webrtc_local_audio_track_unittest.cc revision d0247b1b59f9c528cb6df88b4f2b9afaf80d181e 
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.
270TEST_F(WebRtcLocalAudioTrackTest, MultipleAudioTracks) {
271  EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(Return());
272  RTCMediaConstraints constraints;
273  scoped_refptr<WebRtcLocalAudioTrack> track_1 =
274    WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
275        &constraints);
276  track_1->Start();
277  static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
278      GetRenderer()->AddChannel(0);
279  EXPECT_TRUE(track_1->enabled());
280  scoped_ptr<MockWebRtcAudioCapturerSink> sink_1(
281      new MockWebRtcAudioCapturerSink());
282  const media::AudioParameters params = capturer_->audio_parameters();
283  base::WaitableEvent event_1(false, false);
284  EXPECT_CALL(*sink_1, SetCaptureFormat(_)).WillOnce(Return());
285  EXPECT_CALL(*sink_1,
286      CaptureData(1,
287                  params.sample_rate(),
288                  params.channels(),
289                  params.sample_rate() / 100,
290                  0,
291                  0,
292                  false,
293                  false)).Times(AtLeast(1))
294      .WillRepeatedly(SignalEvent(&event_1));
295  track_1->AddSink(sink_1.get());
296  EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
297
298  scoped_refptr<WebRtcLocalAudioTrack> track_2 =
299    WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
300        &constraints);
301  track_2->Start();
302  static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
303      GetRenderer()->AddChannel(1);
304  EXPECT_TRUE(track_2->enabled());
305
306  // Verify both |sink_1| and |sink_2| get data.
307  event_1.Reset();
308  base::WaitableEvent event_2(false, false);
309
310  scoped_ptr<MockWebRtcAudioCapturerSink> sink_2(
311        new MockWebRtcAudioCapturerSink());
312  EXPECT_CALL(*sink_2, SetCaptureFormat(_)).WillOnce(Return());
313  EXPECT_CALL(*sink_1,
314      CaptureData(1,
315                  params.sample_rate(),
316                  params.channels(),
317                  params.sample_rate() / 100,
318                  0,
319                  0,
320                  false,
321                  false)).Times(AtLeast(1))
322      .WillRepeatedly(SignalEvent(&event_1));
323  EXPECT_CALL(*sink_2,
324      CaptureData(1,
325                  params.sample_rate(),
326                  params.channels(),
327                  params.sample_rate() / 100,
328                  0,
329                  0,
330                  false,
331                  false)).Times(AtLeast(1))
332      .WillRepeatedly(SignalEvent(&event_2));
333  track_2->AddSink(sink_2.get());
334  EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
335  EXPECT_TRUE(event_2.TimedWait(TestTimeouts::tiny_timeout()));
336
337  track_1->RemoveSink(sink_1.get());
338  track_1->Stop();
339  track_1 = NULL;
340
341  EXPECT_CALL(*capturer_source_.get(), Stop()).WillOnce(Return());
342  track_2->RemoveSink(sink_2.get());
343  track_2->Stop();
344  track_2 = NULL;
345}
346
347
348// Start one track and verify the capturer is correctly starting its source.
349// And it should be fine to not to call Stop() explicitly.
350TEST_F(WebRtcLocalAudioTrackTest, StartOneAudioTrack) {
351  EXPECT_CALL(*capturer_source_.get(), Start()).Times(1);
352  RTCMediaConstraints constraints;
353  scoped_refptr<WebRtcLocalAudioTrack> track =
354      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
355          &constraints);
356  track->Start();
357
358  // When the track goes away, it will automatically stop the
359  // |capturer_source_|.
360  EXPECT_CALL(*capturer_source_.get(), Stop());
361  track->Stop();
362  track = NULL;
363}
364
365// Start/Stop tracks and verify the capturer is correctly starting/stopping
366// its source.
367TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
368  // Starting the first audio track will start the |capturer_source_|.
369  base::WaitableEvent event(false, false);
370  EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(SignalEvent(&event));
371  RTCMediaConstraints constraints;
372  scoped_refptr<WebRtcLocalAudioTrack> track_1 =
373      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
374          &constraints);
375  static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
376      GetRenderer()->AddChannel(0);
377  track_1->Start();
378  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
379
380  // Verify the data flow by connecting the sink to |track_1|.
381  scoped_ptr<MockWebRtcAudioCapturerSink> sink(
382      new MockWebRtcAudioCapturerSink());
383  event.Reset();
384  EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false, false))
385      .Times(AnyNumber()).WillRepeatedly(Return());
386  EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(1);
387  track_1->AddSink(sink.get());
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(), Start()).Times(0);
392  scoped_refptr<WebRtcLocalAudioTrack> track_2 =
393      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
394          &constraints);
395  track_2->Start();
396  static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
397      GetRenderer()->AddChannel(1);
398
399  // Stop the first audio track will not stop the |capturer_source_|.
400  EXPECT_CALL(*capturer_source_.get(), Stop()).Times(0);
401  track_1->RemoveSink(sink.get());
402  track_1->Stop();
403  track_1 = NULL;
404
405  EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false, false))
406      .Times(AnyNumber()).WillRepeatedly(Return());
407  EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(1);
408  track_2->AddSink(sink.get());
409
410  // Stop the last audio track will stop the |capturer_source_|.
411  event.Reset();
412  EXPECT_CALL(*capturer_source_.get(), Stop())
413      .Times(1).WillOnce(SignalEvent(&event));
414  track_2->Stop();
415  track_2->RemoveSink(sink.get());
416  track_2 = NULL;
417  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
418}
419
420// Set new source to the existing capturer.
421TEST_F(WebRtcLocalAudioTrackTest, SetNewSourceForCapturerAfterStartTrack) {
422  // Setup the audio track and start the track.
423  EXPECT_CALL(*capturer_source_.get(), Start()).Times(1);
424  RTCMediaConstraints constraints;
425  scoped_refptr<WebRtcLocalAudioTrack> track =
426      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
427          &constraints);
428  track->Start();
429
430  // Setting new source to the capturer and the track should still get packets.
431  scoped_refptr<MockCapturerSource> new_source(new MockCapturerSource());
432  EXPECT_CALL(*capturer_source_.get(), Stop());
433  EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(false));
434  EXPECT_CALL(*new_source.get(), Initialize(_, capturer_.get(), 0))
435      .WillOnce(Return());
436  EXPECT_CALL(*new_source.get(), Start()).WillOnce(Return());
437  capturer_->SetCapturerSource(new_source,
438                               params_.channel_layout(),
439                               params_.sample_rate());
440
441  // Stop the track.
442  EXPECT_CALL(*new_source.get(), Stop());
443  track->Stop();
444  track = NULL;
445}
446
447// Create a new capturer with new source, connect it to a new audio track.
448TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) {
449  // Setup the first audio track and start it.
450  EXPECT_CALL(*capturer_source_.get(), Start()).Times(1);
451  RTCMediaConstraints constraints;
452  scoped_refptr<WebRtcLocalAudioTrack> track_1 =
453      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
454          &constraints);
455  track_1->Start();
456
457  // Connect a number of network channels to the |track_1|.
458  static const int kNumberOfNetworkChannelsForTrack1 = 2;
459  for (int i = 0; i < kNumberOfNetworkChannelsForTrack1; ++i) {
460    static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
461        GetRenderer()->AddChannel(i);
462  }
463  // Verify the data flow by connecting the |sink_1| to |track_1|.
464  scoped_ptr<MockWebRtcAudioCapturerSink> sink_1(
465      new MockWebRtcAudioCapturerSink());
466  EXPECT_CALL(
467      *sink_1.get(),
468      CaptureData(
469          kNumberOfNetworkChannelsForTrack1, 48000, 2, _, 0, 0, false, false))
470      .Times(AnyNumber()).WillRepeatedly(Return());
471  EXPECT_CALL(*sink_1.get(), SetCaptureFormat(_)).Times(1);
472  track_1->AddSink(sink_1.get());
473
474  // Create a new capturer with new source with different audio format.
475  scoped_refptr<WebRtcAudioCapturer> new_capturer(
476      WebRtcAudioCapturer::CreateCapturer());
477  WebRtcLocalAudioSourceProvider* source_provider =
478      static_cast<WebRtcLocalAudioSourceProvider*>(
479          new_capturer->audio_source_provider());
480  source_provider->SetSinkParamsForTesting(params_);
481  scoped_refptr<MockCapturerSource> new_source(new MockCapturerSource());
482  EXPECT_CALL(*new_source.get(), Initialize(_, new_capturer.get(), 0))
483      .WillOnce(Return());
484  EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(false))
485      .WillOnce(Return());
486  new_capturer->SetCapturerSource(new_source,
487                                  media::CHANNEL_LAYOUT_MONO,
488                                  44100);
489
490  // Start the audio thread used by the new source.
491  scoped_ptr<FakeAudioThread> audio_thread(new FakeAudioThread(new_capturer));
492  audio_thread->Start();
493
494  // Setup the second audio track, connect it to the new capturer and start it.
495  EXPECT_CALL(*new_source.get(), Start()).Times(1);
496  scoped_refptr<WebRtcLocalAudioTrack> track_2 =
497      WebRtcLocalAudioTrack::Create(std::string(), new_capturer, NULL, NULL,
498          &constraints);
499  track_2->Start();
500
501  // Connect a number of network channels to the |track_2|.
502  static const int kNumberOfNetworkChannelsForTrack2 = 3;
503  for (int i = 0; i < kNumberOfNetworkChannelsForTrack2; ++i) {
504    static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
505        GetRenderer()->AddChannel(i);
506  }
507  // Verify the data flow by connecting the |sink_2| to |track_2|.
508  scoped_ptr<MockWebRtcAudioCapturerSink> sink_2(
509      new MockWebRtcAudioCapturerSink());
510  EXPECT_CALL(
511      *sink_2,
512      CaptureData(
513          kNumberOfNetworkChannelsForTrack2, 44100, 1, _, 0, 0, false, false))
514      .Times(AnyNumber()).WillRepeatedly(Return());
515  EXPECT_CALL(*sink_2, SetCaptureFormat(_)).Times(1);
516  track_2->AddSink(sink_2.get());
517
518  // Stop the second audio track will stop the new source.
519  base::WaitableEvent event(false, false);
520  EXPECT_CALL(*new_source.get(), Stop()).Times(1).WillOnce(SignalEvent(&event));
521  track_2->Stop();
522  track_2->RemoveSink(sink_2.get());
523  track_2 = NULL;
524  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
525  audio_thread->Stop();
526  audio_thread.reset();
527
528  // Stop the first audio track.
529  EXPECT_CALL(*capturer_source_.get(), Stop());
530  track_1->Stop();
531  track_1 = NULL;
532}
533
534}  // namespace content
535