected into objects that use them at creation time. class FieldTrials : public FieldTrialsRegistry { public: + // Creates field trials from a valid field trial string. + // Returns nullptr if the string is invalid. + // E.g., valid string: + // "WebRTC-ExperimentFoo/Enabled/WebRTC-ExperimentBar/Enabled100kbps/" + // Assigns to group "Enabled" on WebRTC-ExperimentFoo trial + // and to group "Enabled100kbps" on WebRTC-ExperimentBar. + // + // E.g., invalid string: + // "WebRTC-experiment1/Enabled" (note missing / separator at the end). + static absl_nullable std::unique_ptr Create(absl::string_view s); + + // Creates field trials from a string. + // It is an error to call the constructor with an invalid field trial string. explicit FieldTrials(absl::string_view s); + + FieldTrials(const FieldTrials&) = default; FieldTrials(FieldTrials&&) = default; + FieldTrials& operator=(const FieldTrials&) = default; FieldTrials& operator=(FieldTrials&&) = default; - ~FieldTrials() = default; + + ~FieldTrials() override = default; + + template + friend void AbslStringify(Sink& sink, const FieldTrials& self); + + // Merges field trials from the `other` into this. + // + // If a key (trial) exists twice with conflicting values (groups), the value + // in `other` takes precedence. + void Merge(const FieldTrials& other); + + // Sets value (`group`) for an indvidual `trial`. + // It is an error to call this function with an invalid `trial` or `group`. + // Setting empty `group` is valid and removes the `trial`. + void Set(absl::string_view trial, absl::string_view group); // TODO: bugs.webrtc.org/42220378 - Deprecate and inline once no longer used // within webrtc. @@ -44,11 +77,27 @@ class FieldTrials : public FieldTrialsRegistry { } private: + explicit FieldTrials(flat_map key_value_map) + : key_value_map_(std::move(key_value_map)) {} + std::string GetValue(absl::string_view key) const override; flat_map key_value_map_; }; +template +void AbslStringify(Sink& sink, const FieldTrials& self) { + for (const auto& [trial, group] : self.key_value_map_) { + sink.Append(trial); + sink.Append("/"); + sink.Append(group); + // Intentionally output a string that is not a valid field trial string. + // Stringification is intended only for human readable logs, and is not + // intended for reusing as `FieldTrials` construction parameter. + sink.Append("//"); + } +} + } // namespace webrtc #endif // API_FIELD_TRIALS_H_ diff --git a/api/field_trials_unittest.cc b/api/field_trials_unittest.cc index 9d492529d1..d3521b49ca 100644 --- a/api/field_trials_unittest.cc +++ b/api/field_trials_unittest.cc @@ -10,10 +10,11 @@ #include "api/field_trials.h" - +#include "absl/strings/str_cat.h" #include "rtc_base/containers/flat_set.h" #include "system_wrappers/include/field_trial.h" #include "test/field_trial.h" +#include "test/gmock.h" #include "test/gtest.h" namespace webrtc { @@ -21,6 +22,11 @@ namespace { using field_trial::FieldTrialsAllowedInScopeForTesting; using test::ScopedFieldTrials; +using ::testing::AllOf; +using ::testing::HasSubstr; +using ::testing::IsNull; +using ::testing::Not; +using ::testing::NotNull; TEST(FieldTrialsTest, EmptyStringHasNoEffect) { FieldTrialsAllowedInScopeForTesting k({"MyCoolTrial"}); @@ -89,5 +95,103 @@ TEST(FieldTrialsTest, GlobalAndNonGlobalFieldTrialsAreDisjoint) { EXPECT_TRUE(f.IsEnabled("SomeOtherString")); } +TEST(FieldTrialsTest, CreateAcceptsValidInputs) { + EXPECT_THAT(FieldTrials::Create(""), NotNull()); + EXPECT_THAT(FieldTrials::Create("Audio/Enabled/"), NotNull()); + EXPECT_THAT(FieldTrials::Create("Audio/Enabled/Video/Disabled/"), NotNull()); + + // Duplicate trials with the same value is fine + EXPECT_THAT(FieldTrials::Create("Audio/Enabled/Audio/Enabled/"), NotNull()); + EXPECT_THAT(FieldTrials::Create("Audio/Enabled/B/C/Audio/Enabled/"), + NotNull()); +} + +TEST(FieldTrialsTest, CreateRejectsBadInputs) { + // Bad delimiters + EXPECT_THAT(FieldTrials::Create("Audio/EnabledVideo/Disabled/"), IsNull()); + EXPECT_THAT(FieldTrials::Create("Audio/Enabled//Video/Disabled/"), IsNull()); + EXPECT_THAT(FieldTrials::Create("/Audio/Enabled/Video/Disabled/"), IsNull()); + EXPECT_THAT(FieldTrials::Create("Audio/Enabled/Video/Disabled"), IsNull()); + EXPECT_THAT(FieldTrials::Create("Audio/Enabled/Video/Disabled/garbage"), + IsNull()); + + // Empty trial or group + EXPECT_THAT(FieldTrials::Create("Audio//"), IsNull()); + EXPECT_THAT(FieldTrials::Create("/Enabled/"), IsNull()); + EXPECT_THAT(FieldTrials::Create("//"), IsNull()); + EXPECT_THAT(FieldTrials::Create("//Enabled"), IsNull()); + + // Duplicate trials with different values is not fine + EXPECT_THAT(FieldTrials::Create("Audio/Enabled/Audio/Disabled/"), IsNull()); + EXPECT_THAT(FieldTrials::Create("Audio/Enabled/B/C/Audio/Disabled/"), + IsNull()); + EXPECT_THAT(FieldTrials::Create("Audio/Enabled/Audio/Disabled/"), IsNull()); + EXPECT_THAT(FieldTrials::Create("Audio/Enabled/B/C/Audio/Disabled/"), + IsNull()); +} + +TEST(FieldTrialsTest, StringfiyMentionsKeysAndValues) { + // Exact format of the stringification is undefined. + EXPECT_THAT(absl::StrCat(FieldTrials("Audio/Enabled/Video/Value/")), + AllOf(HasSubstr("Audio"), HasSubstr("Enabled"), + HasSubstr("Video"), HasSubstr("Value"))); +} + +TEST(FieldTrialsTest, MergeCombinesFieldTrials) { + FieldTrials f("Video/Value1/"); + FieldTrials other("Audio/Value2/"); + + f.Merge(other); + + f.RegisterKeysForTesting({"Audio", "Video"}); + EXPECT_EQ(f.Lookup("Video"), "Value1"); + EXPECT_EQ(f.Lookup("Audio"), "Value2"); +} + +TEST(FieldTrialsTest, MergeGivesPrecedenceToOther) { + FieldTrials f("Audio/Disabled/Video/Enabled/"); + FieldTrials other("Audio/Enabled/"); + + f.Merge(other); + + f.RegisterKeysForTesting({"Audio"}); + EXPECT_EQ(f.Lookup("Audio"), "Enabled"); +} + +TEST(FieldTrialsTest, MergeDoesntChangeTrialAbsentInOther) { + FieldTrials f("Audio/Enabled/Video/Enabled/"); + FieldTrials other("Audio/Enabled/"); + + f.Merge(other); + + f.RegisterKeysForTesting({"Video"}); + EXPECT_EQ(f.Lookup("Video"), "Enabled"); +} + +TEST(FieldTrialsTest, SetUpdatesTrial) { + FieldTrials f("Audio/Enabled/Video/Enabled/"); + + f.Set("Audio", "Disabled"); + + f.RegisterKeysForTesting({"Audio"}); + EXPECT_EQ(f.Lookup("Audio"), "Disabled"); +} + +TEST(FieldTrialsTest, SettingEmptyValueRemovesFieldTrial) { + FieldTrials f("Audio/Enabled/Video/Enabled/"); + + f.Set("Audio", ""); + + f.RegisterKeysForTesting({"Audio"}); + EXPECT_EQ(f.Lookup("Audio"), ""); + EXPECT_THAT(absl::StrCat(f), Not(HasSubstr("Audio"))); + + // Absent field trials shouldn't override previous value during merge. + FieldTrials f2("Audio/Disabled/"); + f2.Merge(f); + f2.RegisterKeysForTesting({"Audio"}); + EXPECT_EQ(f2.Lookup("Audio"), "Disabled"); +} + } // namespace } // namespace webrtc From 4930228a3eaca3411d54b315ed29e5a504ed61da Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Thu, 24 Apr 2025 09:42:27 -0700 Subject: [PATCH 229/292] pc: limit TURN username to 509 bytes which is the limit for the STUN username field defined in https://www.rfc-editor.org/rfc/rfc8489#section-14.3 This is already silently enforced at the lower layer. Spec issue: https://github.com/w3c/webrtc-pc/issues/3049 BUG=webrtc:413259647 Change-Id: I34ffcf26bdc7ca772c41565f5a9a6fce088e49c9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/388360 Reviewed-by: Guido Urdaneta Commit-Queue: Philipp Hancke Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#44701} --- p2p/base/p2p_constants.h | 2 ++ p2p/base/turn_port.h | 3 +-- pc/BUILD.gn | 1 + pc/ice_server_parsing.cc | 7 +++++++ pc/ice_server_parsing_unittest.cc | 30 ++++++++++++++++++++++++++++++ 5 files changed, 41 insertions(+), 2 deletions(-) diff --git a/p2p/base/p2p_constants.h b/p2p/base/p2p_constants.h index fae92bfb27..80aa37639b 100644 --- a/p2p/base/p2p_constants.h +++ b/p2p/base/p2p_constants.h @@ -124,6 +124,8 @@ enum IcePriorityValue : uint8_t { ICE_TYPE_PREFERENCE_HOST = 126 }; +const int kMaxTurnUsernameLength = 509; // RFC 8489 section 14.3 + } // namespace webrtc // Re-export symbols from the webrtc namespace for backwards compatibility. diff --git a/p2p/base/turn_port.h b/p2p/base/turn_port.h index 44a4563942..794e4a41ad 100644 --- a/p2p/base/turn_port.h +++ b/p2p/base/turn_port.h @@ -27,6 +27,7 @@ #include "api/task_queue/pending_task_safety_flag.h" #include "api/transport/stun.h" #include "p2p/base/connection.h" +#include "p2p/base/p2p_constants.h" #include "p2p/base/port.h" #include "p2p/base/port_allocator.h" #include "p2p/base/port_interface.h" @@ -46,8 +47,6 @@ namespace webrtc { class TurnCustomizer; -const int kMaxTurnUsernameLength = 509; // RFC 8489 section 14.3 - extern const int STUN_ATTR_TURN_LOGGING_ID; extern const char TURN_PORT_TYPE[]; class TurnAllocateRequest; diff --git a/pc/BUILD.gn b/pc/BUILD.gn index 6d5d0ca9da..7241255ce4 100644 --- a/pc/BUILD.gn +++ b/pc/BUILD.gn @@ -1483,6 +1483,7 @@ rtc_library("ice_server_parsing") { "../api:candidate", "../api:libjingle_peerconnection_api", "../api:rtc_error", + "../p2p:p2p_constants", "../p2p:port", "../p2p:port_allocator", "../p2p:port_interface", diff --git a/pc/ice_server_parsing.cc b/pc/ice_server_parsing.cc index 02612eb23a..c929a2f1cd 100644 --- a/pc/ice_server_parsing.cc +++ b/pc/ice_server_parsing.cc @@ -22,6 +22,7 @@ #include "api/candidate.h" #include "api/peer_connection_interface.h" #include "api/rtc_error.h" +#include "p2p/base/p2p_constants.h" #include "p2p/base/port.h" #include "p2p/base/port_allocator.h" #include "p2p/base/port_interface.h" @@ -276,6 +277,12 @@ RTCError ParseIceServerUrl(const PeerConnectionInterface::IceServer& server, "ICE server parsing failed: TURN server with empty " "username or password"); } + // RFC 8489 limits the size of the STUN username field to 509 characters. + if (server.username.size() > kMaxTurnUsernameLength) { + LOG_AND_RETURN_ERROR( + RTCErrorType::INVALID_PARAMETER, + "ICE server parsing failed: TURN server username is too long"); + } // If the hostname field is not empty, then the server address must be // the resolved IP for that host, the hostname is needed later for TLS // handshake (SNI and Certificate verification). diff --git a/pc/ice_server_parsing_unittest.cc b/pc/ice_server_parsing_unittest.cc index df459e6ddd..4bc4f266f1 100644 --- a/pc/ice_server_parsing_unittest.cc +++ b/pc/ice_server_parsing_unittest.cc @@ -14,6 +14,7 @@ #include #include "api/peer_connection_interface.h" +#include "api/rtc_error.h" #include "p2p/base/port.h" #include "p2p/base/port_allocator.h" #include "p2p/base/port_interface.h" @@ -239,4 +240,33 @@ TEST_F(IceServerParsingTest, ParseMultipleUrls) { EXPECT_EQ(1U, turn_servers_.size()); } +TEST_F(IceServerParsingTest, TooLongUsername) { + // 509 characters is the maximum length of the username field in RFC 8489. + std::string k509Characters = + "aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa" + "aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa" + "aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa" + "aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa" + "aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa" + "aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa" + "aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa" + "aaaaa"; + ASSERT_EQ(k509Characters.size(), 509u); + // 509 characters is supported. + EXPECT_TRUE(ParseUrl("turn:hostname", k509Characters, "password")); + + // 510 is not supported. + PeerConnectionInterface::IceServers servers; + PeerConnectionInterface::IceServer server; + server.urls.push_back("turn:hostname"); + server.username = k509Characters + "b"; + server.password = "password"; + servers.push_back(server); + stun_servers_.clear(); + turn_servers_.clear(); + auto error = ParseIceServersOrError(servers, &stun_servers_, &turn_servers_); + EXPECT_FALSE(error.ok()); + EXPECT_EQ(error.type(), RTCErrorType::INVALID_PARAMETER); +} + } // namespace webrtc From db02d9cc3330ede59c2de495ced898c198091817 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Tue, 20 May 2025 15:05:49 -0700 Subject: [PATCH 230/292] Roll chromium_revision 6924e57e1f..b019ea34ab (1462834:1463060) Change log: https://chromium.googlesource.com/chromium/src/+log/6924e57e1f..b019ea34ab Full diff: https://chromium.googlesource.com/chromium/src/+/6924e57e1f..b019ea34ab Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/4078802f3b..436fa45edc * src/build: https://chromium.googlesource.com/chromium/src/build/+log/2263856e4c..03eb9166dd * src/buildtools/linux64: git_revision:0c25d1bbde6ef17e1e6d3938011f56a00c02e457..git_revision:cad8f67e2dd0cea00bbe4566efe2fdf4c1821563 * src/buildtools/mac: git_revision:0c25d1bbde6ef17e1e6d3938011f56a00c02e457..git_revision:cad8f67e2dd0cea00bbe4566efe2fdf4c1821563 * src/buildtools/win: git_revision:0c25d1bbde6ef17e1e6d3938011f56a00c02e457..git_revision:cad8f67e2dd0cea00bbe4566efe2fdf4c1821563 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/8e8a804b47..c2f6abd4e2 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/583ee76993..74a3da3c0c * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/316ff32605..4c53f1652a * src/third_party/android_deps/autorolled/cipd: 5jhxywV7wAc6P8KUwq9kM4caswnIJ4Y0n8PEaEAIpNMC..gbOkrsK3-7xG1DV1l2XCjQfdKOR_9eY9G2JQ08TEQkwC * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/196d941610..0888983e47 * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/fd420b420d..a58a5ffb47 * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/5a75e4faa9..188242925b * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/f253b1ac5d..369ac5168a DEPS diff: https://chromium.googlesource.com/chromium/src/+/6924e57e1f..b019ea34ab/DEPS BUG=None Change-Id: Ia268e60d8708001ee3eaa7ef2f0081140295f5f4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392761 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44702} --- DEPS | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/DEPS b/DEPS index f7b7b9a296..baf17d85d2 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '6924e57e1fd64a1b3e2d24a61ea19f377663e2c3', + 'chromium_revision': 'b019ea34ab2070eac0fb94345d2e63bf12267f32', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@4078802f3b1e87162d796280163fae164d0e2d16', + 'https://chromium.googlesource.com/chromium/src/base@436fa45edc5ad3de19e2e46f6a86b0394b79411f', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@2263856e4c6de03a8d6cfddebb4e5ef573af2d2e', + 'https://chromium.googlesource.com/chromium/src/build@03eb9166dd46ed90f914635ab989893ec5a7f6e0', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@cd169314af80e77a75eb25e2c0df8798f35bc04a', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,19 +84,19 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@8e8a804b47b15254bad16a8789beda5a21d277f0', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@c2f6abd4e2fa164e03f01c157659c75a3481b606', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@583ee76993914517f7d7dc700b55a5ea771ff72c', + 'https://chromium.googlesource.com/chromium/src/testing@74a3da3c0c5523157bb7bb34a573f93ca473f007', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@316ff32605d5eb94f2c0bd4b06f375ab0c765edc', + 'https://chromium.googlesource.com/chromium/src/third_party@4c53f1652a97b52d478780faeea4273bacef1f3e', 'src/buildtools/linux64': { 'packages': [ { 'package': 'gn/gn/linux-${{arch}}', - 'version': 'git_revision:0c25d1bbde6ef17e1e6d3938011f56a00c02e457', + 'version': 'git_revision:cad8f67e2dd0cea00bbe4566efe2fdf4c1821563', } ], 'dep_type': 'cipd', @@ -106,7 +106,7 @@ deps = { 'packages': [ { 'package': 'gn/gn/mac-${{arch}}', - 'version': 'git_revision:0c25d1bbde6ef17e1e6d3938011f56a00c02e457', + 'version': 'git_revision:cad8f67e2dd0cea00bbe4566efe2fdf4c1821563', } ], 'dep_type': 'cipd', @@ -116,7 +116,7 @@ deps = { 'packages': [ { 'package': 'gn/gn/windows-amd64', - 'version': 'git_revision:0c25d1bbde6ef17e1e6d3938011f56a00c02e457', + 'version': 'git_revision:cad8f67e2dd0cea00bbe4566efe2fdf4c1821563', } ], 'dep_type': 'cipd', @@ -497,13 +497,13 @@ deps = { 'src/third_party/crc32c/src': 'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558', 'src/third_party/depot_tools': - 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@196d9416106a6ace0729b09acda46546c4a12ce1', + 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@0888983e473613d853c419245b97294e46497798', 'src/third_party/ffmpeg': 'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@dcdd0fa51b65a0b1688ff6b8f0cc81908f09ded2', 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@fd420b420d04ec3b274af33a7089ea44fa5de9e1', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@a58a5ffb4763e0f922461efbaa493752361b4435', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '5a75e4faa948b13d3bb6567ea2dd6347cabac610', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '188242925b83331305ff6f8f146b2a6667f21ac2', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@f253b1ac5dfc2a163e271132a45064eacd4d4179', + 'https://chromium.googlesource.com/chromium/src/tools@369ac5168aef86aaa88feb29e5aa603e181ff932', 'src/third_party/espresso': { 'packages': [ @@ -827,7 +827,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_deps/autorolled', - 'version': '5jhxywV7wAc6P8KUwq9kM4caswnIJ4Y0n8PEaEAIpNMC', + 'version': 'gbOkrsK3-7xG1DV1l2XCjQfdKOR_9eY9G2JQ08TEQkwC', }, ], 'condition': 'checkout_android and non_git_source', From 3b4bc39c58ce73db5a14663bac05c249234ad847 Mon Sep 17 00:00:00 2001 From: webrtc-version-updater Date: Tue, 20 May 2025 21:07:13 -0700 Subject: [PATCH 231/292] Update WebRTC code version (2025-05-21T04:06:33). Bug: None Change-Id: I2e94a5fe5526691808ac1d9901325bfaa874c730 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392765 Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Cr-Commit-Position: refs/heads/main@{#44703} --- call/version.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/call/version.cc b/call/version.cc index 8c9ae5dc4c..04ef9a8d93 100644 --- a/call/version.cc +++ b/call/version.cc @@ -13,7 +13,7 @@ namespace webrtc { // The timestamp is always in UTC. -const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-20T04:05:45"; +const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-21T04:06:33"; void LoadWebRTCVersionInRegister() { // Using volatile to instruct the compiler to not optimize `p` away even From b16527f9ac6634622a251a4c749fde6cd89b29b1 Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Tue, 20 May 2025 14:07:29 -0700 Subject: [PATCH 232/292] tweak video payload parse failure logging to show the RTP ssrc and timestamp of the packet. Bug: webrtc:417857735 Change-Id: I997742096d8eb7ab036f51d37556ec2887d67ec6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392240 Reviewed-by: Philip Eliasson Reviewed-by: Ilya Nikolaevskiy Commit-Queue: Philipp Hancke Cr-Commit-Position: refs/heads/main@{#44704} --- video/rtp_video_stream_receiver2.cc | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc index 3e41758685..2fccb605cb 100644 --- a/video/rtp_video_stream_receiver2.cc +++ b/video/rtp_video_stream_receiver2.cc @@ -634,9 +634,9 @@ bool RtpVideoStreamReceiver2::OnReceivedPayloadData( Timestamp now = env_.clock().CurrentTime(); if (now - last_logged_failed_to_parse_dd_ > TimeDelta::Seconds(1)) { last_logged_failed_to_parse_dd_ = now; - RTC_LOG(LS_WARNING) << "ssrc: " << rtp_packet.Ssrc() - << ", timestamp: " << rtp_packet.Timestamp() - << " Failed to parse dependency descriptor."; + RTC_LOG(LS_WARNING) << "Failed to parse dependency descriptor for " + << "ssrc: " << rtp_packet.Ssrc() + << ", timestamp: " << rtp_packet.Timestamp(); } if (video_structure_ == nullptr && next_keyframe_request_for_missing_video_structure_ < now) { @@ -1231,7 +1231,9 @@ void RtpVideoStreamReceiver2::ReceivePacket(const RtpPacketReceived& packet) { std::optional parsed_payload = type_it->second->Parse(packet.PayloadBuffer()); if (parsed_payload == std::nullopt) { - RTC_LOG(LS_WARNING) << "Failed parsing payload."; + RTC_LOG(LS_WARNING) << " Failed to parse payload for " + << "ssrc: " << packet.Ssrc() + << ", timestamp: " << packet.Timestamp(); return false; } From b2f9a5858ab1b297cc7edaba151f949f865f7379 Mon Sep 17 00:00:00 2001 From: Jonas Oreland Date: Wed, 21 May 2025 07:21:21 +0200 Subject: [PATCH 233/292] improve error message for compile w/o WEBRTC_HAVE_SCTP To reduce time-to-fix-testcase for uninformed webrtc developer (https://webrtc-review.googlesource.com/c/src/+/392680). BUG=webrtc:0 Change-Id: Ibb20d1611918e2611a81d7bff4968d8e84d4d8cb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392900 Commit-Queue: Tomas Gunnarsson Auto-Submit: Jonas Oreland Reviewed-by: Tomas Gunnarsson Cr-Commit-Position: refs/heads/main@{#44705} --- pc/peer_connection.cc | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc index 829fb10ff9..630ab4db09 100644 --- a/pc/peer_connection.cc +++ b/pc/peer_connection.cc @@ -2506,9 +2506,16 @@ std::optional PeerConnection::SetupDataChannelTransport_n( DataChannelTransportInterface* transport = transport_controller_->GetDataChannelTransport(*sctp_mid_n_); if (!transport) { +#ifndef WEBRTC_HAVE_SCTP + RTC_LOG(LS_ERROR) << "Data channel transport is not available" + << " as WebRTC has been compiled without SCTP support " + "(WEBRTC_HAVE_SCTP), mid=" + << mid; +#else RTC_LOG(LS_ERROR) << "Data channel transport is not available for data channels, mid=" << mid; +#endif sctp_mid_n_ = std::nullopt; return std::nullopt; } From 0c83665d27a680ea98ed585a98a3c95fbe78d27b Mon Sep 17 00:00:00 2001 From: Mirko Bonadei Date: Tue, 20 May 2025 15:03:43 +0000 Subject: [PATCH 234/292] Remove tools_webrtc/network_emulator. This code has not been used for almost 10 years. Bug: webrtc:418969535 Change-Id: I79dcb020323a66df62780b53f5e476c29d53feb4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392660 Auto-Submit: Mirko Bonadei Reviewed-by: Christoffer Dewerin Commit-Queue: Christoffer Dewerin Cr-Commit-Position: refs/heads/main@{#44706} --- tools_webrtc/network_emulator/config.py | 36 --- tools_webrtc/network_emulator/emulate.py | 209 ------------------ .../network_emulator/network_emulator.py | 195 ---------------- 3 files changed, 440 deletions(-) delete mode 100644 tools_webrtc/network_emulator/config.py delete mode 100755 tools_webrtc/network_emulator/emulate.py delete mode 100644 tools_webrtc/network_emulator/network_emulator.py diff --git a/tools_webrtc/network_emulator/config.py b/tools_webrtc/network_emulator/config.py deleted file mode 100644 index 9a18bdce45..0000000000 --- a/tools_webrtc/network_emulator/config.py +++ /dev/null @@ -1,36 +0,0 @@ -#!/usr/bin/env vpython3 - -# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. -# -# Use of this source code is governed by a BSD-style license -# that can be found in the LICENSE file in the root of the source -# tree. An additional intellectual property rights grant can be found -# in the file PATENTS. All contributing project authors may -# be found in the AUTHORS file in the root of the source tree. -"""Configuration class for network emulation.""" - - -class ConnectionConfig: - """Configuration containing the characteristics of a network connection.""" - - def __init__(self, num, name, receive_bw_kbps, send_bw_kbps, delay_ms, - packet_loss_percent, queue_slots): - self.num = num - self.name = name - self.receive_bw_kbps = receive_bw_kbps - self.send_bw_kbps = send_bw_kbps - self.delay_ms = delay_ms - self.packet_loss_percent = packet_loss_percent - self.queue_slots = queue_slots - - def __str__(self): - """String representing the configuration. - - Returns: - A string formatted and padded like this example: - 12 Name 375 kbps 375 kbps 10 145 ms 0.1 % - """ - left_aligned_name = self.name.ljust(24, ' ') - return '%2s %24s %5s kbps %5s kbps %4s %5s ms %3s %%' % ( - self.num, left_aligned_name, self.receive_bw_kbps, self.send_bw_kbps, - self.queue_slots, self.delay_ms, self.packet_loss_percent) diff --git a/tools_webrtc/network_emulator/emulate.py b/tools_webrtc/network_emulator/emulate.py deleted file mode 100755 index a35ccd36c8..0000000000 --- a/tools_webrtc/network_emulator/emulate.py +++ /dev/null @@ -1,209 +0,0 @@ -#!/usr/bin/env vpython3 - -# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. -# -# Use of this source code is governed by a BSD-style license -# that can be found in the LICENSE file in the root of the source -# tree. An additional intellectual property rights grant can be found -# in the file PATENTS. All contributing project authors may -# be found in the AUTHORS file in the root of the source tree. -"""Script for constraining traffic on the local machine.""" - -import logging -import optparse -import socket -import sys - -import config -import network_emulator - -_DEFAULT_LOG_LEVEL = logging.INFO - -# Default port range to apply network constraints on. -_DEFAULT_PORT_RANGE = (32768, 65535) - -# The numbers below are gathered from Google stats from the presets of the Apple -# developer tool called Network Link Conditioner. -_PRESETS = [ - config.ConnectionConfig(1, 'Generic, Bad', 95, 95, 250, 2, 100), - config.ConnectionConfig(2, 'Generic, Average', 375, 375, 145, 0.1, 100), - config.ConnectionConfig(3, 'Generic, Good', 1000, 1000, 35, 0, 100), - config.ConnectionConfig(4, '3G, Average Case', 780, 330, 100, 0, 100), - config.ConnectionConfig(5, '3G, Good', 850, 420, 90, 0, 100), - config.ConnectionConfig(6, '3G, Lossy Network', 780, 330, 100, 1, 100), - config.ConnectionConfig(7, 'Cable Modem', 6000, 1000, 2, 0, 10), - config.ConnectionConfig(8, 'DSL', 2000, 256, 5, 0, 10), - config.ConnectionConfig(9, 'Edge, Average Case', 240, 200, 400, 0, 100), - config.ConnectionConfig(10, 'Edge, Good', 250, 200, 350, 0, 100), - config.ConnectionConfig(11, 'Edge, Lossy Network', 240, 200, 400, 1, 100), - config.ConnectionConfig(12, 'Wifi, Average Case', 40000, 33000, 1, 0, 100), - config.ConnectionConfig(13, 'Wifi, Good', 45000, 40000, 1, 0, 100), - config.ConnectionConfig(14, 'Wifi, Lossy', 40000, 33000, 1, 0, 100), -] -_PRESETS_DICT = dict((p.num, p) for p in _PRESETS) - -_DEFAULT_PRESET_ID = 2 -_DEFAULT_PRESET = _PRESETS_DICT[_DEFAULT_PRESET_ID] - - -class NonStrippingEpilogOptionParser(optparse.OptionParser): - """Custom parser to let us show the epilog without weird line breaking.""" - - def format_epilog(self, formatter): - return self.epilog - - -def _GetExternalIp(): - """Finds out the machine's external IP by connecting to google.com.""" - external_socket = socket.socket(socket.AF_INET, socket.SOCK_DGRAM) - external_socket.connect(('google.com', 80)) - return external_socket.getsockname()[0] - - -def _ParseArgs(): - """Define and parse the command-line arguments.""" - presets_string = '\n'.join(str(p) for p in _PRESETS) - parser = NonStrippingEpilogOptionParser(epilog=( - '\nAvailable presets:\n' - ' Bandwidth (kbps) Packet\n' - 'ID Name Receive Send Queue Delay loss \n' - '-- ---- --------- -------- ----- ------- ------\n' - '%s\n' % presets_string)) - parser.add_option('-p', - '--preset', - type='int', - default=_DEFAULT_PRESET_ID, - help=('ConnectionConfig configuration, specified by ID. ' - 'Default: %default')) - parser.add_option('-r', - '--receive-bw', - type='int', - default=_DEFAULT_PRESET.receive_bw_kbps, - help=('Receive bandwidth in kilobit/s. Default: %default')) - parser.add_option('-s', - '--send-bw', - type='int', - default=_DEFAULT_PRESET.send_bw_kbps, - help=('Send bandwidth in kilobit/s. Default: %default')) - parser.add_option('-d', - '--delay', - type='int', - default=_DEFAULT_PRESET.delay_ms, - help=('Delay in ms. Default: %default')) - parser.add_option('-l', - '--packet-loss', - type='float', - default=_DEFAULT_PRESET.packet_loss_percent, - help=('Packet loss in %. Default: %default')) - parser.add_option('-q', - '--queue', - type='int', - default=_DEFAULT_PRESET.queue_slots, - help=('Queue size as number of slots. Default: %default')) - parser.add_option('--port-range', - default='%s,%s' % _DEFAULT_PORT_RANGE, - help=('Range of ports for constrained network. Specify as ' - 'two comma separated integers. Default: %default')) - parser.add_option('--target-ip', - default=None, - help=('The interface IP address to apply the rules for. ' - 'Default: the external facing interface IP address.')) - parser.add_option('-v', - '--verbose', - action='store_true', - default=False, - help=('Turn on verbose output. Will print all \'ipfw\' ' - 'commands that are executed.')) - - options = parser.parse_args()[0] - - # Find preset by ID, if specified. - if options.preset and options.preset not in _PRESETS_DICT: - parser.error('Invalid preset: %s' % options.preset) - - # Simple validation of the IP address, if supplied. - if options.target_ip: - try: - socket.inet_aton(options.target_ip) - except socket.error: - parser.error('Invalid IP address specified: %s' % options.target_ip) - - # Convert port range into the desired tuple format. - try: - if isinstance(options.port_range, str): - options.port_range = tuple( - int(port) for port in options.port_range.split(',')) - if len(options.port_range) != 2: - parser.error('Invalid port range specified, please specify two ' - 'integers separated by a comma.') - except ValueError: - parser.error('Invalid port range specified.') - - _InitLogging(options.verbose) - return options - - -def _InitLogging(verbose): - """Setup logging.""" - log_level = _DEFAULT_LOG_LEVEL - if verbose: - log_level = logging.DEBUG - logging.basicConfig(level=log_level, format='%(message)s') - - -def main(): - options = _ParseArgs() - - # Build a configuration object. Override any preset configuration settings if - # a value of a setting was also given as a flag. - connection_config = _PRESETS_DICT[options.preset] - if options.receive_bw is not _DEFAULT_PRESET.receive_bw_kbps: - connection_config.receive_bw_kbps = options.receive_bw - if options.send_bw is not _DEFAULT_PRESET.send_bw_kbps: - connection_config.send_bw_kbps = options.send_bw - if options.delay is not _DEFAULT_PRESET.delay_ms: - connection_config.delay_ms = options.delay - if options.packet_loss is not _DEFAULT_PRESET.packet_loss_percent: - connection_config.packet_loss_percent = options.packet_loss - if options.queue is not _DEFAULT_PRESET.queue_slots: - connection_config.queue_slots = options.queue - emulator = network_emulator.NetworkEmulator(connection_config, - options.port_range) - try: - emulator.CheckPermissions() - except network_emulator.NetworkEmulatorError as e: - logging.error('Error: %s\n\nCause: %s', e.fail_msg, e.error) - return -1 - - if not options.target_ip: - external_ip = _GetExternalIp() - else: - external_ip = options.target_ip - - logging.info('Constraining traffic to/from IP: %s', external_ip) - try: - emulator.Emulate(external_ip) - logging.info( - 'Started network emulation with the following configuration:\n' - ' Receive bandwidth: %s kbps (%s kB/s)\n' - ' Send bandwidth : %s kbps (%s kB/s)\n' - ' Delay : %s ms\n' - ' Packet loss : %s %%\n' - ' Queue slots : %s', connection_config.receive_bw_kbps, - connection_config.receive_bw_kbps / 8, connection_config.send_bw_kbps, - connection_config.send_bw_kbps / 8, connection_config.delay_ms, - connection_config.packet_loss_percent, connection_config.queue_slots) - logging.info('Affected traffic: IP traffic on ports %s-%s', - options.port_range[0], options.port_range[1]) - input('Press Enter to abort Network Emulation...') - logging.info('Flushing all Dummynet rules...') - network_emulator.Cleanup() - logging.info('Completed Network Emulation.') - return 0 - except network_emulator.NetworkEmulatorError as e: - logging.error('Error: %s\n\nCause: %s', e.fail_msg, e.error) - return -2 - - -if __name__ == '__main__': - sys.exit(main()) diff --git a/tools_webrtc/network_emulator/network_emulator.py b/tools_webrtc/network_emulator/network_emulator.py deleted file mode 100644 index a7776a5f92..0000000000 --- a/tools_webrtc/network_emulator/network_emulator.py +++ /dev/null @@ -1,195 +0,0 @@ -#!/usr/bin/env vpython3 - -# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. -# -# Use of this source code is governed by a BSD-style license -# that can be found in the LICENSE file in the root of the source -# tree. An additional intellectual property rights grant can be found -# in the file PATENTS. All contributing project authors may -# be found in the AUTHORS file in the root of the source tree. -"""Script for constraining traffic on the local machine.""" - -import ctypes -import logging -import os -import subprocess -import sys - - -class NetworkEmulatorError(BaseException): - """Exception raised for errors in the network emulator. - - Attributes: - fail_msg: User defined error message. - cmd: Command for which the exception was raised. - returncode: Return code of running the command. - stdout: Output of running the command. - stderr: Error output of running the command. - """ - - def __init__(self, - fail_msg, - cmd=None, - returncode=None, - output=None, - error=None): - BaseException.__init__(self, fail_msg) - self.fail_msg = fail_msg - self.cmd = cmd - self.returncode = returncode - self.output = output - self.error = error - - -class NetworkEmulator: - """A network emulator that can constrain the network using Dummynet.""" - - def __init__(self, connection_config, port_range): - """Constructor. - - Args: - connection_config: A config.ConnectionConfig object containing the - characteristics for the connection to be emulation. - port_range: Tuple containing two integers defining the port range. - """ - self._pipe_counter = 0 - self._rule_counter = 0 - self._port_range = port_range - self._connection_config = connection_config - - def Emulate(self, target_ip): - """Starts a network emulation by setting up Dummynet rules. - - Args: - target_ip: The IP address of the interface that shall be that have the - network constraints applied to it. - """ - receive_pipe_id = self._CreateDummynetPipe( - self._connection_config.receive_bw_kbps, - self._connection_config.delay_ms, - self._connection_config.packet_loss_percent, - self._connection_config.queue_slots) - logging.debug('Created receive pipe: %s', receive_pipe_id) - send_pipe_id = self._CreateDummynetPipe( - self._connection_config.send_bw_kbps, self._connection_config.delay_ms, - self._connection_config.packet_loss_percent, - self._connection_config.queue_slots) - logging.debug('Created send pipe: %s', send_pipe_id) - - # Adding the rules will start the emulation. - incoming_rule_id = self._CreateDummynetRule(receive_pipe_id, 'any', - target_ip, self._port_range) - logging.debug('Created incoming rule: %s', incoming_rule_id) - outgoing_rule_id = self._CreateDummynetRule(send_pipe_id, target_ip, 'any', - self._port_range) - logging.debug('Created outgoing rule: %s', outgoing_rule_id) - - @staticmethod - def CheckPermissions(): - """Checks if permissions are available to run Dummynet commands. - - Raises: - NetworkEmulatorError: If permissions to run Dummynet commands are not - available. - """ - try: - if os.getuid() != 0: - raise NetworkEmulatorError('You must run this script with sudo.') - except AttributeError as permission_error: - - # AttributeError will be raised on Windows. - if ctypes.windll.shell32.IsUserAnAdmin() == 0: - raise NetworkEmulatorError('You must run this script with administrator' - ' privileges.') from permission_error - - def _CreateDummynetRule(self, pipe_id, from_address, to_address, port_range): - """Creates a network emulation rule and returns its ID. - - Args: - pipe_id: integer ID of the pipe. - from_address: The IP address to match source address. May be an IP or - 'any'. - to_address: The IP address to match destination address. May be an IP or - 'any'. - port_range: The range of ports the rule shall be applied on. Must be - specified as a tuple of with two integers. - Returns: - The ID of the rule, starting at 100. The rule ID increments with 100 for - each rule being added. - """ - self._rule_counter += 100 - add_part = [ - 'add', self._rule_counter, 'pipe', pipe_id, 'ip', 'from', from_address, - 'to', to_address - ] - _RunIpfwCommand(add_part + ['src-port', '%s-%s' % port_range], - 'Failed to add Dummynet src-port rule.') - _RunIpfwCommand(add_part + ['dst-port', '%s-%s' % port_range], - 'Failed to add Dummynet dst-port rule.') - return self._rule_counter - - def _CreateDummynetPipe(self, bandwidth_kbps, delay_ms, packet_loss_percent, - queue_slots): - """Creates a Dummynet pipe and return its ID. - - Args: - bandwidth_kbps: Bandwidth. - delay_ms: Delay for a one-way trip of a packet. - packet_loss_percent: Float value of packet loss, in percent. - queue_slots: Size of the queue. - Returns: - The ID of the pipe, starting at 1. - """ - self._pipe_counter += 1 - cmd = [ - 'pipe', self._pipe_counter, 'config', 'bw', - str(bandwidth_kbps / 8) + 'KByte/s', 'delay', - '%sms' % delay_ms, 'plr', (packet_loss_percent / 100.0), 'queue', - queue_slots - ] - error_message = 'Failed to create Dummynet pipe. ' - if sys.platform.startswith('linux'): - error_message += ('Make sure you have loaded the ipfw_mod.ko module to ' - 'your kernel (sudo insmod /path/to/ipfw_mod.ko).') - _RunIpfwCommand(cmd, error_message) - return self._pipe_counter - - -def Cleanup(): - """Stops the network emulation by flushing all Dummynet rules. - - Notice that this will flush any rules that may have been created previously - before starting the emulation. - """ - _RunIpfwCommand(['-f', 'flush'], 'Failed to flush Dummynet rules!') - _RunIpfwCommand(['-f', 'pipe', 'flush'], 'Failed to flush Dummynet pipes!') - - -def _RunIpfwCommand(command, fail_msg=None): - """Executes a command and prefixes the appropriate command for - Windows or Linux/UNIX. - - Args: - command: Command list to execute. - fail_msg: Message describing the error in case the command fails. - - Raises: - NetworkEmulatorError: If command fails a message is set by the fail_msg - parameter. - """ - if sys.platform == 'win32': - ipfw_command = ['ipfw.exe'] - else: - ipfw_command = ['sudo', '-n', 'ipfw'] - - cmd_list = ipfw_command[:] + [str(x) for x in command] - cmd_string = ' '.join(cmd_list) - logging.debug('Running command: %s', cmd_string) - process = subprocess.Popen(cmd_list, - stdout=subprocess.PIPE, - stderr=subprocess.PIPE) - output, error = process.communicate() - if process.returncode != 0: - raise NetworkEmulatorError(fail_msg, cmd_string, process.returncode, output, - error) - return output.strip() From 65ce9656f3fe01d8ca9f913976b128fdf203fb4c Mon Sep 17 00:00:00 2001 From: Junji Watanabe Date: Wed, 21 May 2025 16:37:17 +0900 Subject: [PATCH 235/292] infra: Remove $build/reclient properties from builders Reclient isn't used by bots anymore. Bug: b/412968361 Change-Id: I556ea9b4c11a705f6a804f91711464da75fe1d21 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392920 Reviewed-by: Mirko Bonadei Reviewed-by: Christoffer Dewerin Auto-Submit: Junji Watanabe Commit-Queue: Christoffer Dewerin Cr-Commit-Position: refs/heads/main@{#44707} --- infra/config/config.star | 16 +- infra/config/cr-buildbucket.cfg | 460 -------------------------------- 2 files changed, 5 insertions(+), 471 deletions(-) diff --git a/infra/config/config.star b/infra/config/config.star index 2ee74ee097..314f6d5aaa 100755 --- a/infra/config/config.star +++ b/infra/config/config.star @@ -18,19 +18,15 @@ DEFAULT_CPU = "x86-64" # Helpers: -def make_rbe_properties(instance, jobs = None): +def make_siso_properties(instance, jobs = None): """Makes a default RBE property with the specified argument. Args: instance: RBE insatnce name. jobs: Number of jobs to be used by the builder. Returns: - A dictonary with the reclient properties. + A dictonary with the siso properties. """ - reclient_props = { - "instance": instance, - "metrics_project": "chromium-reclient-metrics", - } siso_props = { "project": instance, "configs": ["builder"], @@ -39,10 +35,8 @@ def make_rbe_properties(instance, jobs = None): "enable_monitoring": True, } if jobs: - reclient_props["jobs"] = jobs siso_props["remote_jobs"] = jobs props = { - "$build/reclient": reclient_props, "$build/siso": siso_props, } return props @@ -600,7 +594,7 @@ def ci_builder( dimensions["builderless"] = "1" properties = properties or {} properties["builder_group"] = "client.webrtc" - properties.update(make_rbe_properties("rbe-webrtc-trusted")) + properties.update(make_siso_properties("rbe-webrtc-trusted")) notifies = ["post_submit_failure_notifier", "infra_failure_notifier"] notifies += ["webrtc_tree_closer"] if name not in skipped_lkgr_bots else [] @@ -645,7 +639,7 @@ def try_builder( dimensions["builderless"] = "1" properties = properties or {} properties["builder_group"] = "tryserver.webrtc" - properties.update(make_rbe_properties("rbe-webrtc-untrusted")) + properties.update(make_siso_properties("rbe-webrtc-untrusted")) if cq != None: luci.cq_tryjob_verifier(name, cq_group = "cq", **cq) if branch_cq: @@ -674,7 +668,7 @@ def perf_builder(name, perf_cat, **kwargs): Notifications are also disabled. """ add_milo(name, {"perf": perf_cat}) - properties = make_rbe_properties("rbe-webrtc-trusted") + properties = make_siso_properties("rbe-webrtc-trusted") properties["builder_group"] = "client.webrtc.perf" dimensions = {"pool": "luci.webrtc.perf", "os": "Linux"} return webrtc_builder( diff --git a/infra/config/cr-buildbucket.cfg b/infra/config/cr-buildbucket.cfg index 8866e25b77..8ff8135014 100644 --- a/infra/config/cr-buildbucket.cfg +++ b/infra/config/cr-buildbucket.cfg @@ -29,10 +29,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -85,10 +81,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -141,10 +133,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -197,10 +185,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -253,10 +237,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -309,10 +289,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -365,10 +341,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -421,10 +393,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -477,10 +445,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -533,10 +497,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -589,10 +549,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -645,10 +601,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -701,10 +653,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -757,10 +705,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -813,10 +757,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -869,10 +809,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -925,10 +861,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -981,10 +913,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -1037,10 +965,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -1093,10 +1017,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -1149,10 +1069,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -1205,10 +1121,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -1261,10 +1173,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -1317,10 +1225,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -1373,10 +1277,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -1429,10 +1329,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -1485,10 +1381,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -1541,10 +1433,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -1597,10 +1485,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -1653,10 +1537,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -1709,10 +1589,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -1765,10 +1641,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -1821,10 +1693,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -1877,10 +1745,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -1938,10 +1802,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -1999,10 +1859,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -2060,10 +1916,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -2345,10 +2197,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -2401,10 +2249,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -2457,10 +2301,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -2513,10 +2353,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -2569,10 +2405,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -2625,10 +2457,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -2679,10 +2507,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -2733,10 +2557,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -2787,10 +2607,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -2841,10 +2657,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -2895,10 +2707,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -2949,10 +2757,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -3003,10 +2807,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -3059,10 +2859,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-trusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -3135,10 +2931,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -3191,10 +2983,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -3247,10 +3035,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -3303,10 +3087,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -3359,10 +3139,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -3415,10 +3191,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -3471,10 +3243,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -3527,10 +3295,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -3583,10 +3347,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -3639,10 +3399,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -3695,10 +3451,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -3751,10 +3503,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -3807,10 +3555,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -3863,10 +3607,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -3919,10 +3659,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -3975,10 +3711,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -4036,10 +3768,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -4097,10 +3825,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -4158,10 +3882,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -4219,10 +3939,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -4275,10 +3991,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -4331,10 +4043,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -4387,10 +4095,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -4443,10 +4147,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -4499,10 +4199,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -4555,10 +4251,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -4611,10 +4303,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -4667,10 +4355,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -4723,10 +4407,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -4779,10 +4459,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -4835,10 +4511,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -4891,10 +4563,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -4947,10 +4615,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -5003,10 +4667,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -5059,10 +4719,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -5115,10 +4771,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -5171,10 +4823,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -5227,10 +4875,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -5283,10 +4927,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -5339,10 +4979,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -5395,10 +5031,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -5451,10 +5083,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -5507,10 +5135,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -5563,10 +5187,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -5619,10 +5239,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -5675,10 +5291,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -5731,10 +5343,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -5787,10 +5395,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -5843,10 +5447,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -5901,10 +5501,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -5957,10 +5553,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -6013,10 +5605,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -6069,10 +5657,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -6125,10 +5709,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -6181,10 +5761,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -6237,10 +5813,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -6293,10 +5865,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -6349,10 +5917,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -6405,10 +5969,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -6461,10 +6021,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -6517,10 +6073,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -6573,10 +6125,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -6629,10 +6177,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' @@ -6685,10 +6229,6 @@ buckets { } properties: '{' - ' "$build/reclient": {' - ' "instance": "rbe-webrtc-untrusted",' - ' "metrics_project": "chromium-reclient-metrics"' - ' },' ' "$build/siso": {' ' "configs": [' ' "builder"' From cf3ef74a9a8ad40b9d10eeed239588d0f1940bb7 Mon Sep 17 00:00:00 2001 From: Alexander Cooper Date: Tue, 20 May 2025 13:50:21 -0700 Subject: [PATCH 236/292] Speculative fix for ScreencastResult crash There are low amounts of a crash on this line. While it is possible that this is due to a CFI error in `StartScreenCastStream`, the low hit rate could also be indicative of a rare failure where the screencast_stream may be null. Speculatively add a null check to this location. Bug: chromium:354145778 Change-Id: Ibc71abd93f862fd78a0f96e85125fe974201967d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392760 Commit-Queue: Ilya Nikolaevskiy Reviewed-by: Ilya Nikolaevskiy Auto-Submit: Alexander Cooper Cr-Commit-Position: refs/heads/main@{#44708} --- modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc | 1 + 1 file changed, 1 insertion(+) diff --git a/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc b/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc index 7228063401..76340ced41 100644 --- a/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc +++ b/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc @@ -73,6 +73,7 @@ void BaseCapturerPipeWire::OnScreenCastRequestResult(RequestResponse result, // then it'll set it to the right value again soon enough. capturer_failed_ = false; if (result != RequestResponse::kSuccess || + !options_.screencast_stream() || !options_.screencast_stream()->StartScreenCastStream( stream_node_id, fd, options_.get_width(), options_.get_height(), options_.prefer_cursor_embedded(), From 068e6ba6962bccea53c826df06a9d79db5101e69 Mon Sep 17 00:00:00 2001 From: Diep Bui Date: Tue, 20 May 2025 12:33:24 +0000 Subject: [PATCH 237/292] Update new default loss based configs. MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Also update and removed duplicated unit tests. Bug: webrtc:12707 Change-Id: I0faa207424ac87aabcdc2ffbff6b3ab5a23a764c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392102 Reviewed-by: Per Kjellander Commit-Queue: Diep Bui Reviewed-by: Åsa Persson Cr-Commit-Position: refs/heads/main@{#44709} --- .../goog_cc/loss_based_bwe_v2.cc | 30 ++- .../goog_cc/loss_based_bwe_v2_test.cc | 207 ++++-------------- video/video_send_stream_tests.cc | 5 +- 3 files changed, 65 insertions(+), 177 deletions(-) diff --git a/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc b/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc index 92150715cb..c2f3018dec 100644 --- a/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc +++ b/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc @@ -402,9 +402,9 @@ std::optional LossBasedBweV2::CreateConfig( const FieldTrialsView* key_value_config) { FieldTrialParameter enabled("Enabled", true); FieldTrialParameter bandwidth_rampup_upper_bound_factor( - "BwRampupUpperBoundFactor", 1000000.0); + "BwRampupUpperBoundFactor", 1.5); FieldTrialParameter bandwidth_rampup_upper_bound_factor_in_hold( - "BwRampupUpperBoundInHoldFactor", 1000000.0); + "BwRampupUpperBoundInHoldFactor", 1.2); FieldTrialParameter bandwidth_rampup_hold_threshold( "BwRampupUpperBoundHoldThreshold", 1.3); FieldTrialParameter rampup_acceleration_max_factor( @@ -420,11 +420,11 @@ std::optional LossBasedBweV2::CreateConfig( FieldTrialParameter inherent_loss_lower_bound( "InherentLossLowerBound", 1.0e-3); FieldTrialParameter loss_threshold_of_high_bandwidth_preference( - "LossThresholdOfHighBandwidthPreference", 0.15); + "LossThresholdOfHighBandwidthPreference", 0.2); FieldTrialParameter bandwidth_preference_smoothing_factor( "BandwidthPreferenceSmoothingFactor", 0.002); FieldTrialParameter inherent_loss_upper_bound_bandwidth_balance( - "InherentLossUpperBoundBwBalance", DataRate::KilobitsPerSec(75.0)); + "InherentLossUpperBoundBwBalance", DataRate::KilobitsPerSec(100.0)); FieldTrialParameter inherent_loss_upper_bound_offset( "InherentLossUpperBoundOffset", 0.05); FieldTrialParameter initial_inherent_loss_estimate( @@ -439,13 +439,13 @@ std::optional LossBasedBweV2::CreateConfig( "UpperBoundCandidateInAlr", false); FieldTrialParameter observation_duration_lower_bound( "ObservationDurationLowerBound", TimeDelta::Millis(250)); - FieldTrialParameter observation_window_size("ObservationWindowSize", 20); + FieldTrialParameter observation_window_size("ObservationWindowSize", 15); FieldTrialParameter sending_rate_smoothing_factor( "SendingRateSmoothingFactor", 0.0); FieldTrialParameter instant_upper_bound_temporal_weight_factor( "InstantUpperBoundTemporalWeightFactor", 0.9); FieldTrialParameter instant_upper_bound_bandwidth_balance( - "InstantUpperBoundBwBalance", DataRate::KilobitsPerSec(75.0)); + "InstantUpperBoundBwBalance", DataRate::KilobitsPerSec(100.0)); FieldTrialParameter instant_upper_bound_loss_offset( "InstantUpperBoundLossOffset", 0.05); FieldTrialParameter temporal_weight_factor("TemporalWeightFactor", @@ -460,15 +460,15 @@ std::optional LossBasedBweV2::CreateConfig( "NotIncreaseIfInherentLossLessThanAverageLoss", true); FieldTrialParameter not_use_acked_rate_in_alr("NotUseAckedRateInAlr", true); - FieldTrialParameter use_in_start_phase("UseInStartPhase", false); + FieldTrialParameter use_in_start_phase("UseInStartPhase", true); FieldTrialParameter min_num_observations("MinNumObservations", 3); FieldTrialParameter lower_bound_by_acked_rate_factor( - "LowerBoundByAckedRateFactor", 0.0); - FieldTrialParameter hold_duration_factor("HoldDurationFactor", 0.0); - FieldTrialParameter use_byte_loss_rate("UseByteLossRate", false); + "LowerBoundByAckedRateFactor", 1.0); + FieldTrialParameter hold_duration_factor("HoldDurationFactor", 2.0); + FieldTrialParameter use_byte_loss_rate("UseByteLossRate", true); FieldTrialParameter padding_duration("PaddingDuration", - TimeDelta::Zero()); - FieldTrialParameter bound_best_candidate("BoundBestCandidate", false); + TimeDelta::Seconds(2)); + FieldTrialParameter bound_best_candidate("BoundBestCandidate", true); FieldTrialParameter median_sending_rate_factor( "MedianSendingRateFactor", 2.0); if (key_value_config) { @@ -854,6 +854,12 @@ double LossBasedBweV2::CalculateAverageReportedByteLossRatio() const { // the loss rate might not be due to a spike. return lost_bytes / total_bytes; } + + if (total_bytes == max_bytes_received + min_bytes_received) { + // It could happen if the observation window was 2. + return lost_bytes / total_bytes; + } + return (lost_bytes - min_lost_bytes - max_lost_bytes) / (total_bytes - max_bytes_received - min_bytes_received); } diff --git a/modules/congestion_controller/goog_cc/loss_based_bwe_v2_test.cc b/modules/congestion_controller/goog_cc/loss_based_bwe_v2_test.cc index 0550e4acd3..e308ff8730 100644 --- a/modules/congestion_controller/goog_cc/loss_based_bwe_v2_test.cc +++ b/modules/congestion_controller/goog_cc/loss_based_bwe_v2_test.cc @@ -642,17 +642,12 @@ TEST_F(LossBasedBweV2Test, delay_based_estimate); } -TEST_F(LossBasedBweV2Test, - IncreaseByMaxIncreaseFactorAfterLossBasedBweBacksOff) { - ExplicitKeyValueConfig key_value_config(ShortObservationConfig( - "CandidateFactors:1.2|1|0.5," - "InstantUpperBoundBwBalance:10000kbps," - "MaxIncreaseFactor:1.5,NotIncreaseIfInherentLossLessThanAverageLoss:" - "false")); +TEST_F(LossBasedBweV2Test, IncreaseByHoldFactorAfterLossBasedBweBacksOff) { + ExplicitKeyValueConfig key_value_config(ShortObservationConfig("")); LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config); DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000); - DataRate acked_rate = DataRate::KilobitsPerSec(300); + DataRate acked_rate = DataRate::KilobitsPerSec(100); loss_based_bandwidth_estimator.SetBandwidthEstimate( DataRate::KilobitsPerSec(600)); loss_based_bandwidth_estimator.SetAcknowledgedBitrate(acked_rate); @@ -668,20 +663,23 @@ TEST_F(LossBasedBweV2Test, loss_based_bandwidth_estimator.GetLossBasedResult(); // Network recovers after loss. - std::vector enough_feedback_2 = - CreatePacketResultsWithReceivedPackets( - /*first_packet_timestamp=*/Timestamp::Zero() + - kObservationDurationLowerBound); - loss_based_bandwidth_estimator.SetAcknowledgedBitrate( - DataRate::KilobitsPerSec(600)); - loss_based_bandwidth_estimator.UpdateBandwidthEstimate(enough_feedback_2, - delay_based_estimate, - /*in_alr=*/false); + int feedback_count = 2; + while (loss_based_bandwidth_estimator.GetLossBasedResult().state != + LossBasedState::kIncreaseUsingPadding) { + loss_based_bandwidth_estimator.UpdateBandwidthEstimate( + CreatePacketResultsWithReceivedPackets( + /*first_packet_timestamp=*/Timestamp::Zero() + + kObservationDurationLowerBound * feedback_count), + delay_based_estimate, + /*in_alr=*/false); + feedback_count++; + } LossBasedBweV2::Result result_after_recovery = loss_based_bandwidth_estimator.GetLossBasedResult(); + // BwRampupUpperBoundInHoldFactor is 1.2. EXPECT_EQ(result_after_recovery.bandwidth_estimate, - result_at_loss.bandwidth_estimate * 1.5); + result_at_loss.bandwidth_estimate * 1.2); } TEST_F(LossBasedBweV2Test, @@ -803,7 +801,7 @@ TEST_F(LossBasedBweV2Test, loss_based_bandwidth_estimator.SetAcknowledgedBitrate(estimate_1 * 0.9); int feedback_count = 1; - while (feedback_count < 5 && result.state != LossBasedState::kIncreasing) { + while (result.state != LossBasedState::kIncreaseUsingPadding) { loss_based_bandwidth_estimator.UpdateBandwidthEstimate( CreatePacketResultsWithReceivedPackets( /*first_packet_timestamp=*/Timestamp::Zero() + @@ -812,7 +810,7 @@ TEST_F(LossBasedBweV2Test, /*in_alr=*/false); result = loss_based_bandwidth_estimator.GetLossBasedResult(); } - ASSERT_EQ(result.state, LossBasedState::kIncreasing); + ASSERT_EQ(result.state, LossBasedState::kIncreaseUsingPadding); // The estimate is capped by acked_bitrate * BwRampupUpperBoundFactor. EXPECT_EQ(result.bandwidth_estimate, estimate_1 * 0.9 * 1.2); @@ -863,7 +861,7 @@ TEST_F(LossBasedBweV2Test, EnsureIncreaseEvenIfAckedBitrateBound) { loss_based_bandwidth_estimator.SetAcknowledgedBitrate(estimate_1 / 2); int feedback_count = 1; - while (feedback_count < 5 && result.state != LossBasedState::kIncreasing) { + while (result.state != LossBasedState::kIncreaseUsingPadding) { loss_based_bandwidth_estimator.UpdateBandwidthEstimate( CreatePacketResultsWithReceivedPackets( /*first_packet_timestamp=*/Timestamp::Zero() + @@ -873,7 +871,7 @@ TEST_F(LossBasedBweV2Test, EnsureIncreaseEvenIfAckedBitrateBound) { result = loss_based_bandwidth_estimator.GetLossBasedResult(); } - ASSERT_EQ(result.state, LossBasedState::kIncreasing); + ASSERT_EQ(result.state, LossBasedState::kIncreaseUsingPadding); // The estimate increases by 1kbps. EXPECT_EQ(result.bandwidth_estimate, estimate_1 + DataRate::BitsPerSec(1)); } @@ -1238,7 +1236,7 @@ TEST_F(LossBasedBweV2Test, HasDecreaseStateBecauseOfUpperBound) { loss_based_bandwidth_estimator.SetBandwidthEstimate( DataRate::KilobitsPerSec(500)); loss_based_bandwidth_estimator.SetAcknowledgedBitrate( - DataRate::KilobitsPerSec(500)); + DataRate::KilobitsPerSec(100)); std::vector enough_feedback_10p_loss_1 = CreatePacketResultsWith10pPacketLossRate( @@ -1287,19 +1285,19 @@ TEST_F(LossBasedBweV2Test, HasIncreaseStateBecauseOfLowerBound) { std::vector enough_feedback_50p_loss_2 = CreatePacketResultsWith50pPacketLossRate( /*first_packet_timestamp=*/Timestamp::Zero() + - kObservationDurationLowerBound); + kObservationDurationLowerBound * 2); loss_based_bandwidth_estimator.UpdateBandwidthEstimate( enough_feedback_50p_loss_2, /*delay_based_estimate=*/DataRate::PlusInfinity(), /*in_alr=*/false); // Verify that the instant lower bound increases the estimate, and state is - // updated to kIncreasing. + // updated to kIncreaseUsingPadding. EXPECT_EQ( loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate, DataRate::KilobitsPerSec(200) * 10); EXPECT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state, - LossBasedState::kIncreasing); + LossBasedState::kIncreaseUsingPadding); } TEST_F(LossBasedBweV2Test, @@ -1365,8 +1363,7 @@ TEST_F(LossBasedBweV2Test, HasDelayBasedStateIfLossBasedBweIsMax) { // Eventually the estimator recovers to delay based state. int feedback_count = 2; - while (feedback_count < 5 && - result.state != LossBasedState::kDelayBasedEstimate) { + while (result.state != LossBasedState::kDelayBasedEstimate) { loss_based_bandwidth_estimator.UpdateBandwidthEstimate( /*feedback = */ CreatePacketResultsWithReceivedPackets( /*first_packet_timestamp=*/Timestamp::Zero() + @@ -1394,13 +1391,17 @@ TEST_F(LossBasedBweV2Test, IncreaseUsingPaddingStateIfFieldTrial) { /*in_alr=*/false); ASSERT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state, LossBasedState::kDecreasing); - - loss_based_bandwidth_estimator.UpdateBandwidthEstimate( - CreatePacketResultsWithReceivedPackets( - /*first_packet_timestamp=*/Timestamp::Zero() + - kObservationDurationLowerBound), - /*delay_based_estimate=*/DataRate::PlusInfinity(), - /*in_alr=*/false); + int feedback_count = 1; + while (loss_based_bandwidth_estimator.GetLossBasedResult().state != + LossBasedState::kIncreaseUsingPadding) { + loss_based_bandwidth_estimator.UpdateBandwidthEstimate( + CreatePacketResultsWithReceivedPackets( + /*first_packet_timestamp=*/Timestamp::Zero() + + kObservationDurationLowerBound * feedback_count), + /*delay_based_estimate=*/DataRate::PlusInfinity(), + /*in_alr=*/false); + feedback_count++; + } EXPECT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state, LossBasedState::kIncreaseUsingPadding); } @@ -1546,131 +1547,6 @@ TEST_F(LossBasedBweV2Test, DecreaseAfterPadding) { EXPECT_EQ(start_decreasing - estimate_increased, TimeDelta::Seconds(1)); } -TEST_F(LossBasedBweV2Test, IncreaseEstimateIfNotHold) { - ExplicitKeyValueConfig key_value_config( - ShortObservationConfig("HoldDurationFactor:0")); - LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config); - loss_based_bandwidth_estimator.SetBandwidthEstimate( - DataRate::KilobitsPerSec(2500)); - loss_based_bandwidth_estimator.UpdateBandwidthEstimate( - CreatePacketResultsWith50pPacketLossRate( - /*first_packet_timestamp=*/Timestamp::Zero()), - /*delay_based_estimate=*/DataRate::PlusInfinity(), - /*in_alr=*/false); - ASSERT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state, - LossBasedState::kDecreasing); - DataRate estimate = - loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate; - - loss_based_bandwidth_estimator.UpdateBandwidthEstimate( - CreatePacketResultsWithReceivedPackets( - /*first_packet_timestamp=*/Timestamp::Zero() + - kObservationDurationLowerBound), - /*delay_based_estimate=*/DataRate::PlusInfinity(), - /*in_alr=*/false); - EXPECT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state, - LossBasedState::kIncreasing); - EXPECT_GT( - loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate, - estimate); -} - -TEST_F(LossBasedBweV2Test, IncreaseEstimateAfterHoldDuration) { - ExplicitKeyValueConfig key_value_config( - ShortObservationConfig("HoldDurationFactor:10")); - LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config); - loss_based_bandwidth_estimator.SetBandwidthEstimate( - DataRate::KilobitsPerSec(2500)); - loss_based_bandwidth_estimator.UpdateBandwidthEstimate( - CreatePacketResultsWith50pPacketLossRate( - /*first_packet_timestamp=*/Timestamp::Zero()), - /*delay_based_estimate=*/DataRate::PlusInfinity(), - /*in_alr=*/false); - ASSERT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state, - LossBasedState::kDecreasing); - DataRate estimate = - loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate; - - // During the hold duration, e.g. first 300ms, the estimate cannot increase. - loss_based_bandwidth_estimator.UpdateBandwidthEstimate( - CreatePacketResultsWithReceivedPackets( - /*first_packet_timestamp=*/Timestamp::Zero() + - kObservationDurationLowerBound), - /*delay_based_estimate=*/DataRate::PlusInfinity(), - /*in_alr=*/false); - EXPECT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state, - LossBasedState::kDecreasing); - EXPECT_EQ( - loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate, - estimate); - - // After the hold duration, the estimate can increase. - loss_based_bandwidth_estimator.UpdateBandwidthEstimate( - CreatePacketResultsWithReceivedPackets( - /*first_packet_timestamp=*/Timestamp::Zero() + - kObservationDurationLowerBound * 2), - /*delay_based_estimate=*/DataRate::PlusInfinity(), - /*in_alr=*/false); - EXPECT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state, - LossBasedState::kIncreasing); - EXPECT_GE( - loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate, - estimate); - - // Get another 50p packet loss. - loss_based_bandwidth_estimator.UpdateBandwidthEstimate( - CreatePacketResultsWith50pPacketLossRate( - /*first_packet_timestamp=*/Timestamp::Zero() + - kObservationDurationLowerBound * 3), - /*delay_based_estimate=*/DataRate::PlusInfinity(), - /*in_alr=*/false); - EXPECT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state, - LossBasedState::kDecreasing); - DataRate estimate_at_hold = - loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate; - - // In the hold duration, e.g. next 3s, the estimate cannot increase above the - // hold rate. Get some lost packets to get lower estimate than the HOLD rate. - for (int i = 4; i <= 6; ++i) { - loss_based_bandwidth_estimator.UpdateBandwidthEstimate( - CreatePacketResultsWith100pLossRate( - /*first_packet_timestamp=*/Timestamp::Zero() + - kObservationDurationLowerBound * i), - /*delay_based_estimate=*/DataRate::PlusInfinity(), - /*in_alr=*/false); - EXPECT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state, - LossBasedState::kDecreasing); - EXPECT_LT( - loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate, - estimate_at_hold); - } - - int feedback_id = 7; - while (loss_based_bandwidth_estimator.GetLossBasedResult().state != - LossBasedState::kIncreasing) { - loss_based_bandwidth_estimator.UpdateBandwidthEstimate( - CreatePacketResultsWithReceivedPackets( - /*first_packet_timestamp=*/Timestamp::Zero() + - kObservationDurationLowerBound * feedback_id), - /*delay_based_estimate=*/DataRate::PlusInfinity(), - /*in_alr=*/false); - if (loss_based_bandwidth_estimator.GetLossBasedResult().state == - LossBasedState::kDecreasing) { - // In the hold duration, the estimate can not go higher than estimate at - // hold. - EXPECT_LE(loss_based_bandwidth_estimator.GetLossBasedResult() - .bandwidth_estimate, - estimate_at_hold); - } else if (loss_based_bandwidth_estimator.GetLossBasedResult().state == - LossBasedState::kIncreasing) { - // After the hold duration, the estimate can increase again. - EXPECT_GT(loss_based_bandwidth_estimator.GetLossBasedResult() - .bandwidth_estimate, - estimate_at_hold); - } - feedback_id++; - } -} TEST_F(LossBasedBweV2Test, HoldRateNotLowerThanAckedRate) { ExplicitKeyValueConfig key_value_config(ShortObservationConfig( @@ -1749,8 +1625,7 @@ TEST_F(LossBasedBweV2Test, EstimateNotLowerThanAckedRate) { } TEST_F(LossBasedBweV2Test, EndHoldDurationIfDelayBasedEstimateWorks) { - ExplicitKeyValueConfig key_value_config( - ShortObservationConfig("HoldDurationFactor:3")); + ExplicitKeyValueConfig key_value_config(ShortObservationConfig("")); LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config); loss_based_bandwidth_estimator.SetBandwidthEstimate( DataRate::KilobitsPerSec(2500)); @@ -1770,6 +1645,12 @@ TEST_F(LossBasedBweV2Test, EndHoldDurationIfDelayBasedEstimateWorks) { kObservationDurationLowerBound), /*delay_based_estimate=*/estimate + DataRate::KilobitsPerSec(10), /*in_alr=*/false); + loss_based_bandwidth_estimator.UpdateBandwidthEstimate( + CreatePacketResultsWithReceivedPackets( + /*first_packet_timestamp=*/Timestamp::Zero() + + kObservationDurationLowerBound * 2), + /*delay_based_estimate=*/estimate + DataRate::KilobitsPerSec(10), + /*in_alr=*/false); EXPECT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state, LossBasedState::kDelayBasedEstimate); EXPECT_EQ( @@ -1795,7 +1676,7 @@ TEST_F(LossBasedBweV2Test, UseByteLossRate) { // The estimate is bounded by the instant upper bound due to high loss. EXPECT_LT( loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate, - DataRate::KilobitsPerSec(150)); + DataRate::KilobitsPerSec(160)); } TEST_F(LossBasedBweV2Test, UseByteLossRateIgnoreLossSpike) { diff --git a/video/video_send_stream_tests.cc b/video/video_send_stream_tests.cc index d73601ebc2..7003ffe933 100644 --- a/video/video_send_stream_tests.cc +++ b/video/video_send_stream_tests.cc @@ -1537,8 +1537,9 @@ TEST_F(VideoSendStreamTest, PaddingIsPrimarilyRetransmissions) { SleepMs(5000); { MutexLock lock(&mutex_); - // Expect padding to be a small percentage of total bytes sent. - EXPECT_LT(padding_length_, .1 * total_length_); + // Expect padding to be some percentage of total bytes sent. + // BWE uses padding to ramp up after network has recovered from loss. + EXPECT_LT(padding_length_, .6 * total_length_); } } From 955a8474e674068b397720e69297281535969a30 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Wed, 21 May 2025 01:03:13 -0700 Subject: [PATCH 238/292] Roll chromium_revision b019ea34ab..c1450c64cd (1463060:1463258) Change log: https://chromium.googlesource.com/chromium/src/+log/b019ea34ab..c1450c64cd Full diff: https://chromium.googlesource.com/chromium/src/+/b019ea34ab..c1450c64cd Changed dependencies * siso_version: git_revision:3753a55d0911b4ea3dd000263088f60edb8d947b..git_revision:d9393c2115244b6e4a797189055e4a2b6769a64d * src/base: https://chromium.googlesource.com/chromium/src/base/+log/436fa45edc..763bbf5746 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/03eb9166dd..3a75313211 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/c2f6abd4e2..c1e8fc5602 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/74a3da3c0c..a19ccd2f8a * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/4c53f1652a..885028080a * src/third_party/android_build_tools/error_prone/cipd: bd18Z8-1Y85m4C8xYA3z1J_VIhAaE2qRbkrBzn4_200C..2jzgic7at7ZxQds7qXCsmkfC4bO7d3qnWU5Q7SFUiJwC * src/third_party/android_build_tools/lint/cipd: 7xZFl5M3w3LIJvhKsJYDQoTh5JZ3vr3I_-VWYXiGJC0C..PaYB6553MH9GJfamUZLduJESRbN13Clv2N2beHR6IQAC * src/third_party/android_build_tools/manifest_merger/cipd: Ym1eWXywKFRvxsgK0yC4bFj9SwZViDqGEPb_w3mAKqMC..0L0N3_u2ypIxEEtf0k_l36g_2CykW2BNuZBXm-v5qAUC * src/third_party/android_deps/autorolled/cipd: gbOkrsK3-7xG1DV1l2XCjQfdKOR_9eY9G2JQ08TEQkwC.._jfsXVgk5GB8GWdnHRUcV__xkPxOGvUXpabqaMcVBfcC * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/a934ee9e1f..be17ebd7c1 * src/third_party/googletest/src: https://chromium.googlesource.com/external/github.com/google/googletest.git/+log/fa8438ae6b..16d4f8eff6 * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/a58a5ffb47..c19a471bea * src/third_party/kotlin_stdlib/cipd: 8JF4DFDjaasGcuqxNRqw0DePRizH5yR1-Al191P4K9wC..GUpKElqF0PYGB-SP4D5w6p_MuMYQSBrRkGqFGjPhsIYC * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/188242925b..410e8a174f * src/third_party/siso/cipd: git_revision:3753a55d0911b4ea3dd000263088f60edb8d947b..git_revision:d9393c2115244b6e4a797189055e4a2b6769a64d * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/369ac5168a..609e781abb DEPS diff: https://chromium.googlesource.com/chromium/src/+/b019ea34ab..c1450c64cd/DEPS BUG=None Change-Id: Ifc1d6941b981de491146ff61efd2444034be6c35 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392842 Commit-Queue: Autoroller Bot-Commit: Autoroller Cr-Commit-Position: refs/heads/main@{#44710} --- DEPS | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) diff --git a/DEPS b/DEPS index baf17d85d2..0841a712c8 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': 'b019ea34ab2070eac0fb94345d2e63bf12267f32', + 'chromium_revision': 'c1450c64cd894812bf4e7928f1bfa64b19e59f01', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -53,7 +53,7 @@ vars = { # reclient CIPD package version 'reclient_version': 're_client_version:0.178.0.5ee9d3e8-gomaip', # siso CIPD package version. - 'siso_version': 'git_revision:3753a55d0911b4ea3dd000263088f60edb8d947b', + 'siso_version': 'git_revision:d9393c2115244b6e4a797189055e4a2b6769a64d', # ninja CIPD package. 'ninja_package': 'infra/3pp/tools/ninja/', @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@436fa45edc5ad3de19e2e46f6a86b0394b79411f', + 'https://chromium.googlesource.com/chromium/src/base@763bbf5746302b010dbdd996a38a204fb0eea0a5', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@03eb9166dd46ed90f914635ab989893ec5a7f6e0', + 'https://chromium.googlesource.com/chromium/src/build@3a7531321173cb50efa60fadb0b41fbef9bda5d9', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@cd169314af80e77a75eb25e2c0df8798f35bc04a', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@c2f6abd4e2fa164e03f01c157659c75a3481b606', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@c1e8fc5602a81ef96015103aab8304eeff044338', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@74a3da3c0c5523157bb7bb34a573f93ca473f007', + 'https://chromium.googlesource.com/chromium/src/testing@a19ccd2f8a23db0c358ff03e6229e6fb3d4e51d0', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@4c53f1652a97b52d478780faeea4273bacef1f3e', + 'https://chromium.googlesource.com/chromium/src/third_party@885028080a17f763a8e6fa90f190f546ff20a4ab', 'src/buildtools/linux64': { 'packages': [ @@ -427,7 +427,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_build_tools/error_prone', - 'version': 'bd18Z8-1Y85m4C8xYA3z1J_VIhAaE2qRbkrBzn4_200C', + 'version': '2jzgic7at7ZxQds7qXCsmkfC4bO7d3qnWU5Q7SFUiJwC', }, ], 'condition': 'checkout_android', @@ -449,7 +449,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_build_tools/lint', - 'version': '7xZFl5M3w3LIJvhKsJYDQoTh5JZ3vr3I_-VWYXiGJC0C', + 'version': 'PaYB6553MH9GJfamUZLduJESRbN13Clv2N2beHR6IQAC', }, ], 'condition': 'checkout_android and non_git_source', @@ -480,7 +480,7 @@ deps = { }, 'src/third_party/boringssl/src': - 'https://boringssl.googlesource.com/boringssl.git@a934ee9e1fe4397e682f9f18b1f4f061d7400f9d', + 'https://boringssl.googlesource.com/boringssl.git@be17ebd7c19cd9f60ff684a953329a25a531a688', 'src/third_party/breakpad/breakpad': 'https://chromium.googlesource.com/breakpad/breakpad.git@fad82477312f786994151f01c352813fabf924c0', 'src/third_party/catapult': @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@a58a5ffb4763e0f922461efbaa493752361b4435', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@c19a471beacf6fc3e8914dcaae7ef331986eba6a', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -525,7 +525,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/googletest/src': - 'https://chromium.googlesource.com/external/github.com/google/googletest.git@fa8438ae6b70c57010177de47a9f13d7041a6328', + 'https://chromium.googlesource.com/external/github.com/google/googletest.git@16d4f8eff6d7cefca6975d82a53f8fc995a6feb7', 'src/third_party/icu': { 'url': 'https://chromium.googlesource.com/chromium/deps/icu.git@b929596baebf0ab4ac7ec07f38365db4c50a559d', }, @@ -563,7 +563,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/kotlin_stdlib', - 'version': '8JF4DFDjaasGcuqxNRqw0DePRizH5yR1-Al191P4K9wC', + 'version': 'GUpKElqF0PYGB-SP4D5w6p_MuMYQSBrRkGqFGjPhsIYC', }, ], 'condition': 'checkout_android', @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '188242925b83331305ff6f8f146b2a6667f21ac2', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '410e8a174f36c6293d3aebd3833d04ac3f367b40', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@369ac5168aef86aaa88feb29e5aa603e181ff932', + 'https://chromium.googlesource.com/chromium/src/tools@609e781abb9a942151d56a37f4e78eada7438862', 'src/third_party/espresso': { 'packages': [ @@ -706,7 +706,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_build_tools/manifest_merger', - 'version': 'Ym1eWXywKFRvxsgK0yC4bFj9SwZViDqGEPb_w3mAKqMC', + 'version': '0L0N3_u2ypIxEEtf0k_l36g_2CykW2BNuZBXm-v5qAUC', }, ], 'condition': 'checkout_android', @@ -827,7 +827,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_deps/autorolled', - 'version': 'gbOkrsK3-7xG1DV1l2XCjQfdKOR_9eY9G2JQ08TEQkwC', + 'version': '_jfsXVgk5GB8GWdnHRUcV__xkPxOGvUXpabqaMcVBfcC', }, ], 'condition': 'checkout_android and non_git_source', From 29aac9e9da09c53cd6951a0338f0ac9f92b8ea0d Mon Sep 17 00:00:00 2001 From: Jonas Oreland Date: Wed, 21 May 2025 08:51:53 +0200 Subject: [PATCH 239/292] WebRTC/PQC - Add ssl_group_id to DtlsTransport::Information 1) So that it is detectable which ssl group was used for handshake. 2) Add PeerConnectionIntegration test 3) Fix minor bug in CryptoOptions::Update (wrt field trials) where it did not keep the expected order... BUG=webrtc:404763475 Change-Id: I817295e9783f8fb7c7bd31382d409675a4e730f9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392680 Reviewed-by: Tomas Gunnarsson Commit-Queue: Jonas Oreland Cr-Commit-Position: refs/heads/main@{#44711} --- api/crypto/crypto_options.cc | 25 ++++-- api/dtls_transport_interface.cc | 10 ++- api/dtls_transport_interface.h | 5 +- p2p/dtls/dtls_transport.cc | 4 + p2p/dtls/dtls_transport.h | 3 + p2p/dtls/dtls_transport_internal.h | 3 + p2p/dtls/fake_dtls_transport.h | 1 + pc/BUILD.gn | 1 + pc/dtls_transport.cc | 6 +- pc/peer_connection_integrationtest.cc | 92 +++++++++++++++++++++ pc/slow_peer_connection_integration_test.cc | 4 +- pc/test/integration_test_helpers.h | 33 ++++---- 12 files changed, 159 insertions(+), 28 deletions(-) diff --git a/api/crypto/crypto_options.cc b/api/crypto/crypto_options.cc index 249bc34105..e939f390e5 100644 --- a/api/crypto/crypto_options.cc +++ b/api/crypto/crypto_options.cc @@ -13,6 +13,7 @@ #include #include #include +#include #include #include "api/field_trials_view.h" @@ -114,11 +115,16 @@ void CryptoOptions::EphemeralKeyExchangeCipherGroups::Update( const std::vector* disabled_groups) { // Note: assumption is that these lists contains few elements...so converting // to set<> is not worth it. - std::vector current; - enabled_ = SSLStreamAdapter::GetDefaultEphemeralKeyExchangeCipherGroups( - field_trials); + std::vector default_groups = + SSLStreamAdapter::GetDefaultEphemeralKeyExchangeCipherGroups( + field_trials); // Remove all disabled. if (disabled_groups) { + default_groups.erase(std::remove_if( + default_groups.begin(), default_groups.end(), [&](uint16_t val) { + return std::find(disabled_groups->begin(), disabled_groups->end(), + val) != disabled_groups->end(); + })); enabled_.erase( std::remove_if(enabled_.begin(), enabled_.end(), [&](uint16_t val) { return std::find(disabled_groups->begin(), disabled_groups->end(), @@ -126,10 +132,17 @@ void CryptoOptions::EphemeralKeyExchangeCipherGroups::Update( })); } - // Add all current not already present to end of list. - auto end = enabled_.end(); + // Add those enabled by field-trials first. + std::vector current = std::move(enabled_); + for (auto val : default_groups) { + if (std::find(current.begin(), current.end(), val) == current.end()) { + enabled_.push_back(val); + } + } + + // Then re-add those present (unless already there). for (auto val : current) { - if (std::find(enabled_.begin(), end, val) == end) { + if (std::find(enabled_.begin(), enabled_.end(), val) == enabled_.end()) { enabled_.push_back(val); } } diff --git a/api/dtls_transport_interface.cc b/api/dtls_transport_interface.cc index 4e672e2201..88d137e8ed 100644 --- a/api/dtls_transport_interface.cc +++ b/api/dtls_transport_interface.cc @@ -30,13 +30,15 @@ DtlsTransportInformation::DtlsTransportInformation( std::optional tls_version, std::optional ssl_cipher_suite, std::optional srtp_cipher_suite, - std::unique_ptr remote_ssl_certificates) + std::unique_ptr remote_ssl_certificates, + std::optional ssl_group_id) : state_(state), role_(role), tls_version_(tls_version), ssl_cipher_suite_(ssl_cipher_suite), srtp_cipher_suite_(srtp_cipher_suite), - remote_ssl_certificates_(std::move(remote_ssl_certificates)) {} + remote_ssl_certificates_(std::move(remote_ssl_certificates)), + ssl_group_id_(ssl_group_id) {} // Deprecated version DtlsTransportInformation::DtlsTransportInformation( @@ -61,7 +63,8 @@ DtlsTransportInformation::DtlsTransportInformation( srtp_cipher_suite_(c.srtp_cipher_suite_), remote_ssl_certificates_(c.remote_ssl_certificates() ? c.remote_ssl_certificates()->Clone() - : nullptr) {} + : nullptr), + ssl_group_id_(c.ssl_group_id_) {} DtlsTransportInformation& DtlsTransportInformation::operator=( const DtlsTransportInformation& c) { @@ -73,6 +76,7 @@ DtlsTransportInformation& DtlsTransportInformation::operator=( remote_ssl_certificates_ = c.remote_ssl_certificates() ? c.remote_ssl_certificates()->Clone() : nullptr; + ssl_group_id_ = c.ssl_group_id_; return *this; } diff --git a/api/dtls_transport_interface.h b/api/dtls_transport_interface.h index 10fc174643..c444313a5d 100644 --- a/api/dtls_transport_interface.h +++ b/api/dtls_transport_interface.h @@ -53,7 +53,8 @@ class RTC_EXPORT DtlsTransportInformation { std::optional tls_version, std::optional ssl_cipher_suite, std::optional srtp_cipher_suite, - std::unique_ptr remote_ssl_certificates); + std::unique_ptr remote_ssl_certificates, + std::optional ssl_group_id); ABSL_DEPRECATED("Use version with role parameter") DtlsTransportInformation( DtlsTransportState state, @@ -75,6 +76,7 @@ class RTC_EXPORT DtlsTransportInformation { std::optional tls_version() const { return tls_version_; } std::optional ssl_cipher_suite() const { return ssl_cipher_suite_; } std::optional srtp_cipher_suite() const { return srtp_cipher_suite_; } + std::optional ssl_group_id() const { return ssl_group_id_; } // The accessor returns a temporary pointer, it does not release ownership. const SSLCertChain* remote_ssl_certificates() const { return remote_ssl_certificates_.get(); @@ -87,6 +89,7 @@ class RTC_EXPORT DtlsTransportInformation { std::optional ssl_cipher_suite_; std::optional srtp_cipher_suite_; std::unique_ptr remote_ssl_certificates_; + std::optional ssl_group_id_; }; class DtlsTransportObserverInterface { diff --git a/p2p/dtls/dtls_transport.cc b/p2p/dtls/dtls_transport.cc index 61a136fddb..0d8f263e11 100644 --- a/p2p/dtls/dtls_transport.cc +++ b/p2p/dtls/dtls_transport.cc @@ -519,6 +519,10 @@ bool DtlsTransportInternalImpl::GetSslVersionBytes(int* version) const { return dtls_->GetSslVersionBytes(version); } +uint16_t DtlsTransportInternalImpl::GetSslGroupId() const { + return dtls_->GetSslGroupId(); +} + uint16_t DtlsTransportInternalImpl::GetSslPeerSignatureAlgorithm() const { if (dtls_state() != DtlsTransportState::kConnected) { return kSslSignatureAlgorithmUnknown; // "not applicable" diff --git a/p2p/dtls/dtls_transport.h b/p2p/dtls/dtls_transport.h index cacd77f451..a7fb055adb 100644 --- a/p2p/dtls/dtls_transport.h +++ b/p2p/dtls/dtls_transport.h @@ -178,6 +178,9 @@ class DtlsTransportInternalImpl : public webrtc::DtlsTransportInternal { // Find out which TLS version was negotiated bool GetSslVersionBytes(int* version) const override; + // Return the the ID of the group used by the adapters most recently + // completed handshake, or 0 if not applicable (e.g. before the handshake). + uint16_t GetSslGroupId() const override; // Find out which DTLS-SRTP cipher was negotiated bool GetSrtpCryptoSuite(int* cipher) const override; diff --git a/p2p/dtls/dtls_transport_internal.h b/p2p/dtls/dtls_transport_internal.h index df17ac600f..c5df3c51cf 100644 --- a/p2p/dtls/dtls_transport_internal.h +++ b/p2p/dtls/dtls_transport_internal.h @@ -64,6 +64,9 @@ class DtlsTransportInternal : public PacketTransportInternal { // Finds out which TLS/DTLS version is running. virtual bool GetSslVersionBytes(int* version) const = 0; + // Return the the ID of the group used by the adapters most recently + // completed handshake, or 0 if not applicable (e.g. before the handshake). + virtual uint16_t GetSslGroupId() const = 0; // Finds out which DTLS-SRTP cipher was negotiated. // TODO(zhihuang): Remove this once all dependencies implement this. virtual bool GetSrtpCryptoSuite(int* cipher) const = 0; diff --git a/p2p/dtls/fake_dtls_transport.h b/p2p/dtls/fake_dtls_transport.h index e75e5e6bca..a0f7596439 100644 --- a/p2p/dtls/fake_dtls_transport.h +++ b/p2p/dtls/fake_dtls_transport.h @@ -207,6 +207,7 @@ class FakeDtlsTransport : public DtlsTransportInternal { *version = 0x0102; return true; } + uint16_t GetSslGroupId() const override { return 0; } bool GetSrtpCryptoSuite(int* crypto_suite) const override { if (!do_dtls_) { return false; diff --git a/pc/BUILD.gn b/pc/BUILD.gn index 7241255ce4..cc366601e5 100644 --- a/pc/BUILD.gn +++ b/pc/BUILD.gn @@ -2683,6 +2683,7 @@ if (rtc_include_tests && !build_with_chromium) { ":video_track_source", "../api:audio_options_api", "../api:candidate", + "../api:dtls_transport_interface", "../api:enable_media_with_defaults", "../api:field_trials", "../api:field_trials_view", diff --git a/pc/dtls_transport.cc b/pc/dtls_transport.cc index f79f1bb203..e95a5cfe67 100644 --- a/pc/dtls_transport.cc +++ b/pc/dtls_transport.cc @@ -134,14 +134,16 @@ void DtlsTransport::UpdateInformation() { set_info(DtlsTransportInformation( internal_dtls_transport_->dtls_state(), role, tls_version, ssl_cipher_suite, srtp_cipher, - internal_dtls_transport_->GetRemoteSSLCertChain())); + internal_dtls_transport_->GetRemoteSSLCertChain(), + internal_dtls_transport_->GetSslGroupId())); } else { RTC_LOG(LS_ERROR) << "DtlsTransport in connected state has incomplete " "TLS information"; set_info(DtlsTransportInformation( internal_dtls_transport_->dtls_state(), role, std::nullopt, std::nullopt, std::nullopt, - internal_dtls_transport_->GetRemoteSSLCertChain())); + internal_dtls_transport_->GetRemoteSSLCertChain(), + /* ssl_group_id= */ std::nullopt)); } } else { set_info( diff --git a/pc/peer_connection_integrationtest.cc b/pc/peer_connection_integrationtest.cc index b083a75fbb..506495b616 100644 --- a/pc/peer_connection_integrationtest.cc +++ b/pc/peer_connection_integrationtest.cc @@ -4997,6 +4997,98 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan, } } +#ifdef WEBRTC_HAVE_SCTP + +TEST_P(PeerConnectionIntegrationTest, DtlsPqc) { + CryptoOptions crypto_options; + crypto_options.ephemeral_key_exchange_cipher_groups.AddFirst( + webrtc::CryptoOptions::EphemeralKeyExchangeCipherGroups:: + kX25519_MLKEM768); + + PeerConnectionInterface::RTCConfiguration config; + config.crypto_options = crypto_options; + + PeerConnectionFactoryInterface::Options options; + options.ssl_max_version = SSL_PROTOCOL_DTLS_13; + + const bool create_media_engine = false; + SetCallerPcWrapperAndReturnCurrent(CreatePeerConnectionWrapper( + "caller", &options, &config, PeerConnectionDependencies(nullptr), + /* event_log_factory= */ nullptr, + /* reset_encoder_factory= */ false, + /* reset_decoder_factory= */ false, create_media_engine)); + SetCalleePcWrapperAndReturnCurrent(CreatePeerConnectionWrapper( + "callee", &options, &config, PeerConnectionDependencies(nullptr), + /* event_log_factory= */ nullptr, + /* reset_encoder_factory= */ false, + /* reset_decoder_factory= */ false, create_media_engine)); + + ConnectFakeSignaling(); + + caller()->CreateDataChannel(); + + caller()->CreateAndSetAndSignalOffer(); + ASSERT_THAT( + WaitUntil( + [&] { + return PeerConnectionStateIs( + PeerConnectionInterface::PeerConnectionState::kConnected); + }, + ::testing::IsTrue()), + IsRtcOk()); + + uint16_t expected = + webrtc::CryptoOptions::EphemeralKeyExchangeCipherGroups::kX25519_MLKEM768; + if (!SSLStreamAdapter::IsBoringSsl()) { + expected = webrtc::CryptoOptions::EphemeralKeyExchangeCipherGroups::kX25519; + } + EXPECT_EQ(caller()->dtls_transport_information().ssl_group_id(), expected); + EXPECT_EQ(caller()->dtls_transport_information().ssl_group_id(), expected); +} + +TEST_P(PeerConnectionIntegrationTest, DtlsPqcFieldTrial) { + SetFieldTrials("WebRTC-EnableDtlsPqc/Enabled/"); + PeerConnectionInterface::RTCConfiguration config; + PeerConnectionFactoryInterface::Options options; + options.ssl_max_version = SSL_PROTOCOL_DTLS_13; + + const bool create_media_engine = false; + SetCallerPcWrapperAndReturnCurrent(CreatePeerConnectionWrapper( + "caller", &options, &config, PeerConnectionDependencies(nullptr), + /* event_log_factory= */ nullptr, + /* reset_encoder_factory= */ false, + /* reset_decoder_factory= */ false, create_media_engine)); + SetCalleePcWrapperAndReturnCurrent(CreatePeerConnectionWrapper( + "callee", &options, &config, PeerConnectionDependencies(nullptr), + /* event_log_factory= */ nullptr, + /* reset_encoder_factory= */ false, + /* reset_decoder_factory= */ false, create_media_engine)); + + ConnectFakeSignaling(); + + caller()->CreateDataChannel(); + + caller()->CreateAndSetAndSignalOffer(); + ASSERT_THAT( + WaitUntil( + [&] { + return PeerConnectionStateIs( + PeerConnectionInterface::PeerConnectionState::kConnected); + }, + ::testing::IsTrue()), + IsRtcOk()); + + uint16_t expected = + webrtc::CryptoOptions::EphemeralKeyExchangeCipherGroups::kX25519_MLKEM768; + if (!SSLStreamAdapter::IsBoringSsl()) { + expected = webrtc::CryptoOptions::EphemeralKeyExchangeCipherGroups::kX25519; + } + EXPECT_EQ(caller()->dtls_transport_information().ssl_group_id(), expected); + EXPECT_EQ(caller()->dtls_transport_information().ssl_group_id(), expected); +} + +#endif // WEBRTC_HAVE_SCTP + } // namespace } // namespace webrtc diff --git a/pc/slow_peer_connection_integration_test.cc b/pc/slow_peer_connection_integration_test.cc index d306989bc9..8666edc077 100644 --- a/pc/slow_peer_connection_integration_test.cc +++ b/pc/slow_peer_connection_integration_test.cc @@ -471,7 +471,7 @@ TEST_P(PeerConnectionIntegrationTest, CallTransferredForCallee) { // receiving client. These SRTP packets will be dropped. std::unique_ptr original_peer( SetCallerPcWrapperAndReturnCurrent( - CreatePeerConnectionWrapperWithAlternateKey().release())); + CreatePeerConnectionWrapperWithAlternateKey())); // TODO(deadbeef): Why do we call Close here? That goes against the comment // directly above. original_peer->pc()->Close(); @@ -504,7 +504,7 @@ TEST_P(PeerConnectionIntegrationTest, CallTransferredForCaller) { // receiving client. These SRTP packets will be dropped. std::unique_ptr original_peer( SetCalleePcWrapperAndReturnCurrent( - CreatePeerConnectionWrapperWithAlternateKey().release())); + CreatePeerConnectionWrapperWithAlternateKey())); // TODO(deadbeef): Why do we call Close here? That goes against the comment // directly above. original_peer->pc()->Close(); diff --git a/pc/test/integration_test_helpers.h b/pc/test/integration_test_helpers.h index 63ba2108b4..8eda1f5783 100644 --- a/pc/test/integration_test_helpers.h +++ b/pc/test/integration_test_helpers.h @@ -33,6 +33,7 @@ #include "api/candidate.h" #include "api/crypto/crypto_options.h" #include "api/data_channel_interface.h" +#include "api/dtls_transport_interface.h" #include "api/field_trials.h" #include "api/field_trials_view.h" #include "api/ice_transport_interface.h" @@ -739,18 +740,16 @@ class PeerConnectionIntegrationWrapper : public PeerConnectionObserver, } std::optional tls_version() { - return network_thread_->BlockingCall([&] { - return pc() - ->GetSctpTransport() - ->dtls_transport() - ->Information() - .tls_version(); - }); + return dtls_transport_information().tls_version(); } std::optional dtls_transport_role() { + return dtls_transport_information().role(); + } + + DtlsTransportInformation dtls_transport_information() { return network_thread_->BlockingCall([&] { - return pc()->GetSctpTransport()->dtls_transport()->Information().role(); + return pc()->GetSctpTransport()->dtls_transport()->Information(); }); } @@ -1385,12 +1384,12 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test { if (caller_) { caller_->set_signaling_message_receiver(nullptr); caller_->pc()->Close(); - delete SetCallerPcWrapperAndReturnCurrent(nullptr); + caller_.reset(); } if (callee_) { callee_->set_signaling_message_receiver(nullptr); callee_->pc()->Close(); - delete SetCalleePcWrapperAndReturnCurrent(nullptr); + callee_.reset(); } // If turn servers were created for the test they need to be destroyed on @@ -1405,6 +1404,12 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test { return caller_->SignalingStateStable() && callee_->SignalingStateStable(); } + bool PeerConnectionStateIs( + PeerConnectionInterface::PeerConnectionState state) { + return caller_->pc()->peer_connection_state() == state && + callee_->pc()->peer_connection_state() == state; + } + bool DtlsConnected() { // TODO(deadbeef): kIceConnectionConnected currently means both ICE and DTLS // are connected. This is an important distinction. Once we have separate @@ -1720,9 +1725,9 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test { // Set the `caller_` to the `wrapper` passed in and return the // original `caller_`. PeerConnectionIntegrationWrapper* SetCallerPcWrapperAndReturnCurrent( - PeerConnectionIntegrationWrapper* wrapper) { + std::unique_ptr wrapper) { PeerConnectionIntegrationWrapper* old = caller_.release(); - caller_.reset(wrapper); + caller_ = std::move(wrapper); return old; } @@ -1731,9 +1736,9 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test { // Set the `callee_` to the `wrapper` passed in and return the // original `callee_`. PeerConnectionIntegrationWrapper* SetCalleePcWrapperAndReturnCurrent( - PeerConnectionIntegrationWrapper* wrapper) { + std::unique_ptr wrapper) { PeerConnectionIntegrationWrapper* old = callee_.release(); - callee_.reset(wrapper); + callee_ = std::move(wrapper); return old; } From ca21f9e2de5a24908d4e7c5d0f2637f692ea0b27 Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Tue, 20 May 2025 11:02:47 +0200 Subject: [PATCH 240/292] Delete deprecated function to create android AudioDeviceModule Bug: webrtc:413413572 Change-Id: I6bd930f5d3e078fa5d528a5a21bf35c4a17e96dc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392580 Commit-Queue: Danil Chapovalov Reviewed-by: Henrik Andreassson Cr-Commit-Position: refs/heads/main@{#44712} --- sdk/android/BUILD.gn | 22 ------------ .../audio_device_android.cc | 36 ------------------- .../audio_device_android.h | 31 ---------------- .../jni/audio_device/audio_device_module.cc | 15 -------- .../jni/audio_device/audio_device_module.h | 9 ----- 5 files changed, 113 deletions(-) diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn index 06d81a9719..a2ae393859 100644 --- a/sdk/android/BUILD.gn +++ b/sdk/android/BUILD.gn @@ -867,10 +867,6 @@ if (current_os == "linux" || is_android) { visibility = [ "*" ] sources = [ "src/jni/audio_device/java_audio_device_module.cc" ] - # TODO: bugs.webrtc.org/413413572 - Remove poison when - # 'audio_device_module_base' is no longer poisonous. - allow_poison = [ "environment_construction" ] - deps = [ ":audio_device_module_base", ":generated_java_audio_jni", @@ -1035,10 +1031,6 @@ if (current_os == "linux" || is_android) { rtc_library("native_api_audio_device_module") { visibility = [ "*" ] - # TODO: bugs.webrtc.org/413413572 - Remove poison when Environment is - # propagated - allow_poison = [ "environment_construction" ] - sources = [ "native_api/audio_device_module/audio_device_android.cc", "native_api/audio_device_module/audio_device_android.h", @@ -1053,7 +1045,6 @@ if (current_os == "linux" || is_android) { "../../api:scoped_refptr", "../../api/audio:audio_device", "../../api/environment", - "../../api/environment:environment_factory", "../../rtc_base:checks", "../../rtc_base:logging", "../../rtc_base:refcount", @@ -1230,10 +1221,6 @@ if (current_os == "linux" || is_android) { rtc_library("audio_device_module_base") { visibility = [ "*" ] - # TODO: bugs.webrtc.org/413413572 - Remove poison when Environment is - # propagated - allow_poison = [ "environment_construction" ] - sources = [ "src/jni/audio_device/audio_common.h", "src/jni/audio_device/audio_device_module.cc", @@ -1249,7 +1236,6 @@ if (current_os == "linux" || is_android) { "../../api:sequence_checker", "../../api/audio:audio_device", "../../api/environment", - "../../api/environment:environment_factory", "../../api/task_queue", "../../modules/audio_device:audio_device_buffer", "../../rtc_base:checks", @@ -1261,10 +1247,6 @@ if (current_os == "linux" || is_android) { rtc_library("java_audio_device_module") { visibility = [ "*" ] - # TODO: bugs.webrtc.org/413413572 - Remove poison when - # 'audio_device_module_base' is no longer poisonous. - allow_poison = [ "environment_construction" ] - sources = [ "src/jni/audio_device/audio_record_jni.cc", "src/jni/audio_device/audio_record_jni.h", @@ -1325,10 +1307,6 @@ if (current_os == "linux" || is_android) { rtc_library("opensles_audio_device_module") { visibility = [ "*" ] - # TODO: bugs.webrtc.org/413413572 - Remove poison when - # 'audio_device_module_base' is no longer poisonous. - allow_poison = [ "environment_construction" ] - sources = [ "src/jni/audio_device/opensles_common.cc", "src/jni/audio_device/opensles_common.h", diff --git a/sdk/android/native_api/audio_device_module/audio_device_android.cc b/sdk/android/native_api/audio_device_module/audio_device_android.cc index 200c37aa8b..06a66df61c 100644 --- a/sdk/android/native_api/audio_device_module/audio_device_android.cc +++ b/sdk/android/native_api/audio_device_module/audio_device_android.cc @@ -16,7 +16,6 @@ #include #include "api/environment/environment.h" -#include "api/environment/environment_factory.h" #include "api/scoped_refptr.h" #include "rtc_base/logging.h" #include "rtc_base/ref_count.h" @@ -246,39 +245,4 @@ scoped_refptr CreateAndroidAudioDeviceModule( } } -#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO) -scoped_refptr CreateAAudioAudioDeviceModule( - JNIEnv* env, - jobject application_context) { - return CreateAAudioAudioDeviceModule(env, CreateEnvironment(), - application_context); -} -#endif - -scoped_refptr CreateJavaAudioDeviceModule( - JNIEnv* env, - jobject application_context) { - return CreateJavaAudioDeviceModule(env, CreateEnvironment(), - application_context); -} - -scoped_refptr CreateOpenSLESAudioDeviceModule( - JNIEnv* env, - jobject application_context) { - return CreateOpenSLESAudioDeviceModule(env, CreateEnvironment(), - application_context); -} - -scoped_refptr -CreateJavaInputAndOpenSLESOutputAudioDeviceModule(JNIEnv* env, - jobject application_context) { - return CreateJavaInputAndOpenSLESOutputAudioDeviceModule( - env, CreateEnvironment(), application_context); -} - -scoped_refptr CreateAndroidAudioDeviceModule( - AudioDeviceModule::AudioLayer audio_layer) { - return CreateAndroidAudioDeviceModule(CreateEnvironment(), audio_layer); -} - } // namespace webrtc diff --git a/sdk/android/native_api/audio_device_module/audio_device_android.h b/sdk/android/native_api/audio_device_module/audio_device_android.h index 88ecbb0d7e..5cf4e51014 100644 --- a/sdk/android/native_api/audio_device_module/audio_device_android.h +++ b/sdk/android/native_api/audio_device_module/audio_device_android.h @@ -28,37 +28,6 @@ scoped_refptr CreateAndroidAudioDeviceModule( const Environment& webrtc_env, AudioDeviceModule::AudioLayer audio_layer); -#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO) -[[deprecated("bugs.webrtc.org/413413572")]] -scoped_refptr CreateAAudioAudioDeviceModule( - JNIEnv* env, - jobject application_context); -#endif - -[[deprecated("bugs.webrtc.org/413413572")]] -scoped_refptr CreateJavaAudioDeviceModule( - JNIEnv* env, - jobject application_context); - -[[deprecated("bugs.webrtc.org/413413572")]] -scoped_refptr CreateOpenSLESAudioDeviceModule( - JNIEnv* env, - jobject application_context); - -[[deprecated("bugs.webrtc.org/413413572")]] -scoped_refptr -CreateJavaInputAndOpenSLESOutputAudioDeviceModule(JNIEnv* env, - jobject application_context); - -[[deprecated("bugs.webrtc.org/413413572")]] -scoped_refptr -CreateJavaInputAndAAudioOutputAudioDeviceModule(JNIEnv* env, - jobject application_context); - -[[deprecated("bugs.webrtc.org/413413572")]] -scoped_refptr CreateAndroidAudioDeviceModule( - AudioDeviceModule::AudioLayer audio_layer); - } // namespace webrtc #endif // SDK_ANDROID_NATIVE_API_AUDIO_DEVICE_MODULE_AUDIO_DEVICE_ANDROID_H_ diff --git a/sdk/android/src/jni/audio_device/audio_device_module.cc b/sdk/android/src/jni/audio_device/audio_device_module.cc index 732e8d4607..4331db2112 100644 --- a/sdk/android/src/jni/audio_device/audio_device_module.cc +++ b/sdk/android/src/jni/audio_device/audio_device_module.cc @@ -14,7 +14,6 @@ #include #include "api/environment/environment.h" -#include "api/environment/environment_factory.h" #include "api/make_ref_counted.h" #include "api/sequence_checker.h" #include "modules/audio_device/audio_device_buffer.h" @@ -648,20 +647,6 @@ bool IsLowLatencyOutputSupported(JNIEnv* env, return Java_WebRtcAudioManager_isLowLatencyOutputSupported(env, j_context); } -scoped_refptr CreateAudioDeviceModuleFromInputAndOutput( - AudioDeviceModule::AudioLayer audio_layer, - bool is_stereo_playout_supported, - bool is_stereo_record_supported, - uint16_t playout_delay_ms, - std::unique_ptr audio_input, - std::unique_ptr audio_output) { - RTC_DLOG(LS_INFO) << __FUNCTION__; - return make_ref_counted( - CreateEnvironment(), audio_layer, is_stereo_playout_supported, - is_stereo_record_supported, playout_delay_ms, std::move(audio_input), - std::move(audio_output)); -} - scoped_refptr CreateAudioDeviceModuleFromInputAndOutput( const Environment& env, AudioDeviceModule::AudioLayer audio_layer, diff --git a/sdk/android/src/jni/audio_device/audio_device_module.h b/sdk/android/src/jni/audio_device/audio_device_module.h index f60432543e..dc7c73d587 100644 --- a/sdk/android/src/jni/audio_device/audio_device_module.h +++ b/sdk/android/src/jni/audio_device/audio_device_module.h @@ -97,15 +97,6 @@ bool IsLowLatencyOutputSupported(JNIEnv* env, const JavaRef& j_context); // Glue together an audio input and audio output to get an AudioDeviceModule. -[[deprecated("bugs.webrtc.org/413413572")]] -scoped_refptr CreateAudioDeviceModuleFromInputAndOutput( - AudioDeviceModule::AudioLayer audio_layer, - bool is_stereo_playout_supported, - bool is_stereo_record_supported, - uint16_t playout_delay_ms, - std::unique_ptr audio_input, - std::unique_ptr audio_output); - scoped_refptr CreateAudioDeviceModuleFromInputAndOutput( const Environment& env, AudioDeviceModule::AudioLayer audio_layer, From d36191dda9614f1670217aa54b75427ddd37c28d Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Wed, 21 May 2025 13:20:03 +0200 Subject: [PATCH 241/292] Avoid global field trial string in audio processing fuzzer Bug: webrtc:42225026 Change-Id: I5f2076720f084144bd01edf1526589e37e0278bf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392960 Commit-Queue: Danil Chapovalov Reviewed-by: Sam Zackrisson Cr-Commit-Position: refs/heads/main@{#44713} --- test/fuzzers/BUILD.gn | 12 ++-------- .../audio_processing_configs_fuzzer.cc | 23 +++++++++---------- 2 files changed, 13 insertions(+), 22 deletions(-) diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn index c57a1409a3..acf6db2a6a 100644 --- a/test/fuzzers/BUILD.gn +++ b/test/fuzzers/BUILD.gn @@ -553,6 +553,7 @@ webrtc_fuzzer_test("audio_processing_fuzzer") { deps = [ ":audio_processing_fuzzer_helper", "../../api:array_view", + "../../api:field_trials", "../../api:scoped_refptr", "../../api/audio:aec3_factory", "../../api/audio:audio_processing", @@ -562,19 +563,10 @@ webrtc_fuzzer_test("audio_processing_fuzzer") { "../../api/environment", "../../api/environment:environment_factory", "../../api/task_queue", - "../../api/task_queue:default_task_queue_factory", - "../../modules/audio_processing", - "../../modules/audio_processing:audio_buffer", - "../../modules/audio_processing:audioproc_test_utils", - "../../modules/audio_processing/aec3", "../../modules/audio_processing/aec_dump", - "../../modules/audio_processing/aec_dump:aec_dump_impl", "../../rtc_base:checks", - "../../rtc_base:macromagic", - "../../rtc_base:safe_minmax", - "../../system_wrappers:field_trial", "//third_party/abseil-cpp/absl/base:nullability", - "//third_party/abseil-cpp/absl/memory", + "//third_party/abseil-cpp/absl/strings:string_view", ] seed_corpus = "corpora/audio_processing-corpus" } diff --git a/test/fuzzers/audio_processing_configs_fuzzer.cc b/test/fuzzers/audio_processing_configs_fuzzer.cc index 80a5f626dc..04a50e21ae 100644 --- a/test/fuzzers/audio_processing_configs_fuzzer.cc +++ b/test/fuzzers/audio_processing_configs_fuzzer.cc @@ -16,6 +16,7 @@ #include #include "absl/base/nullability.h" +#include "absl/strings/string_view.h" #include "api/array_view.h" #include "api/audio/audio_processing.h" #include "api/audio/builtin_audio_processing_builder.h" @@ -24,20 +25,19 @@ #include "api/audio/echo_detector_creator.h" #include "api/environment/environment.h" #include "api/environment/environment_factory.h" +#include "api/field_trials.h" #include "api/scoped_refptr.h" #include "api/task_queue/task_queue_base.h" #include "api/task_queue/task_queue_factory.h" #include "modules/audio_processing/aec_dump/aec_dump_factory.h" -#include "rtc_base/arraysize.h" #include "rtc_base/checks.h" -#include "system_wrappers/include/field_trial.h" #include "test/fuzzers/audio_processing_fuzzer_helper.h" #include "test/fuzzers/fuzz_data_helper.h" namespace webrtc { namespace { -const std::string kFieldTrialNames[] = { +constexpr absl::string_view kFieldTrialNames[] = { "WebRTC-Aec3MinErleDuringOnsetsKillSwitch", "WebRTC-Aec3ShortHeadroomKillSwitch", }; @@ -49,7 +49,6 @@ const Environment& GetEnvironment() { webrtc::scoped_refptr CreateApm( test::FuzzDataHelper* fuzz_data, - std::string* field_trial_string, TaskQueueBase* absl_nonnull worker_queue) { // Parse boolean values for optionally enabling different // configurable public components of APM. @@ -69,19 +68,22 @@ webrtc::scoped_refptr CreateApm( const float gain_controller2_gain_db = fuzz_data->ReadOrDefaultValue(0) % 50; - constexpr size_t kNumFieldTrials = arraysize(kFieldTrialNames); + constexpr size_t kNumFieldTrials = std::size(kFieldTrialNames); // Verify that the read data type has enough bits to fuzz the field trials. using FieldTrialBitmaskType = uint64_t; static_assert(kNumFieldTrials <= sizeof(FieldTrialBitmaskType) * 8, "FieldTrialBitmaskType is not large enough."); std::bitset field_trial_bitmask( fuzz_data->ReadOrDefaultValue(0)); + auto field_trials = std::make_unique(""); for (size_t i = 0; i < kNumFieldTrials; ++i) { if (field_trial_bitmask[i]) { - *field_trial_string += kFieldTrialNames[i] + "/Enabled/"; + field_trials->Set(kFieldTrialNames[i], "Enabled"); } } - field_trial::InitFieldTrialsFromString(field_trial_string->c_str()); + EnvironmentFactory env_factory(GetEnvironment()); + env_factory.Set(std::move(field_trials)); + const Environment env = env_factory.Create(); // Ignore a few bytes. Bytes from this segment will be used for // future config flag changes. We assume 40 bytes is enough for @@ -123,7 +125,7 @@ webrtc::scoped_refptr CreateApm( .SetEchoControlFactory(std::move(echo_control_factory)) .SetEchoDetector(use_red ? CreateEchoDetector() : nullptr) .SetConfig(apm_config) - .Build(GetEnvironment()); + .Build(env); #ifdef WEBRTC_LINUX apm->AttachAecDump(AecDumpFactory::Create("/dev/null", -1, worker_queue)); @@ -139,14 +141,11 @@ void FuzzOneInput(const uint8_t* data, size_t size) { return; } test::FuzzDataHelper fuzz_data(webrtc::ArrayView(data, size)); - // This string must be in scope during execution, according to documentation - // for field_trial.h. Hence it's created here and not in CreateApm. - std::string field_trial_string = ""; std::unique_ptr worker_queue = GetEnvironment().task_queue_factory().CreateTaskQueue( "rtc-low-prio", TaskQueueFactory::Priority::LOW); - auto apm = CreateApm(&fuzz_data, &field_trial_string, worker_queue.get()); + auto apm = CreateApm(&fuzz_data, worker_queue.get()); if (apm) { FuzzAudioProcessing(&fuzz_data, std::move(apm)); From 328241aeb1ada060a4ddc4d14bca42f792f39557 Mon Sep 17 00:00:00 2001 From: Harald Alvestrand Date: Wed, 21 May 2025 10:14:36 +0000 Subject: [PATCH 242/292] Expand pc/webrtc_sdp.cc OWNERS Ensure that someone (who knows about standards) is available even if hta and hbos are out. No-Try: true Bug: None Change-Id: I10717d013f69d65543605ec9dfbcad9ba916c0c8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392901 Reviewed-by: Tomas Gunnarsson Commit-Queue: Tomas Gunnarsson Cr-Commit-Position: refs/heads/main@{#44714} --- pc/OWNERS | 3 +++ 1 file changed, 3 insertions(+) diff --git a/pc/OWNERS b/pc/OWNERS index 4b662f76a0..8f054a6427 100644 --- a/pc/OWNERS +++ b/pc/OWNERS @@ -8,3 +8,6 @@ deadbeef@webrtc.org per-file webrtc_sdp.cc = set noparent per-file webrtc_sdp.cc = hta@webrtc.org per-file webrtc_sdp.cc = hbos@webrtc.org +# If none of the above are present, may also try one of these +per-file webrtc_sdp.cc = tommi@webrtc.org +per-file webrtc_sdp.cc = guidou@webrtc.org From 4ecf1117a8189a772e68d98e74062780e065e6a3 Mon Sep 17 00:00:00 2001 From: Fredrik Solenberg Date: Wed, 21 May 2025 11:05:56 +0200 Subject: [PATCH 243/292] Remove system_wrappers/../sleep.* There is an equivalent function (SleepMs) in rtc_base/thread.h Bug: webrtc:42228262 Change-Id: I19bf136b97518fb62064b31ef1058041655ea59c No-Iwyu: Avoid unrelated changes in CL. Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392681 Commit-Queue: Tomas Gunnarsson Auto-Submit: Fredrik Solenberg Reviewed-by: Tomas Gunnarsson Cr-Commit-Position: refs/heads/main@{#44715} --- audio/test/audio_end_to_end_test.cc | 1 - audio/test/audio_stats_test.cc | 7 +- audio/test/nack_test.cc | 4 +- modules/audio_coding/BUILD.gn | 1 + .../acm2/audio_coding_module_unittest.cc | 8 +- modules/audio_device/BUILD.gn | 4 +- .../audio_device/dummy/file_audio_device.cc | 6 +- .../linux/audio_device_alsa_linux.cc | 6 +- .../audio_device/win/audio_device_core_win.cc | 4 +- .../audio_processing_impl_locking_unittest.cc | 4 +- modules/desktop_capture/BUILD.gn | 11 ++- .../desktop_capture/screen_drawer_linux.cc | 6 +- .../desktop_capture/screen_drawer_unittest.cc | 12 +-- modules/desktop_capture/screen_drawer_win.cc | 6 +- .../win/dxgi_duplicator_controller.cc | 4 +- .../win/wgc_capture_session.cc | 4 +- .../win/wgc_capturer_win_unittest.cc | 3 +- modules/video_coding/BUILD.gn | 2 +- .../test/videocodec_test_fixture_impl.cc | 6 +- rtc_base/BUILD.gn | 2 +- rtc_base/cpu_time_unittest.cc | 4 +- rtc_base/platform_thread_unittest.cc | 1 - rtc_base/task_queue_stdlib_unittest.cc | 4 +- rtc_tools/video_replay.cc | 3 +- sdk/android/BUILD.gn | 1 - .../stacktrace/stacktrace_unittest.cc | 4 +- system_wrappers/BUILD.gn | 2 - system_wrappers/include/sleep.h | 24 ------ system_wrappers/source/sleep.cc | 36 -------- test/BUILD.gn | 3 +- test/fake_encoder.cc | 4 +- test/pc/e2e/analyzer/video/BUILD.gn | 3 +- ...ideo_quality_analyzer_metric_names_test.cc | 4 +- .../default_video_quality_analyzer_test.cc | 84 +++++++++---------- test/video_codec_tester.cc | 4 +- video/end_to_end_tests/bandwidth_tests.cc | 6 +- .../end_to_end_tests/call_operation_tests.cc | 4 +- .../corruption_detection_tests.cc | 4 +- video/end_to_end_tests/network_state_tests.cc | 6 +- video/end_to_end_tests/stats_tests.cc | 6 +- video/frame_cadence_adapter_unittest.cc | 6 +- video/video_send_stream_tests.cc | 3 +- 42 files changed, 126 insertions(+), 191 deletions(-) delete mode 100644 system_wrappers/include/sleep.h delete mode 100644 system_wrappers/source/sleep.cc diff --git a/audio/test/audio_end_to_end_test.cc b/audio/test/audio_end_to_end_test.cc index d9c4594935..95bc86dcca 100644 --- a/audio/test/audio_end_to_end_test.cc +++ b/audio/test/audio_end_to_end_test.cc @@ -16,7 +16,6 @@ #include "api/task_queue/task_queue_base.h" #include "call/fake_network_pipe.h" #include "modules/audio_device/include/test_audio_device.h" -#include "system_wrappers/include/sleep.h" #include "test/gtest.h" #include "test/video_test_constants.h" diff --git a/audio/test/audio_stats_test.cc b/audio/test/audio_stats_test.cc index e8521cfe99..d8f1d406a5 100644 --- a/audio/test/audio_stats_test.cc +++ b/audio/test/audio_stats_test.cc @@ -10,7 +10,7 @@ #include "audio/test/audio_end_to_end_test.h" #include "rtc_base/numerics/safe_compare.h" -#include "system_wrappers/include/sleep.h" +#include "rtc_base/thread.h" #include "test/gtest.h" namespace webrtc { @@ -42,10 +42,11 @@ class NoLossTest : public AudioEndToEndTest { } void PerformTest() override { - SleepMs(kTestDurationMs); + Thread::SleepMs(kTestDurationMs); send_audio_device()->StopRecording(); // and some extra time to account for network delay. - SleepMs(GetSendTransportConfig().queue_delay_ms + kExtraRecordTimeMs); + Thread::SleepMs(GetSendTransportConfig().queue_delay_ms + + kExtraRecordTimeMs); } void OnStreamsStopped() override { diff --git a/audio/test/nack_test.cc b/audio/test/nack_test.cc index b36adf8991..05cd11b2ce 100644 --- a/audio/test/nack_test.cc +++ b/audio/test/nack_test.cc @@ -9,7 +9,7 @@ */ #include "audio/test/audio_end_to_end_test.h" -#include "system_wrappers/include/sleep.h" +#include "rtc_base/thread.h" #include "test/gtest.h" namespace webrtc { @@ -40,7 +40,7 @@ TEST_F(NackTest, ShouldNackInLossyNetwork) { AudioEndToEndTest::ModifyAudioConfigs(send_config, receive_configs); } - void PerformTest() override { SleepMs(kTestDurationMs); } + void PerformTest() override { Thread::SleepMs(kTestDurationMs); } void OnStreamsStopped() override { AudioReceiveStreamInterface::Stats recv_stats = diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn index dd4500f744..46c659c928 100644 --- a/modules/audio_coding/BUILD.gn +++ b/modules/audio_coding/BUILD.gn @@ -1560,6 +1560,7 @@ if (rtc_include_tests) { "../../rtc_base:safe_conversions", "../../rtc_base:sanitizer", "../../rtc_base:stringutils", + "../../rtc_base:threading", "../../rtc_base:timeutils", "../../rtc_base/synchronization:mutex", "../../rtc_base/system:arch", diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc index 9d5916931b..ba8ca870d1 100644 --- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc +++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc @@ -63,9 +63,9 @@ #include "rtc_base/string_encode.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/system/arch.h" +#include "rtc_base/thread.h" #include "rtc_base/thread_annotations.h" #include "system_wrappers/include/clock.h" -#include "system_wrappers/include/sleep.h" #include "test/audio_decoder_proxy_factory.h" #include "test/gmock.h" #include "test/gtest.h" @@ -441,7 +441,7 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi { // The send thread doesn't have to care about the current simulated time, // since only the AcmReceiver is using the clock. void CbSendImpl() { - SleepMs(1); + Thread::SleepMs(1); if (HasFatalFailure()) { // End the test early if a fatal failure (ASSERT_*) has occurred. test_complete_.Set(); @@ -454,7 +454,7 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi { } void CbInsertPacketImpl() { - SleepMs(1); + Thread::SleepMs(1); { MutexLock lock(&mutex_); if (env_.clock().TimeInMilliseconds() < next_insert_packet_time_ms_) { @@ -468,7 +468,7 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi { } void CbPullAudioImpl() { - SleepMs(1); + Thread::SleepMs(1); { MutexLock lock(&mutex_); // Don't let the insert thread fall behind. diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn index 1bae2579b7..9bf81c315f 100644 --- a/modules/audio_device/BUILD.gn +++ b/modules/audio_device/BUILD.gn @@ -246,10 +246,10 @@ if (!build_with_chromium) { "../../rtc_base:logging", "../../rtc_base:platform_thread", "../../rtc_base:stringutils", + "../../rtc_base:threading", "../../rtc_base:timeutils", "../../rtc_base/synchronization:mutex", "../../rtc_base/system:file_wrapper", - "../../system_wrappers", "//third_party/abseil-cpp/absl/strings:string_view", ] } @@ -285,12 +285,12 @@ rtc_library("audio_device_impl") { "../../rtc_base:rtc_event", "../../rtc_base:safe_conversions", "../../rtc_base:stringutils", + "../../rtc_base:threading", "../../rtc_base:timeutils", "../../rtc_base/synchronization:mutex", "../../rtc_base/system:arch", "../../rtc_base/system:file_wrapper", "../../rtc_base/task_utils:repeating_task", - "../../system_wrappers", "../../system_wrappers:metrics", "../utility", "//third_party/abseil-cpp/absl/base:core_headers", diff --git a/modules/audio_device/dummy/file_audio_device.cc b/modules/audio_device/dummy/file_audio_device.cc index 7c31d6bdd3..e33036b315 100644 --- a/modules/audio_device/dummy/file_audio_device.cc +++ b/modules/audio_device/dummy/file_audio_device.cc @@ -16,8 +16,8 @@ #include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/platform_thread.h" +#include "rtc_base/thread.h" #include "rtc_base/time_utils.h" -#include "system_wrappers/include/sleep.h" namespace webrtc { @@ -466,7 +466,7 @@ bool FileAudioDevice::PlayThreadProcess() { int64_t deltaTimeMillis = TimeMillis() - currentTime; if (deltaTimeMillis < 10) { - SleepMs(10 - deltaTimeMillis); + Thread::SleepMs(10 - deltaTimeMillis); } return true; @@ -499,7 +499,7 @@ bool FileAudioDevice::RecThreadProcess() { int64_t deltaTimeMillis = TimeMillis() - currentTime; if (deltaTimeMillis < 10) { - SleepMs(10 - deltaTimeMillis); + Thread::SleepMs(10 - deltaTimeMillis); } return true; diff --git a/modules/audio_device/linux/audio_device_alsa_linux.cc b/modules/audio_device/linux/audio_device_alsa_linux.cc index e4895392cf..5667df1aa0 100644 --- a/modules/audio_device/linux/audio_device_alsa_linux.cc +++ b/modules/audio_device/linux/audio_device_alsa_linux.cc @@ -13,7 +13,7 @@ #include "modules/audio_device/audio_device_config.h" #include "rtc_base/logging.h" #include "rtc_base/system/arch.h" -#include "system_wrappers/include/sleep.h" +#include "rtc_base/thread.h" WebRTCAlsaSymbolTable* GetAlsaSymbolTable() { static WebRTCAlsaSymbolTable* alsa_symbol_table = new WebRTCAlsaSymbolTable(); @@ -782,7 +782,7 @@ int32_t AudioDeviceLinuxALSA::InitPlayoutLocked() { if (errVal == -EBUSY) // Device busy - try some more! { for (int i = 0; i < 5; i++) { - SleepMs(1000); + Thread::SleepMs(1000); errVal = LATE(snd_pcm_open)(&_handlePlayout, deviceName, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); if (errVal == 0) { @@ -906,7 +906,7 @@ int32_t AudioDeviceLinuxALSA::InitRecordingLocked() { if (errVal == -EBUSY) // Device busy - try some more! { for (int i = 0; i < 5; i++) { - SleepMs(1000); + Thread::SleepMs(1000); errVal = LATE(snd_pcm_open)(&_handleRecord, deviceName, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK); if (errVal == 0) { diff --git a/modules/audio_device/win/audio_device_core_win.cc b/modules/audio_device/win/audio_device_core_win.cc index 015a0a394f..de17a0f647 100644 --- a/modules/audio_device/win/audio_device_core_win.cc +++ b/modules/audio_device/win/audio_device_core_win.cc @@ -45,8 +45,8 @@ #include "rtc_base/logging.h" #include "rtc_base/platform_thread.h" #include "rtc_base/string_utils.h" +#include "rtc_base/thread.h" #include "rtc_base/thread_annotations.h" -#include "system_wrappers/include/sleep.h" // Macro that calls a COM method returning HRESULT value. #define EXIT_ON_ERROR(hres) \ @@ -2878,7 +2878,7 @@ DWORD AudioDeviceWindowsCore::DoRenderThread() { // ------------------ THREAD LOOP ------------------ << - SleepMs(static_cast(endpointBufferSizeMS + 0.5)); + Thread::SleepMs(static_cast(endpointBufferSizeMS + 0.5)); hr = _ptrClientOut->Stop(); Exit: diff --git a/modules/audio_processing/audio_processing_impl_locking_unittest.cc b/modules/audio_processing/audio_processing_impl_locking_unittest.cc index dc1bf1287b..d9d75dccd6 100644 --- a/modules/audio_processing/audio_processing_impl_locking_unittest.cc +++ b/modules/audio_processing/audio_processing_impl_locking_unittest.cc @@ -21,7 +21,7 @@ #include "rtc_base/platform_thread.h" #include "rtc_base/random.h" #include "rtc_base/synchronization/mutex.h" -#include "system_wrappers/include/sleep.h" +#include "rtc_base/thread.h" #include "test/gtest.h" namespace webrtc { @@ -441,7 +441,7 @@ class AudioProcessingImplLockTest // Sleeps a random time between 0 and max_sleep milliseconds. void SleepRandomMs(int max_sleep, RandomGenerator* rand_gen) { int sleeptime = rand_gen->RandInt(0, max_sleep); - SleepMs(sleeptime); + Thread::SleepMs(sleeptime); } // Populates a float audio frame with random data. diff --git a/modules/desktop_capture/BUILD.gn b/modules/desktop_capture/BUILD.gn index 4b004c6527..f00de0c3cb 100644 --- a/modules/desktop_capture/BUILD.gn +++ b/modules/desktop_capture/BUILD.gn @@ -66,7 +66,7 @@ if (rtc_include_tests) { ":screen_drawer", "../../api:array_view", "../../rtc_base:base64", - "../../system_wrappers", + "../../rtc_base:threading", "../../test:test_support", ] sources += [ @@ -158,12 +158,9 @@ if (rtc_include_tests) { "../../rtc_base:logging", "../../rtc_base:macromagic", "../../rtc_base:random", - "../../rtc_base:timeutils", - - # TODO(bugs.webrtc.org/9987): Remove this dep on rtc_base:rtc_base once - # rtc_base:threading is fully defined. "../../rtc_base:task_queue_for_test", "../../rtc_base:threading", + "../../rtc_base:timeutils", "../../system_wrappers", "../../test:test_support", ] @@ -235,7 +232,7 @@ if (rtc_include_tests) { "../../api:scoped_refptr", "../../rtc_base:checks", "../../rtc_base:logging", - "../../system_wrappers", + "../../rtc_base:threading", "//third_party/abseil-cpp/absl/strings:string_view", ] @@ -527,6 +524,7 @@ rtc_library("desktop_capture") { "shcore.lib", ] deps += [ + "../../rtc_base:threading", "../../rtc_base:win32", "../../rtc_base/containers:flat_map", "../../rtc_base/win:create_direct3d_device", @@ -595,6 +593,7 @@ rtc_library("desktop_capture") { libs += [ "dwmapi.lib" ] deps += [ "../../rtc_base:rtc_event", + "../../rtc_base:threading", "../../rtc_base/win:hstring", ] } diff --git a/modules/desktop_capture/screen_drawer_linux.cc b/modules/desktop_capture/screen_drawer_linux.cc index 55b9570d9c..5a04d37cde 100644 --- a/modules/desktop_capture/screen_drawer_linux.cc +++ b/modules/desktop_capture/screen_drawer_linux.cc @@ -22,7 +22,7 @@ #include "modules/desktop_capture/screen_drawer.h" #include "modules/desktop_capture/screen_drawer_lock_posix.h" #include "rtc_base/checks.h" -#include "system_wrappers/include/sleep.h" +#include "rtc_base/thread.h" namespace webrtc { @@ -97,7 +97,7 @@ ScreenDrawerLinux::ScreenDrawerLinux() { colormap_ = DefaultColormap(display_->display(), screen_num_); BringToFront(); // Wait for window animations. - SleepMs(200); + Thread::SleepMs(200); } ScreenDrawerLinux::~ScreenDrawerLinux() { @@ -134,7 +134,7 @@ void ScreenDrawerLinux::Clear() { // TODO(zijiehe): Find the right signal from X11 to indicate the finish of all // pending paintings. void ScreenDrawerLinux::WaitForPendingDraws() { - SleepMs(50); + Thread::SleepMs(50); } bool ScreenDrawerLinux::MayDrawIncompleteShapes() { diff --git a/modules/desktop_capture/screen_drawer_unittest.cc b/modules/desktop_capture/screen_drawer_unittest.cc index f674355eb0..b832b1752c 100644 --- a/modules/desktop_capture/screen_drawer_unittest.cc +++ b/modules/desktop_capture/screen_drawer_unittest.cc @@ -20,8 +20,8 @@ #include "rtc_base/logging.h" #include "rtc_base/platform_thread.h" #include "rtc_base/random.h" +#include "rtc_base/thread.h" #include "rtc_base/time_utils.h" -#include "system_wrappers/include/sleep.h" #include "test/gtest.h" #if defined(WEBRTC_POSIX) @@ -54,7 +54,7 @@ void TestScreenDrawerLock( created_->store(true); // Wait for the main thread to get the signal of created_. while (!ready_.load()) { - SleepMs(1); + Thread::SleepMs(1); } // At this point, main thread should begin to create a second lock. Though // it's still possible the second lock won't be created before the @@ -66,7 +66,7 @@ void TestScreenDrawerLock( // But we need to ensure at least 100 ms has been passed before unlocking // `lock`. while (TimeMillis() - current_ms < kLockDurationMs) { - SleepMs(kLockDurationMs - (TimeMillis() - current_ms)); + Thread::SleepMs(kLockDurationMs - (TimeMillis() - current_ms)); } } @@ -84,7 +84,7 @@ void TestScreenDrawerLock( // lock. See // https://chromium-review.googlesource.com/c/607688/13/webrtc/modules/desktop_capture/screen_drawer_unittest.cc while (!created.load()) { - SleepMs(1); + Thread::SleepMs(1); } const int64_t start_ms = TimeMillis(); @@ -130,11 +130,11 @@ TEST(ScreenDrawerTest, DISABLED_DrawRectangles) { random.Rand(), random.Rand())); if (i == 50) { - SleepMs(10000); + Thread::SleepMs(10000); } } - SleepMs(10000); + Thread::SleepMs(10000); } #if defined(THREAD_SANITIZER) // bugs.webrtc.org/10019 diff --git a/modules/desktop_capture/screen_drawer_win.cc b/modules/desktop_capture/screen_drawer_win.cc index 7cf634fe89..76c6081a4d 100644 --- a/modules/desktop_capture/screen_drawer_win.cc +++ b/modules/desktop_capture/screen_drawer_win.cc @@ -13,7 +13,7 @@ #include #include "modules/desktop_capture/screen_drawer.h" -#include "system_wrappers/include/sleep.h" +#include "rtc_base/thread.h" namespace webrtc { @@ -40,7 +40,7 @@ ScreenDrawerLockWin::ScreenDrawerLockWin() { if (mutex_) { CloseHandle(mutex_); } - SleepMs(1000); + Thread::SleepMs(1000); } } } @@ -152,7 +152,7 @@ void ScreenDrawerWin::Clear() { // paintings. void ScreenDrawerWin::WaitForPendingDraws() { BringToFront(); - SleepMs(50); + Thread::SleepMs(50); } bool ScreenDrawerWin::MayDrawIncompleteShapes() { diff --git a/modules/desktop_capture/win/dxgi_duplicator_controller.cc b/modules/desktop_capture/win/dxgi_duplicator_controller.cc index bee0ba72f2..24d8ad18f0 100644 --- a/modules/desktop_capture/win/dxgi_duplicator_controller.cc +++ b/modules/desktop_capture/win/dxgi_duplicator_controller.cc @@ -20,8 +20,8 @@ #include "modules/desktop_capture/win/screen_capture_utils.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" +#include "rtc_base/thread.h" #include "rtc_base/time_utils.h" -#include "system_wrappers/include/sleep.h" namespace webrtc { @@ -531,7 +531,7 @@ bool DxgiDuplicatorController::EnsureFrameCaptured(Context* context, // Sleep `ms_per_frame` before attempting to capture the next frame to // ensure the video adapter has time to update the screen. - webrtc::SleepMs(ms_per_frame); + Thread::SleepMs(ms_per_frame); } // When capturing multiple monitors, we need to update the captured region to // prevent flickering by re-setting context. See diff --git a/modules/desktop_capture/win/wgc_capture_session.cc b/modules/desktop_capture/win/wgc_capture_session.cc index 63a6a15229..da9054e0eb 100644 --- a/modules/desktop_capture/win/wgc_capture_session.cc +++ b/modules/desktop_capture/win/wgc_capture_session.cc @@ -24,12 +24,12 @@ #include "modules/desktop_capture/win/wgc_desktop_frame.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" +#include "rtc_base/thread.h" #include "rtc_base/time_utils.h" #include "rtc_base/win/create_direct3d_device.h" #include "rtc_base/win/get_activation_factory.h" #include "rtc_base/win/windows_version.h" #include "system_wrappers/include/metrics.h" -#include "system_wrappers/include/sleep.h" using Microsoft::WRL::ComPtr; namespace WGC = ABI::Windows::Graphics::Capture; @@ -256,7 +256,7 @@ void WgcCaptureSession::EnsureFrame() { int sleep_count = 0; while (!queue_.current_frame() && sleep_count < max_sleep_count) { sleep_count++; - webrtc::SleepMs(sleep_time_ms); + Thread::SleepMs(sleep_time_ms); hr = ProcessFrame(); if (FAILED(hr)) { RTC_DLOG(LS_WARNING) << "ProcessFrame failed during startup: " << hr; diff --git a/modules/desktop_capture/win/wgc_capturer_win_unittest.cc b/modules/desktop_capture/win/wgc_capturer_win_unittest.cc index 5d0884a398..6190579753 100644 --- a/modules/desktop_capture/win/wgc_capturer_win_unittest.cc +++ b/modules/desktop_capture/win/wgc_capturer_win_unittest.cc @@ -28,7 +28,6 @@ #include "rtc_base/win/scoped_com_initializer.h" #include "rtc_base/win/windows_version.h" #include "system_wrappers/include/metrics.h" -#include "system_wrappers/include/sleep.h" #include "test/gtest.h" namespace webrtc { @@ -548,7 +547,7 @@ TEST_F(WgcCapturerWindowTest, CloseWindowMidCapture) { // Unlike GetMessage, PeekMessage will not hang if there are no messages in // the queue. PeekMessage(&msg, 0, 0, 0, PM_REMOVE); - SleepMs(1); + Thread::SleepMs(1); } EXPECT_FALSE(wgc_capturer->IsSourceBeingCaptured(source_id_)); diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn index 6314840119..52c391dc3f 100644 --- a/modules/video_coding/BUILD.gn +++ b/modules/video_coding/BUILD.gn @@ -986,9 +986,9 @@ if (rtc_include_tests) { "../../rtc_base:rtc_base_tests_utils", "../../rtc_base:stringutils", "../../rtc_base:task_queue_for_test", + "../../rtc_base:threading", "../../rtc_base:timeutils", "../../rtc_base/system:file_wrapper", - "../../system_wrappers", "../../test:fileutils", "../../test:test_support", "../../test:video_frame_writer", diff --git a/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc b/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc index 7372508ecb..5732dccc6d 100644 --- a/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc +++ b/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc @@ -72,8 +72,8 @@ #include "rtc_base/system/file_wrapper.h" #include "rtc_base/system_time.h" #include "rtc_base/task_queue_for_test.h" +#include "rtc_base/thread.h" #include "rtc_base/time_utils.h" -#include "system_wrappers/include/sleep.h" #include "test/gtest.h" #include "test/testsupport/file_utils.h" #include "test/testsupport/frame_reader.h" @@ -539,7 +539,7 @@ void VideoCodecTestFixtureImpl::ProcessAllFrames( // Roughly pace the frames. const int frame_duration_ms = std::ceil(kNumMillisecsPerSec / rate_profile->input_fps); - SleepMs(frame_duration_ms); + Thread::SleepMs(frame_duration_ms); } } @@ -550,7 +550,7 @@ void VideoCodecTestFixtureImpl::ProcessAllFrames( // Give the VideoProcessor pipeline some time to process the last frame, // and then release the codecs. - SleepMs(1 * kNumMillisecsPerSec); + Thread::SleepMs(1 * kNumMillisecsPerSec); cpu_process_time_->Stop(); } diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn index bf217ee16a..b019bb01f3 100644 --- a/rtc_base/BUILD.gn +++ b/rtc_base/BUILD.gn @@ -716,10 +716,10 @@ if (rtc_include_tests) { ":logging", ":rtc_event", ":rtc_task_queue_stdlib", + ":threading", "../api/task_queue", "../api/task_queue:task_queue_test", "../api/units:time_delta", - "../system_wrappers", "../test:test_main", "../test:test_support", ] diff --git a/rtc_base/cpu_time_unittest.cc b/rtc_base/cpu_time_unittest.cc index 9d2f4adac4..591ae8001d 100644 --- a/rtc_base/cpu_time_unittest.cc +++ b/rtc_base/cpu_time_unittest.cc @@ -13,8 +13,8 @@ #include #include "rtc_base/platform_thread.h" +#include "rtc_base/thread.h" #include "rtc_base/time_utils.h" -#include "system_wrappers/include/sleep.h" #include "test/gtest.h" // Only run these tests on non-instrumented builds, because timing on @@ -91,7 +91,7 @@ TEST(CpuTimeTest, MAYBE_TEST(TwoThreads)) { TEST(CpuTimeTest, MAYBE_TEST(Sleeping)) { int64_t process_start_time_nanos = GetProcessCpuTimeNanos(); - SleepMs(kProcessingTimeMillisecs); + Thread::SleepMs(kProcessingTimeMillisecs); int64_t process_duration_nanos = GetProcessCpuTimeNanos() - process_start_time_nanos; // Sleeping should not introduce any additional CPU time. diff --git a/rtc_base/platform_thread_unittest.cc b/rtc_base/platform_thread_unittest.cc index b25e06a67f..921138de22 100644 --- a/rtc_base/platform_thread_unittest.cc +++ b/rtc_base/platform_thread_unittest.cc @@ -13,7 +13,6 @@ #include #include "rtc_base/event.h" -#include "system_wrappers/include/sleep.h" #include "test/gmock.h" namespace webrtc { diff --git a/rtc_base/task_queue_stdlib_unittest.cc b/rtc_base/task_queue_stdlib_unittest.cc index ca27f82219..3218a08cb7 100644 --- a/rtc_base/task_queue_stdlib_unittest.cc +++ b/rtc_base/task_queue_stdlib_unittest.cc @@ -18,7 +18,7 @@ #include "api/units/time_delta.h" #include "rtc_base/event.h" #include "rtc_base/logging.h" -#include "system_wrappers/include/sleep.h" +#include "rtc_base/thread.h" #include "test/gtest.h" namespace webrtc { @@ -54,7 +54,7 @@ TEST(TaskQueueStdlib, AvoidsSpammingLogOnInactivity) { auto task_queue = CreateTaskQueueStdlibFactory()->CreateTaskQueue( "test", TaskQueueFactory::Priority::NORMAL); auto wait_duration = Event::kDefaultWarnDuration + TimeDelta::Seconds(1); - SleepMs(wait_duration.ms()); + Thread::SleepMs(wait_duration.ms()); EXPECT_EQ(log_output.length(), 0u); task_queue = nullptr; LogMessage::RemoveLogToStream(&stream); diff --git a/rtc_tools/video_replay.cc b/rtc_tools/video_replay.cc index 56a0e2187d..d281a8c316 100644 --- a/rtc_tools/video_replay.cc +++ b/rtc_tools/video_replay.cc @@ -60,7 +60,6 @@ #include "rtc_base/system/file_wrapper.h" #include "rtc_base/thread.h" #include "system_wrappers/include/clock.h" -#include "system_wrappers/include/sleep.h" #include "test/call_config_utils.h" #include "test/encoder_settings.h" #include "test/fake_decoder.h" @@ -703,7 +702,7 @@ class RtpReplayer final { if (time_sim_) { time_sim_->AdvanceTime(TimeDelta::Millis(duration_ms)); } else if (duration_ms > 0) { - SleepMs(duration_ms); + Thread::SleepMs(duration_ms); } } diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn index a2ae393859..9c9cd415db 100644 --- a/sdk/android/BUILD.gn +++ b/sdk/android/BUILD.gn @@ -1749,7 +1749,6 @@ if (is_android) { "../../rtc_base:timeutils", "../../rtc_base/synchronization:mutex", "../../rtc_base/system:inline", - "../../system_wrappers", "../../test:fileutils", "../../test:scoped_key_value_config", "../../test:test_support", diff --git a/sdk/android/native_unittests/stacktrace/stacktrace_unittest.cc b/sdk/android/native_unittests/stacktrace/stacktrace_unittest.cc index 0d81cd6918..1ecd2d6dff 100644 --- a/sdk/android/native_unittests/stacktrace/stacktrace_unittest.cc +++ b/sdk/android/native_unittests/stacktrace/stacktrace_unittest.cc @@ -24,7 +24,7 @@ #include "rtc_base/strings/string_builder.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/system/inline.h" -#include "system_wrappers/include/sleep.h" +#include "rtc_base/thread.h" #include "test/gtest.h" namespace webrtc { @@ -176,7 +176,7 @@ void TestStacktrace(std::unique_ptr deadlock_impl) { // Wait until the thread has entered the deadlock region, and take a very // brief nap to give it time to reach the actual deadlock. params.deadlock_start_event.Wait(); - SleepMs(1); + Thread::SleepMs(1); // Acquire the stack trace of the thread which should now be deadlocking. std::vector stack_trace = GetStackTrace(params.tid); diff --git a/system_wrappers/BUILD.gn b/system_wrappers/BUILD.gn index 318bbaff90..a36adfa47b 100644 --- a/system_wrappers/BUILD.gn +++ b/system_wrappers/BUILD.gn @@ -19,11 +19,9 @@ rtc_library("system_wrappers") { "include/cpu_features_wrapper.h", "include/cpu_info.h", "include/ntp_time.h", - "include/sleep.h", "source/clock.cc", "source/cpu_features.cc", "source/cpu_info.cc", - "source/sleep.cc", ] defines = [] diff --git a/system_wrappers/include/sleep.h b/system_wrappers/include/sleep.h deleted file mode 100644 index 3bf8df219f..0000000000 --- a/system_wrappers/include/sleep.h +++ /dev/null @@ -1,24 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ -// An OS-independent sleep function. - -#ifndef SYSTEM_WRAPPERS_INCLUDE_SLEEP_H_ -#define SYSTEM_WRAPPERS_INCLUDE_SLEEP_H_ - -namespace webrtc { - -// This function sleeps for the specified number of milliseconds. -// It may return early if the thread is woken by some other event, -// such as the delivery of a signal on Unix. -void SleepMs(int msecs); - -} // namespace webrtc - -#endif // SYSTEM_WRAPPERS_INCLUDE_SLEEP_H_ diff --git a/system_wrappers/source/sleep.cc b/system_wrappers/source/sleep.cc deleted file mode 100644 index e2fa486118..0000000000 --- a/system_wrappers/source/sleep.cc +++ /dev/null @@ -1,36 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ -// An OS-independent sleep function. - -#include "system_wrappers/include/sleep.h" - -#ifdef _WIN32 -// For Sleep() -#include -#else -// For nanosleep() -#include -#endif - -namespace webrtc { - -void SleepMs(int msecs) { -#ifdef _WIN32 - Sleep(msecs); -#else - struct timespec short_wait; - struct timespec remainder; - short_wait.tv_sec = msecs / 1000; - short_wait.tv_nsec = (msecs % 1000) * 1000 * 1000; - nanosleep(&short_wait, &remainder); -#endif -} - -} // namespace webrtc diff --git a/test/BUILD.gn b/test/BUILD.gn index eb8bd0ddf5..85e65b3785 100644 --- a/test/BUILD.gn +++ b/test/BUILD.gn @@ -1013,6 +1013,7 @@ rtc_library("fake_video_codecs") { "../rtc_base:checks", "../rtc_base:criticalsection", "../rtc_base:macromagic", + "../rtc_base:threading", "../rtc_base:timeutils", "../rtc_base/synchronization:mutex", "../system_wrappers", @@ -1435,10 +1436,10 @@ rtc_library("video_codec_tester") { "../rtc_base:rtc_event", "../rtc_base:stringutils", "../rtc_base:task_queue_for_test", + "../rtc_base:threading", "../rtc_base:timeutils", "../rtc_base/synchronization:mutex", "../rtc_base/system:file_wrapper", - "../system_wrappers", "../test:fileutils", "../video/config:encoder_config", "../video/config:streams_config", diff --git a/test/fake_encoder.cc b/test/fake_encoder.cc index 15e2702c79..e069b4b4a3 100644 --- a/test/fake_encoder.cc +++ b/test/fake_encoder.cc @@ -24,7 +24,7 @@ #include "modules/video_coding/include/video_codec_interface.h" #include "modules/video_coding/include/video_error_codes.h" #include "rtc_base/checks.h" -#include "system_wrappers/include/sleep.h" +#include "rtc_base/thread.h" namespace webrtc { namespace test { @@ -380,7 +380,7 @@ int32_t DelayedEncoder::Encode(const VideoFrame& input_image, const std::vector* frame_types) { RTC_DCHECK_RUN_ON(&sequence_checker_); - SleepMs(delay_ms_); + Thread::SleepMs(delay_ms_); return FakeEncoder::Encode(input_image, frame_types); } diff --git a/test/pc/e2e/analyzer/video/BUILD.gn b/test/pc/e2e/analyzer/video/BUILD.gn index 72b2796976..8eb5c1acfb 100644 --- a/test/pc/e2e/analyzer/video/BUILD.gn +++ b/test/pc/e2e/analyzer/video/BUILD.gn @@ -514,8 +514,8 @@ if (rtc_include_tests) { "../../../../../api/video:video_frame", "../../../../../common_video", "../../../../../rtc_base:stringutils", + "../../../../../rtc_base:threading", "../../../../../rtc_tools:video_quality_analysis", - "../../../../../system_wrappers", "../../../../time_controller", ] } @@ -540,6 +540,7 @@ if (rtc_include_tests) { "../../../../../api/video:encoded_image", "../../../../../api/video:video_frame", "../../../../../common_video", + "../../../../../rtc_base:threading", "../../../../../rtc_tools:video_quality_analysis", "../../../../../system_wrappers", "//third_party/abseil-cpp/absl/flags:flag", diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_metric_names_test.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_metric_names_test.cc index 6bc493bf79..6dfe510b98 100644 --- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_metric_names_test.cc +++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_metric_names_test.cc @@ -28,8 +28,8 @@ #include "api/video/encoded_image.h" #include "api/video/i420_buffer.h" #include "api/video/video_frame.h" +#include "rtc_base/thread.h" #include "system_wrappers/include/clock.h" -#include "system_wrappers/include/sleep.h" #include "test/gmock.h" #include "test/gtest.h" #include "test/pc/e2e/analyzer/video/default_video_quality_analyzer.h" @@ -123,7 +123,7 @@ void PassFramesThroughAnalyzer(DefaultVideoQualityAnalyzer& analyzer, analyzer.OnFrameRendered(receiver, received_frame); } if (i < frames_count - 1 && interframe_delay_ms > 0) { - SleepMs(interframe_delay_ms); + Thread::SleepMs(interframe_delay_ms); } } } diff --git a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc index 94dad5782e..449247f3a7 100644 --- a/test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc +++ b/test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc @@ -27,8 +27,8 @@ #include "api/video/video_frame.h" #include "common_video/libyuv/include/webrtc_libyuv.h" #include "rtc_base/strings/string_builder.h" +#include "rtc_base/thread.h" #include "rtc_tools/frame_analyzer/video_geometry_aligner.h" -#include "system_wrappers/include/sleep.h" #include "test/gmock.h" #include "test/gtest.h" #include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.h" @@ -163,7 +163,7 @@ void PassFramesThroughAnalyzerSenderOnly( } if (i < frames_count - 1 && interframe_delay_ms > 0) { if (time_controller == nullptr) { - SleepMs(interframe_delay_ms); + Thread::SleepMs(interframe_delay_ms); } else { time_controller->AdvanceTime(TimeDelta::Millis(interframe_delay_ms)); } @@ -200,7 +200,7 @@ void PassFramesThroughAnalyzer(DefaultVideoQualityAnalyzer& analyzer, } if (i < frames_count - 1 && interframe_delay_ms > 0) { if (time_controller == nullptr) { - SleepMs(interframe_delay_ms); + Thread::SleepMs(interframe_delay_ms); } else { time_controller->AdvanceTime(TimeDelta::Millis(interframe_delay_ms)); } @@ -248,7 +248,7 @@ TEST(DefaultVideoQualityAnalyzerTest, NormalScenario) { // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); AnalyzerStats stats = analyzer.GetAnalyzerStats(); @@ -306,7 +306,7 @@ TEST(DefaultVideoQualityAnalyzerTest, OneFrameReceivedTwice) { // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); AnalyzerStats stats = analyzer.GetAnalyzerStats(); @@ -345,23 +345,23 @@ TEST(DefaultVideoQualityAnalyzerTest, NormalScenario2Receivers) { frames_order.push_back(frame.id()); captured_frames.insert({frame.id(), frame}); analyzer.OnFramePreEncode(kAlice, frame); - SleepMs(20); + Thread::SleepMs(20); analyzer.OnFrameEncoded(kAlice, frame.id(), FakeEncode(frame), VideoQualityAnalyzerInterface::EncoderStats(), false); } - SleepMs(50); + Thread::SleepMs(50); for (size_t i = 1; i < frames_order.size(); i += 2) { uint16_t frame_id = frames_order.at(i); VideoFrame received_frame = DeepCopy(captured_frames.at(frame_id)); analyzer.OnFramePreDecode(kBob, received_frame.id(), FakeEncode(received_frame)); - SleepMs(30); + Thread::SleepMs(30); analyzer.OnFrameDecoded(kBob, received_frame, VideoQualityAnalyzerInterface::DecoderStats()); - SleepMs(10); + Thread::SleepMs(10); analyzer.OnFrameRendered(kBob, received_frame); } @@ -370,17 +370,17 @@ TEST(DefaultVideoQualityAnalyzerTest, NormalScenario2Receivers) { VideoFrame received_frame = DeepCopy(captured_frames.at(frame_id)); analyzer.OnFramePreDecode(kCharlie, received_frame.id(), FakeEncode(received_frame)); - SleepMs(40); + Thread::SleepMs(40); analyzer.OnFrameDecoded(kCharlie, received_frame, VideoQualityAnalyzerInterface::DecoderStats()); - SleepMs(5); + Thread::SleepMs(5); analyzer.OnFrameRendered(kCharlie, received_frame); } // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); AnalyzerStats analyzer_stats = analyzer.GetAnalyzerStats(); @@ -494,7 +494,7 @@ TEST(DefaultVideoQualityAnalyzerTest, // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); AnalyzerStats stats = analyzer.GetAnalyzerStats(); @@ -548,7 +548,7 @@ TEST(DefaultVideoQualityAnalyzerTest, HeavyQualityMetricsFromEqualFrames) { // Give analyzer some time to process frames on async thread. Heavy metrics // computation is turned on, so giving some extra time to be sure that // computatio have ended. - SleepMs(500); + Thread::SleepMs(500); analyzer.Stop(); AnalyzerStats stats = analyzer.GetAnalyzerStats(); @@ -613,7 +613,7 @@ TEST(DefaultVideoQualityAnalyzerTest, // Give analyzer some time to process frames on async thread. Heavy metrics // computation is turned on, so giving some extra time to be sure that // computatio have ended. - SleepMs(500); + Thread::SleepMs(500); analyzer.Stop(); AnalyzerStats stats = analyzer.GetAnalyzerStats(); @@ -678,13 +678,13 @@ TEST(DefaultVideoQualityAnalyzerTest, CpuUsage) { // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); double cpu_usage = analyzer.GetCpuUsagePercent(); ASSERT_GT(cpu_usage, 0); - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); EXPECT_EQ(analyzer.GetCpuUsagePercent(), cpu_usage); @@ -788,7 +788,7 @@ TEST(DefaultVideoQualityAnalyzerTest, RuntimeParticipantsAdding) { // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); AnalyzerStats stats = analyzer.GetAnalyzerStats(); @@ -891,7 +891,7 @@ TEST(DefaultVideoQualityAnalyzerTest, // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); AnalyzerStats stats = analyzer.GetAnalyzerStats(); @@ -966,7 +966,7 @@ TEST(DefaultVideoQualityAnalyzerTest, // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); analyzer_stats = analyzer.GetAnalyzerStats(); @@ -1063,7 +1063,7 @@ TEST(DefaultVideoQualityAnalyzerTest, // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); analyzer_stats = analyzer.GetAnalyzerStats(); @@ -1147,7 +1147,7 @@ TEST(DefaultVideoQualityAnalyzerTest, CodecTrackedCorrectly) { // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); std::map stats = analyzer.GetStats(); @@ -1228,7 +1228,7 @@ TEST(DefaultVideoQualityAnalyzerTest, // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); AnalyzerStats analyzer_stats = analyzer.GetAnalyzerStats(); @@ -1326,7 +1326,7 @@ TEST( // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); AnalyzerStats analyzer_stats = analyzer.GetAnalyzerStats(); @@ -1410,7 +1410,7 @@ TEST(DefaultVideoQualityAnalyzerTest, GetStreamFrames) { // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); EXPECT_EQ(analyzer.GetStreamFrames(), stream_to_frame_ids); @@ -1445,7 +1445,7 @@ TEST(DefaultVideoQualityAnalyzerTest, ReceiverReceivedFramesWhenSenderRemoved) { // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); FrameCounters stream_conters = @@ -1489,7 +1489,7 @@ TEST(DefaultVideoQualityAnalyzerTest, // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); FrameCounters stream_conters = @@ -1533,7 +1533,7 @@ TEST(DefaultVideoQualityAnalyzerTest, // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); FrameCounters stream_conters = @@ -1582,7 +1582,7 @@ TEST(DefaultVideoQualityAnalyzerTest, // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); FrameCounters alice_alice_stream_conters = @@ -1625,7 +1625,7 @@ TEST(DefaultVideoQualityAnalyzerTest, ReceiverRemovedBeforeCapturing2ndFrame) { // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); FrameCounters global_stream_conters = analyzer.GetGlobalCounters(); @@ -1668,7 +1668,7 @@ TEST(DefaultVideoQualityAnalyzerTest, ReceiverRemovedBeforePreEncoded) { // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); FrameCounters global_stream_conters = analyzer.GetGlobalCounters(); @@ -1711,7 +1711,7 @@ TEST(DefaultVideoQualityAnalyzerTest, ReceiverRemovedBeforeEncoded) { // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); FrameCounters global_stream_conters = analyzer.GetGlobalCounters(); @@ -1759,7 +1759,7 @@ TEST(DefaultVideoQualityAnalyzerTest, // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); FrameCounters global_stream_conters = analyzer.GetGlobalCounters(); @@ -1804,7 +1804,7 @@ TEST(DefaultVideoQualityAnalyzerTest, UnregisterOneAndRegisterAnother) { // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); FrameCounters global_stream_conters = analyzer.GetGlobalCounters(); @@ -1868,7 +1868,7 @@ TEST(DefaultVideoQualityAnalyzerTest, // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); FrameCounters global_stream_conters = analyzer.GetGlobalCounters(); @@ -1914,7 +1914,7 @@ TEST(DefaultVideoQualityAnalyzerTest, uint16_t frame_id = analyzer.OnFrameCaptured("alice", "alice_video", frame); frame.set_id(frame_id); analyzer.OnFramePreEncode("alice", frame); - SleepMs(10); + Thread::SleepMs(10); analyzer.OnFrameEncoded("alice", frame.id(), FakeEncode(frame), VideoQualityAnalyzerInterface::EncoderStats(), false); @@ -1923,7 +1923,7 @@ TEST(DefaultVideoQualityAnalyzerTest, // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); StreamStats stats = analyzer.GetStats().at(StatsKey("alice_video", "bob")); @@ -1950,7 +1950,7 @@ TEST(DefaultVideoQualityAnalyzerTest, InfraMetricsAreReportedWhenRequested) { // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); AnalyzerStats stats = analyzer.GetAnalyzerStats(); @@ -1982,7 +1982,7 @@ TEST(DefaultVideoQualityAnalyzerTest, InfraMetricsNotCollectedByDefault) { // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); AnalyzerStats stats = analyzer.GetAnalyzerStats(); @@ -2027,7 +2027,7 @@ TEST(DefaultVideoQualityAnalyzerTest, // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); StreamStats stats = analyzer.GetStats().at(StatsKey("alice_video", "bob")); @@ -2060,7 +2060,7 @@ TEST_P(DefaultVideoQualityAnalyzerTimeBetweenFreezesTest, // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(50); + Thread::SleepMs(50); analyzer.Stop(); StreamStats stats = analyzer.GetStats().at(StatsKey("alice_video", "bob")); @@ -2401,7 +2401,7 @@ TEST(DefaultVideoQualityAnalyzerTest, CheckFrameSenderPeerName) { // Give analyzer some time to process frames on async thread. The computations // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it // means we have an issue! - SleepMs(100); + Thread::SleepMs(100); analyzer.Stop(); EXPECT_EQ(sender_alice, kAlice); diff --git a/test/video_codec_tester.cc b/test/video_codec_tester.cc index edd917f878..7c8d21019d 100644 --- a/test/video_codec_tester.cc +++ b/test/video_codec_tester.cc @@ -82,9 +82,9 @@ #include "rtc_base/synchronization/mutex.h" #include "rtc_base/system/file_wrapper.h" #include "rtc_base/task_queue_for_test.h" +#include "rtc_base/thread.h" #include "rtc_base/thread_annotations.h" #include "rtc_base/time_utils.h" -#include "system_wrappers/include/sleep.h" #include "test/testsupport/file_utils.h" #include "test/testsupport/frame_reader.h" #include "test/testsupport/video_frame_writer.h" @@ -276,7 +276,7 @@ class LimitedTaskQueue { int64_t wait_ms = (scheduled - now).ms(); if (wait_ms > 0) { RTC_CHECK_LT(wait_ms, 10000) << "Too high wait_ms " << wait_ms; - SleepMs(wait_ms); + Thread::SleepMs(wait_ms); } std::move(task)(); --queue_size_; diff --git a/video/end_to_end_tests/bandwidth_tests.cc b/video/end_to_end_tests/bandwidth_tests.cc index 2f6ec09f6d..b073e6745d 100644 --- a/video/end_to_end_tests/bandwidth_tests.cc +++ b/video/end_to_end_tests/bandwidth_tests.cc @@ -35,9 +35,9 @@ #include "rtc_base/rate_limiter.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/task_queue_for_test.h" +#include "rtc_base/thread.h" #include "rtc_base/thread_annotations.h" #include "system_wrappers/include/field_trial.h" -#include "system_wrappers/include/sleep.h" #include "test/call_test.h" #include "test/fake_encoder.h" #include "test/field_trial.h" @@ -392,7 +392,7 @@ TEST_F(BandwidthEndToEndTest, ReportsSetEncoderRates) { return; } } - SleepMs(1); + Thread::SleepMs(1); } FAIL() << "Timed out waiting for stats reporting the currently set bitrate."; @@ -403,7 +403,7 @@ TEST_F(BandwidthEndToEndTest, ReportsSetEncoderRates) { if (send_stream_->GetStats().target_media_bitrate_bps == 0) { return; } - SleepMs(1); + Thread::SleepMs(1); } FAIL() << "Timed out waiting for stats reporting zero bitrate."; } diff --git a/video/end_to_end_tests/call_operation_tests.cc b/video/end_to_end_tests/call_operation_tests.cc index 610b871b0d..272f66196f 100644 --- a/video/end_to_end_tests/call_operation_tests.cc +++ b/video/end_to_end_tests/call_operation_tests.cc @@ -19,7 +19,7 @@ #include "api/video/video_sink_interface.h" #include "rtc_base/event.h" #include "rtc_base/task_queue_for_test.h" -#include "system_wrappers/include/sleep.h" +#include "rtc_base/thread.h" #include "test/call_test.h" #include "test/direct_transport.h" #include "test/frame_forwarder.h" @@ -88,7 +88,7 @@ TEST_F(CallOperationEndToEndTest, RendersSingleDelayedFrame) { class Renderer : public VideoSinkInterface { public: void OnFrame(const VideoFrame& video_frame) override { - SleepMs(kRenderDelayMs); + Thread::SleepMs(kRenderDelayMs); event_.Set(); } diff --git a/video/end_to_end_tests/corruption_detection_tests.cc b/video/end_to_end_tests/corruption_detection_tests.cc index 098d230622..3398929f16 100644 --- a/video/end_to_end_tests/corruption_detection_tests.cc +++ b/video/end_to_end_tests/corruption_detection_tests.cc @@ -23,7 +23,7 @@ #include "modules/video_coding/codecs/vp8/include/vp8.h" #include "rtc_base/checks.h" #include "rtc_base/task_queue_for_test.h" -#include "system_wrappers/include/sleep.h" +#include "rtc_base/thread.h" #include "test/call_test.h" #include "test/gtest.h" #include "test/video_test_constants.h" @@ -91,7 +91,7 @@ TEST_F( constexpr int kMaxIterations = 200; bool corruption_score_reported = false; for (int i = 0; i < kMaxIterations; ++i) { - SleepMs(10); + Thread::SleepMs(10); VideoReceiveStreamInterface::Stats stats; SendTask(task_queue_, [&]() { ASSERT_EQ(receive_streams_.size(), 1u); diff --git a/video/end_to_end_tests/network_state_tests.cc b/video/end_to_end_tests/network_state_tests.cc index 1f9b33664a..35b802b7e2 100644 --- a/video/end_to_end_tests/network_state_tests.cc +++ b/video/end_to_end_tests/network_state_tests.cc @@ -33,8 +33,8 @@ #include "rtc_base/event.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/task_queue_for_test.h" +#include "rtc_base/thread.h" #include "rtc_base/thread_annotations.h" -#include "system_wrappers/include/sleep.h" #include "test/call_test.h" #include "test/fake_encoder.h" #include "test/gtest.h" @@ -125,7 +125,7 @@ void NetworkStateEndToEndTest::VerifyNewVideoSendStreamsRespectNetworkState( Start(); }); - SleepMs(kSilenceTimeoutMs); + Thread::SleepMs(kSilenceTimeoutMs); SendTask(task_queue(), [this]() { Stop(); @@ -152,7 +152,7 @@ void NetworkStateEndToEndTest::VerifyNewVideoReceiveStreamsRespectNetworkState( Start(); }); - SleepMs(kSilenceTimeoutMs); + Thread::SleepMs(kSilenceTimeoutMs); SendTask(task_queue(), [this]() { Stop(); diff --git a/video/end_to_end_tests/stats_tests.cc b/video/end_to_end_tests/stats_tests.cc index 8952e9f366..13008f9878 100644 --- a/video/end_to_end_tests/stats_tests.cc +++ b/video/end_to_end_tests/stats_tests.cc @@ -45,9 +45,9 @@ #include "rtc_base/strings/string_builder.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/task_queue_for_test.h" +#include "rtc_base/thread.h" #include "rtc_base/thread_annotations.h" #include "system_wrappers/include/metrics.h" -#include "system_wrappers/include/sleep.h" #include "test/call_test.h" #include "test/fake_decoder.h" #include "test/fake_encoder.h" @@ -424,7 +424,7 @@ TEST_F(StatsEndToEndTest, TimingFramesAreReported) { } }); // Wait for at least one timing frame to be sent with 100ms grace period. - SleepMs(kDefaultTimingFramesDelayMs + 100); + Thread::SleepMs(kDefaultTimingFramesDelayMs + 100); // Check that timing frames are reported for each stream. SendTask(task_queue_, [&]() { for (const auto& receive_stream : receive_streams_) { @@ -775,7 +775,7 @@ TEST_F(StatsEndToEndTest, CallReportsRttForSender) { EXPECT_GE(stats.rtt_ms, kSendDelayMs + kReceiveDelayMs - kAllowedErrorMs); break; } - SleepMs(10); + Thread::SleepMs(10); } SendTask(task_queue(), [this]() { diff --git a/video/frame_cadence_adapter_unittest.cc b/video/frame_cadence_adapter_unittest.cc index 285b917eb7..37f73eedfb 100644 --- a/video/frame_cadence_adapter_unittest.cc +++ b/video/frame_cadence_adapter_unittest.cc @@ -33,11 +33,11 @@ #include "rtc_base/logging.h" #include "rtc_base/rate_statistics.h" #include "rtc_base/task_queue_for_test.h" +#include "rtc_base/thread.h" #include "rtc_base/time_utils.h" #include "system_wrappers/include/clock.h" #include "system_wrappers/include/metrics.h" #include "system_wrappers/include/ntp_time.h" -#include "system_wrappers/include/sleep.h" #include "test/gmock.h" #include "test/gtest.h" #include "test/scoped_key_value_config.h" @@ -1082,7 +1082,7 @@ TEST(FrameCadenceAdapterRealTimeTest, TimestampsDoNotDrift) { ++frame_counter; // Avoid the first OnFrame and sleep on the second. if (frame_counter == 2) { - SleepMs(kSleepMs); + Thread::SleepMs(kSleepMs); } else if (frame_counter == 3) { EXPECT_GE(incoming_frame.ntp_time_ms(), original_ntp_time_ms + kSleepMs); @@ -1134,7 +1134,7 @@ TEST(FrameCadenceAdapterRealTimeTest, ScheduledRepeatAllowsForSlowEncode) { // Avoid the first OnFrame and sleep on the second. if (frame_counter == 2) { start_time = clock->CurrentTime(); - SleepMs(kSleepMs); + Thread::SleepMs(kSleepMs); } else if (frame_counter == 3) { TimeDelta diff = clock->CurrentTime() - (*start_time + TimeDelta::Millis(500)); diff --git a/video/video_send_stream_tests.cc b/video/video_send_stream_tests.cc index 7003ffe933..9b0b57763d 100644 --- a/video/video_send_stream_tests.cc +++ b/video/video_send_stream_tests.cc @@ -101,7 +101,6 @@ #include "rtc_base/thread_annotations.h" #include "rtc_base/time_utils.h" #include "rtc_base/unique_id_generator.h" -#include "system_wrappers/include/sleep.h" #include "test/call_test.h" #include "test/configurable_frame_size_encoder.h" #include "test/encoder_settings.h" @@ -1534,7 +1533,7 @@ TEST_F(VideoSendStreamTest, PaddingIsPrimarilyRetransmissions) { // TODO(isheriff): Some platforms do not ramp up as expected to full // capacity due to packet scheduling delays. Fix that before getting // rid of this. - SleepMs(5000); + Thread::SleepMs(5000); { MutexLock lock(&mutex_); // Expect padding to be some percentage of total bytes sent. From c3d052470facecd09af479f9e1affc742b2dbe87 Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Tue, 20 May 2025 09:15:30 -0700 Subject: [PATCH 244/292] Improve SDP SCTP handling Bug: chromium:417930289 Change-Id: I48cf8ca617ff7d1f443a7654eb37b62d8805c9de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392540 Commit-Queue: Harald Alvestrand Reviewed-by: Tomas Gunnarsson Reviewed-by: Harald Alvestrand Reviewed-by: Jonas Oreland Cr-Commit-Position: refs/heads/main@{#44716} --- pc/webrtc_sdp.cc | 13 +++++++++---- pc/webrtc_sdp_unittest.cc | 24 ++++++++++++++++++++++++ 2 files changed, 33 insertions(+), 4 deletions(-) diff --git a/pc/webrtc_sdp.cc b/pc/webrtc_sdp.cc index e0fc9afdb4..f6dabc69a9 100644 --- a/pc/webrtc_sdp.cc +++ b/pc/webrtc_sdp.cc @@ -1292,8 +1292,14 @@ bool ParseExtmap(absl::string_view line, } static void BuildSctpContentAttributes( - std::string* message, - const SctpDataContentDescription* data_desc) { + const MediaContentDescription* media_desc, + std::string* message) { + const SctpDataContentDescription* data_desc = media_desc->as_sctp(); + if (!data_desc) { + // Ignore unsupported media types with the SCTP protocol. + return; + } + StringBuilder os; if (data_desc->use_sctpmap()) { // draft-ietf-mmusic-sctp-sdp-04 @@ -1524,8 +1530,7 @@ void BuildMediaDescription(const ContentInfo* content_info, AddLine(os.str(), message); if (IsDtlsSctp(media_desc->protocol())) { - const SctpDataContentDescription* data_desc = media_desc->as_sctp(); - BuildSctpContentAttributes(message, data_desc); + BuildSctpContentAttributes(media_desc, message); } else if (IsRtpProtocol(media_desc->protocol())) { BuildRtpContentAttributes(media_desc, media_type, msid_signaling, message); } diff --git a/pc/webrtc_sdp_unittest.cc b/pc/webrtc_sdp_unittest.cc index 3906027e6e..844973a53f 100644 --- a/pc/webrtc_sdp_unittest.cc +++ b/pc/webrtc_sdp_unittest.cc @@ -5076,5 +5076,29 @@ TEST_F(WebRtcSdpTest, ParsesNonKeyValueFmtpParameterSet) { EXPECT_EQ(codec_params[""], "not-in-key-value-format"); } +TEST_F(WebRtcSdpTest, SctpProtocolWithNonApplication) { + std::string sdp = + "v=0\r\n" + "o=- 0 3 IN IP4 127.0.0.1\r\n" + "s=-\r\n" + "t=0 0\r\n" + "a=group:BUNDLE 0\r\n" + "a=fingerprint:sha-1 " + "4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB\r\n" + "a=setup:actpass\r\n" + "a=ice-ufrag:ETEn\r\n" + "a=ice-pwd:OtSK0WpNtpUjkY4+86js7Z/l\r\n" + "m=unsupported 9 UDP/DTLS/SCTP webrtc-datachannel\r\n" + "c=IN IP4 0.0.0.0\r\n" + "a=mid:0\r\n" + "a=sctp-port:5000\r\n" + "a=max-message-size:262144\r\n"; + + auto desc = CreateSessionDescription(SdpType::kOffer, sdp); + ASSERT_NE(desc, nullptr); + std::string serialized; + EXPECT_TRUE(desc->ToString(&serialized)); +} + } // namespace } // namespace webrtc From 7d7dcb9650d67a6085a4f4a038edd45c2f4b65b5 Mon Sep 17 00:00:00 2001 From: Tommi Date: Sun, 18 May 2025 10:21:40 +0200 Subject: [PATCH 245/292] [AudioProcessingImpl] Don't use AudioFrameView with aec dump MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit In AudioProcessingImpl we receive externally provided channel data that may not be byte aligned (e.g AudioBus in chromium is aligned on a 16 byte boundary). Bug: chromium:409777951 Change-Id: I7ad9afda6f4bbfa67cfaf2ea89a40bebf4624e8b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392322 Commit-Queue: Tomas Gunnarsson Reviewed-by: Per Åhgren Cr-Commit-Position: refs/heads/main@{#44717} --- modules/audio_processing/BUILD.gn | 3 ++ modules/audio_processing/aec_dump/BUILD.gn | 8 ++++ .../aec_dump/aec_dump_impl.cc | 35 ++++++++++++++ .../audio_processing/aec_dump/aec_dump_impl.h | 13 ++++-- .../aec_dump/aec_dump_integration_test.cc | 10 +++- .../aec_dump/capture_stream_info.cc | 27 +++++++---- .../aec_dump/capture_stream_info.h | 5 ++ .../audio_processing/aec_dump/mock_aec_dump.h | 19 +++++++- modules/audio_processing/agc2/BUILD.gn | 1 + .../audio_processing/audio_processing_impl.cc | 46 +++++++++++++++---- modules/audio_processing/include/aec_dump.h | 6 +++ 11 files changed, 149 insertions(+), 24 deletions(-) diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn index b42ded6f23..d36c3c4982 100644 --- a/modules/audio_processing/BUILD.gn +++ b/modules/audio_processing/BUILD.gn @@ -91,6 +91,7 @@ rtc_library("aec_dump_interface") { deps = [ ":audio_frame_view", + "../../api/audio:audio_frame_api", "../../api/audio:audio_processing", "//third_party/abseil-cpp/absl/base:core_headers", ] @@ -156,6 +157,7 @@ rtc_library("audio_processing") { "../../api:field_trials_view", "../../api:function_view", "../../api:make_ref_counted", + "../../api:scoped_refptr", "../../api/audio:aec3_config", "../../api/audio:audio_frame_api", "../../api/audio:audio_processing", @@ -189,6 +191,7 @@ rtc_library("audio_processing") { "agc", "agc:gain_control_interface", "agc:legacy_agc", + "agc2:input_volume_controller", "agc2:input_volume_stats_reporter", "capture_levels_adjuster", "ns", diff --git a/modules/audio_processing/aec_dump/BUILD.gn b/modules/audio_processing/aec_dump/BUILD.gn index 5fb29422d8..b747aaab8f 100644 --- a/modules/audio_processing/aec_dump/BUILD.gn +++ b/modules/audio_processing/aec_dump/BUILD.gn @@ -34,6 +34,9 @@ if (rtc_include_tests) { "..:aec_dump_interface", "..:audioproc_test_utils", "../", + "../../../api/audio:audio_frame_api", + "../../../api/audio:audio_processing", + "../../../modules/audio_processing:audio_frame_view", "../../../test:test_support", ] } @@ -47,9 +50,12 @@ if (rtc_include_tests) { ":mock_aec_dump", "..:audioproc_test_utils", "../", + "../../../api:scoped_refptr", "../../../api/audio:audio_processing", "../../../api/audio:builtin_audio_processing_builder", "../../../api/environment:environment_factory", + "../../../rtc_base:checks", + "../../../test:test_support", "//testing/gtest", ] } @@ -68,7 +74,9 @@ if (rtc_enable_protobuf) { ":aec_dump", "..:aec_dump_interface", "../../../api/audio:audio_frame_api", + "../../../api/audio:audio_processing", "../../../api/task_queue", + "../../../modules/audio_processing:audio_frame_view", "../../../rtc_base:checks", "../../../rtc_base:logging", "../../../rtc_base:macromagic", diff --git a/modules/audio_processing/aec_dump/aec_dump_impl.cc b/modules/audio_processing/aec_dump/aec_dump_impl.cc index 31826ff824..92f3bd3eb3 100644 --- a/modules/audio_processing/aec_dump/aec_dump_impl.cc +++ b/modules/audio_processing/aec_dump/aec_dump_impl.cc @@ -10,15 +10,26 @@ #include "modules/audio_processing/aec_dump/aec_dump_impl.h" +#include +#include +#include #include +#include #include #include "absl/base/nullability.h" #include "absl/strings/string_view.h" +#include "api/audio/audio_processing.h" +#include "api/audio/audio_view.h" #include "api/task_queue/task_queue_base.h" #include "modules/audio_processing/aec_dump/aec_dump_factory.h" +#include "modules/audio_processing/debug.pb.h" +#include "modules/audio_processing/include/aec_dump.h" +#include "modules/audio_processing/include/audio_frame_view.h" #include "rtc_base/checks.h" #include "rtc_base/event.h" +#include "rtc_base/race_checker.h" +#include "rtc_base/system/file_wrapper.h" namespace webrtc { @@ -105,11 +116,19 @@ void AecDumpImpl::AddCaptureStreamInput( capture_stream_info_.AddInput(src); } +void AecDumpImpl::AddCaptureStreamInput(MonoView channel) { + capture_stream_info_.AddInputChannel(channel); +} + void AecDumpImpl::AddCaptureStreamOutput( const AudioFrameView& src) { capture_stream_info_.AddOutput(src); } +void AecDumpImpl::AddCaptureStreamOutput(MonoView channel) { + capture_stream_info_.AddOutputChannel(channel); +} + void AecDumpImpl::AddCaptureStreamInput(const int16_t* const data, int num_channels, int samples_per_channel) { @@ -157,6 +176,22 @@ void AecDumpImpl::WriteRenderStreamMessage( PostWriteToFileTask(std::move(event)); } +void AecDumpImpl::WriteRenderStreamMessage(const float* const* data, + int num_channels, + int samples_per_channel) { + auto event = std::make_unique(); + event->set_type(audioproc::Event::REVERSE_STREAM); + + audioproc::ReverseStream* msg = event->mutable_reverse_stream(); + + for (int i = 0; i < num_channels; ++i) { + MonoView channel_view(data[i], samples_per_channel); + msg->add_channel(channel_view.begin(), sizeof(float) * channel_view.size()); + } + + PostWriteToFileTask(std::move(event)); +} + void AecDumpImpl::WriteConfig(const InternalAPMConfig& config) { RTC_DCHECK_RUNS_SERIALIZED(&race_checker_); auto event = std::make_unique(); diff --git a/modules/audio_processing/aec_dump/aec_dump_impl.h b/modules/audio_processing/aec_dump/aec_dump_impl.h index 7638758dee..aa25eae4da 100644 --- a/modules/audio_processing/aec_dump/aec_dump_impl.h +++ b/modules/audio_processing/aec_dump/aec_dump_impl.h @@ -11,16 +11,18 @@ #ifndef MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_ #define MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_ +#include #include -#include -#include +#include "absl/base/nullability.h" +#include "api/audio/audio_processing.h" +#include "api/audio/audio_view.h" #include "api/task_queue/task_queue_base.h" #include "modules/audio_processing/aec_dump/capture_stream_info.h" #include "modules/audio_processing/include/aec_dump.h" +#include "modules/audio_processing/include/audio_frame_view.h" #include "rtc_base/race_checker.h" #include "rtc_base/system/file_wrapper.h" -#include "rtc_base/thread_annotations.h" // Files generated at build-time by the protobuf compiler. #ifdef WEBRTC_ANDROID_PLATFORM_BUILD @@ -47,7 +49,9 @@ class AecDumpImpl : public AecDump { void WriteInitMessage(const ProcessingConfig& api_format, int64_t time_now_ms) override; void AddCaptureStreamInput(const AudioFrameView& src) override; + void AddCaptureStreamInput(MonoView channel) override; void AddCaptureStreamOutput(const AudioFrameView& src) override; + void AddCaptureStreamOutput(MonoView channel) override; void AddCaptureStreamInput(const int16_t* const data, int num_channels, int samples_per_channel) override; @@ -62,6 +66,9 @@ class AecDumpImpl : public AecDump { int samples_per_channel) override; void WriteRenderStreamMessage( const AudioFrameView& src) override; + void WriteRenderStreamMessage(const float* const* data, + int num_channels, + int samples_per_channel) override; void WriteConfig(const InternalAPMConfig& config) override; diff --git a/modules/audio_processing/aec_dump/aec_dump_integration_test.cc b/modules/audio_processing/aec_dump/aec_dump_integration_test.cc index 5c23c9678f..7701877199 100644 --- a/modules/audio_processing/aec_dump/aec_dump_integration_test.cc +++ b/modules/audio_processing/aec_dump/aec_dump_integration_test.cc @@ -9,18 +9,23 @@ */ #include +#include #include #include #include "api/audio/audio_processing.h" #include "api/audio/builtin_audio_processing_builder.h" #include "api/environment/environment_factory.h" +#include "api/scoped_refptr.h" #include "modules/audio_processing/aec_dump/mock_aec_dump.h" -#include "modules/audio_processing/audio_processing_impl.h" +#include "rtc_base/checks.h" +#include "test/gmock.h" +#include "test/gtest.h" using ::testing::_; using ::testing::AtLeast; using ::testing::Exactly; +using ::testing::Matcher; using ::testing::StrictMock; namespace { @@ -59,7 +64,8 @@ TEST(AecDumpIntegration, frame.fill(0.f); webrtc::StreamConfig stream_config(kNumSampleRateHz, kNumChannels); - EXPECT_CALL(*mock_aec_dump.get(), WriteRenderStreamMessage(_, _, _)) + EXPECT_CALL(*mock_aec_dump.get(), + WriteRenderStreamMessage(Matcher(_), _, _)) .Times(Exactly(1)); apm->AttachAecDump(std::move(mock_aec_dump)); diff --git a/modules/audio_processing/aec_dump/capture_stream_info.cc b/modules/audio_processing/aec_dump/capture_stream_info.cc index 7d82a39729..993386658e 100644 --- a/modules/audio_processing/aec_dump/capture_stream_info.cc +++ b/modules/audio_processing/aec_dump/capture_stream_info.cc @@ -10,28 +10,37 @@ #include "modules/audio_processing/aec_dump/capture_stream_info.h" +#include +#include + +#include "api/audio/audio_view.h" +#include "modules/audio_processing/include/aec_dump.h" +#include "modules/audio_processing/include/audio_frame_view.h" + namespace webrtc { void CaptureStreamInfo::AddInput(const AudioFrameView& src) { - auto* stream = event_->mutable_stream(); - for (int i = 0; i < src.num_channels(); ++i) { - const auto& channel_view = src.channel(i); - stream->add_input_channel(channel_view.begin(), - sizeof(float) * channel_view.size()); + AddInputChannel(src.channel(i)); } } -void CaptureStreamInfo::AddOutput(const AudioFrameView& src) { +void CaptureStreamInfo::AddInputChannel(MonoView channel) { auto* stream = event_->mutable_stream(); + stream->add_input_channel(channel.begin(), sizeof(float) * channel.size()); +} +void CaptureStreamInfo::AddOutput(const AudioFrameView& src) { for (int i = 0; i < src.num_channels(); ++i) { - const auto& channel_view = src.channel(i); - stream->add_output_channel(channel_view.begin(), - sizeof(float) * channel_view.size()); + AddOutputChannel(src.channel(i)); } } +void CaptureStreamInfo::AddOutputChannel(MonoView channel) { + auto* stream = event_->mutable_stream(); + stream->add_output_channel(channel.begin(), sizeof(float) * channel.size()); +} + void CaptureStreamInfo::AddInput(const int16_t* const data, int num_channels, int samples_per_channel) { diff --git a/modules/audio_processing/aec_dump/capture_stream_info.h b/modules/audio_processing/aec_dump/capture_stream_info.h index 572990c150..bbbbb22a11 100644 --- a/modules/audio_processing/aec_dump/capture_stream_info.h +++ b/modules/audio_processing/aec_dump/capture_stream_info.h @@ -11,10 +11,13 @@ #ifndef MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_ #define MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_ +#include #include #include +#include "api/audio/audio_view.h" #include "modules/audio_processing/include/aec_dump.h" +#include "modules/audio_processing/include/audio_frame_view.h" // Files generated at build-time by the protobuf compiler. #ifdef WEBRTC_ANDROID_PLATFORM_BUILD @@ -33,7 +36,9 @@ class CaptureStreamInfo { ~CaptureStreamInfo() = default; void AddInput(const AudioFrameView& src); + void AddInputChannel(MonoView channel); void AddOutput(const AudioFrameView& src); + void AddOutputChannel(MonoView channel); void AddInput(const int16_t* const data, int num_channels, diff --git a/modules/audio_processing/aec_dump/mock_aec_dump.h b/modules/audio_processing/aec_dump/mock_aec_dump.h index b396739de4..c192d4613f 100644 --- a/modules/audio_processing/aec_dump/mock_aec_dump.h +++ b/modules/audio_processing/aec_dump/mock_aec_dump.h @@ -11,9 +11,12 @@ #ifndef MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_ #define MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_ -#include +#include +#include "api/audio/audio_processing.h" +#include "api/audio/audio_view.h" #include "modules/audio_processing/include/aec_dump.h" +#include "modules/audio_processing/include/audio_frame_view.h" #include "test/gmock.h" namespace webrtc { @@ -34,10 +37,18 @@ class MockAecDump : public AecDump { AddCaptureStreamInput, (const AudioFrameView& src), (override)); + MOCK_METHOD(void, + AddCaptureStreamInput, + (MonoView channel), + (override)); MOCK_METHOD(void, AddCaptureStreamOutput, (const AudioFrameView& src), (override)); + MOCK_METHOD(void, + AddCaptureStreamOutput, + (MonoView channel), + (override)); MOCK_METHOD(void, AddCaptureStreamInput, (const int16_t* const data, @@ -66,6 +77,12 @@ class MockAecDump : public AecDump { WriteRenderStreamMessage, (const AudioFrameView& src), (override)); + MOCK_METHOD(void, + WriteRenderStreamMessage, + (const float* const* data, + int num_channels, + int samples_per_channel), + (override)); MOCK_METHOD(void, WriteConfig, (const InternalAPMConfig& config), (override)); diff --git a/modules/audio_processing/agc2/BUILD.gn b/modules/audio_processing/agc2/BUILD.gn index 86609504b3..e9d1743f4e 100644 --- a/modules/audio_processing/agc2/BUILD.gn +++ b/modules/audio_processing/agc2/BUILD.gn @@ -196,6 +196,7 @@ rtc_library("input_volume_controller") { ] visibility = [ + "..:audio_processing", "..:gain_controller2", "./*", ] diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc index dfca465b63..b5b43e2462 100644 --- a/modules/audio_processing/audio_processing_impl.cc +++ b/modules/audio_processing/audio_processing_impl.cc @@ -11,30 +11,54 @@ #include "modules/audio_processing/audio_processing_impl.h" #include +#include +#include #include +#include #include #include #include #include -#include #include +#include #include "absl/base/nullability.h" #include "absl/strings/string_view.h" #include "api/array_view.h" -#include "api/audio/audio_frame.h" +#include "api/audio/audio_processing.h" +#include "api/audio/audio_processing_statistics.h" +#include "api/audio/audio_view.h" +#include "api/audio/echo_canceller3_config.h" +#include "api/audio/echo_control.h" #include "api/environment/environment.h" #include "api/field_trials_view.h" +#include "api/scoped_refptr.h" #include "api/task_queue/task_queue_base.h" #include "common_audio/audio_converter.h" #include "common_audio/include/audio_util.h" +#include "modules/audio_processing/aec3/echo_canceller3.h" #include "modules/audio_processing/aec_dump/aec_dump_factory.h" +#include "modules/audio_processing/agc/agc_manager_direct.h" +#include "modules/audio_processing/agc/gain_control.h" +#include "modules/audio_processing/agc2/input_volume_controller.h" +#include "modules/audio_processing/agc2/input_volume_stats_reporter.h" #include "modules/audio_processing/audio_buffer.h" -#include "modules/audio_processing/include/audio_frame_view.h" +#include "modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.h" +#include "modules/audio_processing/echo_control_mobile_impl.h" +#include "modules/audio_processing/gain_control_impl.h" +#include "modules/audio_processing/gain_controller2.h" +#include "modules/audio_processing/high_pass_filter.h" +#include "modules/audio_processing/include/aec_dump.h" #include "modules/audio_processing/logging/apm_data_dumper.h" +#include "modules/audio_processing/ns/noise_suppressor.h" +#include "modules/audio_processing/ns/ns_config.h" +#include "modules/audio_processing/render_queue_item_verifier.h" +#include "modules/audio_processing/rms_level.h" #include "rtc_base/checks.h" #include "rtc_base/denormal_disabler.h" #include "rtc_base/logging.h" +#include "rtc_base/swap_queue.h" +#include "rtc_base/synchronization/mutex.h" #include "rtc_base/time_utils.h" #include "rtc_base/trace_event.h" #include "system_wrappers/include/metrics.h" @@ -1587,8 +1611,7 @@ int AudioProcessingImpl::AnalyzeReverseStreamLocked( formats_.api_format.reverse_input_stream().num_frames(); const size_t num_channels = formats_.api_format.reverse_input_stream().num_channels(); - aec_dump_->WriteRenderStreamMessage( - AudioFrameView(src, num_channels, channel_size)); + aec_dump_->WriteRenderStreamMessage(src, num_channels, channel_size); } render_.render_audio->CopyFrom(src, formats_.api_format.reverse_input_stream()); @@ -2201,8 +2224,10 @@ void AudioProcessingImpl::RecordUnprocessedCaptureStream( const size_t channel_size = formats_.api_format.input_stream().num_frames(); const size_t num_channels = formats_.api_format.input_stream().num_channels(); - aec_dump_->AddCaptureStreamInput( - AudioFrameView(src, num_channels, channel_size)); + for (size_t ch = 0u; ch < num_channels; ++ch) { + aec_dump_->AddCaptureStreamInput( + MonoView(src[ch], channel_size)); + } RecordAudioProcessingState(); } @@ -2224,8 +2249,11 @@ void AudioProcessingImpl::RecordProcessedCaptureStream( const size_t channel_size = formats_.api_format.output_stream().num_frames(); const size_t num_channels = formats_.api_format.output_stream().num_channels(); - aec_dump_->AddCaptureStreamOutput(AudioFrameView( - processed_capture_stream, num_channels, channel_size)); + for (size_t ch = 0u; ch < num_channels; ++ch) { + aec_dump_->AddCaptureStreamOutput( + MonoView(processed_capture_stream[ch], channel_size)); + } + aec_dump_->WriteCaptureStreamMessage(); } diff --git a/modules/audio_processing/include/aec_dump.h b/modules/audio_processing/include/aec_dump.h index 532fa210c7..4e273d4cde 100644 --- a/modules/audio_processing/include/aec_dump.h +++ b/modules/audio_processing/include/aec_dump.h @@ -18,6 +18,7 @@ #include "absl/base/attributes.h" #include "api/audio/audio_processing.h" +#include "api/audio/audio_view.h" #include "modules/audio_processing/include/audio_frame_view.h" namespace webrtc { @@ -87,8 +88,10 @@ class AecDump { // by a WriteCaptureStreamMessage call. virtual void AddCaptureStreamInput( const AudioFrameView& src) = 0; + virtual void AddCaptureStreamInput(MonoView channel) = 0; virtual void AddCaptureStreamOutput( const AudioFrameView& src) = 0; + virtual void AddCaptureStreamOutput(MonoView channel) = 0; virtual void AddCaptureStreamInput(const int16_t* const data, int num_channels, int samples_per_channel) = 0; @@ -104,6 +107,9 @@ class AecDump { int samples_per_channel) = 0; virtual void WriteRenderStreamMessage( const AudioFrameView& src) = 0; + virtual void WriteRenderStreamMessage(const float* const* data, + int num_channels, + int samples_per_channel) = 0; virtual void WriteRuntimeSetting( const AudioProcessing::RuntimeSetting& runtime_setting) = 0; From 969486666bcec6690066a3e883b97081934396a7 Mon Sep 17 00:00:00 2001 From: Tommi Date: Wed, 21 May 2025 15:55:38 +0200 Subject: [PATCH 246/292] Add stricter checks for valid NetEq configs. MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This is a step towards moving config checks away from the audio path and closer to the initial config/construction path. Construction of decoder instances is similarly more rigorously checked. * NetEq can no longer accept 1000 audio channels as a valid config. (see NoCrashWith1000Channels test) * Improve checking for valid configurations when constructing encoders/decoders. The checks are now more aligned with the limits of the internal classes such as NetEqImpl and AudioFrame rather than e.g. using a value of 255 for channel count. * Consolidate constant definitions of "max number of audio channels". This CL adds one constant with a default value of 24. * Updating similar constants for encoders and decoders to refer to that value. * Updating AudioFrame to use the new value for sanity checking. Since AudioFrame has public member variables, those checks were being, and continue to be, bypassed unfortunately, but this is some progress. Bug: chromium:335805780, chromium:40057791 Change-Id: Ib015d1223f2c3fc00c66a831b1fa851fa50a940d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392380 Commit-Queue: Tomas Gunnarsson Reviewed-by: Per Åhgren Reviewed-by: Jakob Ivarsson‎ Cr-Commit-Position: refs/heads/main@{#44718} --- api/audio/audio_frame.cc | 6 +-- api/audio/audio_frame.h | 4 ++ api/audio/audio_view.h | 10 ++++- api/audio_codecs/BUILD.gn | 1 + api/audio_codecs/audio_decoder.cc | 3 ++ api/audio_codecs/audio_decoder.h | 3 +- api/audio_codecs/audio_encoder.cc | 3 ++ api/audio_codecs/audio_encoder.h | 3 +- api/audio_codecs/opus/BUILD.gn | 5 ++- .../audio_decoder_multi_channel_opus_config.h | 2 +- ...audio_encoder_multi_channel_opus_config.cc | 4 +- .../opus/audio_encoder_opus_config.cc | 4 +- ...audio_decoder_factory_template_unittest.cc | 9 ++++ .../codecs/g711/audio_decoder_pcm.h | 2 + .../codecs/g711/audio_encoder_pcm.cc | 3 +- .../codecs/pcm16b/audio_decoder_pcm16b.cc | 1 + .../audio_coding/neteq/audio_multi_vector.cc | 3 ++ .../audio_coding/neteq/audio_multi_vector.h | 1 + modules/audio_coding/neteq/neteq_impl.cc | 42 +++++++++++++------ modules/audio_coding/neteq/neteq_impl.h | 12 +++--- .../audio_coding/neteq/neteq_impl_unittest.cc | 28 ++++++++++--- .../neteq/neteq_network_stats_unittest.cc | 6 ++- modules/audio_mixer/BUILD.gn | 1 + .../audio_mixer/frame_combiner_unittest.cc | 11 ++--- pc/BUILD.gn | 1 + pc/webrtc_sdp.cc | 11 ++--- 26 files changed, 133 insertions(+), 46 deletions(-) diff --git a/api/audio/audio_frame.cc b/api/audio/audio_frame.cc index f578b719ff..4fc224bb52 100644 --- a/api/audio/audio_frame.cc +++ b/api/audio/audio_frame.cc @@ -38,7 +38,7 @@ AudioFrame::AudioFrame(int sample_rate_hz, channel_layout_(layout == CHANNEL_LAYOUT_UNSUPPORTED ? GuessChannelLayout(num_channels) : layout) { - RTC_DCHECK_LE(num_channels_, kMaxConcurrentChannels); + RTC_DCHECK_LE(num_channels_, kMaxNumberOfAudioChannels); RTC_DCHECK_GT(sample_rate_hz_, 0); RTC_DCHECK_GT(samples_per_channel_, 0u); } @@ -72,7 +72,7 @@ void AudioFrame::UpdateFrame(uint32_t timestamp, SpeechType speech_type, VADActivity vad_activity, size_t num_channels) { - RTC_CHECK_LE(num_channels, kMaxConcurrentChannels); + RTC_CHECK_LE(num_channels, kMaxNumberOfAudioChannels); timestamp_ = timestamp; samples_per_channel_ = samples_per_channel; sample_rate_hz_ = sample_rate_hz; @@ -172,7 +172,7 @@ InterleavedView AudioFrame::mutable_data(size_t samples_per_channel, size_t num_channels) { const size_t total_samples = samples_per_channel * num_channels; RTC_CHECK_LE(total_samples, data_.size()); - RTC_CHECK_LE(num_channels, kMaxConcurrentChannels); + RTC_CHECK_LE(num_channels, kMaxNumberOfAudioChannels); // Sanity check for valid argument values during development. // If `samples_per_channel` is < `num_channels` but larger than 0, // then chances are the order of arguments is incorrect. diff --git a/api/audio/audio_frame.h b/api/audio/audio_frame.h index 456bd2885c..f6f7abe22b 100644 --- a/api/audio/audio_frame.h +++ b/api/audio/audio_frame.h @@ -64,6 +64,10 @@ class AudioFrame { enum : size_t { // Stereo, 32 kHz, 120 ms (2 * 32 * 120) // Stereo, 192 kHz, 20 ms (2 * 192 * 20) + // 8 channels (kMaxConcurrentChannels), 48 kHz, 20 ms (8 * 48 * 20). + // 24 channels (kMaxNumberOfAudioChannels), 32 kHz kHz, 10 ms (24 * 32 * 10) + // At 48 kHz, 10 ms buffers, the maximum number of channels AudioFrame can + // hold, is 16. (16 * 48 * 10). kMaxDataSizeSamples = 7680, kMaxDataSizeBytes = kMaxDataSizeSamples * sizeof(int16_t), }; diff --git a/api/audio/audio_view.h b/api/audio/audio_view.h index 719d60896c..c241b928b6 100644 --- a/api/audio/audio_view.h +++ b/api/audio/audio_view.h @@ -15,7 +15,6 @@ #include #include "api/array_view.h" -#include "api/audio/channel_layout.h" #include "rtc_base/checks.h" namespace webrtc { @@ -45,6 +44,13 @@ namespace webrtc { template using MonoView = ArrayView; +// The maximum number of audio channels supported by WebRTC encoders, decoders +// and the AudioFrame class. +// TODO(peah, tommi): Should kMaxNumberOfAudioChannels be 16 rather than 24? +// The reason is that AudioFrame's max number of samples is 7680, which can +// hold 16 10ms 16bit channels at 48 kHz (and not 24 channels). +static constexpr size_t kMaxNumberOfAudioChannels = 24; + // InterleavedView<> is a view over an interleaved audio buffer (e.g. from // AudioFrame). template @@ -59,7 +65,7 @@ class InterleavedView { : num_channels_(num_channels), samples_per_channel_(samples_per_channel), data_(data, num_channels * samples_per_channel) { - RTC_DCHECK_LE(num_channels_, kMaxConcurrentChannels); + RTC_DCHECK_LE(num_channels_, kMaxNumberOfAudioChannels); RTC_DCHECK(num_channels_ == 0u || samples_per_channel_ != 0u); } diff --git a/api/audio_codecs/BUILD.gn b/api/audio_codecs/BUILD.gn index 492a44aa2f..25ea12557b 100644 --- a/api/audio_codecs/BUILD.gn +++ b/api/audio_codecs/BUILD.gn @@ -35,6 +35,7 @@ rtc_library("audio_codecs_api") { "..:ref_count", "..:scoped_refptr", "../../api:rtp_parameters", + "../../api/audio:audio_frame_api", "../../rtc_base:buffer", "../../rtc_base:checks", "../../rtc_base:event_tracer", diff --git a/api/audio_codecs/audio_decoder.cc b/api/audio_codecs/audio_decoder.cc index 83d09bb1e5..dbdaff1d37 100644 --- a/api/audio_codecs/audio_decoder.cc +++ b/api/audio_codecs/audio_decoder.cc @@ -27,6 +27,9 @@ namespace webrtc { namespace { +// TODO(peah): Rationale +static_assert(AudioDecoder::kMaxNumberOfChannels <= 255, ""); + class OldStyleEncodedFrame final : public AudioDecoder::EncodedAudioFrame { public: OldStyleEncodedFrame(AudioDecoder* decoder, Buffer&& payload) diff --git a/api/audio_codecs/audio_decoder.h b/api/audio_codecs/audio_decoder.h index d2d5e7b30c..a085d3b059 100644 --- a/api/audio_codecs/audio_decoder.h +++ b/api/audio_codecs/audio_decoder.h @@ -19,6 +19,7 @@ #include #include "api/array_view.h" +#include "api/audio/audio_view.h" #include "rtc_base/buffer.h" namespace webrtc { @@ -173,7 +174,7 @@ class AudioDecoder { virtual size_t Channels() const = 0; // The maximum number of audio channels supported by WebRTC decoders. - static constexpr int kMaxNumberOfChannels = 24; + static constexpr int kMaxNumberOfChannels = kMaxNumberOfAudioChannels; protected: static SpeechType ConvertSpeechType(int16_t type); diff --git a/api/audio_codecs/audio_encoder.cc b/api/audio_codecs/audio_encoder.cc index 377db93c5e..b79d0a8a1e 100644 --- a/api/audio_codecs/audio_encoder.cc +++ b/api/audio_codecs/audio_encoder.cc @@ -24,6 +24,9 @@ namespace webrtc { +// TODO(peah): Rationale +static_assert(AudioEncoder::kMaxNumberOfChannels <= 255, ""); + ANAStats::ANAStats() = default; ANAStats::~ANAStats() = default; ANAStats::ANAStats(const ANAStats&) = default; diff --git a/api/audio_codecs/audio_encoder.h b/api/audio_codecs/audio_encoder.h index 56ea287319..8d45eb6201 100644 --- a/api/audio_codecs/audio_encoder.h +++ b/api/audio_codecs/audio_encoder.h @@ -22,6 +22,7 @@ #include "absl/base/attributes.h" #include "api/array_view.h" +#include "api/audio/audio_view.h" #include "api/call/bitrate_allocation.h" #include "api/units/data_rate.h" #include "api/units/time_delta.h" @@ -256,7 +257,7 @@ class AudioEncoder { } // The maximum number of audio channels supported by WebRTC encoders. - static constexpr int kMaxNumberOfChannels = 24; + static constexpr int kMaxNumberOfChannels = kMaxNumberOfAudioChannels; protected: // Subclasses implement this to perform the actual encoding. Called by diff --git a/api/audio_codecs/opus/BUILD.gn b/api/audio_codecs/opus/BUILD.gn index 2035a791be..c24e3d779b 100644 --- a/api/audio_codecs/opus/BUILD.gn +++ b/api/audio_codecs/opus/BUILD.gn @@ -20,7 +20,10 @@ rtc_library("audio_encoder_opus_config") { "audio_encoder_opus_config.cc", "audio_encoder_opus_config.h", ] - deps = [ "../../../rtc_base/system:rtc_export" ] + deps = [ + "..:audio_codecs_api", + "../../../rtc_base/system:rtc_export", + ] defines = [] if (rtc_opus_variable_complexity) { defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=1" ] diff --git a/api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h b/api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h index a24e28e036..0bb7c431e5 100644 --- a/api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h +++ b/api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h @@ -55,7 +55,7 @@ struct AudioDecoderMultiChannelOpusConfig { } } - if (num_channels > 255 || max_coded_channel >= 255) { + if (max_coded_channel >= 255) { return false; } return true; diff --git a/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.cc b/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.cc index d8367b45c9..a8123cb17a 100644 --- a/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.cc +++ b/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.cc @@ -13,6 +13,8 @@ #include #include +#include "api/audio_codecs/audio_encoder.h" + namespace webrtc { namespace { @@ -42,7 +44,7 @@ AudioEncoderMultiChannelOpusConfig::operator=( bool AudioEncoderMultiChannelOpusConfig::IsOk() const { if (frame_size_ms <= 0 || frame_size_ms % 10 != 0) return false; - if (num_channels >= 255) { + if (num_channels > AudioEncoder::kMaxNumberOfChannels) { return false; } if (bitrate_bps < kMinBitrateBps || bitrate_bps > kMaxBitrateBps) diff --git a/api/audio_codecs/opus/audio_encoder_opus_config.cc b/api/audio_codecs/opus/audio_encoder_opus_config.cc index a9ab924b38..92ee7bd66c 100644 --- a/api/audio_codecs/opus/audio_encoder_opus_config.cc +++ b/api/audio_codecs/opus/audio_encoder_opus_config.cc @@ -10,6 +10,8 @@ #include "api/audio_codecs/opus/audio_encoder_opus_config.h" +#include "api/audio_codecs/audio_encoder.h" + namespace webrtc { namespace { @@ -59,7 +61,7 @@ bool AudioEncoderOpusConfig::IsOk() const { // well; we can add support for them when needed.) return false; } - if (num_channels >= 255) { + if (num_channels > AudioEncoder::kMaxNumberOfChannels) { return false; } if (!bitrate_bps) diff --git a/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc b/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc index 50768b930a..f2dfe93608 100644 --- a/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc +++ b/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc @@ -281,5 +281,14 @@ TEST(AudioDecoderFactoryTemplateTest, Opus) { EXPECT_EQ(48000, dec->SampleRateHz()); } +TEST(AudioDecoderFactoryTemplateTest, G711TooManyChannels) { + auto factory = CreateAudioDecoderFactory(); + const Environment env = CreateEnvironment(); + EXPECT_EQ(nullptr, factory->Create(env, + {"pcmu", 16000, + /* num_channels=*/1000}, + std::nullopt)); +} + } // namespace } // namespace webrtc diff --git a/modules/audio_coding/codecs/g711/audio_decoder_pcm.h b/modules/audio_coding/codecs/g711/audio_decoder_pcm.h index 7eb2835be4..6071ef1e46 100644 --- a/modules/audio_coding/codecs/g711/audio_decoder_pcm.h +++ b/modules/audio_coding/codecs/g711/audio_decoder_pcm.h @@ -26,6 +26,7 @@ class AudioDecoderPcmU final : public AudioDecoder { public: explicit AudioDecoderPcmU(size_t num_channels) : num_channels_(num_channels) { RTC_DCHECK_GE(num_channels, 1); + RTC_DCHECK_LE(num_channels, AudioDecoder::kMaxNumberOfChannels); } AudioDecoderPcmU(const AudioDecoderPcmU&) = delete; @@ -55,6 +56,7 @@ class AudioDecoderPcmA final : public AudioDecoder { public: explicit AudioDecoderPcmA(size_t num_channels) : num_channels_(num_channels) { RTC_DCHECK_GE(num_channels, 1); + RTC_DCHECK_LE(num_channels, AudioDecoder::kMaxNumberOfChannels); } AudioDecoderPcmA(const AudioDecoderPcmA&) = delete; diff --git a/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc index 5aa27d0e8e..281be597cf 100644 --- a/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc +++ b/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc @@ -25,7 +25,8 @@ namespace webrtc { bool AudioEncoderPcm::Config::IsOk() const { - return (frame_size_ms % 10 == 0) && (num_channels >= 1); + return (frame_size_ms % 10 == 0) && (num_channels >= 1) && + (num_channels <= AudioEncoder::kMaxNumberOfChannels); } AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz) diff --git a/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc b/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc index f732d03bd1..8007b322a2 100644 --- a/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc +++ b/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc @@ -29,6 +29,7 @@ AudioDecoderPcm16B::AudioDecoderPcm16B(int sample_rate_hz, size_t num_channels) sample_rate_hz == 32000 || sample_rate_hz == 48000) << "Unsupported sample rate " << sample_rate_hz; RTC_DCHECK_GE(num_channels, 1); + RTC_DCHECK_LE(num_channels, AudioDecoder::kMaxNumberOfChannels); } void AudioDecoderPcm16B::Reset() {} diff --git a/modules/audio_coding/neteq/audio_multi_vector.cc b/modules/audio_coding/neteq/audio_multi_vector.cc index 0b68d6b9b3..baad7720eb 100644 --- a/modules/audio_coding/neteq/audio_multi_vector.cc +++ b/modules/audio_coding/neteq/audio_multi_vector.cc @@ -16,6 +16,7 @@ #include #include "api/array_view.h" +#include "api/audio/audio_view.h" #include "modules/audio_coding/neteq/audio_vector.h" #include "rtc_base/checks.h" @@ -23,6 +24,7 @@ namespace webrtc { AudioMultiVector::AudioMultiVector(size_t N) : channels_(N) { RTC_DCHECK_GT(N, 0); + RTC_DCHECK_LE(N, kMaxNumberOfAudioChannels); for (auto& c : channels_) { c.reset(new AudioVector()); } @@ -31,6 +33,7 @@ AudioMultiVector::AudioMultiVector(size_t N) : channels_(N) { AudioMultiVector::AudioMultiVector(size_t N, size_t initial_size) : channels_(N) { RTC_DCHECK_GT(N, 0); + RTC_DCHECK_LE(N, kMaxNumberOfAudioChannels); for (auto& c : channels_) { c.reset(new AudioVector(initial_size)); } diff --git a/modules/audio_coding/neteq/audio_multi_vector.h b/modules/audio_coding/neteq/audio_multi_vector.h index b87a4432b8..b98e870f82 100644 --- a/modules/audio_coding/neteq/audio_multi_vector.h +++ b/modules/audio_coding/neteq/audio_multi_vector.h @@ -22,6 +22,7 @@ namespace webrtc { +// TODO: b/335805780 - Update to use InterleavedView. class AudioMultiVector { public: // Creates an empty AudioMultiVector with `N` audio channels. `N` must be diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc index 204cde0c12..4141b76868 100644 --- a/modules/audio_coding/neteq/neteq_impl.cc +++ b/modules/audio_coding/neteq/neteq_impl.cc @@ -22,6 +22,8 @@ #include "absl/strings/str_cat.h" #include "api/array_view.h" +#include "api/audio/audio_frame.h" +#include "api/audio/audio_view.h" #include "api/audio_codecs/audio_decoder.h" #include "api/audio_codecs/audio_decoder_factory.h" #include "api/audio_codecs/audio_format.h" @@ -188,7 +190,7 @@ int NetEqImpl::InsertPacket(const RTPHeader& rtp_header, MsanCheckInitialized(payload); TRACE_EVENT0("webrtc", "NetEqImpl::InsertPacket"); MutexLock lock(&mutex_); - if (InsertPacketInternal(rtp_header, payload, packet_info) != 0) { + if (InsertPacketInternal(rtp_header, payload, packet_info) != kNoError) { return kFail; } return kOK; @@ -209,7 +211,7 @@ int NetEqImpl::GetAudio(AudioFrame* audio_frame, std::optional action_override) { TRACE_EVENT0("webrtc", "NetEqImpl::GetAudio"); MutexLock lock(&mutex_); - if (GetAudioInternal(audio_frame, action_override) != 0) { + if (GetAudioInternal(audio_frame, action_override) != kNoError) { return kFail; } stats_->IncreaseCounter(output_size_samples_, fs_hz_); @@ -452,9 +454,10 @@ NetEq::Operation NetEqImpl::last_operation_for_test() const { // Methods below this line are private. -int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header, - ArrayView payload, - const RtpPacketInfo& packet_info) { +NetEqImpl::Error NetEqImpl::InsertPacketInternal( + const RTPHeader& rtp_header, + ArrayView payload, + const RtpPacketInfo& packet_info) { if (payload.empty()) { RTC_LOG_F(LS_ERROR) << "payload is empty"; return kInvalidPointer; @@ -693,12 +696,14 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header, AudioDecoder* decoder = decoder_database_->GetDecoder(payload_type); RTC_DCHECK(decoder); // Payloads are already checked to be valid. channels = decoder->Channels(); + RTC_DCHECK_LE(channels, kMaxNumberOfAudioChannels); } const DecoderDatabase::DecoderInfo* decoder_info = decoder_database_->GetDecoderInfo(payload_type); RTC_DCHECK(decoder_info); if (decoder_info->SampleRateHz() != fs_hz_ || channels != algorithm_buffer_->Channels()) { + RTC_DCHECK_LE(channels, kMaxNumberOfAudioChannels); SetSampleRateAndChannels(decoder_info->SampleRateHz(), channels); } if (nack_enabled_) { @@ -708,7 +713,7 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header, } } - return 0; + return kNoError; } bool NetEqImpl::MaybeChangePayloadType(uint8_t payload_type) { @@ -755,6 +760,9 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame, // Make sure the total number of samples fits in the AudioFrame. if (output_size_samples_ * sync_buffer_->Channels() > AudioFrame::kMaxDataSizeSamples) { + // TODO(tommi): Remove this check. This should no longer happen + // after stricter checks were added to SetSampleRateAndChannels(). + RTC_DCHECK_NOTREACHED(); return kSampleUnderrun; } audio_frame->samples_per_channel_ = output_size_samples_; @@ -881,6 +889,9 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame, size_t num_output_samples_per_channel = output_size_samples_; size_t num_output_samples = output_size_samples_ * sync_buffer_->Channels(); if (num_output_samples > AudioFrame::kMaxDataSizeSamples) { + // TODO(tommi): Remove this check. This should no longer happen + // after stricter checks were added to SetSampleRateAndChannels(). + RTC_DCHECK_NOTREACHED(); RTC_LOG(LS_WARNING) << "Output array is too short. " << AudioFrame::kMaxDataSizeSamples << " < " << output_size_samples_ << " * " @@ -1302,7 +1313,6 @@ int NetEqImpl::Decode(PacketList* packet_list, // a reset. if (decoder_info->SampleRateHz() != fs_hz_ || decoder->Channels() != algorithm_buffer_->Channels()) { - // TODO(tlegrand): Add unittest to cover this event. SetSampleRateAndChannels(decoder_info->SampleRateHz(), decoder->Channels()); } @@ -2000,16 +2010,24 @@ void NetEqImpl::UpdatePlcComponents(int fs_hz, size_t channels) { void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) { RTC_LOG(LS_VERBOSE) << "SetSampleRateAndChannels " << fs_hz << " " << channels; - // TODO(hlundin): Change to an enumerator and skip assert. - RTC_DCHECK(fs_hz == 8000 || fs_hz == 16000 || fs_hz == 32000 || - fs_hz == 48000); - RTC_DCHECK_GT(channels, 0); + RTC_CHECK(fs_hz == 8000 || fs_hz == 16000 || fs_hz == 32000 || + fs_hz == 48000); + RTC_CHECK_GT(channels, 0); + RTC_CHECK_LE(channels, kMaxNumberOfAudioChannels); + + // The format must fit in an AudioFrame. Situations where this could + // theoratically happen but aren't supported is e.g. if receiving 24 channels + // of 10ms 48 kHz buffers. + output_size_samples_ = SampleRateToDefaultChannelSize(fs_hz); + RTC_CHECK_LE(channels * output_size_samples_, + AudioFrame::kMaxDataSizeSamples); // Before changing the sample rate, end and report any ongoing expand event. stats_->EndExpandEvent(fs_hz_); fs_hz_ = fs_hz; fs_mult_ = fs_hz / 8000; - output_size_samples_ = static_cast(kOutputSizeMs * 8 * fs_mult_); + RTC_DCHECK_EQ(output_size_samples_, + static_cast(kOutputSizeMs * 8 * fs_mult_)); decoder_frame_length_ = 3 * output_size_samples_; // Initialize to 30ms. last_mode_ = Mode::kNormal; diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h index 722fa876d3..2a5b4acdc0 100644 --- a/modules/audio_coding/neteq/neteq_impl.h +++ b/modules/audio_coding/neteq/neteq_impl.h @@ -75,7 +75,7 @@ class NetEqImpl : public webrtc::NetEq { kCodecPLC }; - enum ErrorCodes { + enum Error : int { kNoError = 0, kOtherError, kUnknownRtpPayloadType, @@ -92,7 +92,7 @@ class NetEqImpl : public webrtc::NetEq { kSampleUnderrun, kDecodedTooMuch, kRedundancySplitError, - kPacketBufferCorruption + kPacketBufferCorruption, }; struct Dependencies { @@ -217,9 +217,9 @@ class NetEqImpl : public webrtc::NetEq { // Inserts a new packet into NetEq. This is used by the InsertPacket method // above. Returns 0 on success, otherwise an error code. // TODO(hlundin): Merge this with InsertPacket above? - int InsertPacketInternal(const RTPHeader& rtp_header, - ArrayView payload, - const RtpPacketInfo& packet_info) + Error InsertPacketInternal(const RTPHeader& rtp_header, + ArrayView payload, + const RtpPacketInfo& packet_info) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_); // Returns true if the payload type changed (this should be followed by @@ -334,6 +334,8 @@ class NetEqImpl : public webrtc::NetEq { // Resets various variables and objects to new values based on the sample rate // `fs_hz` and `channels` number audio channels. + // If the sample rate, the number of channels or a combination thereof aren't + // supported, the function will fail on an RTC_CHECK. void SetSampleRateAndChannels(int fs_hz, size_t channels) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_); diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc index 999f1952ca..bc9e84ae24 100644 --- a/modules/audio_coding/neteq/neteq_impl_unittest.cc +++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc @@ -18,6 +18,7 @@ #include #include +#include "api/audio/audio_view.h" #include "api/audio_codecs/audio_decoder.h" #include "api/audio_codecs/audio_decoder_factory.h" #include "api/audio_codecs/audio_format.h" @@ -1597,8 +1598,17 @@ TEST_F(NetEqImplTest, NotifyControllerOfReorderedPacket) { EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload)); } -// When using a codec with 1000 channels, there should be no crashes. -TEST_F(NetEqImplTest, NoCrashWith1000Channels) { +#if RTC_DCHECK_IS_ON +#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) +TEST(NetEqImplDeathTest, CrashWith1000Channels) { + EXPECT_DEATH(std::make_unique(1000), ""); +} +#endif // GTEST_HAS_DEATH_TEST +#endif + +// When using a codec with kMaxNumberOfAudioChannels channels, there should be +// no crashes. +TEST_F(NetEqImplTest, NoCrashWithMaxChannels) { using ::testing::AllOf; using ::testing::Field; UseNoMocks(); @@ -1628,7 +1638,8 @@ TEST_F(NetEqImplTest, NoCrashWith1000Channels) { EXPECT_CALL(*mock_decoder_factory, Create) .WillOnce(WithArg<1>([&](const SdpAudioFormat& format) { EXPECT_EQ("pcmu", format.name); - auto dec = std::make_unique(1000); + auto dec = + std::make_unique(kMaxNumberOfAudioChannels); decoder = dec.get(); return dec; })); @@ -1651,13 +1662,18 @@ TEST_F(NetEqImplTest, NoCrashWith1000Channels) { neteq_->InsertPacket(rtp_header, payload); AudioFrame audio_frame; - bool muted; + bool muted = false; + bool got_error = false; // Repeat 40 times to ensure we enter muted state. - for (int i = 0; i < 40; i++) { + for (int i = 0; i < 40 && !muted; i++) { // GetAudio should return an error, and not crash, even in muted state. - EXPECT_NE(0, neteq_->GetAudio(&audio_frame, &muted)); + // EXPECT_NE(0, neteq_->GetAudio(&audio_frame, &muted)); + if (neteq_->GetAudio(&audio_frame, &muted) == -1) + got_error = true; } + EXPECT_TRUE(got_error); + EXPECT_TRUE(muted); } // The test first inserts a packet with narrow-band CNG, then a packet with diff --git a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc index a87590d899..4f2306dc00 100644 --- a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc +++ b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc @@ -30,6 +30,7 @@ #include "api/units/timestamp.h" #include "modules/audio_coding/neteq/tools/rtp_generator.h" #include "rtc_base/buffer.h" +#include "rtc_base/checks.h" #include "test/audio_decoder_proxy_factory.h" #include "test/gmock.h" #include "test/gtest.h" @@ -67,7 +68,10 @@ class MockAudioDecoder final : public AudioDecoder { class MockFrame : public AudioDecoder::EncodedAudioFrame { public: - MockFrame(size_t num_channels) : num_channels_(num_channels) {} + MockFrame(size_t num_channels) : num_channels_(num_channels) { + RTC_DCHECK_GE(num_channels_, 1); + RTC_DCHECK_LE(num_channels_, AudioDecoder::kMaxNumberOfChannels); + } size_t Duration() const override { return kPacketDuration; } diff --git a/modules/audio_mixer/BUILD.gn b/modules/audio_mixer/BUILD.gn index 38c4e95abb..37c82ff731 100644 --- a/modules/audio_mixer/BUILD.gn +++ b/modules/audio_mixer/BUILD.gn @@ -116,6 +116,7 @@ if (rtc_include_tests) { ":audio_mixer_test_utils", "../../api:array_view", "../../api:rtp_packet_info", + "../../api/audio:audio_frame_api", "../../api/audio:audio_mixer_api", "../../api/units:timestamp", "../../audio/utility:audio_frame_operations", diff --git a/modules/audio_mixer/frame_combiner_unittest.cc b/modules/audio_mixer/frame_combiner_unittest.cc index b481035b02..904417c25a 100644 --- a/modules/audio_mixer/frame_combiner_unittest.cc +++ b/modules/audio_mixer/frame_combiner_unittest.cc @@ -10,15 +10,17 @@ #include "modules/audio_mixer/frame_combiner.h" +#include #include #include #include -#include #include -#include #include #include "api/array_view.h" +#include "api/audio/audio_frame.h" +#include "api/audio/audio_view.h" +#include "api/audio/channel_layout.h" #include "api/rtp_packet_info.h" #include "api/rtp_packet_infos.h" #include "api/units/timestamp.h" @@ -145,14 +147,13 @@ TEST(FrameCombinerDeathTest, BuildCrashesWithManyChannels) { FrameCombiner combiner(true); for (const int rate : {8000, 18000, 34000, 48000}) { for (const int number_of_channels : {10, 20, 21}) { + RTC_DCHECK_LE(number_of_channels, kMaxNumberOfAudioChannels); if (static_cast(rate / 100 * number_of_channels) > AudioFrame::kMaxDataSizeSamples) { continue; } const std::vector all_frames = {&frame1, &frame2}; - // With an unsupported channel count, this will crash in - // `AudioFrame::UpdateFrame`. - EXPECT_DEATH(SetUpFrames(rate, number_of_channels), ""); + SetUpFrames(rate, number_of_channels); const int number_of_frames = 2; SCOPED_TRACE( diff --git a/pc/BUILD.gn b/pc/BUILD.gn index cc366601e5..704aa2ad87 100644 --- a/pc/BUILD.gn +++ b/pc/BUILD.gn @@ -1408,6 +1408,7 @@ rtc_library("webrtc_sdp") { "../api:rtc_error", "../api:rtp_parameters", "../api:rtp_transceiver_direction", + "../api/audio:audio_frame_api", "../media:codec", "../media:media_constants", "../media:rid_description", diff --git a/pc/webrtc_sdp.cc b/pc/webrtc_sdp.cc index f6dabc69a9..958287c5e7 100644 --- a/pc/webrtc_sdp.cc +++ b/pc/webrtc_sdp.cc @@ -30,6 +30,7 @@ #include "absl/strings/match.h" #include "absl/strings/str_cat.h" #include "absl/strings/string_view.h" +#include "api/audio/audio_view.h" #include "api/candidate.h" #include "api/jsep.h" #include "api/jsep_ice_candidate.h" @@ -226,9 +227,6 @@ static const char kDefaultSctpmapProtocol[] = "webrtc-datachannel"; // types. const int kWildcardPayloadType = -1; -// Maximum number of channels allowed. -static const size_t kMaxNumberOfChannels = 24; - struct SsrcInfo { uint32_t ssrc_id; std::string cname; @@ -3589,8 +3587,11 @@ bool ParseRtpmapAttribute(absl::string_view line, return false; } } - if (channels > kMaxNumberOfChannels) { - return ParseFailed(line, "At most 24 channels are supported.", error); + if (channels > kMaxNumberOfAudioChannels) { + StringBuilder description; + description << "At most " << kMaxNumberOfAudioChannels + << " channels are supported."; + return ParseFailed(line, description.Release(), error); } for (const Codec& existing_codec : media_desc->codecs()) { From e192822a3017f7fdd99514a648fafc901077287b Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Wed, 21 May 2025 07:03:23 -0700 Subject: [PATCH 247/292] Roll chromium_revision c1450c64cd..eb2ed5afec (1463258:1463392) Change log: https://chromium.googlesource.com/chromium/src/+log/c1450c64cd..eb2ed5afec Full diff: https://chromium.googlesource.com/chromium/src/+/c1450c64cd..eb2ed5afec Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/763bbf5746..480f5cd716 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/3a75313211..95e9dbf0d0 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/c1e8fc5602..472994d857 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/a19ccd2f8a..50d5bc3b7f * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/885028080a..d146714b66 * src/third_party/freetype/src: https://chromium.googlesource.com/chromium/src/third_party/freetype2.git/+log/b87f48a73e..dc8efeff2a * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/c19a471bea..f71fac7747 * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/410e8a174f..9ad2b0da83 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/609e781abb..74e0076769 DEPS diff: https://chromium.googlesource.com/chromium/src/+/c1450c64cd..eb2ed5afec/DEPS BUG=None Change-Id: Ic27d6aa1cc35e5e87145e65cb5b353cca532a0ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392768 Commit-Queue: Autoroller Bot-Commit: Autoroller Cr-Commit-Position: refs/heads/main@{#44719} --- DEPS | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/DEPS b/DEPS index 0841a712c8..1f582fcdb2 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': 'c1450c64cd894812bf4e7928f1bfa64b19e59f01', + 'chromium_revision': 'eb2ed5afeca2a0dfcc239a84d0094b82b149f252', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@763bbf5746302b010dbdd996a38a204fb0eea0a5', + 'https://chromium.googlesource.com/chromium/src/base@480f5cd716276942eda711930c7d7c820ff02506', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@3a7531321173cb50efa60fadb0b41fbef9bda5d9', + 'https://chromium.googlesource.com/chromium/src/build@95e9dbf0d04b36f7990b3ea2ac8d6d7f1f6af40b', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@cd169314af80e77a75eb25e2c0df8798f35bc04a', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@c1e8fc5602a81ef96015103aab8304eeff044338', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@472994d857b939cfe84f792da93e3ff2c0b1eb14', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@a19ccd2f8a23db0c358ff03e6229e6fb3d4e51d0', + 'https://chromium.googlesource.com/chromium/src/testing@50d5bc3b7fcaba1c59bb1c3822627f8e14bb2a03', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@885028080a17f763a8e6fa90f190f546ff20a4ab', + 'https://chromium.googlesource.com/chromium/src/third_party@d146714b665ba056fd9b56a718cb374fd0f0463a', 'src/buildtools/linux64': { 'packages': [ @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@c19a471beacf6fc3e8914dcaae7ef331986eba6a', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@f71fac77473e06142bcc6805e471ceaca1671869', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -511,7 +511,7 @@ deps = { 'condition': 'checkout_linux', }, 'src/third_party/freetype/src': - 'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@b87f48a73e3ac66c0a2b4155209a62cdcbea61b2', + 'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@dc8efeff2a69595a971609489d7ec0517fa54e9b', 'src/third_party/harfbuzz-ng/src': 'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@9f83bbbe64654b45ba5bb06927ff36c2e7588495', 'src/third_party/google_benchmark/src': { @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '410e8a174f36c6293d3aebd3833d04ac3f367b40', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '9ad2b0da836bb48d1dc3d128800b544214272ec2', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@609e781abb9a942151d56a37f4e78eada7438862', + 'https://chromium.googlesource.com/chromium/src/tools@74e0076769c32fce25a0e0594ab68b9883035615', 'src/third_party/espresso': { 'packages': [ From b0564726862508c078a0006f90d83a30eedbdd7c Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Wed, 21 May 2025 15:43:20 +0200 Subject: [PATCH 248/292] Remove unused mention of InitFieldTrialsFromString in media unittests Bug: webrtc:42225026 Change-Id: I7f63d9c23145287022342f9ef3345856633dd013 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392981 Auto-Submit: Danil Chapovalov Reviewed-by: Per Kjellander Commit-Queue: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#44720} --- media/BUILD.gn | 1 - media/engine/internal_decoder_factory_unittest.cc | 3 --- 2 files changed, 4 deletions(-) diff --git a/media/BUILD.gn b/media/BUILD.gn index a11de124ce..47782316a6 100644 --- a/media/BUILD.gn +++ b/media/BUILD.gn @@ -1039,7 +1039,6 @@ if (rtc_include_tests) { "../rtc_base/experiments:min_video_bitrate_experiment", "../rtc_base/system:file_wrapper", "../system_wrappers", - "../system_wrappers:field_trial", "../test:audio_codec_mocks", "../test:fake_video_codecs", "../test:field_trial", diff --git a/media/engine/internal_decoder_factory_unittest.cc b/media/engine/internal_decoder_factory_unittest.cc index 17c3b1381f..d5a30c1acc 100644 --- a/media/engine/internal_decoder_factory_unittest.cc +++ b/media/engine/internal_decoder_factory_unittest.cc @@ -18,7 +18,6 @@ #include "api/video_codecs/video_decoder.h" #include "api/video_codecs/video_decoder_factory.h" #include "media/base/media_constants.h" -#include "system_wrappers/include/field_trial.h" #include "test/gmock.h" #include "test/gtest.h" @@ -28,8 +27,6 @@ using ::testing::Contains; using ::testing::Field; using ::testing::Not; -using field_trial::InitFieldTrialsFromString; - #ifdef RTC_ENABLE_VP9 constexpr bool kVp9Enabled = true; #else From 514c90857dd4562894d53f5161f9825d752cb661 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Wed, 21 May 2025 11:06:23 -0700 Subject: [PATCH 249/292] Roll chromium_revision eb2ed5afec..963e5bc411 (1463392:1463586) Change log: https://chromium.googlesource.com/chromium/src/+log/eb2ed5afec..963e5bc411 Full diff: https://chromium.googlesource.com/chromium/src/+/eb2ed5afec..963e5bc411 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/480f5cd716..b263126c0d * src/build: https://chromium.googlesource.com/chromium/src/build/+log/95e9dbf0d0..bab0217478 * src/buildtools/linux64: git_revision:cad8f67e2dd0cea00bbe4566efe2fdf4c1821563..git_revision:ebc8f16ca7b0d36a3e532ee90896f9eb48e5423b * src/buildtools/mac: git_revision:cad8f67e2dd0cea00bbe4566efe2fdf4c1821563..git_revision:ebc8f16ca7b0d36a3e532ee90896f9eb48e5423b * src/buildtools/win: git_revision:cad8f67e2dd0cea00bbe4566efe2fdf4c1821563..git_revision:ebc8f16ca7b0d36a3e532ee90896f9eb48e5423b * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/472994d857..51ac1cd0db * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/50d5bc3b7f..8bf19bfbb6 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/d146714b66..894344ce34 * src/third_party/android_deps/autorolled/cipd: _jfsXVgk5GB8GWdnHRUcV__xkPxOGvUXpabqaMcVBfcC..p_9xkMekw1ax7bzdkzI9gjCXkONcxJb-xhJDLgyxxlUC * src/third_party/androidx/cipd: soQYlPZv5PZNVpLURHERMQJ9xpNLVvNZ1uzgNs2ZoEIC..MeWFyffHP4p_HRbUyL4Ry3MFkz-AyFsU82kR5iCGeDsC * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/be17ebd7c1..3369878187 * src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/2a15f4c283..a060a8eafc * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/0888983e47..f73c587323 * src/third_party/freetype/src: https://chromium.googlesource.com/chromium/src/third_party/freetype2.git/+log/dc8efeff2a..013adc82b9 * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/f71fac7747..b8cc89c389 * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/9ad2b0da83..b4fbf1b930 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/74e0076769..df3f80ae96 DEPS diff: https://chromium.googlesource.com/chromium/src/+/eb2ed5afec..963e5bc411/DEPS BUG=None Change-Id: I92d76da2e9e3659723ecd451ace213dff64b162d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393020 Commit-Queue: Autoroller Bot-Commit: Autoroller Cr-Commit-Position: refs/heads/main@{#44721} --- DEPS | 36 ++++++++++++++++++------------------ 1 file changed, 18 insertions(+), 18 deletions(-) diff --git a/DEPS b/DEPS index 1f582fcdb2..a28e2d8660 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': 'eb2ed5afeca2a0dfcc239a84d0094b82b149f252', + 'chromium_revision': '963e5bc41111d4f9a00600123af8202d7d982d15', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@480f5cd716276942eda711930c7d7c820ff02506', + 'https://chromium.googlesource.com/chromium/src/base@b263126c0dd2045ef17f6ed73554c6053955bdc7', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@95e9dbf0d04b36f7990b3ea2ac8d6d7f1f6af40b', + 'https://chromium.googlesource.com/chromium/src/build@bab02174789b858096626a887eb2b75a04f462ed', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@cd169314af80e77a75eb25e2c0df8798f35bc04a', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,19 +84,19 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@472994d857b939cfe84f792da93e3ff2c0b1eb14', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@51ac1cd0dbb0b3f152e1d4b20e27abf8a175ff29', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@50d5bc3b7fcaba1c59bb1c3822627f8e14bb2a03', + 'https://chromium.googlesource.com/chromium/src/testing@8bf19bfbb614a7940b12d17837dcf208b1e58afe', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@d146714b665ba056fd9b56a718cb374fd0f0463a', + 'https://chromium.googlesource.com/chromium/src/third_party@894344ce349723d80d87c28c323cdc94b1a927a5', 'src/buildtools/linux64': { 'packages': [ { 'package': 'gn/gn/linux-${{arch}}', - 'version': 'git_revision:cad8f67e2dd0cea00bbe4566efe2fdf4c1821563', + 'version': 'git_revision:ebc8f16ca7b0d36a3e532ee90896f9eb48e5423b', } ], 'dep_type': 'cipd', @@ -106,7 +106,7 @@ deps = { 'packages': [ { 'package': 'gn/gn/mac-${{arch}}', - 'version': 'git_revision:cad8f67e2dd0cea00bbe4566efe2fdf4c1821563', + 'version': 'git_revision:ebc8f16ca7b0d36a3e532ee90896f9eb48e5423b', } ], 'dep_type': 'cipd', @@ -116,7 +116,7 @@ deps = { 'packages': [ { 'package': 'gn/gn/windows-amd64', - 'version': 'git_revision:cad8f67e2dd0cea00bbe4566efe2fdf4c1821563', + 'version': 'git_revision:ebc8f16ca7b0d36a3e532ee90896f9eb48e5423b', } ], 'dep_type': 'cipd', @@ -323,7 +323,7 @@ deps = { 'src/third_party/clang-format/script': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@37f6e68a107df43b7d7e044fd36a13cbae3413f2', 'src/third_party/compiler-rt/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@2a15f4c28336302d95a664d99c18d3d0a011bd09', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@a060a8eafc4db8a156e34820b3add404f5a9c16b', 'src/third_party/libc++/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@a01c02c9d4acbdae3b7e8a2f3ee58579a9c29f96', 'src/third_party/libc++abi/src': @@ -480,7 +480,7 @@ deps = { }, 'src/third_party/boringssl/src': - 'https://boringssl.googlesource.com/boringssl.git@be17ebd7c19cd9f60ff684a953329a25a531a688', + 'https://boringssl.googlesource.com/boringssl.git@33698781873bab6685da95cde0a821bcd2e17302', 'src/third_party/breakpad/breakpad': 'https://chromium.googlesource.com/breakpad/breakpad.git@fad82477312f786994151f01c352813fabf924c0', 'src/third_party/catapult': @@ -497,13 +497,13 @@ deps = { 'src/third_party/crc32c/src': 'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558', 'src/third_party/depot_tools': - 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@0888983e473613d853c419245b97294e46497798', + 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@f73c587323cae09c7b82f149b9d30c70218fff01', 'src/third_party/ffmpeg': 'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@dcdd0fa51b65a0b1688ff6b8f0cc81908f09ded2', 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@f71fac77473e06142bcc6805e471ceaca1671869', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@b8cc89c38984ba3e38ef812ebe8f581265e0136f', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -511,7 +511,7 @@ deps = { 'condition': 'checkout_linux', }, 'src/third_party/freetype/src': - 'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@dc8efeff2a69595a971609489d7ec0517fa54e9b', + 'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@013adc82b953d8462a24e4060db41dce548ac071', 'src/third_party/harfbuzz-ng/src': 'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@9f83bbbe64654b45ba5bb06927ff36c2e7588495', 'src/third_party/google_benchmark/src': { @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '9ad2b0da836bb48d1dc3d128800b544214272ec2', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'b4fbf1b9302d8e1fcc869098c54453a659c61211', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@74e0076769c32fce25a0e0594ab68b9883035615', + 'https://chromium.googlesource.com/chromium/src/tools@df3f80ae96f62c345c0bd176d125084ab082a00e', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'soQYlPZv5PZNVpLURHERMQJ9xpNLVvNZ1uzgNs2ZoEIC', + 'version': 'MeWFyffHP4p_HRbUyL4Ry3MFkz-AyFsU82kR5iCGeDsC', }, ], 'condition': 'checkout_android and non_git_source', @@ -827,7 +827,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_deps/autorolled', - 'version': '_jfsXVgk5GB8GWdnHRUcV__xkPxOGvUXpabqaMcVBfcC', + 'version': 'p_9xkMekw1ax7bzdkzI9gjCXkONcxJb-xhJDLgyxxlUC', }, ], 'condition': 'checkout_android and non_git_source', From 1cb2ef0e06059b3d44ce9e51911f43a323b57d2c Mon Sep 17 00:00:00 2001 From: Andreas Pehrson Date: Tue, 20 May 2025 23:37:23 +0200 Subject: [PATCH 250/292] Make SckPickerProxy thread-safe with a Mutex Single-threaded access doesn't work because SckPickerProxy is a singleton and clients typically use different capturers for enumeration and capture, and runs them for those use cases on different threads. Bug: webrtc:367915807 Change-Id: I8695ea38a21bf3a307dfd21d169ef5ca2fe43809 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392780 Reviewed-by: Alexander Cooper Reviewed-by: Tomas Gunnarsson Commit-Queue: Andreas Pehrson Cr-Commit-Position: refs/heads/main@{#44722} --- .../desktop_capture/mac/sck_picker_handle.mm | 36 ++++++++++--------- .../mac/screen_capturer_sck.mm | 32 ++++++++++------- 2 files changed, 40 insertions(+), 28 deletions(-) diff --git a/modules/desktop_capture/mac/sck_picker_handle.mm b/modules/desktop_capture/mac/sck_picker_handle.mm index c1b4db19f6..dd5391f3f9 100644 --- a/modules/desktop_capture/mac/sck_picker_handle.mm +++ b/modules/desktop_capture/mac/sck_picker_handle.mm @@ -29,10 +29,7 @@ class API_AVAILABLE(macos(14.0)) SckPickerProxy { return g_picker; } - SckPickerProxy() : thread_checker_(SequenceChecker::kDetached) {} - - bool AtCapacity() const { - RTC_DCHECK_RUN_ON(&thread_checker_); + bool AtCapacityLocked() const RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_) { return handle_count_ == kMaximumStreamCount; } @@ -42,8 +39,8 @@ bool AtCapacity() const { ABSL_MUST_USE_RESULT std::optional AcquireSourceId() { - RTC_DCHECK_RUN_ON(&thread_checker_); - if (AtCapacity()) { + MutexLock lock(&mutex_); + if (AtCapacityLocked()) { return std::nullopt; } if (handle_count_ == 0) { @@ -58,25 +55,26 @@ bool AtCapacity() const { } void RelinquishSourceId(DesktopCapturer::SourceId source) { - RTC_DCHECK_RUN_ON(&thread_checker_); + MutexLock lock(&mutex_); handle_count_ -= 1; if (handle_count_ > 0) { return; } - // Detach now in case the next user (possibly after a long time) uses a - // different thread. - thread_checker_.Detach(); GetPicker().active = NO; } private: - webrtc::SequenceChecker thread_checker_; + // SckPickerProxy is a process-wide singleton. ScreenCapturerSck and + // SckPickerHandle are largely single-threaded but may be used on different + // threads. For instance some clients use a capturer on one thread for + // enumeration and on another for frame capture. Since all those capturers + // share the same SckPickerProxy instance, it must be thread-safe. + Mutex mutex_; // 100 is an arbitrary number that seems high enough to never get reached, // while still providing a reasonably low upper bound. static constexpr size_t kMaximumStreamCount = 100; - size_t handle_count_ RTC_GUARDED_BY(thread_checker_) = 0; - DesktopCapturer::SourceId unique_source_id_ RTC_GUARDED_BY(thread_checker_) = - 0; + size_t handle_count_ RTC_GUARDED_BY(mutex_) = 0; + DesktopCapturer::SourceId unique_source_id_ RTC_GUARDED_BY(mutex_) = 0; }; class API_AVAILABLE(macos(14.0)) SckPickerHandle @@ -90,7 +88,10 @@ class API_AVAILABLE(macos(14.0)) SckPickerHandle return std::unique_ptr(new SckPickerHandle(proxy, *id)); } - ~SckPickerHandle() { proxy_->RelinquishSourceId(source_); } + ~SckPickerHandle() { + RTC_DCHECK_RUN_ON(&thread_checker_); + proxy_->RelinquishSourceId(source_); + } SCContentSharingPicker* GetPicker() const override { return proxy_->GetPicker(); @@ -100,8 +101,11 @@ class API_AVAILABLE(macos(14.0)) SckPickerHandle private: SckPickerHandle(SckPickerProxy* proxy, DesktopCapturer::SourceId source) - : proxy_(proxy), source_(source) {} + : proxy_(proxy), source_(source) { + RTC_DCHECK_RUN_ON(&thread_checker_); + } + webrtc::SequenceChecker thread_checker_; SckPickerProxy* const proxy_; const DesktopCapturer::SourceId source_; }; diff --git a/modules/desktop_capture/mac/screen_capturer_sck.mm b/modules/desktop_capture/mac/screen_capturer_sck.mm index d6e3c2eeee..49c923e56b 100644 --- a/modules/desktop_capture/mac/screen_capturer_sck.mm +++ b/modules/desktop_capture/mac/screen_capturer_sck.mm @@ -72,6 +72,8 @@ class API_AVAILABLE(macos(14.0)) ScreenCapturerSck final void CaptureFrame() override; bool GetSourceList(SourceList* sources) override; bool SelectSource(SourceId id) override; + // Creates the SckPickerHandle if needed and not already done. + void EnsurePickerHandle(); // Prep for implementing DelegatedSourceListController interface, for now used // by Start(). Triggers SCContentSharingPicker. Runs on the caller's thread. void EnsureVisible(); @@ -218,15 +220,6 @@ friend void AbslStringify(Sink& sink, : api_checker_(SequenceChecker::kDetached), capture_options_(options), picker_modes_(modes) { - if (capture_options_.allow_sck_system_picker()) { - picker_handle_ = CreateSckPickerHandle(); - } - RTC_LOG(LS_INFO) << "ScreenCapturerSck " << this - << " created. allow_sck_system_picker=" - << capture_options_.allow_sck_system_picker() << ", source=" - << (picker_handle_ ? picker_handle_->Source() : -1) - << ", modes=" - << StringifiableSCContentSharingPickerMode{.modes_ = modes}; helper_ = [[SckHelper alloc] initWithCapturer:this]; } @@ -322,9 +315,25 @@ friend void AbslStringify(Sink& sink, } } +void ScreenCapturerSck::EnsurePickerHandle() { + RTC_DCHECK_RUN_ON(&api_checker_); + if (!picker_handle_ && capture_options_.allow_sck_system_picker()) { + picker_handle_ = CreateSckPickerHandle(); + RTC_LOG(LS_INFO) << "ScreenCapturerSck " << this + << " Created picker handle. allow_sck_system_picker=" + << capture_options_.allow_sck_system_picker() + << ", source=" + << (picker_handle_ ? picker_handle_->Source() : -1) + << ", modes=" + << StringifiableSCContentSharingPickerMode{ + .modes_ = picker_modes_}; + } +} + void ScreenCapturerSck::EnsureVisible() { RTC_DCHECK_RUN_ON(&api_checker_); RTC_LOG(LS_INFO) << "ScreenCapturerSck " << this << " " << __func__ << "."; + EnsurePickerHandle(); if (picker_handle_) { if (!picker_handle_registered_) { picker_handle_registered_ = true; @@ -433,15 +442,14 @@ friend void AbslStringify(Sink& sink, bool ScreenCapturerSck::GetSourceList(SourceList* sources) { RTC_DCHECK_RUN_ON(&api_checker_); sources->clear(); - if (capture_options_.allow_sck_system_picker() && picker_handle_) { + EnsurePickerHandle(); + if (picker_handle_) { sources->push_back({picker_handle_->Source(), std::string()}); } return true; } bool ScreenCapturerSck::SelectSource(SourceId id) { - RTC_DCHECK_RUN_ON(&api_checker_); - if (capture_options_.allow_sck_system_picker()) { return true; } From c2af9bfde87cb6662d2be7a581236c345e454c53 Mon Sep 17 00:00:00 2001 From: Tommi Date: Wed, 21 May 2025 23:10:43 +0200 Subject: [PATCH 251/292] Use RunLoop::Flush instead of Sleep This hopefully addresses flakiness in ResolveAfterDeleteDoesNotReturn (at least on Windows) Example run: https://ci.chromium.org/ui/p/webrtc/builders/ci/Win64%20ASan/15860/overview Bug: webrtc:0 Change-Id: I1090dba592c15d2764c523a371bdc41aadaa0526 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393080 Reviewed-by: Mirko Bonadei Auto-Submit: Tomas Gunnarsson Commit-Queue: Tomas Gunnarsson Cr-Commit-Position: refs/heads/main@{#44723} --- rtc_base/async_dns_resolver_unittest.cc | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/rtc_base/async_dns_resolver_unittest.cc b/rtc_base/async_dns_resolver_unittest.cc index ae03b204b2..5c4d65415d 100644 --- a/rtc_base/async_dns_resolver_unittest.cc +++ b/rtc_base/async_dns_resolver_unittest.cc @@ -17,7 +17,6 @@ #include "rtc_base/logging.h" #include "rtc_base/net_helpers.h" #include "rtc_base/socket_address.h" -#include "rtc_base/thread.h" #include "test/gmock.h" #include "test/gtest.h" #include "test/run_loop.h" @@ -64,7 +63,7 @@ TEST(AsyncDnsResolver, ResolveAfterDeleteDoesNotReturn) { bool done = false; resolver->Start(address, [&done] { done = true; }); resolver.reset(); // Deletes resolver. - Thread::Current()->SleepMs(1); // Allows callback to execute + loop.Flush(); // Allows callback to execute EXPECT_FALSE(done); // Expect no result. } From 203527e4ad261bf8eff61a66c5fae67018a0f277 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Wed, 21 May 2025 15:02:40 -0700 Subject: [PATCH 252/292] Roll chromium_revision 963e5bc411..c6667425ae (1463586:1463743) Change log: https://chromium.googlesource.com/chromium/src/+log/963e5bc411..c6667425ae Full diff: https://chromium.googlesource.com/chromium/src/+/963e5bc411..c6667425ae Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/b263126c0d..c55ff1f35c * src/build: https://chromium.googlesource.com/chromium/src/build/+log/bab0217478..e2ccff1243 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/51ac1cd0db..1c44c20c54 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/8bf19bfbb6..ca7c530527 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/894344ce34..66545bbd50 * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/41104509db..938fc9953b * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/b4fbf1b930..c10f213cb4 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/df3f80ae96..627bbcf8b2 DEPS diff: https://chromium.googlesource.com/chromium/src/+/963e5bc411..c6667425ae/DEPS BUG=None Change-Id: I0fba398847c4a31cfbf2854011392ca5b400cd93 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392771 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44724} --- DEPS | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/DEPS b/DEPS index a28e2d8660..b7f0d02f26 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '963e5bc41111d4f9a00600123af8202d7d982d15', + 'chromium_revision': 'c6667425ae853faf1db7fd4745d07ce36e324738', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@b263126c0dd2045ef17f6ed73554c6053955bdc7', + 'https://chromium.googlesource.com/chromium/src/base@c55ff1f35c5107a28f54d97906bf651eb094fe0b', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@bab02174789b858096626a887eb2b75a04f462ed', + 'https://chromium.googlesource.com/chromium/src/build@e2ccff1243019df5ebfa6a2f505872f3ffec6869', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@cd169314af80e77a75eb25e2c0df8798f35bc04a', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@51ac1cd0dbb0b3f152e1d4b20e27abf8a175ff29', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@1c44c20c546a76f233eb494a888a10a9f1f42dcb', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@8bf19bfbb614a7940b12d17837dcf208b1e58afe', + 'https://chromium.googlesource.com/chromium/src/testing@ca7c530527162e979de01b4bb88f1b0d948b793f', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@894344ce349723d80d87c28c323cdc94b1a927a5', + 'https://chromium.googlesource.com/chromium/src/third_party@66545bbd5099785b865809980ac774a2308c1e72', 'src/buildtools/linux64': { 'packages': [ @@ -484,7 +484,7 @@ deps = { 'src/third_party/breakpad/breakpad': 'https://chromium.googlesource.com/breakpad/breakpad.git@fad82477312f786994151f01c352813fabf924c0', 'src/third_party/catapult': - 'https://chromium.googlesource.com/catapult.git@41104509db3116fd5a5e3af4f8242b59de7adc67', + 'https://chromium.googlesource.com/catapult.git@938fc9953b4159851a52022b4f2a90cddedba2fc', 'src/third_party/ced/src': { 'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5', }, @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'b4fbf1b9302d8e1fcc869098c54453a659c61211', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'c10f213cb474e756222a165a75e8fa008e3c769b', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@df3f80ae96f62c345c0bd176d125084ab082a00e', + 'https://chromium.googlesource.com/chromium/src/tools@627bbcf8b23e50bbfdeb5f86b7f0d3f7d7c7e8f1', 'src/third_party/espresso': { 'packages': [ From 76e7925f95db199cf3d2e18caabe656661d21456 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Wed, 21 May 2025 21:02:52 -0700 Subject: [PATCH 253/292] Roll chromium_revision c6667425ae..918b47ff10 (1463743:1463861) Change log: https://chromium.googlesource.com/chromium/src/+log/c6667425ae..918b47ff10 Full diff: https://chromium.googlesource.com/chromium/src/+/c6667425ae..918b47ff10 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/c55ff1f35c..5238e7ee71 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/e2ccff1243..69dc3b19a1 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/ca7c530527..6a20ae4677 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/66545bbd50..6db5132348 * src/third_party/androidx/cipd: MeWFyffHP4p_HRbUyL4Ry3MFkz-AyFsU82kR5iCGeDsC..RPIZQSHBlkhsLX97i1zrh3lwBoksi4vxanP-y5OVJM8C * src/third_party/libvpx/source/libvpx: https://chromium.googlesource.com/webm/libvpx.git/+log/6bb2886336..b84ca9b637 * src/third_party/protobuf-javascript/src: https://chromium.googlesource.com/external/github.com/protocolbuffers/protobuf-javascript/+log/eb785a9363..28bf5df73e * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/627bbcf8b2..7299b0e228 DEPS diff: https://chromium.googlesource.com/chromium/src/+/c6667425ae..918b47ff10/DEPS BUG=None Change-Id: I137b9262ef811539c2dcd077291a4b39cd51e2dd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392773 Commit-Queue: Autoroller Bot-Commit: Autoroller Cr-Commit-Position: refs/heads/main@{#44725} --- DEPS | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/DEPS b/DEPS index b7f0d02f26..578765b924 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': 'c6667425ae853faf1db7fd4745d07ce36e324738', + 'chromium_revision': '918b47ff10eaaed2312cb07b44537b7f277fd255', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@c55ff1f35c5107a28f54d97906bf651eb094fe0b', + 'https://chromium.googlesource.com/chromium/src/base@5238e7ee7167a8f9f4b7b564720f0361d7f24e93', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@e2ccff1243019df5ebfa6a2f505872f3ffec6869', + 'https://chromium.googlesource.com/chromium/src/build@69dc3b19a1c8707994411ae29dcaddf3330965b3', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@cd169314af80e77a75eb25e2c0df8798f35bc04a', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -88,9 +88,9 @@ deps = { 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@ca7c530527162e979de01b4bb88f1b0d948b793f', + 'https://chromium.googlesource.com/chromium/src/testing@6a20ae4677a3ee6ed783b576a8586f2566356d22', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@66545bbd5099785b865809980ac774a2308c1e72', + 'https://chromium.googlesource.com/chromium/src/third_party@6db51323482ae97a2eb560c1e71c257fad364907', 'src/buildtools/linux64': { 'packages': [ @@ -599,9 +599,9 @@ deps = { 'src/third_party/perfetto': Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'c10f213cb474e756222a165a75e8fa008e3c769b', 'src/third_party/protobuf-javascript/src': - Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'eb785a9363664a402b6336dfe96aad27fb33ffa8', + Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + '28bf5df73ef2f345a936d9cc95d64ba8ed426a53', 'src/third_party/libvpx/source/libvpx': - 'https://chromium.googlesource.com/webm/libvpx.git@6bb288633613c613c87d23b32e5f1d23102b1a43', + 'https://chromium.googlesource.com/webm/libvpx.git@b84ca9b63730e7d4563573a56a66317eb0087ebf', 'src/third_party/libyuv': 'https://chromium.googlesource.com/libyuv/libyuv.git@61bdaee13a701d2b52c6dc943ccc5c888077a591', 'src/third_party/lss': { @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@627bbcf8b23e50bbfdeb5f86b7f0d3f7d7c7e8f1', + 'https://chromium.googlesource.com/chromium/src/tools@7299b0e228bde04946e912d997d517f43b167e48', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'MeWFyffHP4p_HRbUyL4Ry3MFkz-AyFsU82kR5iCGeDsC', + 'version': 'RPIZQSHBlkhsLX97i1zrh3lwBoksi4vxanP-y5OVJM8C', }, ], 'condition': 'checkout_android and non_git_source', From 6ffc178f12ec2adf5d847a26a76a9f44d5a5b94e Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Lionel=20Koenig=20G=C3=A9las?= Date: Thu, 15 May 2025 17:00:23 +0200 Subject: [PATCH 254/292] Add a PostFilter to compensate processing shortcomings MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Bug: webrtc:417439718 Change-Id: Iaea63c6cbd7d5ae556fa0371e68ed7b4ea32b16f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391864 Reviewed-by: Per Åhgren Auto-Submit: Lionel Koenig Commit-Queue: Lionel Koenig Reviewed-by: Gustaf Ullberg Cr-Commit-Position: refs/heads/main@{#44726} --- experiments/field_trials.py | 3 + modules/audio_processing/BUILD.gn | 21 +++ .../audio_processing/audio_processing_impl.cc | 12 ++ .../audio_processing/audio_processing_impl.h | 12 +- modules/audio_processing/post_filter.cc | 69 ++++++++ modules/audio_processing/post_filter.h | 49 ++++++ .../audio_processing/post_filter_unittest.cc | 164 ++++++++++++++++++ 7 files changed, 325 insertions(+), 5 deletions(-) create mode 100644 modules/audio_processing/post_filter.cc create mode 100644 modules/audio_processing/post_filter.h create mode 100644 modules/audio_processing/post_filter_unittest.cc diff --git a/experiments/field_trials.py b/experiments/field_trials.py index ec7e88a673..1ccb44be20 100755 --- a/experiments/field_trials.py +++ b/experiments/field_trials.py @@ -134,6 +134,9 @@ def bug_url(self) -> str: FieldTrial('WebRTC-Pacer-KeyframeFlushing', 42221435, date(2024, 4, 1)), + FieldTrial('WebRTC-PostFilterKillSwitch', + 417439718, + date(2025, 9, 1)), FieldTrial('WebRTC-QCM-Dynamic-AV1', 349860657, date(2025, 7, 1)), diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn index d36c3c4982..eac4a66376 100644 --- a/modules/audio_processing/BUILD.gn +++ b/modules/audio_processing/BUILD.gn @@ -82,6 +82,24 @@ rtc_library("high_pass_filter") { ] } +rtc_library("post_filter") { + visibility = [ "*" ] + + sources = [ + "post_filter.cc", + "post_filter.h", + ] + + defines = [] + + deps = [ + ":audio_buffer", + "../../api:array_view", + "../../rtc_base:checks", + "utility:cascaded_biquad_filter", + ] +} + rtc_library("aec_dump_interface") { visibility = [ "*" ] sources = [ @@ -152,6 +170,7 @@ rtc_library("audio_processing") { ":audio_frame_view", ":gain_controller2", ":high_pass_filter", + ":post_filter", ":rms_level", "../../api:array_view", "../../api:field_trials_view", @@ -337,6 +356,7 @@ if (rtc_include_tests) { ":gain_controller2", ":high_pass_filter", ":mocks", + ":post_filter", "../../api:array_view", "../../api:make_ref_counted", "../../api:scoped_refptr", @@ -436,6 +456,7 @@ if (rtc_include_tests) { "echo_detector/normalized_covariance_estimator_unittest.cc", "gain_control_unittest.cc", "high_pass_filter_unittest.cc", + "post_filter_unittest.cc", "residual_echo_detector_unittest.cc", "rms_level_unittest.cc", "test/debug_dump_replayer.cc", diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc index b5b43e2462..9a6f785657 100644 --- a/modules/audio_processing/audio_processing_impl.cc +++ b/modules/audio_processing/audio_processing_impl.cc @@ -52,6 +52,7 @@ #include "modules/audio_processing/logging/apm_data_dumper.h" #include "modules/audio_processing/ns/noise_suppressor.h" #include "modules/audio_processing/ns/ns_config.h" +#include "modules/audio_processing/post_filter.h" #include "modules/audio_processing/render_queue_item_verifier.h" #include "modules/audio_processing/rms_level.h" #include "rtc_base/checks.h" @@ -1475,6 +1476,10 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() { capture_.applied_input_volume_changed, capture_buffer); } + if (submodules_.post_filter) { + submodules_.post_filter->Process(*capture_buffer); + } + if (submodules_.capture_post_processor) { submodules_.capture_post_processor->Process(capture_buffer); } @@ -1937,6 +1942,13 @@ void AudioProcessingImpl::InitializeEchoController() { capture_nonlocked_.echo_controller_enabled = true; + if (!env_.field_trials().IsEnabled("WebRTC-PostFilterKillSwitch")) { + // Only creates a PostFilter if current sample-rate is high enough to + // require a filter. + submodules_.post_filter = PostFilter::CreateIfNeeded( + proc_sample_rate_hz(), num_proc_channels()); + } + submodules_.echo_control_mobile.reset(); aecm_render_signal_queue_.reset(); return; diff --git a/modules/audio_processing/audio_processing_impl.h b/modules/audio_processing/audio_processing_impl.h index c148d9c8a8..ffaca0151e 100644 --- a/modules/audio_processing/audio_processing_impl.h +++ b/modules/audio_processing/audio_processing_impl.h @@ -13,11 +13,12 @@ #include +#include #include -#include +#include #include #include -#include +#include #include #include "absl/base/nullability.h" @@ -25,12 +26,11 @@ #include "api/array_view.h" #include "api/audio/audio_processing.h" #include "api/audio/audio_processing_statistics.h" +#include "api/audio/echo_control.h" #include "api/environment/environment.h" -#include "api/function_view.h" +#include "api/scoped_refptr.h" #include "api/task_queue/task_queue_base.h" -#include "modules/audio_processing/aec3/echo_canceller3.h" #include "modules/audio_processing/agc/agc_manager_direct.h" -#include "modules/audio_processing/agc/gain_control.h" #include "modules/audio_processing/agc2/input_volume_stats_reporter.h" #include "modules/audio_processing/audio_buffer.h" #include "modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.h" @@ -41,6 +41,7 @@ #include "modules/audio_processing/include/aec_dump.h" #include "modules/audio_processing/include/audio_frame_proxies.h" #include "modules/audio_processing/ns/noise_suppressor.h" +#include "modules/audio_processing/post_filter.h" #include "modules/audio_processing/render_queue_item_verifier.h" #include "modules/audio_processing/rms_level.h" #include "rtc_base/gtest_prod_util.h" @@ -384,6 +385,7 @@ class AudioProcessingImpl : public AudioProcessing { std::unique_ptr echo_controller; std::unique_ptr echo_control_mobile; std::unique_ptr noise_suppressor; + std::unique_ptr post_filter; std::unique_ptr capture_levels_adjuster; } submodules_; diff --git a/modules/audio_processing/post_filter.cc b/modules/audio_processing/post_filter.cc new file mode 100644 index 0000000000..1fa94cb5c3 --- /dev/null +++ b/modules/audio_processing/post_filter.cc @@ -0,0 +1,69 @@ +/* + * Copyright (c) 2025 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "modules/audio_processing/post_filter.h" + +#include +#include +#include +#include + +#include "api/array_view.h" +#include "modules/audio_processing/audio_buffer.h" +#include "modules/audio_processing/utility/cascaded_biquad_filter.h" +#include "rtc_base/checks.h" + +namespace webrtc { + +namespace { + +// Removes frequencies above 19.5kHz. +// sos = signal.iirdesign( +// 19200 * 2 / 48000, 19500 * 2 / 48000, +// 3, 20, ftype='cheby2', output="sos") +constexpr std::array + kPostFilterCoefficients48kHz = {{ + {{0.56142156f, 1.11499931f, 0.56142156f}, {1.57914249f, 0.63379496f}}, + {{1.00000000f, 1.88944170f, 1.00000000f}, {1.55130066f, 0.68708719f}}, + {{1.00000000f, 1.76057310f, 1.00000000f}, {1.53001328f, 0.78591224f}}, + {{1.00000000f, 1.67448535f, 1.00000000f}, {1.56506670f, 0.92096576f}}, + }}; +} // namespace + +std::unique_ptr PostFilter::CreateIfNeeded(int sample_rate_hz, + size_t num_channels) { + if (sample_rate_hz != 48000) { + return nullptr; + } + + return std::unique_ptr( + new PostFilter(kPostFilterCoefficients48kHz, num_channels)); +} + +PostFilter::PostFilter( + ArrayView coefficients, + size_t num_channels) { + RTC_DCHECK(!coefficients.empty()); + + filters_.resize(num_channels); + for (size_t k = 0; k < filters_.size(); ++k) { + filters_[k].reset(new CascadedBiQuadFilter(coefficients)); + } +} + +void PostFilter::Process(AudioBuffer& audio) { + RTC_DCHECK_EQ(filters_.size(), audio.num_channels()); + for (size_t k = 0; k < audio.num_channels(); ++k) { + ArrayView channel_data = + ArrayView(&audio.channels()[k][0], audio.num_frames()); + filters_[k]->Process(channel_data); + } +} +} // namespace webrtc diff --git a/modules/audio_processing/post_filter.h b/modules/audio_processing/post_filter.h new file mode 100644 index 0000000000..cce42608a6 --- /dev/null +++ b/modules/audio_processing/post_filter.h @@ -0,0 +1,49 @@ +/* + * Copyright (c) 2025 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef MODULES_AUDIO_PROCESSING_POST_FILTER_H_ +#define MODULES_AUDIO_PROCESSING_POST_FILTER_H_ + +#include +#include +#include + +#include "api/array_view.h" +#include "modules/audio_processing/audio_buffer.h" +#include "modules/audio_processing/utility/cascaded_biquad_filter.h" + +namespace webrtc { + +// Provides functionality for general enhancement and compensation of +// artefacts/shortcomings introduced by prior processing. The processing is +// applied to the fullband signal. +class PostFilter { + public: + // Creates a Post Processing Filter. + // Returns nullptr if sample_rate is low enough that no filter are required. + static std::unique_ptr CreateIfNeeded(int sample_rate_hz, + size_t num_channels); + + ~PostFilter() = default; + PostFilter(const PostFilter&) = delete; + PostFilter& operator=(const PostFilter&) = delete; + + void Process(AudioBuffer& audio); + + private: + PostFilter( + ArrayView coefficients, + size_t num_channels); + + std::vector> filters_; +}; +} // namespace webrtc + +#endif // MODULES_AUDIO_PROCESSING_POST_FILTER_H_ diff --git a/modules/audio_processing/post_filter_unittest.cc b/modules/audio_processing/post_filter_unittest.cc new file mode 100644 index 0000000000..096d734a93 --- /dev/null +++ b/modules/audio_processing/post_filter_unittest.cc @@ -0,0 +1,164 @@ +/* + * Copyright (c) 2025 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ +#include "modules/audio_processing/post_filter.h" + +#include + +#include +#include +#include +#include + +#include "api/array_view.h" +#include "api/audio/audio_processing.h" +#include "modules/audio_processing/audio_buffer.h" +#include "modules/audio_processing/test/audio_buffer_tools.h" +#include "test/gmock.h" +#include "test/gtest.h" + +namespace webrtc { +namespace { + +using ::testing::FloatNear; +using ::testing::Le; +using ::testing::NotNull; + +// Process one frame of data via the AudioBuffer interface and produce the +// output. +std::vector ProcessOneFrameAsAudioBuffer( + const std::vector& frame_input, + const StreamConfig& stream_config, + PostFilter* filter) { + AudioBuffer audio_buffer( + stream_config.sample_rate_hz(), stream_config.num_channels(), + stream_config.sample_rate_hz(), stream_config.num_channels(), + stream_config.sample_rate_hz(), stream_config.num_channels()); + + test::CopyVectorToAudioBuffer(stream_config, frame_input, &audio_buffer); + filter->Process(audio_buffer); + std::vector frame_output; + test::ExtractVectorFromAudioBuffer(stream_config, &audio_buffer, + &frame_output); + return frame_output; +} + +float ComputePower(ArrayView audio) { + double energy = 0.0; + std::for_each(audio.begin(), audio.end(), + [&energy](float x) { energy += x * x; }); + return energy / audio.size(); +} + +float PowerDecibelToLinear(float db) { + return std::powf(10.0, db / 10.0); +} +} // namespace + +class PostFilterCreateTest : public ::testing::TestWithParam {}; + +TEST_P(PostFilterCreateTest, CreateOnlyFor48k) { + const int sample_rate_hz = GetParam(); + std::unique_ptr filter = + PostFilter::CreateIfNeeded(sample_rate_hz, /*num_channels=*/2); + if (sample_rate_hz == 48000) { + EXPECT_NE(filter, nullptr); + } else { + EXPECT_EQ(filter, nullptr); + } +} + +INSTANTIATE_TEST_SUITE_P( + _, + PostFilterCreateTest, + ::testing::Values(8000, 16000, 24000, 32000, 44100, 48000)); + +TEST(PostFilterTest, Tone19p8kHzSignalAttenuation48k) { + constexpr int sample_rate_hz = 48000; + + std::unique_ptr filter = + PostFilter::CreateIfNeeded(sample_rate_hz, 1); + ASSERT_THAT(filter, NotNull()); + + const StreamConfig stream_config(sample_rate_hz, 1); + + constexpr int num_frames = sample_rate_hz * 10 / 1000; // 10ms; + constexpr double tone_frequency = 19800; // Hz + + const double phase_increment = tone_frequency * 2.0 * M_PI / sample_rate_hz; + double phase = 0.0; + + std::vector audio_input(num_frames); + + // Prime the filter to avoid initial state effect. + std::generate(audio_input.begin(), audio_input.end(), + [&phase, phase_increment]() { + phase += phase_increment; + return std::sinf(phase); + }); + ProcessOneFrameAsAudioBuffer(audio_input, stream_config, filter.get()); + + for (int n = 0; n < 20; ++n) { + std::generate(audio_input.begin(), audio_input.end(), + [&phase, phase_increment]() { + phase += phase_increment; + return std::sinf(phase); + }); + const float input_power = ComputePower(audio_input); + + std::vector audio_output = + ProcessOneFrameAsAudioBuffer(audio_input, stream_config, filter.get()); + + float output_power = ComputePower(audio_output); + EXPECT_THAT(output_power / input_power, Le(PowerDecibelToLinear(-20.0))); + } +} + +TEST(PostFilterTest, Tone17kHzSignalNoAttenuation48k) { + constexpr int sample_rate_hz = 48000; + + std::unique_ptr filter = + PostFilter::CreateIfNeeded(sample_rate_hz, 1); + ASSERT_THAT(filter, NotNull()); + + const StreamConfig stream_config(sample_rate_hz, 1); + + constexpr int num_frames = sample_rate_hz * 10 / 1000; // 10ms; + constexpr double tone_frequency = 16800; // Hz + + const double phase_increment = tone_frequency * 2.0 * M_PI / sample_rate_hz; + double phase = 0.0; + + std::vector audio_input(num_frames); + + // Prime the filter to avoid initial state effect. + std::generate(audio_input.begin(), audio_input.end(), + [&phase, phase_increment]() { + phase += phase_increment; + return std::sinf(phase); + }); + ProcessOneFrameAsAudioBuffer(audio_input, stream_config, filter.get()); + + for (int n = 0; n < 20; ++n) { + std::generate(audio_input.begin(), audio_input.end(), + [&phase, phase_increment]() { + phase += phase_increment; + return std::sinf(phase); + }); + const float input_power = ComputePower(audio_input); + + std::vector audio_output = + ProcessOneFrameAsAudioBuffer(audio_input, stream_config, filter.get()); + + float output_power = ComputePower(audio_output); + EXPECT_THAT(10 * std::log10f(output_power / input_power), + FloatNear(0.0, 0.5f)); + } +} +} // namespace webrtc From 46ba194f5b3588ba1eefa7f9ca4d06f839eac3ff Mon Sep 17 00:00:00 2001 From: Jeremy Leconte Date: Thu, 22 May 2025 09:00:18 +0200 Subject: [PATCH 255/292] Reenable WebRTC FYI bots for LKGR. Change-Id: I93667f2ac8f277e05a6744c63b06e4b0a694a40a Bug: chromium:418717425 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393100 Commit-Queue: Jeremy Leconte Reviewed-by: Johannes Kron Cr-Commit-Position: refs/heads/main@{#44727} --- infra/config/config.star | 5 ++--- infra/config/cr-buildbucket.cfg | 2 ++ 2 files changed, 4 insertions(+), 3 deletions(-) diff --git a/infra/config/config.star b/infra/config/config.star index 314f6d5aaa..c9e5361484 100755 --- a/infra/config/config.star +++ b/infra/config/config.star @@ -898,9 +898,8 @@ lkgr_config = { "WebRTC Chromium FYI Android Builder (dbg)", "WebRTC Chromium FYI Android Builder ARM64 (dbg)", "WebRTC Chromium FYI Android Builder", - # TODO: b/418717425 - Reenable when flakiness is fixed. - # "WebRTC Chromium FYI Android Tests (dbg)", - # "WebRTC Chromium FYI Android Tests ARM64 (dbg)", + "WebRTC Chromium FYI Android Tests (dbg)", + "WebRTC Chromium FYI Android Tests ARM64 (dbg)", "WebRTC Chromium FYI Linux Builder (dbg)", "WebRTC Chromium FYI Linux Builder", "WebRTC Chromium FYI Linux Tester", diff --git a/infra/config/cr-buildbucket.cfg b/infra/config/cr-buildbucket.cfg index 8ff8135014..3630d8abd1 100644 --- a/infra/config/cr-buildbucket.cfg +++ b/infra/config/cr-buildbucket.cfg @@ -2041,6 +2041,8 @@ buckets { ' "WebRTC Chromium FYI Android Builder (dbg)",' ' "WebRTC Chromium FYI Android Builder ARM64 (dbg)",' ' "WebRTC Chromium FYI Android Builder",' + ' "WebRTC Chromium FYI Android Tests (dbg)",' + ' "WebRTC Chromium FYI Android Tests ARM64 (dbg)",' ' "WebRTC Chromium FYI Linux Builder (dbg)",' ' "WebRTC Chromium FYI Linux Builder",' ' "WebRTC Chromium FYI Linux Tester",' From 54c8230a472603aa97770e34b5e891eba4dde7b8 Mon Sep 17 00:00:00 2001 From: Jeremy Leconte Date: Thu, 22 May 2025 09:19:52 +0200 Subject: [PATCH 256/292] Fix the perf waterfall. From https://ci.chromium.org/p/webrtc/g/perf/console: protobuf.runtime_version.VersionError: Detected incompatible Protobuf Gencode/Runtime versions when loading histogram.proto: gencode 6.31.0 runtime 6.30.2. Change-Id: Ia909b36accc8f7ae66dd2da731d2aa76f3a9977e Bug: None Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393081 Reviewed-by: Johannes Kron Commit-Queue: Jeremy Leconte Cr-Commit-Position: refs/heads/main@{#44728} --- .vpython3 | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/.vpython3 b/.vpython3 index 6d52ebcba6..dfbd4323ef 100644 --- a/.vpython3 +++ b/.vpython3 @@ -80,7 +80,7 @@ wheel: < > wheel: < name: "infra/python/wheels/protobuf-py3" - version: "version:6.30.2" + version: "version:6.31.0" > wheel: < name: "infra/python/wheels/requests-py3" From cad50b5a1445a344dffaae0f57d5fbde95910b98 Mon Sep 17 00:00:00 2001 From: Tommi Date: Tue, 20 May 2025 20:54:36 +0200 Subject: [PATCH 257/292] [ResamplerHelper] Avoid resampling muted audio frames. MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Bug: none Change-Id: I1787b5ea6b9b33a0d37992287a4adc287fbb2ce5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392662 Commit-Queue: Tomas Gunnarsson Reviewed-by: Jakob Ivarsson‎ Cr-Commit-Position: refs/heads/main@{#44729} --- modules/audio_coding/acm2/acm_resampler.cc | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/modules/audio_coding/acm2/acm_resampler.cc b/modules/audio_coding/acm2/acm_resampler.cc index 9daa04d14b..7d42d8fa1d 100644 --- a/modules/audio_coding/acm2/acm_resampler.cc +++ b/modules/audio_coding/acm2/acm_resampler.cc @@ -58,8 +58,10 @@ bool ResamplerHelper::MaybeResample(int desired_sample_rate_hz, audio_frame->SetSampleRateAndChannelSize(desired_sample_rate_hz); InterleavedView dst = audio_frame->mutable_data( audio_frame->samples_per_channel(), audio_frame->num_channels()); - // TODO(tommi): Don't resample muted audio frames. - resampler_.Resample(src, dst); + // Muted frames will already have the audio buffer zeroed out. + if (!audio_frame->muted()) { + resampler_.Resample(src, dst); + } resampled_last_output_frame_ = true; } else { resampled_last_output_frame_ = false; From b0348979caadc534dbf5d5dc6a1a9598c6a9a628 Mon Sep 17 00:00:00 2001 From: Tommi Date: Thu, 22 May 2025 11:15:36 +0200 Subject: [PATCH 258/292] Address flakiness due to FakeNetworkManager This fixes crashes due to pending tasks e.g. triggered in tests such as VerifyUfragPwdLength. Example run: https://ci.chromium.org/ui/p/webrtc/builders/ci/Win64%20ASan/15860/overview Bug: webrtc:0 Change-Id: I99c02fee58d221a00cfb717e2d6bbb04f1873cbb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393060 Commit-Queue: Tomas Gunnarsson Auto-Submit: Tomas Gunnarsson Reviewed-by: Mirko Bonadei Cr-Commit-Position: refs/heads/main@{#44730} --- p2p/base/p2p_transport_channel_unittest.cc | 5 ++-- p2p/client/basic_port_allocator_unittest.cc | 1 + p2p/dtls/dtls_ice_integrationtest.cc | 12 +++++--- pc/peer_connection_bundle_unittest.cc | 3 ++ pc/peer_connection_histogram_unittest.cc | 10 +++---- pc/peer_connection_ice_unittest.cc | 3 +- pc/peer_connection_rampup_tests.cc | 3 +- pc/test/integration_test_helpers.cc | 2 +- rtc_base/BUILD.gn | 2 ++ rtc_base/fake_network.h | 31 ++++++++++++++++++--- 10 files changed, 54 insertions(+), 18 deletions(-) diff --git a/p2p/base/p2p_transport_channel_unittest.cc b/p2p/base/p2p_transport_channel_unittest.cc index acebe9a6c4..cceb4cc33c 100644 --- a/p2p/base/p2p_transport_channel_unittest.cc +++ b/p2p/base/p2p_transport_channel_unittest.cc @@ -65,6 +65,7 @@ #include "p2p/test/stun_server.h" #include "p2p/test/test_stun_server.h" #include "p2p/test/test_turn_server.h" +#include "rtc_base/async_packet_socket.h" #include "rtc_base/buffer.h" #include "rtc_base/byte_buffer.h" #include "rtc_base/checks.h" @@ -434,7 +435,7 @@ class P2PTransportChannelTestBase : public ::testing::Test, return ice_regathering_counter_[reason]; } - FakeNetworkManager network_manager_; + FakeNetworkManager network_manager_{Thread::Current()}; std::unique_ptr allocator_; AsyncDnsResolverFactoryInterface* async_dns_resolver_factory_ = nullptr; ChannelData cd1_; @@ -5693,7 +5694,7 @@ class P2PTransportChannelMostLikelyToWorkFirstTest private: std::unique_ptr port_allocator_; - FakeNetworkManager network_manager_; + FakeNetworkManager network_manager_{Thread::Current()}; TestTurnServer turn_server_; std::unique_ptr channel_; }; diff --git a/p2p/client/basic_port_allocator_unittest.cc b/p2p/client/basic_port_allocator_unittest.cc index 7c4f834f95..c022efe33f 100644 --- a/p2p/client/basic_port_allocator_unittest.cc +++ b/p2p/client/basic_port_allocator_unittest.cc @@ -170,6 +170,7 @@ class BasicPortAllocatorTestBase : public ::testing::Test, fss_.get(), kTurnUdpIntAddr, kTurnUdpExtAddr), + network_manager_(&thread_), candidate_allocation_done_(false) { allocator_.emplace(env_, &network_manager_, &socket_factory_); allocator_->SetConfiguration({kStunAddr}, {}, 0, NO_PRUNE, nullptr); diff --git a/p2p/dtls/dtls_ice_integrationtest.cc b/p2p/dtls/dtls_ice_integrationtest.cc index 01ebb8bd42..8a638faab1 100644 --- a/p2p/dtls/dtls_ice_integrationtest.cc +++ b/p2p/dtls/dtls_ice_integrationtest.cc @@ -148,9 +148,13 @@ class DtlsIceIntegrationTest : public ::testing::TestWithParam client_certificate, const scoped_refptr server_certificate) { thread(ep)->BlockingCall([&]() { + if (!network_manager_) { + network_manager_ = + std::make_unique(Thread::Current()); + } if (network_emulation_manager_ == nullptr) { ep.allocator = std::make_unique( - ep.env, &network_manager_, socket_factory_.get()); + ep.env, network_manager_.get(), socket_factory_.get()); } else { ep.network_manager = ep.emulated_network_manager->ReleaseNetworkManager(); @@ -234,7 +238,7 @@ class DtlsIceIntegrationTest : public ::testing::TestWithParamAddInterface(webrtc::SocketAddress("192.168.1.1", 0)); } client_thread()->BlockingCall([&]() { client_.allocator->Initialize(); }); @@ -335,11 +339,11 @@ class DtlsIceIntegrationTest : public ::testing::TestWithParam ss_; std::unique_ptr socket_factory_; std::unique_ptr network_emulation_manager_; std::unique_ptr thread_; + std::unique_ptr network_manager_; Endpoint client_; Endpoint server_; @@ -386,7 +390,7 @@ TEST_P(DtlsIceIntegrationTest, SmokeTest) { } // Validate that we can add new Connections (that become writable). - network_manager_.AddInterface(webrtc::SocketAddress("192.168.2.1", 0)); + network_manager_->AddInterface(webrtc::SocketAddress("192.168.2.1", 0)); EXPECT_THAT(webrtc::WaitUntil( [&] { return CountWritableConnections(client_.ice.get()) > 1 && diff --git a/pc/peer_connection_bundle_unittest.cc b/pc/peer_connection_bundle_unittest.cc index 56df0f92c6..b3be6d85bc 100644 --- a/pc/peer_connection_bundle_unittest.cc +++ b/pc/peer_connection_bundle_unittest.cc @@ -88,6 +88,9 @@ using ::testing::Values; class FakeNetworkManagerWithNoAnyNetwork : public FakeNetworkManager { public: + FakeNetworkManagerWithNoAnyNetwork() + : FakeNetworkManager(Thread::Current()) {} + std::vector GetAnyAddressNetworks() override { // This function allocates networks that are owned by the // NetworkManager. But some tests assume that they can release diff --git a/pc/peer_connection_histogram_unittest.cc b/pc/peer_connection_histogram_unittest.cc index 20d8ff823e..5b1da41863 100644 --- a/pc/peer_connection_histogram_unittest.cc +++ b/pc/peer_connection_histogram_unittest.cc @@ -226,8 +226,7 @@ class PeerConnectionUsageHistogramTest : public ::testing::Test { PeerConnectionDependencies deps(nullptr /* observer_in */); deps.async_dns_resolver_factory = std::make_unique>(); - - auto fake_network = std::make_unique(); + auto fake_network = std::make_unique(Thread::Current()); fake_network->set_mdns_responder( std::make_unique(Thread::Current())); fake_network->AddInterface(NextLocalAddress()); @@ -245,7 +244,7 @@ class PeerConnectionUsageHistogramTest : public ::testing::Test { } WrapperPtr CreatePeerConnectionWithPrivateLocalAddresses() { - auto fake_network = std::make_unique(); + auto fake_network = std::make_unique(Thread::Current()); fake_network->AddInterface(NextLocalAddress()); fake_network->AddInterface(kPrivateLocalAddress); @@ -257,7 +256,7 @@ class PeerConnectionUsageHistogramTest : public ::testing::Test { } WrapperPtr CreatePeerConnectionWithPrivateIpv6LocalAddresses() { - auto fake_network = std::make_unique(); + auto fake_network = std::make_unique(Thread::Current()); fake_network->AddInterface(NextLocalAddress()); fake_network->AddInterface(kPrivateIpv6LocalAddress); @@ -283,7 +282,8 @@ class PeerConnectionUsageHistogramTest : public ::testing::Test { } else { // If no network manager is provided, one will be created that uses the // host network. This doesn't work on all trybots. - auto fake_network = std::make_unique(); + auto fake_network = + std::make_unique(pcf_deps.network_thread); fake_network->AddInterface(NextLocalAddress()); pcf_deps.network_manager = std::move(fake_network); } diff --git a/pc/peer_connection_ice_unittest.cc b/pc/peer_connection_ice_unittest.cc index 9872a9a79d..9e70705544 100644 --- a/pc/peer_connection_ice_unittest.cc +++ b/pc/peer_connection_ice_unittest.cc @@ -158,7 +158,8 @@ class PeerConnectionIceBaseTest : public ::testing::Test { pcf_deps.worker_thread = Thread::Current(); pcf_deps.signaling_thread = Thread::Current(); pcf_deps.socket_factory = &vss_; - auto network_manager = std::make_unique(); + auto network_manager = + std::make_unique(pcf_deps.network_thread); auto* fake_network = network_manager.get(); pcf_deps.network_manager = std::move(network_manager); pcf_deps.adm = FakeAudioCaptureModule::Create(); diff --git a/pc/peer_connection_rampup_tests.cc b/pc/peer_connection_rampup_tests.cc index 4cf6bacaae..8b928531c6 100644 --- a/pc/peer_connection_rampup_tests.cc +++ b/pc/peer_connection_rampup_tests.cc @@ -178,7 +178,8 @@ class PeerConnectionRampUpTest : public ::testing::Test { pcf_deps.worker_thread = worker_thread_.get(); pcf_deps.signaling_thread = Thread::Current(); pcf_deps.socket_factory = &firewall_socket_server_; - auto network_manager = std::make_unique(); + auto network_manager = + std::make_unique(network_thread()); network_manager->AddInterface(kDefaultLocalAddress); pcf_deps.network_manager = std::move(network_manager); pcf_deps.adm = FakeAudioCaptureModule::Create(); diff --git a/pc/test/integration_test_helpers.cc b/pc/test/integration_test_helpers.cc index 4fc57c3bf5..ecd2aa311e 100644 --- a/pc/test/integration_test_helpers.cc +++ b/pc/test/integration_test_helpers.cc @@ -231,7 +231,7 @@ bool PeerConnectionIntegrationWrapper::Init( RTC_DCHECK(!peer_connection_); RTC_DCHECK(!peer_connection_factory_); - auto network_manager = std::make_unique(); + auto network_manager = std::make_unique(network_thread); fake_network_manager_ = network_manager.get(); fake_network_manager_->AddInterface(kDefaultLocalAddress); diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn index b019bb01f3..1211f0c856 100644 --- a/rtc_base/BUILD.gn +++ b/rtc_base/BUILD.gn @@ -1757,6 +1757,7 @@ rtc_library("rtc_base_tests_utils") { "../api:scoped_refptr", "../api:sequence_checker", "../api/task_queue", + "../api/task_queue:pending_task_safety_flag", "../api/units:time_delta", "../api/units:timestamp", "../test:scoped_key_value_config", @@ -1769,6 +1770,7 @@ rtc_library("rtc_base_tests_utils") { "//third_party/abseil-cpp/absl/base:core_headers", "//third_party/abseil-cpp/absl/memory", "//third_party/abseil-cpp/absl/strings:string_view", + "//third_party/abseil-cpp/absl/strings:strings", ] if (is_fuchsia) { deps += [ "//third_party/fuchsia-sdk/sdk/pkg/zx" ] diff --git a/rtc_base/fake_network.h b/rtc_base/fake_network.h index 6e1521a54c..b8c22136d9 100644 --- a/rtc_base/fake_network.h +++ b/rtc_base/fake_network.h @@ -17,14 +17,17 @@ #include #include +#include "absl/strings/str_cat.h" #include "absl/strings/string_view.h" +#include "api/scoped_refptr.h" +#include "api/sequence_checker.h" +#include "api/task_queue/pending_task_safety_flag.h" #include "rtc_base/ip_address.h" #include "rtc_base/mdns_responder_interface.h" #include "rtc_base/net_helpers.h" #include "rtc_base/network.h" #include "rtc_base/network_constants.h" #include "rtc_base/socket_address.h" -#include "rtc_base/string_encode.h" #include "rtc_base/thread.h" namespace webrtc { @@ -35,7 +38,18 @@ const int kFakeIPv6NetworkPrefixLength = 64; // Fake network manager that allows us to manually specify the IPs to use. class FakeNetworkManager : public NetworkManagerBase { public: - FakeNetworkManager() {} + explicit FakeNetworkManager(Thread* network_thread) + : network_thread_(network_thread) {} + ~FakeNetworkManager() override { + if (network_thread_) { + network_thread_->BlockingCall([this]() { + if (safety_flag_) { + safety_flag_->SetNotAlive(); + safety_flag_ = nullptr; + } + }); + } + } struct Iface { SocketAddress socket_address; @@ -75,12 +89,18 @@ class FakeNetworkManager : public NetworkManagerBase { } void StartUpdating() override { + RTC_DCHECK_RUN_ON(network_thread_); + if (!safety_flag_) { + safety_flag_ = PendingTaskSafetyFlag::Create(); + } ++start_count_; if (start_count_ == 1) { sent_first_update_ = false; - Thread::Current()->PostTask([this] { DoUpdateNetworks(); }); + network_thread_->PostTask( + SafeTask(safety_flag_, [this] { DoUpdateNetworks(); })); } else if (sent_first_update_) { - Thread::Current()->PostTask([this] { SignalNetworksChanged(); }); + network_thread_->PostTask( + SafeTask(safety_flag_, [this] { SignalNetworksChanged(); })); } } @@ -131,6 +151,9 @@ class FakeNetworkManager : public NetworkManagerBase { } } + Thread* const network_thread_; + scoped_refptr safety_flag_; + IfaceList ifaces_; int next_index_ = 0; int start_count_ = 0; From 84ed63cc123a7dae988fe1b94bbb7046cad28305 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Thu, 22 May 2025 03:02:45 -0700 Subject: [PATCH 259/292] Roll chromium_revision 918b47ff10..09b6d02991 (1463861:1463976) Change log: https://chromium.googlesource.com/chromium/src/+log/918b47ff10..09b6d02991 Full diff: https://chromium.googlesource.com/chromium/src/+/918b47ff10..09b6d02991 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/5238e7ee71..ad8012919b * src/buildtools: https://chromium.googlesource.com/chromium/src/buildtools/+log/cd169314af..813bee86ee * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/1c44c20c54..b46a37e0ed * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/6a20ae4677..a1165ea8b0 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/6db5132348..e68e9e5786 * src/third_party/androidx/cipd: RPIZQSHBlkhsLX97i1zrh3lwBoksi4vxanP-y5OVJM8C..pTEOsaDcd32WtfN-pVyjazgeRGUSf2p_vv-4DtFKR_IC * src/third_party/freetype/src: https://chromium.googlesource.com/chromium/src/third_party/freetype2.git/+log/013adc82b9..42a649be49 * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/c10f213cb4..04c0a05c24 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/7299b0e228..b4b877b178 DEPS diff: https://chromium.googlesource.com/chromium/src/+/918b47ff10..09b6d02991/DEPS BUG=None Change-Id: I1efdc3d4c6f4976b824fa1c812f4608e5a495593 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392775 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44731} --- DEPS | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/DEPS b/DEPS index 578765b924..6ec65a223c 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '918b47ff10eaaed2312cb07b44537b7f277fd255', + 'chromium_revision': '09b6d029916e7d3478d82e48819f0e9b73ac301d', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,24 +73,24 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@5238e7ee7167a8f9f4b7b564720f0361d7f24e93', + 'https://chromium.googlesource.com/chromium/src/base@ad8012919ba72154284cb291a643602ec9d01d65', 'src/build': 'https://chromium.googlesource.com/chromium/src/build@69dc3b19a1c8707994411ae29dcaddf3330965b3', 'src/buildtools': - 'https://chromium.googlesource.com/chromium/src/buildtools@cd169314af80e77a75eb25e2c0df8798f35bc04a', + 'https://chromium.googlesource.com/chromium/src/buildtools@813bee86eeb7d00fa170b81e221120576a18517a', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. 'src/examples/androidtests/third_party/gradle': { 'url': 'https://chromium.googlesource.com/external/github.com/gradle/gradle.git@f2d1fb54a951d8b11d25748e4711bec8d128d7e3', 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@1c44c20c546a76f233eb494a888a10a9f1f42dcb', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@b46a37e0ed529b9bdd2413f3ebc3724ad3963ec4', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@6a20ae4677a3ee6ed783b576a8586f2566356d22', + 'https://chromium.googlesource.com/chromium/src/testing@a1165ea8b0d4aa95bb11659a49351cce045217e8', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@6db51323482ae97a2eb560c1e71c257fad364907', + 'https://chromium.googlesource.com/chromium/src/third_party@e68e9e578676278ceab90de4b13c624a7f989ad3', 'src/buildtools/linux64': { 'packages': [ @@ -511,7 +511,7 @@ deps = { 'condition': 'checkout_linux', }, 'src/third_party/freetype/src': - 'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@013adc82b953d8462a24e4060db41dce548ac071', + 'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@42a649be4929444f4c42763ac8704a3c2f22f9f7', 'src/third_party/harfbuzz-ng/src': 'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@9f83bbbe64654b45ba5bb06927ff36c2e7588495', 'src/third_party/google_benchmark/src': { @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'c10f213cb474e756222a165a75e8fa008e3c769b', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '04c0a05c24c9b6c27a8e01a71e996c146f85dffd', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + '28bf5df73ef2f345a936d9cc95d64ba8ed426a53', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@7299b0e228bde04946e912d997d517f43b167e48', + 'https://chromium.googlesource.com/chromium/src/tools@b4b877b178dbfce980161a7b6025592644443f4a', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'RPIZQSHBlkhsLX97i1zrh3lwBoksi4vxanP-y5OVJM8C', + 'version': 'pTEOsaDcd32WtfN-pVyjazgeRGUSf2p_vv-4DtFKR_IC', }, ], 'condition': 'checkout_android and non_git_source', From 9284fbd0ad2401f067f1110c1eca59df0b92cb96 Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Thu, 22 May 2025 13:07:54 +0200 Subject: [PATCH 260/292] Cleanup arraysize usage in call and p2p/ Bug: webrtc:407433458 Change-Id: Ib3f5eff97f6cbeebbd5cf119d999712be92b2d9e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393120 Reviewed-by: Per Kjellander Commit-Queue: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#44732} --- call/rtp_demuxer_unittest.cc | 25 +++++++++++----------- call/rtp_payload_params.cc | 4 ++-- p2p/BUILD.gn | 3 --- p2p/base/port_unittest.cc | 4 +--- p2p/base/pseudo_tcp.cc | 2 +- p2p/base/transport_description.cc | 3 +-- p2p/test/nat_server.cc | 15 +++++-------- p2p/test/nat_socket_factory.cc | 35 +++++++++++++++---------------- p2p/test/nat_socket_factory.h | 5 ++--- 9 files changed, 41 insertions(+), 55 deletions(-) diff --git a/call/rtp_demuxer_unittest.cc b/call/rtp_demuxer_unittest.cc index e2702267b4..5cbc1bc641 100644 --- a/call/rtp_demuxer_unittest.cc +++ b/call/rtp_demuxer_unittest.cc @@ -20,7 +20,6 @@ #include "call/test/mock_rtp_packet_sink_interface.h" #include "modules/rtp_rtcp/source/rtp_header_extensions.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" -#include "rtc_base/arraysize.h" #include "rtc_base/checks.h" #include "rtc_base/numerics/safe_conversions.h" #include "test/gmock.h" @@ -290,12 +289,12 @@ TEST_F(RtpDemuxerTest, DISABLED_RejectAddSinkForSamePayloadTypes) { TEST_F(RtpDemuxerTest, OnRtpPacketCalledOnCorrectSinkBySsrc) { constexpr uint32_t ssrcs[] = {101, 202, 303}; - MockRtpPacketSink sinks[arraysize(ssrcs)]; - for (size_t i = 0; i < arraysize(ssrcs); i++) { + MockRtpPacketSink sinks[std::size(ssrcs)]; + for (size_t i = 0; i < std::size(ssrcs); i++) { AddSinkOnlySsrc(ssrcs[i], &sinks[i]); } - for (size_t i = 0; i < arraysize(ssrcs); i++) { + for (size_t i = 0; i < std::size(ssrcs); i++) { auto packet = CreatePacketWithSsrc(ssrcs[i]); EXPECT_CALL(sinks[i], OnRtpPacket(SamePacketAs(*packet))).Times(1); EXPECT_TRUE(demuxer_.OnRtpPacket(*packet)); @@ -304,12 +303,12 @@ TEST_F(RtpDemuxerTest, OnRtpPacketCalledOnCorrectSinkBySsrc) { TEST_F(RtpDemuxerTest, OnRtpPacketCalledOnCorrectSinkByRsid) { const std::string rsids[] = {"a", "b", "c"}; - MockRtpPacketSink sinks[arraysize(rsids)]; - for (size_t i = 0; i < arraysize(rsids); i++) { + MockRtpPacketSink sinks[std::size(rsids)]; + for (size_t i = 0; i < std::size(rsids); i++) { AddSinkOnlyRsid(rsids[i], &sinks[i]); } - for (size_t i = 0; i < arraysize(rsids); i++) { + for (size_t i = 0; i < std::size(rsids); i++) { auto packet = CreatePacketWithSsrcRsid(checked_cast(i), rsids[i]); EXPECT_CALL(sinks[i], OnRtpPacket(SamePacketAs(*packet))).Times(1); EXPECT_TRUE(demuxer_.OnRtpPacket(*packet)); @@ -318,12 +317,12 @@ TEST_F(RtpDemuxerTest, OnRtpPacketCalledOnCorrectSinkByRsid) { TEST_F(RtpDemuxerTest, OnRtpPacketCalledOnCorrectSinkByMid) { const std::string mids[] = {"a", "v", "s"}; - MockRtpPacketSink sinks[arraysize(mids)]; - for (size_t i = 0; i < arraysize(mids); i++) { + MockRtpPacketSink sinks[std::size(mids)]; + for (size_t i = 0; i < std::size(mids); i++) { AddSinkOnlyMid(mids[i], &sinks[i]); } - for (size_t i = 0; i < arraysize(mids); i++) { + for (size_t i = 0; i < std::size(mids); i++) { auto packet = CreatePacketWithSsrcMid(checked_cast(i), mids[i]); EXPECT_CALL(sinks[i], OnRtpPacket(SamePacketAs(*packet))).Times(1); EXPECT_TRUE(demuxer_.OnRtpPacket(*packet)); @@ -376,7 +375,7 @@ TEST_F(RtpDemuxerTest, PacketsDeliveredInRightOrder) { AddSinkOnlySsrc(ssrc, &sink); std::unique_ptr packets[5]; - for (size_t i = 0; i < arraysize(packets); i++) { + for (size_t i = 0; i < std::size(packets); i++) { packets[i] = CreatePacketWithSsrc(ssrc); packets[i]->SetSequenceNumber(checked_cast(i)); } @@ -511,7 +510,7 @@ TEST_F(RtpDemuxerTest, RsidLearnedAndLaterPacketsDeliveredWithOnlySsrc) { std::unique_ptr packets[5]; constexpr uint32_t rsid_ssrc = 111; packets[0] = CreatePacketWithSsrcRsid(rsid_ssrc, rsid); - for (size_t i = 1; i < arraysize(packets); i++) { + for (size_t i = 1; i < std::size(packets); i++) { packets[i] = CreatePacketWithSsrc(rsid_ssrc); } @@ -663,7 +662,7 @@ TEST_F(RtpDemuxerTest, MultipleRsidsOnSameSink) { } InSequence sequence; - for (size_t i = 0; i < arraysize(rsids); i++) { + for (size_t i = 0; i < std::size(rsids); i++) { // Assign different SSRCs and sequence numbers to all packets. const uint32_t ssrc = 1000 + static_cast(i); const uint16_t sequence_number = 50 + static_cast(i); diff --git a/call/rtp_payload_params.cc b/call/rtp_payload_params.cc index 2941f6375c..a3151814c3 100644 --- a/call/rtp_payload_params.cc +++ b/call/rtp_payload_params.cc @@ -14,6 +14,7 @@ #include #include +#include #include #include "absl/container/inlined_vector.h" @@ -36,7 +37,6 @@ #include "modules/video_coding/codecs/vp9/include/vp9_globals.h" #include "modules/video_coding/frame_dependencies_calculator.h" #include "modules/video_coding/include/video_codec_interface.h" -#include "rtc_base/arraysize.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/random.h" @@ -789,7 +789,7 @@ void RtpPayloadParams::SetDependenciesVp8New( RTC_DCHECK_GT(vp8_info.referencedBuffersCount, 0u); RTC_DCHECK_LE(vp8_info.referencedBuffersCount, - arraysize(vp8_info.referencedBuffers)); + std::size(vp8_info.referencedBuffers)); for (size_t i = 0; i < vp8_info.referencedBuffersCount; ++i) { const size_t referenced_buffer = vp8_info.referencedBuffers[i]; diff --git a/p2p/BUILD.gn b/p2p/BUILD.gn index 6971391435..69d876d957 100644 --- a/p2p/BUILD.gn +++ b/p2p/BUILD.gn @@ -847,7 +847,6 @@ rtc_library("transport_description") { ":p2p_constants", "../api:rtc_error", "../rtc_base:logging", - "../rtc_base:macromagic", "../rtc_base:ssl", "../rtc_base:stringutils", "../rtc_base/system:rtc_export", @@ -1127,7 +1126,6 @@ if (rtc_include_tests) { "../rtc_base:gunit_helpers", "../rtc_base:ip_address", "../rtc_base:logging", - "../rtc_base:macromagic", "../rtc_base:net_helpers", "../rtc_base:network_route", "../rtc_base:rtc_base_tests_utils", @@ -1255,7 +1253,6 @@ if (rtc_include_tests) { "../rtc_base:gunit_helpers", "../rtc_base:ip_address", "../rtc_base:logging", - "../rtc_base:macromagic", "../rtc_base:mdns_responder_interface", "../rtc_base:net_helper", "../rtc_base:net_helpers", diff --git a/p2p/base/port_unittest.cc b/p2p/base/port_unittest.cc index 72db2f8ca1..5bf4b43347 100644 --- a/p2p/base/port_unittest.cc +++ b/p2p/base/port_unittest.cc @@ -51,8 +51,6 @@ #include "p2p/test/stun_server.h" #include "p2p/test/test_stun_server.h" #include "p2p/test/test_turn_server.h" -#include "p2p/test/turn_server.h" -#include "rtc_base/arraysize.h" #include "rtc_base/async_packet_socket.h" #include "rtc_base/buffer.h" #include "rtc_base/byte_buffer.h" @@ -3178,7 +3176,7 @@ TEST_F(PortTest, TestWritableState) { // Data should be sendable before the connection is accepted. char data[] = "abcd"; - int data_size = arraysize(data); + int data_size = std::ssize(data); AsyncSocketPacketOptions options; EXPECT_EQ(data_size, ch1.conn()->Send(data, data_size, options)); diff --git a/p2p/base/pseudo_tcp.cc b/p2p/base/pseudo_tcp.cc index 1470fd0b36..8564afe9a0 100644 --- a/p2p/base/pseudo_tcp.cc +++ b/p2p/base/pseudo_tcp.cc @@ -188,7 +188,7 @@ void ReportStats() { char buffer[256]; size_t len = 0; for (int i = 0; i < S_NUM_STATS; ++i) { - len += snprintf(buffer, arraysize(buffer), "%s%s:%d", + len += snprintf(buffer, std::size(buffer), "%s%s:%d", (i == 0) ? "" : ",", STAT_NAMES[i], g_stats[i]); g_stats[i] = 0; } diff --git a/p2p/base/transport_description.cc b/p2p/base/transport_description.cc index 0a5621a16d..b6013cdc44 100644 --- a/p2p/base/transport_description.cc +++ b/p2p/base/transport_description.cc @@ -21,7 +21,6 @@ #include "absl/strings/string_view.h" #include "api/rtc_error.h" #include "p2p/base/p2p_constants.h" -#include "rtc_base/arraysize.h" #include "rtc_base/logging.h" #include "rtc_base/ssl_fingerprint.h" #include "rtc_base/strings/string_builder.h" @@ -123,7 +122,7 @@ std::optional StringToConnectionRole( CONNECTIONROLE_ACTIVE_STR, CONNECTIONROLE_PASSIVE_STR, CONNECTIONROLE_ACTPASS_STR, CONNECTIONROLE_HOLDCONN_STR}; - for (size_t i = 0; i < arraysize(roles); ++i) { + for (size_t i = 0; i < std::size(roles); ++i) { if (absl::EqualsIgnoreCase(roles[i], role_str)) { return static_cast(CONNECTIONROLE_ACTIVE + i); } diff --git a/p2p/test/nat_server.cc b/p2p/test/nat_server.cc index d1ab308ab8..7aa4007ba7 100644 --- a/p2p/test/nat_server.cc +++ b/p2p/test/nat_server.cc @@ -13,13 +13,13 @@ #include #include #include -#include #include "api/array_view.h" #include "p2p/test/nat_socket_factory.h" #include "p2p/test/nat_types.h" #include "rtc_base/async_packet_socket.h" #include "rtc_base/async_udp_socket.h" +#include "rtc_base/buffer.h" #include "rtc_base/checks.h" #include "rtc_base/ip_address.h" #include "rtc_base/logging.h" @@ -229,17 +229,12 @@ void NATServer::OnExternalUDPPacket(AsyncPacketSocket* socket, // Forward this packet to the internal address. // First prepend the address in a quasi-STUN format. - std::unique_ptr real_buf( - new char[packet.payload().size() + kNATEncodedIPv6AddressSize]); - size_t addrlength = PackAddressForNAT( - real_buf.get(), packet.payload().size() + kNATEncodedIPv6AddressSize, - packet.source_address()); + Buffer real_buf(packet.payload().size() + kNATEncodedIPv6AddressSize); + PackAddressForNAT(packet.source_address(), real_buf); // Copy the data part after the address. AsyncSocketPacketOptions options; - memcpy(real_buf.get() + addrlength, packet.payload().data(), - packet.payload().size()); - udp_server_socket_->SendTo(real_buf.get(), - packet.payload().size() + addrlength, + real_buf.AppendData(packet.payload()); + udp_server_socket_->SendTo(real_buf.data(), real_buf.size(), iter->second->route.source(), options); } diff --git a/p2p/test/nat_socket_factory.cc b/p2p/test/nat_socket_factory.cc index ba580c6020..a38b4bb8d9 100644 --- a/p2p/test/nat_socket_factory.cc +++ b/p2p/test/nat_socket_factory.cc @@ -21,7 +21,6 @@ #include "api/units/timestamp.h" #include "p2p/test/nat_server.h" #include "p2p/test/nat_types.h" -#include "rtc_base/arraysize.h" #include "rtc_base/buffer.h" #include "rtc_base/byte_order.h" #include "rtc_base/checks.h" @@ -41,9 +40,8 @@ namespace webrtc { // Packs the given socketaddress into the buffer in buf, in the quasi-STUN // format that the natserver uses. // Returns 0 if an invalid address is passed. -size_t PackAddressForNAT(char* buf, - size_t buf_size, - const SocketAddress& remote_addr) { +void PackAddressForNAT(const SocketAddress& remote_addr, Buffer& buf) { + RTC_DCHECK_GE(buf.capacity(), 4); const IPAddress& ip = remote_addr.ipaddr(); int family = ip.family(); buf[0] = 0; @@ -51,17 +49,18 @@ size_t PackAddressForNAT(char* buf, // Writes the port. *(reinterpret_cast(&buf[2])) = HostToNetwork16(remote_addr.port()); if (family == AF_INET) { - RTC_DCHECK(buf_size >= kNATEncodedIPv4AddressSize); + RTC_DCHECK_GE(buf.capacity(), kNATEncodedIPv4AddressSize); in_addr v4addr = ip.ipv4_address(); memcpy(&buf[4], &v4addr, kNATEncodedIPv4AddressSize - 4); - return kNATEncodedIPv4AddressSize; + buf.SetSize(kNATEncodedIPv4AddressSize); } else if (family == AF_INET6) { - RTC_DCHECK(buf_size >= kNATEncodedIPv6AddressSize); + RTC_DCHECK_GE(buf.capacity(), kNATEncodedIPv6AddressSize); in6_addr v6addr = ip.ipv6_address(); memcpy(&buf[4], &v6addr, kNATEncodedIPv6AddressSize - 4); - return kNATEncodedIPv6AddressSize; + buf.SetSize(kNATEncodedIPv6AddressSize); + } else { + buf.SetSize(0); } - return 0U; } // Decodes the remote address from a packet that has been encoded with the nat's @@ -152,12 +151,12 @@ class NATSocket : public Socket, public sigslot::has_slots<> { return socket_->SendTo(data, size, addr); } // This array will be too large for IPv4 packets, but only by 12 bytes. - std::unique_ptr buf(new char[size + kNATEncodedIPv6AddressSize]); - size_t addrlength = - PackAddressForNAT(buf.get(), size + kNATEncodedIPv6AddressSize, addr); - size_t encoded_size = size + addrlength; - memcpy(buf.get() + addrlength, data, size); - int result = socket_->SendTo(buf.get(), encoded_size, server_addr_); + Buffer buf(/*size=*/size + kNATEncodedIPv6AddressSize); + PackAddressForNAT(addr, buf); + size_t addrlength = buf.size(); + buf.AppendData(static_cast(data), size); + size_t encoded_size = buf.size(); + int result = socket_->SendTo(buf.data(), buf.size(), server_addr_); if (result >= 0) { RTC_DCHECK(result == static_cast(encoded_size)); result = result - static_cast(addrlength); @@ -298,9 +297,9 @@ class NATSocket : public Socket, public sigslot::has_slots<> { // Sends the destination address to the server to tell it to connect. void SendConnectRequest() { - char buf[kNATEncodedIPv6AddressSize]; - size_t length = PackAddressForNAT(buf, arraysize(buf), remote_addr_); - socket_->Send(buf, length); + Buffer buf(kNATEncodedIPv6AddressSize); + PackAddressForNAT(remote_addr_, buf); + socket_->Send(buf.data(), buf.size()); } // Handles the byte sent back from the server and fires the appropriate event. diff --git a/p2p/test/nat_socket_factory.h b/p2p/test/nat_socket_factory.h index d73cc99ea6..2da761dfb8 100644 --- a/p2p/test/nat_socket_factory.h +++ b/p2p/test/nat_socket_factory.h @@ -22,6 +22,7 @@ #include "api/units/time_delta.h" #include "p2p/test/nat_server.h" #include "p2p/test/nat_types.h" +#include "rtc_base/buffer.h" #include "rtc_base/socket.h" #include "rtc_base/socket_address.h" #include "rtc_base/socket_factory.h" @@ -172,9 +173,7 @@ class NATSocketServer : public SocketServer, public NATInternalSocketFactory { }; // Free-standing NAT helper functions. -size_t PackAddressForNAT(char* buf, - size_t buf_size, - const SocketAddress& remote_addr); +void PackAddressForNAT(const SocketAddress& remote_addr, Buffer& buf); size_t UnpackAddressFromNAT(ArrayView buf, SocketAddress* remote_addr); } // namespace webrtc From 0f1742e458d2b187a6bf4470352acb6d7387274e Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Thu, 22 May 2025 11:02:19 -0700 Subject: [PATCH 261/292] Roll chromium_revision 09b6d02991..9eb35d16c3 (1463976:1464233) Change log: https://chromium.googlesource.com/chromium/src/+log/09b6d02991..9eb35d16c3 Full diff: https://chromium.googlesource.com/chromium/src/+/09b6d02991..9eb35d16c3 Changed dependencies * fuchsia_version: version:28.20250515.3.1..version:28.20250522.3.1 * src/base: https://chromium.googlesource.com/chromium/src/base/+log/ad8012919b..6589a38594 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/69dc3b19a1..b2a7af3c71 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/b46a37e0ed..b1395fbaa7 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/a1165ea8b0..781771e2ac * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/e68e9e5786..7fc69a484b * src/third_party/androidx/cipd: pTEOsaDcd32WtfN-pVyjazgeRGUSf2p_vv-4DtFKR_IC..j7iNI3myTyI0EMBVQIEhOLRfEWiicoOLLXtRaela9s0C * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/f73c587323..d4e0bd4085 * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/b8cc89c389..d56eca0267 * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/04c0a05c24..d15aa8fb5b * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/b4b877b178..8ea8d0e620 DEPS diff: https://chromium.googlesource.com/chromium/src/+/09b6d02991..9eb35d16c3/DEPS BUG=None Change-Id: I6924d2e4ebe0335640fa77c259fa1df86152d326 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393240 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44733} --- DEPS | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/DEPS b/DEPS index 6ec65a223c..858833f736 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '09b6d029916e7d3478d82e48819f0e9b73ac301d', + 'chromium_revision': '9eb35d16c339fadb20cc668b0d4beb678536fd29', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -36,7 +36,7 @@ vars = { # By default, download the fuchsia sdk from the public sdk directory. 'fuchsia_sdk_cipd_prefix': 'fuchsia/sdk/core/', - 'fuchsia_version': 'version:28.20250515.3.1', + 'fuchsia_version': 'version:28.20250522.3.1', # By default, download the fuchsia images from the fuchsia GCS bucket. 'fuchsia_images_bucket': 'fuchsia', 'checkout_fuchsia': False, @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@ad8012919ba72154284cb291a643602ec9d01d65', + 'https://chromium.googlesource.com/chromium/src/base@6589a38594eb9f004f3e0d26d859adbc782297cc', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@69dc3b19a1c8707994411ae29dcaddf3330965b3', + 'https://chromium.googlesource.com/chromium/src/build@b2a7af3c7135440016cdaf8601702828dd668eab', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@813bee86eeb7d00fa170b81e221120576a18517a', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@b46a37e0ed529b9bdd2413f3ebc3724ad3963ec4', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@b1395fbaa7f1b1ad1fbd746aaf9f68e2ae030b1a', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@a1165ea8b0d4aa95bb11659a49351cce045217e8', + 'https://chromium.googlesource.com/chromium/src/testing@781771e2acbe0e7c8214529a9d602b0080503fef', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@e68e9e578676278ceab90de4b13c624a7f989ad3', + 'https://chromium.googlesource.com/chromium/src/third_party@7fc69a484b0bc8b630e4c780267d88948c21e25c', 'src/buildtools/linux64': { 'packages': [ @@ -497,13 +497,13 @@ deps = { 'src/third_party/crc32c/src': 'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558', 'src/third_party/depot_tools': - 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@f73c587323cae09c7b82f149b9d30c70218fff01', + 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@d4e0bd40852ad583ca00754142ff48b0f3bc2431', 'src/third_party/ffmpeg': 'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@dcdd0fa51b65a0b1688ff6b8f0cc81908f09ded2', 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@b8cc89c38984ba3e38ef812ebe8f581265e0136f', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@d56eca02673c8032dbdf1ac9706f8125ffec5d8a', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '04c0a05c24c9b6c27a8e01a71e996c146f85dffd', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'd15aa8fb5bed6a9e04f7f93ebdc3573b6f7a366e', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + '28bf5df73ef2f345a936d9cc95d64ba8ed426a53', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@b4b877b178dbfce980161a7b6025592644443f4a', + 'https://chromium.googlesource.com/chromium/src/tools@8ea8d0e6204ac3fd1a84f83816be0fc232afcb70', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'pTEOsaDcd32WtfN-pVyjazgeRGUSf2p_vv-4DtFKR_IC', + 'version': 'j7iNI3myTyI0EMBVQIEhOLRfEWiicoOLLXtRaela9s0C', }, ], 'condition': 'checkout_android and non_git_source', From 22a8c5cf42e5f8ed8ba650ad9b2fb13e6f4953c7 Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Thu, 22 May 2025 08:55:02 -0700 Subject: [PATCH 262/292] Notify dtls-stun-piggyback controller of DTLS failure which will stop it from embedded further DTLS packets or ACKs. Manually tested that the log for DTLS-STUN piggybacking not supported by peer. shows up when running rtc_p2p_unittests --gtest_filter="*.TestEventOrdering/59" TestEventOrdering/59 # GetParam() = ({ 2, 1, 3, 0 }, false, 3, true) Bug: webrtc:367395350 Change-Id: I2233abe0da2f2d29f5eded726fd614aaa75e3be2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391221 Reviewed-by: Jonas Oreland Reviewed-by: Elad Alon Reviewed-by: Sameer Vijaykar Commit-Queue: Philipp Hancke Cr-Commit-Position: refs/heads/main@{#44734} --- p2p/dtls/dtls_stun_piggyback_controller.cc | 11 +++++++++++ p2p/dtls/dtls_stun_piggyback_controller.h | 4 +++- p2p/dtls/dtls_transport.cc | 3 +++ p2p/dtls/dtls_transport_unittest.cc | 4 ++-- 4 files changed, 19 insertions(+), 3 deletions(-) diff --git a/p2p/dtls/dtls_stun_piggyback_controller.cc b/p2p/dtls/dtls_stun_piggyback_controller.cc index 5da36d48ad..4cc5ffc612 100644 --- a/p2p/dtls/dtls_stun_piggyback_controller.cc +++ b/p2p/dtls/dtls_stun_piggyback_controller.cc @@ -58,6 +58,17 @@ void DtlsStunPiggybackController::SetDtlsHandshakeComplete(bool is_dtls_client, state_ = State::PENDING; } +void DtlsStunPiggybackController::SetDtlsFailed() { + RTC_DCHECK_RUN_ON(&sequence_checker_); + + if (state_ == State::TENTATIVE || state_ == State::CONFIRMED || + state_ == State::PENDING) { + RTC_LOG(LS_INFO) + << "DTLS-STUN piggybacking DTLS failed during negotiation."; + } + state_ = State::OFF; +} + void DtlsStunPiggybackController::CapturePacket(ArrayView data) { RTC_DCHECK_RUN_ON(&sequence_checker_); if (!IsDtlsPacket(data)) { diff --git a/p2p/dtls/dtls_stun_piggyback_controller.h b/p2p/dtls/dtls_stun_piggyback_controller.h index 13a422d72e..9f9c3e9868 100644 --- a/p2p/dtls/dtls_stun_piggyback_controller.h +++ b/p2p/dtls/dtls_stun_piggyback_controller.h @@ -62,8 +62,10 @@ class DtlsStunPiggybackController { return state_; } - // Called by DtlsTransport when handshake is complete. + // Called by DtlsTransport when the handshake is complete. void SetDtlsHandshakeComplete(bool is_dtls_client, bool is_dtls13); + // Called by DtlsTransport when DTLS failed. + void SetDtlsFailed(); // Intercepts DTLS packets which should go into the STUN packets during the // handshake. diff --git a/p2p/dtls/dtls_transport.cc b/p2p/dtls/dtls_transport.cc index 0d8f263e11..bcbf7ec681 100644 --- a/p2p/dtls/dtls_transport.cc +++ b/p2p/dtls/dtls_transport.cc @@ -1032,6 +1032,9 @@ void DtlsTransportInternalImpl::set_dtls_state(DtlsTransportState state) { << static_cast(dtls_state_) << " to " << static_cast(state); dtls_state_ = state; + if (dtls_state_ == DtlsTransportState::kFailed) { + dtls_stun_piggyback_controller_.SetDtlsFailed(); + } SendDtlsState(this, state); } diff --git a/p2p/dtls/dtls_transport_unittest.cc b/p2p/dtls/dtls_transport_unittest.cc index d7a2f7fdbf..0fc2814016 100644 --- a/p2p/dtls/dtls_transport_unittest.cc +++ b/p2p/dtls/dtls_transport_unittest.cc @@ -1489,9 +1489,9 @@ INSTANTIATE_TEST_SUITE_P( std::vector{ CALLER_RECEIVES_CLIENTHELLO, CALLER_WRITABLE, HANDSHAKE_FINISHES, CALLER_RECEIVES_FINGERPRINT}), - ::testing::Bool(), + /*valid_fingerprint=*/::testing::Bool(), ::testing::Values(SSL_PROTOCOL_DTLS_12, SSL_PROTOCOL_DTLS_13), - ::testing::Bool())); + /*pqc=*/::testing::Bool())); class DtlsTransportInternalImplDtlsInStunTest : public DtlsTransportInternalImplVersionTest { From 59d578881fea163b8bd63aa056ed38feefd273de Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Thu, 22 May 2025 15:07:38 -0700 Subject: [PATCH 263/292] Roll chromium_revision 9eb35d16c3..8eb28e52ae (1464233:1464391) Change log: https://chromium.googlesource.com/chromium/src/+log/9eb35d16c3..8eb28e52ae Full diff: https://chromium.googlesource.com/chromium/src/+/9eb35d16c3..8eb28e52ae Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/6589a38594..b4035c6dd6 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/b2a7af3c71..a6102280a1 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/b1395fbaa7..e88df4a784 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/781771e2ac..d90b68398c * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/7fc69a484b..9724639de5 * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/3369878187..c5795a3288 * src/third_party/breakpad/breakpad: https://chromium.googlesource.com/breakpad/breakpad.git/+log/fad8247731..2625edb085 * src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/a060a8eafc..a6f7c1ae19 * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/d4e0bd4085..2ce4c87bda * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/d56eca0267..255938a7dd * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/8ea8d0e620..b6847763c0 DEPS diff: https://chromium.googlesource.com/chromium/src/+/9eb35d16c3..8eb28e52ae/DEPS BUG=None Change-Id: I6cd39ccf7baa756763e4ddcc772a2cfebc3cdf2e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393260 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44735} --- DEPS | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/DEPS b/DEPS index 858833f736..8406d2ad4a 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '9eb35d16c339fadb20cc668b0d4beb678536fd29', + 'chromium_revision': '8eb28e52aec4f7aeb861c2f186b7150bf6b3f082', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@6589a38594eb9f004f3e0d26d859adbc782297cc', + 'https://chromium.googlesource.com/chromium/src/base@b4035c6dd644584fb872b4178350d0bfac4fe214', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@b2a7af3c7135440016cdaf8601702828dd668eab', + 'https://chromium.googlesource.com/chromium/src/build@a6102280a183a0a89c1e24950ad93ed849ccb5d5', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@813bee86eeb7d00fa170b81e221120576a18517a', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@b1395fbaa7f1b1ad1fbd746aaf9f68e2ae030b1a', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@e88df4a784fc0a864361326bb49c47b376164d7e', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@781771e2acbe0e7c8214529a9d602b0080503fef', + 'https://chromium.googlesource.com/chromium/src/testing@d90b68398cee99b45f20b52015a6311dcdd4d65e', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@7fc69a484b0bc8b630e4c780267d88948c21e25c', + 'https://chromium.googlesource.com/chromium/src/third_party@9724639de5599c5c180020766a6f51fc8c8c6980', 'src/buildtools/linux64': { 'packages': [ @@ -323,7 +323,7 @@ deps = { 'src/third_party/clang-format/script': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@37f6e68a107df43b7d7e044fd36a13cbae3413f2', 'src/third_party/compiler-rt/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@a060a8eafc4db8a156e34820b3add404f5a9c16b', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@a6f7c1ae19c301452a66fdfeceef4d7c47d86107', 'src/third_party/libc++/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@a01c02c9d4acbdae3b7e8a2f3ee58579a9c29f96', 'src/third_party/libc++abi/src': @@ -480,9 +480,9 @@ deps = { }, 'src/third_party/boringssl/src': - 'https://boringssl.googlesource.com/boringssl.git@33698781873bab6685da95cde0a821bcd2e17302', + 'https://boringssl.googlesource.com/boringssl.git@c5795a328880eadb879088fc59cd0087edc8dc20', 'src/third_party/breakpad/breakpad': - 'https://chromium.googlesource.com/breakpad/breakpad.git@fad82477312f786994151f01c352813fabf924c0', + 'https://chromium.googlesource.com/breakpad/breakpad.git@2625edb085169e92cf036c236ac79ab594a7b1cc', 'src/third_party/catapult': 'https://chromium.googlesource.com/catapult.git@938fc9953b4159851a52022b4f2a90cddedba2fc', 'src/third_party/ced/src': { @@ -497,13 +497,13 @@ deps = { 'src/third_party/crc32c/src': 'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558', 'src/third_party/depot_tools': - 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@d4e0bd40852ad583ca00754142ff48b0f3bc2431', + 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@2ce4c87bda23e5188f218c57571e347b35df8f26', 'src/third_party/ffmpeg': 'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@dcdd0fa51b65a0b1688ff6b8f0cc81908f09ded2', 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@d56eca02673c8032dbdf1ac9706f8125ffec5d8a', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@255938a7dd0d67c1909d9e3196522406e900a548', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@8ea8d0e6204ac3fd1a84f83816be0fc232afcb70', + 'https://chromium.googlesource.com/chromium/src/tools@b6847763c05423fb13b61af217ee0ba9c10c8508', 'src/third_party/espresso': { 'packages': [ From 23949a3bac65cb1a72c1be1dd7e11485966d6929 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Thu, 22 May 2025 19:10:54 -0700 Subject: [PATCH 264/292] Roll chromium_revision 8eb28e52ae..b6610e2a20 (1464391:1464515) Change log: https://chromium.googlesource.com/chromium/src/+log/8eb28e52ae..b6610e2a20 Full diff: https://chromium.googlesource.com/chromium/src/+/8eb28e52ae..b6610e2a20 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/b4035c6dd6..9b193b2b34 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/a6102280a1..b228bdd418 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/e88df4a784..75a6898270 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/d90b68398c..9276bb98bf * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/9724639de5..cb9dd5e734 * src/third_party/android_build_tools/lint/cipd: PaYB6553MH9GJfamUZLduJESRbN13Clv2N2beHR6IQAC..Rh_qNy2kyeA9GGIdsjnPMvQa1WpHOEoUBZqV9opGGgIC * src/third_party/android_build_tools/manifest_merger/cipd: 0L0N3_u2ypIxEEtf0k_l36g_2CykW2BNuZBXm-v5qAUC..i1CvLtWlkB9QDx0DL_52AZNLpuQc2d6MbpRsCbzgEtEC * src/third_party/androidx/cipd: j7iNI3myTyI0EMBVQIEhOLRfEWiicoOLLXtRaela9s0C..35sx6LkLBBzbBXpfWIfXVI8tvIsqtu2T6tWdqcT-g94C * src/third_party/googletest/src: https://chromium.googlesource.com/external/github.com/google/googletest.git/+log/16d4f8eff6..6aa03e6774 * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/255938a7dd..36b3a363a7 * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/d15aa8fb5b..fa368a66f7 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/b6847763c0..85b444a3f5 DEPS diff: https://chromium.googlesource.com/chromium/src/+/8eb28e52ae..b6610e2a20/DEPS BUG=None Change-Id: I75e1ef803faa9d2716a48854f91284028de9c5bb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393242 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44736} --- DEPS | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) diff --git a/DEPS b/DEPS index 8406d2ad4a..345a0b69de 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '8eb28e52aec4f7aeb861c2f186b7150bf6b3f082', + 'chromium_revision': 'b6610e2a2071e974f01757d30f4b761b2bda48ef', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@b4035c6dd644584fb872b4178350d0bfac4fe214', + 'https://chromium.googlesource.com/chromium/src/base@9b193b2b3460d87025ac76f2d2e39b275a749342', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@a6102280a183a0a89c1e24950ad93ed849ccb5d5', + 'https://chromium.googlesource.com/chromium/src/build@b228bdd41836c0fac86a729ef5ebbe3047999f7b', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@813bee86eeb7d00fa170b81e221120576a18517a', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@e88df4a784fc0a864361326bb49c47b376164d7e', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@75a6898270bef70bbf4291a9d33ae2bed20ece87', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@d90b68398cee99b45f20b52015a6311dcdd4d65e', + 'https://chromium.googlesource.com/chromium/src/testing@9276bb98bfd4252d56436616e1a0c9afe7cd1a55', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@9724639de5599c5c180020766a6f51fc8c8c6980', + 'https://chromium.googlesource.com/chromium/src/third_party@cb9dd5e73451cb0115ed24f0cc62e58c46fb088b', 'src/buildtools/linux64': { 'packages': [ @@ -449,7 +449,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_build_tools/lint', - 'version': 'PaYB6553MH9GJfamUZLduJESRbN13Clv2N2beHR6IQAC', + 'version': 'Rh_qNy2kyeA9GGIdsjnPMvQa1WpHOEoUBZqV9opGGgIC', }, ], 'condition': 'checkout_android and non_git_source', @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@255938a7dd0d67c1909d9e3196522406e900a548', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@36b3a363a7e8ff99798d12efff4dce55f7f8ed85', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -525,7 +525,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/googletest/src': - 'https://chromium.googlesource.com/external/github.com/google/googletest.git@16d4f8eff6d7cefca6975d82a53f8fc995a6feb7', + 'https://chromium.googlesource.com/external/github.com/google/googletest.git@6aa03e6774f8cb70da277c56efb24b44ce29d8d7', 'src/third_party/icu': { 'url': 'https://chromium.googlesource.com/chromium/deps/icu.git@b929596baebf0ab4ac7ec07f38365db4c50a559d', }, @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'd15aa8fb5bed6a9e04f7f93ebdc3573b6f7a366e', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'fa368a66f714203d1232ff62ece403bd2f5e7c10', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + '28bf5df73ef2f345a936d9cc95d64ba8ed426a53', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@b6847763c05423fb13b61af217ee0ba9c10c8508', + 'https://chromium.googlesource.com/chromium/src/tools@85b444a3f5e8f8817164490b52b050ebb90d2018', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'j7iNI3myTyI0EMBVQIEhOLRfEWiicoOLLXtRaela9s0C', + 'version': '35sx6LkLBBzbBXpfWIfXVI8tvIsqtu2T6tWdqcT-g94C', }, ], 'condition': 'checkout_android and non_git_source', @@ -706,7 +706,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/android_build_tools/manifest_merger', - 'version': '0L0N3_u2ypIxEEtf0k_l36g_2CykW2BNuZBXm-v5qAUC', + 'version': 'i1CvLtWlkB9QDx0DL_52AZNLpuQc2d6MbpRsCbzgEtEC', }, ], 'condition': 'checkout_android', From e4e818a13258085dd36fcfab2cc31959028265b8 Mon Sep 17 00:00:00 2001 From: webrtc-version-updater Date: Thu, 22 May 2025 21:04:33 -0700 Subject: [PATCH 265/292] Update WebRTC code version (2025-05-23T04:04:31). Bug: None Change-Id: Id01e02e22ec573775b97b4c1da31f76d839a97b7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393262 Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Cr-Commit-Position: refs/heads/main@{#44737} --- call/version.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/call/version.cc b/call/version.cc index 04ef9a8d93..511e4b882e 100644 --- a/call/version.cc +++ b/call/version.cc @@ -13,7 +13,7 @@ namespace webrtc { // The timestamp is always in UTC. -const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-21T04:06:33"; +const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-23T04:04:31"; void LoadWebRTCVersionInRegister() { // Using volatile to instruct the compiler to not optimize `p` away even From 8b8874ac6a92d31eed8cc40db852b2be4129c188 Mon Sep 17 00:00:00 2001 From: Tomas Gunnarsson Date: Thu, 22 May 2025 09:10:22 -0700 Subject: [PATCH 266/292] Revert "[ResamplerHelper] Avoid resampling muted audio frames." MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This reverts commit cad50b5a1445a344dffaae0f57d5fbde95910b98. Reason for revert: This might introduce unwanted side effects and also does not seem to be worth it for the most common case. Bug: none Original change's description: > [ResamplerHelper] Avoid resampling muted audio frames. > > Bug: none > Change-Id: I1787b5ea6b9b33a0d37992287a4adc287fbb2ce5 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392662 > Commit-Queue: Tomas Gunnarsson > Reviewed-by: Jakob Ivarsson‎ > Cr-Commit-Position: refs/heads/main@{#44729} Bug: none No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I8ff3fc77aab4c25f5e76b23946b8a3c2f2edc47f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393201 Commit-Queue: Jakob Ivarsson‎ Auto-Submit: Tomas Gunnarsson Reviewed-by: Jakob Ivarsson‎ Cr-Commit-Position: refs/heads/main@{#44738} --- modules/audio_coding/acm2/acm_resampler.cc | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/modules/audio_coding/acm2/acm_resampler.cc b/modules/audio_coding/acm2/acm_resampler.cc index 7d42d8fa1d..9daa04d14b 100644 --- a/modules/audio_coding/acm2/acm_resampler.cc +++ b/modules/audio_coding/acm2/acm_resampler.cc @@ -58,10 +58,8 @@ bool ResamplerHelper::MaybeResample(int desired_sample_rate_hz, audio_frame->SetSampleRateAndChannelSize(desired_sample_rate_hz); InterleavedView dst = audio_frame->mutable_data( audio_frame->samples_per_channel(), audio_frame->num_channels()); - // Muted frames will already have the audio buffer zeroed out. - if (!audio_frame->muted()) { - resampler_.Resample(src, dst); - } + // TODO(tommi): Don't resample muted audio frames. + resampler_.Resample(src, dst); resampled_last_output_frame_ = true; } else { resampled_last_output_frame_ = false; From 737e0fd10666b6142a9e1717bd7af06a478156e9 Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Thu, 22 May 2025 17:56:55 +0200 Subject: [PATCH 267/292] Cleanup reference to WebRTC-Audio-FecAdaptation field trial MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Field trial itself was already cleaned up from the code Bug: webrtc:42233254 Change-Id: I5a8c842eea84f1e210c38aa9bc7f5b7deccab310 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393200 Reviewed-by: Jakob Ivarsson‎ Commit-Queue: Jakob Ivarsson‎ Auto-Submit: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#44739} --- experiments/field_trials.py | 5 +---- modules/audio_coding/BUILD.gn | 1 - .../audio_network_adaptor_impl_unittest.cc | 5 ----- 3 files changed, 1 insertion(+), 10 deletions(-) diff --git a/experiments/field_trials.py b/experiments/field_trials.py index 1ccb44be20..ec83ca5615 100755 --- a/experiments/field_trials.py +++ b/experiments/field_trials.py @@ -499,9 +499,6 @@ def bug_url(self) -> str: FieldTrial('WebRTC-Audio-AlrProbing', 42220234, date(2024, 4, 1)), - FieldTrial('WebRTC-Audio-FecAdaptation', - 42233254, - date(2024, 4, 1)), FieldTrial('WebRTC-Audio-LegacyOverhead', 42221084, INDEFINITE), @@ -914,7 +911,7 @@ def bug_url(self) -> str: ]) # yapf: disable POLICY_EXEMPT_FIELD_TRIALS_DIGEST: str = \ - 'e10a1af33d0a4cf81a6dc87d963b82cf6092ada3' + '8e2640561e5ff33824da8ecf4f37b34eeaad92f3' REGISTERED_FIELD_TRIALS: FrozenSet[FieldTrial] = ACTIVE_FIELD_TRIALS.union( POLICY_EXEMPT_FIELD_TRIALS) diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn index 46c659c928..3a8d0db866 100644 --- a/modules/audio_coding/BUILD.gn +++ b/modules/audio_coding/BUILD.gn @@ -1568,7 +1568,6 @@ if (rtc_include_tests) { "../../test:audio_codec_mocks", "../../test:audio_test_common", "../../test:explicit_key_value_config", - "../../test:field_trial", "../../test:fileutils", "../../test:rtc_expect_death", "../../test:rtp_test_utils", diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc index ddabcfeba7..339b968512 100644 --- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc +++ b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc @@ -25,7 +25,6 @@ #include "modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h" #include "modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h" #include "rtc_base/fake_clock.h" -#include "test/field_trial.h" #include "test/gmock.h" #include "test/gtest.h" @@ -180,8 +179,6 @@ TEST(AudioNetworkAdaptorImplTest, TEST(AudioNetworkAdaptorImplTest, DumpEncoderRuntimeConfigIsCalledOnGetEncoderRuntimeConfig) { - test::ScopedFieldTrials override_field_trials( - "WebRTC-Audio-FecAdaptation/Enabled/"); ScopedFakeClock fake_clock; fake_clock.AdvanceTime(TimeDelta::Millis(kClockInitialTimeMs)); auto states = CreateAudioNetworkAdaptor(); @@ -252,8 +249,6 @@ TEST(AudioNetworkAdaptorImplTest, } TEST(AudioNetworkAdaptorImplTest, LogRuntimeConfigOnGetEncoderRuntimeConfig) { - test::ScopedFieldTrials override_field_trials( - "WebRTC-Audio-FecAdaptation/Enabled/"); auto states = CreateAudioNetworkAdaptor(); AudioEncoderRuntimeConfig config; From 36284511c8850c76bbbbf89ed10285eada8bf770 Mon Sep 17 00:00:00 2001 From: Evan Shrubsole Date: Wed, 23 Apr 2025 14:53:18 +0000 Subject: [PATCH 268/292] Reland "Default disallow deprecated namespaces in Chromium" This is a reland of commit e6fb7d69dad1760b424731a5947918d9b39c2288 Reland safe: Upstream is fixed. Original change's description: > Default disallow deprecated namespaces in Chromium > > Bug: webrtc:42232595 > Change-Id: Iff6bc474d2bd5d40bf9652b5d36440c2c993e930 > No-Iwyu: No relevant changes > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/388081 > Auto-Submit: Evan Shrubsole > Reviewed-by: Mirko Bonadei > Commit-Queue: Mirko Bonadei > Cr-Commit-Position: refs/heads/main@{#44445} Bug: webrtc:42232595 Change-Id: I985ca42b36f421902f0ad56a6c4457022128fd48 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389540 Commit-Queue: Mirko Bonadei Reviewed-by: Mirko Bonadei Cr-Commit-Position: refs/heads/main@{#44740} --- webrtc.gni | 1 + 1 file changed, 1 insertion(+) diff --git a/webrtc.gni b/webrtc.gni index 450476539c..103983dec4 100644 --- a/webrtc.gni +++ b/webrtc.gni @@ -384,6 +384,7 @@ if (is_mac || is_ios) { if (build_with_chromium) { rtc_use_perfetto = true + rtc_allow_deprecated_namespaces = false } # Global public configuration that should be applied to all WebRTC targets. You From 14ab61e4dfa3b73a3bebbd431a177c8f39dd4506 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Fri, 23 May 2025 03:03:05 -0700 Subject: [PATCH 269/292] Roll chromium_revision b6610e2a20..d85e0e6296 (1464515:1464635) Change log: https://chromium.googlesource.com/chromium/src/+log/b6610e2a20..d85e0e6296 Full diff: https://chromium.googlesource.com/chromium/src/+/b6610e2a20..d85e0e6296 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/9b193b2b34..50dc228e38 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/75a6898270..e23f79e492 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/9276bb98bf..f2c7b7b7f9 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/cb9dd5e734..101505f3a7 * src/third_party/androidx/cipd: 35sx6LkLBBzbBXpfWIfXVI8tvIsqtu2T6tWdqcT-g94C..Q86wYH3YCLvRC3zxlUGeKoryYeDIjW1gNT3ECAilwZUC * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/2ce4c87bda..e0ece52cfb * src/third_party/freetype/src: https://chromium.googlesource.com/chromium/src/third_party/freetype2.git/+log/42a649be49..5987a9f51d * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/36b3a363a7..6079d62bf9 * src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/378f15d1cb..2cca4aba03 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/85b444a3f5..c76ac7bd93 * src/tools/luci-go: git_revision:45d1c0a0168f06a2bbde9eca9a03087ed1da523e..git_revision:14c1dfd95e07b23eeaff304bd29a2eb80b06fe0b * src/tools/luci-go: git_revision:45d1c0a0168f06a2bbde9eca9a03087ed1da523e..git_revision:14c1dfd95e07b23eeaff304bd29a2eb80b06fe0b * src/tools/luci-go: git_revision:45d1c0a0168f06a2bbde9eca9a03087ed1da523e..git_revision:14c1dfd95e07b23eeaff304bd29a2eb80b06fe0b DEPS diff: https://chromium.googlesource.com/chromium/src/+/b6610e2a20..d85e0e6296/DEPS BUG=None Change-Id: I771f17088a9de8010e2dde24f6261e719651c9a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393282 Commit-Queue: Autoroller Bot-Commit: Autoroller Cr-Commit-Position: refs/heads/main@{#44741} --- DEPS | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/DEPS b/DEPS index 345a0b69de..1297061441 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': 'b6610e2a2071e974f01757d30f4b761b2bda48ef', + 'chromium_revision': 'd85e0e62960bdf9988ec18dc4b4e6e1897fa7664', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,7 +73,7 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@9b193b2b3460d87025ac76f2d2e39b275a749342', + 'https://chromium.googlesource.com/chromium/src/base@50dc228e3885dea149d3cee5e31f326c91c8649d', 'src/build': 'https://chromium.googlesource.com/chromium/src/build@b228bdd41836c0fac86a729ef5ebbe3047999f7b', 'src/buildtools': @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@75a6898270bef70bbf4291a9d33ae2bed20ece87', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@e23f79e492a90f81d91bdbfdfc9472c12f9d8ab7', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@9276bb98bfd4252d56436616e1a0c9afe7cd1a55', + 'https://chromium.googlesource.com/chromium/src/testing@f2c7b7b7f9477c00c9c05c6207427da6dfa39ac8', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@cb9dd5e73451cb0115ed24f0cc62e58c46fb088b', + 'https://chromium.googlesource.com/chromium/src/third_party@101505f3a724a53020ada0861e6672c67023de4f', 'src/buildtools/linux64': { 'packages': [ @@ -497,13 +497,13 @@ deps = { 'src/third_party/crc32c/src': 'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558', 'src/third_party/depot_tools': - 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@2ce4c87bda23e5188f218c57571e347b35df8f26', + 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@e0ece52cfb4faaf6d4468a94068265452ef6de13', 'src/third_party/ffmpeg': 'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@dcdd0fa51b65a0b1688ff6b8f0cc81908f09ded2', 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@36b3a363a7e8ff99798d12efff4dce55f7f8ed85', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@6079d62bf9b04cfd0390a07802f148d526d459a3', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -511,7 +511,7 @@ deps = { 'condition': 'checkout_linux', }, 'src/third_party/freetype/src': - 'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@42a649be4929444f4c42763ac8704a3c2f22f9f7', + 'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@5987a9f51d7002f7ca375a34c2b857841d82a91c', 'src/third_party/harfbuzz-ng/src': 'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@9f83bbbe64654b45ba5bb06927ff36c2e7588495', 'src/third_party/google_benchmark/src': { @@ -591,7 +591,7 @@ deps = { 'src/third_party/dav1d/libdav1d': 'https://chromium.googlesource.com/external/github.com/videolan/dav1d.git@8d956180934f16244bdb58b39175824775125e55', 'src/third_party/libaom/source/libaom': - 'https://aomedia.googlesource.com/aom.git@378f15d1cbd5102f89a52881bcdcf2c010f3c5c8', + 'https://aomedia.googlesource.com/aom.git@2cca4aba034f99842c2e6cdc173f83801d289764', 'src/third_party/libunwindstack': { 'url': 'https://chromium.googlesource.com/chromium/src/third_party/libunwindstack.git@0d758dd57f42564acecdd7a1e7ac5c8521c1b01a', 'condition': 'checkout_android', @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@85b444a3f5e8f8817164490b52b050ebb90d2018', + 'https://chromium.googlesource.com/chromium/src/tools@c76ac7bd937400b9ed8c1d7c64d60665159fc560', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': '35sx6LkLBBzbBXpfWIfXVI8tvIsqtu2T6tWdqcT-g94C', + 'version': 'Q86wYH3YCLvRC3zxlUGeKoryYeDIjW1gNT3ECAilwZUC', }, ], 'condition': 'checkout_android and non_git_source', @@ -793,15 +793,15 @@ deps = { 'packages': [ { 'package': 'infra/tools/luci/cas/${{platform}}', - 'version': 'git_revision:45d1c0a0168f06a2bbde9eca9a03087ed1da523e', + 'version': 'git_revision:14c1dfd95e07b23eeaff304bd29a2eb80b06fe0b', }, { 'package': 'infra/tools/luci/isolate/${{platform}}', - 'version': 'git_revision:45d1c0a0168f06a2bbde9eca9a03087ed1da523e', + 'version': 'git_revision:14c1dfd95e07b23eeaff304bd29a2eb80b06fe0b', }, { 'package': 'infra/tools/luci/swarming/${{platform}}', - 'version': 'git_revision:45d1c0a0168f06a2bbde9eca9a03087ed1da523e', + 'version': 'git_revision:14c1dfd95e07b23eeaff304bd29a2eb80b06fe0b', } ], 'dep_type': 'cipd', From aeac045cca685c72cf0522df14379a478c474ea6 Mon Sep 17 00:00:00 2001 From: Sergey Silkin Date: Fri, 23 May 2025 09:34:44 +0200 Subject: [PATCH 270/292] Disable "follow preferred order for codec fallback" by default MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The feature causes "no video" issue in M135+. See webrtc:416659453 for details. Bug: webrtc:416659453 Change-Id: Id7509b5e9c8f99c5914f74911878daf83459d5da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393180 Reviewed-by: Erik Språng Commit-Queue: Sergey Silkin Cr-Commit-Position: refs/heads/main@{#44742} --- media/engine/webrtc_video_engine_unittest.cc | 2 ++ video/video_stream_encoder.cc | 2 +- video/video_stream_encoder_unittest.cc | 11 ++++++----- 3 files changed, 9 insertions(+), 6 deletions(-) diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc index 31b9b4f455..596c555f15 100644 --- a/media/engine/webrtc_video_engine_unittest.cc +++ b/media/engine/webrtc_video_engine_unittest.cc @@ -2556,6 +2556,8 @@ TEST_F(WebRtcVideoChannelBaseTest, TwoStreamsSendAndReceive) { TEST_F(WebRtcVideoChannelBaseTest, RequestEncoderFallbackNextCodecFollowNegotiatedOrder) { + webrtc::test::ScopedKeyValueConfig field_trials( + field_trials_, "WebRTC-SwitchEncoderFollowCodecPreferenceOrder/Enabled/"); VideoSenderParameters parameters; parameters.codecs.push_back(GetEngineCodec("VP9")); parameters.codecs.push_back(GetEngineCodec("AV1")); diff --git a/video/video_stream_encoder.cc b/video/video_stream_encoder.cc index 212108ed06..53125ff373 100644 --- a/video/video_stream_encoder.cc +++ b/video/video_stream_encoder.cc @@ -1530,7 +1530,7 @@ void VideoStreamEncoder::RequestEncoderSwitch() { } if (!preferred_fallback_encoder) { - if (!env_.field_trials().IsDisabled( + if (env_.field_trials().IsEnabled( kSwitchEncoderFollowCodecPreferenceOrderFieldTrial)) { encoder_fallback_requested_ = true; settings_.encoder_switch_request_callback->RequestEncoderFallback(); diff --git a/video/video_stream_encoder_unittest.cc b/video/video_stream_encoder_unittest.cc index ea47223c53..bbae05d0df 100644 --- a/video/video_stream_encoder_unittest.cc +++ b/video/video_stream_encoder_unittest.cc @@ -8314,7 +8314,9 @@ TEST_F(VideoStreamEncoderTest, .WillByDefault(Return(WEBRTC_VIDEO_CODEC_ENCODER_FAILURE)); Event encode_attempted; - EXPECT_CALL(switch_callback, RequestEncoderFallback()) + EXPECT_CALL(switch_callback, + RequestEncoderSwitch(Field(&SdpVideoFormat::name, "VP8"), + /*allow_default_fallback=*/true)) .WillOnce([&encode_attempted]() { encode_attempted.Set(); }); video_source_.IncomingCapturedFrame(CreateFrame(1, nullptr)); @@ -8387,6 +8389,9 @@ TEST_F(VideoStreamEncoderTest, NoPreferenceDefaultFallbackToVP8Disabled) { constexpr int kDontCare = 100; constexpr int kNumFrames = 8; + webrtc::test::ScopedKeyValueConfig field_trials( + field_trials_, "WebRTC-SwitchEncoderFollowCodecPreferenceOrder/Enabled/"); + NiceMock video_encoder; StrictMock switch_callback; video_send_config_.encoder_settings.encoder_switch_request_callback = @@ -8444,10 +8449,6 @@ TEST_F(VideoStreamEncoderTest, NoPreferenceDefaultFallbackToVP8Enabled) { constexpr int kSufficientBitrateToNotDrop = 1000; constexpr int kDontCare = 100; - webrtc::test::ScopedKeyValueConfig field_trials( - field_trials_, - "WebRTC-SwitchEncoderFollowCodecPreferenceOrder/Disabled/"); - NiceMock video_encoder; StrictMock switch_callback; video_send_config_.encoder_settings.encoder_switch_request_callback = From 60342300228c4b8915e876e30a38865032e4b1af Mon Sep 17 00:00:00 2001 From: Per Kjellander Date: Fri, 23 May 2025 10:33:32 +0000 Subject: [PATCH 271/292] Allow padding packets to trigger probe MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Bug: webrtc:419715927, b/404742427 Change-Id: I0e1f6c77489d2e2eedd0918450199ec5307260a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393300 Commit-Queue: Per Kjellander Reviewed-by: Erik Språng Cr-Commit-Position: refs/heads/main@{#44743} --- modules/pacing/pacing_controller.cc | 2 +- modules/pacing/pacing_controller_unittest.cc | 48 +++++++++++++++++++- 2 files changed, 47 insertions(+), 3 deletions(-) diff --git a/modules/pacing/pacing_controller.cc b/modules/pacing/pacing_controller.cc index 7e1fb3c60a..d328c8692e 100644 --- a/modules/pacing/pacing_controller.cc +++ b/modules/pacing/pacing_controller.cc @@ -221,7 +221,7 @@ void PacingController::EnqueuePacket(std::unique_ptr packet) { } } - prober_.OnIncomingPacket(DataSize::Bytes(packet->payload_size())); + prober_.OnIncomingPacket(DataSize::Bytes(packet->size())); const Timestamp now = CurrentTime(); if (packet_queue_.Empty()) { diff --git a/modules/pacing/pacing_controller_unittest.cc b/modules/pacing/pacing_controller_unittest.cc index a51e7047c1..27fdbd1aba 100644 --- a/modules/pacing/pacing_controller_unittest.cc +++ b/modules/pacing/pacing_controller_unittest.cc @@ -66,13 +66,13 @@ std::unique_ptr BuildPacket(RtpPacketMediaType type, uint32_t ssrc, uint16_t sequence_number, int64_t capture_time_ms, - size_t size) { + size_t payload_size) { auto packet = std::make_unique(nullptr); packet->set_packet_type(type); packet->SetSsrc(ssrc); packet->SetSequenceNumber(sequence_number); packet->set_capture_time(Timestamp::Millis(capture_time_ms)); - packet->SetPayloadSize(size); + packet->SetPayloadSize(payload_size); return packet; } @@ -1367,6 +1367,50 @@ TEST_F(PacingControllerTest, ProbingWithPaddingSupport) { kFirstClusterRate.bps(), kProbingErrorMargin.bps()); } +TEST_F(PacingControllerTest, PaddingPacketCanTriggerProbe) { + const int kInitialBitrateBps = 300000; + PacingControllerProbing packet_sender; + auto pacer = + std::make_unique(&clock_, &packet_sender, trials_); + + pacer->SetPacingRates( + DataRate::BitsPerSec(kInitialBitrateBps * kPaceMultiplier), + /*padding_rate*/ DataRate::KilobitsPerSec(300)); + + pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, + /*ssrc=*/123, /*sequence_number=*/1, + clock_.TimeInMilliseconds(), + /*payload_size=*/50)); + + for (int i = 0; i < 5; ++i) { + AdvanceTimeUntil(pacer->NextSendTime()); + pacer->ProcessPackets(); + EXPECT_EQ(packet_sender.last_pacing_info().probe_cluster_id, + PacedPacketInfo::kNotAProbe); + } + ASSERT_GT(packet_sender.packets_sent(), 0); + ASSERT_GT(packet_sender.padding_sent(), 0); + + const int kProbeClusterId = 1; + std::vector probe_clusters = { + {.at_time = clock_.CurrentTime(), + .target_data_rate = DataRate::KilobitsPerSec(1000), + .target_duration = TimeDelta::Millis(15), + .target_probe_count = 5, + .id = kProbeClusterId}}; + pacer->CreateProbeClusters(probe_clusters); + bool probe_packet_seen = false; + for (int i = 0; i < 5; ++i) { + AdvanceTimeUntil(pacer->NextSendTime()); + pacer->ProcessPackets(); + if (packet_sender.last_pacing_info().probe_cluster_id == kProbeClusterId) { + probe_packet_seen = true; + break; + } + } + EXPECT_TRUE(probe_packet_seen); +} + TEST_F(PacingControllerTest, CanProbeWithPaddingBeforeFirstMediaPacket) { // const size_t kPacketSize = 1200; const int kInitialBitrateBps = 300000; From 4fd769cfb6ff91d2e4957c884c367cad1e35161f Mon Sep 17 00:00:00 2001 From: Tommi Date: Fri, 23 May 2025 12:37:39 +0200 Subject: [PATCH 272/292] Add support to DeInterleavedView for channel pointers MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This adds support for mapping distinct channel pointers into a view and addresses the incompatibility with AudioFrameView where we receive externally provided channel pointers. Mapping a single buffer for multiple channels continues to be supported, which is how some of the internal webrtc code manages multiple channels. The implementation does not do any heap allocations and relies on any buffers being externally owned. Bug: chromium:409777951 Change-Id: I9929e89884d1037c90bda1c63038af054c058c08 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392520 Commit-Queue: Tomas Gunnarsson Reviewed-by: Per Åhgren Cr-Commit-Position: refs/heads/main@{#44744} --- api/audio/audio_view.h | 71 +++++++++++++++---- api/audio/test/BUILD.gn | 1 + api/audio/test/audio_view_unittest.cc | 29 ++++++++ modules/audio_mixer/frame_combiner.cc | 15 ++-- modules/audio_processing/BUILD.gn | 6 +- modules/audio_processing/agc2/limiter.cc | 3 + modules/audio_processing/agc2/limiter.h | 7 +- modules/audio_processing/audio_buffer.h | 3 +- .../audio_processing_unittest.cc | 4 +- modules/audio_processing/gain_controller2.cc | 16 ++++- .../include/audio_frame_view.h | 3 +- 11 files changed, 125 insertions(+), 33 deletions(-) diff --git a/api/audio/audio_view.h b/api/audio/audio_view.h index c241b928b6..9c4396fd8e 100644 --- a/api/audio/audio_view.h +++ b/api/audio/audio_view.h @@ -13,6 +13,8 @@ #include #include +#include +#include #include "api/array_view.h" #include "rtc_base/checks.h" @@ -135,31 +137,65 @@ class DeinterleavedView { DeinterleavedView() = default; + // Construct a view where all the channels are coallocated in a single buffer. template DeinterleavedView(U* data, size_t samples_per_channel, size_t num_channels) : num_channels_(num_channels), samples_per_channel_(samples_per_channel), - data_(data, num_channels * samples_per_channel_) {} + data_(data) {} + // Construct a view from an array of channel pointers where the channels + // may all be allocated seperately. + template + DeinterleavedView(U* const* channels, + size_t samples_per_channel, + size_t num_channels) + : num_channels_(num_channels), + samples_per_channel_(samples_per_channel), + data_(channels) {} + + // Construct a view from an array of channel pointers where the pointers are + // helt in a `std::vector<>`. + template + DeinterleavedView(const std::vector& channels, size_t samples_per_channel) + : num_channels_(channels.size()), + samples_per_channel_(samples_per_channel), + data_(channels.data()) {} + + // Construct a view from another view. Note that the type of + // the other view may be different from the current type and + // therefore the internal data types may not be exactly the + // same, but still compatible. + // E.g.: + // DeinterleavedView mutable_view; + // DeinterleavedView const_view(mutable_view); template DeinterleavedView(const DeinterleavedView& other) - : num_channels_(other.num_channels()), - samples_per_channel_(other.samples_per_channel()), - data_(other.data()) {} + : num_channels_(other.num_channels_), + samples_per_channel_(other.samples_per_channel()) { + if (other.is_ptr_array()) { + data_ = std::get(other.data_); + } else { + data_ = std::get(other.data_); + } + } // Returns a deinterleaved channel where `idx` is the zero based index, // in the range [0 .. num_channels()-1]. MonoView operator[](size_t idx) const { - RTC_DCHECK_LT(idx, num_channels_); - return MonoView(&data_[idx * samples_per_channel_], + RTC_DCHECK_LT(idx, num_channels()); + if (is_ptr_array()) + return MonoView(std::get(data_)[idx], samples_per_channel_); + return MonoView(&std::get(data_)[idx * samples_per_channel_], samples_per_channel_); } size_t num_channels() const { return num_channels_; } size_t samples_per_channel() const { return samples_per_channel_; } - ArrayView data() const { return data_; } - bool empty() const { return data_.empty(); } - size_t size() const { return data_.size(); } + bool empty() const { + return num_channels_ == 0u || samples_per_channel_ == 0u; + } + size_t size() const { return num_channels_ * samples_per_channel_; } // Returns the first (and possibly only) channel. MonoView AsMono() const { @@ -167,12 +203,23 @@ class DeinterleavedView { return (*this)[0]; } + // Zeros out all samples in channels represented by the view. + void Clear() { + for (size_t i = 0u; i < num_channels_; ++i) { + MonoView view = (*this)[i]; + ClearSamples(view); + } + } + private: - // TODO(tommi): Consider having these be stored as uint16_t to save a few - // bytes per view. Use `dchecked_cast` to support size_t during construction. + bool is_ptr_array() const { return std::holds_alternative(data_); } + + template + friend class DeinterleavedView; + size_t num_channels_ = 0u; size_t samples_per_channel_ = 0u; - ArrayView data_; + std::variant data_; }; template diff --git a/api/audio/test/BUILD.gn b/api/audio/test/BUILD.gn index b1060faae3..73762fc54a 100644 --- a/api/audio/test/BUILD.gn +++ b/api/audio/test/BUILD.gn @@ -26,6 +26,7 @@ if (rtc_include_tests) { "../..:array_view", "../../../modules/audio_processing:aec3_config_json", "../../../rtc_base:checks", + "../../../rtc_base:macromagic", "../../../test:test_support", ] } diff --git a/api/audio/test/audio_view_unittest.cc b/api/audio/test/audio_view_unittest.cc index 1d3f5f17b5..55e9d87f63 100644 --- a/api/audio/test/audio_view_unittest.cc +++ b/api/audio/test/audio_view_unittest.cc @@ -13,8 +13,10 @@ #include #include #include +#include #include "api/array_view.h" +#include "rtc_base/arraysize.h" #include "test/gtest.h" namespace webrtc { @@ -190,4 +192,31 @@ TEST(AudioViewTest, ClearSamples) { } } } + +TEST(AudioViewTest, DeinterleavedViewPointerArray) { + // Create vectors of varying sizes to guarantee that they don't end up + // aligned in memory. + std::vector v1(100), v2(200), v3(300), v4(400); + std::vector channels = {&v1[0], &v2[0], &v3[0], &v4[0]}; + + DeinterleavedView di(channels, v1.size()); + EXPECT_EQ(NumChannels(di), channels.size()); + EXPECT_EQ(SamplesPerChannel(di), v1.size()); + EXPECT_EQ(di[0].data(), v1.data()); + EXPECT_EQ(di[1].data(), v2.data()); + EXPECT_EQ(di[2].data(), v3.data()); + EXPECT_EQ(di[3].data(), v4.data()); + + // Test that the same thing works with T* const *. + float* channel_array[] = {&v1[0], &v2[0], &v3[0], &v4[0]}; + di = DeinterleavedView(channel_array, v1.size(), + arraysize(channel_array)); + EXPECT_EQ(NumChannels(di), channels.size()); + EXPECT_EQ(SamplesPerChannel(di), v1.size()); + EXPECT_EQ(di[0].data(), v1.data()); + EXPECT_EQ(di[1].data(), v2.data()); + EXPECT_EQ(di[2].data(), v3.data()); + EXPECT_EQ(di[3].data(), v4.data()); +} + } // namespace webrtc diff --git a/modules/audio_mixer/frame_combiner.cc b/modules/audio_mixer/frame_combiner.cc index b801ca7f8e..e9d5f0c9cc 100644 --- a/modules/audio_mixer/frame_combiner.cc +++ b/modules/audio_mixer/frame_combiner.cc @@ -12,26 +12,22 @@ #include #include +#include #include -#include #include -#include #include #include #include "api/array_view.h" -#include "api/audio/audio_processing.h" +#include "api/audio/audio_frame.h" +#include "api/audio/audio_view.h" #include "api/rtp_packet_info.h" #include "api/rtp_packet_infos.h" #include "common_audio/include/audio_util.h" #include "modules/audio_mixer/audio_frame_manipulator.h" -#include "modules/audio_mixer/audio_mixer_impl.h" -#include "modules/audio_processing/include/audio_frame_view.h" +#include "modules/audio_processing/agc2/limiter.h" #include "modules/audio_processing/logging/apm_data_dumper.h" -#include "rtc_base/arraysize.h" #include "rtc_base/checks.h" -#include "rtc_base/numerics/safe_conversions.h" -#include "system_wrappers/include/metrics.h" namespace webrtc { namespace { @@ -90,8 +86,7 @@ void MixToFloatFrame(ArrayView mix_list, DeinterleavedView& mixing_buffer) { const size_t number_of_channels = NumChannels(mixing_buffer); // Clear the mixing buffer. - ArrayView raw_data = mixing_buffer.data(); - ClearSamples(raw_data); + mixing_buffer.Clear(); // Convert to FloatS16 and mix. for (size_t i = 0; i < mix_list.size(); ++i) { diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn index eac4a66376..2d92bc5178 100644 --- a/modules/audio_processing/BUILD.gn +++ b/modules/audio_processing/BUILD.gn @@ -61,6 +61,7 @@ rtc_library("audio_buffer") { "../../common_audio", "../../common_audio:common_audio_c", "../../rtc_base:checks", + "../../rtc_base:gtest_prod", ] } @@ -271,7 +272,10 @@ rtc_library("audio_processing_statistics") { rtc_source_set("audio_frame_view") { sources = [ "include/audio_frame_view.h" ] - deps = [ "../../api/audio:audio_frame_api" ] + deps = [ + "../../api/audio:audio_frame_api", + "../../rtc_base:checks", + ] } if (rtc_enable_protobuf) { diff --git a/modules/audio_processing/agc2/limiter.cc b/modules/audio_processing/agc2/limiter.cc index 072b79ed48..c4d594ae40 100644 --- a/modules/audio_processing/agc2/limiter.cc +++ b/modules/audio_processing/agc2/limiter.cc @@ -13,10 +13,13 @@ #include #include #include +#include #include "absl/strings/string_view.h" #include "api/array_view.h" +#include "api/audio/audio_view.h" #include "modules/audio_processing/agc2/agc2_common.h" +#include "modules/audio_processing/agc2/interpolated_gain_curve.h" #include "modules/audio_processing/logging/apm_data_dumper.h" #include "rtc_base/checks.h" #include "rtc_base/numerics/safe_conversions.h" diff --git a/modules/audio_processing/agc2/limiter.h b/modules/audio_processing/agc2/limiter.h index 55cb1a5b15..f3e4a54179 100644 --- a/modules/audio_processing/agc2/limiter.h +++ b/modules/audio_processing/agc2/limiter.h @@ -11,13 +11,14 @@ #ifndef MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_ #define MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_ -#include +#include +#include #include "absl/strings/string_view.h" -#include "api/audio/audio_frame.h" +#include "api/audio/audio_view.h" +#include "modules/audio_processing/agc2/agc2_common.h" #include "modules/audio_processing/agc2/fixed_digital_level_estimator.h" #include "modules/audio_processing/agc2/interpolated_gain_curve.h" -#include "modules/audio_processing/include/audio_frame_view.h" namespace webrtc { class ApmDataDumper; diff --git a/modules/audio_processing/audio_buffer.h b/modules/audio_processing/audio_buffer.h index 9369572af8..a876aa8741 100644 --- a/modules/audio_processing/audio_buffer.h +++ b/modules/audio_processing/audio_buffer.h @@ -21,6 +21,7 @@ #include "api/audio/audio_view.h" #include "common_audio/channel_buffer.h" #include "common_audio/include/audio_util.h" +#include "rtc_base/gtest_prod_util.h" namespace webrtc { @@ -62,7 +63,7 @@ class AudioBuffer { // Returns a DeinterleavedView<> over the channel data. DeinterleavedView view() { return DeinterleavedView( - num_channels_ && buffer_num_frames_ ? channels()[0] : nullptr, + num_channels_ && buffer_num_frames_ ? channels() : nullptr, buffer_num_frames_, num_channels_); } diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc index 86d049ee63..1f4a9eda1a 100644 --- a/modules/audio_processing/audio_processing_unittest.cc +++ b/modules/audio_processing/audio_processing_unittest.cc @@ -2031,7 +2031,7 @@ class AudioProcessingTest // Dump forward output to file. RTC_DCHECK_EQ(out_cb.num_bands(), 1u); // Assumes full frequency band. DeinterleavedView deinterleaved_src( - out_cb.channels()[0], out_cb.num_frames(), out_cb.num_channels()); + out_cb.channels(), out_cb.num_frames(), out_cb.num_channels()); InterleavedView interleaved_dst( float_data.get(), out_cb.num_frames(), out_cb.num_channels()); Interleave(deinterleaved_src, interleaved_dst); @@ -2043,7 +2043,7 @@ class AudioProcessingTest // Dump reverse output to file. RTC_DCHECK_EQ(rev_out_cb.num_bands(), 1u); deinterleaved_src = DeinterleavedView( - rev_out_cb.channels()[0], rev_out_cb.num_frames(), + rev_out_cb.channels(), rev_out_cb.num_frames(), rev_out_cb.num_channels()); interleaved_dst = InterleavedView( float_data.get(), rev_out_cb.num_frames(), rev_out_cb.num_channels()); diff --git a/modules/audio_processing/gain_controller2.cc b/modules/audio_processing/gain_controller2.cc index 6e4a251456..8704d7bee1 100644 --- a/modules/audio_processing/gain_controller2.cc +++ b/modules/audio_processing/gain_controller2.cc @@ -10,21 +10,31 @@ #include "modules/audio_processing/gain_controller2.h" +#include +#include +#include #include -#include +#include #include "api/audio/audio_frame.h" +#include "api/audio/audio_processing.h" +#include "api/audio/audio_view.h" #include "api/environment/environment.h" #include "api/field_trials_view.h" #include "common_audio/include/audio_util.h" +#include "modules/audio_processing/agc2/adaptive_digital_gain_controller.h" #include "modules/audio_processing/agc2/agc2_common.h" #include "modules/audio_processing/agc2/cpu_features.h" +#include "modules/audio_processing/agc2/input_volume_controller.h" +#include "modules/audio_processing/agc2/interpolated_gain_curve.h" +#include "modules/audio_processing/agc2/noise_level_estimator.h" +#include "modules/audio_processing/agc2/saturation_protector.h" +#include "modules/audio_processing/agc2/speech_level_estimator.h" +#include "modules/audio_processing/agc2/vad_wrapper.h" #include "modules/audio_processing/audio_buffer.h" -#include "modules/audio_processing/include/audio_frame_view.h" #include "modules/audio_processing/logging/apm_data_dumper.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" -#include "rtc_base/strings/string_builder.h" namespace webrtc { namespace { diff --git a/modules/audio_processing/include/audio_frame_view.h b/modules/audio_processing/include/audio_frame_view.h index 27e2009067..d742e93125 100644 --- a/modules/audio_processing/include/audio_frame_view.h +++ b/modules/audio_processing/include/audio_frame_view.h @@ -12,6 +12,7 @@ #define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_VIEW_H_ #include "api/audio/audio_view.h" +#include "rtc_base/checks.h" namespace webrtc { @@ -30,7 +31,7 @@ class AudioFrameView { // channel pointers can be calculated based on that (which is what the class // now uses `DeinterleavedView<>` internally for). AudioFrameView(T* const* audio_samples, int num_channels, int channel_size) - : view_(num_channels && channel_size ? audio_samples[0] : nullptr, + : view_(num_channels && channel_size ? audio_samples : nullptr, channel_size, num_channels) { RTC_DCHECK_GE(view_.num_channels(), 0); From 5099aef700c439ca6091e33333664fd7fc725a04 Mon Sep 17 00:00:00 2001 From: Tommi Date: Thu, 22 May 2025 23:34:04 +0200 Subject: [PATCH 273/292] Add AllocationCounter for common_audio_unittests MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This test utility can be used for performance critical code that must either perform no or a known quantity of heap allocations. AllocationCounter can be used to count the number of allocations done within a specific scope and pass/fail a test based on the expected number of heap allocations. To begin with there's one test included: common_audio_unittests \ --gtest_filter=AllocationCounterTest.CountsHeapAllocations Bug: webrtc:0 Change-Id: Id349b33d10e147388b65319c63233bd3652d833c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393040 Reviewed-by: Per Åhgren Reviewed-by: Jakob Ivarsson‎ Commit-Queue: Tomas Gunnarsson Cr-Commit-Position: refs/heads/main@{#44745} --- common_audio/BUILD.gn | 3 ++ common_audio/allocation_counter.cc | 83 ++++++++++++++++++++++++++++++ common_audio/allocation_counter.h | 52 +++++++++++++++++++ 3 files changed, 138 insertions(+) create mode 100644 common_audio/allocation_counter.cc create mode 100644 common_audio/allocation_counter.h diff --git a/common_audio/BUILD.gn b/common_audio/BUILD.gn index 9a7478d8db..d40cf4e2b5 100644 --- a/common_audio/BUILD.gn +++ b/common_audio/BUILD.gn @@ -337,6 +337,8 @@ if (rtc_include_tests && !build_with_chromium) { testonly = true sources = [ + "allocation_counter.cc", + "allocation_counter.h", "audio_converter_unittest.cc", "audio_util_unittest.cc", "channel_buffer_unittest.cc", @@ -387,6 +389,7 @@ if (rtc_include_tests && !build_with_chromium) { "../test:test_main", "../test:test_support", "//testing/gtest", + "//third_party/abseil-cpp/absl/base:core_headers", ] if (is_android) { diff --git a/common_audio/allocation_counter.cc b/common_audio/allocation_counter.cc new file mode 100644 index 0000000000..2a60d2b13a --- /dev/null +++ b/common_audio/allocation_counter.cc @@ -0,0 +1,83 @@ +/* + * Copyright (c) 2025 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "common_audio/allocation_counter.h" + +#if defined(WEBRTC_ALLOCATION_COUNTER_AVAILABLE) + +#include + +#include +#include + +#include "absl/base/attributes.h" +#include "test/gtest.h" + +namespace { +#if defined(ABSL_HAVE_THREAD_LOCAL) +ABSL_CONST_INIT thread_local size_t g_new_count = 0u; +ABSL_CONST_INIT thread_local size_t g_delete_count = 0u; +#elif defined(WEBRTC_POSIX) +#error Handle WEBRTC_POSIX +#else +#error Unsupported platform +#endif +} // namespace + +void* operator new(size_t s) { + ++g_new_count; + return malloc(s); +} + +void* operator new[](size_t s) { + ++g_new_count; + return malloc(s); +} + +void operator delete(void* p) throw() { + ++g_delete_count; + return free(p); +} + +void operator delete[](void* p) throw() { + ++g_delete_count; + return free(p); +} + +namespace webrtc { + +AllocationCounter::AllocationCounter() + : initial_new_count_(g_new_count), initial_delete_count_(g_delete_count) {} + +size_t AllocationCounter::new_count() const { + return g_new_count - initial_new_count_; +} + +size_t AllocationCounter::delete_count() const { + return g_delete_count - initial_delete_count_; +} + +TEST(AllocationCounterTest, CountsHeapAllocations) { + std::vector v; + AllocationCounter counter; + EXPECT_EQ(counter.new_count(), 0u); + EXPECT_EQ(counter.delete_count(), 0u); + v.resize(1000); + EXPECT_EQ(counter.new_count(), 1u); + EXPECT_EQ(counter.delete_count(), 0u); + v.clear(); + v.shrink_to_fit(); + EXPECT_EQ(counter.new_count(), 1u); + EXPECT_EQ(counter.delete_count(), 1u); +} + +} // namespace webrtc + +#endif // defined(WEBRTC_ALLOCATION_COUNTER_AVAILABLE) diff --git a/common_audio/allocation_counter.h b/common_audio/allocation_counter.h new file mode 100644 index 0000000000..1d1caf3edd --- /dev/null +++ b/common_audio/allocation_counter.h @@ -0,0 +1,52 @@ +/* + * Copyright (c) 2025 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef COMMON_AUDIO_ALLOCATION_COUNTER_H_ +#define COMMON_AUDIO_ALLOCATION_COUNTER_H_ + +#if defined(ADDRESS_SANITIZER) || defined(LEAK_SANITIZER) || \ + defined(MEMORY_SANITIZER) || defined(THREAD_SANITIZER) || \ + defined(UNDEFINED_SANITIZER) +// The allocator override mechanism is not available since the +// sanitizers override the allocators themselves. +#else +#define WEBRTC_ALLOCATION_COUNTER_AVAILABLE 1 + +#include + +namespace webrtc { + +// Use to count the number of heap allocations that have been performed on the +// current thread within the scope of the AllocationCounter. +// +// * Note1: This class is a test-only utility. In order to be able to count +// allocations, AllocationCounter overrides the global new and delete +// operators for the test binary. +// +// * Note2: An AllocationCounter instance must always be used from the same +// thread. +class AllocationCounter { + public: + AllocationCounter(); + ~AllocationCounter() = default; + + // Returns the number of heap allocations that have been made since + // construction. + size_t new_count() const; + size_t delete_count() const; + + private: + const size_t initial_new_count_; + const size_t initial_delete_count_; +}; +} // namespace webrtc + +#endif // all the sanitizers +#endif // COMMON_AUDIO_ALLOCATION_COUNTER_H_ From 447950dc42025d50e9789a022bbed3a0c8a74be0 Mon Sep 17 00:00:00 2001 From: Diep Bui Date: Fri, 23 May 2025 12:45:38 +0000 Subject: [PATCH 274/292] Remove loss based bwe log printing This log is printed when network is healthy without loss, and loss based estimate could be higher than delay based estimate. So the log is verbose. Bug: webrtc:12707 Change-Id: I3eeb2c41653d193201cafa4f58c772abfa6e9aa6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393360 Commit-Queue: Diep Bui Reviewed-by: Per Kjellander Cr-Commit-Position: refs/heads/main@{#44746} --- modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc b/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc index c2f3018dec..bd0a81b80c 100644 --- a/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc +++ b/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc @@ -299,10 +299,9 @@ void LossBasedBweV2::UpdateBandwidthEstimate( } if (config_->bound_best_candidate && bounded_bandwidth_estimate < best_candidate.loss_limited_bandwidth) { - RTC_LOG(LS_INFO) << "Resetting loss based BWE to " - << bounded_bandwidth_estimate.kbps() - << "due to loss. Avg loss rate: " - << average_reported_loss_ratio_; + // If network is lossy, cap the best estimate by the instant upper bound, + // e.g. 450kbps if loss rate is 50%. + // Otherwise, cap the estimate by the delay-based estimate or max_bitrate. current_best_estimate_.loss_limited_bandwidth = bounded_bandwidth_estimate; current_best_estimate_.inherent_loss = 0; } else { From e4b0bdec9e8a72d536b50cfe1565c128dda38d46 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Fri, 23 May 2025 09:03:10 -0700 Subject: [PATCH 275/292] Roll chromium_revision d85e0e6296..40a123ec1c (1464635:1464787) Change log: https://chromium.googlesource.com/chromium/src/+log/d85e0e6296..40a123ec1c Full diff: https://chromium.googlesource.com/chromium/src/+/d85e0e6296..40a123ec1c Changed dependencies * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/e23f79e492..0bbbe8c0b1 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/f2c7b7b7f9..7221d98d92 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/101505f3a7..f6339ec07d * src/third_party/androidx/cipd: Q86wYH3YCLvRC3zxlUGeKoryYeDIjW1gNT3ECAilwZUC..dp2RQWmveEINZyA9NNi-6HB9ilkVW2dfJ7m4g3YvzJgC * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/6079d62bf9..04edb99d08 * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/fa368a66f7..82c60ea0d2 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/c76ac7bd93..f3c7973853 DEPS diff: https://chromium.googlesource.com/chromium/src/+/d85e0e6296..40a123ec1c/DEPS BUG=None Change-Id: I296f5c6eeca86339cd994d34ce8888476f093887 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393321 Commit-Queue: Autoroller Bot-Commit: Autoroller Cr-Commit-Position: refs/heads/main@{#44747} --- DEPS | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/DEPS b/DEPS index 1297061441..5bbb67c34d 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': 'd85e0e62960bdf9988ec18dc4b4e6e1897fa7664', + 'chromium_revision': '40a123ec1c34a54730b3ae9c8f4cbae364e73f66', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@e23f79e492a90f81d91bdbfdfc9472c12f9d8ab7', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@0bbbe8c0b1a84a1d6d2a04efadfbb21170670898', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@f2c7b7b7f9477c00c9c05c6207427da6dfa39ac8', + 'https://chromium.googlesource.com/chromium/src/testing@7221d98d9210df9d75561fb3fdb14618996b3c0f', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@101505f3a724a53020ada0861e6672c67023de4f', + 'https://chromium.googlesource.com/chromium/src/third_party@f6339ec07d0642c32eb3bf18c8407785c08926b3', 'src/buildtools/linux64': { 'packages': [ @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@6079d62bf9b04cfd0390a07802f148d526d459a3', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@04edb99d08d7e47cd3562d1d64ef2de64e3bb1e1', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'fa368a66f714203d1232ff62ece403bd2f5e7c10', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '82c60ea0d2b453aded71857c93b9e6a2637da79b', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + '28bf5df73ef2f345a936d9cc95d64ba8ed426a53', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@c76ac7bd937400b9ed8c1d7c64d60665159fc560', + 'https://chromium.googlesource.com/chromium/src/tools@f3c797385333da22bb2a7f2e9d10f163d61a8948', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'Q86wYH3YCLvRC3zxlUGeKoryYeDIjW1gNT3ECAilwZUC', + 'version': 'dp2RQWmveEINZyA9NNi-6HB9ilkVW2dfJ7m4g3YvzJgC', }, ], 'condition': 'checkout_android and non_git_source', From 41ab5e8dff4789108a704f859c952de780fe9e25 Mon Sep 17 00:00:00 2001 From: Guido Urdaneta Date: Fri, 23 May 2025 15:29:23 +0200 Subject: [PATCH 276/292] Allow setting audio level in receiver transformer frames MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit No-Iwyu: No output from iwyu/apply-include-cleaner Bug: chromium:402391734 Change-Id: Ia50cfe8bc5bdc7b6b848b9330b86d957e6384b55 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393340 Commit-Queue: Guido Urdaneta Reviewed-by: Jakob Ivarsson‎ Reviewed-by: Guido Urdaneta Cr-Commit-Position: refs/heads/main@{#44748} --- ...nnel_receive_frame_transformer_delegate.cc | 18 ++++++ ...ive_frame_transformer_delegate_unittest.cc | 61 +++++++++++++++++++ 2 files changed, 79 insertions(+) diff --git a/audio/channel_receive_frame_transformer_delegate.cc b/audio/channel_receive_frame_transformer_delegate.cc index 42f40b764a..a9a52990c0 100644 --- a/audio/channel_receive_frame_transformer_delegate.cc +++ b/audio/channel_receive_frame_transformer_delegate.cc @@ -10,6 +10,7 @@ #include "audio/channel_receive_frame_transformer_delegate.h" +#include #include #include #include @@ -92,6 +93,17 @@ class TransformableIncomingAudioFrame return std::nullopt; } + bool CanSetAudioLevel() const override { return true; } + + void SetAudioLevel(std::optional audio_level_dbov) override { + header_.extension.set_audio_level( + audio_level_dbov.has_value() + ? std::make_optional(webrtc::AudioLevel( + /*voice_activity=*/false, + std::min(*audio_level_dbov, static_cast(127u)))) + : std::nullopt); + } + std::optional ReceiveTime() const override { return receive_time_ == Timestamp::MinusInfinity() ? std::nullopt @@ -200,6 +212,12 @@ void ChannelReceiveFrameTransformerDelegate::ReceiveFrame( header.extension.absolute_capture_time->absolute_capture_timestamp = transformed_frame->AbsoluteCaptureTimestamp().value(); } + if (transformed_frame->AudioLevel().has_value()) { + // TODO(crbug.com/webrtc/419746427): Add support for voice activity in + // TransformableAudioFrameInterface. + header.extension.set_audio_level(AudioLevel( + /*voice_activity=*/false, *transformed_frame->AudioLevel())); + } } else { auto* transformed_incoming_frame = static_cast(frame.get()); diff --git a/audio/channel_receive_frame_transformer_delegate_unittest.cc b/audio/channel_receive_frame_transformer_delegate_unittest.cc index 4b60ea210a..5e8d35fc5d 100644 --- a/audio/channel_receive_frame_transformer_delegate_unittest.cc +++ b/audio/channel_receive_frame_transformer_delegate_unittest.cc @@ -34,6 +34,7 @@ namespace { using ::testing::_; using ::testing::ElementsAre; using ::testing::NiceMock; +using ::testing::Return; using ::testing::SaveArg; constexpr Timestamp kFakeReceiveTimestamp = Timestamp::Millis(1234567); @@ -308,5 +309,65 @@ TEST(ChannelReceiveFrameTransformerDelegateTest, } } +TEST(ChannelReceiveFrameTransformerDelegateTest, SetAudioLevel) { + AutoThread main_thread; + scoped_refptr mock_frame_transformer = + make_ref_counted>(); + scoped_refptr delegate = + make_ref_counted( + /*receive_frame_callback=*/nullptr, mock_frame_transformer, + Thread::Current()); + delegate->Init(); + const uint8_t data[] = {1, 2, 3, 4}; + ArrayView packet(data, sizeof(data)); + std::unique_ptr frame; + ON_CALL(*mock_frame_transformer, Transform) + .WillByDefault( + [&](std::unique_ptr transform_frame) { + frame = std::move(transform_frame); + }); + delegate->Transform(packet, RTPHeader(), /*ssrc=*/1111, + /*mimeType=*/"audio/opus", kFakeReceiveTimestamp); + + EXPECT_TRUE(frame); + auto* audio_frame = + static_cast(frame.get()); + EXPECT_TRUE(audio_frame->CanSetAudioLevel()); + EXPECT_FALSE(audio_frame->AudioLevel().has_value()); + + audio_frame->SetAudioLevel(67u); + EXPECT_EQ(audio_frame->AudioLevel(), 67u); + + // Audio level is clamped to the range [0, 127]. + audio_frame->SetAudioLevel(128u); + EXPECT_EQ(audio_frame->AudioLevel(), 127u); +} + +TEST(ChannelReceiveFrameTransformerDelegateTest, + ReceivingSenderFrameWithAudioValueSetsAudioLevelInHeader) { + AutoThread main_thread; + scoped_refptr mock_frame_transformer = + make_ref_counted>(); + MockChannelReceive mock_channel; + scoped_refptr delegate = + make_ref_counted( + mock_channel.callback(), mock_frame_transformer, Thread::Current()); + delegate->Init(); + + std::unique_ptr audio_frame = + std::make_unique>(); + ON_CALL(*audio_frame, GetDirection()) + .WillByDefault(Return(TransformableFrameInterface::Direction::kSender)); + ON_CALL(*audio_frame, AudioLevel()).WillByDefault(Return(111u)); + + RTPHeader header; + EXPECT_CALL(mock_channel, ReceiveFrame).WillOnce(SaveArg<1>(&header)); + delegate->OnTransformedFrame(std::move(audio_frame)); + ThreadManager::ProcessAllMessageQueuesForTesting(); + + ASSERT_TRUE(header.extension.audio_level().has_value()); + EXPECT_EQ(header.extension.audio_level()->level(), 111); +} + } // namespace } // namespace webrtc From a0627fe0e93d67295b996e4448c179b5d11376d9 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Fri, 23 May 2025 13:03:58 -0700 Subject: [PATCH 277/292] Roll chromium_revision 40a123ec1c..8bd819a6dc (1464787:1464938) Change log: https://chromium.googlesource.com/chromium/src/+log/40a123ec1c..8bd819a6dc Full diff: https://chromium.googlesource.com/chromium/src/+/40a123ec1c..8bd819a6dc Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/50dc228e38..4da407f0ac * src/build: https://chromium.googlesource.com/chromium/src/build/+log/b228bdd418..57810addcd * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/0bbbe8c0b1..d29a81b816 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/7221d98d92..186b283e5e * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/f6339ec07d..9eaedc200b * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/c5795a3288..9295969e1d * src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/a6f7c1ae19..689e2e2268 * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/04edb99d08..72ea42a9ed * src/third_party/perfetto: https://chromium.googlesource.com/external/github.com/google/perfetto.git/+log/82c60ea0d2..dd35b295cd * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/f3c7973853..2d74c1f5b9 DEPS diff: https://chromium.googlesource.com/chromium/src/+/40a123ec1c..8bd819a6dc/DEPS BUG=None Change-Id: I4ccbe0eed3392afe9668d71bec15464c3ddbf5f5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393383 Commit-Queue: Autoroller Bot-Commit: Autoroller Cr-Commit-Position: refs/heads/main@{#44749} --- DEPS | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/DEPS b/DEPS index 5bbb67c34d..6c3a84579e 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '40a123ec1c34a54730b3ae9c8f4cbae364e73f66', + 'chromium_revision': '8bd819a6dc464c3568a8d35efaf73fc80ec58270', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@50dc228e3885dea149d3cee5e31f326c91c8649d', + 'https://chromium.googlesource.com/chromium/src/base@4da407f0aca0d5cd6c5d4999eb9df3763486601d', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@b228bdd41836c0fac86a729ef5ebbe3047999f7b', + 'https://chromium.googlesource.com/chromium/src/build@57810addcdd7dfda8bf7c15f6456ea1ff9283815', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@813bee86eeb7d00fa170b81e221120576a18517a', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@0bbbe8c0b1a84a1d6d2a04efadfbb21170670898', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@d29a81b816e460bb309200e18ac3ae5a641ce19d', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@7221d98d9210df9d75561fb3fdb14618996b3c0f', + 'https://chromium.googlesource.com/chromium/src/testing@186b283e5e14d7e52347a7c7f851bda6201b1e2c', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@f6339ec07d0642c32eb3bf18c8407785c08926b3', + 'https://chromium.googlesource.com/chromium/src/third_party@9eaedc200b114025d63f87f2caeec4e571055766', 'src/buildtools/linux64': { 'packages': [ @@ -323,7 +323,7 @@ deps = { 'src/third_party/clang-format/script': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@37f6e68a107df43b7d7e044fd36a13cbae3413f2', 'src/third_party/compiler-rt/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@a6f7c1ae19c301452a66fdfeceef4d7c47d86107', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@689e2e2268b19ff47d6a7e3a0cedd6d5ac19d0a7', 'src/third_party/libc++/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@a01c02c9d4acbdae3b7e8a2f3ee58579a9c29f96', 'src/third_party/libc++abi/src': @@ -480,7 +480,7 @@ deps = { }, 'src/third_party/boringssl/src': - 'https://boringssl.googlesource.com/boringssl.git@c5795a328880eadb879088fc59cd0087edc8dc20', + 'https://boringssl.googlesource.com/boringssl.git@9295969e1dad2c31d0d99481734c1c68dcbc6403', 'src/third_party/breakpad/breakpad': 'https://chromium.googlesource.com/breakpad/breakpad.git@2625edb085169e92cf036c236ac79ab594a7b1cc', 'src/third_party/catapult': @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@04edb99d08d7e47cd3562d1d64ef2de64e3bb1e1', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@72ea42a9ed880182ae3a9e09cf633921398155d5', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -597,7 +597,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/perfetto': - Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + '82c60ea0d2b453aded71857c93b9e6a2637da79b', + Var('chromium_git') + '/external/github.com/google/perfetto.git' + '@' + 'dd35b295cd359ba094404218414955f961a0d6ae', 'src/third_party/protobuf-javascript/src': Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + '28bf5df73ef2f345a936d9cc95d64ba8ed426a53', 'src/third_party/libvpx/source/libvpx': @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@f3c797385333da22bb2a7f2e9d10f163d61a8948', + 'https://chromium.googlesource.com/chromium/src/tools@2d74c1f5b933b37f292cfce591a5df00386e1ba9', 'src/third_party/espresso': { 'packages': [ From 5400eaf45a36a0b5db81fe278e5e10963f938800 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Fri, 23 May 2025 17:02:53 -0700 Subject: [PATCH 278/292] Roll chromium_revision 8bd819a6dc..fb867d2bd0 (1464938:1465065) Change log: https://chromium.googlesource.com/chromium/src/+log/8bd819a6dc..fb867d2bd0 Full diff: https://chromium.googlesource.com/chromium/src/+/8bd819a6dc..fb867d2bd0 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/4da407f0ac..8a3bd4e294 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/57810addcd..74c1cea432 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/d29a81b816..48b69c9ca1 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/186b283e5e..739fbc1a64 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/9eaedc200b..d3abd9af0d * src/third_party/androidx/cipd: dp2RQWmveEINZyA9NNi-6HB9ilkVW2dfJ7m4g3YvzJgC..YTyVPkY5O2nkIxZIlN-Z4vbTlcfp8XK1K8EaaHMIQA4C * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/72ea42a9ed..f736a43230 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/2d74c1f5b9..681d318af2 DEPS diff: https://chromium.googlesource.com/chromium/src/+/8bd819a6dc..fb867d2bd0/DEPS BUG=None Change-Id: I16a13823f6d2d9dc6c9e57b68815c50d2134da0a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393323 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44750} --- DEPS | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/DEPS b/DEPS index 6c3a84579e..9d96c122f2 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '8bd819a6dc464c3568a8d35efaf73fc80ec58270', + 'chromium_revision': 'fb867d2bd07eb80640f5c7765dfdb8da1114a02c', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@4da407f0aca0d5cd6c5d4999eb9df3763486601d', + 'https://chromium.googlesource.com/chromium/src/base@8a3bd4e29427ba9d3747fd0d3a2189595fcfcf2a', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@57810addcdd7dfda8bf7c15f6456ea1ff9283815', + 'https://chromium.googlesource.com/chromium/src/build@74c1cea4320f6744444fff7ceeff2e6b0370eb2d', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@813bee86eeb7d00fa170b81e221120576a18517a', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@d29a81b816e460bb309200e18ac3ae5a641ce19d', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@48b69c9ca1c8ac6f3db80ddc5dbd55d4575fdde3', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@186b283e5e14d7e52347a7c7f851bda6201b1e2c', + 'https://chromium.googlesource.com/chromium/src/testing@739fbc1a64b4a6413515887b85e73ac22fb9ef44', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@9eaedc200b114025d63f87f2caeec4e571055766', + 'https://chromium.googlesource.com/chromium/src/third_party@d3abd9af0d01d3e4da0ab97eb669fc0dc7660c8c', 'src/buildtools/linux64': { 'packages': [ @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@72ea42a9ed880182ae3a9e09cf633921398155d5', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@f736a432308c3d9630b1e77813ded44cbead096c', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@2d74c1f5b933b37f292cfce591a5df00386e1ba9', + 'https://chromium.googlesource.com/chromium/src/tools@681d318af28a626a6be9c2a809badc0db28c5b28', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'dp2RQWmveEINZyA9NNi-6HB9ilkVW2dfJ7m4g3YvzJgC', + 'version': 'YTyVPkY5O2nkIxZIlN-Z4vbTlcfp8XK1K8EaaHMIQA4C', }, ], 'condition': 'checkout_android and non_git_source', From 54f2bd06922d0859e05d4307fcfef4c59e72b6ae Mon Sep 17 00:00:00 2001 From: webrtc-version-updater Date: Fri, 23 May 2025 21:09:14 -0700 Subject: [PATCH 279/292] Update WebRTC code version (2025-05-24T04:08:54). Bug: None Change-Id: Ifbde01a35951fd616bb07bf89a9c2a584d57398f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393326 Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Cr-Commit-Position: refs/heads/main@{#44751} --- call/version.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/call/version.cc b/call/version.cc index 511e4b882e..a2fefb3bc0 100644 --- a/call/version.cc +++ b/call/version.cc @@ -13,7 +13,7 @@ namespace webrtc { // The timestamp is always in UTC. -const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-23T04:04:31"; +const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-24T04:08:54"; void LoadWebRTCVersionInRegister() { // Using volatile to instruct the compiler to not optimize `p` away even From 6733a59a44ee089e9be6fc4b1f69a7e03efb5353 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Sun, 25 May 2025 15:02:04 -0700 Subject: [PATCH 280/292] Roll chromium_revision fb867d2bd0..4d8cdcc0a8 (1465065:1465279) Change log: https://chromium.googlesource.com/chromium/src/+log/fb867d2bd0..4d8cdcc0a8 Full diff: https://chromium.googlesource.com/chromium/src/+/fb867d2bd0..4d8cdcc0a8 Changed dependencies * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/48b69c9ca1..c61efe1d6d * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/d3abd9af0d..d3778078c8 * src/third_party/androidx/cipd: YTyVPkY5O2nkIxZIlN-Z4vbTlcfp8XK1K8EaaHMIQA4C..jxj98gNBLn43NAe-JF3GXSRicC44jGfwPTmq4vJvFHYC * src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/689e2e2268..9c6d38aa28 * src/third_party/googletest/src: https://chromium.googlesource.com/external/github.com/google/googletest.git/+log/6aa03e6774..09ffd00153 * src/third_party/grpc/src: https://chromium.googlesource.com/external/github.com/grpc/grpc.git/+log/f736a43230..cadf3c8329 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/681d318af2..806658eb4d DEPS diff: https://chromium.googlesource.com/chromium/src/+/fb867d2bd0..4d8cdcc0a8/DEPS BUG=None Change-Id: I5c9565cbd9ad58697af69f11ce42380937a0390d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393337 Commit-Queue: Autoroller Bot-Commit: Autoroller Cr-Commit-Position: refs/heads/main@{#44752} --- DEPS | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/DEPS b/DEPS index 9d96c122f2..6f1bc9f36e 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': 'fb867d2bd07eb80640f5c7765dfdb8da1114a02c', + 'chromium_revision': '4d8cdcc0a83f879cbcee987cb82ba5b4f28f86c5', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@48b69c9ca1c8ac6f3db80ddc5dbd55d4575fdde3', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@c61efe1d6df4173b3fa0c17b1ee947b114d4c075', 'condition': 'checkout_ios', }, 'src/testing': 'https://chromium.googlesource.com/chromium/src/testing@739fbc1a64b4a6413515887b85e73ac22fb9ef44', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@d3abd9af0d01d3e4da0ab97eb669fc0dc7660c8c', + 'https://chromium.googlesource.com/chromium/src/third_party@d3778078c8788c51d42be013d4128ccd57067c18', 'src/buildtools/linux64': { 'packages': [ @@ -323,7 +323,7 @@ deps = { 'src/third_party/clang-format/script': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@37f6e68a107df43b7d7e044fd36a13cbae3413f2', 'src/third_party/compiler-rt/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@689e2e2268b19ff47d6a7e3a0cedd6d5ac19d0a7', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@9c6d38aa28100761bc2d6dcd2a590a37f8519adf', 'src/third_party/libc++/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@a01c02c9d4acbdae3b7e8a2f3ee58579a9c29f96', 'src/third_party/libc++abi/src': @@ -503,7 +503,7 @@ deps = { 'src/third_party/flatbuffers/src': 'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32', 'src/third_party/grpc/src': { - 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@f736a432308c3d9630b1e77813ded44cbead096c', + 'url': 'https://chromium.googlesource.com/external/github.com/grpc/grpc.git@cadf3c8329377e93b1f5e2d6a43d91f7a4becc28', }, # Used for embedded builds. CrOS & Linux use the system version. 'src/third_party/fontconfig/src': { @@ -525,7 +525,7 @@ deps = { 'condition': 'checkout_android', }, 'src/third_party/googletest/src': - 'https://chromium.googlesource.com/external/github.com/google/googletest.git@6aa03e6774f8cb70da277c56efb24b44ce29d8d7', + 'https://chromium.googlesource.com/external/github.com/google/googletest.git@09ffd0015395354774c059a17d9f5bee36177ff9', 'src/third_party/icu': { 'url': 'https://chromium.googlesource.com/chromium/deps/icu.git@b929596baebf0ab4ac7ec07f38365db4c50a559d', }, @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@681d318af28a626a6be9c2a809badc0db28c5b28', + 'https://chromium.googlesource.com/chromium/src/tools@806658eb4d5ccf58c7cf07b5127539886d16ded0', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'YTyVPkY5O2nkIxZIlN-Z4vbTlcfp8XK1K8EaaHMIQA4C', + 'version': 'jxj98gNBLn43NAe-JF3GXSRicC44jGfwPTmq4vJvFHYC', }, ], 'condition': 'checkout_android and non_git_source', From 2ebee7129fffe0bad8d6a1ec18664efb61cd5470 Mon Sep 17 00:00:00 2001 From: webrtc-version-updater Date: Sun, 25 May 2025 21:07:23 -0700 Subject: [PATCH 281/292] Update WebRTC code version (2025-05-26T04:07:21). Bug: None Change-Id: I00219aa95abe6cd553ff39f3ca72a3ac54aec561 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393338 Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Cr-Commit-Position: refs/heads/main@{#44753} --- call/version.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/call/version.cc b/call/version.cc index a2fefb3bc0..f10e6c2f1a 100644 --- a/call/version.cc +++ b/call/version.cc @@ -13,7 +13,7 @@ namespace webrtc { // The timestamp is always in UTC. -const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-24T04:08:54"; +const char* const kSourceTimestamp = "WebRTC source stamp 2025-05-26T04:07:21"; void LoadWebRTCVersionInRegister() { // Using volatile to instruct the compiler to not optimize `p` away even From 8178847af905d8d14ffaee5bbf7d3d1ef1296027 Mon Sep 17 00:00:00 2001 From: Guido Urdaneta Date: Fri, 23 May 2025 16:38:47 +0200 Subject: [PATCH 282/292] Use milliseconds to covert NTP capture time in receiver frames. MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit No-Iwyu: No output from iwyu/apply-include-cleaner Bug: chromium:419755714 Change-Id: Ia5ede3e253902ad288096eb9db3bc63dc7ba15e9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393400 Commit-Queue: Guido Urdaneta Reviewed-by: Jakob Ivarsson‎ Cr-Commit-Position: refs/heads/main@{#44754} --- audio/channel_receive_frame_transformer_delegate.cc | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/audio/channel_receive_frame_transformer_delegate.cc b/audio/channel_receive_frame_transformer_delegate.cc index a9a52990c0..e3a3617b95 100644 --- a/audio/channel_receive_frame_transformer_delegate.cc +++ b/audio/channel_receive_frame_transformer_delegate.cc @@ -112,7 +112,7 @@ class TransformableIncomingAudioFrame std::optional CaptureTime() const override { if (header_.extension.absolute_capture_time) { - return Timestamp::Micros(UQ32x32ToInt64Us( + return Timestamp::Millis(UQ32x32ToInt64Ms( header_.extension.absolute_capture_time->absolute_capture_timestamp)); } return std::nullopt; @@ -122,8 +122,8 @@ class TransformableIncomingAudioFrame if (header_.extension.absolute_capture_time && header_.extension.absolute_capture_time ->estimated_capture_clock_offset) { - return TimeDelta::Micros( - Q32x32ToInt64Us(*header_.extension.absolute_capture_time + return TimeDelta::Millis( + Q32x32ToInt64Ms(*header_.extension.absolute_capture_time ->estimated_capture_clock_offset)); } return std::nullopt; From 66bd0e0cffa97feccb4a1e5c3420d37693c40741 Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Mon, 26 May 2025 10:36:12 +0200 Subject: [PATCH 283/292] Update peerconnection examples not to rely on the global field trials No-Iwyu: questionable suggestion unrelated to this patch. Bug: webrtc:42220378 Change-Id: I02decdef8e0b10357dade329919568a8c0150e29 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393440 Reviewed-by: Tomas Gunnarsson Commit-Queue: Danil Chapovalov Cr-Commit-Position: refs/heads/main@{#44755} --- examples/BUILD.gn | 7 ++---- examples/peerconnection/client/conductor.cc | 9 +++++--- examples/peerconnection/client/conductor.h | 7 ++++-- examples/peerconnection/client/linux/main.cc | 21 +++++++++++------- examples/peerconnection/client/main.cc | 23 +++++++++++++------- examples/peerconnection/server/main.cc | 11 ++++------ 6 files changed, 45 insertions(+), 33 deletions(-) diff --git a/examples/BUILD.gn b/examples/BUILD.gn index e17d40ab0d..b1b5829395 100644 --- a/examples/BUILD.gn +++ b/examples/BUILD.gn @@ -684,8 +684,8 @@ if (is_linux || is_chromeos || is_win) { "../api:async_dns_resolver", "../api:audio_options_api", "../api:create_frame_generator", - "../api:create_peerconnection_factory", "../api:enable_media", + "../api:field_trials", "../api:libjingle_peerconnection_api", "../api:make_ref_counted", "../api:media_stream_interface", @@ -722,12 +722,11 @@ if (is_linux || is_chromeos || is_win) { "../rtc_base:threading", "../rtc_base/third_party/sigslot", "../system_wrappers", - "../system_wrappers:field_trial", - "../test:field_trial", "../test:frame_generator_capturer", "../test:platform_video_capturer", "../test:rtp_test_utils", "../test:test_video_capturer", + "//third_party/abseil-cpp/absl/base:nullability", "//third_party/abseil-cpp/absl/memory", "//third_party/jsoncpp", ] @@ -805,8 +804,6 @@ if (is_linux || is_chromeos || is_win) { "../rtc_base:ip_address", "../rtc_base:net_helpers", "../rtc_base:stringutils", - "../system_wrappers:field_trial", - "../test:field_trial", "//third_party/abseil-cpp/absl/flags:flag", "//third_party/abseil-cpp/absl/flags:parse", "//third_party/abseil-cpp/absl/flags:usage", diff --git a/examples/peerconnection/client/conductor.cc b/examples/peerconnection/client/conductor.cc index dc5ad214a0..2039ea66ee 100644 --- a/examples/peerconnection/client/conductor.cc +++ b/examples/peerconnection/client/conductor.cc @@ -18,12 +18,13 @@ #include #include +#include "absl/base/nullability.h" #include "absl/memory/memory.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" #include "api/audio_options.h" #include "api/enable_media.h" -#include "api/environment/environment_factory.h" +#include "api/environment/environment.h" #include "api/jsep.h" #include "api/make_ref_counted.h" #include "api/media_stream_interface.h" @@ -140,10 +141,12 @@ class CapturerTrackSource : public webrtc::VideoTrackSource { } // namespace -Conductor::Conductor(PeerConnectionClient* client, MainWindow* main_wnd) +Conductor::Conductor(const webrtc::Environment& env, + PeerConnectionClient* absl_nonnull client, + MainWindow* absl_nonnull main_wnd) : peer_id_(-1), loopback_(false), - env_(webrtc::CreateEnvironment()), + env_(env), client_(client), main_wnd_(main_wnd) { client_->RegisterObserver(this); diff --git a/examples/peerconnection/client/conductor.h b/examples/peerconnection/client/conductor.h index 3970963f8e..68b70bb2f6 100644 --- a/examples/peerconnection/client/conductor.h +++ b/examples/peerconnection/client/conductor.h @@ -16,6 +16,7 @@ #include #include +#include "absl/base/nullability.h" #include "api/data_channel_interface.h" #include "api/environment/environment.h" #include "api/jsep.h" @@ -45,7 +46,9 @@ class Conductor : public webrtc::PeerConnectionObserver, TRACK_REMOVED, }; - Conductor(PeerConnectionClient* client, MainWindow* main_wnd); + Conductor(const webrtc::Environment& env, + PeerConnectionClient* absl_nonnull client, + MainWindow* absl_nonnull main_wnd); bool connection_active() const; @@ -124,7 +127,7 @@ class Conductor : public webrtc::PeerConnectionObserver, int peer_id_; bool loopback_; - webrtc::Environment env_; + const webrtc::Environment env_; std::unique_ptr signaling_thread_; webrtc::scoped_refptr peer_connection_; webrtc::scoped_refptr diff --git a/examples/peerconnection/client/linux/main.cc b/examples/peerconnection/client/linux/main.cc index 5e3be8beb9..a0a80e83ac 100644 --- a/examples/peerconnection/client/linux/main.cc +++ b/examples/peerconnection/client/linux/main.cc @@ -12,8 +12,17 @@ #include #include +#include +#include + +#include "absl/flags/flag.h" #include "absl/flags/parse.h" +#include "api/environment/environment.h" +#include "api/environment/environment_factory.h" +#include "api/field_trials.h" +#include "api/make_ref_counted.h" #include "api/scoped_refptr.h" +#include "api/units/time_delta.h" #include "examples/peerconnection/client/conductor.h" #include "examples/peerconnection/client/flag_defs.h" #include "examples/peerconnection/client/linux/main_wnd.h" @@ -21,8 +30,6 @@ #include "rtc_base/physical_socket_server.h" #include "rtc_base/ssl_adapter.h" #include "rtc_base/thread.h" -#include "system_wrappers/include/field_trial.h" -#include "test/field_trial.h" class CustomSocketServer : public webrtc::PhysicalSocketServer { public: @@ -78,11 +85,9 @@ int main(int argc, char* argv[]) { absl::ParseCommandLine(argc, argv); - // InitFieldTrialsFromString stores the char*, so the char array must outlive - // the application. - const std::string forced_field_trials = - absl::GetFlag(FLAGS_force_fieldtrials); - webrtc::field_trial::InitFieldTrialsFromString(forced_field_trials.c_str()); + webrtc::Environment env = + webrtc::CreateEnvironment(std::make_unique( + absl::GetFlag(FLAGS_force_fieldtrials))); // Abort if the user specifies a port that is outside the allowed // range [1, 65535]. @@ -103,7 +108,7 @@ int main(int argc, char* argv[]) { webrtc::InitializeSSL(); // Must be constructed after we set the socketserver. PeerConnectionClient client; - auto conductor = webrtc::make_ref_counted(&client, &wnd); + auto conductor = webrtc::make_ref_counted(env, &client, &wnd); socket_server.set_client(&client); socket_server.set_conductor(conductor.get()); diff --git a/examples/peerconnection/client/main.cc b/examples/peerconnection/client/main.cc index c5ac647f72..87f78120a2 100644 --- a/examples/peerconnection/client/main.cc +++ b/examples/peerconnection/client/main.cc @@ -14,20 +14,29 @@ #include // must come after windows.h // clang-format on +#include +#include +#include +#include #include #include +#include "absl/flags/flag.h" #include "absl/flags/parse.h" +#include "api/environment/environment.h" +#include "api/environment/environment_factory.h" +#include "api/field_trials.h" +#include "api/make_ref_counted.h" #include "examples/peerconnection/client/conductor.h" #include "examples/peerconnection/client/flag_defs.h" #include "examples/peerconnection/client/main_wnd.h" #include "examples/peerconnection/client/peer_connection_client.h" #include "rtc_base/checks.h" +#include "rtc_base/physical_socket_server.h" #include "rtc_base/ssl_adapter.h" #include "rtc_base/string_utils.h" // For ToUtf8 +#include "rtc_base/thread.h" #include "rtc_base/win32_socket_init.h" -#include "system_wrappers/include/field_trial.h" -#include "test/field_trial.h" namespace { // A helper class to translate Windows command line arguments into UTF8, @@ -83,11 +92,9 @@ int PASCAL wWinMain(HINSTANCE instance, absl::ParseCommandLine(argc, argv); - // InitFieldTrialsFromString stores the char*, so the char array must outlive - // the application. - const std::string forced_field_trials = - absl::GetFlag(FLAGS_force_fieldtrials); - webrtc::field_trial::InitFieldTrialsFromString(forced_field_trials.c_str()); + webrtc::Environment env = + webrtc::CreateEnvironment(std::make_unique( + absl::GetFlag(FLAGS_force_fieldtrials))); // Abort if the user specifies a port that is outside the allowed // range [1, 65535]. @@ -106,7 +113,7 @@ int PASCAL wWinMain(HINSTANCE instance, webrtc::InitializeSSL(); PeerConnectionClient client; - auto conductor = webrtc::make_ref_counted(&client, &wnd); + auto conductor = webrtc::make_ref_counted(env, &client, &wnd); // Main loop. MSG msg; diff --git a/examples/peerconnection/server/main.cc b/examples/peerconnection/server/main.cc index 501c7fdaee..0b4e483427 100644 --- a/examples/peerconnection/server/main.cc +++ b/examples/peerconnection/server/main.cc @@ -24,9 +24,11 @@ #include "examples/peerconnection/server/data_socket.h" #include "examples/peerconnection/server/peer_channel.h" #include "rtc_base/checks.h" -#include "system_wrappers/include/field_trial.h" -#include "test/field_trial.h" +// As of now, no components in peerconnection_server rely on WebRTC components +// that change its behavior based on a field trial, so this flag is currently +// unused. See peerconnection_client for example how this command line flag +// can be used and propagated. ABSL_FLAG( std::string, force_fieldtrials, @@ -70,11 +72,6 @@ int main(int argc, char* argv[]) { "Example usage: ./peerconnection_server --port=8888\n"); absl::ParseCommandLine(argc, argv); - // InitFieldTrialsFromString stores the char*, so the char array must outlive - // the application. - const std::string force_field_trials = absl::GetFlag(FLAGS_force_fieldtrials); - webrtc::field_trial::InitFieldTrialsFromString(force_field_trials.c_str()); - int port = absl::GetFlag(FLAGS_port); // Abort if the user specifies a port that is outside the allowed From 2fe2d5190bd3d8766759e3a7de2b648d5ce96aee Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Mon, 26 May 2025 03:02:05 -0700 Subject: [PATCH 284/292] Roll chromium_revision 4d8cdcc0a8..1f52f5a706 (1465279:1465420) Change log: https://chromium.googlesource.com/chromium/src/+log/4d8cdcc0a8..1f52f5a706 Full diff: https://chromium.googlesource.com/chromium/src/+/4d8cdcc0a8..1f52f5a706 Changed dependencies * siso_version: git_revision:d9393c2115244b6e4a797189055e4a2b6769a64d..git_revision:df8adf712c5b5605b438fdfcd44235107811e5ef * src/base: https://chromium.googlesource.com/chromium/src/base/+log/8a3bd4e294..6e9d1a97fc * src/build: https://chromium.googlesource.com/chromium/src/build/+log/74c1cea432..26dfa1e660 * src/buildtools: https://chromium.googlesource.com/chromium/src/buildtools/+log/813bee86ee..1fc7350e65 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/c61efe1d6d..08d976ff4f * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/d3778078c8..fe88e9476e * src/third_party/androidx/cipd: jxj98gNBLn43NAe-JF3GXSRicC44jGfwPTmq4vJvFHYC..N3BICdmBe8aMPRI6ZbZ4tNuSs26ziJ1PctmwDJnJ2p0C * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/e0ece52cfb..a8900cc0f0 * src/third_party/freetype/src: https://chromium.googlesource.com/chromium/src/third_party/freetype2.git/+log/5987a9f51d..57a8730450 * src/third_party/siso/cipd: git_revision:d9393c2115244b6e4a797189055e4a2b6769a64d..git_revision:df8adf712c5b5605b438fdfcd44235107811e5ef * src/third_party/zstd/src: https://chromium.googlesource.com/external/github.com/facebook/zstd.git/+log/d654fca786..f9938c217d * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/806658eb4d..147300be82 DEPS diff: https://chromium.googlesource.com/chromium/src/+/4d8cdcc0a8..1f52f5a706/DEPS BUG=None Change-Id: Ied19f1b034e65dcfafedd1d54bf8f385657bcf2e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393433 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44756} --- DEPS | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/DEPS b/DEPS index 6f1bc9f36e..6c25d17d12 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '4d8cdcc0a83f879cbcee987cb82ba5b4f28f86c5', + 'chromium_revision': '1f52f5a706529fe517aa750a516faa90be92f56e', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -53,7 +53,7 @@ vars = { # reclient CIPD package version 'reclient_version': 're_client_version:0.178.0.5ee9d3e8-gomaip', # siso CIPD package version. - 'siso_version': 'git_revision:d9393c2115244b6e4a797189055e4a2b6769a64d', + 'siso_version': 'git_revision:df8adf712c5b5605b438fdfcd44235107811e5ef', # ninja CIPD package. 'ninja_package': 'infra/3pp/tools/ninja/', @@ -73,24 +73,24 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@8a3bd4e29427ba9d3747fd0d3a2189595fcfcf2a', + 'https://chromium.googlesource.com/chromium/src/base@6e9d1a97fc906921130453717244d39a773ec684', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@74c1cea4320f6744444fff7ceeff2e6b0370eb2d', + 'https://chromium.googlesource.com/chromium/src/build@26dfa1e660573ea7db832c2c3b9eaa1fce398ff7', 'src/buildtools': - 'https://chromium.googlesource.com/chromium/src/buildtools@813bee86eeb7d00fa170b81e221120576a18517a', + 'https://chromium.googlesource.com/chromium/src/buildtools@1fc7350e65e9d7848c083b83aaf67611e74a5654', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. 'src/examples/androidtests/third_party/gradle': { 'url': 'https://chromium.googlesource.com/external/github.com/gradle/gradle.git@f2d1fb54a951d8b11d25748e4711bec8d128d7e3', 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@c61efe1d6df4173b3fa0c17b1ee947b114d4c075', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@08d976ff4f97ccfae36c7a07e361f6a51aea95bd', 'condition': 'checkout_ios', }, 'src/testing': 'https://chromium.googlesource.com/chromium/src/testing@739fbc1a64b4a6413515887b85e73ac22fb9ef44', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@d3778078c8788c51d42be013d4128ccd57067c18', + 'https://chromium.googlesource.com/chromium/src/third_party@fe88e9476e499896a972e9ad2cf227698199e4a8', 'src/buildtools/linux64': { 'packages': [ @@ -497,7 +497,7 @@ deps = { 'src/third_party/crc32c/src': 'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558', 'src/third_party/depot_tools': - 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@e0ece52cfb4faaf6d4468a94068265452ef6de13', + 'https://chromium.googlesource.com/chromium/tools/depot_tools.git@a8900cc0f023d6a662eb66b317e8ddceeb113490', 'src/third_party/ffmpeg': 'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@dcdd0fa51b65a0b1688ff6b8f0cc81908f09ded2', 'src/third_party/flatbuffers/src': @@ -511,7 +511,7 @@ deps = { 'condition': 'checkout_linux', }, 'src/third_party/freetype/src': - 'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@5987a9f51d7002f7ca375a34c2b857841d82a91c', + 'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@57a873045016c7866a298f5e9faa830e6ec24cf9', 'src/third_party/harfbuzz-ng/src': 'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@9f83bbbe64654b45ba5bb06927ff36c2e7588495', 'src/third_party/google_benchmark/src': { @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@806658eb4d5ccf58c7cf07b5127539886d16ded0', + 'https://chromium.googlesource.com/chromium/src/tools@147300be822ca0a93c69afc958d1b169bdcf6bdd', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'jxj98gNBLn43NAe-JF3GXSRicC44jGfwPTmq4vJvFHYC', + 'version': 'N3BICdmBe8aMPRI6ZbZ4tNuSs26ziJ1PctmwDJnJ2p0C', }, ], 'condition': 'checkout_android and non_git_source', @@ -785,7 +785,7 @@ deps = { }, 'src/third_party/zstd/src': { - 'url': Var('chromium_git') + '/external/github.com/facebook/zstd.git' + '@' + 'd654fca78690fa15cceb8058ac47454d914a0e63', + 'url': Var('chromium_git') + '/external/github.com/facebook/zstd.git' + '@' + 'f9938c217da17ec3e9dcd2a2d99c5cf39536aeb9', 'condition': 'checkout_android', }, From ae4c16c1f9bd48896269d2ea9721a55e9a28c250 Mon Sep 17 00:00:00 2001 From: Guido Urdaneta Date: Mon, 26 May 2025 11:11:35 +0200 Subject: [PATCH 285/292] Set voice_activity to true in receiver transformable audio frames. MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit No-Iwyu: No output from iwyu/apply-include-cleaner Bug: chromium:402391734 Change-Id: I220c798a5cf0283084ee73be210925ec88118aee Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393460 Commit-Queue: Guido Urdaneta Reviewed-by: Jakob Ivarsson‎ Cr-Commit-Position: refs/heads/main@{#44757} --- audio/channel_receive_frame_transformer_delegate.cc | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/audio/channel_receive_frame_transformer_delegate.cc b/audio/channel_receive_frame_transformer_delegate.cc index e3a3617b95..7995548043 100644 --- a/audio/channel_receive_frame_transformer_delegate.cc +++ b/audio/channel_receive_frame_transformer_delegate.cc @@ -99,7 +99,7 @@ class TransformableIncomingAudioFrame header_.extension.set_audio_level( audio_level_dbov.has_value() ? std::make_optional(webrtc::AudioLevel( - /*voice_activity=*/false, + /*voice_activity=*/true, std::min(*audio_level_dbov, static_cast(127u)))) : std::nullopt); } @@ -216,7 +216,7 @@ void ChannelReceiveFrameTransformerDelegate::ReceiveFrame( // TODO(crbug.com/webrtc/419746427): Add support for voice activity in // TransformableAudioFrameInterface. header.extension.set_audio_level(AudioLevel( - /*voice_activity=*/false, *transformed_frame->AudioLevel())); + /*voice_activity=*/true, *transformed_frame->AudioLevel())); } } else { auto* transformed_incoming_frame = From 6b8ef9d19d250c8991ff2f3445d32be99efa639c Mon Sep 17 00:00:00 2001 From: Fredrik Solenberg Date: Mon, 26 May 2025 14:08:25 +0200 Subject: [PATCH 286/292] Remove system_wrappers/../cpu_info.* This has been moved to rtc_base/. Bug: webrtc:42228262 Change-Id: I5a8a6afdffc1cde3c2ab8a26568ee1d0f54676d8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392941 Commit-Queue: Fredrik Solenberg Reviewed-by: Tomas Gunnarsson Commit-Queue: Tomas Gunnarsson Auto-Submit: Fredrik Solenberg Cr-Commit-Position: refs/heads/main@{#44758} --- system_wrappers/BUILD.gn | 3 --- system_wrappers/include/cpu_info.h | 29 ----------------------------- system_wrappers/source/cpu_info.cc | 23 ----------------------- 3 files changed, 55 deletions(-) delete mode 100644 system_wrappers/include/cpu_info.h delete mode 100644 system_wrappers/source/cpu_info.cc diff --git a/system_wrappers/BUILD.gn b/system_wrappers/BUILD.gn index a36adfa47b..a1fab28ad8 100644 --- a/system_wrappers/BUILD.gn +++ b/system_wrappers/BUILD.gn @@ -17,11 +17,9 @@ rtc_library("system_wrappers") { sources = [ "include/clock.h", "include/cpu_features_wrapper.h", - "include/cpu_info.h", "include/ntp_time.h", "source/clock.cc", "source/cpu_features.cc", - "source/cpu_info.cc", ] defines = [] @@ -31,7 +29,6 @@ rtc_library("system_wrappers") { "../api/units:timestamp", "../modules:module_api_public", "../rtc_base:checks", - "../rtc_base:cpu_info", "../rtc_base:logging", "../rtc_base:safe_conversions", "../rtc_base:timeutils", diff --git a/system_wrappers/include/cpu_info.h b/system_wrappers/include/cpu_info.h deleted file mode 100644 index 6bc99b6f2d..0000000000 --- a/system_wrappers/include/cpu_info.h +++ /dev/null @@ -1,29 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef SYSTEM_WRAPPERS_INCLUDE_CPU_INFO_H_ -#define SYSTEM_WRAPPERS_INCLUDE_CPU_INFO_H_ - -#include - -namespace webrtc { - -class CpuInfo { - public: - // Wrapper kept while updating dependencies. - static uint32_t DetectNumberOfCores(); - - private: - CpuInfo() {} -}; - -} // namespace webrtc - -#endif // SYSTEM_WRAPPERS_INCLUDE_CPU_INFO_H_ diff --git a/system_wrappers/source/cpu_info.cc b/system_wrappers/source/cpu_info.cc deleted file mode 100644 index 90bf72077a..0000000000 --- a/system_wrappers/source/cpu_info.cc +++ /dev/null @@ -1,23 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "system_wrappers/include/cpu_info.h" - -#include - -#include "rtc_base/cpu_info.h" - -namespace webrtc { - -uint32_t CpuInfo::DetectNumberOfCores() { - return cpu_info::DetectNumberOfCores(); -} - -} // namespace webrtc From 1499c20424d1a2a930990e1ecf06dffcf9677072 Mon Sep 17 00:00:00 2001 From: Danil Chapovalov Date: Mon, 26 May 2025 15:33:12 +0200 Subject: [PATCH 287/292] Cleanup arraysize usage in examples/ No-Iwyu: questionable suggestion unrelated to this patch. Bug: webrtc:407433458 Change-Id: I89978eed82c21c9c574467fc8b7be29257c39d0d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393480 Commit-Queue: Tomas Gunnarsson Auto-Submit: Danil Chapovalov Reviewed-by: Tomas Gunnarsson Cr-Commit-Position: refs/heads/main@{#44759} --- examples/BUILD.gn | 5 +- examples/peerconnection/client/defaults.cc | 8 ++- examples/peerconnection/client/main_wnd.cc | 60 +++++++++++-------- examples/peerconnection/server/data_socket.cc | 47 ++++++++------- .../peerconnection/server/peer_channel.cc | 31 +++++----- examples/peerconnection/server/utils.cc | 25 -------- examples/peerconnection/server/utils.h | 25 -------- 7 files changed, 84 insertions(+), 117 deletions(-) delete mode 100644 examples/peerconnection/server/utils.cc delete mode 100644 examples/peerconnection/server/utils.h diff --git a/examples/BUILD.gn b/examples/BUILD.gn index b1b5829395..4508146c93 100644 --- a/examples/BUILD.gn +++ b/examples/BUILD.gn @@ -713,7 +713,6 @@ if (is_linux || is_chromeos || is_win) { "../rtc_base:buffer", "../rtc_base:checks", "../rtc_base:logging", - "../rtc_base:macromagic", "../rtc_base:net_helpers", "../rtc_base:refcount", "../rtc_base:rtc_certificate_generator", @@ -796,8 +795,6 @@ if (is_linux || is_chromeos || is_win) { "peerconnection/server/main.cc", "peerconnection/server/peer_channel.cc", "peerconnection/server/peer_channel.h", - "peerconnection/server/utils.cc", - "peerconnection/server/utils.h", ] deps = [ "../rtc_base:checks", @@ -807,6 +804,8 @@ if (is_linux || is_chromeos || is_win) { "//third_party/abseil-cpp/absl/flags:flag", "//third_party/abseil-cpp/absl/flags:parse", "//third_party/abseil-cpp/absl/flags:usage", + "//third_party/abseil-cpp/absl/strings", + "//third_party/abseil-cpp/absl/strings:string_view", ] } rtc_executable("turnserver") { diff --git a/examples/peerconnection/client/defaults.cc b/examples/peerconnection/client/defaults.cc index ee3a9e1f0a..9ff7aeedf3 100644 --- a/examples/peerconnection/client/defaults.cc +++ b/examples/peerconnection/client/defaults.cc @@ -12,14 +12,16 @@ #include +#include +#include +#include + #ifdef WIN32 #include #else #include #endif -#include "rtc_base/arraysize.h" - const char kAudioLabel[] = "audio_label"; const char kVideoLabel[] = "video_label"; const char kStreamId[] = "stream_id"; @@ -50,7 +52,7 @@ std::string GetPeerName() { char computer_name[256]; std::string ret(GetEnvVarOrDefault("USERNAME", "user")); ret += '@'; - if (gethostname(computer_name, arraysize(computer_name)) == 0) { + if (gethostname(computer_name, std::size(computer_name)) == 0) { ret += computer_name; } else { ret += "host"; diff --git a/examples/peerconnection/client/main_wnd.cc b/examples/peerconnection/client/main_wnd.cc index 6b834bc466..a9d00d3d2d 100644 --- a/examples/peerconnection/client/main_wnd.cc +++ b/examples/peerconnection/client/main_wnd.cc @@ -12,11 +12,22 @@ #include +#include +#include +#include +#include +#include +#include + +#include "api/media_stream_interface.h" +#include "api/scoped_refptr.h" #include "api/video/i420_buffer.h" -#include "examples/peerconnection/client/defaults.h" -#include "rtc_base/arraysize.h" +#include "api/video/video_frame.h" +#include "api/video/video_frame_buffer.h" +#include "api/video/video_rotation.h" +#include "api/video/video_source_interface.h" +#include "examples/peerconnection/client/peer_connection_client.h" #include "rtc_base/checks.h" -#include "rtc_base/logging.h" #include "third_party/libyuv/include/libyuv/convert_argb.h" ATOM MainWnd::wnd_class_ = 0; @@ -53,7 +64,7 @@ HFONT GetDefaultFont() { std::string GetWindowText(HWND wnd) { char text[MAX_PATH] = {0}; - ::GetWindowTextA(wnd, &text[0], ARRAYSIZE(text)); + ::GetWindowTextA(wnd, &text[0], std::size(text)); return text; } @@ -254,11 +265,10 @@ void MainWnd::OnPaint() { ::SetStretchBltMode(dc_mem, HALFTONE); // Set the map mode so that the ratio will be maintained for us. - HDC all_dc[] = {ps.hdc, dc_mem}; - for (size_t i = 0; i < arraysize(all_dc); ++i) { - SetMapMode(all_dc[i], MM_ISOTROPIC); - SetWindowExtEx(all_dc[i], width, height, NULL); - SetViewportExtEx(all_dc[i], rc.right, rc.bottom, NULL); + for (HDC dc : {ps.hdc, dc_mem}) { + SetMapMode(dc, MM_ISOTROPIC); + SetWindowExtEx(dc, width, height, nullptr); + SetViewportExtEx(dc, rc.right, rc.bottom, nullptr); } HBITMAP bmp_mem = ::CreateCompatibleBitmap(ps.hdc, rc.right, rc.bottom); @@ -503,31 +513,31 @@ void MainWnd::LayoutConnectUI(bool show) { if (show) { const size_t kSeparator = 5; - size_t total_width = (ARRAYSIZE(windows) - 1) * kSeparator; + size_t total_width = (std::size(windows) - 1) * kSeparator; - for (size_t i = 0; i < ARRAYSIZE(windows); ++i) { - CalculateWindowSizeForText(windows[i].wnd, windows[i].text, - &windows[i].width, &windows[i].height); - total_width += windows[i].width; + for (Windows& window : windows) { + CalculateWindowSizeForText(window.wnd, window.text, &window.width, + &window.height); + total_width += window.width; } RECT rc; ::GetClientRect(wnd_, &rc); size_t x = (rc.right / 2) - (total_width / 2); size_t y = rc.bottom / 2; - for (size_t i = 0; i < ARRAYSIZE(windows); ++i) { - size_t top = y - (windows[i].height / 2); - ::MoveWindow(windows[i].wnd, static_cast(x), static_cast(top), - static_cast(windows[i].width), - static_cast(windows[i].height), TRUE); - x += kSeparator + windows[i].width; - if (windows[i].text[0] != 'X') - ::SetWindowTextW(windows[i].wnd, windows[i].text); - ::ShowWindow(windows[i].wnd, SW_SHOWNA); + for (Windows& window : windows) { + size_t top = y - (window.height / 2); + ::MoveWindow(window.wnd, static_cast(x), static_cast(top), + static_cast(window.width), + static_cast(window.height), TRUE); + x += kSeparator + window.width; + if (window.text[0] != 'X') + ::SetWindowTextW(window.wnd, window.text); + ::ShowWindow(window.wnd, SW_SHOWNA); } } else { - for (size_t i = 0; i < ARRAYSIZE(windows); ++i) { - ::ShowWindow(windows[i].wnd, SW_HIDE); + for (Windows& window : windows) { + ::ShowWindow(window.wnd, SW_HIDE); } } } diff --git a/examples/peerconnection/server/data_socket.cc b/examples/peerconnection/server/data_socket.cc index 75835aede3..9ece090a64 100644 --- a/examples/peerconnection/server/data_socket.cc +++ b/examples/peerconnection/server/data_socket.cc @@ -13,13 +13,20 @@ #include #include #include + +#include +#include + +#include "absl/strings/str_cat.h" +#include "absl/strings/string_view.h" +#include "rtc_base/checks.h" +#include "rtc_base/ip_address.h" +#include "rtc_base/net_helpers.h" + #if defined(WEBRTC_POSIX) #include #endif -#include "examples/peerconnection/server/utils.h" -#include "rtc_base/checks.h" - static const char kHeaderTerminator[] = "\r\n\r\n"; static const int kHeaderTerminatorLength = sizeof(kHeaderTerminator) - 1; @@ -138,8 +145,7 @@ bool DataSocket::Send(const std::string& status, if (!content_type.empty()) buffer += "Content-Type: " + content_type + "\r\n"; - buffer += - "Content-Length: " + int2str(static_cast(data.size())) + "\r\n"; + buffer += "Content-Length: " + absl::StrCat(data.size()) + "\r\n"; if (!extra_headers.empty()) { buffer += extra_headers; @@ -198,13 +204,12 @@ bool DataSocket::ParseMethodAndPath(const char* begin, size_t len) { }; const char* path = nullptr; - for (size_t i = 0; i < ARRAYSIZE(supported_methods); ++i) { - if (len > supported_methods[i].method_name_len && - isspace(begin[supported_methods[i].method_name_len]) && - strncmp(begin, supported_methods[i].method_name, - supported_methods[i].method_name_len) == 0) { - method_ = supported_methods[i].id; - path = begin + supported_methods[i].method_name_len; + for (const auto& method : supported_methods) { + if (len > method.method_name_len && + isspace(begin[method.method_name_len]) && + strncmp(begin, method.method_name, method.method_name_len) == 0) { + method_ = method.id; + path = begin + method.method_name_len; break; } } @@ -230,19 +235,17 @@ bool DataSocket::ParseContentLengthAndType(const char* headers, size_t length) { const char* end = headers + length; while (headers && headers < end) { if (!isspace(headers[0])) { - static const char kContentLength[] = "Content-Length:"; - static const char kContentType[] = "Content-Type:"; - if ((headers + ARRAYSIZE(kContentLength)) < end && - strncmp(headers, kContentLength, ARRAYSIZE(kContentLength) - 1) == - 0) { - headers += ARRAYSIZE(kContentLength) - 1; + static constexpr absl::string_view kContentLength = "Content-Length:"; + static constexpr absl::string_view kContentType = "Content-Type:"; + if (absl::string_view(headers, end - headers) + .starts_with(kContentLength)) { + headers += kContentLength.size(); while (headers[0] == ' ') ++headers; content_length_ = atoi(headers); - } else if ((headers + ARRAYSIZE(kContentType)) < end && - strncmp(headers, kContentType, ARRAYSIZE(kContentType) - 1) == - 0) { - headers += ARRAYSIZE(kContentType) - 1; + } else if (absl::string_view(headers, end - headers) + .starts_with(kContentType)) { + headers += kContentType.size(); while (headers[0] == ' ') ++headers; const char* type_end = strstr(headers, "\r\n"); diff --git a/examples/peerconnection/server/peer_channel.cc b/examples/peerconnection/server/peer_channel.cc index bb74124977..4b8a8d5253 100644 --- a/examples/peerconnection/server/peer_channel.cc +++ b/examples/peerconnection/server/peer_channel.cc @@ -14,9 +14,13 @@ #include #include +#include +#include +#include +#include "absl/strings/str_cat.h" +#include "absl/strings/string_view.h" #include "examples/peerconnection/server/data_socket.h" -#include "examples/peerconnection/server/utils.h" #include "rtc_base/checks.h" // Set to the peer id of the originator when messages are being @@ -62,7 +66,7 @@ ChannelMember::ChannelMember(DataSocket* socket) RTC_DCHECK(socket->PathEquals("/sign_in")); name_ = socket->request_arguments(); if (name_.empty()) - name_ = "peer_" + int2str(id_); + name_ = "peer_" + absl::StrCat(id_); else if (name_.length() > kMaxNameLength) name_.resize(kMaxNameLength); @@ -80,8 +84,7 @@ bool ChannelMember::TimedOut() { } std::string ChannelMember::GetPeerIdHeader() const { - std::string ret(kPeerIdHeader + int2str(id_) + "\r\n"); - return ret; + return kPeerIdHeader + absl::StrCat(id_) + "\r\n"; } bool ChannelMember::NotifyOfOtherMember(const ChannelMember& other) { @@ -179,21 +182,21 @@ ChannelMember* PeerChannel::Lookup(DataSocket* ds) const { return nullptr; size_t i = 0; - for (; i < ARRAYSIZE(kRequestPaths); ++i) { + for (; i < std::size(kRequestPaths); ++i) { if (ds->PathEquals(kRequestPaths[i])) break; } - if (i == ARRAYSIZE(kRequestPaths)) + if (i == std::size(kRequestPaths)) return nullptr; std::string args(ds->request_arguments()); - static const char kPeerId[] = "peer_id="; + static constexpr absl::string_view kPeerId = "peer_id="; size_t found = args.find(kPeerId); if (found == std::string::npos) return nullptr; - int id = atoi(&args[found + ARRAYSIZE(kPeerId) - 1]); + int id = atoi(&args[found + kPeerId.size()]); Members::const_iterator iter = members_.begin(); for (; iter != members_.end(); ++iter) { if (id == (*iter)->id()) { @@ -217,16 +220,16 @@ ChannelMember* PeerChannel::IsTargetedRequest(const DataSocket* ds) const { if (args == std::string::npos) return nullptr; size_t found; - const char kTargetPeerIdParam[] = "to="; + static constexpr absl::string_view kTargetPeerIdParam = "to="; do { found = path.find(kTargetPeerIdParam, args); if (found == std::string::npos) return nullptr; if (found == (args + 1) || path[found - 1] == '&') { - found += ARRAYSIZE(kTargetPeerIdParam) - 1; + found += kTargetPeerIdParam.size(); break; } - args = found + ARRAYSIZE(kTargetPeerIdParam) - 1; + args = found + kTargetPeerIdParam.size(); } while (true); int id = atoi(&path[found]); Members::const_iterator i = members_.begin(); @@ -246,8 +249,8 @@ bool PeerChannel::AddMember(DataSocket* ds) { HandleDeliveryFailures(&failures); members_.push_back(new_guy); - printf("New member added (total=%s): %s\n", - size_t2str(members_.size()).c_str(), new_guy->name().c_str()); + printf("New member added (total=%zu): %s\n", members_.size(), + new_guy->name().c_str()); // Let the newly connected peer know about other members of the channel. std::string content_type; @@ -279,7 +282,7 @@ void PeerChannel::OnClosing(DataSocket* ds) { break; } } - printf("Total connected: %s\n", size_t2str(members_.size()).c_str()); + printf("Total connected: %zu\n", members_.size()); } void PeerChannel::CheckForTimeout() { diff --git a/examples/peerconnection/server/utils.cc b/examples/peerconnection/server/utils.cc deleted file mode 100644 index 4cdaa0af15..0000000000 --- a/examples/peerconnection/server/utils.cc +++ /dev/null @@ -1,25 +0,0 @@ -/* - * Copyright 2011 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "examples/peerconnection/server/utils.h" - -#include - -#include "rtc_base/string_encode.h" - -using absl::StrCat; - -std::string int2str(int i) { - return absl::StrCat(i); -} - -std::string size_t2str(size_t i) { - return absl::StrCat(i); -} diff --git a/examples/peerconnection/server/utils.h b/examples/peerconnection/server/utils.h deleted file mode 100644 index 85c04a40e9..0000000000 --- a/examples/peerconnection/server/utils.h +++ /dev/null @@ -1,25 +0,0 @@ -/* - * Copyright 2011 The WebRTC Project Authors. All rights reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef EXAMPLES_PEERCONNECTION_SERVER_UTILS_H_ -#define EXAMPLES_PEERCONNECTION_SERVER_UTILS_H_ - -#include - -#include - -#ifndef ARRAYSIZE -#define ARRAYSIZE(x) (sizeof(x) / sizeof(x[0])) -#endif - -std::string int2str(int i); -std::string size_t2str(size_t i); - -#endif // EXAMPLES_PEERCONNECTION_SERVER_UTILS_H_ From 2bbf9a97f3e5a8ae1daad5746a9ff746a84ed030 Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Mon, 26 May 2025 09:03:09 -0700 Subject: [PATCH 288/292] Roll chromium_revision 1f52f5a706..2e7181a294 (1465420:1465565) Change log: https://chromium.googlesource.com/chromium/src/+log/1f52f5a706..2e7181a294 Full diff: https://chromium.googlesource.com/chromium/src/+/1f52f5a706..2e7181a294 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/6e9d1a97fc..241c958e93 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/26dfa1e660..8e0dfc75fb * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/08d976ff4f..2c371d52c6 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/739fbc1a64..e76c23b4cb * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/fe88e9476e..08b0a5a13e * src/third_party/androidx/cipd: N3BICdmBe8aMPRI6ZbZ4tNuSs26ziJ1PctmwDJnJ2p0C..SRLnRQtsEaeNt_-cy_acdUbX63elq51jmwvif531vHgC * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/938fc9953b..5477c6dfde * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/147300be82..84348bdf72 DEPS diff: https://chromium.googlesource.com/chromium/src/+/1f52f5a706..2e7181a294/DEPS BUG=None Change-Id: I3f63a56b220c14ae69f743ed8e93422ef8738c79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393600 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44760} --- DEPS | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/DEPS b/DEPS index 6c25d17d12..3a87cf49b6 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '1f52f5a706529fe517aa750a516faa90be92f56e', + 'chromium_revision': '2e7181a2946283d061f74a8b31e828a852f704e5', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@6e9d1a97fc906921130453717244d39a773ec684', + 'https://chromium.googlesource.com/chromium/src/base@241c958e9381b1e19abaf03672ba6c67af42a780', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@26dfa1e660573ea7db832c2c3b9eaa1fce398ff7', + 'https://chromium.googlesource.com/chromium/src/build@8e0dfc75fb53d0d2f73bb81ecbb7c01a1f1f17e6', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@1fc7350e65e9d7848c083b83aaf67611e74a5654', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@08d976ff4f97ccfae36c7a07e361f6a51aea95bd', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@2c371d52c671d1a1dd211b05e66b9b66bcb1ff94', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@739fbc1a64b4a6413515887b85e73ac22fb9ef44', + 'https://chromium.googlesource.com/chromium/src/testing@e76c23b4cbe951ed8d686b05f40865f21c7faf0c', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@fe88e9476e499896a972e9ad2cf227698199e4a8', + 'https://chromium.googlesource.com/chromium/src/third_party@08b0a5a13ec33692fb74fb5b83a1bc2bb01d6312', 'src/buildtools/linux64': { 'packages': [ @@ -484,7 +484,7 @@ deps = { 'src/third_party/breakpad/breakpad': 'https://chromium.googlesource.com/breakpad/breakpad.git@2625edb085169e92cf036c236ac79ab594a7b1cc', 'src/third_party/catapult': - 'https://chromium.googlesource.com/catapult.git@938fc9953b4159851a52022b4f2a90cddedba2fc', + 'https://chromium.googlesource.com/catapult.git@5477c6dfde1132b685c73edc16e1bc71449a691d', 'src/third_party/ced/src': { 'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5', }, @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@147300be822ca0a93c69afc958d1b169bdcf6bdd', + 'https://chromium.googlesource.com/chromium/src/tools@84348bdf72caa5f9d669afe2c9a38cd8c3f9a6b8', 'src/third_party/espresso': { 'packages': [ @@ -695,7 +695,7 @@ deps = { 'packages': [ { 'package': 'chromium/third_party/androidx', - 'version': 'N3BICdmBe8aMPRI6ZbZ4tNuSs26ziJ1PctmwDJnJ2p0C', + 'version': 'SRLnRQtsEaeNt_-cy_acdUbX63elq51jmwvif531vHgC', }, ], 'condition': 'checkout_android and non_git_source', From 7ade4a3e976e0c42a5638188b8756c9aa4b96c44 Mon Sep 17 00:00:00 2001 From: Fredrik Solenberg Date: Mon, 26 May 2025 17:56:02 +0200 Subject: [PATCH 289/292] Remove unused function: webrtc::GetCPUInfoNoASM() Bug: webrtc:42228262 Change-Id: I1d0afcee6132c5ee0c41813423d77534d4f71739 No-Iwyu: Avoid unrelated changes Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393580 Reviewed-by: Henrik Andreassson Auto-Submit: Fredrik Solenberg Commit-Queue: Fredrik Solenberg Cr-Commit-Position: refs/heads/main@{#44761} --- system_wrappers/include/cpu_features_wrapper.h | 5 +---- system_wrappers/source/cpu_features.cc | 6 ------ 2 files changed, 1 insertion(+), 10 deletions(-) diff --git a/system_wrappers/include/cpu_features_wrapper.h b/system_wrappers/include/cpu_features_wrapper.h index 254e2d8dda..d73b12abe8 100644 --- a/system_wrappers/include/cpu_features_wrapper.h +++ b/system_wrappers/include/cpu_features_wrapper.h @@ -11,7 +11,7 @@ #ifndef SYSTEM_WRAPPERS_INCLUDE_CPU_FEATURES_WRAPPER_H_ #define SYSTEM_WRAPPERS_INCLUDE_CPU_FEATURES_WRAPPER_H_ -#include +#include namespace webrtc { @@ -29,9 +29,6 @@ enum { // Returns true if the CPU supports the feature. int GetCPUInfo(CPUFeature feature); -// No CPU feature is available => straight C path. -int GetCPUInfoNoASM(CPUFeature feature); - // Return the features in an ARM device. // It detects the features in the hardware platform, and returns supported // values in the above enum definition as a bitmask. diff --git a/system_wrappers/source/cpu_features.cc b/system_wrappers/source/cpu_features.cc index 4a6170c520..3050ad8f09 100644 --- a/system_wrappers/source/cpu_features.cc +++ b/system_wrappers/source/cpu_features.cc @@ -19,12 +19,6 @@ namespace webrtc { -// No CPU feature is available => straight C path. -int GetCPUInfoNoASM(CPUFeature feature) { - (void)feature; - return 0; -} - #if defined(WEBRTC_ARCH_X86_FAMILY) #if defined(WEBRTC_ENABLE_AVX2) From b2fa2b875d6929ff0ba92217daf741172e29bd64 Mon Sep 17 00:00:00 2001 From: Per K Date: Fri, 23 May 2025 14:40:31 +0000 Subject: [PATCH 290/292] Use TransportPacketFeedback as input to TransportFeedbackDemuxer MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This change refactor TransportFeedbackDemuxer to directly use the TransportPacketFeedback from TransporFeedbackAdapter instead of using an internal history to map a transport sequence number to RTP packets. The purpose is to ensure TransportFeedbackDemuxer works the same way regardless of if RFC8888 feedback or TWCC is used. Bug: webrtc:42225697 Change-Id: I7af2dbbfab1af89e9cd25c6304ec6154430e2127 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393341 Commit-Queue: Per Kjellander Reviewed-by: Erik Språng Cr-Commit-Position: refs/heads/main@{#44762} --- api/transport/network_types.h | 10 +- call/rtp_transport_controller_send.cc | 16 +--- .../rtp/transport_feedback_adapter.cc | 20 +++- .../rtp/transport_feedback_adapter.h | 5 +- .../transport_feedback_adapter_unittest.cc | 5 + .../rtp/transport_feedback_demuxer.cc | 47 +++------- .../rtp/transport_feedback_demuxer.h | 11 +-- .../transport_feedback_demuxer_unittest.cc | 92 +++++++++---------- 8 files changed, 87 insertions(+), 119 deletions(-) diff --git a/api/transport/network_types.h b/api/transport/network_types.h index d229d77bfc..79bcc12d44 100644 --- a/api/transport/network_types.h +++ b/api/transport/network_types.h @@ -154,13 +154,18 @@ struct RTC_EXPORT TransportLossReport { }; // Packet level feedback - struct RTC_EXPORT PacketResult { class ReceiveTimeOrder { public: bool operator()(const PacketResult& lhs, const PacketResult& rhs); }; + struct RtpPacketInfo { + uint32_t ssrc = 0; + uint16_t rtp_sequence_number = 0; + bool is_retransmission = false; + }; + PacketResult(); PacketResult(const PacketResult&); ~PacketResult(); @@ -170,6 +175,9 @@ struct RTC_EXPORT PacketResult { SentPacket sent_packet; Timestamp receive_time = Timestamp::PlusInfinity(); EcnMarking ecn = EcnMarking::kNotEct; + + // `rtp_packet_info` is only set if the feedback is related to a RTP packet. + std::optional rtp_packet_info; }; struct RTC_EXPORT TransportPacketsFeedback { diff --git a/call/rtp_transport_controller_send.cc b/call/rtp_transport_controller_send.cc index 992e3a436c..dcf12b2159 100644 --- a/call/rtp_transport_controller_send.cc +++ b/call/rtp_transport_controller_send.cc @@ -50,7 +50,6 @@ #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtcp_packet/congestion_control_feedback.h" #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" -#include "modules/rtp_rtcp/source/rtp_header_extensions.h" #include "modules/rtp_rtcp/source/rtp_rtcp_interface.h" #include "rtc_base/checks.h" #include "rtc_base/experiments/field_trial_parser.h" @@ -624,15 +623,6 @@ void RtpTransportControllerSend::NotifyBweOfPacedSentPacket( RTC_DCHECK_NOTREACHED() << "Unknown packet type"; return; } - if (packet.HasExtension()) { - // TODO: bugs.webrtc.org/42225697 - Refactor TransportFeedbackDemuxer to use - // TransportPacketsFeedback instead of directly using - // rtcp::TransportFeedback. For now, only use it if TransportSeqeunce number - // header extension is used. - RtpPacketSendInfo packet_info = - RtpPacketSendInfo::From(packet, pacing_info); - feedback_demuxer_.AddPacket(packet_info); - } Timestamp creation_time = Timestamp::Millis(env_.clock().TimeInMilliseconds()); transport_feedback_adapter_.AddPacket( @@ -652,7 +642,6 @@ void RtpTransportControllerSend::OnTransportFeedback( const rtcp::TransportFeedback& feedback) { RTC_DCHECK_RUN_ON(&sequence_checker_); ++transport_cc_feedback_count_; - feedback_demuxer_.OnTransportFeedback(feedback); std::optional feedback_msg = transport_feedback_adapter_.ProcessTransportFeedback(feedback, receive_time); @@ -666,9 +655,6 @@ void RtpTransportControllerSend::OnCongestionControlFeedback( const rtcp::CongestionControlFeedback& feedback) { RTC_DCHECK_RUN_ON(&sequence_checker_); ++feedback_count_; - // TODO: bugs.webrtc.org/42225697 - update feedback demuxer for RFC 8888. - // Suggest feedback_demuxer_.OnTransportFeedback use TransportPacketFeedback - // instead. See usage in OnTransportFeedback. std::optional feedback_msg = transport_feedback_adapter_.ProcessCongestionControlFeedback( feedback, receive_time); @@ -689,6 +675,8 @@ void RtpTransportControllerSend::HandleTransportPacketsFeedback( << " ECN capable. Stop sending ECT(1)."; packet_router_.ConfigureForRfc8888Feedback(sending_packets_as_ect1_); } + + feedback_demuxer_.OnTransportFeedback(feedback); if (controller_) PostUpdates(controller_->OnTransportPacketsFeedback(feedback)); diff --git a/modules/congestion_controller/rtp/transport_feedback_adapter.cc b/modules/congestion_controller/rtp/transport_feedback_adapter.cc index 53a12f1e67..3b38b6dfa6 100644 --- a/modules/congestion_controller/rtp/transport_feedback_adapter.cc +++ b/modules/congestion_controller/rtp/transport_feedback_adapter.cc @@ -116,6 +116,8 @@ void TransportFeedbackAdapter::AddPacket(const RtpPacketToSend& packet_to_send, feedback.sent.pacing_info = pacing_info; feedback.ssrc = packet_to_send.Ssrc(); feedback.rtp_sequence_number = packet_to_send.SequenceNumber(); + feedback.is_retransmission = + packet_to_send.packet_type() == RtpPacketMediaType::kRetransmission; while (!history_.empty() && creation_time - history_.begin()->second.creation_time > @@ -223,14 +225,18 @@ TransportFeedbackAdapter::ProcessTransportFeedback( ++failed_lookups; return; } - if (delta_since_base.IsFinite()) { - packet_feedback->receive_time = - current_offset_ + delta_since_base.RoundDownTo(TimeDelta::Millis(1)); - } + if (packet_feedback->network_route == network_route_) { PacketResult result; result.sent_packet = packet_feedback->sent; - result.receive_time = packet_feedback->receive_time; + if (delta_since_base.IsFinite()) { + result.receive_time = current_offset_ + delta_since_base.RoundDownTo( + TimeDelta::Millis(1)); + } + result.rtp_packet_info = { + .ssrc = packet_feedback->ssrc, + .rtp_sequence_number = packet_feedback->rtp_sequence_number, + .is_retransmission = packet_feedback->is_retransmission}; packet_result_vector.push_back(result); } else { ++ignored; @@ -301,6 +307,10 @@ TransportFeedbackAdapter::ProcessCongestionControlFeedback( supports_ecn &= packet_info.ecn != EcnMarking::kNotEct; } result.ecn = packet_info.ecn; + result.rtp_packet_info = { + .ssrc = packet_feedback->ssrc, + .rtp_sequence_number = packet_feedback->rtp_sequence_number, + .is_retransmission = packet_feedback->is_retransmission}; packet_result_vector.push_back(result); } diff --git a/modules/congestion_controller/rtp/transport_feedback_adapter.h b/modules/congestion_controller/rtp/transport_feedback_adapter.h index 7f6e9427ad..10e06cae93 100644 --- a/modules/congestion_controller/rtp/transport_feedback_adapter.h +++ b/modules/congestion_controller/rtp/transport_feedback_adapter.h @@ -35,16 +35,13 @@ struct PacketFeedback { // Time corresponding to when this object was created. Timestamp creation_time = Timestamp::MinusInfinity(); SentPacket sent; - // Time corresponding to when the packet was received. Timestamped with the - // receiver's clock. For unreceived packet, Timestamp::PlusInfinity() is - // used. - Timestamp receive_time = Timestamp::PlusInfinity(); // The network route that this packet is associated with. NetworkRoute network_route; uint32_t ssrc = 0; uint16_t rtp_sequence_number = 0; + bool is_retransmission = false; }; class InFlightBytesTracker { diff --git a/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc b/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc index 2ceceaea5d..813edcc84a 100644 --- a/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc +++ b/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc @@ -112,6 +112,11 @@ void ComparePacketFeedbackVectors(const std::vector& truth, EXPECT_EQ(truth[i].packet_size, input[i].sent_packet.size); EXPECT_EQ(truth[i].pacing_info, input[i].sent_packet.pacing_info); EXPECT_EQ(truth[i].is_audio, input[i].sent_packet.audio); + EXPECT_EQ(input[i].rtp_packet_info->rtp_sequence_number, + truth[i].rtp_sequence_number); + EXPECT_EQ(input[i].rtp_packet_info->ssrc, truth[i].ssrc); + EXPECT_EQ(input[i].rtp_packet_info->is_retransmission, + truth[i].media_type == RtpPacketMediaType::kRetransmission); } } diff --git a/modules/congestion_controller/rtp/transport_feedback_demuxer.cc b/modules/congestion_controller/rtp/transport_feedback_demuxer.cc index 1c29108ffb..4b4f921395 100644 --- a/modules/congestion_controller/rtp/transport_feedback_demuxer.cc +++ b/modules/congestion_controller/rtp/transport_feedback_demuxer.cc @@ -9,22 +9,17 @@ */ #include "modules/congestion_controller/rtp/transport_feedback_demuxer.h" -#include #include #include #include #include "absl/algorithm/container.h" #include "api/sequence_checker.h" -#include "api/units/time_delta.h" +#include "api/transport/network_types.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" -#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" #include "rtc_base/checks.h" namespace webrtc { -namespace { -static const size_t kMaxPacketsInHistory = 5000; -} TransportFeedbackDemuxer::TransportFeedbackDemuxer() { // In case the construction thread is different from where the registration @@ -53,40 +48,20 @@ void TransportFeedbackDemuxer::DeRegisterStreamFeedbackObserver( observers_.erase(it); } -void TransportFeedbackDemuxer::AddPacket(const RtpPacketSendInfo& packet_info) { - RTC_DCHECK_RUN_ON(&observer_checker_); - - StreamFeedbackObserver::StreamPacketInfo info; - info.ssrc = packet_info.media_ssrc; - info.rtp_sequence_number = packet_info.rtp_sequence_number; - info.received = false; - info.is_retransmission = - packet_info.packet_type == RtpPacketMediaType::kRetransmission; - history_.insert( - {seq_num_unwrapper_.Unwrap(packet_info.transport_sequence_number), info}); - - while (history_.size() > kMaxPacketsInHistory) { - history_.erase(history_.begin()); - } -} - void TransportFeedbackDemuxer::OnTransportFeedback( - const rtcp::TransportFeedback& feedback) { + const TransportPacketsFeedback& feedback) { RTC_DCHECK_RUN_ON(&observer_checker_); std::vector stream_feedbacks; - feedback.ForAllPackets( - [&](uint16_t sequence_number, TimeDelta delta_since_base) { - RTC_DCHECK_RUN_ON(&observer_checker_); - auto it = history_.find(seq_num_unwrapper_.PeekUnwrap(sequence_number)); - if (it != history_.end()) { - auto packet_info = it->second; - packet_info.received = delta_since_base.IsFinite(); - stream_feedbacks.push_back(std::move(packet_info)); - if (delta_since_base.IsFinite()) - history_.erase(it); - } - }); + for (const PacketResult& packet : feedback.packet_feedbacks) { + if (packet.rtp_packet_info.has_value()) { + stream_feedbacks.push_back( + {.received = packet.receive_time.IsFinite(), + .ssrc = packet.rtp_packet_info->ssrc, + .rtp_sequence_number = packet.rtp_packet_info->rtp_sequence_number, + .is_retransmission = packet.rtp_packet_info->is_retransmission}); + } + } for (auto& observer : observers_) { std::vector selected_feedback; diff --git a/modules/congestion_controller/rtp/transport_feedback_demuxer.h b/modules/congestion_controller/rtp/transport_feedback_demuxer.h index 80661176d3..8999f3c301 100644 --- a/modules/congestion_controller/rtp/transport_feedback_demuxer.h +++ b/modules/congestion_controller/rtp/transport_feedback_demuxer.h @@ -11,13 +11,12 @@ #define MODULES_CONGESTION_CONTROLLER_RTP_TRANSPORT_FEEDBACK_DEMUXER_H_ #include -#include #include #include #include "api/sequence_checker.h" +#include "api/transport/network_types.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" -#include "rtc_base/numerics/sequence_number_unwrapper.h" #include "rtc_base/system/no_unique_address.h" #include "rtc_base/thread_annotations.h" @@ -42,16 +41,10 @@ class TransportFeedbackDemuxer final : public StreamFeedbackProvider { StreamFeedbackObserver* observer) override; void DeRegisterStreamFeedbackObserver( StreamFeedbackObserver* observer) override; - void AddPacket(const RtpPacketSendInfo& packet_info); - void OnTransportFeedback(const rtcp::TransportFeedback& feedback); + void OnTransportFeedback(const TransportPacketsFeedback& feedback); private: RTC_NO_UNIQUE_ADDRESS SequenceChecker observer_checker_; - RtpSequenceNumberUnwrapper seq_num_unwrapper_ - RTC_GUARDED_BY(&observer_checker_); - std::map history_ - RTC_GUARDED_BY(&observer_checker_); - // Maps a set of ssrcs to corresponding observer. Vectors are used rather than // set/map to ensure that the processing order is consistent independently of // the randomized ssrcs. diff --git a/modules/congestion_controller/rtp/transport_feedback_demuxer_unittest.cc b/modules/congestion_controller/rtp/transport_feedback_demuxer_unittest.cc index 204ce328d1..f37f43817d 100644 --- a/modules/congestion_controller/rtp/transport_feedback_demuxer_unittest.cc +++ b/modules/congestion_controller/rtp/transport_feedback_demuxer_unittest.cc @@ -12,9 +12,9 @@ #include #include +#include "api/transport/network_types.h" #include "api/units/timestamp.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" -#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" #include "test/gmock.h" #include "test/gtest.h" @@ -36,16 +36,19 @@ class MockStreamFeedbackObserver : public StreamFeedbackObserver { (override)); }; -RtpPacketSendInfo CreatePacket(uint32_t ssrc, - uint16_t rtp_sequence_number, - int64_t transport_sequence_number, - bool is_retransmission) { - RtpPacketSendInfo res; - res.media_ssrc = ssrc; - res.transport_sequence_number = transport_sequence_number; - res.rtp_sequence_number = rtp_sequence_number; - res.packet_type = is_retransmission ? RtpPacketMediaType::kRetransmission - : RtpPacketMediaType::kVideo; +PacketResult CreatePacket(uint32_t ssrc, + uint16_t rtp_sequence_number, + bool received, + bool is_retransmission) { + PacketResult res; + res.rtp_packet_info = { + .ssrc = ssrc, + .rtp_sequence_number = rtp_sequence_number, + .is_retransmission = is_retransmission, + }; + res.receive_time = + received ? Timestamp::Seconds(123) : Timestamp::PlusInfinity(); + return res; } } // namespace @@ -55,49 +58,38 @@ TEST(TransportFeedbackDemuxerTest, ObserverSanity) { MockStreamFeedbackObserver mock; demuxer.RegisterStreamFeedbackObserver({kSsrc}, &mock); - const uint16_t kRtpStartSeq = 55; - const int64_t kTransportStartSeq = 1; - demuxer.AddPacket(CreatePacket(kSsrc, kRtpStartSeq, kTransportStartSeq, - /*is_retransmission=*/false)); - demuxer.AddPacket(CreatePacket(kSsrc, kRtpStartSeq + 1, - kTransportStartSeq + 1, - /*is_retransmission=*/false)); - demuxer.AddPacket(CreatePacket(kSsrc, kRtpStartSeq + 2, - kTransportStartSeq + 2, - /*is_retransmission=*/true)); - - rtcp::TransportFeedback feedback; - feedback.SetBase(kTransportStartSeq, Timestamp::Millis(1)); - ASSERT_TRUE( - feedback.AddReceivedPacket(kTransportStartSeq, Timestamp::Millis(1))); - // Drop middle packet. - ASSERT_TRUE( - feedback.AddReceivedPacket(kTransportStartSeq + 2, Timestamp::Millis(3))); - - EXPECT_CALL( - mock, OnPacketFeedbackVector(ElementsAre( - AllOf(Field(&PacketInfo::received, true), - Field(&PacketInfo::ssrc, kSsrc), - Field(&PacketInfo::rtp_sequence_number, kRtpStartSeq), - Field(&PacketInfo::is_retransmission, false)), - AllOf(Field(&PacketInfo::received, false), - Field(&PacketInfo::ssrc, kSsrc), - Field(&PacketInfo::rtp_sequence_number, kRtpStartSeq + 1), - Field(&PacketInfo::is_retransmission, false)), - AllOf(Field(&PacketInfo::received, true), - Field(&PacketInfo::ssrc, kSsrc), - Field(&PacketInfo::rtp_sequence_number, kRtpStartSeq + 2), - Field(&PacketInfo::is_retransmission, true))))); + TransportPacketsFeedback feedback; + feedback.packet_feedbacks = {CreatePacket(kSsrc, /*rtp_sequence_number=*/55, + /*received=*/true, + /*is_retransmission=*/false), + CreatePacket(kSsrc, /*rtp_sequence_number=*/56, + /*received=*/false, + /*is_retransmission=*/false), + CreatePacket(kSsrc, /*rtp_sequence_number=*/57, + /*received=*/true, + /*is_retransmission=*/true)}; + EXPECT_CALL(mock, OnPacketFeedbackVector(ElementsAre( + AllOf(Field(&PacketInfo::received, true), + Field(&PacketInfo::ssrc, kSsrc), + Field(&PacketInfo::rtp_sequence_number, 55), + Field(&PacketInfo::is_retransmission, false)), + AllOf(Field(&PacketInfo::received, false), + Field(&PacketInfo::ssrc, kSsrc), + Field(&PacketInfo::rtp_sequence_number, 56), + Field(&PacketInfo::is_retransmission, false)), + AllOf(Field(&PacketInfo::received, true), + Field(&PacketInfo::ssrc, kSsrc), + Field(&PacketInfo::rtp_sequence_number, 57), + Field(&PacketInfo::is_retransmission, true))))); demuxer.OnTransportFeedback(feedback); demuxer.DeRegisterStreamFeedbackObserver(&mock); - demuxer.AddPacket( - CreatePacket(kSsrc, kRtpStartSeq + 3, kTransportStartSeq + 3, false)); - rtcp::TransportFeedback second_feedback; - second_feedback.SetBase(kTransportStartSeq + 3, Timestamp::Millis(4)); - ASSERT_TRUE(second_feedback.AddReceivedPacket(kTransportStartSeq + 3, - Timestamp::Millis(4))); + TransportPacketsFeedback second_feedback; + second_feedback.packet_feedbacks = { + CreatePacket(kSsrc, /*rtp_sequence_number=*/58, + /*received=*/true, + /*is_retransmission=*/false)}; EXPECT_CALL(mock, OnPacketFeedbackVector).Times(0); demuxer.OnTransportFeedback(second_feedback); From 9909b30d5e3c104d4dce7d066fc8422eef1dcc59 Mon Sep 17 00:00:00 2001 From: Palak Agarwal Date: Fri, 23 May 2025 21:42:19 +0200 Subject: [PATCH 291/292] Add tests for FullScreenPowerPointHandler These tests check if string manipulation is handled correctly by the FullScreenPowerPointHandler class. Bug: chromium:409473386 Change-Id: I2dad1627918a4ee240813de72013b8b18ec4d5ac Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/387880 Reviewed-by: Ilya Nikolaevskiy Commit-Queue: Palak Agarwal Reviewed-by: Elad Alon Reviewed-by: Henrik Andreassson Cr-Commit-Position: refs/heads/main@{#44763} --- modules/desktop_capture/BUILD.gn | 1 + .../full_screen_win_application_handler.cc | 193 ++++++------ .../win/full_screen_win_application_handler.h | 33 ++ ...screen_win_application_handler_unittest.cc | 293 ++++++++++++++++++ .../win/test_support/test_window.cc | 19 +- .../win/test_support/test_window.h | 5 +- 6 files changed, 436 insertions(+), 108 deletions(-) create mode 100644 modules/desktop_capture/win/full_screen_win_application_handler_unittest.cc diff --git a/modules/desktop_capture/BUILD.gn b/modules/desktop_capture/BUILD.gn index f00de0c3cb..a49c24fe46 100644 --- a/modules/desktop_capture/BUILD.gn +++ b/modules/desktop_capture/BUILD.gn @@ -170,6 +170,7 @@ if (rtc_include_tests) { "win/cursor_unittest.cc", "win/cursor_unittest_resources.h", "win/cursor_unittest_resources.rc", + "win/full_screen_win_application_handler_unittest.cc", "win/screen_capture_utils_unittest.cc", "win/screen_capturer_win_directx_unittest.cc", "win/test_support/test_window.cc", diff --git a/modules/desktop_capture/win/full_screen_win_application_handler.cc b/modules/desktop_capture/win/full_screen_win_application_handler.cc index 88ab8f617e..fa15b3e7e0 100644 --- a/modules/desktop_capture/win/full_screen_win_application_handler.cc +++ b/modules/desktop_capture/win/full_screen_win_application_handler.cc @@ -38,23 +38,27 @@ bool CheckWindowClassName(HWND window, const wchar_t* class_name) { WCHAR buffer[kMaxClassNameLength]; const int length = ::GetClassNameW(window, buffer, kMaxClassNameLength); - if (length <= 0) + if (length <= 0) { return false; + } - if (static_cast(length) != classNameLength) + if (static_cast(length) != classNameLength) { return false; + } return wcsncmp(buffer, class_name, classNameLength) == 0; } std::string WindowText(HWND window) { size_t len = ::GetWindowTextLength(window); - if (len == 0) + if (len == 0) { return std::string(); + } std::vector buffer(len + 1, 0); size_t copied = ::GetWindowTextW(window, buffer.data(), buffer.size()); - if (copied == 0) + if (copied == 0) { return std::string(); + } return webrtc::ToUtf8(buffer.data(), copied); } @@ -66,8 +70,9 @@ DWORD WindowProcessId(HWND window) { std::wstring FileNameFromPath(const std::wstring& path) { auto found = path.rfind(L"\\"); - if (found == std::string::npos) + if (found == std::string::npos) { return path; + } return path.substr(found + 1); } @@ -88,111 +93,101 @@ DesktopCapturer::SourceList GetProcessWindows( }); return result; } +} // namespace -class FullScreenPowerPointHandler : public FullScreenApplicationHandler { - public: - explicit FullScreenPowerPointHandler(DesktopCapturer::SourceId sourceId) - : FullScreenApplicationHandler(sourceId) {} - - ~FullScreenPowerPointHandler() override {} +FullScreenPowerPointHandler::FullScreenPowerPointHandler( + DesktopCapturer::SourceId sourceId) + : FullScreenApplicationHandler(sourceId) {} - DesktopCapturer::SourceId FindFullScreenWindow( - const DesktopCapturer::SourceList& window_list, - int64_t timestamp) const override { - if (!UseHeuristicFullscreenPowerPointWindows()) { - return 0; - } +DesktopCapturer::SourceId FullScreenPowerPointHandler::FindFullScreenWindow( + const DesktopCapturer::SourceList& window_list, + int64_t timestamp) const { + if (!UseHeuristicFullscreenPowerPointWindows() || window_list.empty()) { + return 0; + } - if (window_list.empty()) - return 0; + HWND original_window = reinterpret_cast(GetSourceId()); + if (GetWindowType(original_window) != WindowType::kEditor) { + return 0; + } - HWND original_window = reinterpret_cast(GetSourceId()); - if (GetWindowType(original_window) != WindowType::kEditor) - return 0; + DesktopCapturer::SourceList powerpoint_windows = GetProcessWindows( + window_list, WindowProcessId(original_window), original_window); - DesktopCapturer::SourceList powerpoint_windows = GetProcessWindows( - window_list, WindowProcessId(original_window), original_window); + // No relevant windows with the same process id as the `original_window` were + // found. + if (powerpoint_windows.empty()) { + return 0; + } - // No relevant window with the same process id as the `original_window` was - // found. - if (powerpoint_windows.empty()) - return 0; + const std::string original_document_title = + GetDocumentTitleFromEditor(original_window); + for (const auto& source : powerpoint_windows) { + HWND window = reinterpret_cast(source.id); - const std::string original_document_title = - GetDocumentTitleFromEditor(original_window); - for (const auto& source : powerpoint_windows) { - HWND window = reinterpret_cast(source.id); - - // Looking for fullscreen slide show window for the corresponding editor - // document - if (GetWindowType(window) == WindowType::kSlideShow && - GetDocumentTitleFromSlideShow(window) == original_document_title) { - return source.id; - } + // Looking for fullscreen slide show window for the corresponding editor + // document. + if (GetWindowType(window) == WindowType::kSlideShow && + GetDocumentTitleFromSlideShow(window) == original_document_title) { + return source.id; } - return 0; } + return 0; +} - private: - enum class WindowType { kEditor, kSlideShow, kOther }; - - WindowType GetWindowType(HWND window) const { - if (IsEditorWindow(window)) - return WindowType::kEditor; - else if (IsSlideShowWindow(window)) - return WindowType::kSlideShow; - else - return WindowType::kOther; +FullScreenPowerPointHandler::WindowType +FullScreenPowerPointHandler::GetWindowType(HWND window) const { + if (IsEditorWindow(window)) { + return WindowType::kEditor; + } else if (IsSlideShowWindow(window)) { + return WindowType::kSlideShow; } - constexpr static char kDocumentTitleSeparator = '-'; + return WindowType::kOther; +} - // This function extracts the title from the editor. It needs to be - // updated everytime PowerPoint changes its editor title format. Currently, it - // supports editor title in the format "Window - Title - PowerPoint". - std::string GetDocumentTitleFromEditor(HWND window) const { - std::string title = WindowText(window); - return std::string( - absl::StripAsciiWhitespace(absl::string_view(title).substr( - 0, title.rfind(kDocumentTitleSeparator)))); - } +constexpr static char kDocumentTitleSeparator = '-'; - // This function extracts the title from the slideshow when PowerPoint goes - // fullscreen. This function needs to be updated whenever PowerPoint changes - // its title format. Currently, it supports Fullscreen titles of the format - // "PowerPoint Slide Show - [Window - Title]" or "PowerPoint Slide Show - - // Window - Title". - std::string GetDocumentTitleFromSlideShow(HWND window) const { - std::string title = WindowText(window); - auto position = title.find(kDocumentTitleSeparator); - if (position != std::string::npos) { - title = std::string(absl::StripAsciiWhitespace( - absl::string_view(title).substr(position + 1, std::wstring::npos))); - } - - auto left_bracket_pos = title.find("["); - auto right_bracket_pos = title.rfind("]"); - if (left_bracket_pos == std::string::npos || - right_bracket_pos == std::string::npos || - right_bracket_pos <= left_bracket_pos) { - return title; - } +std::string FullScreenPowerPointHandler::GetDocumentTitleFromEditor( + HWND window) const { + std::string title = WindowText(window); + return std::string(absl::StripAsciiWhitespace(absl::string_view(title).substr( + 0, title.rfind(kDocumentTitleSeparator)))); +} - return std::string(absl::StripAsciiWhitespace(title.substr( - left_bracket_pos + 1, right_bracket_pos - left_bracket_pos - 1))); +std::string FullScreenPowerPointHandler::GetDocumentTitleFromSlideShow( + HWND window) const { + std::string title = WindowText(window); + size_t position = title.find(kDocumentTitleSeparator); + if (position != std::string::npos) { + title = absl::StripAsciiWhitespace( + absl::string_view(title).substr(position + 1, std::wstring::npos)); } - bool IsEditorWindow(HWND window) const { - return CheckWindowClassName(window, L"PPTFrameClass"); + size_t left_bracket_pos = title.find("["); + size_t right_bracket_pos = title.rfind("]"); + if (left_bracket_pos == std::string::npos || + right_bracket_pos == std::string::npos || + right_bracket_pos <= left_bracket_pos) { + return title; } - bool IsSlideShowWindow(HWND window) const { - const LONG style = ::GetWindowLong(window, GWL_STYLE); - const bool min_box = WS_MINIMIZEBOX & style; - const bool max_box = WS_MAXIMIZEBOX & style; - return !min_box && !max_box; - } -}; + return std::string(absl::StripAsciiWhitespace(title.substr( + left_bracket_pos + 1, right_bracket_pos - left_bracket_pos - 1))); +} + +bool FullScreenPowerPointHandler::IsEditorWindow(HWND window) const { + return CheckWindowClassName(window, L"PPTFrameClass"); +} + +bool FullScreenPowerPointHandler::IsSlideShowWindow(HWND window) const { + // TODO(https://crbug.com/409473386): Change this to use GetWindowLongPtr + // instead as recommended in the MS Windows API. + // https://learn.microsoft.com/en-us/windows/win32/api/winuser/nf-winuser-getwindowlongptra + const bool has_minimize_or_maximize_buttons = + ::GetWindowLong(window, GWL_STYLE) & (WS_MINIMIZEBOX | WS_MAXIMIZEBOX); + return !has_minimize_or_maximize_buttons; +} class OpenOfficeApplicationHandler : public FullScreenApplicationHandler { public: @@ -202,8 +197,9 @@ class OpenOfficeApplicationHandler : public FullScreenApplicationHandler { DesktopCapturer::SourceId FindFullScreenWindow( const DesktopCapturer::SourceList& window_list, int64_t timestamp) const override { - if (window_list.empty()) + if (window_list.empty()) { return 0; + } DWORD process_id = WindowProcessId(reinterpret_cast(GetSourceId())); @@ -235,8 +231,9 @@ class OpenOfficeApplicationHandler : public FullScreenApplicationHandler { return IsSlideShowWindow(x); }); - if (slide_show_window == app_windows.end()) + if (slide_show_window == app_windows.end()) { return 0; + } return slide_show_window->id; } @@ -266,22 +263,22 @@ std::wstring GetPathByWindowId(HWND window_id) { DWORD process_id = WindowProcessId(window_id); HANDLE process = ::OpenProcess(PROCESS_QUERY_LIMITED_INFORMATION, FALSE, process_id); - if (process == NULL) + if (process == NULL) { return L""; + } DWORD path_len = MAX_PATH; WCHAR path[MAX_PATH]; std::wstring result; - if (::QueryFullProcessImageNameW(process, 0, path, &path_len)) + if (::QueryFullProcessImageNameW(process, 0, path, &path_len)) { result = std::wstring(path, path_len); - else + } else { RTC_LOG_GLE(LS_ERROR) << "QueryFullProcessImageName failed."; + } ::CloseHandle(process); return result; } -} // namespace - std::unique_ptr CreateFullScreenWinApplicationHandler(DesktopCapturer::SourceId source_id) { std::unique_ptr result; diff --git a/modules/desktop_capture/win/full_screen_win_application_handler.h b/modules/desktop_capture/win/full_screen_win_application_handler.h index 286e8e5cb9..706e96eaef 100644 --- a/modules/desktop_capture/win/full_screen_win_application_handler.h +++ b/modules/desktop_capture/win/full_screen_win_application_handler.h @@ -14,9 +14,42 @@ #include #include "modules/desktop_capture/full_screen_application_handler.h" +#include "modules/desktop_capture/win/window_capture_utils.h" namespace webrtc { +class FullScreenPowerPointHandler : public FullScreenApplicationHandler { + public: + enum class WindowType { kEditor, kSlideShow, kOther }; + + explicit FullScreenPowerPointHandler(DesktopCapturer::SourceId sourceId); + + ~FullScreenPowerPointHandler() override {} + + DesktopCapturer::SourceId FindFullScreenWindow( + const DesktopCapturer::SourceList& window_list, + int64_t timestamp) const override; + + private: + WindowType GetWindowType(HWND window) const; + + // This function extracts the title from the editor. It needs to be + // updated every time PowerPoint changes its editor title format. Currently, + // it supports editor title in the format "Window - Title - PowerPoint". + std::string GetDocumentTitleFromEditor(HWND window) const; + + // This function extracts the title from the slideshow when PowerPoint goes + // fullscreen. This function needs to be updated whenever PowerPoint changes + // its title format. Currently, it supports Fullscreen titles of the format + // "PowerPoint Slide Show - [Window - Title]" or "PowerPoint Slide Show - + // Window - Title". + std::string GetDocumentTitleFromSlideShow(HWND window) const; + + bool IsEditorWindow(HWND window) const; + + bool IsSlideShowWindow(HWND window) const; +}; + std::unique_ptr CreateFullScreenWinApplicationHandler(DesktopCapturer::SourceId sourceId); diff --git a/modules/desktop_capture/win/full_screen_win_application_handler_unittest.cc b/modules/desktop_capture/win/full_screen_win_application_handler_unittest.cc new file mode 100644 index 0000000000..1fcd13bff0 --- /dev/null +++ b/modules/desktop_capture/win/full_screen_win_application_handler_unittest.cc @@ -0,0 +1,293 @@ +/* + * Copyright (c) 2025 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "modules/desktop_capture/win/full_screen_win_application_handler.h" + +#include + +#include "modules/desktop_capture/win/test_support/test_window.h" +#include "modules/desktop_capture/win/window_capture_utils.h" +#include "test/gmock.h" +#include "test/gtest.h" + +namespace webrtc { + +WindowInfo CreateTestWindow(const WCHAR* window_title, + const WCHAR* window_class) { + return CreateTestWindow(window_title, /*height=*/240, /*width=*/320, + /*extended_styles=*/0, window_class); +} + +class FullScreenWinApplicationHandlerTest : public ::testing::Test { + public: + void CreateEditorWindow(const WCHAR* title, + const WCHAR* window_class = L"PPTFrameClass") { + editor_window_info_ = CreateTestWindow(title, window_class); + full_screen_ppt_handler_ = std::make_unique( + reinterpret_cast(editor_window_info_.hwnd)); + } + + HWND CreateSlideShowWindow(const WCHAR* title) { + slide_show_window_info_ = + CreateTestWindow(title, /*window_class=*/L"screenClass"); + return slide_show_window_info_.hwnd; + } + + // FindFullScreenWindow returns a non-zero value when a full screen slide show + // is found. It returns NULL when it fails to find a corresponding full screen + // slide show. + HWND FindFullScreenWindow() { + DesktopCapturer::SourceList window_list; + EXPECT_TRUE(GetWindowList(GetWindowListFlags::kNone, &window_list)); + EXPECT_GT(window_list.size(), 0u); // Otherwise, faulty test. + + return reinterpret_cast( + full_screen_ppt_handler_->FindFullScreenWindow(window_list, + /*timestamp=*/0)); + } + + void TearDown() override { + DestroyTestWindow(editor_window_info_); + DestroyTestWindow(slide_show_window_info_); + } + + protected: + WindowInfo editor_window_info_; + WindowInfo slide_show_window_info_; + std::unique_ptr full_screen_ppt_handler_; +}; + +TEST_F(FullScreenWinApplicationHandlerTest, FullScreenWindowFoundForEditor) { + CreateEditorWindow(L"My - Title - PowerPoint"); + HWND slide_show = + CreateSlideShowWindow(L"PowerPoint Slide Show - [My - Title]"); + + EXPECT_EQ(FindFullScreenWindow(), slide_show); +} + +TEST_F(FullScreenWinApplicationHandlerTest, + FullScreenWindowFoundWhenEditorTitleHasExtraSpaces) { + CreateEditorWindow(L"My Title - PowerPoint"); + HWND slide_show = + CreateSlideShowWindow(L"PowerPoint Slide Show - [My Title ]"); + + EXPECT_EQ(FindFullScreenWindow(), slide_show); +} + +// This can happen in older PowerPoint versions, where full screen slide show +// title did not had brackets around the document name. +TEST_F(FullScreenWinApplicationHandlerTest, + FullScreenWindowFoundWhenSlideShowTitleHasNoBrackets) { + CreateEditorWindow(L"My - Title - PowerPoint"); + HWND slide_show = + CreateSlideShowWindow(L"PowerPoint Slide Show - My - Title"); + + EXPECT_EQ(FindFullScreenWindow(), slide_show); +} + +TEST_F(FullScreenWinApplicationHandlerTest, + FullScreenWindowFoundWhenEditorTitleHasUnevenlySpacedDashes) { + CreateEditorWindow(L"My -Test - Title - PowerPoint"); + HWND slide_show = + CreateSlideShowWindow(L"PowerPoint Slide Show - [My -Test - Title]"); + + EXPECT_EQ(FindFullScreenWindow(), slide_show); +} + +TEST_F(FullScreenWinApplicationHandlerTest, + FullScreenWindowFoundWhenEditorTitleHasMatchingBrackets) { + CreateEditorWindow(L"[My - Title] - PowerPoint"); + HWND slide_show = + CreateSlideShowWindow(L"PowerPoint Slide Show - [[My - Title]]"); + + EXPECT_EQ(FindFullScreenWindow(), slide_show); +} + +TEST_F(FullScreenWinApplicationHandlerTest, + FullScreenWindowFoundWhenEditorTitleHasAFrontRightBracket) { + CreateEditorWindow(L"[My - Title - PowerPoint"); + HWND slide_show = + CreateSlideShowWindow(L"PowerPoint Slide Show - [[My - Title]"); + + EXPECT_EQ(FindFullScreenWindow(), slide_show); +} + +TEST_F(FullScreenWinApplicationHandlerTest, + FullScreenWindowFoundWhenEditorTitleHasABackRightBracket) { + CreateEditorWindow(L"My - Title[ - PowerPoint"); + HWND slide_show = + CreateSlideShowWindow(L"PowerPoint Slide Show - [My - Title[]"); + + EXPECT_EQ(FindFullScreenWindow(), slide_show); +} + +TEST_F(FullScreenWinApplicationHandlerTest, + FullScreenWindowFoundWhenEditorTitleHasAFrontLeftBracket) { + CreateEditorWindow(L"]My - Title - PowerPoint"); + HWND slide_show = + CreateSlideShowWindow(L"PowerPoint Slide Show - []My - Title]"); + + EXPECT_EQ(FindFullScreenWindow(), slide_show); +} + +TEST_F(FullScreenWinApplicationHandlerTest, + FullScreenWindowFoundWhenEditorTitleHasABackLeftBracket) { + CreateEditorWindow(L"My - Title] - PowerPoint"); + HWND slide_show = + CreateSlideShowWindow(L"PowerPoint Slide Show - [My - Title]]"); + + EXPECT_EQ(FindFullScreenWindow(), slide_show); +} + +TEST_F(FullScreenWinApplicationHandlerTest, + FullScreenWindowFoundWhenEditorTitleHasMismatchingBrackets) { + CreateEditorWindow(L"]My - Title[ - PowerPoint"); + HWND slide_show = + CreateSlideShowWindow(L"PowerPoint Slide Show - []My - Title[]"); + + EXPECT_EQ(FindFullScreenWindow(), slide_show); +} + +TEST_F(FullScreenWinApplicationHandlerTest, + FullScreenWindowFoundWhenEditorTitleHasOnlyLeftBrackets) { + CreateEditorWindow(L"[[My - Title[ - PowerPoint"); + HWND slide_show = + CreateSlideShowWindow(L"PowerPoint Slide Show - [[[My - Title[]"); + + EXPECT_EQ(FindFullScreenWindow(), slide_show); +} + +TEST_F(FullScreenWinApplicationHandlerTest, + FullScreenWindowFoundWhenEditorTitleHasOnlyRightBrackets) { + CreateEditorWindow(L"]My - Title]] - PowerPoint"); + HWND slide_show = + CreateSlideShowWindow(L"PowerPoint Slide Show - []My - Title]]]"); + + EXPECT_EQ(FindFullScreenWindow(), slide_show); +} + +TEST_F(FullScreenWinApplicationHandlerTest, + FullScreenWindowFoundWhenEditorTitleHasNestedBrackets) { + CreateEditorWindow(L"[[My - Title]] - PowerPoint"); + HWND slide_show = + CreateSlideShowWindow(L"PowerPoint Slide Show - [[[My - Title]]]"); + + EXPECT_EQ(FindFullScreenWindow(), slide_show); +} + +TEST_F(FullScreenWinApplicationHandlerTest, + FullScreenWindowFoundWhenEditorTitleHasNestedBracketsInFront) { + CreateEditorWindow(L"[[My] - Title] - PowerPoint"); + HWND slide_show = + CreateSlideShowWindow(L"PowerPoint Slide Show - [[[My] - Title]]"); + + EXPECT_EQ(FindFullScreenWindow(), slide_show); +} + +TEST_F(FullScreenWinApplicationHandlerTest, + FullScreenWindowFoundWhenEditorTitleHasNestedBracketsInBack) { + CreateEditorWindow(L"[My - [Title]] - PowerPoint"); + HWND slide_show = + CreateSlideShowWindow(L"PowerPoint Slide Show - [[My - [Title]]]"); + + EXPECT_EQ(FindFullScreenWindow(), slide_show); +} + +TEST_F(FullScreenWinApplicationHandlerTest, + FullScreenWindowFoundWhenEditorWindowHasNoPowerPointInTitle) { + CreateEditorWindow(L"My - Title - Power"); + HWND slide_show = + CreateSlideShowWindow(L"PowerPoint Slide Show - [My - Title]"); + + EXPECT_EQ(FindFullScreenWindow(), slide_show); +} + +TEST_F(FullScreenWinApplicationHandlerTest, + FullScreenWindowFoundWhenSlideShowWindowHasNoPowerPointInTitle) { + CreateEditorWindow(L"My - Title - PowerPoint"); + HWND slide_show = CreateSlideShowWindow(L"Slide Show - [My - Title]"); + + EXPECT_EQ(FindFullScreenWindow(), slide_show); +} + +TEST_F(FullScreenWinApplicationHandlerTest, + FullScreenWindowNotFoundWhenEditorWindowHasWrongWindowClass) { + // Create editor window with class name "WrongClass" instead of + // "PPTFrameClass". + CreateEditorWindow(L"My - Title - PowerPoint", + /*window_class=*/L"WrongClass"); + HWND slide_show = + CreateSlideShowWindow(L"PowerPoint Slide Show - [My - Title]"); + + EXPECT_NE(FindFullScreenWindow(), slide_show); +} + +TEST_F(FullScreenWinApplicationHandlerTest, + FullScreenWindowNotFoundWhenFullScreenPPTHandlerUsesSlideShowWindow) { + CreateEditorWindow(L"My - Title - PowerPoint"); + + // Create FullScreenPowerPointHandler using the slide show WindowInfo. + HWND slide_show = + CreateSlideShowWindow(L"PowerPoint Slide Show - [My - Title]"); + full_screen_ppt_handler_.reset(new FullScreenPowerPointHandler( + reinterpret_cast(slide_show))); + + EXPECT_NE(FindFullScreenWindow(), slide_show); +} + +TEST_F(FullScreenWinApplicationHandlerTest, + CorrectFullScreenWindowFoundWhenMultipleSlideShowsHaveSimilarTitles) { + CreateEditorWindow(L"My - Title - PowerPoint"); + HWND correct_slide_show = + CreateSlideShowWindow(L"PowerPoint Slide Show - [My - Title]"); + std::vector wrong_slide_shows = { + CreateSlideShowWindow(L"PowerPoint Slide Show - [My - Title]"), + CreateSlideShowWindow(L"PowerPoint Slide Show - [My Title]"), + CreateSlideShowWindow(L"PowerPoint Slide Show - [My Title -]")}; + + ASSERT_THAT(wrong_slide_shows, + testing::Not(testing::Contains(correct_slide_show))); + EXPECT_EQ(FindFullScreenWindow(), correct_slide_show); +} + +TEST_F(FullScreenWinApplicationHandlerTest, + FullScreenWindowsFoundWhenMultipleEditorsAndSlideShowsExist) { + std::vector editors = { + CreateTestWindow(L"My - Title - PowerPoint", + /*window_class=*/L"PPTFrameClass"), + CreateTestWindow(L"[[My Title] - PowerPoint", + /*window_class=*/L"PPTFrameClass"), + CreateTestWindow(L"My Ttile - PowerPoint", + /*window_class=*/L"PPTFrameClass")}; + + std::vector> handlers; + for (auto& editor : editors) { + handlers.push_back(std::make_unique( + reinterpret_cast(editor.hwnd))); + } + + std::vector slide_shows = { + CreateSlideShowWindow(L"PowerPoint Slide Show - [My - Title]"), + CreateSlideShowWindow(L"PowerPoint Slide Show - [[[My Title]]"), + CreateSlideShowWindow(L"PowerPoint Slide Show - [My Ttile]")}; + + DesktopCapturer::SourceList window_list; + EXPECT_TRUE(GetWindowList(GetWindowListFlags::kNone, &window_list)); + EXPECT_GT(window_list.size(), 0u); + + for (size_t count = 0; count < handlers.size(); count++) { + EXPECT_EQ(reinterpret_cast( + handlers[count]->FindFullScreenWindow(window_list, + /*timestamp=*/0)), + slide_shows[count]); + } +} + +} // namespace webrtc diff --git a/modules/desktop_capture/win/test_support/test_window.cc b/modules/desktop_capture/win/test_support/test_window.cc index c07ff74aa5..2b7d132630 100644 --- a/modules/desktop_capture/win/test_support/test_window.cc +++ b/modules/desktop_capture/win/test_support/test_window.cc @@ -13,7 +13,6 @@ namespace webrtc { namespace { -const WCHAR kWindowClass[] = L"DesktopCaptureTestWindowClass"; const int kWindowHeight = 200; const int kWindowWidth = 300; @@ -42,7 +41,8 @@ LRESULT CALLBACK WindowProc(HWND hwnd, WindowInfo CreateTestWindow(const WCHAR* window_title, const int height, const int width, - const LONG extended_styles) { + const LONG extended_styles, + const WCHAR* window_class) { WindowInfo info; ::GetModuleHandleExW(GET_MODULE_HANDLE_EX_FLAG_FROM_ADDRESS | GET_MODULE_HANDLE_EX_FLAG_UNCHANGED_REFCOUNT, @@ -55,19 +55,20 @@ WindowInfo CreateTestWindow(const WCHAR* window_title, wcex.style = CS_HREDRAW | CS_VREDRAW; wcex.hInstance = info.window_instance; wcex.lpfnWndProc = &WindowProc; - wcex.lpszClassName = kWindowClass; + wcex.lpszClassName = window_class; info.window_class = ::RegisterClassExW(&wcex); // Use the default height and width if the caller did not supply the optional // height and width parameters, or if they supplied invalid values. int window_height = height <= 0 ? kWindowHeight : height; int window_width = width <= 0 ? kWindowWidth : width; - info.hwnd = - ::CreateWindowExW(extended_styles, kWindowClass, window_title, - WS_OVERLAPPEDWINDOW, CW_USEDEFAULT, CW_USEDEFAULT, - window_width, window_height, /*parent_window=*/nullptr, - /*menu_bar=*/nullptr, info.window_instance, - /*additional_params=*/nullptr); + info.hwnd = ::CreateWindowExW( + extended_styles, window_class, window_title, + (window_class == kWindowClass) ? WS_OVERLAPPEDWINDOW : WS_OVERLAPPED, + CW_USEDEFAULT, CW_USEDEFAULT, window_width, window_height, + /*parent_window=*/nullptr, + /*menu_bar=*/nullptr, info.window_instance, + /*additional_params=*/nullptr); ::ShowWindow(info.hwnd, SW_SHOWNORMAL); ::UpdateWindow(info.hwnd); diff --git a/modules/desktop_capture/win/test_support/test_window.h b/modules/desktop_capture/win/test_support/test_window.h index b055da7ccd..a30d4532c4 100644 --- a/modules/desktop_capture/win/test_support/test_window.h +++ b/modules/desktop_capture/win/test_support/test_window.h @@ -25,6 +25,8 @@ const uint8_t kTestWindowRValue = 191; const uint8_t kTestWindowGValue = 99; const uint8_t kTestWindowBValue = 12; +const WCHAR kWindowClass[] = L"DesktopCaptureTestWindowClass"; + struct WindowInfo { HWND hwnd; HINSTANCE window_instance; @@ -34,7 +36,8 @@ struct WindowInfo { WindowInfo CreateTestWindow(const WCHAR* window_title, int height = 0, int width = 0, - LONG extended_styles = 0); + LONG extended_styles = 0, + const WCHAR* window_class = kWindowClass); void ResizeTestWindow(HWND hwnd, int width, int height); From e4445e46a910eb407571ec0b0b8b7043562678cf Mon Sep 17 00:00:00 2001 From: chromium-webrtc-autoroll Date: Mon, 26 May 2025 15:01:56 -0700 Subject: [PATCH 292/292] Roll chromium_revision 2e7181a294..8a36d47cdc (1465565:1465671) Change log: https://chromium.googlesource.com/chromium/src/+log/2e7181a294..8a36d47cdc Full diff: https://chromium.googlesource.com/chromium/src/+/2e7181a294..8a36d47cdc Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/241c958e93..4ba67f727a * src/build: https://chromium.googlesource.com/chromium/src/build/+log/8e0dfc75fb..7adbc7e326 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/2c371d52c6..28bb2043d5 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/e76c23b4cb..44b0a8d794 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/08b0a5a13e..8a150db896 * src/third_party/compiler-rt/src: https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git/+log/9c6d38aa28..57196dd146 * src/third_party/freetype/src: https://chromium.googlesource.com/chromium/src/third_party/freetype2.git/+log/57a8730450..e07e56c7f1 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/84348bdf72..901b847ded DEPS diff: https://chromium.googlesource.com/chromium/src/+/2e7181a294..8a36d47cdc/DEPS BUG=None Change-Id: Ifd594169d3ae6f574dbca390f31ecf6243272b73 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/393621 Bot-Commit: Autoroller Commit-Queue: Autoroller Cr-Commit-Position: refs/heads/main@{#44764} --- DEPS | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/DEPS b/DEPS index 3a87cf49b6..2ae7d6b92a 100644 --- a/DEPS +++ b/DEPS @@ -10,7 +10,7 @@ vars = { # chromium waterfalls. More info at: crbug.com/570091. 'checkout_configuration': 'default', 'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"', - 'chromium_revision': '2e7181a2946283d061f74a8b31e828a852f704e5', + 'chromium_revision': '8a36d47cdc4cb85d99099ba00f864008f8a38dbe', # Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to # process the raw profiles produced by instrumented targets (built with @@ -73,9 +73,9 @@ vars = { deps = { # TODO(kjellander): Move this to be Android-only. 'src/base': - 'https://chromium.googlesource.com/chromium/src/base@241c958e9381b1e19abaf03672ba6c67af42a780', + 'https://chromium.googlesource.com/chromium/src/base@4ba67f727a84a10e32a417dc7e194f4fc6a23390', 'src/build': - 'https://chromium.googlesource.com/chromium/src/build@8e0dfc75fb53d0d2f73bb81ecbb7c01a1f1f17e6', + 'https://chromium.googlesource.com/chromium/src/build@7adbc7e3263f3ab427ba7c5ac7839a69082ff7fb', 'src/buildtools': 'https://chromium.googlesource.com/chromium/src/buildtools@1fc7350e65e9d7848c083b83aaf67611e74a5654', # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC. @@ -84,13 +84,13 @@ deps = { 'condition': 'checkout_android', }, 'src/ios': { - 'url': 'https://chromium.googlesource.com/chromium/src/ios@2c371d52c671d1a1dd211b05e66b9b66bcb1ff94', + 'url': 'https://chromium.googlesource.com/chromium/src/ios@28bb2043d52518c3358080a865271d72b7478e06', 'condition': 'checkout_ios', }, 'src/testing': - 'https://chromium.googlesource.com/chromium/src/testing@e76c23b4cbe951ed8d686b05f40865f21c7faf0c', + 'https://chromium.googlesource.com/chromium/src/testing@44b0a8d794b28dbd74614e5f5e7da2b407030647', 'src/third_party': - 'https://chromium.googlesource.com/chromium/src/third_party@08b0a5a13ec33692fb74fb5b83a1bc2bb01d6312', + 'https://chromium.googlesource.com/chromium/src/third_party@8a150db896356cd9b47f8c1a6d916347393f90f2', 'src/buildtools/linux64': { 'packages': [ @@ -323,7 +323,7 @@ deps = { 'src/third_party/clang-format/script': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@37f6e68a107df43b7d7e044fd36a13cbae3413f2', 'src/third_party/compiler-rt/src': - 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@9c6d38aa28100761bc2d6dcd2a590a37f8519adf', + 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt.git@57196dd146582915c955f6d388e31aea93220c51', 'src/third_party/libc++/src': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@a01c02c9d4acbdae3b7e8a2f3ee58579a9c29f96', 'src/third_party/libc++abi/src': @@ -511,7 +511,7 @@ deps = { 'condition': 'checkout_linux', }, 'src/third_party/freetype/src': - 'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@57a873045016c7866a298f5e9faa830e6ec24cf9', + 'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@e07e56c7f106b600262ab653d696b7b57f320127', 'src/third_party/harfbuzz-ng/src': 'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@9f83bbbe64654b45ba5bb06927ff36c2e7588495', 'src/third_party/google_benchmark/src': { @@ -656,7 +656,7 @@ deps = { 'condition': 'checkout_android', }, 'src/tools': - 'https://chromium.googlesource.com/chromium/src/tools@84348bdf72caa5f9d669afe2c9a38cd8c3f9a6b8', + 'https://chromium.googlesource.com/chromium/src/tools@901b847deda65d44f1bba16a9f47e2ea68a805be', 'src/third_party/espresso': { 'packages': [