-
Notifications
You must be signed in to change notification settings - Fork 6.6k
Commit
This commit does not belong to any branch on this repository, and may belong to a fork outside of the repository.
Implement AudioDeviceModule::GetStats API from WebRTC.
Should only be landed once https://webrtc-review.googlesource.com/c/src/+/291040 is landed. Bug: webrtc:14653 Change-Id: I31471064e8b0f88ad0904a400641586bf2382822 Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/4154699 Reviewed-by: Henrik Boström <hbos@chromium.org> Reviewed-by: Jeremy Roman <jbroman@chromium.org> Commit-Queue: Fredrik Hernqvist <fhernqvist@google.com> Cr-Commit-Position: refs/heads/main@{#1096068}
- Loading branch information
Fredrik Hernqvist
authored and
Chromium LUCI CQ
committed
Jan 24, 2023
1 parent
907555d
commit f639a71
Showing
12 changed files
with
257 additions
and
14 deletions.
There are no files selected for viewing
This file contains bidirectional Unicode text that may be interpreted or compiled differently than what appears below. To review, open the file in an editor that reveals hidden Unicode characters.
Learn more about bidirectional Unicode characters
This file contains bidirectional Unicode text that may be interpreted or compiled differently than what appears below. To review, open the file in an editor that reveals hidden Unicode characters.
Learn more about bidirectional Unicode characters
This file contains bidirectional Unicode text that may be interpreted or compiled differently than what appears below. To review, open the file in an editor that reveals hidden Unicode characters.
Learn more about bidirectional Unicode characters
This file contains bidirectional Unicode text that may be interpreted or compiled differently than what appears below. To review, open the file in an editor that reveals hidden Unicode characters.
Learn more about bidirectional Unicode characters
This file contains bidirectional Unicode text that may be interpreted or compiled differently than what appears below. To review, open the file in an editor that reveals hidden Unicode characters.
Learn more about bidirectional Unicode characters
This file contains bidirectional Unicode text that may be interpreted or compiled differently than what appears below. To review, open the file in an editor that reveals hidden Unicode characters.
Learn more about bidirectional Unicode characters
124 changes: 124 additions & 0 deletions
124
third_party/blink/renderer/modules/webrtc/webrtc_audio_device_impl_test.cc
This file contains bidirectional Unicode text that may be interpreted or compiled differently than what appears below. To review, open the file in an editor that reveals hidden Unicode characters.
Learn more about bidirectional Unicode characters
| Original file line number | Diff line number | Diff line change |
|---|---|---|
| @@ -0,0 +1,124 @@ | ||
| // Copyright 2023 The Chromium Authors | ||
| // Use of this source code is governed by a BSD-style license that can be | ||
| // found in the LICENSE file. | ||
|
|
||
| #include "third_party/blink/renderer/modules/webrtc/webrtc_audio_device_impl.h" | ||
|
|
||
| #include <memory> | ||
|
|
||
| #include "base/time/time.h" | ||
| #include "media/base/audio_bus.h" | ||
| #include "media/base/audio_glitch_info.h" | ||
| #include "media/base/audio_parameters.h" | ||
| #include "media/base/audio_timestamp_helper.h" | ||
| #include "testing/gmock/include/gmock/gmock.h" | ||
| #include "testing/gtest/include/gtest/gtest.h" | ||
| #include "third_party/blink/renderer/platform/webrtc/webrtc_source.h" | ||
| #include "third_party/webrtc/rtc_base/ref_counted_object.h" | ||
|
|
||
| namespace blink { | ||
|
|
||
| namespace { | ||
|
|
||
| class MockAudioTransport : public webrtc::AudioTransport { | ||
| public: | ||
| MockAudioTransport() = default; | ||
|
|
||
| MockAudioTransport(const MockAudioTransport&) = delete; | ||
| MockAudioTransport& operator=(const MockAudioTransport&) = delete; | ||
|
|
||
| MOCK_METHOD10(RecordedDataIsAvailable, | ||
| int32_t(const void* audioSamples, | ||
| size_t nSamples, | ||
| size_t nBytesPerSample, | ||
| size_t nChannels, | ||
| uint32_t samplesPerSec, | ||
| uint32_t totalDelayMS, | ||
| int32_t clockDrift, | ||
| uint32_t currentMicLevel, | ||
| bool keyPressed, | ||
| uint32_t& newMicLevel)); | ||
|
|
||
| MOCK_METHOD8(NeedMorePlayData, | ||
| int32_t(size_t nSamples, | ||
| size_t nBytesPerSample, | ||
| size_t nChannels, | ||
| uint32_t samplesPerSec, | ||
| void* audioSamples, | ||
| size_t& nSamplesOut, | ||
| int64_t* elapsed_time_ms, | ||
| int64_t* ntp_time_ms)); | ||
|
|
||
| MOCK_METHOD7(PullRenderData, | ||
| void(int bits_per_sample, | ||
| int sample_rate, | ||
| size_t number_of_channels, | ||
| size_t number_of_frames, | ||
| void* audio_data, | ||
| int64_t* elapsed_time_ms, | ||
| int64_t* ntp_time_ms)); | ||
| }; | ||
|
|
||
| const int kHardwareSampleRate = 44100; | ||
| const int kHardwareBufferSize = 512; | ||
|
|
||
| const media::AudioParameters kAudioParameters = | ||
| media::AudioParameters(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, | ||
| media::ChannelLayoutConfig::Stereo(), | ||
| kHardwareSampleRate, | ||
| kHardwareBufferSize); | ||
|
|
||
| } // namespace | ||
|
|
||
| class WebRtcAudioDeviceImplTest : public testing::Test { | ||
| public: | ||
| WebRtcAudioDeviceImplTest() | ||
| : audio_device_( | ||
| new rtc::RefCountedObject<blink::WebRtcAudioDeviceImpl>()), | ||
| audio_transport_(new MockAudioTransport()) { | ||
| audio_device_module()->Init(); | ||
| audio_device_module()->RegisterAudioCallback(audio_transport_.get()); | ||
| } | ||
|
|
||
| ~WebRtcAudioDeviceImplTest() override { audio_device_module()->Terminate(); } | ||
|
|
||
| protected: | ||
| webrtc::AudioDeviceModule* audio_device_module() { | ||
| return static_cast<webrtc::AudioDeviceModule*>(audio_device_.get()); | ||
| } | ||
|
|
||
| scoped_refptr<blink::WebRtcAudioDeviceImpl> audio_device_; | ||
| std::unique_ptr<MockAudioTransport> audio_transport_; | ||
| }; | ||
|
|
||
| // Verify that stats are accumulated during calls to RenderData and are | ||
| // available through GetStats(). | ||
| TEST_F(WebRtcAudioDeviceImplTest, GetStats) { | ||
| auto audio_bus = media::AudioBus::Create(kAudioParameters); | ||
| int sample_rate = kAudioParameters.sample_rate(); | ||
| auto audio_delay = base::Seconds(1); | ||
| base::TimeDelta current_time; | ||
| media::AudioGlitchInfo glitch_info; | ||
| glitch_info.duration = base::Seconds(2); | ||
| glitch_info.count = 3; | ||
|
|
||
| for (int i = 0; i < 10; i++) { | ||
| webrtc::AudioDeviceModule::Stats stats = *audio_device_->GetStats(); | ||
| EXPECT_EQ(stats.synthesized_samples_duration_s, | ||
| (base::Seconds(2) * i).InSecondsF()); | ||
| EXPECT_EQ(stats.synthesized_samples_events, 3ull * i); | ||
| EXPECT_EQ(stats.total_samples_count, | ||
| static_cast<uint64_t>(audio_bus->frames() * i)); | ||
| EXPECT_EQ(stats.total_playout_delay_s, | ||
| (audio_bus->frames() * i * base::Seconds(1)).InSecondsF()); | ||
| EXPECT_EQ(stats.total_samples_duration_s, | ||
| (media::AudioTimestampHelper::FramesToTime(audio_bus->frames(), | ||
| sample_rate) * | ||
| i) | ||
| .InSecondsF()); | ||
| audio_device_->RenderData(audio_bus.get(), sample_rate, audio_delay, | ||
| ¤t_time, glitch_info); | ||
| } | ||
| } | ||
|
|
||
| } // namespace blink |
This file contains bidirectional Unicode text that may be interpreted or compiled differently than what appears below. To review, open the file in an editor that reveals hidden Unicode characters.
Learn more about bidirectional Unicode characters
Oops, something went wrong.