OLD | NEW |
| (Empty) |
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/macros.h" | |
6 #include "base/synchronization/waitable_event.h" | |
7 #include "base/test/test_timeouts.h" | |
8 #include "build/build_config.h" | |
9 #include "content/public/renderer/media_stream_audio_sink.h" | |
10 #include "content/renderer/media/media_stream_audio_source.h" | |
11 #include "content/renderer/media/mock_media_constraint_factory.h" | |
12 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" | |
13 #include "content/renderer/media/webrtc_audio_capturer.h" | |
14 #include "content/renderer/media/webrtc_local_audio_track.h" | |
15 #include "media/audio/audio_parameters.h" | |
16 #include "media/base/audio_bus.h" | |
17 #include "media/base/audio_capturer_source.h" | |
18 #include "testing/gmock/include/gmock/gmock.h" | |
19 #include "testing/gtest/include/gtest/gtest.h" | |
20 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" | |
21 #include "third_party/WebKit/public/web/WebHeap.h" | |
22 #include "third_party/webrtc/api/mediastreaminterface.h" | |
23 | |
24 using ::testing::_; | |
25 using ::testing::AnyNumber; | |
26 using ::testing::AtLeast; | |
27 using ::testing::Return; | |
28 | |
29 namespace content { | |
30 | |
31 namespace { | |
32 | |
33 ACTION_P(SignalEvent, event) { | |
34 event->Signal(); | |
35 } | |
36 | |
37 // A simple thread that we use to fake the audio thread which provides data to | |
38 // the |WebRtcAudioCapturer|. | |
39 class FakeAudioThread : public base::PlatformThread::Delegate { | |
40 public: | |
41 FakeAudioThread(WebRtcAudioCapturer* capturer, | |
42 const media::AudioParameters& params) | |
43 : capturer_(capturer), | |
44 thread_(), | |
45 closure_(false, false) { | |
46 DCHECK(capturer); | |
47 audio_bus_ = media::AudioBus::Create(params); | |
48 } | |
49 | |
50 ~FakeAudioThread() override { DCHECK(thread_.is_null()); } | |
51 | |
52 // base::PlatformThread::Delegate: | |
53 void ThreadMain() override { | |
54 while (true) { | |
55 if (closure_.IsSignaled()) | |
56 return; | |
57 | |
58 media::AudioCapturerSource::CaptureCallback* callback = | |
59 static_cast<media::AudioCapturerSource::CaptureCallback*>( | |
60 capturer_); | |
61 audio_bus_->Zero(); | |
62 callback->Capture(audio_bus_.get(), 0, 0, false); | |
63 | |
64 // Sleep 1ms to yield the resource for the main thread. | |
65 base::PlatformThread::Sleep(base::TimeDelta::FromMilliseconds(1)); | |
66 } | |
67 } | |
68 | |
69 void Start() { | |
70 base::PlatformThread::CreateWithPriority( | |
71 0, this, &thread_, base::ThreadPriority::REALTIME_AUDIO); | |
72 CHECK(!thread_.is_null()); | |
73 } | |
74 | |
75 void Stop() { | |
76 closure_.Signal(); | |
77 base::PlatformThread::Join(thread_); | |
78 thread_ = base::PlatformThreadHandle(); | |
79 } | |
80 | |
81 private: | |
82 scoped_ptr<media::AudioBus> audio_bus_; | |
83 WebRtcAudioCapturer* capturer_; | |
84 base::PlatformThreadHandle thread_; | |
85 base::WaitableEvent closure_; | |
86 DISALLOW_COPY_AND_ASSIGN(FakeAudioThread); | |
87 }; | |
88 | |
89 class MockCapturerSource : public media::AudioCapturerSource { | |
90 public: | |
91 explicit MockCapturerSource(WebRtcAudioCapturer* capturer) | |
92 : capturer_(capturer) {} | |
93 MOCK_METHOD3(OnInitialize, void(const media::AudioParameters& params, | |
94 CaptureCallback* callback, | |
95 int session_id)); | |
96 MOCK_METHOD0(OnStart, void()); | |
97 MOCK_METHOD0(OnStop, void()); | |
98 MOCK_METHOD1(SetVolume, void(double volume)); | |
99 MOCK_METHOD1(SetAutomaticGainControl, void(bool enable)); | |
100 | |
101 void Initialize(const media::AudioParameters& params, | |
102 CaptureCallback* callback, | |
103 int session_id) override { | |
104 DCHECK(params.IsValid()); | |
105 params_ = params; | |
106 OnInitialize(params, callback, session_id); | |
107 } | |
108 void Start() override { | |
109 audio_thread_.reset(new FakeAudioThread(capturer_, params_)); | |
110 audio_thread_->Start(); | |
111 OnStart(); | |
112 } | |
113 void Stop() override { | |
114 audio_thread_->Stop(); | |
115 audio_thread_.reset(); | |
116 OnStop(); | |
117 } | |
118 | |
119 protected: | |
120 ~MockCapturerSource() override {} | |
121 | |
122 private: | |
123 scoped_ptr<FakeAudioThread> audio_thread_; | |
124 WebRtcAudioCapturer* capturer_; | |
125 media::AudioParameters params_; | |
126 }; | |
127 | |
128 class MockMediaStreamAudioSink : public MediaStreamAudioSink { | |
129 public: | |
130 MockMediaStreamAudioSink() {} | |
131 ~MockMediaStreamAudioSink() {} | |
132 void OnData(const media::AudioBus& audio_bus, | |
133 base::TimeTicks estimated_capture_time) override { | |
134 EXPECT_EQ(params_.channels(), audio_bus.channels()); | |
135 EXPECT_EQ(params_.frames_per_buffer(), audio_bus.frames()); | |
136 EXPECT_FALSE(estimated_capture_time.is_null()); | |
137 CaptureData(); | |
138 } | |
139 MOCK_METHOD0(CaptureData, void()); | |
140 void OnSetFormat(const media::AudioParameters& params) { | |
141 params_ = params; | |
142 FormatIsSet(); | |
143 } | |
144 MOCK_METHOD0(FormatIsSet, void()); | |
145 | |
146 const media::AudioParameters& audio_params() const { return params_; } | |
147 | |
148 private: | |
149 media::AudioParameters params_; | |
150 }; | |
151 | |
152 } // namespace | |
153 | |
154 class WebRtcLocalAudioTrackTest : public ::testing::Test { | |
155 protected: | |
156 void SetUp() override { | |
157 params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, | |
158 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 480); | |
159 MockMediaConstraintFactory constraint_factory; | |
160 blink_source_.initialize("dummy", blink::WebMediaStreamSource::TypeAudio, | |
161 "dummy", | |
162 false /* remote */, true /* readonly */); | |
163 MediaStreamAudioSource* audio_source = new MediaStreamAudioSource(); | |
164 blink_source_.setExtraData(audio_source); | |
165 | |
166 StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE, | |
167 std::string(), std::string()); | |
168 capturer_ = WebRtcAudioCapturer::CreateCapturer( | |
169 -1, device, constraint_factory.CreateWebMediaConstraints(), NULL, | |
170 audio_source); | |
171 audio_source->SetAudioCapturer(capturer_.get()); | |
172 capturer_source_ = new MockCapturerSource(capturer_.get()); | |
173 EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_.get(), -1)) | |
174 .WillOnce(Return()); | |
175 EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); | |
176 EXPECT_CALL(*capturer_source_.get(), OnStart()); | |
177 capturer_->SetCapturerSource(capturer_source_, params_); | |
178 } | |
179 | |
180 void TearDown() override { | |
181 blink_source_.reset(); | |
182 blink::WebHeap::collectAllGarbageForTesting(); | |
183 } | |
184 | |
185 media::AudioParameters params_; | |
186 blink::WebMediaStreamSource blink_source_; | |
187 scoped_refptr<MockCapturerSource> capturer_source_; | |
188 scoped_refptr<WebRtcAudioCapturer> capturer_; | |
189 }; | |
190 | |
191 // Creates a capturer and audio track, fakes its audio thread, and | |
192 // connect/disconnect the sink to the audio track on the fly, the sink should | |
193 // get data callback when the track is connected to the capturer but not when | |
194 // the track is disconnected from the capturer. | |
195 TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) { | |
196 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( | |
197 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
198 scoped_ptr<WebRtcLocalAudioTrack> track( | |
199 new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL)); | |
200 track->Start(); | |
201 EXPECT_TRUE(track->GetAudioAdapter()->enabled()); | |
202 | |
203 scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink()); | |
204 base::WaitableEvent event(false, false); | |
205 EXPECT_CALL(*sink, FormatIsSet()); | |
206 EXPECT_CALL(*sink, | |
207 CaptureData()).Times(AtLeast(1)) | |
208 .WillRepeatedly(SignalEvent(&event)); | |
209 track->AddSink(sink.get()); | |
210 EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); | |
211 track->RemoveSink(sink.get()); | |
212 | |
213 EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return()); | |
214 capturer_->Stop(); | |
215 } | |
216 | |
217 // The same setup as ConnectAndDisconnectOneSink, but enable and disable the | |
218 // audio track on the fly. When the audio track is disabled, there is no data | |
219 // callback to the sink; when the audio track is enabled, there comes data | |
220 // callback. | |
221 // TODO(xians): Enable this test after resolving the racing issue that TSAN | |
222 // reports on MediaStreamTrack::enabled(); | |
223 TEST_F(WebRtcLocalAudioTrackTest, DISABLED_DisableEnableAudioTrack) { | |
224 EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); | |
225 EXPECT_CALL(*capturer_source_.get(), OnStart()); | |
226 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( | |
227 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
228 scoped_ptr<WebRtcLocalAudioTrack> track( | |
229 new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL)); | |
230 track->Start(); | |
231 EXPECT_TRUE(track->GetAudioAdapter()->enabled()); | |
232 EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(false)); | |
233 scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink()); | |
234 const media::AudioParameters params = capturer_->source_audio_parameters(); | |
235 base::WaitableEvent event(false, false); | |
236 EXPECT_CALL(*sink, FormatIsSet()).Times(1); | |
237 EXPECT_CALL(*sink, CaptureData()).Times(0); | |
238 EXPECT_EQ(sink->audio_params().frames_per_buffer(), | |
239 params.sample_rate() / 100); | |
240 track->AddSink(sink.get()); | |
241 EXPECT_FALSE(event.TimedWait(TestTimeouts::tiny_timeout())); | |
242 | |
243 event.Reset(); | |
244 EXPECT_CALL(*sink, CaptureData()).Times(AtLeast(1)) | |
245 .WillRepeatedly(SignalEvent(&event)); | |
246 EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(true)); | |
247 EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); | |
248 track->RemoveSink(sink.get()); | |
249 | |
250 EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return()); | |
251 capturer_->Stop(); | |
252 track.reset(); | |
253 } | |
254 | |
255 // Create multiple audio tracks and enable/disable them, verify that the audio | |
256 // callbacks appear/disappear. | |
257 // Flaky due to a data race, see http://crbug.com/295418 | |
258 TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) { | |
259 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1( | |
260 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
261 scoped_ptr<WebRtcLocalAudioTrack> track_1( | |
262 new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL)); | |
263 track_1->Start(); | |
264 EXPECT_TRUE(track_1->GetAudioAdapter()->enabled()); | |
265 scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink()); | |
266 const media::AudioParameters params = capturer_->source_audio_parameters(); | |
267 base::WaitableEvent event_1(false, false); | |
268 EXPECT_CALL(*sink_1, FormatIsSet()).WillOnce(Return()); | |
269 EXPECT_CALL(*sink_1, CaptureData()).Times(AtLeast(1)) | |
270 .WillRepeatedly(SignalEvent(&event_1)); | |
271 EXPECT_EQ(sink_1->audio_params().frames_per_buffer(), | |
272 params.sample_rate() / 100); | |
273 track_1->AddSink(sink_1.get()); | |
274 EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout())); | |
275 | |
276 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2( | |
277 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
278 scoped_ptr<WebRtcLocalAudioTrack> track_2( | |
279 new WebRtcLocalAudioTrack(adapter_2.get(), capturer_, NULL)); | |
280 track_2->Start(); | |
281 EXPECT_TRUE(track_2->GetAudioAdapter()->enabled()); | |
282 | |
283 // Verify both |sink_1| and |sink_2| get data. | |
284 event_1.Reset(); | |
285 base::WaitableEvent event_2(false, false); | |
286 | |
287 scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink()); | |
288 EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(Return()); | |
289 EXPECT_CALL(*sink_1, CaptureData()).Times(AtLeast(1)) | |
290 .WillRepeatedly(SignalEvent(&event_1)); | |
291 EXPECT_EQ(sink_1->audio_params().frames_per_buffer(), | |
292 params.sample_rate() / 100); | |
293 EXPECT_CALL(*sink_2, CaptureData()).Times(AtLeast(1)) | |
294 .WillRepeatedly(SignalEvent(&event_2)); | |
295 EXPECT_EQ(sink_2->audio_params().frames_per_buffer(), | |
296 params.sample_rate() / 100); | |
297 track_2->AddSink(sink_2.get()); | |
298 EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout())); | |
299 EXPECT_TRUE(event_2.TimedWait(TestTimeouts::tiny_timeout())); | |
300 | |
301 track_1->RemoveSink(sink_1.get()); | |
302 track_1->Stop(); | |
303 track_1.reset(); | |
304 | |
305 EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return()); | |
306 track_2->RemoveSink(sink_2.get()); | |
307 track_2->Stop(); | |
308 track_2.reset(); | |
309 } | |
310 | |
311 | |
312 // Start one track and verify the capturer is correctly starting its source. | |
313 // And it should be fine to not to call Stop() explicitly. | |
314 TEST_F(WebRtcLocalAudioTrackTest, StartOneAudioTrack) { | |
315 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( | |
316 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
317 scoped_ptr<WebRtcLocalAudioTrack> track( | |
318 new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL)); | |
319 track->Start(); | |
320 | |
321 // When the track goes away, it will automatically stop the | |
322 // |capturer_source_|. | |
323 EXPECT_CALL(*capturer_source_.get(), OnStop()); | |
324 track.reset(); | |
325 } | |
326 | |
327 // Start two tracks and verify the capturer is correctly starting its source. | |
328 // When the last track connected to the capturer is stopped, the source is | |
329 // stopped. | |
330 TEST_F(WebRtcLocalAudioTrackTest, StartTwoAudioTracks) { | |
331 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter1( | |
332 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
333 scoped_ptr<WebRtcLocalAudioTrack> track1( | |
334 new WebRtcLocalAudioTrack(adapter1.get(), capturer_, NULL)); | |
335 track1->Start(); | |
336 | |
337 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter2( | |
338 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
339 scoped_ptr<WebRtcLocalAudioTrack> track2( | |
340 new WebRtcLocalAudioTrack(adapter2.get(), capturer_, NULL)); | |
341 track2->Start(); | |
342 | |
343 track1->Stop(); | |
344 // When the last track is stopped, it will automatically stop the | |
345 // |capturer_source_|. | |
346 EXPECT_CALL(*capturer_source_.get(), OnStop()); | |
347 track2->Stop(); | |
348 } | |
349 | |
350 // Start/Stop tracks and verify the capturer is correctly starting/stopping | |
351 // its source. | |
352 TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) { | |
353 base::WaitableEvent event(false, false); | |
354 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1( | |
355 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
356 scoped_ptr<WebRtcLocalAudioTrack> track_1( | |
357 new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL)); | |
358 track_1->Start(); | |
359 | |
360 // Verify the data flow by connecting the sink to |track_1|. | |
361 scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink()); | |
362 event.Reset(); | |
363 EXPECT_CALL(*sink, FormatIsSet()).WillOnce(SignalEvent(&event)); | |
364 EXPECT_CALL(*sink, CaptureData()) | |
365 .Times(AnyNumber()).WillRepeatedly(Return()); | |
366 track_1->AddSink(sink.get()); | |
367 EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); | |
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(), OnStart()).Times(0); | |
372 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2( | |
373 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
374 scoped_ptr<WebRtcLocalAudioTrack> track_2( | |
375 new WebRtcLocalAudioTrack(adapter_2.get(), capturer_, NULL)); | |
376 track_2->Start(); | |
377 | |
378 // Stop the capturer will clear up the track lists in the capturer. | |
379 EXPECT_CALL(*capturer_source_.get(), OnStop()); | |
380 capturer_->Stop(); | |
381 | |
382 // Adding a new track to the capturer. | |
383 track_2->AddSink(sink.get()); | |
384 EXPECT_CALL(*sink, FormatIsSet()).Times(0); | |
385 | |
386 // Stop the capturer again will not trigger stopping the source of the | |
387 // capturer again.. | |
388 event.Reset(); | |
389 EXPECT_CALL(*capturer_source_.get(), OnStop()).Times(0); | |
390 capturer_->Stop(); | |
391 } | |
392 | |
393 // Create a new capturer with new source, connect it to a new audio track. | |
394 #if defined(THREAD_SANITIZER) | |
395 // Fails under TSan, see https://crbug.com/576634. | |
396 #define MAYBE_ConnectTracksToDifferentCapturers \ | |
397 DISABLED_ConnectTracksToDifferentCapturers | |
398 #else | |
399 #define MAYBE_ConnectTracksToDifferentCapturers \ | |
400 ConnectTracksToDifferentCapturers | |
401 #endif | |
402 TEST_F(WebRtcLocalAudioTrackTest, MAYBE_ConnectTracksToDifferentCapturers) { | |
403 // Setup the first audio track and start it. | |
404 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1( | |
405 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
406 scoped_ptr<WebRtcLocalAudioTrack> track_1( | |
407 new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL)); | |
408 track_1->Start(); | |
409 | |
410 // Verify the data flow by connecting the |sink_1| to |track_1|. | |
411 scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink()); | |
412 EXPECT_CALL(*sink_1.get(), CaptureData()) | |
413 .Times(AnyNumber()).WillRepeatedly(Return()); | |
414 EXPECT_CALL(*sink_1.get(), FormatIsSet()).Times(AnyNumber()); | |
415 track_1->AddSink(sink_1.get()); | |
416 | |
417 // Create a new capturer with new source with different audio format. | |
418 MockMediaConstraintFactory constraint_factory; | |
419 StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE, | |
420 std::string(), std::string()); | |
421 scoped_refptr<WebRtcAudioCapturer> new_capturer( | |
422 WebRtcAudioCapturer::CreateCapturer( | |
423 -1, device, constraint_factory.CreateWebMediaConstraints(), NULL, | |
424 NULL)); | |
425 scoped_refptr<MockCapturerSource> new_source( | |
426 new MockCapturerSource(new_capturer.get())); | |
427 EXPECT_CALL(*new_source.get(), OnInitialize(_, new_capturer.get(), -1)); | |
428 EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(true)); | |
429 EXPECT_CALL(*new_source.get(), OnStart()); | |
430 | |
431 media::AudioParameters new_param( | |
432 media::AudioParameters::AUDIO_PCM_LOW_LATENCY, | |
433 media::CHANNEL_LAYOUT_MONO, 44100, 16, 441); | |
434 new_capturer->SetCapturerSource(new_source, new_param); | |
435 | |
436 // Setup the second audio track, connect it to the new capturer and start it. | |
437 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2( | |
438 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
439 scoped_ptr<WebRtcLocalAudioTrack> track_2( | |
440 new WebRtcLocalAudioTrack(adapter_2.get(), new_capturer, NULL)); | |
441 track_2->Start(); | |
442 | |
443 // Verify the data flow by connecting the |sink_2| to |track_2|. | |
444 scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink()); | |
445 base::WaitableEvent event(false, false); | |
446 EXPECT_CALL(*sink_2, CaptureData()) | |
447 .Times(AnyNumber()).WillRepeatedly(Return()); | |
448 EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(SignalEvent(&event)); | |
449 track_2->AddSink(sink_2.get()); | |
450 EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); | |
451 | |
452 // Stopping the new source will stop the second track. | |
453 event.Reset(); | |
454 EXPECT_CALL(*new_source.get(), OnStop()) | |
455 .Times(1).WillOnce(SignalEvent(&event)); | |
456 new_capturer->Stop(); | |
457 EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); | |
458 | |
459 // Stop the capturer of the first audio track. | |
460 EXPECT_CALL(*capturer_source_.get(), OnStop()); | |
461 capturer_->Stop(); | |
462 } | |
463 | |
464 // Make sure a audio track can deliver packets with a buffer size smaller than | |
465 // 10ms when it is not connected with a peer connection. | |
466 TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) { | |
467 // Setup a capturer which works with a buffer size smaller than 10ms. | |
468 media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, | |
469 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128); | |
470 | |
471 // Create a capturer with new source which works with the format above. | |
472 MockMediaConstraintFactory factory; | |
473 factory.DisableDefaultAudioConstraints(); | |
474 scoped_refptr<WebRtcAudioCapturer> capturer( | |
475 WebRtcAudioCapturer::CreateCapturer( | |
476 -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", "", | |
477 params.sample_rate(), params.channel_layout(), | |
478 params.frames_per_buffer()), | |
479 factory.CreateWebMediaConstraints(), NULL, NULL)); | |
480 scoped_refptr<MockCapturerSource> source( | |
481 new MockCapturerSource(capturer.get())); | |
482 EXPECT_CALL(*source.get(), OnInitialize(_, capturer.get(), -1)); | |
483 EXPECT_CALL(*source.get(), SetAutomaticGainControl(true)); | |
484 EXPECT_CALL(*source.get(), OnStart()); | |
485 capturer->SetCapturerSource(source, params); | |
486 | |
487 // Setup a audio track, connect it to the capturer and start it. | |
488 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( | |
489 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
490 scoped_ptr<WebRtcLocalAudioTrack> track( | |
491 new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL)); | |
492 track->Start(); | |
493 | |
494 // Verify the data flow by connecting the |sink| to |track|. | |
495 scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink()); | |
496 base::WaitableEvent event(false, false); | |
497 EXPECT_CALL(*sink, FormatIsSet()).Times(1); | |
498 // Verify the sinks are getting the packets with an expecting buffer size. | |
499 #if defined(OS_ANDROID) | |
500 const int expected_buffer_size = params.sample_rate() / 100; | |
501 #else | |
502 const int expected_buffer_size = params.frames_per_buffer(); | |
503 #endif | |
504 EXPECT_CALL(*sink, CaptureData()) | |
505 .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event)); | |
506 track->AddSink(sink.get()); | |
507 EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); | |
508 EXPECT_EQ(expected_buffer_size, sink->audio_params().frames_per_buffer()); | |
509 | |
510 // Stopping the new source will stop the second track. | |
511 EXPECT_CALL(*source.get(), OnStop()).Times(1); | |
512 capturer->Stop(); | |
513 | |
514 // Even though this test don't use |capturer_source_| it will be stopped | |
515 // during teardown of the test harness. | |
516 EXPECT_CALL(*capturer_source_.get(), OnStop()); | |
517 } | |
518 | |
519 } // namespace content | |
OLD | NEW |