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/mock_media_constraint_factory.h"
   8 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
   9 #include "content/renderer/media/webrtc_audio_capturer.h"
  10 #include "content/renderer/media/webrtc_audio_device_impl.h"
  11 #include "content/renderer/media/webrtc_local_audio_source_provider.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(const scoped_refptr<WebRtcAudioCapturer>& capturer,
  39                   const media::AudioParameters& params)
  40     : capturer_(capturer),
  41       thread_(),
  42       closure_(false, false) {
  43     DCHECK(capturer.get());
  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_.get());
  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   scoped_refptr<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     blink::WebMediaConstraints constraints;
 174     StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
 175                             std::string(), std::string());
 176     capturer_ = WebRtcAudioCapturer::CreateCapturer(-1, device,
 177                                                     constraints, NULL);
 178     capturer_source_ = new MockCapturerSource(capturer_.get());
 179     EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_.get(), -1))
 180         .WillOnce(Return());
 181     capturer_->SetCapturerSourceForTesting(capturer_source_, params_);
 182   }
 183
 184   media::AudioParameters params_;
 185   scoped_refptr<MockCapturerSource> capturer_source_;
 186   scoped_refptr<WebRtcAudioCapturer> capturer_;
 187 };
 188
 189 // Creates a capturer and audio track, fakes its audio thread, and
 190 // connect/disconnect the sink to the audio track on the fly, the sink should
 191 // get data callback when the track is connected to the capturer but not when
 192 // the track is disconnected from the capturer.
 193 TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) {
 194   EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
 195   EXPECT_CALL(*capturer_source_.get(), OnStart());
 196   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
 197       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 198   scoped_ptr<WebRtcLocalAudioTrack> track(
 199       new WebRtcLocalAudioTrack(adapter, capturer_, NULL));
 200   static_cast<WebRtcLocalAudioSourceProvider*>(
 201       track->audio_source_provider())->SetSinkParamsForTesting(params_);
 202   track->Start();
 203   EXPECT_TRUE(track->GetAudioAdapter()->enabled());
 204
 205   // Connect a number of network channels to the audio track.
 206   static const int kNumberOfNetworkChannels = 4;
 207   for (int i = 0; i < kNumberOfNetworkChannels; ++i) {
 208     static_cast<webrtc::AudioTrackInterface*>(
 209         adapter.get())->GetRenderer()->AddChannel(i);
 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(kNumberOfNetworkChannels,
 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   static_cast<WebRtcLocalAudioSourceProvider*>(
 243       track->audio_source_provider())->SetSinkParamsForTesting(params_);
 244   track->Start();
 245   static_cast<webrtc::AudioTrackInterface*>(
 246       adapter.get())->GetRenderer()->AddChannel(0);
 247   EXPECT_TRUE(track->GetAudioAdapter()->enabled());
 248   EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(false));
 249   scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
 250   const media::AudioParameters params = capturer_->source_audio_parameters();
 251   base::WaitableEvent event(false, false);
 252   EXPECT_CALL(*sink, FormatIsSet()).Times(1);
 253   EXPECT_CALL(*sink,
 254               CaptureData(1, 0, 0, _, false)).Times(0);
 255   EXPECT_EQ(sink->audio_params().frames_per_buffer(),
 256             params.sample_rate() / 100);
 257   track->AddSink(sink.get());
 258   EXPECT_FALSE(event.TimedWait(TestTimeouts::tiny_timeout()));
 259
 260   event.Reset();
 261   EXPECT_CALL(*sink,
 262               CaptureData(1, 0, 0, _, false)).Times(AtLeast(1))
 263       .WillRepeatedly(SignalEvent(&event));
 264   EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(true));
 265   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 266   track->RemoveSink(sink.get());
 267
 268   EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return());
 269   capturer_->Stop();
 270   track.reset();
 271 }
 272
 273 // Create multiple audio tracks and enable/disable them, verify that the audio
 274 // callbacks appear/disappear.
 275 // Flaky due to a data race, see http://crbug.com/295418
 276 TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) {
 277   EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
 278   EXPECT_CALL(*capturer_source_.get(), OnStart());
 279   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
 280       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 281   scoped_ptr<WebRtcLocalAudioTrack> track_1(
 282     new WebRtcLocalAudioTrack(adapter_1, capturer_, NULL));
 283   static_cast<WebRtcLocalAudioSourceProvider*>(
 284       track_1->audio_source_provider())->SetSinkParamsForTesting(params_);
 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   static_cast<WebRtcLocalAudioSourceProvider*>(
 306       track_2->audio_source_provider())->SetSinkParamsForTesting(params_);
 307   track_2->Start();
 308   static_cast<webrtc::AudioTrackInterface*>(
 309       adapter_2.get())->GetRenderer()->AddChannel(1);
 310   EXPECT_TRUE(track_2->GetAudioAdapter()->enabled());
 311
 312   // Verify both |sink_1| and |sink_2| get data.
 313   event_1.Reset();
 314   base::WaitableEvent event_2(false, false);
 315
 316   scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink());
 317   EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(Return());
 318   EXPECT_CALL(*sink_1, CaptureData(1, 0, 0, _, false)).Times(AtLeast(1))
 319       .WillRepeatedly(SignalEvent(&event_1));
 320   EXPECT_EQ(sink_1->audio_params().frames_per_buffer(),
 321             params.sample_rate() / 100);
 322   EXPECT_CALL(*sink_2, CaptureData(1, 0, 0, _, false)).Times(AtLeast(1))
 323       .WillRepeatedly(SignalEvent(&event_2));
 324   EXPECT_EQ(sink_2->audio_params().frames_per_buffer(),
 325             params.sample_rate() / 100);
 326   track_2->AddSink(sink_2.get());
 327   EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
 328   EXPECT_TRUE(event_2.TimedWait(TestTimeouts::tiny_timeout()));
 329
 330   track_1->RemoveSink(sink_1.get());
 331   track_1->Stop();
 332   track_1.reset();
 333
 334   EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return());
 335   track_2->RemoveSink(sink_2.get());
 336   track_2->Stop();
 337   track_2.reset();
 338
 339   capturer_->Stop();
 340 }
 341
 342
 343 // Start one track and verify the capturer is correctly starting its source.
 344 // And it should be fine to not to call Stop() explicitly.
 345 TEST_F(WebRtcLocalAudioTrackTest, StartOneAudioTrack) {
 346   EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
 347   EXPECT_CALL(*capturer_source_.get(), OnStart());
 348   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
 349       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 350   scoped_ptr<WebRtcLocalAudioTrack> track(
 351       new WebRtcLocalAudioTrack(adapter, capturer_, NULL));
 352   static_cast<WebRtcLocalAudioSourceProvider*>(
 353       track->audio_source_provider())->SetSinkParamsForTesting(params_);
 354   track->Start();
 355
 356   // When the track goes away, it will automatically stop the
 357   // |capturer_source_|.
 358   EXPECT_CALL(*capturer_source_.get(), OnStop());
 359   capturer_->Stop();
 360   track.reset();
 361 }
 362
 363 // Start/Stop tracks and verify the capturer is correctly starting/stopping
 364 // its source.
 365 TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
 366   // Starting the first audio track will start the |capturer_source_|.
 367   base::WaitableEvent event(false, false);
 368   EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
 369   EXPECT_CALL(*capturer_source_.get(), OnStart()).WillOnce(SignalEvent(&event));
 370   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
 371       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 372   scoped_ptr<WebRtcLocalAudioTrack> track_1(
 373       new WebRtcLocalAudioTrack(adapter_1, capturer_, NULL));
 374   static_cast<webrtc::AudioTrackInterface*>(
 375       adapter_1.get())->GetRenderer()->AddChannel(0);
 376   static_cast<WebRtcLocalAudioSourceProvider*>(
 377       track_1->audio_source_provider())->SetSinkParamsForTesting(params_);
 378   track_1->Start();
 379   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 380
 381   // Verify the data flow by connecting the sink to |track_1|.
 382   scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
 383   event.Reset();
 384   EXPECT_CALL(*sink, FormatIsSet()).WillOnce(SignalEvent(&event));
 385   EXPECT_CALL(*sink, CaptureData(_, 0, 0, _, false))
 386       .Times(AnyNumber()).WillRepeatedly(Return());
 387   track_1->AddSink(sink.get());
 388   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 389
 390   // Start the second audio track will not start the |capturer_source_|
 391   // since it has been started.
 392   EXPECT_CALL(*capturer_source_.get(), OnStart()).Times(0);
 393   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
 394       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 395   scoped_ptr<WebRtcLocalAudioTrack> track_2(
 396       new WebRtcLocalAudioTrack(adapter_2, capturer_, NULL));
 397   static_cast<WebRtcLocalAudioSourceProvider*>(
 398       track_2->audio_source_provider())->SetSinkParamsForTesting(params_);
 399   track_2->Start();
 400   static_cast<webrtc::AudioTrackInterface*>(
 401       adapter_2.get())->GetRenderer()->AddChannel(1);
 402
 403   // Stop the capturer will clear up the track lists in the capturer.
 404   EXPECT_CALL(*capturer_source_.get(), OnStop());
 405   capturer_->Stop();
 406
 407   // Adding a new track to the capturer.
 408   track_2->AddSink(sink.get());
 409   EXPECT_CALL(*sink, FormatIsSet()).Times(0);
 410
 411   // Stop the capturer again will not trigger stopping the source of the
 412   // capturer again..
 413   event.Reset();
 414   EXPECT_CALL(*capturer_source_.get(), OnStop()).Times(0);
 415   capturer_->Stop();
 416 }
 417
 418 // Create a new capturer with new source, connect it to a new audio track.
 419 TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) {
 420   // Setup the first audio track and start it.
 421   EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
 422   EXPECT_CALL(*capturer_source_.get(), OnStart());
 423   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
 424       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 425   scoped_ptr<WebRtcLocalAudioTrack> track_1(
 426       new WebRtcLocalAudioTrack(adapter_1, capturer_, NULL));
 427   static_cast<WebRtcLocalAudioSourceProvider*>(
 428       track_1->audio_source_provider())->SetSinkParamsForTesting(params_);
 429   track_1->Start();
 430
 431   // Connect a number of network channels to the |track_1|.
 432   static const int kNumberOfNetworkChannelsForTrack1 = 2;
 433   for (int i = 0; i < kNumberOfNetworkChannelsForTrack1; ++i) {
 434     static_cast<webrtc::AudioTrackInterface*>(
 435         adapter_1.get())->GetRenderer()->AddChannel(i);
 436   }
 437   // Verify the data flow by connecting the |sink_1| to |track_1|.
 438   scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink());
 439   EXPECT_CALL(*sink_1.get(),
 440               CaptureData(kNumberOfNetworkChannelsForTrack1,
 441                           0, 0, _, false))
 442       .Times(AnyNumber()).WillRepeatedly(Return());
 443   EXPECT_CALL(*sink_1.get(), FormatIsSet()).Times(AnyNumber());
 444   track_1->AddSink(sink_1.get());
 445
 446   // Create a new capturer with new source with different audio format.
 447   blink::WebMediaConstraints constraints;
 448   StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
 449                           std::string(), std::string());
 450   scoped_refptr<WebRtcAudioCapturer> new_capturer(
 451       WebRtcAudioCapturer::CreateCapturer(-1, device, constraints, NULL));
 452   scoped_refptr<MockCapturerSource> new_source(
 453       new MockCapturerSource(new_capturer.get()));
 454   EXPECT_CALL(*new_source.get(), OnInitialize(_, new_capturer.get(), -1));
 455   media::AudioParameters new_param(
 456       media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
 457       media::CHANNEL_LAYOUT_MONO, 44100, 16, 441);
 458   new_capturer->SetCapturerSourceForTesting(new_source, new_param);
 459
 460   // Setup the second audio track, connect it to the new capturer and start it.
 461   EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(true));
 462   EXPECT_CALL(*new_source.get(), OnStart());
 463   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
 464       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 465   scoped_ptr<WebRtcLocalAudioTrack> track_2(
 466       new WebRtcLocalAudioTrack(adapter_2, new_capturer, NULL));
 467   static_cast<WebRtcLocalAudioSourceProvider*>(
 468       track_2->audio_source_provider())->SetSinkParamsForTesting(params_);
 469   track_2->Start();
 470
 471   // Connect a number of network channels to the |track_2|.
 472   static const int kNumberOfNetworkChannelsForTrack2 = 3;
 473   for (int i = 0; i < kNumberOfNetworkChannelsForTrack2; ++i) {
 474     static_cast<webrtc::AudioTrackInterface*>(
 475         adapter_2.get())->GetRenderer()->AddChannel(i);
 476   }
 477   // Verify the data flow by connecting the |sink_2| to |track_2|.
 478   scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink());
 479   base::WaitableEvent event(false, false);
 480   EXPECT_CALL(*sink_2,
 481               CaptureData(kNumberOfNetworkChannelsForTrack2, 0, 0, _, false))
 482       .Times(AnyNumber()).WillRepeatedly(Return());
 483   EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(SignalEvent(&event));
 484   track_2->AddSink(sink_2.get());
 485   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 486
 487   // Stopping the new source will stop the second track.
 488   event.Reset();
 489   EXPECT_CALL(*new_source.get(), OnStop())
 490       .Times(1).WillOnce(SignalEvent(&event));
 491   new_capturer->Stop();
 492   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 493
 494   // Stop the capturer of the first audio track.
 495   EXPECT_CALL(*capturer_source_.get(), OnStop());
 496   capturer_->Stop();
 497 }
 498
 499
 500 // Make sure a audio track can deliver packets with a buffer size smaller than
 501 // 10ms when it is not connected with a peer connection.
 502 TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) {
 503   // Setup a capturer which works with a buffer size smaller than 10ms.
 504   media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
 505                                 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128);
 506
 507   // Create a capturer with new source which works with the format above.
 508   MockMediaConstraintFactory factory;
 509   factory.DisableDefaultAudioConstraints();
 510   scoped_refptr<WebRtcAudioCapturer> capturer(
 511       WebRtcAudioCapturer::CreateCapturer(
 512           -1,
 513           StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE,
 514                            "", "", params.sample_rate(),
 515                            params.channel_layout(),
 516                            params.frames_per_buffer()),
 517           factory.CreateWebMediaConstraints(),
 518           NULL));
 519   scoped_refptr<MockCapturerSource> source(
 520       new MockCapturerSource(capturer.get()));
 521   EXPECT_CALL(*source.get(), OnInitialize(_, capturer.get(), -1));
 522   capturer->SetCapturerSourceForTesting(source, params);
 523
 524   // Setup a audio track, connect it to the capturer and start it.
 525   EXPECT_CALL(*source.get(), SetAutomaticGainControl(true));
 526   EXPECT_CALL(*source.get(), OnStart());
 527   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
 528       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 529   scoped_ptr<WebRtcLocalAudioTrack> track(
 530       new WebRtcLocalAudioTrack(adapter, capturer, NULL));
 531   static_cast<WebRtcLocalAudioSourceProvider*>(
 532       track->audio_source_provider())->SetSinkParamsForTesting(params);
 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
 557 }  // namespace content