content/renderer/media/webrtc_local_audio_track_unittest.cc - fp2-dev/platform/external/chromium_org - Gitiles

 // 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/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/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:
   explicit FakeAudioThread(const scoped_refptr<WebRtcAudioCapturer>& capturer)
     : capturer_(capturer),
       thread_(),
       closure_(false, false) {
     DCHECK(capturer.get());
     audio_bus_ = media::AudioBus::Create(capturer_->audio_parameters());
   }

   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);

       // 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:
   MockCapturerSource() {}
   MOCK_METHOD3(Initialize, void(const media::AudioParameters& params,
                                 CaptureCallback* callback,
                                 int session_id));
   MOCK_METHOD0(Start, void());
   MOCK_METHOD0(Stop, void());
   MOCK_METHOD1(SetVolume, void(double volume));
   MOCK_METHOD1(SetAutomaticGainControl, void(bool enable));

  protected:
   virtual ~MockCapturerSource() {}
 };

 class MockWebRtcAudioCapturerSink : public WebRtcAudioCapturerSink {
  public:
   MockWebRtcAudioCapturerSink() {}
   ~MockWebRtcAudioCapturerSink() {}
   int CaptureData(const std::vector<int>& channels,
                   const int16* audio_data,
                   int sample_rate,
                   int number_of_channels,
                   int number_of_frames,
                   int audio_delay_milliseconds,
                   int current_volume,
                   bool need_audio_processing) OVERRIDE {
     CaptureData(channels.size(), sample_rate, number_of_channels,
                 number_of_frames, audio_delay_milliseconds, current_volume,
                 need_audio_processing);
     return 0;
   }
   MOCK_METHOD7(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));

   MOCK_METHOD1(SetCaptureFormat, void(const media::AudioParameters& params));
 };

 }  // namespace

 class WebRtcLocalAudioTrackTest : public ::testing::Test {
  protected:
   virtual void SetUp() OVERRIDE {
     capturer_ = WebRtcAudioCapturer::CreateCapturer();
     capturer_source_ = new MockCapturerSource();
     EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), 0))
         .WillOnce(Return());
     capturer_->SetCapturerSource(capturer_source_,
                                  media::CHANNEL_LAYOUT_STEREO,
                                  48000);

     EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(false))
         .WillOnce(Return());

     // Start the audio thread used by the |capturer_source_|.
     audio_thread_.reset(new FakeAudioThread(capturer_));
     audio_thread_->Start();
   }

   virtual void TearDown() {
     audio_thread_->Stop();
     audio_thread_.reset();
   }

   scoped_refptr<MockCapturerSource> capturer_source_;
   scoped_refptr<WebRtcAudioCapturer> capturer_;
   scoped_ptr<FakeAudioThread> audio_thread_;
 };

 // 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(), Start()).WillOnce(Return());
   scoped_refptr<WebRtcLocalAudioTrack> track =
       WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
   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<MockWebRtcAudioCapturerSink> sink(
       new MockWebRtcAudioCapturerSink());
   const media::AudioParameters params = capturer_->audio_parameters();
   base::WaitableEvent event(false, false);
   EXPECT_CALL(*sink, SetCaptureFormat(_)).WillOnce(Return());
   EXPECT_CALL(*sink, CaptureData(
       kNumberOfNetworkChannels, params.sample_rate(), params.channels(),
       params.frames_per_buffer(), 0, 0, 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(), Stop()).WillOnce(Return());
   track->Stop();
   track = NULL;
 }

 // 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(), Start()).WillOnce(Return());
   scoped_refptr<WebRtcLocalAudioTrack> track =
     WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
   track->Start();
   static_cast<webrtc::AudioTrackInterface*>(track.get())->
       GetRenderer()->AddChannel(0);
   EXPECT_TRUE(track->enabled());
   EXPECT_TRUE(track->set_enabled(false));
   scoped_ptr<MockWebRtcAudioCapturerSink> sink(
       new MockWebRtcAudioCapturerSink());
   const media::AudioParameters params = capturer_->audio_parameters();
   base::WaitableEvent event(false, false);
   EXPECT_CALL(*sink, SetCaptureFormat(_)).WillOnce(Return());
   EXPECT_CALL(*sink, CaptureData(
       1, params.sample_rate(), params.channels(),
       params.frames_per_buffer(), 0, 0, 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.frames_per_buffer(), 0, 0, 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(), Stop()).WillOnce(Return());
   track->Stop();
   track = NULL;
 }

 // Create multiple audio tracks and enable/disable them, verify that the audio
 // callbacks appear/disappear.
 TEST_F(WebRtcLocalAudioTrackTest, MultipleAudioTracks) {
   EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(Return());
   scoped_refptr<WebRtcLocalAudioTrack> track_1 =
     WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
   track_1->Start();
   static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
       GetRenderer()->AddChannel(0);
   EXPECT_TRUE(track_1->enabled());
   scoped_ptr<MockWebRtcAudioCapturerSink> sink_1(
       new MockWebRtcAudioCapturerSink());
   const media::AudioParameters params = capturer_->audio_parameters();
   base::WaitableEvent event_1(false, false);
   EXPECT_CALL(*sink_1, SetCaptureFormat(_)).WillOnce(Return());
   EXPECT_CALL(*sink_1, CaptureData(
       1, params.sample_rate(), params.channels(),
       params.frames_per_buffer(), 0, 0, 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);
   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<MockWebRtcAudioCapturerSink> sink_2(
         new MockWebRtcAudioCapturerSink());
   EXPECT_CALL(*sink_2, SetCaptureFormat(_)).WillOnce(Return());
   EXPECT_CALL(*sink_1, CaptureData(
       1, params.sample_rate(), params.channels(),
       params.frames_per_buffer(), 0, 0, false))
       .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event_1));
   EXPECT_CALL(*sink_2, CaptureData(
       1, params.sample_rate(), params.channels(),
       params.frames_per_buffer(), 0, 0, 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(), Stop()).WillOnce(Return());
   track_2->RemoveSink(sink_2.get());
   track_2->Stop();
   track_2 = NULL;
 }


 // 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(), Start()).Times(1);
   scoped_refptr<WebRtcLocalAudioTrack> track =
       WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
   track->Start();

   // When the track goes away, it will automatically stop the
   // |capturer_source_|.
   EXPECT_CALL(*capturer_source_.get(), Stop());
   track->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(), Start()).WillOnce(SignalEvent(&event));
   scoped_refptr<WebRtcLocalAudioTrack> track_1 =
       WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
   static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
       GetRenderer()->AddChannel(0);
   track_1->Start();
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));

   // Verify the data flow by connecting the sink to |track_1|.
   scoped_ptr<MockWebRtcAudioCapturerSink> sink(
       new MockWebRtcAudioCapturerSink());
   event.Reset();
   EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false))
       .Times(AnyNumber()).WillRepeatedly(Return());
   EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(1);
   track_1->AddSink(sink.get());

   // Start the second audio track will not start the |capturer_source_|
   // since it has been started.
   EXPECT_CALL(*capturer_source_.get(), Start()).Times(0);
   scoped_refptr<WebRtcLocalAudioTrack> track_2 =
       WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
   track_2->Start();
   static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
       GetRenderer()->AddChannel(1);

   // Stop the first audio track will not stop the |capturer_source_|.
   EXPECT_CALL(*capturer_source_.get(), Stop()).Times(0);
   track_1->RemoveSink(sink.get());
   track_1->Stop();
   track_1 = NULL;

   EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false))
       .Times(AnyNumber()).WillRepeatedly(Return());
   EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(1);
   track_2->AddSink(sink.get());

   // Stop the last audio track will stop the |capturer_source_|.
   event.Reset();
   EXPECT_CALL(*capturer_source_.get(), Stop())
       .Times(1).WillOnce(SignalEvent(&event));
   track_2->Stop();
   track_2->RemoveSink(sink.get());
   track_2 = NULL;
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 }

 // Set new source to the existing capturer.
 TEST_F(WebRtcLocalAudioTrackTest, SetNewSourceForCapturerAfterStartTrack) {
   // Setup the audio track and start the track.
   EXPECT_CALL(*capturer_source_.get(), Start()).Times(1);
   scoped_refptr<WebRtcLocalAudioTrack> track =
       WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
   track->Start();

   // Setting new source to the capturer and the track should still get packets.
   scoped_refptr<MockCapturerSource> new_source(new MockCapturerSource());
   EXPECT_CALL(*capturer_source_.get(), Stop());
   EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(false));
   EXPECT_CALL(*new_source.get(), Initialize(_, capturer_.get(), 0))
       .WillOnce(Return());
   EXPECT_CALL(*new_source.get(), Start()).WillOnce(Return());
   capturer_->SetCapturerSource(new_source,
                                media::CHANNEL_LAYOUT_STEREO,
                                48000);

   // Stop the track.
   EXPECT_CALL(*new_source.get(), Stop());
   track->Stop();
   track = NULL;
 }

 // 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(), Start()).Times(1);
   scoped_refptr<WebRtcLocalAudioTrack> track_1 =
       WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
   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<MockWebRtcAudioCapturerSink> sink_1(
       new MockWebRtcAudioCapturerSink());
   EXPECT_CALL(*sink_1.get(), CaptureData(kNumberOfNetworkChannelsForTrack1,
                                          48000, 2, _, 0, 0, false))
       .Times(AnyNumber()).WillRepeatedly(Return());
   EXPECT_CALL(*sink_1.get(), SetCaptureFormat(_)).Times(1);
   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());
   EXPECT_CALL(*new_source.get(), Initialize(_, new_capturer.get(), 0))
       .WillOnce(Return());
   EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(false))
       .WillOnce(Return());
   new_capturer->SetCapturerSource(new_source,
                                   media::CHANNEL_LAYOUT_MONO,
                                   44100);

   // Start the audio thread used by the new source.
   scoped_ptr<FakeAudioThread> audio_thread(new FakeAudioThread(new_capturer));
   audio_thread->Start();

   // Setup the second audio track, connect it to the new capturer and start it.
   EXPECT_CALL(*new_source.get(), Start()).Times(1);
   scoped_refptr<WebRtcLocalAudioTrack> track_2 =
       WebRtcLocalAudioTrack::Create(std::string(), new_capturer, NULL);
   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<MockWebRtcAudioCapturerSink> sink_2(
       new MockWebRtcAudioCapturerSink());
   EXPECT_CALL(*sink_2, CaptureData(kNumberOfNetworkChannelsForTrack2,
                                      44100, 1, _, 0, 0, false))
       .Times(AnyNumber()).WillRepeatedly(Return());
   EXPECT_CALL(*sink_2, SetCaptureFormat(_)).Times(1);
   track_2->AddSink(sink_2.get());

   // Stop the second audio track will stop the new source.
   base::WaitableEvent event(false, false);
   EXPECT_CALL(*new_source.get(), Stop()).Times(1).WillOnce(SignalEvent(&event));
   track_2->Stop();
   track_2->RemoveSink(sink_2.get());
   track_2 = NULL;
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
   audio_thread->Stop();
   audio_thread.reset();

   // Stop the first audio track.
   EXPECT_CALL(*capturer_source_.get(), Stop());
   track_1->Stop();
   track_1 = NULL;
 }

 }  // namespace content
	// 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/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/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:
	explicit FakeAudioThread(const scoped_refptr<WebRtcAudioCapturer>& capturer)
	: capturer_(capturer),
	thread_(),
	closure_(false, false) {
	DCHECK(capturer.get());
	audio_bus_ = media::AudioBus::Create(capturer_->audio_parameters());
	}

	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);

	// 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:
	MockCapturerSource() {}
	MOCK_METHOD3(Initialize, void(const media::AudioParameters& params,
	CaptureCallback* callback,
	int session_id));
	MOCK_METHOD0(Start, void());
	MOCK_METHOD0(Stop, void());
	MOCK_METHOD1(SetVolume, void(double volume));
	MOCK_METHOD1(SetAutomaticGainControl, void(bool enable));

	protected:
	virtual ~MockCapturerSource() {}
	};

	class MockWebRtcAudioCapturerSink : public WebRtcAudioCapturerSink {
	public:
	MockWebRtcAudioCapturerSink() {}
	~MockWebRtcAudioCapturerSink() {}
	int CaptureData(const std::vector<int>& channels,
	const int16* audio_data,
	int sample_rate,
	int number_of_channels,
	int number_of_frames,
	int audio_delay_milliseconds,
	int current_volume,
	bool need_audio_processing) OVERRIDE {
	CaptureData(channels.size(), sample_rate, number_of_channels,
	number_of_frames, audio_delay_milliseconds, current_volume,
	need_audio_processing);
	return 0;
	}
	MOCK_METHOD7(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));

	MOCK_METHOD1(SetCaptureFormat, void(const media::AudioParameters& params));
	};

	} // namespace

	class WebRtcLocalAudioTrackTest : public ::testing::Test {
	protected:
	virtual void SetUp() OVERRIDE {
	capturer_ = WebRtcAudioCapturer::CreateCapturer();
	capturer_source_ = new MockCapturerSource();
	EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), 0))
	.WillOnce(Return());
	capturer_->SetCapturerSource(capturer_source_,
	media::CHANNEL_LAYOUT_STEREO,
	48000);

	EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(false))
	.WillOnce(Return());

	// Start the audio thread used by the \|capturer_source_\|.
	audio_thread_.reset(new FakeAudioThread(capturer_));
	audio_thread_->Start();
	}

	virtual void TearDown() {
	audio_thread_->Stop();
	audio_thread_.reset();
	}

	scoped_refptr<MockCapturerSource> capturer_source_;
	scoped_refptr<WebRtcAudioCapturer> capturer_;
	scoped_ptr<FakeAudioThread> audio_thread_;
	};

	// 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(), Start()).WillOnce(Return());
	scoped_refptr<WebRtcLocalAudioTrack> track =
	WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
	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<MockWebRtcAudioCapturerSink> sink(
	new MockWebRtcAudioCapturerSink());
	const media::AudioParameters params = capturer_->audio_parameters();
	base::WaitableEvent event(false, false);
	EXPECT_CALL(*sink, SetCaptureFormat(_)).WillOnce(Return());
	EXPECT_CALL(*sink, CaptureData(
	kNumberOfNetworkChannels, params.sample_rate(), params.channels(),
	params.frames_per_buffer(), 0, 0, 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(), Stop()).WillOnce(Return());
	track->Stop();
	track = NULL;
	}

	// 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(), Start()).WillOnce(Return());
	scoped_refptr<WebRtcLocalAudioTrack> track =
	WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
	track->Start();
	static_cast<webrtc::AudioTrackInterface*>(track.get())->
	GetRenderer()->AddChannel(0);
	EXPECT_TRUE(track->enabled());
	EXPECT_TRUE(track->set_enabled(false));
	scoped_ptr<MockWebRtcAudioCapturerSink> sink(
	new MockWebRtcAudioCapturerSink());
	const media::AudioParameters params = capturer_->audio_parameters();
	base::WaitableEvent event(false, false);
	EXPECT_CALL(*sink, SetCaptureFormat(_)).WillOnce(Return());
	EXPECT_CALL(*sink, CaptureData(
	1, params.sample_rate(), params.channels(),
	params.frames_per_buffer(), 0, 0, 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.frames_per_buffer(), 0, 0, 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(), Stop()).WillOnce(Return());
	track->Stop();
	track = NULL;
	}

	// Create multiple audio tracks and enable/disable them, verify that the audio
	// callbacks appear/disappear.
	TEST_F(WebRtcLocalAudioTrackTest, MultipleAudioTracks) {
	EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(Return());
	scoped_refptr<WebRtcLocalAudioTrack> track_1 =
	WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
	track_1->Start();
	static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
	GetRenderer()->AddChannel(0);
	EXPECT_TRUE(track_1->enabled());
	scoped_ptr<MockWebRtcAudioCapturerSink> sink_1(
	new MockWebRtcAudioCapturerSink());
	const media::AudioParameters params = capturer_->audio_parameters();
	base::WaitableEvent event_1(false, false);
	EXPECT_CALL(*sink_1, SetCaptureFormat(_)).WillOnce(Return());
	EXPECT_CALL(*sink_1, CaptureData(
	1, params.sample_rate(), params.channels(),
	params.frames_per_buffer(), 0, 0, 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);
	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<MockWebRtcAudioCapturerSink> sink_2(
	new MockWebRtcAudioCapturerSink());
	EXPECT_CALL(*sink_2, SetCaptureFormat(_)).WillOnce(Return());
	EXPECT_CALL(*sink_1, CaptureData(
	1, params.sample_rate(), params.channels(),
	params.frames_per_buffer(), 0, 0, false))
	.Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event_1));
	EXPECT_CALL(*sink_2, CaptureData(
	1, params.sample_rate(), params.channels(),
	params.frames_per_buffer(), 0, 0, 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(), Stop()).WillOnce(Return());
	track_2->RemoveSink(sink_2.get());
	track_2->Stop();
	track_2 = NULL;
	}


	// 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(), Start()).Times(1);
	scoped_refptr<WebRtcLocalAudioTrack> track =
	WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
	track->Start();

	// When the track goes away, it will automatically stop the
	// \|capturer_source_\|.
	EXPECT_CALL(*capturer_source_.get(), Stop());
	track->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(), Start()).WillOnce(SignalEvent(&event));
	scoped_refptr<WebRtcLocalAudioTrack> track_1 =
	WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
	static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
	GetRenderer()->AddChannel(0);
	track_1->Start();
	EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));

	// Verify the data flow by connecting the sink to \|track_1\|.
	scoped_ptr<MockWebRtcAudioCapturerSink> sink(
	new MockWebRtcAudioCapturerSink());
	event.Reset();
	EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false))
	.Times(AnyNumber()).WillRepeatedly(Return());
	EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(1);
	track_1->AddSink(sink.get());

	// Start the second audio track will not start the \|capturer_source_\|
	// since it has been started.
	EXPECT_CALL(*capturer_source_.get(), Start()).Times(0);
	scoped_refptr<WebRtcLocalAudioTrack> track_2 =
	WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
	track_2->Start();
	static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
	GetRenderer()->AddChannel(1);

	// Stop the first audio track will not stop the \|capturer_source_\|.
	EXPECT_CALL(*capturer_source_.get(), Stop()).Times(0);
	track_1->RemoveSink(sink.get());
	track_1->Stop();
	track_1 = NULL;

	EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false))
	.Times(AnyNumber()).WillRepeatedly(Return());
	EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(1);
	track_2->AddSink(sink.get());

	// Stop the last audio track will stop the \|capturer_source_\|.
	event.Reset();
	EXPECT_CALL(*capturer_source_.get(), Stop())
	.Times(1).WillOnce(SignalEvent(&event));
	track_2->Stop();
	track_2->RemoveSink(sink.get());
	track_2 = NULL;
	EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
	}

	// Set new source to the existing capturer.
	TEST_F(WebRtcLocalAudioTrackTest, SetNewSourceForCapturerAfterStartTrack) {
	// Setup the audio track and start the track.
	EXPECT_CALL(*capturer_source_.get(), Start()).Times(1);
	scoped_refptr<WebRtcLocalAudioTrack> track =
	WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
	track->Start();

	// Setting new source to the capturer and the track should still get packets.
	scoped_refptr<MockCapturerSource> new_source(new MockCapturerSource());
	EXPECT_CALL(*capturer_source_.get(), Stop());
	EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(false));
	EXPECT_CALL(*new_source.get(), Initialize(_, capturer_.get(), 0))
	.WillOnce(Return());
	EXPECT_CALL(*new_source.get(), Start()).WillOnce(Return());
	capturer_->SetCapturerSource(new_source,
	media::CHANNEL_LAYOUT_STEREO,
	48000);

	// Stop the track.
	EXPECT_CALL(*new_source.get(), Stop());
	track->Stop();
	track = NULL;
	}

	// 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(), Start()).Times(1);
	scoped_refptr<WebRtcLocalAudioTrack> track_1 =
	WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
	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<MockWebRtcAudioCapturerSink> sink_1(
	new MockWebRtcAudioCapturerSink());
	EXPECT_CALL(*sink_1.get(), CaptureData(kNumberOfNetworkChannelsForTrack1,
	48000, 2, _, 0, 0, false))
	.Times(AnyNumber()).WillRepeatedly(Return());
	EXPECT_CALL(*sink_1.get(), SetCaptureFormat(_)).Times(1);
	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());
	EXPECT_CALL(*new_source.get(), Initialize(_, new_capturer.get(), 0))
	.WillOnce(Return());
	EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(false))
	.WillOnce(Return());
	new_capturer->SetCapturerSource(new_source,
	media::CHANNEL_LAYOUT_MONO,
	44100);

	// Start the audio thread used by the new source.
	scoped_ptr<FakeAudioThread> audio_thread(new FakeAudioThread(new_capturer));
	audio_thread->Start();

	// Setup the second audio track, connect it to the new capturer and start it.
	EXPECT_CALL(*new_source.get(), Start()).Times(1);
	scoped_refptr<WebRtcLocalAudioTrack> track_2 =
	WebRtcLocalAudioTrack::Create(std::string(), new_capturer, NULL);
	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<MockWebRtcAudioCapturerSink> sink_2(
	new MockWebRtcAudioCapturerSink());
	EXPECT_CALL(*sink_2, CaptureData(kNumberOfNetworkChannelsForTrack2,
	44100, 1, _, 0, 0, false))
	.Times(AnyNumber()).WillRepeatedly(Return());
	EXPECT_CALL(*sink_2, SetCaptureFormat(_)).Times(1);
	track_2->AddSink(sink_2.get());

	// Stop the second audio track will stop the new source.
	base::WaitableEvent event(false, false);
	EXPECT_CALL(*new_source.get(), Stop()).Times(1).WillOnce(SignalEvent(&event));
	track_2->Stop();
	track_2->RemoveSink(sink_2.get());
	track_2 = NULL;
	EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
	audio_thread->Stop();
	audio_thread.reset();

	// Stop the first audio track.
	EXPECT_CALL(*capturer_source_.get(), Stop());
	track_1->Stop();
	track_1 = NULL;
	}

	} // namespace content