webrtc_local_audio_track_unittest.cc revision 3551c9c881056c480085172ff9840cab31610854 
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/webrtc_audio_capturer.h"
8#include "content/renderer/media/webrtc_local_audio_track.h"
9#include "media/audio/audio_parameters.h"
10#include "media/base/audio_bus.h"
11#include "media/base/audio_capturer_source.h"
12#include "testing/gmock/include/gmock/gmock.h"
13#include "testing/gtest/include/gtest/gtest.h"
14#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
15
16using ::testing::_;
17using ::testing::AnyNumber;
18using ::testing::AtLeast;
19using ::testing::Return;
20
21namespace content {
22
23namespace {
24
25ACTION_P(SignalEvent, event) {
26  event->Signal();
27}
28
29// A simple thread that we use to fake the audio thread which provides data to
30// the |WebRtcAudioCapturer|.
31class FakeAudioThread : public base::PlatformThread::Delegate {
32 public:
33  explicit FakeAudioThread(const scoped_refptr<WebRtcAudioCapturer>& capturer)
34    : capturer_(capturer),
35      thread_(),
36      closure_(false, false) {
37    DCHECK(capturer.get());
38    audio_bus_ = media::AudioBus::Create(capturer_->audio_parameters());
39  }
40
41  virtual ~FakeAudioThread() { DCHECK(thread_.is_null()); }
42
43  // base::PlatformThread::Delegate:
44  virtual void ThreadMain() OVERRIDE {
45    while (true) {
46      if (closure_.IsSignaled())
47        return;
48
49      media::AudioCapturerSource::CaptureCallback* callback =
50          static_cast<media::AudioCapturerSource::CaptureCallback*>(
51              capturer_.get());
52      audio_bus_->Zero();
53      callback->Capture(audio_bus_.get(), 0, 0, false);
54
55      // Sleep 1ms to yield the resource for the main thread.
56      base::PlatformThread::Sleep(base::TimeDelta::FromMilliseconds(1));
57    }
58  }
59
60  void Start() {
61    base::PlatformThread::CreateWithPriority(
62        0, this, &thread_, base::kThreadPriority_RealtimeAudio);
63    CHECK(!thread_.is_null());
64  }
65
66  void Stop() {
67    closure_.Signal();
68    base::PlatformThread::Join(thread_);
69    thread_ = base::PlatformThreadHandle();
70  }
71
72 private:
73  scoped_ptr<media::AudioBus> audio_bus_;
74  scoped_refptr<WebRtcAudioCapturer> capturer_;
75  base::PlatformThreadHandle thread_;
76  base::WaitableEvent closure_;
77  DISALLOW_COPY_AND_ASSIGN(FakeAudioThread);
78};
79
80class MockCapturerSource : public media::AudioCapturerSource {
81 public:
82  MockCapturerSource() {}
83  MOCK_METHOD3(Initialize, void(const media::AudioParameters& params,
84                                CaptureCallback* callback,
85                                int session_id));
86  MOCK_METHOD0(Start, void());
87  MOCK_METHOD0(Stop, void());
88  MOCK_METHOD1(SetVolume, void(double volume));
89  MOCK_METHOD1(SetAutomaticGainControl, void(bool enable));
90
91 protected:
92  virtual ~MockCapturerSource() {}
93};
94
95class MockWebRtcAudioCapturerSink : public WebRtcAudioCapturerSink {
96 public:
97  MockWebRtcAudioCapturerSink() {}
98  ~MockWebRtcAudioCapturerSink() {}
99  int CaptureData(const std::vector<int>& channels,
100                  const int16* audio_data,
101                  int sample_rate,
102                  int number_of_channels,
103                  int number_of_frames,
104                  int audio_delay_milliseconds,
105                  int current_volume,
106                  bool need_audio_processing,
107                  bool key_pressed) OVERRIDE {
108    CaptureData(channels.size(),
109                sample_rate,
110                number_of_channels,
111                number_of_frames,
112                audio_delay_milliseconds,
113                current_volume,
114                need_audio_processing,
115                key_pressed);
116    return 0;
117  }
118  MOCK_METHOD8(CaptureData,
119               void(int number_of_network_channels,
120                    int sample_rate,
121                    int number_of_channels,
122                    int number_of_frames,
123                    int audio_delay_milliseconds,
124                    int current_volume,
125                    bool need_audio_processing,
126                    bool key_pressed));
127  MOCK_METHOD1(SetCaptureFormat, void(const media::AudioParameters& params));
128};
129
130}  // namespace
131
132class WebRtcLocalAudioTrackTest : public ::testing::Test {
133 protected:
134  virtual void SetUp() OVERRIDE {
135    capturer_ = WebRtcAudioCapturer::CreateCapturer();
136    capturer_source_ = new MockCapturerSource();
137    EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), 0))
138        .WillOnce(Return());
139    capturer_->SetCapturerSource(capturer_source_,
140                                 media::CHANNEL_LAYOUT_STEREO,
141                                 48000);
142
143    EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(false))
144        .WillOnce(Return());
145
146    // Start the audio thread used by the |capturer_source_|.
147    audio_thread_.reset(new FakeAudioThread(capturer_));
148    audio_thread_->Start();
149  }
150
151  virtual void TearDown() {
152    audio_thread_->Stop();
153    audio_thread_.reset();
154  }
155
156  scoped_refptr<MockCapturerSource> capturer_source_;
157  scoped_refptr<WebRtcAudioCapturer> capturer_;
158  scoped_ptr<FakeAudioThread> audio_thread_;
159};
160
161// Creates a capturer and audio track, fakes its audio thread, and
162// connect/disconnect the sink to the audio track on the fly, the sink should
163// get data callback when the track is connected to the capturer but not when
164// the track is disconnected from the capturer.
165TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) {
166  EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(Return());
167  scoped_refptr<WebRtcLocalAudioTrack> track =
168      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
169  track->Start();
170  EXPECT_TRUE(track->enabled());
171
172  // Connect a number of network channels to the audio track.
173  static const int kNumberOfNetworkChannels = 4;
174  for (int i = 0; i < kNumberOfNetworkChannels; ++i) {
175    static_cast<webrtc::AudioTrackInterface*>(track.get())->
176        GetRenderer()->AddChannel(i);
177  }
178  scoped_ptr<MockWebRtcAudioCapturerSink> sink(
179      new MockWebRtcAudioCapturerSink());
180  const media::AudioParameters params = capturer_->audio_parameters();
181  base::WaitableEvent event(false, false);
182  EXPECT_CALL(*sink, SetCaptureFormat(_)).WillOnce(Return());
183  EXPECT_CALL(*sink,
184              CaptureData(kNumberOfNetworkChannels,
185                          params.sample_rate(),
186                          params.channels(),
187                          params.frames_per_buffer(),
188                          0,
189                          0,
190                          false,
191                          false)).Times(AtLeast(1))
192      .WillRepeatedly(SignalEvent(&event));
193  track->AddSink(sink.get());
194
195  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
196  track->RemoveSink(sink.get());
197
198  EXPECT_CALL(*capturer_source_.get(), Stop()).WillOnce(Return());
199  track->Stop();
200  track = NULL;
201}
202
203// The same setup as ConnectAndDisconnectOneSink, but enable and disable the
204// audio track on the fly. When the audio track is disabled, there is no data
205// callback to the sink; when the audio track is enabled, there comes data
206// callback.
207// TODO(xians): Enable this test after resolving the racing issue that TSAN
208// reports on MediaStreamTrack::enabled();
209TEST_F(WebRtcLocalAudioTrackTest, DISABLED_DisableEnableAudioTrack) {
210  EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(Return());
211  scoped_refptr<WebRtcLocalAudioTrack> track =
212    WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
213  track->Start();
214  static_cast<webrtc::AudioTrackInterface*>(track.get())->
215      GetRenderer()->AddChannel(0);
216  EXPECT_TRUE(track->enabled());
217  EXPECT_TRUE(track->set_enabled(false));
218  scoped_ptr<MockWebRtcAudioCapturerSink> sink(
219      new MockWebRtcAudioCapturerSink());
220  const media::AudioParameters params = capturer_->audio_parameters();
221  base::WaitableEvent event(false, false);
222  EXPECT_CALL(*sink, SetCaptureFormat(_)).WillOnce(Return());
223  EXPECT_CALL(*sink,
224              CaptureData(1,
225                          params.sample_rate(),
226                          params.channels(),
227                          params.frames_per_buffer(),
228                          0,
229                          0,
230                          false,
231                          false)).Times(0);
232  track->AddSink(sink.get());
233  EXPECT_FALSE(event.TimedWait(TestTimeouts::tiny_timeout()));
234
235  event.Reset();
236  EXPECT_CALL(*sink,
237              CaptureData(1,
238                          params.sample_rate(),
239                          params.channels(),
240                          params.frames_per_buffer(),
241                          0,
242                          0,
243                          false,
244                          false)).Times(AtLeast(1))
245      .WillRepeatedly(SignalEvent(&event));
246  EXPECT_TRUE(track->set_enabled(true));
247  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
248  track->RemoveSink(sink.get());
249
250  EXPECT_CALL(*capturer_source_.get(), Stop()).WillOnce(Return());
251  track->Stop();
252  track = NULL;
253}
254
255// Create multiple audio tracks and enable/disable them, verify that the audio
256// callbacks appear/disappear.
257TEST_F(WebRtcLocalAudioTrackTest, MultipleAudioTracks) {
258  EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(Return());
259  scoped_refptr<WebRtcLocalAudioTrack> track_1 =
260    WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
261  track_1->Start();
262  static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
263      GetRenderer()->AddChannel(0);
264  EXPECT_TRUE(track_1->enabled());
265  scoped_ptr<MockWebRtcAudioCapturerSink> sink_1(
266      new MockWebRtcAudioCapturerSink());
267  const media::AudioParameters params = capturer_->audio_parameters();
268  base::WaitableEvent event_1(false, false);
269  EXPECT_CALL(*sink_1, SetCaptureFormat(_)).WillOnce(Return());
270  EXPECT_CALL(*sink_1,
271              CaptureData(1,
272                          params.sample_rate(),
273                          params.channels(),
274                          params.frames_per_buffer(),
275                          0,
276                          0,
277                          false,
278                          false)).Times(AtLeast(1))
279      .WillRepeatedly(SignalEvent(&event_1));
280  track_1->AddSink(sink_1.get());
281  EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
282
283  scoped_refptr<WebRtcLocalAudioTrack> track_2 =
284    WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
285  track_2->Start();
286  static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
287      GetRenderer()->AddChannel(1);
288  EXPECT_TRUE(track_2->enabled());
289
290  // Verify both |sink_1| and |sink_2| get data.
291  event_1.Reset();
292  base::WaitableEvent event_2(false, false);
293
294  scoped_ptr<MockWebRtcAudioCapturerSink> sink_2(
295        new MockWebRtcAudioCapturerSink());
296  EXPECT_CALL(*sink_2, SetCaptureFormat(_)).WillOnce(Return());
297  EXPECT_CALL(*sink_1,
298              CaptureData(1,
299                          params.sample_rate(),
300                          params.channels(),
301                          params.frames_per_buffer(),
302                          0,
303                          0,
304                          false,
305                          false)).Times(AtLeast(1))
306      .WillRepeatedly(SignalEvent(&event_1));
307  EXPECT_CALL(*sink_2,
308              CaptureData(1,
309                          params.sample_rate(),
310                          params.channels(),
311                          params.frames_per_buffer(),
312                          0,
313                          0,
314                          false,
315                          false)).Times(AtLeast(1))
316      .WillRepeatedly(SignalEvent(&event_2));
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 = NULL;
324
325  EXPECT_CALL(*capturer_source_.get(), Stop()).WillOnce(Return());
326  track_2->RemoveSink(sink_2.get());
327  track_2->Stop();
328  track_2 = NULL;
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.
334TEST_F(WebRtcLocalAudioTrackTest, StartOneAudioTrack) {
335  EXPECT_CALL(*capturer_source_.get(), Start()).Times(1);
336  scoped_refptr<WebRtcLocalAudioTrack> track =
337      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
338  track->Start();
339
340  // When the track goes away, it will automatically stop the
341  // |capturer_source_|.
342  EXPECT_CALL(*capturer_source_.get(), Stop());
343  track->Stop();
344  track = NULL;
345}
346
347// Start/Stop tracks and verify the capturer is correctly starting/stopping
348// its source.
349TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
350  // Starting the first audio track will start the |capturer_source_|.
351  base::WaitableEvent event(false, false);
352  EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(SignalEvent(&event));
353  scoped_refptr<WebRtcLocalAudioTrack> track_1 =
354      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
355  static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
356      GetRenderer()->AddChannel(0);
357  track_1->Start();
358  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
359
360  // Verify the data flow by connecting the sink to |track_1|.
361  scoped_ptr<MockWebRtcAudioCapturerSink> sink(
362      new MockWebRtcAudioCapturerSink());
363  event.Reset();
364  EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false, false))
365      .Times(AnyNumber()).WillRepeatedly(Return());
366  EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(1);
367  track_1->AddSink(sink.get());
368
369  // Start the second audio track will not start the |capturer_source_|
370  // since it has been started.
371  EXPECT_CALL(*capturer_source_.get(), Start()).Times(0);
372  scoped_refptr<WebRtcLocalAudioTrack> track_2 =
373      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
374  track_2->Start();
375  static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
376      GetRenderer()->AddChannel(1);
377
378  // Stop the first audio track will not stop the |capturer_source_|.
379  EXPECT_CALL(*capturer_source_.get(), Stop()).Times(0);
380  track_1->RemoveSink(sink.get());
381  track_1->Stop();
382  track_1 = NULL;
383
384  EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false, false))
385      .Times(AnyNumber()).WillRepeatedly(Return());
386  EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(1);
387  track_2->AddSink(sink.get());
388
389  // Stop the last audio track will stop the |capturer_source_|.
390  event.Reset();
391  EXPECT_CALL(*capturer_source_.get(), Stop())
392      .Times(1).WillOnce(SignalEvent(&event));
393  track_2->Stop();
394  track_2->RemoveSink(sink.get());
395  track_2 = NULL;
396  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
397}
398
399// Set new source to the existing capturer.
400TEST_F(WebRtcLocalAudioTrackTest, SetNewSourceForCapturerAfterStartTrack) {
401  // Setup the audio track and start the track.
402  EXPECT_CALL(*capturer_source_.get(), Start()).Times(1);
403  scoped_refptr<WebRtcLocalAudioTrack> track =
404      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
405  track->Start();
406
407  // Setting new source to the capturer and the track should still get packets.
408  scoped_refptr<MockCapturerSource> new_source(new MockCapturerSource());
409  EXPECT_CALL(*capturer_source_.get(), Stop());
410  EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(false));
411  EXPECT_CALL(*new_source.get(), Initialize(_, capturer_.get(), 0))
412      .WillOnce(Return());
413  EXPECT_CALL(*new_source.get(), Start()).WillOnce(Return());
414  capturer_->SetCapturerSource(new_source,
415                               media::CHANNEL_LAYOUT_STEREO,
416                               48000);
417
418  // Stop the track.
419  EXPECT_CALL(*new_source.get(), Stop());
420  track->Stop();
421  track = NULL;
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  EXPECT_CALL(*capturer_source_.get(), Start()).Times(1);
428  scoped_refptr<WebRtcLocalAudioTrack> track_1 =
429      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL);
430  track_1->Start();
431
432  // Connect a number of network channels to the |track_1|.
433  static const int kNumberOfNetworkChannelsForTrack1 = 2;
434  for (int i = 0; i < kNumberOfNetworkChannelsForTrack1; ++i) {
435    static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
436        GetRenderer()->AddChannel(i);
437  }
438  // Verify the data flow by connecting the |sink_1| to |track_1|.
439  scoped_ptr<MockWebRtcAudioCapturerSink> sink_1(
440      new MockWebRtcAudioCapturerSink());
441  EXPECT_CALL(
442      *sink_1.get(),
443      CaptureData(
444          kNumberOfNetworkChannelsForTrack1, 48000, 2, _, 0, 0, false, false))
445      .Times(AnyNumber()).WillRepeatedly(Return());
446  EXPECT_CALL(*sink_1.get(), SetCaptureFormat(_)).Times(1);
447  track_1->AddSink(sink_1.get());
448
449  // Create a new capturer with new source with different audio format.
450  scoped_refptr<WebRtcAudioCapturer> new_capturer(
451      WebRtcAudioCapturer::CreateCapturer());
452  scoped_refptr<MockCapturerSource> new_source(new MockCapturerSource());
453  EXPECT_CALL(*new_source.get(), Initialize(_, new_capturer.get(), 0))
454      .WillOnce(Return());
455  EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(false))
456      .WillOnce(Return());
457  new_capturer->SetCapturerSource(new_source,
458                                  media::CHANNEL_LAYOUT_MONO,
459                                  44100);
460
461  // Start the audio thread used by the new source.
462  scoped_ptr<FakeAudioThread> audio_thread(new FakeAudioThread(new_capturer));
463  audio_thread->Start();
464
465  // Setup the second audio track, connect it to the new capturer and start it.
466  EXPECT_CALL(*new_source.get(), Start()).Times(1);
467  scoped_refptr<WebRtcLocalAudioTrack> track_2 =
468      WebRtcLocalAudioTrack::Create(std::string(), new_capturer, NULL);
469  track_2->Start();
470
471  // Connect a number of network channels to the |track_2|.
472  static const int kNumberOfNetworkChannelsForTrack2 = 3;
473  for (int i = 0; i < kNumberOfNetworkChannelsForTrack2; ++i) {
474    static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
475        GetRenderer()->AddChannel(i);
476  }
477  // Verify the data flow by connecting the |sink_2| to |track_2|.
478  scoped_ptr<MockWebRtcAudioCapturerSink> sink_2(
479      new MockWebRtcAudioCapturerSink());
480  EXPECT_CALL(
481      *sink_2,
482      CaptureData(
483          kNumberOfNetworkChannelsForTrack2, 44100, 1, _, 0, 0, false, false))
484      .Times(AnyNumber()).WillRepeatedly(Return());
485  EXPECT_CALL(*sink_2, SetCaptureFormat(_)).Times(1);
486  track_2->AddSink(sink_2.get());
487
488  // Stop the second audio track will stop the new source.
489  base::WaitableEvent event(false, false);
490  EXPECT_CALL(*new_source.get(), Stop()).Times(1).WillOnce(SignalEvent(&event));
491  track_2->Stop();
492  track_2->RemoveSink(sink_2.get());
493  track_2 = NULL;
494  EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
495  audio_thread->Stop();
496  audio_thread.reset();
497
498  // Stop the first audio track.
499  EXPECT_CALL(*capturer_source_.get(), Stop());
500  track_1->Stop();
501  track_1 = NULL;
502}
503
504}  // namespace content
505