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/WebKit/public/web/WebHeap.h"
  20 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
  21
  22 using ::testing::_;
  23 using ::testing::AnyNumber;
  24 using ::testing::AtLeast;
  25 using ::testing::Return;
  26
  27 namespace content {
  28
  29 namespace {
  30
  31 ACTION_P(SignalEvent, event) {
  32   event->Signal();
  33 }
  34
  35 // A simple thread that we use to fake the audio thread which provides data to
  36 // the |WebRtcAudioCapturer|.
  37 class FakeAudioThread : public base::PlatformThread::Delegate {
  38  public:
  39   FakeAudioThread(WebRtcAudioCapturer* capturer,
  40                   const media::AudioParameters& params)
  41     : capturer_(capturer),
  42       thread_(),
  43       closure_(false, false) {
  44     DCHECK(capturer);
  45     audio_bus_ = media::AudioBus::Create(params);
  46   }
  47
  48   virtual ~FakeAudioThread() { DCHECK(thread_.is_null()); }
  49
  50   // base::PlatformThread::Delegate:
  51   virtual void ThreadMain() OVERRIDE {
  52     while (true) {
  53       if (closure_.IsSignaled())
  54         return;
  55
  56       media::AudioCapturerSource::CaptureCallback* callback =
  57           static_cast<media::AudioCapturerSource::CaptureCallback*>(
  58               capturer_);
  59       audio_bus_->Zero();
  60       callback->Capture(audio_bus_.get(), 0, 0, false);
  61
  62       // Sleep 1ms to yield the resource for the main thread.
  63       base::PlatformThread::Sleep(base::TimeDelta::FromMilliseconds(1));
  64     }
  65   }
  66
  67   void Start() {
  68     base::PlatformThread::CreateWithPriority(
  69         0, this, &thread_, base::kThreadPriority_RealtimeAudio);
  70     CHECK(!thread_.is_null());
  71   }
  72
  73   void Stop() {
  74     closure_.Signal();
  75     base::PlatformThread::Join(thread_);
  76     thread_ = base::PlatformThreadHandle();
  77   }
  78
  79  private:
  80   scoped_ptr<media::AudioBus> audio_bus_;
  81   WebRtcAudioCapturer* capturer_;
  82   base::PlatformThreadHandle thread_;
  83   base::WaitableEvent closure_;
  84   DISALLOW_COPY_AND_ASSIGN(FakeAudioThread);
  85 };
  86
  87 class MockCapturerSource : public media::AudioCapturerSource {
  88  public:
  89   explicit MockCapturerSource(WebRtcAudioCapturer* capturer)
  90       : capturer_(capturer) {}
  91   MOCK_METHOD3(OnInitialize, void(const media::AudioParameters& params,
  92                                   CaptureCallback* callback,
  93                                   int session_id));
  94   MOCK_METHOD0(OnStart, void());
  95   MOCK_METHOD0(OnStop, void());
  96   MOCK_METHOD1(SetVolume, void(double volume));
  97   MOCK_METHOD1(SetAutomaticGainControl, void(bool enable));
  98
  99   virtual void Initialize(const media::AudioParameters& params,
 100                           CaptureCallback* callback,
 101                           int session_id) OVERRIDE {
 102     DCHECK(params.IsValid());
 103     params_ = params;
 104     OnInitialize(params, callback, session_id);
 105   }
 106   virtual void Start() OVERRIDE {
 107     audio_thread_.reset(new FakeAudioThread(capturer_, params_));
 108     audio_thread_->Start();
 109     OnStart();
 110   }
 111   virtual void Stop() OVERRIDE {
 112     audio_thread_->Stop();
 113     audio_thread_.reset();
 114     OnStop();
 115   }
 116  protected:
 117   virtual ~MockCapturerSource() {}
 118
 119  private:
 120   scoped_ptr<FakeAudioThread> audio_thread_;
 121   WebRtcAudioCapturer* capturer_;
 122   media::AudioParameters params_;
 123 };
 124
 125 // TODO(xians): Use MediaStreamAudioSink.
 126 class MockMediaStreamAudioSink : public PeerConnectionAudioSink {
 127  public:
 128   MockMediaStreamAudioSink() {}
 129   ~MockMediaStreamAudioSink() {}
 130   int OnData(const int16* audio_data,
 131              int sample_rate,
 132              int number_of_channels,
 133              int number_of_frames,
 134              const std::vector<int>& channels,
 135              int audio_delay_milliseconds,
 136              int current_volume,
 137              bool need_audio_processing,
 138              bool key_pressed) OVERRIDE {
 139     EXPECT_EQ(params_.sample_rate(), sample_rate);
 140     EXPECT_EQ(params_.channels(), number_of_channels);
 141     EXPECT_EQ(params_.frames_per_buffer(), number_of_frames);
 142     CaptureData(channels.size(),
 143                 audio_delay_milliseconds,
 144                 current_volume,
 145                 need_audio_processing,
 146                 key_pressed);
 147     return 0;
 148   }
 149   MOCK_METHOD5(CaptureData,
 150                void(int number_of_network_channels,
 151                     int audio_delay_milliseconds,
 152                     int current_volume,
 153                     bool need_audio_processing,
 154                     bool key_pressed));
 155   void OnSetFormat(const media::AudioParameters& params) {
 156     params_ = params;
 157     FormatIsSet();
 158   }
 159   MOCK_METHOD0(FormatIsSet, void());
 160
 161   const media::AudioParameters& audio_params() const { return params_; }
 162
 163  private:
 164   media::AudioParameters params_;
 165 };
 166
 167 }  // namespace
 168
 169 class WebRtcLocalAudioTrackTest : public ::testing::Test {
 170  protected:
 171   virtual void SetUp() OVERRIDE {
 172     params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
 173                   media::CHANNEL_LAYOUT_STEREO, 2, 48000, 16, 480);
 174     MockMediaConstraintFactory constraint_factory;
 175     blink_source_.initialize("dummy", blink::WebMediaStreamSource::TypeAudio,
 176                              "dummy");
 177     MediaStreamAudioSource* audio_source = new MediaStreamAudioSource();
 178     blink_source_.setExtraData(audio_source);
 179
 180     StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
 181                             std::string(), std::string());
 182     capturer_ = WebRtcAudioCapturer::CreateCapturer(
 183         -1, device, constraint_factory.CreateWebMediaConstraints(), NULL,
 184         audio_source);
 185     audio_source->SetAudioCapturer(capturer_.get());
 186     capturer_source_ = new MockCapturerSource(capturer_.get());
 187     EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_.get(), -1))
 188         .WillOnce(Return());
 189     EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
 190     EXPECT_CALL(*capturer_source_.get(), OnStart());
 191     capturer_->SetCapturerSourceForTesting(capturer_source_, params_);
 192   }
 193
 194   virtual void TearDown() OVERRIDE {
 195     blink_source_.reset();
 196     blink::WebHeap::collectAllGarbageForTesting();
 197   }
 198
 199   media::AudioParameters params_;
 200   blink::WebMediaStreamSource blink_source_;
 201   scoped_refptr<MockCapturerSource> capturer_source_;
 202   scoped_refptr<WebRtcAudioCapturer> capturer_;
 203 };
 204
 205 // Creates a capturer and audio track, fakes its audio thread, and
 206 // connect/disconnect the sink to the audio track on the fly, the sink should
 207 // get data callback when the track is connected to the capturer but not when
 208 // the track is disconnected from the capturer.
 209 TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) {
 210   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
 211       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 212   scoped_ptr<WebRtcLocalAudioTrack> track(
 213       new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL));
 214   track->Start();
 215   EXPECT_TRUE(track->GetAudioAdapter()->enabled());
 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(0,
 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.get(), capturer_, NULL));
 248   track->Start();
 249   EXPECT_TRUE(track->GetAudioAdapter()->enabled());
 250   EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(false));
 251   scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
 252   const media::AudioParameters params = capturer_->source_audio_parameters();
 253   base::WaitableEvent event(false, false);
 254   EXPECT_CALL(*sink, FormatIsSet()).Times(1);
 255   EXPECT_CALL(*sink,
 256               CaptureData(0, 0, 0, _, false)).Times(0);
 257   EXPECT_EQ(sink->audio_params().frames_per_buffer(),
 258             params.sample_rate() / 100);
 259   track->AddSink(sink.get());
 260   EXPECT_FALSE(event.TimedWait(TestTimeouts::tiny_timeout()));
 261
 262   event.Reset();
 263   EXPECT_CALL(*sink, CaptureData(0, 0, 0, _, false)).Times(AtLeast(1))
 264       .WillRepeatedly(SignalEvent(&event));
 265   EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(true));
 266   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 267   track->RemoveSink(sink.get());
 268
 269   EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return());
 270   capturer_->Stop();
 271   track.reset();
 272 }
 273
 274 // Create multiple audio tracks and enable/disable them, verify that the audio
 275 // callbacks appear/disappear.
 276 // Flaky due to a data race, see http://crbug.com/295418
 277 TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) {
 278   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
 279       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 280   scoped_ptr<WebRtcLocalAudioTrack> track_1(
 281       new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL));
 282   track_1->Start();
 283   EXPECT_TRUE(track_1->GetAudioAdapter()->enabled());
 284   scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink());
 285   const media::AudioParameters params = capturer_->source_audio_parameters();
 286   base::WaitableEvent event_1(false, false);
 287   EXPECT_CALL(*sink_1, FormatIsSet()).WillOnce(Return());
 288   EXPECT_CALL(*sink_1,
 289       CaptureData(0, 0, 0, _, false)).Times(AtLeast(1))
 290       .WillRepeatedly(SignalEvent(&event_1));
 291   EXPECT_EQ(sink_1->audio_params().frames_per_buffer(),
 292             params.sample_rate() / 100);
 293   track_1->AddSink(sink_1.get());
 294   EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
 295
 296   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
 297       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 298   scoped_ptr<WebRtcLocalAudioTrack> track_2(
 299       new WebRtcLocalAudioTrack(adapter_2.get(), capturer_, NULL));
 300   track_2->Start();
 301   EXPECT_TRUE(track_2->GetAudioAdapter()->enabled());
 302
 303   // Verify both |sink_1| and |sink_2| get data.
 304   event_1.Reset();
 305   base::WaitableEvent event_2(false, false);
 306
 307   scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink());
 308   EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(Return());
 309   EXPECT_CALL(*sink_1, CaptureData(0, 0, 0, _, false)).Times(AtLeast(1))
 310       .WillRepeatedly(SignalEvent(&event_1));
 311   EXPECT_EQ(sink_1->audio_params().frames_per_buffer(),
 312             params.sample_rate() / 100);
 313   EXPECT_CALL(*sink_2, CaptureData(0, 0, 0, _, false)).Times(AtLeast(1))
 314       .WillRepeatedly(SignalEvent(&event_2));
 315   EXPECT_EQ(sink_2->audio_params().frames_per_buffer(),
 316             params.sample_rate() / 100);
 317   track_2->AddSink(sink_2.get());
 318   EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
 319   EXPECT_TRUE(event_2.TimedWait(TestTimeouts::tiny_timeout()));
 320
 321   track_1->RemoveSink(sink_1.get());
 322   track_1->Stop();
 323   track_1.reset();
 324
 325   EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return());
 326   track_2->RemoveSink(sink_2.get());
 327   track_2->Stop();
 328   track_2.reset();
 329 }
 330
 331
 332 // Start one track and verify the capturer is correctly starting its source.
 333 // And it should be fine to not to call Stop() explicitly.
 334 TEST_F(WebRtcLocalAudioTrackTest, StartOneAudioTrack) {
 335   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
 336       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 337   scoped_ptr<WebRtcLocalAudioTrack> track(
 338       new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL));
 339   track->Start();
 340
 341   // When the track goes away, it will automatically stop the
 342   // |capturer_source_|.
 343   EXPECT_CALL(*capturer_source_.get(), OnStop());
 344   track.reset();
 345 }
 346
 347 // Start two tracks and verify the capturer is correctly starting its source.
 348 // When the last track connected to the capturer is stopped, the source is
 349 // stopped.
 350 TEST_F(WebRtcLocalAudioTrackTest, StartTwoAudioTracks) {
 351   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter1(
 352       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 353   scoped_ptr<WebRtcLocalAudioTrack> track1(
 354       new WebRtcLocalAudioTrack(adapter1.get(), capturer_, NULL));
 355   track1->Start();
 356
 357   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter2(
 358         WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 359   scoped_ptr<WebRtcLocalAudioTrack> track2(
 360       new WebRtcLocalAudioTrack(adapter2.get(), capturer_, NULL));
 361   track2->Start();
 362
 363   track1->Stop();
 364   // When the last track is stopped, it will automatically stop the
 365   // |capturer_source_|.
 366   EXPECT_CALL(*capturer_source_.get(), OnStop());
 367   track2->Stop();
 368 }
 369
 370 // Start/Stop tracks and verify the capturer is correctly starting/stopping
 371 // its source.
 372 TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
 373   base::WaitableEvent event(false, false);
 374   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
 375       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 376   scoped_ptr<WebRtcLocalAudioTrack> track_1(
 377       new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL));
 378   track_1->Start();
 379
 380   // Verify the data flow by connecting the sink to |track_1|.
 381   scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
 382   event.Reset();
 383   EXPECT_CALL(*sink, FormatIsSet()).WillOnce(SignalEvent(&event));
 384   EXPECT_CALL(*sink, CaptureData(_, 0, 0, _, false))
 385       .Times(AnyNumber()).WillRepeatedly(Return());
 386   track_1->AddSink(sink.get());
 387   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 388
 389   // Start the second audio track will not start the |capturer_source_|
 390   // since it has been started.
 391   EXPECT_CALL(*capturer_source_.get(), OnStart()).Times(0);
 392   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
 393       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 394   scoped_ptr<WebRtcLocalAudioTrack> track_2(
 395       new WebRtcLocalAudioTrack(adapter_2.get(), capturer_, NULL));
 396   track_2->Start();
 397
 398   // Stop the capturer will clear up the track lists in the capturer.
 399   EXPECT_CALL(*capturer_source_.get(), OnStop());
 400   capturer_->Stop();
 401
 402   // Adding a new track to the capturer.
 403   track_2->AddSink(sink.get());
 404   EXPECT_CALL(*sink, FormatIsSet()).Times(0);
 405
 406   // Stop the capturer again will not trigger stopping the source of the
 407   // capturer again..
 408   event.Reset();
 409   EXPECT_CALL(*capturer_source_.get(), OnStop()).Times(0);
 410   capturer_->Stop();
 411 }
 412
 413 // Contains data races reported by tsan: crbug.com/404133
 414 #if defined(THREAD_SANITIZER)
 415   #define DISABLE_ON_TSAN(function) DISABLED_##function
 416 #else
 417   #define DISABLE_ON_TSAN(function) function
 418 #endif
 419
 420 // Create a new capturer with new source, connect it to a new audio track.
 421 TEST_F(WebRtcLocalAudioTrackTest,
 422        DISABLE_ON_TSAN(ConnectTracksToDifferentCapturers)) {
 423   // Setup the first audio track and start it.
 424   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
 425       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 426   scoped_ptr<WebRtcLocalAudioTrack> track_1(
 427       new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL));
 428   track_1->Start();
 429
 430   // Verify the data flow by connecting the |sink_1| to |track_1|.
 431   scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink());
 432   EXPECT_CALL(*sink_1.get(), CaptureData(0, 0, 0, _, false))
 433       .Times(AnyNumber()).WillRepeatedly(Return());
 434   EXPECT_CALL(*sink_1.get(), FormatIsSet()).Times(AnyNumber());
 435   track_1->AddSink(sink_1.get());
 436
 437   // Create a new capturer with new source with different audio format.
 438   MockMediaConstraintFactory constraint_factory;
 439   StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
 440                           std::string(), std::string());
 441   scoped_refptr<WebRtcAudioCapturer> new_capturer(
 442       WebRtcAudioCapturer::CreateCapturer(
 443           -1, device, constraint_factory.CreateWebMediaConstraints(), NULL,
 444           NULL));
 445   scoped_refptr<MockCapturerSource> new_source(
 446       new MockCapturerSource(new_capturer.get()));
 447   EXPECT_CALL(*new_source.get(), OnInitialize(_, new_capturer.get(), -1));
 448   EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(true));
 449   EXPECT_CALL(*new_source.get(), OnStart());
 450
 451   media::AudioParameters new_param(
 452       media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
 453       media::CHANNEL_LAYOUT_MONO, 44100, 16, 441);
 454   new_capturer->SetCapturerSourceForTesting(new_source, new_param);
 455
 456   // Setup the second audio track, connect it to the new capturer and start it.
 457   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
 458       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 459   scoped_ptr<WebRtcLocalAudioTrack> track_2(
 460       new WebRtcLocalAudioTrack(adapter_2.get(), new_capturer, NULL));
 461   track_2->Start();
 462
 463   // Verify the data flow by connecting the |sink_2| to |track_2|.
 464   scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink());
 465   base::WaitableEvent event(false, false);
 466   EXPECT_CALL(*sink_2, CaptureData(0, 0, 0, _, false))
 467       .Times(AnyNumber()).WillRepeatedly(Return());
 468   EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(SignalEvent(&event));
 469   track_2->AddSink(sink_2.get());
 470   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 471
 472   // Stopping the new source will stop the second track.
 473   event.Reset();
 474   EXPECT_CALL(*new_source.get(), OnStop())
 475       .Times(1).WillOnce(SignalEvent(&event));
 476   new_capturer->Stop();
 477   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 478
 479   // Stop the capturer of the first audio track.
 480   EXPECT_CALL(*capturer_source_.get(), OnStop());
 481   capturer_->Stop();
 482 }
 483
 484 // Make sure a audio track can deliver packets with a buffer size smaller than
 485 // 10ms when it is not connected with a peer connection.
 486 TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) {
 487   // Setup a capturer which works with a buffer size smaller than 10ms.
 488   media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
 489                                 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128);
 490
 491   // Create a capturer with new source which works with the format above.
 492   MockMediaConstraintFactory factory;
 493   factory.DisableDefaultAudioConstraints();
 494   scoped_refptr<WebRtcAudioCapturer> capturer(
 495       WebRtcAudioCapturer::CreateCapturer(
 496           -1,
 497           StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE,
 498                            "", "", params.sample_rate(),
 499                            params.channel_layout(),
 500                            params.frames_per_buffer()),
 501           factory.CreateWebMediaConstraints(),
 502           NULL, NULL));
 503   scoped_refptr<MockCapturerSource> source(
 504       new MockCapturerSource(capturer.get()));
 505   EXPECT_CALL(*source.get(), OnInitialize(_, capturer.get(), -1));
 506   EXPECT_CALL(*source.get(), SetAutomaticGainControl(true));
 507   EXPECT_CALL(*source.get(), OnStart());
 508   capturer->SetCapturerSourceForTesting(source, params);
 509
 510   // Setup a audio track, connect it to the capturer and start it.
 511   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
 512       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
 513   scoped_ptr<WebRtcLocalAudioTrack> track(
 514       new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL));
 515   track->Start();
 516
 517   // Verify the data flow by connecting the |sink| to |track|.
 518   scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
 519   base::WaitableEvent event(false, false);
 520   EXPECT_CALL(*sink, FormatIsSet()).Times(1);
 521   // Verify the sinks are getting the packets with an expecting buffer size.
 522 #if defined(OS_ANDROID)
 523   const int expected_buffer_size = params.sample_rate() / 100;
 524 #else
 525   const int expected_buffer_size = params.frames_per_buffer();
 526 #endif
 527   EXPECT_CALL(*sink, CaptureData(
 528       0, 0, 0, _, false))
 529       .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event));
 530   track->AddSink(sink.get());
 531   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 532   EXPECT_EQ(expected_buffer_size, sink->audio_params().frames_per_buffer());
 533
 534   // Stopping the new source will stop the second track.
 535   EXPECT_CALL(*source.get(), OnStop()).Times(1);
 536   capturer->Stop();
 537
 538   // Even though this test don't use |capturer_source_| it will be stopped
 539   // during teardown of the test harness.
 540   EXPECT_CALL(*capturer_source_.get(), OnStop());
 541 }
 542
 543 }  // namespace content