src/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   scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
 212   base::WaitableEvent event(false, false);
 213   EXPECT_CALL(*sink, FormatIsSet());
 214   EXPECT_CALL(*sink,
 215       CaptureData(0,
 216                   0,
 217                   0,
 218                   _,
 219                   false)).Times(AtLeast(1))
 220       .WillRepeatedly(SignalEvent(&event));
 221   track->AddSink(sink.get());
 222   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 223   track->RemoveSink(sink.get());
 224
 225   EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return());
 226   capturer_->Stop();
 227 }
 228
 229 // The same setup as ConnectAndDisconnectOneSink, but enable and disable the
 230 // audio track on the fly. When the audio track is disabled, there is no data
 231 // callback to the sink; when the audio track is enabled, there comes data
 232 // callback.
 233 // TODO(xians): Enable this test after resolving the racing issue that TSAN
 234 // reports on MediaStreamTrack::enabled();
 235 TEST_F(WebRtcLocalAudioTrackTest,  DISABLED_DisableEnableAudioTrack) {
 236   EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
 237   EXPECT_CALL(*capturer_source_.get(), OnStart());
 238   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
 239       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 240   scoped_ptr<WebRtcLocalAudioTrack> track(
 241       new WebRtcLocalAudioTrack(adapter, capturer_, NULL));
 242   track->Start();
 243   EXPECT_TRUE(track->GetAudioAdapter()->enabled());
 244   EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(false));
 245   scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
 246   const media::AudioParameters params = capturer_->source_audio_parameters();
 247   base::WaitableEvent event(false, false);
 248   EXPECT_CALL(*sink, FormatIsSet()).Times(1);
 249   EXPECT_CALL(*sink,
 250               CaptureData(0, 0, 0, _, false)).Times(0);
 251   EXPECT_EQ(sink->audio_params().frames_per_buffer(),
 252             params.sample_rate() / 100);
 253   track->AddSink(sink.get());
 254   EXPECT_FALSE(event.TimedWait(TestTimeouts::tiny_timeout()));
 255
 256   event.Reset();
 257   EXPECT_CALL(*sink, CaptureData(0, 0, 0, _, false)).Times(AtLeast(1))
 258       .WillRepeatedly(SignalEvent(&event));
 259   EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(true));
 260   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 261   track->RemoveSink(sink.get());
 262
 263   EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return());
 264   capturer_->Stop();
 265   track.reset();
 266 }
 267
 268 // Create multiple audio tracks and enable/disable them, verify that the audio
 269 // callbacks appear/disappear.
 270 // Flaky due to a data race, see http://crbug.com/295418
 271 TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) {
 272   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
 273       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 274   scoped_ptr<WebRtcLocalAudioTrack> track_1(
 275     new WebRtcLocalAudioTrack(adapter_1, capturer_, NULL));
 276   track_1->Start();
 277   EXPECT_TRUE(track_1->GetAudioAdapter()->enabled());
 278   scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink());
 279   const media::AudioParameters params = capturer_->source_audio_parameters();
 280   base::WaitableEvent event_1(false, false);
 281   EXPECT_CALL(*sink_1, FormatIsSet()).WillOnce(Return());
 282   EXPECT_CALL(*sink_1,
 283       CaptureData(0, 0, 0, _, false)).Times(AtLeast(1))
 284       .WillRepeatedly(SignalEvent(&event_1));
 285   EXPECT_EQ(sink_1->audio_params().frames_per_buffer(),
 286             params.sample_rate() / 100);
 287   track_1->AddSink(sink_1.get());
 288   EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
 289
 290   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
 291       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 292   scoped_ptr<WebRtcLocalAudioTrack> track_2(
 293     new WebRtcLocalAudioTrack(adapter_2, capturer_, NULL));
 294   track_2->Start();
 295   EXPECT_TRUE(track_2->GetAudioAdapter()->enabled());
 296
 297   // Verify both |sink_1| and |sink_2| get data.
 298   event_1.Reset();
 299   base::WaitableEvent event_2(false, false);
 300
 301   scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink());
 302   EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(Return());
 303   EXPECT_CALL(*sink_1, CaptureData(0, 0, 0, _, false)).Times(AtLeast(1))
 304       .WillRepeatedly(SignalEvent(&event_1));
 305   EXPECT_EQ(sink_1->audio_params().frames_per_buffer(),
 306             params.sample_rate() / 100);
 307   EXPECT_CALL(*sink_2, CaptureData(0, 0, 0, _, false)).Times(AtLeast(1))
 308       .WillRepeatedly(SignalEvent(&event_2));
 309   EXPECT_EQ(sink_2->audio_params().frames_per_buffer(),
 310             params.sample_rate() / 100);
 311   track_2->AddSink(sink_2.get());
 312   EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
 313   EXPECT_TRUE(event_2.TimedWait(TestTimeouts::tiny_timeout()));
 314
 315   track_1->RemoveSink(sink_1.get());
 316   track_1->Stop();
 317   track_1.reset();
 318
 319   EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return());
 320   track_2->RemoveSink(sink_2.get());
 321   track_2->Stop();
 322   track_2.reset();
 323 }
 324
 325
 326 // Start one track and verify the capturer is correctly starting its source.
 327 // And it should be fine to not to call Stop() explicitly.
 328 TEST_F(WebRtcLocalAudioTrackTest, StartOneAudioTrack) {
 329   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
 330       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 331   scoped_ptr<WebRtcLocalAudioTrack> track(
 332       new WebRtcLocalAudioTrack(adapter, capturer_, NULL));
 333   track->Start();
 334
 335   // When the track goes away, it will automatically stop the
 336   // |capturer_source_|.
 337   EXPECT_CALL(*capturer_source_.get(), OnStop());
 338   track.reset();
 339 }
 340
 341 // Start two tracks and verify the capturer is correctly starting its source.
 342 // When the last track connected to the capturer is stopped, the source is
 343 // stopped.
 344 TEST_F(WebRtcLocalAudioTrackTest, StartTwoAudioTracks) {
 345   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter1(
 346       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 347   scoped_ptr<WebRtcLocalAudioTrack> track1(
 348       new WebRtcLocalAudioTrack(adapter1, capturer_, NULL));
 349   track1->Start();
 350
 351   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter2(
 352         WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 353   scoped_ptr<WebRtcLocalAudioTrack> track2(
 354       new WebRtcLocalAudioTrack(adapter2, capturer_, NULL));
 355   track2->Start();
 356
 357   track1->Stop();
 358   // When the last track is stopped, it will automatically stop the
 359   // |capturer_source_|.
 360   EXPECT_CALL(*capturer_source_.get(), OnStop());
 361   track2->Stop();
 362 }
 363
 364 // Start/Stop tracks and verify the capturer is correctly starting/stopping
 365 // its source.
 366 TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
 367   base::WaitableEvent event(false, false);
 368   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
 369       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 370   scoped_ptr<WebRtcLocalAudioTrack> track_1(
 371       new WebRtcLocalAudioTrack(adapter_1, capturer_, NULL));
 372   track_1->Start();
 373
 374   // Verify the data flow by connecting the sink to |track_1|.
 375   scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
 376   event.Reset();
 377   EXPECT_CALL(*sink, FormatIsSet()).WillOnce(SignalEvent(&event));
 378   EXPECT_CALL(*sink, CaptureData(_, 0, 0, _, false))
 379       .Times(AnyNumber()).WillRepeatedly(Return());
 380   track_1->AddSink(sink.get());
 381   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 382
 383   // Start the second audio track will not start the |capturer_source_|
 384   // since it has been started.
 385   EXPECT_CALL(*capturer_source_.get(), OnStart()).Times(0);
 386   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
 387       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 388   scoped_ptr<WebRtcLocalAudioTrack> track_2(
 389       new WebRtcLocalAudioTrack(adapter_2, capturer_, NULL));
 390   track_2->Start();
 391
 392   // Stop the capturer will clear up the track lists in the capturer.
 393   EXPECT_CALL(*capturer_source_.get(), OnStop());
 394   capturer_->Stop();
 395
 396   // Adding a new track to the capturer.
 397   track_2->AddSink(sink.get());
 398   EXPECT_CALL(*sink, FormatIsSet()).Times(0);
 399
 400   // Stop the capturer again will not trigger stopping the source of the
 401   // capturer again..
 402   event.Reset();
 403   EXPECT_CALL(*capturer_source_.get(), OnStop()).Times(0);
 404   capturer_->Stop();
 405 }
 406
 407 // Contains data races reported by tsan: crbug.com/404133
 408 #if defined(THREAD_SANITIZER)
 409   #define DISABLE_ON_TSAN(function) DISABLED_##function
 410 #else
 411   #define DISABLE_ON_TSAN(function) function
 412 #endif
 413
 414 // Create a new capturer with new source, connect it to a new audio track.
 415 TEST_F(WebRtcLocalAudioTrackTest,
 416        DISABLE_ON_TSAN(ConnectTracksToDifferentCapturers)) {
 417   // Setup the first audio track and start it.
 418   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
 419       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 420   scoped_ptr<WebRtcLocalAudioTrack> track_1(
 421       new WebRtcLocalAudioTrack(adapter_1, capturer_, NULL));
 422   track_1->Start();
 423
 424   // Verify the data flow by connecting the |sink_1| to |track_1|.
 425   scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink());
 426   EXPECT_CALL(*sink_1.get(), CaptureData(0, 0, 0, _, false))
 427       .Times(AnyNumber()).WillRepeatedly(Return());
 428   EXPECT_CALL(*sink_1.get(), FormatIsSet()).Times(AnyNumber());
 429   track_1->AddSink(sink_1.get());
 430
 431   // Create a new capturer with new source with different audio format.
 432   MockMediaConstraintFactory constraint_factory;
 433   StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
 434                           std::string(), std::string());
 435   scoped_refptr<WebRtcAudioCapturer> new_capturer(
 436       WebRtcAudioCapturer::CreateCapturer(
 437           -1, device, constraint_factory.CreateWebMediaConstraints(), NULL,
 438           NULL));
 439   scoped_refptr<MockCapturerSource> new_source(
 440       new MockCapturerSource(new_capturer.get()));
 441   EXPECT_CALL(*new_source.get(), OnInitialize(_, new_capturer.get(), -1));
 442   EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(true));
 443   EXPECT_CALL(*new_source.get(), OnStart());
 444
 445   media::AudioParameters new_param(
 446       media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
 447       media::CHANNEL_LAYOUT_MONO, 44100, 16, 441);
 448   new_capturer->SetCapturerSourceForTesting(new_source, new_param);
 449
 450   // Setup the second audio track, connect it to the new capturer and start it.
 451   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
 452       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 453   scoped_ptr<WebRtcLocalAudioTrack> track_2(
 454       new WebRtcLocalAudioTrack(adapter_2, new_capturer, NULL));
 455   track_2->Start();
 456
 457   // Verify the data flow by connecting the |sink_2| to |track_2|.
 458   scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink());
 459   base::WaitableEvent event(false, false);
 460   EXPECT_CALL(*sink_2, CaptureData(0, 0, 0, _, false))
 461       .Times(AnyNumber()).WillRepeatedly(Return());
 462   EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(SignalEvent(&event));
 463   track_2->AddSink(sink_2.get());
 464   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 465
 466   // Stopping the new source will stop the second track.
 467   event.Reset();
 468   EXPECT_CALL(*new_source.get(), OnStop())
 469       .Times(1).WillOnce(SignalEvent(&event));
 470   new_capturer->Stop();
 471   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 472
 473   // Stop the capturer of the first audio track.
 474   EXPECT_CALL(*capturer_source_.get(), OnStop());
 475   capturer_->Stop();
 476 }
 477
 478 // Make sure a audio track can deliver packets with a buffer size smaller than
 479 // 10ms when it is not connected with a peer connection.
 480 TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) {
 481   // Setup a capturer which works with a buffer size smaller than 10ms.
 482   media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
 483                                 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128);
 484
 485   // Create a capturer with new source which works with the format above.
 486   MockMediaConstraintFactory factory;
 487   factory.DisableDefaultAudioConstraints();
 488   scoped_refptr<WebRtcAudioCapturer> capturer(
 489       WebRtcAudioCapturer::CreateCapturer(
 490           -1,
 491           StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE,
 492                            "", "", params.sample_rate(),
 493                            params.channel_layout(),
 494                            params.frames_per_buffer()),
 495           factory.CreateWebMediaConstraints(),
 496           NULL, NULL));
 497   scoped_refptr<MockCapturerSource> source(
 498       new MockCapturerSource(capturer.get()));
 499   EXPECT_CALL(*source.get(), OnInitialize(_, capturer.get(), -1));
 500   EXPECT_CALL(*source.get(), SetAutomaticGainControl(true));
 501   EXPECT_CALL(*source.get(), OnStart());
 502   capturer->SetCapturerSourceForTesting(source, params);
 503
 504   // Setup a audio track, connect it to the capturer and start it.
 505   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
 506       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 507   scoped_ptr<WebRtcLocalAudioTrack> track(
 508       new WebRtcLocalAudioTrack(adapter, capturer, NULL));
 509   track->Start();
 510
 511   // Verify the data flow by connecting the |sink| to |track|.
 512   scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
 513   base::WaitableEvent event(false, false);
 514   EXPECT_CALL(*sink, FormatIsSet()).Times(1);
 515   // Verify the sinks are getting the packets with an expecting buffer size.
 516 #if defined(OS_ANDROID)
 517   const int expected_buffer_size = params.sample_rate() / 100;
 518 #else
 519   const int expected_buffer_size = params.frames_per_buffer();
 520 #endif
 521   EXPECT_CALL(*sink, CaptureData(
 522       0, 0, 0, _, false))
 523       .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event));
 524   track->AddSink(sink.get());
 525   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 526   EXPECT_EQ(expected_buffer_size, sink->audio_params().frames_per_buffer());
 527
 528   // Stopping the new source will stop the second track.
 529   EXPECT_CALL(*source, OnStop()).Times(1);
 530   capturer->Stop();
 531
 532   // Even though this test don't use |capturer_source_| it will be stopped
 533   // during teardown of the test harness.
 534   EXPECT_CALL(*capturer_source_.get(), OnStop());
 535 }
 536
 537 }  // namespace content