content/renderer/media/webrtc_local_audio_track_unittest.cc

   1 // Copyright 2013 The Chromium Authors. All rights reserved.
   2 // Use of this source code is governed by a BSD-style license that can be
   3 // found in the LICENSE file.
   4
   5 #include "base/synchronization/waitable_event.h"
   6 #include "base/test/test_timeouts.h"
   7 #include "content/renderer/media/media_stream_audio_source.h"
   8 #include "content/renderer/media/mock_media_constraint_factory.h"
   9 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
  10 #include "content/renderer/media/webrtc_audio_capturer.h"
  11 #include "content/renderer/media/webrtc_audio_device_impl.h"
  12 #include "content/renderer/media/webrtc_local_audio_track.h"
  13 #include "media/audio/audio_parameters.h"
  14 #include "media/base/audio_bus.h"
  15 #include "media/base/audio_capturer_source.h"
  16 #include "testing/gmock/include/gmock/gmock.h"
  17 #include "testing/gtest/include/gtest/gtest.h"
  18 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
  19 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
  20
  21 using ::testing::_;
  22 using ::testing::AnyNumber;
  23 using ::testing::AtLeast;
  24 using ::testing::Return;
  25
  26 namespace content {
  27
  28 namespace {
  29
  30 ACTION_P(SignalEvent, event) {
  31   event->Signal();
  32 }
  33
  34 // A simple thread that we use to fake the audio thread which provides data to
  35 // the |WebRtcAudioCapturer|.
  36 class FakeAudioThread : public base::PlatformThread::Delegate {
  37  public:
  38   FakeAudioThread(WebRtcAudioCapturer* capturer,
  39                   const media::AudioParameters& params)
  40     : capturer_(capturer),
  41       thread_(),
  42       closure_(false, false) {
  43     DCHECK(capturer);
  44     audio_bus_ = media::AudioBus::Create(params);
  45   }
  46
  47   virtual ~FakeAudioThread() { DCHECK(thread_.is_null()); }
  48
  49   // base::PlatformThread::Delegate:
  50   virtual void ThreadMain() OVERRIDE {
  51     while (true) {
  52       if (closure_.IsSignaled())
  53         return;
  54
  55       media::AudioCapturerSource::CaptureCallback* callback =
  56           static_cast<media::AudioCapturerSource::CaptureCallback*>(
  57               capturer_);
  58       audio_bus_->Zero();
  59       callback->Capture(audio_bus_.get(), 0, 0, false);
  60
  61       // Sleep 1ms to yield the resource for the main thread.
  62       base::PlatformThread::Sleep(base::TimeDelta::FromMilliseconds(1));
  63     }
  64   }
  65
  66   void Start() {
  67     base::PlatformThread::CreateWithPriority(
  68         0, this, &thread_, base::kThreadPriority_RealtimeAudio);
  69     CHECK(!thread_.is_null());
  70   }
  71
  72   void Stop() {
  73     closure_.Signal();
  74     base::PlatformThread::Join(thread_);
  75     thread_ = base::PlatformThreadHandle();
  76   }
  77
  78  private:
  79   scoped_ptr<media::AudioBus> audio_bus_;
  80   WebRtcAudioCapturer* capturer_;
  81   base::PlatformThreadHandle thread_;
  82   base::WaitableEvent closure_;
  83   DISALLOW_COPY_AND_ASSIGN(FakeAudioThread);
  84 };
  85
  86 class MockCapturerSource : public media::AudioCapturerSource {
  87  public:
  88   explicit MockCapturerSource(WebRtcAudioCapturer* capturer)
  89       : capturer_(capturer) {}
  90   MOCK_METHOD3(OnInitialize, void(const media::AudioParameters& params,
  91                                   CaptureCallback* callback,
  92                                   int session_id));
  93   MOCK_METHOD0(OnStart, void());
  94   MOCK_METHOD0(OnStop, void());
  95   MOCK_METHOD1(SetVolume, void(double volume));
  96   MOCK_METHOD1(SetAutomaticGainControl, void(bool enable));
  97
  98   virtual void Initialize(const media::AudioParameters& params,
  99                           CaptureCallback* callback,
 100                           int session_id) OVERRIDE {
 101     DCHECK(params.IsValid());
 102     params_ = params;
 103     OnInitialize(params, callback, session_id);
 104   }
 105   virtual void Start() OVERRIDE {
 106     audio_thread_.reset(new FakeAudioThread(capturer_, params_));
 107     audio_thread_->Start();
 108     OnStart();
 109   }
 110   virtual void Stop() OVERRIDE {
 111     audio_thread_->Stop();
 112     audio_thread_.reset();
 113     OnStop();
 114   }
 115  protected:
 116   virtual ~MockCapturerSource() {}
 117
 118  private:
 119   scoped_ptr<FakeAudioThread> audio_thread_;
 120   WebRtcAudioCapturer* capturer_;
 121   media::AudioParameters params_;
 122 };
 123
 124 // TODO(xians): Use MediaStreamAudioSink.
 125 class MockMediaStreamAudioSink : public PeerConnectionAudioSink {
 126  public:
 127   MockMediaStreamAudioSink() {}
 128   ~MockMediaStreamAudioSink() {}
 129   int OnData(const int16* audio_data,
 130              int sample_rate,
 131              int number_of_channels,
 132              int number_of_frames,
 133              const std::vector<int>& channels,
 134              int audio_delay_milliseconds,
 135              int current_volume,
 136              bool need_audio_processing,
 137              bool key_pressed) OVERRIDE {
 138     EXPECT_EQ(params_.sample_rate(), sample_rate);
 139     EXPECT_EQ(params_.channels(), number_of_channels);
 140     EXPECT_EQ(params_.frames_per_buffer(), number_of_frames);
 141     CaptureData(channels.size(),
 142                 audio_delay_milliseconds,
 143                 current_volume,
 144                 need_audio_processing,
 145                 key_pressed);
 146     return 0;
 147   }
 148   MOCK_METHOD5(CaptureData,
 149                void(int number_of_network_channels,
 150                     int audio_delay_milliseconds,
 151                     int current_volume,
 152                     bool need_audio_processing,
 153                     bool key_pressed));
 154   void OnSetFormat(const media::AudioParameters& params) {
 155     params_ = params;
 156     FormatIsSet();
 157   }
 158   MOCK_METHOD0(FormatIsSet, void());
 159
 160   const media::AudioParameters& audio_params() const { return params_; }
 161
 162  private:
 163   media::AudioParameters params_;
 164 };
 165
 166 }  // namespace
 167
 168 class WebRtcLocalAudioTrackTest : public ::testing::Test {
 169  protected:
 170   virtual void SetUp() OVERRIDE {
 171     params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
 172                   media::CHANNEL_LAYOUT_STEREO, 2, 0, 48000, 16, 480);
 173     MockMediaConstraintFactory constraint_factory;
 174     blink_source_.initialize("dummy", blink::WebMediaStreamSource::TypeAudio,
 175                              "dummy");
 176     MediaStreamAudioSource* audio_source = new MediaStreamAudioSource();
 177     blink_source_.setExtraData(audio_source);
 178
 179     StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
 180                             std::string(), std::string());
 181     capturer_ = WebRtcAudioCapturer::CreateCapturer(
 182         -1, device, constraint_factory.CreateWebMediaConstraints(), NULL,
 183         audio_source);
 184     audio_source->SetAudioCapturer(capturer_);
 185     capturer_source_ = new MockCapturerSource(capturer_.get());
 186     EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_.get(), -1))
 187         .WillOnce(Return());
 188     EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
 189     EXPECT_CALL(*capturer_source_.get(), OnStart());
 190     capturer_->SetCapturerSourceForTesting(capturer_source_, params_);
 191   }
 192
 193   media::AudioParameters params_;
 194   blink::WebMediaStreamSource blink_source_;
 195   scoped_refptr<MockCapturerSource> capturer_source_;
 196   scoped_refptr<WebRtcAudioCapturer> capturer_;
 197 };
 198
 199 // Creates a capturer and audio track, fakes its audio thread, and
 200 // connect/disconnect the sink to the audio track on the fly, the sink should
 201 // get data callback when the track is connected to the capturer but not when
 202 // the track is disconnected from the capturer.
 203 TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) {
 204   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
 205       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 206   scoped_ptr<WebRtcLocalAudioTrack> track(
 207       new WebRtcLocalAudioTrack(adapter, capturer_, NULL));
 208   track->Start();
 209   EXPECT_TRUE(track->GetAudioAdapter()->enabled());
 210
 211   // Connect a number of network channels to the audio track.
 212   static const int kNumberOfNetworkChannels = 4;
 213   for (int i = 0; i < kNumberOfNetworkChannels; ++i) {
 214     static_cast<webrtc::AudioTrackInterface*>(
 215         adapter.get())->GetRenderer()->AddChannel(i);
 216   }
 217   scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
 218   base::WaitableEvent event(false, false);
 219   EXPECT_CALL(*sink, FormatIsSet());
 220   EXPECT_CALL(*sink,
 221       CaptureData(kNumberOfNetworkChannels,
 222                   0,
 223                   0,
 224                   _,
 225                   false)).Times(AtLeast(1))
 226       .WillRepeatedly(SignalEvent(&event));
 227   track->AddSink(sink.get());
 228   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 229   track->RemoveSink(sink.get());
 230
 231   EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return());
 232   capturer_->Stop();
 233 }
 234
 235 // The same setup as ConnectAndDisconnectOneSink, but enable and disable the
 236 // audio track on the fly. When the audio track is disabled, there is no data
 237 // callback to the sink; when the audio track is enabled, there comes data
 238 // callback.
 239 // TODO(xians): Enable this test after resolving the racing issue that TSAN
 240 // reports on MediaStreamTrack::enabled();
 241 TEST_F(WebRtcLocalAudioTrackTest,  DISABLED_DisableEnableAudioTrack) {
 242   EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
 243   EXPECT_CALL(*capturer_source_.get(), OnStart());
 244   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
 245       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 246   scoped_ptr<WebRtcLocalAudioTrack> track(
 247       new WebRtcLocalAudioTrack(adapter, capturer_, NULL));
 248   track->Start();
 249   static_cast<webrtc::AudioTrackInterface*>(
 250       adapter.get())->GetRenderer()->AddChannel(0);
 251   EXPECT_TRUE(track->GetAudioAdapter()->enabled());
 252   EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(false));
 253   scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
 254   const media::AudioParameters params = capturer_->source_audio_parameters();
 255   base::WaitableEvent event(false, false);
 256   EXPECT_CALL(*sink, FormatIsSet()).Times(1);
 257   EXPECT_CALL(*sink,
 258               CaptureData(1, 0, 0, _, false)).Times(0);
 259   EXPECT_EQ(sink->audio_params().frames_per_buffer(),
 260             params.sample_rate() / 100);
 261   track->AddSink(sink.get());
 262   EXPECT_FALSE(event.TimedWait(TestTimeouts::tiny_timeout()));
 263
 264   event.Reset();
 265   EXPECT_CALL(*sink,
 266               CaptureData(1, 0, 0, _, false)).Times(AtLeast(1))
 267       .WillRepeatedly(SignalEvent(&event));
 268   EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(true));
 269   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 270   track->RemoveSink(sink.get());
 271
 272   EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return());
 273   capturer_->Stop();
 274   track.reset();
 275 }
 276
 277 // Create multiple audio tracks and enable/disable them, verify that the audio
 278 // callbacks appear/disappear.
 279 // Flaky due to a data race, see http://crbug.com/295418
 280 TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) {
 281   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
 282       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 283   scoped_ptr<WebRtcLocalAudioTrack> track_1(
 284     new WebRtcLocalAudioTrack(adapter_1, capturer_, NULL));
 285   track_1->Start();
 286   static_cast<webrtc::AudioTrackInterface*>(
 287       adapter_1.get())->GetRenderer()->AddChannel(0);
 288   EXPECT_TRUE(track_1->GetAudioAdapter()->enabled());
 289   scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink());
 290   const media::AudioParameters params = capturer_->source_audio_parameters();
 291   base::WaitableEvent event_1(false, false);
 292   EXPECT_CALL(*sink_1, FormatIsSet()).WillOnce(Return());
 293   EXPECT_CALL(*sink_1,
 294       CaptureData(1, 0, 0, _, false)).Times(AtLeast(1))
 295       .WillRepeatedly(SignalEvent(&event_1));
 296   EXPECT_EQ(sink_1->audio_params().frames_per_buffer(),
 297             params.sample_rate() / 100);
 298   track_1->AddSink(sink_1.get());
 299   EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
 300
 301   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
 302       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 303   scoped_ptr<WebRtcLocalAudioTrack> track_2(
 304     new WebRtcLocalAudioTrack(adapter_2, capturer_, NULL));
 305   track_2->Start();
 306   static_cast<webrtc::AudioTrackInterface*>(
 307       adapter_2.get())->GetRenderer()->AddChannel(1);
 308   EXPECT_TRUE(track_2->GetAudioAdapter()->enabled());
 309
 310   // Verify both |sink_1| and |sink_2| get data.
 311   event_1.Reset();
 312   base::WaitableEvent event_2(false, false);
 313
 314   scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink());
 315   EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(Return());
 316   EXPECT_CALL(*sink_1, CaptureData(1, 0, 0, _, false)).Times(AtLeast(1))
 317       .WillRepeatedly(SignalEvent(&event_1));
 318   EXPECT_EQ(sink_1->audio_params().frames_per_buffer(),
 319             params.sample_rate() / 100);
 320   EXPECT_CALL(*sink_2, CaptureData(1, 0, 0, _, false)).Times(AtLeast(1))
 321       .WillRepeatedly(SignalEvent(&event_2));
 322   EXPECT_EQ(sink_2->audio_params().frames_per_buffer(),
 323             params.sample_rate() / 100);
 324   track_2->AddSink(sink_2.get());
 325   EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
 326   EXPECT_TRUE(event_2.TimedWait(TestTimeouts::tiny_timeout()));
 327
 328   track_1->RemoveSink(sink_1.get());
 329   track_1->Stop();
 330   track_1.reset();
 331
 332   EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return());
 333   track_2->RemoveSink(sink_2.get());
 334   track_2->Stop();
 335   track_2.reset();
 336 }
 337
 338
 339 // Start one track and verify the capturer is correctly starting its source.
 340 // And it should be fine to not to call Stop() explicitly.
 341 TEST_F(WebRtcLocalAudioTrackTest, StartOneAudioTrack) {
 342   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
 343       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 344   scoped_ptr<WebRtcLocalAudioTrack> track(
 345       new WebRtcLocalAudioTrack(adapter, capturer_, NULL));
 346   track->Start();
 347
 348   // When the track goes away, it will automatically stop the
 349   // |capturer_source_|.
 350   EXPECT_CALL(*capturer_source_.get(), OnStop());
 351   track.reset();
 352 }
 353
 354 // Start two tracks and verify the capturer is correctly starting its source.
 355 // When the last track connected to the capturer is stopped, the source is
 356 // stopped.
 357 TEST_F(WebRtcLocalAudioTrackTest, StartTwoAudioTracks) {
 358   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter1(
 359       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 360   scoped_ptr<WebRtcLocalAudioTrack> track1(
 361       new WebRtcLocalAudioTrack(adapter1, capturer_, NULL));
 362   track1->Start();
 363
 364   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter2(
 365         WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 366   scoped_ptr<WebRtcLocalAudioTrack> track2(
 367       new WebRtcLocalAudioTrack(adapter2, capturer_, NULL));
 368   track2->Start();
 369
 370   track1->Stop();
 371   // When the last track is stopped, it will automatically stop the
 372   // |capturer_source_|.
 373   EXPECT_CALL(*capturer_source_.get(), OnStop());
 374   track2->Stop();
 375 }
 376
 377 // Start/Stop tracks and verify the capturer is correctly starting/stopping
 378 // its source.
 379 TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
 380   base::WaitableEvent event(false, false);
 381   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
 382       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 383   scoped_ptr<WebRtcLocalAudioTrack> track_1(
 384       new WebRtcLocalAudioTrack(adapter_1, capturer_, NULL));
 385   static_cast<webrtc::AudioTrackInterface*>(
 386       adapter_1.get())->GetRenderer()->AddChannel(0);
 387   track_1->Start();
 388
 389   // Verify the data flow by connecting the sink to |track_1|.
 390   scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
 391   event.Reset();
 392   EXPECT_CALL(*sink, FormatIsSet()).WillOnce(SignalEvent(&event));
 393   EXPECT_CALL(*sink, CaptureData(_, 0, 0, _, false))
 394       .Times(AnyNumber()).WillRepeatedly(Return());
 395   track_1->AddSink(sink.get());
 396   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 397
 398   // Start the second audio track will not start the |capturer_source_|
 399   // since it has been started.
 400   EXPECT_CALL(*capturer_source_.get(), OnStart()).Times(0);
 401   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
 402       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 403   scoped_ptr<WebRtcLocalAudioTrack> track_2(
 404       new WebRtcLocalAudioTrack(adapter_2, capturer_, NULL));
 405   track_2->Start();
 406   static_cast<webrtc::AudioTrackInterface*>(
 407       adapter_2.get())->GetRenderer()->AddChannel(1);
 408
 409   // Stop the capturer will clear up the track lists in the capturer.
 410   EXPECT_CALL(*capturer_source_.get(), OnStop());
 411   capturer_->Stop();
 412
 413   // Adding a new track to the capturer.
 414   track_2->AddSink(sink.get());
 415   EXPECT_CALL(*sink, FormatIsSet()).Times(0);
 416
 417   // Stop the capturer again will not trigger stopping the source of the
 418   // capturer again..
 419   event.Reset();
 420   EXPECT_CALL(*capturer_source_.get(), OnStop()).Times(0);
 421   capturer_->Stop();
 422 }
 423
 424 // Create a new capturer with new source, connect it to a new audio track.
 425 TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) {
 426   // Setup the first audio track and start it.
 427   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
 428       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 429   scoped_ptr<WebRtcLocalAudioTrack> track_1(
 430       new WebRtcLocalAudioTrack(adapter_1, capturer_, NULL));
 431   track_1->Start();
 432
 433   // Connect a number of network channels to the |track_1|.
 434   static const int kNumberOfNetworkChannelsForTrack1 = 2;
 435   for (int i = 0; i < kNumberOfNetworkChannelsForTrack1; ++i) {
 436     static_cast<webrtc::AudioTrackInterface*>(
 437         adapter_1.get())->GetRenderer()->AddChannel(i);
 438   }
 439   // Verify the data flow by connecting the |sink_1| to |track_1|.
 440   scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink());
 441   EXPECT_CALL(*sink_1.get(),
 442               CaptureData(kNumberOfNetworkChannelsForTrack1,
 443                           0, 0, _, false))
 444       .Times(AnyNumber()).WillRepeatedly(Return());
 445   EXPECT_CALL(*sink_1.get(), FormatIsSet()).Times(AnyNumber());
 446   track_1->AddSink(sink_1.get());
 447
 448   // Create a new capturer with new source with different audio format.
 449   MockMediaConstraintFactory constraint_factory;
 450   StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
 451                           std::string(), std::string());
 452   scoped_refptr<WebRtcAudioCapturer> new_capturer(
 453       WebRtcAudioCapturer::CreateCapturer(
 454           -1, device, constraint_factory.CreateWebMediaConstraints(), NULL,
 455           NULL));
 456   scoped_refptr<MockCapturerSource> new_source(
 457       new MockCapturerSource(new_capturer.get()));
 458   EXPECT_CALL(*new_source.get(), OnInitialize(_, new_capturer.get(), -1));
 459   EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(true));
 460   EXPECT_CALL(*new_source.get(), OnStart());
 461
 462   media::AudioParameters new_param(
 463       media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
 464       media::CHANNEL_LAYOUT_MONO, 44100, 16, 441);
 465   new_capturer->SetCapturerSourceForTesting(new_source, new_param);
 466
 467   // Setup the second audio track, connect it to the new capturer and start it.
 468   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
 469       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 470   scoped_ptr<WebRtcLocalAudioTrack> track_2(
 471       new WebRtcLocalAudioTrack(adapter_2, new_capturer, NULL));
 472   track_2->Start();
 473
 474   // Connect a number of network channels to the |track_2|.
 475   static const int kNumberOfNetworkChannelsForTrack2 = 3;
 476   for (int i = 0; i < kNumberOfNetworkChannelsForTrack2; ++i) {
 477     static_cast<webrtc::AudioTrackInterface*>(
 478         adapter_2.get())->GetRenderer()->AddChannel(i);
 479   }
 480   // Verify the data flow by connecting the |sink_2| to |track_2|.
 481   scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink());
 482   base::WaitableEvent event(false, false);
 483   EXPECT_CALL(*sink_2,
 484               CaptureData(kNumberOfNetworkChannelsForTrack2, 0, 0, _, false))
 485       .Times(AnyNumber()).WillRepeatedly(Return());
 486   EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(SignalEvent(&event));
 487   track_2->AddSink(sink_2.get());
 488   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 489
 490   // Stopping the new source will stop the second track.
 491   event.Reset();
 492   EXPECT_CALL(*new_source.get(), OnStop())
 493       .Times(1).WillOnce(SignalEvent(&event));
 494   new_capturer->Stop();
 495   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 496
 497   // Stop the capturer of the first audio track.
 498   EXPECT_CALL(*capturer_source_.get(), OnStop());
 499   capturer_->Stop();
 500 }
 501
 502 // Make sure a audio track can deliver packets with a buffer size smaller than
 503 // 10ms when it is not connected with a peer connection.
 504 TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) {
 505   // Setup a capturer which works with a buffer size smaller than 10ms.
 506   media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
 507                                 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128);
 508
 509   // Create a capturer with new source which works with the format above.
 510   MockMediaConstraintFactory factory;
 511   factory.DisableDefaultAudioConstraints();
 512   scoped_refptr<WebRtcAudioCapturer> capturer(
 513       WebRtcAudioCapturer::CreateCapturer(
 514           -1,
 515           StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE,
 516                            "", "", params.sample_rate(),
 517                            params.channel_layout(),
 518                            params.frames_per_buffer()),
 519           factory.CreateWebMediaConstraints(),
 520           NULL, NULL));
 521   scoped_refptr<MockCapturerSource> source(
 522       new MockCapturerSource(capturer.get()));
 523   EXPECT_CALL(*source.get(), OnInitialize(_, capturer.get(), -1));
 524   EXPECT_CALL(*source.get(), SetAutomaticGainControl(true));
 525   EXPECT_CALL(*source.get(), OnStart());
 526   capturer->SetCapturerSourceForTesting(source, params);
 527
 528   // Setup a audio track, connect it to the capturer and start it.
 529   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
 530       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 531   scoped_ptr<WebRtcLocalAudioTrack> track(
 532       new WebRtcLocalAudioTrack(adapter, capturer, NULL));
 533   track->Start();
 534
 535   // Verify the data flow by connecting the |sink| to |track|.
 536   scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
 537   base::WaitableEvent event(false, false);
 538   EXPECT_CALL(*sink, FormatIsSet()).Times(1);
 539   // Verify the sinks are getting the packets with an expecting buffer size.
 540 #if defined(OS_ANDROID)
 541   const int expected_buffer_size = params.sample_rate() / 100;
 542 #else
 543   const int expected_buffer_size = params.frames_per_buffer();
 544 #endif
 545   EXPECT_CALL(*sink, CaptureData(
 546       0, 0, 0, _, false))
 547       .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event));
 548   track->AddSink(sink.get());
 549   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 550   EXPECT_EQ(expected_buffer_size, sink->audio_params().frames_per_buffer());
 551
 552   // Stopping the new source will stop the second track.
 553   EXPECT_CALL(*source, OnStop()).Times(1);
 554   capturer->Stop();
 555
 556   // Even though this test don't use |capturer_source_| it will be stopped
 557   // during teardown of the test harness.
 558   EXPECT_CALL(*capturer_source_.get(), OnStop());
 559 }
 560
 561 }  // namespace content