webrtc_local_audio_track_unittest.cc revision 68043e1e95eeb07d5cae7aca370b26518b0867d6 
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/rtc_media_constraints.h"
8#include "content/renderer/media/webrtc_audio_capturer.h"
9#include "content/renderer/media/webrtc_local_audio_source_provider.h"
10#include "content/renderer/media/webrtc_local_audio_track.h"
11#include "media/audio/audio_parameters.h"
12#include "media/base/audio_bus.h"
13#include "media/base/audio_capturer_source.h"
14#include "testing/gmock/include/gmock/gmock.h"
15#include "testing/gtest/include/gtest/gtest.h"
16#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
17
18using ::testing::_;
19using ::testing::AnyNumber;
20using ::testing::AtLeast;
21using ::testing::Return;
22
23namespace content {
24
25namespace {
26
27ACTION_P(SignalEvent, event) {
28  event->Signal();
29}
30
31// A simple thread that we use to fake the audio thread which provides data to
32// the |WebRtcAudioCapturer|.
33class FakeAudioThread : public base::PlatformThread::Delegate {
34 public:
35  explicit FakeAudioThread(const scoped_refptr<WebRtcAudioCapturer>& capturer)
36    : capturer_(capturer),
37      thread_(),
38      closure_(false, false) {
39    DCHECK(capturer.get());
40    audio_bus_ = media::AudioBus::Create(capturer_->audio_parameters());
41  }
42
43  virtual ~FakeAudioThread() { DCHECK(thread_.is_null()); }
44
45  // base::PlatformThread::Delegate:
46  virtual void ThreadMain() OVERRIDE {
47    while (true) {
48      if (closure_.IsSignaled())
49        return;
50
51      media::AudioCapturerSource::CaptureCallback* callback =
52          static_cast<media::AudioCapturerSource::CaptureCallback*>(
53              capturer_.get());
54      audio_bus_->Zero();
55      callback->Capture(audio_bus_.get(), 0, 0, false);
56
57      // Sleep 1ms to yield the resource for the main thread.
58      base::PlatformThread::Sleep(base::TimeDelta::FromMilliseconds(1));
59    }
60  }
61
62  void Start() {
63    base::PlatformThread::CreateWithPriority(
64        0, this, &thread_, base::kThreadPriority_RealtimeAudio);
65    CHECK(!thread_.is_null());
66  }
67
68  void Stop() {
69    closure_.Signal();
70    base::PlatformThread::Join(thread_);
71    thread_ = base::PlatformThreadHandle();
72  }
73
74 private:
75  scoped_ptr<media::AudioBus> audio_bus_;
76  scoped_refptr<WebRtcAudioCapturer> capturer_;
77  base::PlatformThreadHandle thread_;
78  base::WaitableEvent closure_;
79  DISALLOW_COPY_AND_ASSIGN(FakeAudioThread);
80};
81
82class MockCapturerSource : public media::AudioCapturerSource {
83 public:
84  MockCapturerSource() {}
85  MOCK_METHOD3(Initialize, void(const media::AudioParameters& params,
86                                CaptureCallback* callback,
87                                int session_id));
88  MOCK_METHOD0(Start, void());
89  MOCK_METHOD0(Stop, void());
90  MOCK_METHOD1(SetVolume, void(double volume));
91  MOCK_METHOD1(SetAutomaticGainControl, void(bool enable));
92
93 protected:
94  virtual ~MockCapturerSource() {}
95};
96
97class MockWebRtcAudioCapturerSink : public WebRtcAudioCapturerSink {
98 public:
99  MockWebRtcAudioCapturerSink() {}
100  ~MockWebRtcAudioCapturerSink() {}
101  int CaptureData(const std::vector<int>& channels,
102                  const int16* audio_data,
103                  int sample_rate,
104                  int number_of_channels,
105                  int number_of_frames,
106                  int audio_delay_milliseconds,
107                  int current_volume,
108                  bool need_audio_processing,
109                  bool key_pressed) OVERRIDE {
110    CaptureData(channels.size(),
111                sample_rate,
112                number_of_channels,
113                number_of_frames,
114                audio_delay_milliseconds,
115                current_volume,
116                need_audio_processing,
117                key_pressed);
118    return 0;
119  }
120  MOCK_METHOD8(CaptureData,
121               void(int number_of_network_channels,
122                    int sample_rate,
123                    int number_of_channels,
124                    int number_of_frames,
125                    int audio_delay_milliseconds,
126                    int current_volume,
127                    bool need_audio_processing,
128                    bool key_pressed));
129  MOCK_METHOD1(SetCaptureFormat, void(const media::AudioParameters& params));
130};
131
132}  // namespace
133
134class WebRtcLocalAudioTrackTest : public ::testing::Test {
135 protected:
136  virtual void SetUp() OVERRIDE {
137    params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
138                  media::CHANNEL_LAYOUT_STEREO, 2, 0, 48000, 16, 480);
139    capturer_ = WebRtcAudioCapturer::CreateCapturer();
140    WebRtcLocalAudioSourceProvider* source_provider =
141        static_cast<WebRtcLocalAudioSourceProvider*>(
142            capturer_->audio_source_provider());
143    source_provider->SetSinkParamsForTesting(params_);
144    capturer_source_ = new MockCapturerSource();
145    EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), 0))
146        .WillOnce(Return());
147    capturer_->SetCapturerSource(capturer_source_,
148                                 params_.channel_layout(),
149                                 params_.sample_rate());
150
151    EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(false))
152        .WillOnce(Return());
153
154    // Start the audio thread used by the |capturer_source_|.
155    audio_thread_.reset(new FakeAudioThread(capturer_));
156    audio_thread_->Start();
157  }
158
159  virtual void TearDown() {
160    audio_thread_->Stop();
161    audio_thread_.reset();
162  }
163
164  media::AudioParameters params_;
165  scoped_refptr<MockCapturerSource> capturer_source_;
166  scoped_refptr<WebRtcAudioCapturer> capturer_;
167  scoped_ptr<FakeAudioThread> audio_thread_;
168};
169
170// Creates a capturer and audio track, fakes its audio thread, and
171// connect/disconnect the sink to the audio track on the fly, the sink should
172// get data callback when the track is connected to the capturer but not when
173// the track is disconnected from the capturer.
174TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) {
175  EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(Return());
176  RTCMediaConstraints constraints;
177  scoped_refptr<WebRtcLocalAudioTrack> track =
178      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
179          &constraints);
180  track->Start();
181  EXPECT_TRUE(track->enabled());
182
183  // Connect a number of network channels to the audio track.
184  static const int kNumberOfNetworkChannels = 4;
185  for (int i = 0; i < kNumberOfNetworkChannels; ++i) {
186    static_cast<webrtc::AudioTrackInterface*>(track.get())->
187        GetRenderer()->AddChannel(i);
188  }
189  scoped_ptr<MockWebRtcAudioCapturerSink> sink(
190      new MockWebRtcAudioCapturerSink());
191  const media::AudioParameters params = capturer_->audio_parameters();
192  base::WaitableEvent event(false, false);
193  EXPECT_CALL(*sink, SetCaptureFormat(_)).WillOnce(Return());
194  EXPECT_CALL(*sink,
195      CaptureData(kNumberOfNetworkChannels,
196                  params.sample_rate(),
197                  params.channels(),
198                  params.sample_rate() / 100,
199                  0,
200                  0,
201                  false,
202                  false)).Times(AtLeast(1))
203      .WillRepeatedly(SignalEvent(&event));
204  track->AddSink(sink.get());
205
206  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
207  track->RemoveSink(sink.get());
208
209  EXPECT_CALL(*capturer_source_.get(), Stop()).WillOnce(Return());
210  track->Stop();
211  track = NULL;
212}
213
214// The same setup as ConnectAndDisconnectOneSink, but enable and disable the
215// audio track on the fly. When the audio track is disabled, there is no data
216// callback to the sink; when the audio track is enabled, there comes data
217// callback.
218// TODO(xians): Enable this test after resolving the racing issue that TSAN
219// reports on MediaStreamTrack::enabled();
220TEST_F(WebRtcLocalAudioTrackTest, DISABLED_DisableEnableAudioTrack) {
221  EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(Return());
222  RTCMediaConstraints constraints;
223  scoped_refptr<WebRtcLocalAudioTrack> track =
224    WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
225        &constraints);
226  track->Start();
227  static_cast<webrtc::AudioTrackInterface*>(track.get())->
228      GetRenderer()->AddChannel(0);
229  EXPECT_TRUE(track->enabled());
230  EXPECT_TRUE(track->set_enabled(false));
231  scoped_ptr<MockWebRtcAudioCapturerSink> sink(
232      new MockWebRtcAudioCapturerSink());
233  const media::AudioParameters params = capturer_->audio_parameters();
234  base::WaitableEvent event(false, false);
235  EXPECT_CALL(*sink, SetCaptureFormat(_)).WillOnce(Return());
236  EXPECT_CALL(*sink,
237              CaptureData(1,
238                          params.sample_rate(),
239                          params.channels(),
240                          params.sample_rate() / 100,
241                          0,
242                          0,
243                          false,
244                          false)).Times(0);
245  track->AddSink(sink.get());
246  EXPECT_FALSE(event.TimedWait(TestTimeouts::tiny_timeout()));
247
248  event.Reset();
249  EXPECT_CALL(*sink,
250              CaptureData(1,
251                          params.sample_rate(),
252                          params.channels(),
253                          params.sample_rate() / 100,
254                          0,
255                          0,
256                          false,
257                          false)).Times(AtLeast(1))
258      .WillRepeatedly(SignalEvent(&event));
259  EXPECT_TRUE(track->set_enabled(true));
260  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
261  track->RemoveSink(sink.get());
262
263  EXPECT_CALL(*capturer_source_.get(), Stop()).WillOnce(Return());
264  track->Stop();
265  track = NULL;
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  EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(Return());
273  RTCMediaConstraints constraints;
274  scoped_refptr<WebRtcLocalAudioTrack> track_1 =
275    WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
276        &constraints);
277  track_1->Start();
278  static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
279      GetRenderer()->AddChannel(0);
280  EXPECT_TRUE(track_1->enabled());
281  scoped_ptr<MockWebRtcAudioCapturerSink> sink_1(
282      new MockWebRtcAudioCapturerSink());
283  const media::AudioParameters params = capturer_->audio_parameters();
284  base::WaitableEvent event_1(false, false);
285  EXPECT_CALL(*sink_1, SetCaptureFormat(_)).WillOnce(Return());
286  EXPECT_CALL(*sink_1,
287      CaptureData(1,
288                  params.sample_rate(),
289                  params.channels(),
290                  params.sample_rate() / 100,
291                  0,
292                  0,
293                  false,
294                  false)).Times(AtLeast(1))
295      .WillRepeatedly(SignalEvent(&event_1));
296  track_1->AddSink(sink_1.get());
297  EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
298
299  scoped_refptr<WebRtcLocalAudioTrack> track_2 =
300    WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
301        &constraints);
302  track_2->Start();
303  static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
304      GetRenderer()->AddChannel(1);
305  EXPECT_TRUE(track_2->enabled());
306
307  // Verify both |sink_1| and |sink_2| get data.
308  event_1.Reset();
309  base::WaitableEvent event_2(false, false);
310
311  scoped_ptr<MockWebRtcAudioCapturerSink> sink_2(
312        new MockWebRtcAudioCapturerSink());
313  EXPECT_CALL(*sink_2, SetCaptureFormat(_)).WillOnce(Return());
314  EXPECT_CALL(*sink_1,
315      CaptureData(1,
316                  params.sample_rate(),
317                  params.channels(),
318                  params.sample_rate() / 100,
319                  0,
320                  0,
321                  false,
322                  false)).Times(AtLeast(1))
323      .WillRepeatedly(SignalEvent(&event_1));
324  EXPECT_CALL(*sink_2,
325      CaptureData(1,
326                  params.sample_rate(),
327                  params.channels(),
328                  params.sample_rate() / 100,
329                  0,
330                  0,
331                  false,
332                  false)).Times(AtLeast(1))
333      .WillRepeatedly(SignalEvent(&event_2));
334  track_2->AddSink(sink_2.get());
335  EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
336  EXPECT_TRUE(event_2.TimedWait(TestTimeouts::tiny_timeout()));
337
338  track_1->RemoveSink(sink_1.get());
339  track_1->Stop();
340  track_1 = NULL;
341
342  EXPECT_CALL(*capturer_source_.get(), Stop()).WillOnce(Return());
343  track_2->RemoveSink(sink_2.get());
344  track_2->Stop();
345  track_2 = NULL;
346}
347
348
349// Start one track and verify the capturer is correctly starting its source.
350// And it should be fine to not to call Stop() explicitly.
351TEST_F(WebRtcLocalAudioTrackTest, StartOneAudioTrack) {
352  EXPECT_CALL(*capturer_source_.get(), Start()).Times(1);
353  RTCMediaConstraints constraints;
354  scoped_refptr<WebRtcLocalAudioTrack> track =
355      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
356          &constraints);
357  track->Start();
358
359  // When the track goes away, it will automatically stop the
360  // |capturer_source_|.
361  EXPECT_CALL(*capturer_source_.get(), Stop());
362  track->Stop();
363  track = NULL;
364}
365
366// Start/Stop tracks and verify the capturer is correctly starting/stopping
367// its source.
368TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
369  // Starting the first audio track will start the |capturer_source_|.
370  base::WaitableEvent event(false, false);
371  EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(SignalEvent(&event));
372  RTCMediaConstraints constraints;
373  scoped_refptr<WebRtcLocalAudioTrack> track_1 =
374      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
375          &constraints);
376  static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
377      GetRenderer()->AddChannel(0);
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<MockWebRtcAudioCapturerSink> sink(
383      new MockWebRtcAudioCapturerSink());
384  event.Reset();
385  EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false, false))
386      .Times(AnyNumber()).WillRepeatedly(Return());
387  EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(1);
388  track_1->AddSink(sink.get());
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(), Start()).Times(0);
393  scoped_refptr<WebRtcLocalAudioTrack> track_2 =
394      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
395          &constraints);
396  track_2->Start();
397  static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
398      GetRenderer()->AddChannel(1);
399
400  // Stop the first audio track will not stop the |capturer_source_|.
401  EXPECT_CALL(*capturer_source_.get(), Stop()).Times(0);
402  track_1->RemoveSink(sink.get());
403  track_1->Stop();
404  track_1 = NULL;
405
406  EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false, false))
407      .Times(AnyNumber()).WillRepeatedly(Return());
408  EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(1);
409  track_2->AddSink(sink.get());
410
411  // Stop the last audio track will stop the |capturer_source_|.
412  event.Reset();
413  EXPECT_CALL(*capturer_source_.get(), Stop())
414      .Times(1).WillOnce(SignalEvent(&event));
415  track_2->Stop();
416  track_2->RemoveSink(sink.get());
417  track_2 = NULL;
418  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
419}
420
421// Set new source to the existing capturer.
422TEST_F(WebRtcLocalAudioTrackTest, SetNewSourceForCapturerAfterStartTrack) {
423  // Setup the audio track and start the track.
424  EXPECT_CALL(*capturer_source_.get(), Start()).Times(1);
425  RTCMediaConstraints constraints;
426  scoped_refptr<WebRtcLocalAudioTrack> track =
427      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
428          &constraints);
429  track->Start();
430
431  // Setting new source to the capturer and the track should still get packets.
432  scoped_refptr<MockCapturerSource> new_source(new MockCapturerSource());
433  EXPECT_CALL(*capturer_source_.get(), Stop());
434  EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(false));
435  EXPECT_CALL(*new_source.get(), Initialize(_, capturer_.get(), 0))
436      .WillOnce(Return());
437  EXPECT_CALL(*new_source.get(), Start()).WillOnce(Return());
438  capturer_->SetCapturerSource(new_source,
439                               params_.channel_layout(),
440                               params_.sample_rate());
441
442  // Stop the track.
443  EXPECT_CALL(*new_source.get(), Stop());
444  track->Stop();
445  track = NULL;
446}
447
448// Create a new capturer with new source, connect it to a new audio track.
449TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) {
450  // Setup the first audio track and start it.
451  EXPECT_CALL(*capturer_source_.get(), Start()).Times(1);
452  RTCMediaConstraints constraints;
453  scoped_refptr<WebRtcLocalAudioTrack> track_1 =
454      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
455          &constraints);
456  track_1->Start();
457
458  // Connect a number of network channels to the |track_1|.
459  static const int kNumberOfNetworkChannelsForTrack1 = 2;
460  for (int i = 0; i < kNumberOfNetworkChannelsForTrack1; ++i) {
461    static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
462        GetRenderer()->AddChannel(i);
463  }
464  // Verify the data flow by connecting the |sink_1| to |track_1|.
465  scoped_ptr<MockWebRtcAudioCapturerSink> sink_1(
466      new MockWebRtcAudioCapturerSink());
467  EXPECT_CALL(
468      *sink_1.get(),
469      CaptureData(
470          kNumberOfNetworkChannelsForTrack1, 48000, 2, _, 0, 0, false, false))
471      .Times(AnyNumber()).WillRepeatedly(Return());
472  EXPECT_CALL(*sink_1.get(), SetCaptureFormat(_)).Times(1);
473  track_1->AddSink(sink_1.get());
474
475  // Create a new capturer with new source with different audio format.
476  scoped_refptr<WebRtcAudioCapturer> new_capturer(
477      WebRtcAudioCapturer::CreateCapturer());
478  WebRtcLocalAudioSourceProvider* source_provider =
479      static_cast<WebRtcLocalAudioSourceProvider*>(
480          new_capturer->audio_source_provider());
481  source_provider->SetSinkParamsForTesting(params_);
482  scoped_refptr<MockCapturerSource> new_source(new MockCapturerSource());
483  EXPECT_CALL(*new_source.get(), Initialize(_, new_capturer.get(), 0))
484      .WillOnce(Return());
485  EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(false))
486      .WillOnce(Return());
487  new_capturer->SetCapturerSource(new_source,
488                                  media::CHANNEL_LAYOUT_MONO,
489                                  44100);
490
491  // Start the audio thread used by the new source.
492  scoped_ptr<FakeAudioThread> audio_thread(new FakeAudioThread(new_capturer));
493  audio_thread->Start();
494
495  // Setup the second audio track, connect it to the new capturer and start it.
496  EXPECT_CALL(*new_source.get(), Start()).Times(1);
497  scoped_refptr<WebRtcLocalAudioTrack> track_2 =
498      WebRtcLocalAudioTrack::Create(std::string(), new_capturer, NULL, NULL,
499          &constraints);
500  track_2->Start();
501
502  // Connect a number of network channels to the |track_2|.
503  static const int kNumberOfNetworkChannelsForTrack2 = 3;
504  for (int i = 0; i < kNumberOfNetworkChannelsForTrack2; ++i) {
505    static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
506        GetRenderer()->AddChannel(i);
507  }
508  // Verify the data flow by connecting the |sink_2| to |track_2|.
509  scoped_ptr<MockWebRtcAudioCapturerSink> sink_2(
510      new MockWebRtcAudioCapturerSink());
511  EXPECT_CALL(
512      *sink_2,
513      CaptureData(
514          kNumberOfNetworkChannelsForTrack2, 44100, 1, _, 0, 0, false, false))
515      .Times(AnyNumber()).WillRepeatedly(Return());
516  EXPECT_CALL(*sink_2, SetCaptureFormat(_)).Times(1);
517  track_2->AddSink(sink_2.get());
518
519  // Stop the second audio track will stop the new source.
520  base::WaitableEvent event(false, false);
521  EXPECT_CALL(*new_source.get(), Stop()).Times(1).WillOnce(SignalEvent(&event));
522  track_2->Stop();
523  track_2->RemoveSink(sink_2.get());
524  track_2 = NULL;
525  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
526  audio_thread->Stop();
527  audio_thread.reset();
528
529  // Stop the first audio track.
530  EXPECT_CALL(*capturer_source_.get(), Stop());
531  track_1->Stop();
532  track_1 = NULL;
533}
534
535}  // namespace content
536