Release 1.8.0 for M118 #27
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This CL adds `Y4mFrameGenerator` to support Y4M file input. Bug: webrtc:15210 Change-Id: If21e40a609b3c6f980a413fb183cd4dfb5123aab Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311520 Commit-Queue: Jianhui J Dai <jianhui.j.dai@intel.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40460}
Bug: None Change-Id: I823218c16c64a99353ad03806be22d60ffacbaad Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312765 Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#40461}
Change log: https://chromium.googlesource.com/chromium/src/+log/07494f47d3..264d933fd0 Full diff: https://chromium.googlesource.com/chromium/src/+/07494f47d3..264d933fd0 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/6c61eec692..25e26d80c7 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/926efe92e5..7fb08159d8 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/dbff2922b4..61bbb713a6 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/ec6a729cc9..7a04c5b9df * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/6b136b95fc..1addefcd45 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/fcefc88685..fd83c91087 DEPS diff: https://chromium.googlesource.com/chromium/src/+/07494f47d3..264d933fd0/DEPS No update to Clang. BUG=None Change-Id: Ia5397c7dc17d028dbc61904c3ad2c3ce98759e44 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312820 Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#40462}
test.webrtc.org is gone and webrtc-internals got some updates which make it more clear which dump is used BUG=None No-Try: true Change-Id: I040e54398ced78148345804a4ab4922f67de133d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312360 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Christoffer Jansson <jansson@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40463}
Make sure the callback is reset when tearing down the PipeWireSession and that there is no concurrent access to it, which can potentially lead to a crash. Bug: webrtc:15386 Change-Id: I0b09002fe0479dc1cd946c80684bcc5d8754d54a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311546 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Jan Grulich <grulja@gmail.com> Cr-Commit-Position: refs/heads/main@{#40464}
…trate BitrateTracker uses RateStatistics underneath, thus algorithm is the same, but it provides Timestamp/TimeDelta friendly interface Bug: webrtc:13757 Change-Id: I9f2fcb3d498b2a137b531b94b660d15aa273c4bf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312600 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40465}
Change log: https://chromium.googlesource.com/chromium/src/+log/264d933fd0..10080947c0 Full diff: https://chromium.googlesource.com/chromium/src/+/264d933fd0..10080947c0 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/25e26d80c7..3de7d110cb * src/build: https://chromium.googlesource.com/chromium/src/build/+log/7fb08159d8..3dd34519f9 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/61bbb713a6..a265a85ace * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/7a04c5b9df..85b0f51488 * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/1addefcd45..53a08ec089 * src/third_party/androidx: Bs_fkIRoZaXm-11bg5epoACmu5uzIxUdbAUPlMELw28C..ZIfpMhRlZ2Wm-GCtxgdXmEUojZK4r6xCyO7sLg51fjgC * src/third_party/perfetto: https://android.googlesource.com/platform/external/perfetto.git/+log/c00fefe9a6..e568f2855d * src/third_party/r8: Sz7S7AlqFPYB_t29P5b6i5K80Wq00mpvN2y8aNUAqo0C..O1BBWiBTIeNUcraX8STMtQXVaCleu6SJJjWCcnfhPLkC * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/fd83c91087..1a0f13f46a DEPS diff: https://chromium.googlesource.com/chromium/src/+/264d933fd0..10080947c0/DEPS No update to Clang. BUG=None Change-Id: I8e655d0cd1ff1e0cce4f89234dd046ffa264f98b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312920 Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#40466}
It was discovered that if libvpx reported a scalability mode in getStats (e.g. L3T3_KEY) and we then changed encoder implementation to an RTCVideoEncoder (such as MediaFoundationVideoEncodeAccelerator), getStats continued to report the old scalability mode value. This CL makes sure to clear the scalability mode on encoder implementation change or if the `codec_info` is missing. We should update MediaFoundation to report L1T1 as well, but in the meantime we should clear any old scalability modes values when the implementation changes (if the scalability mode is not known it is better to report nothing than to report an old misleading value). Bug: chromium:1426440 Change-Id: I1b5f324c4d29a00a6c73404cbee0faa2ae9cd843 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312900 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40467}
This CL does 2 things: - Change the DCHECK for payload_type_frequency to a CHECK (so that this error will be a crash not a divide-by-zero) - Change the replay helper that was used by the fuzzer to set the frequency of the packets to the video value (90K). Bug: chromium:1466826 Change-Id: I39941f250b1782b36a3bcddfd347a016591466ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312700 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40468}
BUG=None Change-Id: Ia5c27f0ae752810fabb53aea58f8731c6c314519 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311920 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40469}
Bug: None Change-Id: I1e535f912cbb843122060c26b8c955e8788951a4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313002 Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#40470}
Instead allow RtpPacket to exceed configured capacity when setting payload Bug: None Change-Id: I02fc080ffa3127ffbe0dade1f200dd7456a6a128 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312880 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40471}
…eo frame Bug: webrtc:13757 Change-Id: I0bef9cc17e599382cc2265d61a2a538f2534356f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312860 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40472}
since this may contain sensitive data, just like the address. BUG=None Change-Id: I3faa1512a15467cd5cc4bcbcaebadb736f1bae07 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313040 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40473}
Bug: None Change-Id: If442f10e9c9dfa774d9715932bb2b259665964d5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313141 Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#40474}
remove some of the templating around the Codec-derived types and use more modern C++ loops. BUG=webrtc:15214 Change-Id: I2710628741deca647e7ae88f5966ec7c7f12669a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311260 Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40475}
This CL also adds the prefix RTC_TESTING to `ios_internal_pure_release_bot_arm64` in order to avoid ODR violations. Bug: b/292472934 Change-Id: If63020e679c8670b4c797217eb38fc8c2954d422 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313240 Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40476}
ReconfigureEncoder() is supposed to recreate the send stream when switching between legacy and standard API paths to ensure that the upper and lower layers agree on the number of streams that exist (legacy = 3 encodings but 1 stream, standard = same as encodings). This successfully happened when going from standard to legacy but due to a bug in the condition this did not happen when going from legacy to standard because `scalability_mode_used` is always false here (even though the standard path does use a scalability mode). As a consequence, SetRtpParameters()'s call to UpdateSendState() resulted in a DCHECK-crash. In release builds we still avoid IOOB because active_modules.size() < rtp_streams.size() but to avoid mistakes like this happening again in the future, the DCHECK is promoted to a CHECK. The fix is to remove the scalability mode condition which didn't make sense anyway - changing scalability mode does not require recreation but recreation is necessary when number of streams change, whether or not scalability mode changed. TESTED = Using Simulcast Playground and switching back and forth between standard and legacy and changing scalability modes and confirming from stats, see https://crbug.com/1467455. Bug: chromium:1467455 Change-Id: Ide29742972ba83f2e0a11f135ab9b39c39d4eb49 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313280 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40477}
Bug: b/292167110 Change-Id: Idafa4ef23e87951bdd0276c29dee3e7f8be68476 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312580 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40478}
Bug: None Change-Id: I8efa68729b2ecc0106b6ec6335e660e1118e98f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313401 Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#40479}
This reverts commit 86cfe50. Reason for revert: Breaks roll into Chromium. https://ci.chromium.org/ui/p/chromium/builders/try/android-arm64-rel/264191/overview Original change's description: > Extract HasIPv4Enabled/HasIPv6Enabled. > > Bug: b/292167110 > Change-Id: Idafa4ef23e87951bdd0276c29dee3e7f8be68476 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312580 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#40478} Bug: b/292167110 Change-Id: Id7ebb5a673eac3c83a2236d4dfbaf31cce1eb9b6 No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313262 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40480}
The following can lead to ODR violations with symbols present in the app and in the test module: gn path out/Perf //:webrtc_perf_tests_module //sdk:helpers_objc //:webrtc_perf_tests_module --[public]--> //:webrtc_perf_tests_module_loadable_module --[private]--> //test:google_test_runner_objc --[private]--> //test:test_support_objc --[private]--> //sdk:helpers_objc After this CL: gn path out/Debug/ //:webrtc_perf_tests_module //sdk:helpers_objc No non-data paths found between these two targets. Bug: b/292472934 Change-Id: If8a6ecab9b34bea0f52fe91b3404d1afeca685fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313520 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40481}
EncoderStreamFactory has two code paths for creating a stream: the
"simulcast path" and the "default path". Only the former cares about
encoding paramter's maxBitrate. The latter assumes that
`encoder_config.max_bitrate_bps` already encompasses the maxBitrate of
the first encoding, but this is not always the case.
As of M113, when scalability mode is specified, {active,inactive} does
not count as simulcast stream but as a default stream represented by
encoding[0].
The problem is that `encoder_config.max_bitrate_bps` only includes
`encodings[0].max_bitrate_bps` when `encodings.size() == 1` which isn't
the case here.
This CL fixes the problem by making the "create default stream" code
path look at the first encoding's maxBitrate and remove existing
assumptions that `encoder_config.max_bitrate_bps` encompasses
`encodings[0].max_bitrate_bps`. This is a step in the right direction
since we're trying to remove all special cases and have encodings map
1:1 with SSRCs, so the "max bps of entire stream" should indeed be a
separate limit than the per-encoding limits and it was confusing that
sometimes it included and sometimes it excluded encoding[0]'s limit.
This issue did not happen in {inactive,active} since that code path
counts as "simulcast stream", so "default stream" is only ever
applicable for index 0.
TESTED=Simulcast Playground, see https://crbug.com/1455962.
Bug: chromium:1455962
Change-Id: I7c44925b780623b5979751e8959e972293648a3d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313282
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40482}
…merate instead of plain ints Bug: webrtc:13757 Change-Id: If2df5418dacd2b95387fa74a9bc226426b207aee Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313041 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40483}
Bug: None Change-Id: I101663769852602a5c7cdc72904be230ed2fdd12 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313483 Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#40484}
Bug: webrtc:15054 Change-Id: I23c9008e1979a56bba9421a00e4f0f8ff937d122 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313261 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40485}
Bug: None Change-Id: I872fe20b9ce901e8a5dd2dd814f00bb7d368e1ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313542 Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Christoffer Jansson <jansson@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40486}
Re-enable svc disabled test. Passes with the latest code. Bug: b/288825767 Change-Id: Ie022442ddbd95c8c8b56feecde873208ddec77b0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310449 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Marco Paniconi <marpan@google.com> Cr-Commit-Position: refs/heads/main@{#40487}
Bug: None Change-Id: Iac28ba32ac64485126d46154bc1728756bf4fef0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313780 Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#40488}
Bug: None Change-Id: I7854669de1216385e188bc53ee0cbd26c002c680 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312741 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40489}
Bug: None Change-Id: I1a655de1b6045531d294d4f2eae0d328d1188bda Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318705 Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#40682}
Bug: webrtc:15458 Change-Id: Ib90cb0b9a94e1f358685ed319538654b0c8ed5c4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318581 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40683}
Use Timestamp,TimeDelta, and DataRate types instead of plain integer types. Bug: webrtc:13756 Change-Id: I2a12f4abeeaa653dbd9534c297dbb72db63b012b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314502 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40684}
Bug: None Change-Id: Ibb10492791244ad785677353e32d0f1b0865fc21 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318724 Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#40685}
We get this automatically from the //build checkout now Bug: chromium:1432399 Change-Id: I223d7c5448244ed62821207068f979555617da57 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318560 Auto-Submit: Chong Gu <chonggu@google.com> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Zijie He <zijiehe@chromium.org> Cr-Commit-Position: refs/heads/main@{#40686}
Bug: chromium:1478193 Change-Id: If5207e7f740abcc43f74cf8eab30455a8bb0d5ac Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318622 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40687}
Combine all parameters into single struct so that it is easier to add and remove optional parameters Use Timestamp type instad of plain int to represent capture time Use rtc::ArrayView instead of pointer+size to represent payload Merge passing audio level into send function. Bug: webrtc:13757, webrtc:14870 Change-Id: I0386b710eb99b864334d61235add9abcde9bc69d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317442 Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40688}
Bug: webrtc:15469 Change-Id: Ib42705a49f1a9797edc93d9ca98ef8af173a0cec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318821 Reviewed-by: Nikita Gureev <gureev@google.com> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40689}
The code currently issues frames for encode before scheduling a new repeat. Swap this order to account for time taken by for slow encodes. Bug: webrtc:15456 Change-Id: I74177069e30c1bf65268231ffba033411a0f7b9a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40690}
Bug: webrtc:15469 Change-Id: I1e4149d1df255f393ef842605cb29a3e1d3e5b89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318840 Reviewed-by: Nikita Gureev <gureev@google.com> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40691}
Bug: None Change-Id: Ibb1f4a370822dd57a9296e9e36840d2e1e006c05 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318924 Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#40692}
Add thread checks to TcpPort code Bug: chromium:1478154 Change-Id: I045106c552dfcd8a8ab79218a59873fdc1d4326f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318061 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40693}
disallowing more than one ssrc-group with the same semantic and primary ssrc. BUG=chromium:1477075 Change-Id: I4bce0555cd49834725d9b97693d26c971bc5d5c2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318822 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40694}
Remove internal use of SignalSSLHandshakeError and prepare removal of sigslot dependency from SSLStreamAdapter. Bug: webrtc:11943 Change-Id: I9768e2e31529945620bdd8d0d285042bb2388b7b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318881 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40695}
Bug: chromium:1475195, chromium:1475944, chromium:1475909 Change-Id: Iaa9dc6570a8b70ec58efe0a64d468e1cae4cb484 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317504 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40696}
BitrateTracker uses same implementation as RateStatistics, but provides api using Timestamp and DataRate types instead of plain numbers Bug: webrtc:13756 Change-Id: Ie37fa58ede7590f870ec4376a64e7cf2c94431d7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318841 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40697}
This is following up on a discussion here: https://webrtc-review.googlesource.com/c/src/+/318061 Bug: none Change-Id: Idb572ca6d0aad8d791eb6ba80dc0f48292f9f244 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318883 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40698}
Bug: b/299058719 Change-Id: If356ba92bd49c5e650b3147ee94f28947318c4e5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318961 Reviewed-by: Jeremy Leconte <jleconte@google.com> Commit-Queue: Christoffer Jansson <jansson@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40699}
Bug: webrtc:11943 Change-Id: I8e0839363712d9d8b49c2f6cbdb5f3ac59d79219 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318882 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40700}
move this a bit later in the process since the current handling will consider two ssrc-lines with a cname in the same RTX FID ssrc-group to be part of separate streams due to the different randomly assigned msids. This leads to a misdetection as plan-b SDP. BUG=None Change-Id: Ie8acce9c2c7fb9eabda479b90e8cc7406dcb1696 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318820 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#40701}
Also removing has_slots depdency from OpenSSLStreamAdapter and moving it to the OpenSSLStreamAdapter subclass where it's still needed. Bug: webrtc:11943 Change-Id: Ibcae5ea1efff146d78b32bb0eca63d7f44ed08c1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318885 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40702}
Bug: b/299058719 Change-Id: I1485476a18f4774f3af1ea9254b7c31fdcbd74c4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319060 Reviewed-by: Jeremy Leconte <jleconte@google.com> Commit-Queue: Christoffer Jansson <jansson@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40703}
The FrameCadenceAdapter starts a delayed task to request a new refresh frame on receiving frame drop. However, the resulting RepeatingTaskHandle was not Stop()ed on destruction, leading to UAF. (cherry picked from commit fb98b01) Fixed: chromium:1478944 Change-Id: Iba441420953e989cfc7fcfd2f358b5b30f375786 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320200 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Original-Commit-Position: refs/heads/main@{#40747} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320420 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/branch-heads/5993@{#1} Cr-Branched-From: 5afcec0-refs/heads/main@{#40703}
BUG=chromium:1478690 (cherry picked from commit a8e3111) Change-Id: I7a1caad7bbd2fc82507b61b59be71546494a304c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319580 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Original-Commit-Position: refs/heads/main@{#40724} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320580 Cr-Commit-Position: refs/branch-heads/5993@{#2} Cr-Branched-From: 5afcec0-refs/heads/main@{#40703}
Speculative fix. Writing the test for it is more complex. (cherry picked from commit 83894d3) Bug: chromium:1483874 Change-Id: Icfaf1524b0499c609010753e0b6f3cadbd0e98f9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321480 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Original-Commit-Position: refs/heads/main@{#40820} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322124 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/branch-heads/5993@{#3} Cr-Branched-From: 5afcec0-refs/heads/main@{#40703}
elliot-choic
approved these changes
Feb 15, 2024
Fix camera rotation
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