| // 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/public/renderer/media_stream_audio_sink.h" |
| #include "content/renderer/media/media_stream_audio_source.h" |
| #include "content/renderer/media/mock_media_constraint_factory.h" |
| #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" |
| #include "content/renderer/media/webrtc_audio_capturer.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/WebKit/public/platform/WebMediaConstraints.h" |
| #include "third_party/WebKit/public/web/WebHeap.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(WebRtcAudioCapturer* capturer, |
| const media::AudioParameters& params) |
| : capturer_(capturer), |
| thread_(), |
| closure_(false, false) { |
| DCHECK(capturer); |
| audio_bus_ = media::AudioBus::Create(params); |
| } |
| |
| ~FakeAudioThread() override { DCHECK(thread_.is_null()); } |
| |
| // base::PlatformThread::Delegate: |
| void ThreadMain() override { |
| while (true) { |
| if (closure_.IsSignaled()) |
| return; |
| |
| media::AudioCapturerSource::CaptureCallback* callback = |
| static_cast<media::AudioCapturerSource::CaptureCallback*>( |
| capturer_); |
| 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::ThreadPriority::REALTIME_AUDIO); |
| CHECK(!thread_.is_null()); |
| } |
| |
| void Stop() { |
| closure_.Signal(); |
| base::PlatformThread::Join(thread_); |
| thread_ = base::PlatformThreadHandle(); |
| } |
| |
| private: |
| scoped_ptr<media::AudioBus> audio_bus_; |
| 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)); |
| |
| void Initialize(const media::AudioParameters& params, |
| CaptureCallback* callback, |
| int session_id) override { |
| DCHECK(params.IsValid()); |
| params_ = params; |
| OnInitialize(params, callback, session_id); |
| } |
| void Start() override { |
| audio_thread_.reset(new FakeAudioThread(capturer_, params_)); |
| audio_thread_->Start(); |
| OnStart(); |
| } |
| void Stop() override { |
| audio_thread_->Stop(); |
| audio_thread_.reset(); |
| OnStop(); |
| } |
| protected: |
| ~MockCapturerSource() override {} |
| |
| private: |
| scoped_ptr<FakeAudioThread> audio_thread_; |
| WebRtcAudioCapturer* capturer_; |
| media::AudioParameters params_; |
| }; |
| |
| class MockMediaStreamAudioSink : public MediaStreamAudioSink { |
| public: |
| MockMediaStreamAudioSink() {} |
| ~MockMediaStreamAudioSink() {} |
| void OnData(const media::AudioBus& audio_bus, |
| base::TimeTicks estimated_capture_time) override { |
| EXPECT_EQ(params_.channels(), audio_bus.channels()); |
| EXPECT_EQ(params_.frames_per_buffer(), audio_bus.frames()); |
| EXPECT_FALSE(estimated_capture_time.is_null()); |
| CaptureData(); |
| } |
| MOCK_METHOD0(CaptureData, void()); |
| void OnSetFormat(const media::AudioParameters& params) { |
| params_ = params; |
| FormatIsSet(); |
| } |
| MOCK_METHOD0(FormatIsSet, void()); |
| |
| const media::AudioParameters& audio_params() const { return params_; } |
| |
| private: |
| media::AudioParameters params_; |
| }; |
| |
| } // namespace |
| |
| class WebRtcLocalAudioTrackTest : public ::testing::Test { |
| protected: |
| void SetUp() override { |
| params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
| media::CHANNEL_LAYOUT_STEREO, 2, 48000, 16, 480); |
| MockMediaConstraintFactory constraint_factory; |
| blink_source_.initialize("dummy", blink::WebMediaStreamSource::TypeAudio, |
| "dummy", |
| false /* remote */, true /* readonly */); |
| MediaStreamAudioSource* audio_source = new MediaStreamAudioSource(); |
| blink_source_.setExtraData(audio_source); |
| |
| StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE, |
| std::string(), std::string()); |
| capturer_ = WebRtcAudioCapturer::CreateCapturer( |
| -1, device, constraint_factory.CreateWebMediaConstraints(), NULL, |
| audio_source); |
| audio_source->SetAudioCapturer(capturer_.get()); |
| capturer_source_ = new MockCapturerSource(capturer_.get()); |
| EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_.get(), -1)) |
| .WillOnce(Return()); |
| EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); |
| EXPECT_CALL(*capturer_source_.get(), OnStart()); |
| capturer_->SetCapturerSource(capturer_source_, params_); |
| } |
| |
| void TearDown() override { |
| blink_source_.reset(); |
| blink::WebHeap::collectAllGarbageForTesting(); |
| } |
| |
| media::AudioParameters params_; |
| blink::WebMediaStreamSource blink_source_; |
| 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) { |
| scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
| WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
| scoped_ptr<WebRtcLocalAudioTrack> track( |
| new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL)); |
| track->Start(); |
| EXPECT_TRUE(track->GetAudioAdapter()->enabled()); |
| |
| scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink()); |
| base::WaitableEvent event(false, false); |
| EXPECT_CALL(*sink, FormatIsSet()); |
| EXPECT_CALL(*sink, |
| CaptureData()).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()); |
| scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
| WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
| scoped_ptr<WebRtcLocalAudioTrack> track( |
| new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL)); |
| track->Start(); |
| EXPECT_TRUE(track->GetAudioAdapter()->enabled()); |
| EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(false)); |
| scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink()); |
| const media::AudioParameters params = capturer_->source_audio_parameters(); |
| base::WaitableEvent event(false, false); |
| EXPECT_CALL(*sink, FormatIsSet()).Times(1); |
| EXPECT_CALL(*sink, CaptureData()).Times(0); |
| EXPECT_EQ(sink->audio_params().frames_per_buffer(), |
| params.sample_rate() / 100); |
| track->AddSink(sink.get()); |
| EXPECT_FALSE(event.TimedWait(TestTimeouts::tiny_timeout())); |
| |
| event.Reset(); |
| EXPECT_CALL(*sink, CaptureData()).Times(AtLeast(1)) |
| .WillRepeatedly(SignalEvent(&event)); |
| EXPECT_TRUE(track->GetAudioAdapter()->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.reset(); |
| } |
| |
| // 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) { |
| scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1( |
| WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
| scoped_ptr<WebRtcLocalAudioTrack> track_1( |
| new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL)); |
| track_1->Start(); |
| EXPECT_TRUE(track_1->GetAudioAdapter()->enabled()); |
| scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink()); |
| const media::AudioParameters params = capturer_->source_audio_parameters(); |
| base::WaitableEvent event_1(false, false); |
| EXPECT_CALL(*sink_1, FormatIsSet()).WillOnce(Return()); |
| EXPECT_CALL(*sink_1, CaptureData()).Times(AtLeast(1)) |
| .WillRepeatedly(SignalEvent(&event_1)); |
| EXPECT_EQ(sink_1->audio_params().frames_per_buffer(), |
| params.sample_rate() / 100); |
| track_1->AddSink(sink_1.get()); |
| EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout())); |
| |
| scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2( |
| WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
| scoped_ptr<WebRtcLocalAudioTrack> track_2( |
| new WebRtcLocalAudioTrack(adapter_2.get(), capturer_, NULL)); |
| track_2->Start(); |
| EXPECT_TRUE(track_2->GetAudioAdapter()->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, FormatIsSet()).WillOnce(Return()); |
| EXPECT_CALL(*sink_1, CaptureData()).Times(AtLeast(1)) |
| .WillRepeatedly(SignalEvent(&event_1)); |
| EXPECT_EQ(sink_1->audio_params().frames_per_buffer(), |
| params.sample_rate() / 100); |
| EXPECT_CALL(*sink_2, CaptureData()).Times(AtLeast(1)) |
| .WillRepeatedly(SignalEvent(&event_2)); |
| EXPECT_EQ(sink_2->audio_params().frames_per_buffer(), |
| params.sample_rate() / 100); |
| 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.reset(); |
| |
| EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return()); |
| track_2->RemoveSink(sink_2.get()); |
| track_2->Stop(); |
| track_2.reset(); |
| } |
| |
| |
| // 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) { |
| scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
| WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
| scoped_ptr<WebRtcLocalAudioTrack> track( |
| new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL)); |
| track->Start(); |
| |
| // When the track goes away, it will automatically stop the |
| // |capturer_source_|. |
| EXPECT_CALL(*capturer_source_.get(), OnStop()); |
| track.reset(); |
| } |
| |
| // Start two tracks and verify the capturer is correctly starting its source. |
| // When the last track connected to the capturer is stopped, the source is |
| // stopped. |
| TEST_F(WebRtcLocalAudioTrackTest, StartTwoAudioTracks) { |
| scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter1( |
| WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
| scoped_ptr<WebRtcLocalAudioTrack> track1( |
| new WebRtcLocalAudioTrack(adapter1.get(), capturer_, NULL)); |
| track1->Start(); |
| |
| scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter2( |
| WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
| scoped_ptr<WebRtcLocalAudioTrack> track2( |
| new WebRtcLocalAudioTrack(adapter2.get(), capturer_, NULL)); |
| track2->Start(); |
| |
| track1->Stop(); |
| // When the last track is stopped, it will automatically stop the |
| // |capturer_source_|. |
| EXPECT_CALL(*capturer_source_.get(), OnStop()); |
| track2->Stop(); |
| } |
| |
| // Start/Stop tracks and verify the capturer is correctly starting/stopping |
| // its source. |
| TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) { |
| base::WaitableEvent event(false, false); |
| scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1( |
| WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
| scoped_ptr<WebRtcLocalAudioTrack> track_1( |
| new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL)); |
| track_1->Start(); |
| |
| // Verify the data flow by connecting the sink to |track_1|. |
| scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink()); |
| event.Reset(); |
| EXPECT_CALL(*sink, FormatIsSet()).WillOnce(SignalEvent(&event)); |
| EXPECT_CALL(*sink, CaptureData()) |
| .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<WebRtcLocalAudioTrackAdapter> adapter_2( |
| WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
| scoped_ptr<WebRtcLocalAudioTrack> track_2( |
| new WebRtcLocalAudioTrack(adapter_2.get(), capturer_, NULL)); |
| track_2->Start(); |
| |
| // 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, FormatIsSet()).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(); |
| } |
| |
| // Contains data races reported by tsan: crbug.com/404133 |
| #if defined(THREAD_SANITIZER) |
| #define DISABLE_ON_TSAN(function) DISABLED_##function |
| #else |
| #define DISABLE_ON_TSAN(function) function |
| #endif |
| |
| // Create a new capturer with new source, connect it to a new audio track. |
| TEST_F(WebRtcLocalAudioTrackTest, |
| DISABLE_ON_TSAN(ConnectTracksToDifferentCapturers)) { |
| // Setup the first audio track and start it. |
| scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1( |
| WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
| scoped_ptr<WebRtcLocalAudioTrack> track_1( |
| new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL)); |
| track_1->Start(); |
| |
| // 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()) |
| .Times(AnyNumber()).WillRepeatedly(Return()); |
| EXPECT_CALL(*sink_1.get(), FormatIsSet()).Times(AnyNumber()); |
| track_1->AddSink(sink_1.get()); |
| |
| // Create a new capturer with new source with different audio format. |
| MockMediaConstraintFactory constraint_factory; |
| StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE, |
| std::string(), std::string()); |
| scoped_refptr<WebRtcAudioCapturer> new_capturer( |
| WebRtcAudioCapturer::CreateCapturer( |
| -1, device, constraint_factory.CreateWebMediaConstraints(), NULL, |
| NULL)); |
| scoped_refptr<MockCapturerSource> new_source( |
| new MockCapturerSource(new_capturer.get())); |
| EXPECT_CALL(*new_source.get(), OnInitialize(_, new_capturer.get(), -1)); |
| EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(true)); |
| EXPECT_CALL(*new_source.get(), OnStart()); |
| |
| media::AudioParameters new_param( |
| media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
| media::CHANNEL_LAYOUT_MONO, 44100, 16, 441); |
| new_capturer->SetCapturerSource(new_source, new_param); |
| |
| // Setup the second audio track, connect it to the new capturer and start it. |
| scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2( |
| WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
| scoped_ptr<WebRtcLocalAudioTrack> track_2( |
| new WebRtcLocalAudioTrack(adapter_2.get(), new_capturer, NULL)); |
| track_2->Start(); |
| |
| // 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()) |
| .Times(AnyNumber()).WillRepeatedly(Return()); |
| EXPECT_CALL(*sink_2, FormatIsSet()).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. |
| MockMediaConstraintFactory factory; |
| factory.DisableDefaultAudioConstraints(); |
| scoped_refptr<WebRtcAudioCapturer> capturer( |
| WebRtcAudioCapturer::CreateCapturer( |
| -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", "", |
| params.sample_rate(), params.channel_layout(), |
| params.frames_per_buffer()), |
| factory.CreateWebMediaConstraints(), NULL, NULL)); |
| scoped_refptr<MockCapturerSource> source( |
| new MockCapturerSource(capturer.get())); |
| EXPECT_CALL(*source.get(), OnInitialize(_, capturer.get(), -1)); |
| EXPECT_CALL(*source.get(), SetAutomaticGainControl(true)); |
| EXPECT_CALL(*source.get(), OnStart()); |
| capturer->SetCapturerSource(source, params); |
| |
| // Setup a audio track, connect it to the capturer and start it. |
| scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
| WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
| scoped_ptr<WebRtcLocalAudioTrack> track( |
| new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL)); |
| 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, FormatIsSet()).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()) |
| .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event)); |
| track->AddSink(sink.get()); |
| EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); |
| EXPECT_EQ(expected_buffer_size, sink->audio_params().frames_per_buffer()); |
| |
| // Stopping the new source will stop the second track. |
| EXPECT_CALL(*source.get(), OnStop()).Times(1); |
| capturer->Stop(); |
| |
| // Even though this test don't use |capturer_source_| it will be stopped |
| // during teardown of the test harness. |
| EXPECT_CALL(*capturer_source_.get(), OnStop()); |
| } |
| |
| } // namespace content |