webrtc_local_audio_track_unittest.cc revision 0529e5d033099cbfc42635f6f6183833b09dff6e 
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
21using ::testing::_;
22using ::testing::AnyNumber;
23using ::testing::AtLeast;
24using ::testing::Return;
25
26namespace content {
27
28namespace {
29
30ACTION_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|.
36class 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
86class 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.
125class 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
168class 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    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(-1, device,
182                                                    constraints, 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.
203TEST_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();
241TEST_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
280TEST_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.
341TEST_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.
357TEST_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.
379TEST_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.
425TEST_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  blink::WebMediaConstraints constraints;
450  StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
451                          std::string(), std::string());
452  scoped_refptr<WebRtcAudioCapturer> new_capturer(
453      WebRtcAudioCapturer::CreateCapturer(-1, device, constraints, NULL, NULL));
454  scoped_refptr<MockCapturerSource> new_source(
455      new MockCapturerSource(new_capturer.get()));
456  EXPECT_CALL(*new_source.get(), OnInitialize(_, new_capturer.get(), -1));
457  EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(true));
458  EXPECT_CALL(*new_source.get(), OnStart());
459
460  media::AudioParameters new_param(
461      media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
462      media::CHANNEL_LAYOUT_MONO, 44100, 16, 441);
463  new_capturer->SetCapturerSourceForTesting(new_source, new_param);
464
465  // Setup the second audio track, connect it to the new capturer and start it.
466  scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
467      WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
468  scoped_ptr<WebRtcLocalAudioTrack> track_2(
469      new WebRtcLocalAudioTrack(adapter_2, new_capturer, NULL));
470  track_2->Start();
471
472  // Connect a number of network channels to the |track_2|.
473  static const int kNumberOfNetworkChannelsForTrack2 = 3;
474  for (int i = 0; i < kNumberOfNetworkChannelsForTrack2; ++i) {
475    static_cast<webrtc::AudioTrackInterface*>(
476        adapter_2.get())->GetRenderer()->AddChannel(i);
477  }
478  // Verify the data flow by connecting the |sink_2| to |track_2|.
479  scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink());
480  base::WaitableEvent event(false, false);
481  EXPECT_CALL(*sink_2,
482              CaptureData(kNumberOfNetworkChannelsForTrack2, 0, 0, _, false))
483      .Times(AnyNumber()).WillRepeatedly(Return());
484  EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(SignalEvent(&event));
485  track_2->AddSink(sink_2.get());
486  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
487
488  // Stopping the new source will stop the second track.
489  event.Reset();
490  EXPECT_CALL(*new_source.get(), OnStop())
491      .Times(1).WillOnce(SignalEvent(&event));
492  new_capturer->Stop();
493  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
494
495  // Stop the capturer of the first audio track.
496  EXPECT_CALL(*capturer_source_.get(), OnStop());
497  capturer_->Stop();
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.
502TEST_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, NULL));
519  scoped_refptr<MockCapturerSource> source(
520      new MockCapturerSource(capturer.get()));
521  EXPECT_CALL(*source.get(), OnInitialize(_, capturer.get(), -1));
522  EXPECT_CALL(*source.get(), SetAutomaticGainControl(true));
523  EXPECT_CALL(*source.get(), OnStart());
524  capturer->SetCapturerSourceForTesting(source, params);
525
526  // Setup a audio track, connect it to the capturer and start it.
527  scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
528      WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
529  scoped_ptr<WebRtcLocalAudioTrack> track(
530      new WebRtcLocalAudioTrack(adapter, capturer, NULL));
531  track->Start();
532
533  // Verify the data flow by connecting the |sink| to |track|.
534  scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
535  base::WaitableEvent event(false, false);
536  EXPECT_CALL(*sink, FormatIsSet()).Times(1);
537  // Verify the sinks are getting the packets with an expecting buffer size.
538#if defined(OS_ANDROID)
539  const int expected_buffer_size = params.sample_rate() / 100;
540#else
541  const int expected_buffer_size = params.frames_per_buffer();
542#endif
543  EXPECT_CALL(*sink, CaptureData(
544      0, 0, 0, _, false))
545      .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event));
546  track->AddSink(sink.get());
547  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
548  EXPECT_EQ(expected_buffer_size, sink->audio_params().frames_per_buffer());
549
550  // Stopping the new source will stop the second track.
551  EXPECT_CALL(*source, OnStop()).Times(1);
552  capturer->Stop();
553
554  // Even though this test don't use |capturer_source_| it will be stopped
555  // during teardown of the test harness.
556  EXPECT_CALL(*capturer_source_.get(), OnStop());
557}
558
559}  // namespace content
560