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    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.
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  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();
235TEST_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
271TEST_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.
328TEST_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.
344TEST_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.
366TEST_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// Create a new capturer with new source, connect it to a new audio track.
408TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) {
409  // Setup the first audio track and start it.
410  scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
411      WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
412  scoped_ptr<WebRtcLocalAudioTrack> track_1(
413      new WebRtcLocalAudioTrack(adapter_1, capturer_, NULL));
414  track_1->Start();
415
416  // Verify the data flow by connecting the |sink_1| to |track_1|.
417  scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink());
418  EXPECT_CALL(*sink_1.get(), CaptureData(0, 0, 0, _, false))
419      .Times(AnyNumber()).WillRepeatedly(Return());
420  EXPECT_CALL(*sink_1.get(), FormatIsSet()).Times(AnyNumber());
421  track_1->AddSink(sink_1.get());
422
423  // Create a new capturer with new source with different audio format.
424  MockMediaConstraintFactory constraint_factory;
425  StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
426                          std::string(), std::string());
427  scoped_refptr<WebRtcAudioCapturer> new_capturer(
428      WebRtcAudioCapturer::CreateCapturer(
429          -1, device, constraint_factory.CreateWebMediaConstraints(), NULL,
430          NULL));
431  scoped_refptr<MockCapturerSource> new_source(
432      new MockCapturerSource(new_capturer.get()));
433  EXPECT_CALL(*new_source.get(), OnInitialize(_, new_capturer.get(), -1));
434  EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(true));
435  EXPECT_CALL(*new_source.get(), OnStart());
436
437  media::AudioParameters new_param(
438      media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
439      media::CHANNEL_LAYOUT_MONO, 44100, 16, 441);
440  new_capturer->SetCapturerSourceForTesting(new_source, new_param);
441
442  // Setup the second audio track, connect it to the new capturer and start it.
443  scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
444      WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
445  scoped_ptr<WebRtcLocalAudioTrack> track_2(
446      new WebRtcLocalAudioTrack(adapter_2, new_capturer, NULL));
447  track_2->Start();
448
449  // Verify the data flow by connecting the |sink_2| to |track_2|.
450  scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink());
451  base::WaitableEvent event(false, false);
452  EXPECT_CALL(*sink_2, CaptureData(0, 0, 0, _, false))
453      .Times(AnyNumber()).WillRepeatedly(Return());
454  EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(SignalEvent(&event));
455  track_2->AddSink(sink_2.get());
456  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
457
458  // Stopping the new source will stop the second track.
459  event.Reset();
460  EXPECT_CALL(*new_source.get(), OnStop())
461      .Times(1).WillOnce(SignalEvent(&event));
462  new_capturer->Stop();
463  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
464
465  // Stop the capturer of the first audio track.
466  EXPECT_CALL(*capturer_source_.get(), OnStop());
467  capturer_->Stop();
468}
469
470// Make sure a audio track can deliver packets with a buffer size smaller than
471// 10ms when it is not connected with a peer connection.
472TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) {
473  // Setup a capturer which works with a buffer size smaller than 10ms.
474  media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
475                                media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128);
476
477  // Create a capturer with new source which works with the format above.
478  MockMediaConstraintFactory factory;
479  factory.DisableDefaultAudioConstraints();
480  scoped_refptr<WebRtcAudioCapturer> capturer(
481      WebRtcAudioCapturer::CreateCapturer(
482          -1,
483          StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE,
484                           "", "", params.sample_rate(),
485                           params.channel_layout(),
486                           params.frames_per_buffer()),
487          factory.CreateWebMediaConstraints(),
488          NULL, NULL));
489  scoped_refptr<MockCapturerSource> source(
490      new MockCapturerSource(capturer.get()));
491  EXPECT_CALL(*source.get(), OnInitialize(_, capturer.get(), -1));
492  EXPECT_CALL(*source.get(), SetAutomaticGainControl(true));
493  EXPECT_CALL(*source.get(), OnStart());
494  capturer->SetCapturerSourceForTesting(source, params);
495
496  // Setup a audio track, connect it to the capturer and start it.
497  scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
498      WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
499  scoped_ptr<WebRtcLocalAudioTrack> track(
500      new WebRtcLocalAudioTrack(adapter, capturer, NULL));
501  track->Start();
502
503  // Verify the data flow by connecting the |sink| to |track|.
504  scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
505  base::WaitableEvent event(false, false);
506  EXPECT_CALL(*sink, FormatIsSet()).Times(1);
507  // Verify the sinks are getting the packets with an expecting buffer size.
508#if defined(OS_ANDROID)
509  const int expected_buffer_size = params.sample_rate() / 100;
510#else
511  const int expected_buffer_size = params.frames_per_buffer();
512#endif
513  EXPECT_CALL(*sink, CaptureData(
514      0, 0, 0, _, false))
515      .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event));
516  track->AddSink(sink.get());
517  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
518  EXPECT_EQ(expected_buffer_size, sink->audio_params().frames_per_buffer());
519
520  // Stopping the new source will stop the second track.
521  EXPECT_CALL(*source, OnStop()).Times(1);
522  capturer->Stop();
523
524  // Even though this test don't use |capturer_source_| it will be stopped
525  // during teardown of the test harness.
526  EXPECT_CALL(*capturer_source_.get(), OnStop());
527}
528
529}  // namespace content
530