| // Copyright 2013 The Chromium Authors. All rights reserved. |
| // Use of this source code is governed by a BSD-style license that can be |
| // found in the LICENSE file. |
| |
| #include "base/synchronization/waitable_event.h" |
| #include "base/test/test_timeouts.h" |
| #include "content/renderer/media/rtc_media_constraints.h" |
| #include "content/renderer/media/webrtc_audio_capturer.h" |
| #include "content/renderer/media/webrtc_audio_device_impl.h" |
| #include "content/renderer/media/webrtc_local_audio_source_provider.h" |
| #include "content/renderer/media/webrtc_local_audio_track.h" |
| #include "media/audio/audio_parameters.h" |
| #include "media/base/audio_bus.h" |
| #include "media/base/audio_capturer_source.h" |
| #include "testing/gmock/include/gmock/gmock.h" |
| #include "testing/gtest/include/gtest/gtest.h" |
| #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" |
| |
| using ::testing::_; |
| using ::testing::AnyNumber; |
| using ::testing::AtLeast; |
| using ::testing::Return; |
| |
| namespace content { |
| |
| namespace { |
| |
| ACTION_P(SignalEvent, event) { |
| event->Signal(); |
| } |
| |
| // A simple thread that we use to fake the audio thread which provides data to |
| // the |WebRtcAudioCapturer|. |
| class FakeAudioThread : public base::PlatformThread::Delegate { |
| public: |
| FakeAudioThread(const scoped_refptr<WebRtcAudioCapturer>& capturer, |
| const media::AudioParameters& params) |
| : capturer_(capturer), |
| thread_(), |
| closure_(false, false) { |
| DCHECK(capturer.get()); |
| audio_bus_ = media::AudioBus::Create(params); |
| } |
| |
| virtual ~FakeAudioThread() { DCHECK(thread_.is_null()); } |
| |
| // base::PlatformThread::Delegate: |
| virtual void ThreadMain() OVERRIDE { |
| while (true) { |
| if (closure_.IsSignaled()) |
| return; |
| |
| media::AudioCapturerSource::CaptureCallback* callback = |
| static_cast<media::AudioCapturerSource::CaptureCallback*>( |
| capturer_.get()); |
| audio_bus_->Zero(); |
| callback->Capture(audio_bus_.get(), 0, 0, false); |
| |
| // Sleep 1ms to yield the resource for the main thread. |
| base::PlatformThread::Sleep(base::TimeDelta::FromMilliseconds(1)); |
| } |
| } |
| |
| void Start() { |
| base::PlatformThread::CreateWithPriority( |
| 0, this, &thread_, base::kThreadPriority_RealtimeAudio); |
| CHECK(!thread_.is_null()); |
| } |
| |
| void Stop() { |
| closure_.Signal(); |
| base::PlatformThread::Join(thread_); |
| thread_ = base::PlatformThreadHandle(); |
| } |
| |
| private: |
| scoped_ptr<media::AudioBus> audio_bus_; |
| scoped_refptr<WebRtcAudioCapturer> capturer_; |
| base::PlatformThreadHandle thread_; |
| base::WaitableEvent closure_; |
| DISALLOW_COPY_AND_ASSIGN(FakeAudioThread); |
| }; |
| |
| class MockCapturerSource : public media::AudioCapturerSource { |
| public: |
| explicit MockCapturerSource(WebRtcAudioCapturer* capturer) |
| : capturer_(capturer) {} |
| MOCK_METHOD3(OnInitialize, void(const media::AudioParameters& params, |
| CaptureCallback* callback, |
| int session_id)); |
| MOCK_METHOD0(OnStart, void()); |
| MOCK_METHOD0(OnStop, void()); |
| MOCK_METHOD1(SetVolume, void(double volume)); |
| MOCK_METHOD1(SetAutomaticGainControl, void(bool enable)); |
| |
| virtual void Initialize(const media::AudioParameters& params, |
| CaptureCallback* callback, |
| int session_id) OVERRIDE { |
| DCHECK(params.IsValid()); |
| params_ = params; |
| OnInitialize(params, callback, session_id); |
| } |
| virtual void Start() OVERRIDE { |
| audio_thread_.reset(new FakeAudioThread(capturer_, params_)); |
| audio_thread_->Start(); |
| OnStart(); |
| } |
| virtual void Stop() OVERRIDE { |
| audio_thread_->Stop(); |
| audio_thread_.reset(); |
| OnStop(); |
| } |
| protected: |
| virtual ~MockCapturerSource() {} |
| |
| private: |
| scoped_ptr<FakeAudioThread> audio_thread_; |
| WebRtcAudioCapturer* capturer_; |
| media::AudioParameters params_; |
| }; |
| |
| // TODO(xians): Use MediaStreamAudioSink. |
| class MockMediaStreamAudioSink : public PeerConnectionAudioSink { |
| public: |
| MockMediaStreamAudioSink() {} |
| ~MockMediaStreamAudioSink() {} |
| int OnData(const int16* audio_data, |
| int sample_rate, |
| int number_of_channels, |
| int number_of_frames, |
| const std::vector<int>& channels, |
| int audio_delay_milliseconds, |
| int current_volume, |
| bool need_audio_processing, |
| bool key_pressed) OVERRIDE { |
| CaptureData(channels.size(), |
| sample_rate, |
| number_of_channels, |
| number_of_frames, |
| audio_delay_milliseconds, |
| current_volume, |
| need_audio_processing, |
| key_pressed); |
| return 0; |
| } |
| MOCK_METHOD8(CaptureData, |
| void(int number_of_network_channels, |
| int sample_rate, |
| int number_of_channels, |
| int number_of_frames, |
| int audio_delay_milliseconds, |
| int current_volume, |
| bool need_audio_processing, |
| bool key_pressed)); |
| MOCK_METHOD1(OnSetFormat, void(const media::AudioParameters& params)); |
| }; |
| |
| } // namespace |
| |
| class WebRtcLocalAudioTrackTest : public ::testing::Test { |
| protected: |
| virtual void SetUp() OVERRIDE { |
| params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
| media::CHANNEL_LAYOUT_STEREO, 2, 0, 48000, 16, 480); |
| capturer_ = WebRtcAudioCapturer::CreateCapturer(); |
| capturer_source_ = new MockCapturerSource(capturer_.get()); |
| EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_.get(), 0)) |
| .WillOnce(Return()); |
| capturer_->SetCapturerSource(capturer_source_, |
| params_.channel_layout(), |
| params_.sample_rate()); |
| } |
| |
| media::AudioParameters params_; |
| scoped_refptr<MockCapturerSource> capturer_source_; |
| scoped_refptr<WebRtcAudioCapturer> capturer_; |
| }; |
| |
| // Creates a capturer and audio track, fakes its audio thread, and |
| // connect/disconnect the sink to the audio track on the fly, the sink should |
| // get data callback when the track is connected to the capturer but not when |
| // the track is disconnected from the capturer. |
| TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) { |
| EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); |
| EXPECT_CALL(*capturer_source_.get(), OnStart()); |
| RTCMediaConstraints constraints; |
| scoped_refptr<WebRtcLocalAudioTrack> track = |
| WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL, |
| &constraints); |
| static_cast<WebRtcLocalAudioSourceProvider*>( |
| track->audio_source_provider())->SetSinkParamsForTesting(params_); |
| track->Start(); |
| EXPECT_TRUE(track->enabled()); |
| |
| // Connect a number of network channels to the audio track. |
| static const int kNumberOfNetworkChannels = 4; |
| for (int i = 0; i < kNumberOfNetworkChannels; ++i) { |
| static_cast<webrtc::AudioTrackInterface*>(track.get())-> |
| GetRenderer()->AddChannel(i); |
| } |
| scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink()); |
| const media::AudioParameters params = capturer_->audio_parameters(); |
| base::WaitableEvent event(false, false); |
| EXPECT_CALL(*sink, OnSetFormat(_)).WillOnce(Return()); |
| EXPECT_CALL(*sink, |
| CaptureData(kNumberOfNetworkChannels, |
| params.sample_rate(), |
| params.channels(), |
| params.sample_rate() / 100, |
| 0, |
| 0, |
| false, |
| false)).Times(AtLeast(1)) |
| .WillRepeatedly(SignalEvent(&event)); |
| track->AddSink(sink.get()); |
| |
| EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); |
| track->RemoveSink(sink.get()); |
| |
| EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return()); |
| capturer_->Stop(); |
| } |
| |
| // The same setup as ConnectAndDisconnectOneSink, but enable and disable the |
| // audio track on the fly. When the audio track is disabled, there is no data |
| // callback to the sink; when the audio track is enabled, there comes data |
| // callback. |
| // TODO(xians): Enable this test after resolving the racing issue that TSAN |
| // reports on MediaStreamTrack::enabled(); |
| TEST_F(WebRtcLocalAudioTrackTest, DISABLED_DisableEnableAudioTrack) { |
| EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); |
| EXPECT_CALL(*capturer_source_.get(), OnStart()); |
| RTCMediaConstraints constraints; |
| scoped_refptr<WebRtcLocalAudioTrack> track = |
| WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL, |
| &constraints); |
| static_cast<WebRtcLocalAudioSourceProvider*>( |
| track->audio_source_provider())->SetSinkParamsForTesting(params_); |
| track->Start(); |
| static_cast<webrtc::AudioTrackInterface*>(track.get())-> |
| GetRenderer()->AddChannel(0); |
| EXPECT_TRUE(track->enabled()); |
| EXPECT_TRUE(track->set_enabled(false)); |
| scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink()); |
| const media::AudioParameters params = capturer_->audio_parameters(); |
| base::WaitableEvent event(false, false); |
| EXPECT_CALL(*sink, OnSetFormat(_)).Times(1); |
| EXPECT_CALL(*sink, |
| CaptureData(1, |
| params.sample_rate(), |
| params.channels(), |
| params.sample_rate() / 100, |
| 0, |
| 0, |
| false, |
| false)).Times(0); |
| track->AddSink(sink.get()); |
| EXPECT_FALSE(event.TimedWait(TestTimeouts::tiny_timeout())); |
| |
| event.Reset(); |
| EXPECT_CALL(*sink, |
| CaptureData(1, |
| params.sample_rate(), |
| params.channels(), |
| params.sample_rate() / 100, |
| 0, |
| 0, |
| false, |
| false)).Times(AtLeast(1)) |
| .WillRepeatedly(SignalEvent(&event)); |
| EXPECT_TRUE(track->set_enabled(true)); |
| EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); |
| track->RemoveSink(sink.get()); |
| |
| EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return()); |
| capturer_->Stop(); |
| track = NULL; |
| } |
| |
| // Create multiple audio tracks and enable/disable them, verify that the audio |
| // callbacks appear/disappear. |
| // Flaky due to a data race, see http://crbug.com/295418 |
| TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) { |
| EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); |
| EXPECT_CALL(*capturer_source_.get(), OnStart()); |
| RTCMediaConstraints constraints; |
| scoped_refptr<WebRtcLocalAudioTrack> track_1 = |
| WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL, |
| &constraints); |
| static_cast<WebRtcLocalAudioSourceProvider*>( |
| track_1->audio_source_provider())->SetSinkParamsForTesting(params_); |
| track_1->Start(); |
| static_cast<webrtc::AudioTrackInterface*>(track_1.get())-> |
| GetRenderer()->AddChannel(0); |
| EXPECT_TRUE(track_1->enabled()); |
| scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink()); |
| const media::AudioParameters params = capturer_->audio_parameters(); |
| base::WaitableEvent event_1(false, false); |
| EXPECT_CALL(*sink_1, OnSetFormat(_)).WillOnce(Return()); |
| EXPECT_CALL(*sink_1, |
| CaptureData(1, |
| params.sample_rate(), |
| params.channels(), |
| params.sample_rate() / 100, |
| 0, |
| 0, |
| false, |
| false)).Times(AtLeast(1)) |
| .WillRepeatedly(SignalEvent(&event_1)); |
| track_1->AddSink(sink_1.get()); |
| EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout())); |
| |
| scoped_refptr<WebRtcLocalAudioTrack> track_2 = |
| WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL, |
| &constraints); |
| static_cast<WebRtcLocalAudioSourceProvider*>( |
| track_2->audio_source_provider())->SetSinkParamsForTesting(params_); |
| track_2->Start(); |
| static_cast<webrtc::AudioTrackInterface*>(track_2.get())-> |
| GetRenderer()->AddChannel(1); |
| EXPECT_TRUE(track_2->enabled()); |
| |
| // Verify both |sink_1| and |sink_2| get data. |
| event_1.Reset(); |
| base::WaitableEvent event_2(false, false); |
| |
| scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink()); |
| EXPECT_CALL(*sink_2, OnSetFormat(_)).WillOnce(Return()); |
| EXPECT_CALL(*sink_1, |
| CaptureData(1, |
| params.sample_rate(), |
| params.channels(), |
| params.sample_rate() / 100, |
| 0, |
| 0, |
| false, |
| false)).Times(AtLeast(1)) |
| .WillRepeatedly(SignalEvent(&event_1)); |
| EXPECT_CALL(*sink_2, |
| CaptureData(1, |
| params.sample_rate(), |
| params.channels(), |
| params.sample_rate() / 100, |
| 0, |
| 0, |
| false, |
| false)).Times(AtLeast(1)) |
| .WillRepeatedly(SignalEvent(&event_2)); |
| track_2->AddSink(sink_2.get()); |
| EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout())); |
| EXPECT_TRUE(event_2.TimedWait(TestTimeouts::tiny_timeout())); |
| |
| track_1->RemoveSink(sink_1.get()); |
| track_1->Stop(); |
| track_1 = NULL; |
| |
| EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return()); |
| track_2->RemoveSink(sink_2.get()); |
| track_2->Stop(); |
| track_2 = NULL; |
| |
| capturer_->Stop(); |
| } |
| |
| |
| // Start one track and verify the capturer is correctly starting its source. |
| // And it should be fine to not to call Stop() explicitly. |
| TEST_F(WebRtcLocalAudioTrackTest, StartOneAudioTrack) { |
| EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); |
| EXPECT_CALL(*capturer_source_.get(), OnStart()); |
| RTCMediaConstraints constraints; |
| scoped_refptr<WebRtcLocalAudioTrack> track = |
| WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL, |
| &constraints); |
| static_cast<WebRtcLocalAudioSourceProvider*>( |
| track->audio_source_provider())->SetSinkParamsForTesting(params_); |
| track->Start(); |
| |
| // When the track goes away, it will automatically stop the |
| // |capturer_source_|. |
| EXPECT_CALL(*capturer_source_.get(), OnStop()); |
| capturer_->Stop(); |
| track = NULL; |
| } |
| |
| // Start/Stop tracks and verify the capturer is correctly starting/stopping |
| // its source. |
| TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) { |
| // Starting the first audio track will start the |capturer_source_|. |
| base::WaitableEvent event(false, false); |
| EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); |
| EXPECT_CALL(*capturer_source_.get(), OnStart()).WillOnce(SignalEvent(&event)); |
| RTCMediaConstraints constraints; |
| scoped_refptr<WebRtcLocalAudioTrack> track_1 = |
| WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL, |
| &constraints); |
| static_cast<webrtc::AudioTrackInterface*>(track_1.get())-> |
| GetRenderer()->AddChannel(0); |
| static_cast<WebRtcLocalAudioSourceProvider*>( |
| track_1->audio_source_provider())->SetSinkParamsForTesting(params_); |
| track_1->Start(); |
| EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); |
| |
| // Verify the data flow by connecting the sink to |track_1|. |
| scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink()); |
| event.Reset(); |
| EXPECT_CALL(*sink, OnSetFormat(_)).WillOnce(SignalEvent(&event)); |
| EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false, false)) |
| .Times(AnyNumber()).WillRepeatedly(Return()); |
| track_1->AddSink(sink.get()); |
| EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); |
| |
| // Start the second audio track will not start the |capturer_source_| |
| // since it has been started. |
| EXPECT_CALL(*capturer_source_.get(), OnStart()).Times(0); |
| scoped_refptr<WebRtcLocalAudioTrack> track_2 = |
| WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL, |
| &constraints); |
| static_cast<WebRtcLocalAudioSourceProvider*>( |
| track_2->audio_source_provider())->SetSinkParamsForTesting(params_); |
| track_2->Start(); |
| static_cast<webrtc::AudioTrackInterface*>(track_2.get())-> |
| GetRenderer()->AddChannel(1); |
| |
| // Stop the capturer will clear up the track lists in the capturer. |
| EXPECT_CALL(*capturer_source_.get(), OnStop()); |
| capturer_->Stop(); |
| |
| // Adding a new track to the capturer. |
| track_2->AddSink(sink.get()); |
| EXPECT_CALL(*sink, OnSetFormat(_)).Times(0); |
| |
| // Stop the capturer again will not trigger stopping the source of the |
| // capturer again.. |
| event.Reset(); |
| EXPECT_CALL(*capturer_source_.get(), OnStop()).Times(0); |
| capturer_->Stop(); |
| } |
| |
| // Set new source to the existing capturer. |
| TEST_F(WebRtcLocalAudioTrackTest, SetNewSourceForCapturerAfterStartTrack) { |
| // Setup the audio track and start the track. |
| EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); |
| EXPECT_CALL(*capturer_source_.get(), OnStart()); |
| RTCMediaConstraints constraints; |
| scoped_refptr<WebRtcLocalAudioTrack> track = |
| WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL, |
| &constraints); |
| static_cast<WebRtcLocalAudioSourceProvider*>( |
| track->audio_source_provider())->SetSinkParamsForTesting(params_); |
| track->Start(); |
| |
| // Setting new source to the capturer and the track should still get packets. |
| scoped_refptr<MockCapturerSource> new_source( |
| new MockCapturerSource(capturer_.get())); |
| EXPECT_CALL(*capturer_source_.get(), OnStop()); |
| EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(true)); |
| EXPECT_CALL(*new_source.get(), OnInitialize(_, capturer_.get(), 0)) |
| .WillOnce(Return()); |
| EXPECT_CALL(*new_source.get(), OnStart()); |
| capturer_->SetCapturerSource(new_source, |
| params_.channel_layout(), |
| params_.sample_rate()); |
| |
| // Stop the track. |
| EXPECT_CALL(*new_source.get(), OnStop()); |
| capturer_->Stop(); |
| } |
| |
| // Create a new capturer with new source, connect it to a new audio track. |
| TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) { |
| // Setup the first audio track and start it. |
| EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); |
| EXPECT_CALL(*capturer_source_.get(), OnStart()); |
| RTCMediaConstraints constraints; |
| scoped_refptr<WebRtcLocalAudioTrack> track_1 = |
| WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL, |
| &constraints); |
| static_cast<WebRtcLocalAudioSourceProvider*>( |
| track_1->audio_source_provider())->SetSinkParamsForTesting(params_); |
| track_1->Start(); |
| |
| // Connect a number of network channels to the |track_1|. |
| static const int kNumberOfNetworkChannelsForTrack1 = 2; |
| for (int i = 0; i < kNumberOfNetworkChannelsForTrack1; ++i) { |
| static_cast<webrtc::AudioTrackInterface*>(track_1.get())-> |
| GetRenderer()->AddChannel(i); |
| } |
| // Verify the data flow by connecting the |sink_1| to |track_1|. |
| scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink()); |
| EXPECT_CALL( |
| *sink_1.get(), |
| CaptureData( |
| kNumberOfNetworkChannelsForTrack1, 48000, 2, _, 0, 0, false, false)) |
| .Times(AnyNumber()).WillRepeatedly(Return()); |
| EXPECT_CALL(*sink_1.get(), OnSetFormat(_)).Times(AnyNumber()); |
| track_1->AddSink(sink_1.get()); |
| |
| // Create a new capturer with new source with different audio format. |
| scoped_refptr<WebRtcAudioCapturer> new_capturer( |
| WebRtcAudioCapturer::CreateCapturer()); |
| scoped_refptr<MockCapturerSource> new_source( |
| new MockCapturerSource(new_capturer.get())); |
| EXPECT_CALL(*new_source.get(), OnInitialize(_, new_capturer.get(), 0)); |
| new_capturer->SetCapturerSource(new_source, |
| media::CHANNEL_LAYOUT_MONO, |
| 44100); |
| |
| // Setup the second audio track, connect it to the new capturer and start it. |
| EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(true)); |
| EXPECT_CALL(*new_source.get(), OnStart()); |
| scoped_refptr<WebRtcLocalAudioTrack> track_2 = |
| WebRtcLocalAudioTrack::Create(std::string(), new_capturer, NULL, NULL, |
| &constraints); |
| static_cast<WebRtcLocalAudioSourceProvider*>( |
| track_2->audio_source_provider())->SetSinkParamsForTesting(params_); |
| track_2->Start(); |
| |
| // Connect a number of network channels to the |track_2|. |
| static const int kNumberOfNetworkChannelsForTrack2 = 3; |
| for (int i = 0; i < kNumberOfNetworkChannelsForTrack2; ++i) { |
| static_cast<webrtc::AudioTrackInterface*>(track_2.get())-> |
| GetRenderer()->AddChannel(i); |
| } |
| // Verify the data flow by connecting the |sink_2| to |track_2|. |
| scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink()); |
| base::WaitableEvent event(false, false); |
| EXPECT_CALL( |
| *sink_2, |
| CaptureData( |
| kNumberOfNetworkChannelsForTrack2, 44100, 1, _, 0, 0, false, false)) |
| .Times(AnyNumber()).WillRepeatedly(Return()); |
| EXPECT_CALL(*sink_2, OnSetFormat(_)).WillOnce(SignalEvent(&event)); |
| track_2->AddSink(sink_2.get()); |
| EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); |
| |
| // Stopping the new source will stop the second track. |
| event.Reset(); |
| EXPECT_CALL(*new_source.get(), OnStop()) |
| .Times(1).WillOnce(SignalEvent(&event)); |
| new_capturer->Stop(); |
| EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); |
| |
| // Stop the capturer of the first audio track. |
| EXPECT_CALL(*capturer_source_.get(), OnStop()); |
| capturer_->Stop(); |
| } |
| |
| |
| // Make sure a audio track can deliver packets with a buffer size smaller than |
| // 10ms when it is not connected with a peer connection. |
| TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) { |
| // Setup a capturer which works with a buffer size smaller than 10ms. |
| media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
| media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128); |
| |
| // Create a capturer with new source which works with the format above. |
| scoped_refptr<WebRtcAudioCapturer> capturer( |
| WebRtcAudioCapturer::CreateCapturer()); |
| scoped_refptr<MockCapturerSource> source( |
| new MockCapturerSource(capturer.get())); |
| capturer->Initialize(-1, params.channel_layout(), params.sample_rate(), |
| params.frames_per_buffer(), 0, std::string(), 0, 0); |
| |
| EXPECT_CALL(*source.get(), OnInitialize(_, capturer.get(), 0)); |
| capturer->SetCapturerSource(source, params.channel_layout(), |
| params.sample_rate()); |
| |
| // Setup a audio track, connect it to the capturer and start it. |
| EXPECT_CALL(*source.get(), SetAutomaticGainControl(true)); |
| EXPECT_CALL(*source.get(), OnStart()); |
| RTCMediaConstraints constraints; |
| scoped_refptr<WebRtcLocalAudioTrack> track = |
| WebRtcLocalAudioTrack::Create(std::string(), capturer, NULL, NULL, |
| &constraints); |
| static_cast<WebRtcLocalAudioSourceProvider*>( |
| track->audio_source_provider())->SetSinkParamsForTesting(params); |
| track->Start(); |
| |
| // Verify the data flow by connecting the |sink| to |track|. |
| scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink()); |
| base::WaitableEvent event(false, false); |
| EXPECT_CALL(*sink, OnSetFormat(_)).Times(1); |
| // Verify the sinks are getting the packets with an expecting buffer size. |
| #if defined(OS_ANDROID) |
| const int expected_buffer_size = params.sample_rate() / 100; |
| #else |
| const int expected_buffer_size = params.frames_per_buffer(); |
| #endif |
| EXPECT_CALL(*sink, CaptureData( |
| 0, params.sample_rate(), params.channels(), expected_buffer_size, |
| 0, 0, false, false)) |
| .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event)); |
| track->AddSink(sink.get()); |
| EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); |
| |
| // Stopping the new source will stop the second track. |
| EXPECT_CALL(*source, OnStop()).Times(1); |
| capturer->Stop(); |
| } |
| |
| } // namespace content |
