OLD | NEW |
1 // Copyright 2013 The Chromium Authors. All rights reserved. | 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 | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 #include "base/macros.h" | 5 #include "base/macros.h" |
6 #include "base/synchronization/waitable_event.h" | 6 #include "base/synchronization/waitable_event.h" |
7 #include "base/test/test_timeouts.h" | 7 #include "base/test/test_timeouts.h" |
8 #include "build/build_config.h" | 8 #include "build/build_config.h" |
9 #include "content/public/renderer/media_stream_audio_sink.h" | 9 #include "content/public/renderer/media_stream_audio_sink.h" |
10 #include "content/renderer/media/media_stream_audio_source.h" | 10 #include "content/renderer/media/media_stream_audio_source.h" |
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88 | 88 |
89 class MockCapturerSource : public media::AudioCapturerSource { | 89 class MockCapturerSource : public media::AudioCapturerSource { |
90 public: | 90 public: |
91 explicit MockCapturerSource(WebRtcAudioCapturer* capturer) | 91 explicit MockCapturerSource(WebRtcAudioCapturer* capturer) |
92 : capturer_(capturer) {} | 92 : capturer_(capturer) {} |
93 MOCK_METHOD3(OnInitialize, void(const media::AudioParameters& params, | 93 MOCK_METHOD3(OnInitialize, void(const media::AudioParameters& params, |
94 CaptureCallback* callback, | 94 CaptureCallback* callback, |
95 int session_id)); | 95 int session_id)); |
96 MOCK_METHOD0(OnStart, void()); | 96 MOCK_METHOD0(OnStart, void()); |
97 MOCK_METHOD0(OnStop, void()); | 97 MOCK_METHOD0(OnStop, void()); |
98 void SetVolume(double volume) final {} | 98 MOCK_METHOD1(SetVolume, void(double volume)); |
99 MOCK_METHOD1(SetAutomaticGainControl, void(bool enable)); | 99 MOCK_METHOD1(SetAutomaticGainControl, void(bool enable)); |
100 | 100 |
101 void Initialize(const media::AudioParameters& params, | 101 void Initialize(const media::AudioParameters& params, |
102 CaptureCallback* callback, | 102 CaptureCallback* callback, |
103 int session_id) override { | 103 int session_id) override { |
104 DCHECK(params.IsValid()); | 104 DCHECK(params.IsValid()); |
105 params_ = params; | 105 params_ = params; |
106 OnInitialize(params, callback, session_id); | 106 OnInitialize(params, callback, session_id); |
107 } | 107 } |
108 void Start() override { | 108 void Start() override { |
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158 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 480); | 158 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 480); |
159 MockConstraintFactory constraint_factory; | 159 MockConstraintFactory constraint_factory; |
160 blink_source_.initialize("dummy", blink::WebMediaStreamSource::TypeAudio, | 160 blink_source_.initialize("dummy", blink::WebMediaStreamSource::TypeAudio, |
161 "dummy", | 161 "dummy", |
162 false /* remote */, true /* readonly */); | 162 false /* remote */, true /* readonly */); |
163 MediaStreamAudioSource* audio_source = new MediaStreamAudioSource(); | 163 MediaStreamAudioSource* audio_source = new MediaStreamAudioSource(); |
164 blink_source_.setExtraData(audio_source); | 164 blink_source_.setExtraData(audio_source); |
165 | 165 |
166 StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE, | 166 StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE, |
167 std::string(), std::string()); | 167 std::string(), std::string()); |
168 { | 168 capturer_ = WebRtcAudioCapturer::CreateCapturer( |
169 scoped_ptr<WebRtcAudioCapturer> capturer = | 169 -1, device, constraint_factory.CreateWebMediaConstraints(), NULL, |
170 WebRtcAudioCapturer::CreateCapturer( | 170 audio_source); |
171 -1, device, constraint_factory.CreateWebMediaConstraints(), | 171 audio_source->SetAudioCapturer(capturer_.get()); |
172 nullptr, audio_source); | 172 capturer_source_ = new MockCapturerSource(capturer_.get()); |
173 capturer_ = capturer.get(); | 173 EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_.get(), -1)) |
174 audio_source->SetAudioCapturer(std::move(capturer)); | |
175 } | |
176 capturer_source_ = new MockCapturerSource(capturer_); | |
177 EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_, -1)) | |
178 .WillOnce(Return()); | 174 .WillOnce(Return()); |
179 EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); | 175 EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); |
180 EXPECT_CALL(*capturer_source_.get(), OnStart()); | 176 EXPECT_CALL(*capturer_source_.get(), OnStart()); |
181 capturer_->SetCapturerSource(capturer_source_, params_); | 177 capturer_->SetCapturerSource(capturer_source_, params_); |
182 } | 178 } |
183 | 179 |
184 void TearDown() override { | 180 void TearDown() override { |
185 blink_source_.reset(); | 181 blink_source_.reset(); |
186 blink::WebHeap::collectAllGarbageForTesting(); | 182 blink::WebHeap::collectAllGarbageForTesting(); |
187 } | 183 } |
188 | 184 |
189 media::AudioParameters params_; | 185 media::AudioParameters params_; |
190 blink::WebMediaStreamSource blink_source_; | 186 blink::WebMediaStreamSource blink_source_; |
191 WebRtcAudioCapturer* capturer_; // Owned by |blink_source_|. | |
192 scoped_refptr<MockCapturerSource> capturer_source_; | 187 scoped_refptr<MockCapturerSource> capturer_source_; |
| 188 scoped_refptr<WebRtcAudioCapturer> capturer_; |
193 }; | 189 }; |
194 | 190 |
195 // Creates a capturer and audio track, fakes its audio thread, and | 191 // Creates a capturer and audio track, fakes its audio thread, and |
196 // connect/disconnect the sink to the audio track on the fly, the sink should | 192 // connect/disconnect the sink to the audio track on the fly, the sink should |
197 // get data callback when the track is connected to the capturer but not when | 193 // get data callback when the track is connected to the capturer but not when |
198 // the track is disconnected from the capturer. | 194 // the track is disconnected from the capturer. |
199 TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) { | 195 TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) { |
200 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( | 196 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
201 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | 197 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
202 scoped_ptr<WebRtcLocalAudioTrack> track( | 198 scoped_ptr<WebRtcLocalAudioTrack> track( |
203 new WebRtcLocalAudioTrack(adapter.get())); | 199 new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL)); |
204 track->Start( | 200 track->Start(); |
205 base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo, | |
206 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(), | |
207 track.get())); | |
208 capturer_->AddTrack(track.get()); | |
209 EXPECT_TRUE(track->GetAudioAdapter()->enabled()); | 201 EXPECT_TRUE(track->GetAudioAdapter()->enabled()); |
210 | 202 |
211 scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink()); | 203 scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink()); |
212 base::WaitableEvent event(false, false); | 204 base::WaitableEvent event(false, false); |
213 EXPECT_CALL(*sink, FormatIsSet()); | 205 EXPECT_CALL(*sink, FormatIsSet()); |
214 EXPECT_CALL(*sink, | 206 EXPECT_CALL(*sink, |
215 CaptureData()).Times(AtLeast(1)) | 207 CaptureData()).Times(AtLeast(1)) |
216 .WillRepeatedly(SignalEvent(&event)); | 208 .WillRepeatedly(SignalEvent(&event)); |
217 track->AddSink(sink.get()); | 209 track->AddSink(sink.get()); |
218 EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); | 210 EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); |
219 track->RemoveSink(sink.get()); | 211 track->RemoveSink(sink.get()); |
220 | 212 |
221 EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return()); | 213 EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return()); |
222 capturer_->Stop(); | 214 capturer_->Stop(); |
223 } | 215 } |
224 | 216 |
225 // The same setup as ConnectAndDisconnectOneSink, but enable and disable the | 217 // The same setup as ConnectAndDisconnectOneSink, but enable and disable the |
226 // audio track on the fly. When the audio track is disabled, there is no data | 218 // audio track on the fly. When the audio track is disabled, there is no data |
227 // callback to the sink; when the audio track is enabled, there comes data | 219 // callback to the sink; when the audio track is enabled, there comes data |
228 // callback. | 220 // callback. |
229 // TODO(xians): Enable this test after resolving the racing issue that TSAN | 221 // TODO(xians): Enable this test after resolving the racing issue that TSAN |
230 // reports on MediaStreamTrack::enabled(); | 222 // reports on MediaStreamTrack::enabled(); |
231 TEST_F(WebRtcLocalAudioTrackTest, DISABLED_DisableEnableAudioTrack) { | 223 TEST_F(WebRtcLocalAudioTrackTest, DISABLED_DisableEnableAudioTrack) { |
232 EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); | 224 EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); |
233 EXPECT_CALL(*capturer_source_.get(), OnStart()); | 225 EXPECT_CALL(*capturer_source_.get(), OnStart()); |
234 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( | 226 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
235 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | 227 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
236 scoped_ptr<WebRtcLocalAudioTrack> track( | 228 scoped_ptr<WebRtcLocalAudioTrack> track( |
237 new WebRtcLocalAudioTrack(adapter.get())); | 229 new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL)); |
238 track->Start( | 230 track->Start(); |
239 base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo, | |
240 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(), | |
241 track.get())); | |
242 capturer_->AddTrack(track.get()); | |
243 EXPECT_TRUE(track->GetAudioAdapter()->enabled()); | 231 EXPECT_TRUE(track->GetAudioAdapter()->enabled()); |
244 EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(false)); | 232 EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(false)); |
245 scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink()); | 233 scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink()); |
246 const media::AudioParameters params = capturer_->GetInputFormat(); | 234 const media::AudioParameters params = capturer_->source_audio_parameters(); |
247 base::WaitableEvent event(false, false); | 235 base::WaitableEvent event(false, false); |
248 EXPECT_CALL(*sink, FormatIsSet()).Times(1); | 236 EXPECT_CALL(*sink, FormatIsSet()).Times(1); |
249 EXPECT_CALL(*sink, CaptureData()).Times(0); | 237 EXPECT_CALL(*sink, CaptureData()).Times(0); |
250 EXPECT_EQ(sink->audio_params().frames_per_buffer(), | 238 EXPECT_EQ(sink->audio_params().frames_per_buffer(), |
251 params.sample_rate() / 100); | 239 params.sample_rate() / 100); |
252 track->AddSink(sink.get()); | 240 track->AddSink(sink.get()); |
253 EXPECT_FALSE(event.TimedWait(TestTimeouts::tiny_timeout())); | 241 EXPECT_FALSE(event.TimedWait(TestTimeouts::tiny_timeout())); |
254 | 242 |
255 event.Reset(); | 243 event.Reset(); |
256 EXPECT_CALL(*sink, CaptureData()).Times(AtLeast(1)) | 244 EXPECT_CALL(*sink, CaptureData()).Times(AtLeast(1)) |
257 .WillRepeatedly(SignalEvent(&event)); | 245 .WillRepeatedly(SignalEvent(&event)); |
258 EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(true)); | 246 EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(true)); |
259 EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); | 247 EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); |
260 track->RemoveSink(sink.get()); | 248 track->RemoveSink(sink.get()); |
261 | 249 |
262 EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return()); | 250 EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return()); |
263 capturer_->Stop(); | 251 capturer_->Stop(); |
264 track.reset(); | 252 track.reset(); |
265 } | 253 } |
266 | 254 |
267 // Create multiple audio tracks and enable/disable them, verify that the audio | 255 // Create multiple audio tracks and enable/disable them, verify that the audio |
268 // callbacks appear/disappear. | 256 // callbacks appear/disappear. |
269 // Flaky due to a data race, see http://crbug.com/295418 | 257 // Flaky due to a data race, see http://crbug.com/295418 |
270 TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) { | 258 TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) { |
271 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1( | 259 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1( |
272 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | 260 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
273 scoped_ptr<WebRtcLocalAudioTrack> track_1( | 261 scoped_ptr<WebRtcLocalAudioTrack> track_1( |
274 new WebRtcLocalAudioTrack(adapter_1.get())); | 262 new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL)); |
275 track_1->Start( | 263 track_1->Start(); |
276 base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo, | |
277 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(), | |
278 track_1.get())); | |
279 capturer_->AddTrack(track_1.get()); | |
280 EXPECT_TRUE(track_1->GetAudioAdapter()->enabled()); | 264 EXPECT_TRUE(track_1->GetAudioAdapter()->enabled()); |
281 scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink()); | 265 scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink()); |
282 const media::AudioParameters params = capturer_->GetInputFormat(); | 266 const media::AudioParameters params = capturer_->source_audio_parameters(); |
283 base::WaitableEvent event_1(false, false); | 267 base::WaitableEvent event_1(false, false); |
284 EXPECT_CALL(*sink_1, FormatIsSet()).WillOnce(Return()); | 268 EXPECT_CALL(*sink_1, FormatIsSet()).WillOnce(Return()); |
285 EXPECT_CALL(*sink_1, CaptureData()).Times(AtLeast(1)) | 269 EXPECT_CALL(*sink_1, CaptureData()).Times(AtLeast(1)) |
286 .WillRepeatedly(SignalEvent(&event_1)); | 270 .WillRepeatedly(SignalEvent(&event_1)); |
287 EXPECT_EQ(sink_1->audio_params().frames_per_buffer(), | 271 EXPECT_EQ(sink_1->audio_params().frames_per_buffer(), |
288 params.sample_rate() / 100); | 272 params.sample_rate() / 100); |
289 track_1->AddSink(sink_1.get()); | 273 track_1->AddSink(sink_1.get()); |
290 EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout())); | 274 EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout())); |
291 | 275 |
292 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2( | 276 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2( |
293 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | 277 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
294 scoped_ptr<WebRtcLocalAudioTrack> track_2( | 278 scoped_ptr<WebRtcLocalAudioTrack> track_2( |
295 new WebRtcLocalAudioTrack(adapter_2.get())); | 279 new WebRtcLocalAudioTrack(adapter_2.get(), capturer_, NULL)); |
296 track_2->Start( | 280 track_2->Start(); |
297 base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo, | |
298 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(), | |
299 track_2.get())); | |
300 capturer_->AddTrack(track_2.get()); | |
301 EXPECT_TRUE(track_2->GetAudioAdapter()->enabled()); | 281 EXPECT_TRUE(track_2->GetAudioAdapter()->enabled()); |
302 | 282 |
303 // Verify both |sink_1| and |sink_2| get data. | 283 // Verify both |sink_1| and |sink_2| get data. |
304 event_1.Reset(); | 284 event_1.Reset(); |
305 base::WaitableEvent event_2(false, false); | 285 base::WaitableEvent event_2(false, false); |
306 | 286 |
307 scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink()); | 287 scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink()); |
308 EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(Return()); | 288 EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(Return()); |
309 EXPECT_CALL(*sink_1, CaptureData()).Times(AtLeast(1)) | 289 EXPECT_CALL(*sink_1, CaptureData()).Times(AtLeast(1)) |
310 .WillRepeatedly(SignalEvent(&event_1)); | 290 .WillRepeatedly(SignalEvent(&event_1)); |
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328 track_2.reset(); | 308 track_2.reset(); |
329 } | 309 } |
330 | 310 |
331 | 311 |
332 // Start one track and verify the capturer is correctly starting its source. | 312 // Start one track and verify the capturer is correctly starting its source. |
333 // And it should be fine to not to call Stop() explicitly. | 313 // And it should be fine to not to call Stop() explicitly. |
334 TEST_F(WebRtcLocalAudioTrackTest, StartOneAudioTrack) { | 314 TEST_F(WebRtcLocalAudioTrackTest, StartOneAudioTrack) { |
335 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( | 315 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
336 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | 316 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
337 scoped_ptr<WebRtcLocalAudioTrack> track( | 317 scoped_ptr<WebRtcLocalAudioTrack> track( |
338 new WebRtcLocalAudioTrack(adapter.get())); | 318 new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL)); |
339 track->Start( | 319 track->Start(); |
340 base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo, | |
341 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(), | |
342 track.get())); | |
343 capturer_->AddTrack(track.get()); | |
344 | 320 |
345 // When the track goes away, it will automatically stop the | 321 // When the track goes away, it will automatically stop the |
346 // |capturer_source_|. | 322 // |capturer_source_|. |
347 EXPECT_CALL(*capturer_source_.get(), OnStop()); | 323 EXPECT_CALL(*capturer_source_.get(), OnStop()); |
348 track.reset(); | 324 track.reset(); |
349 } | 325 } |
350 | 326 |
351 // Start two tracks and verify the capturer is correctly starting its source. | 327 // Start two tracks and verify the capturer is correctly starting its source. |
352 // When the last track connected to the capturer is stopped, the source is | 328 // When the last track connected to the capturer is stopped, the source is |
353 // stopped. | 329 // stopped. |
354 TEST_F(WebRtcLocalAudioTrackTest, StartTwoAudioTracks) { | 330 TEST_F(WebRtcLocalAudioTrackTest, StartTwoAudioTracks) { |
355 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter1( | 331 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter1( |
356 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | 332 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
357 scoped_ptr<WebRtcLocalAudioTrack> track1( | 333 scoped_ptr<WebRtcLocalAudioTrack> track1( |
358 new WebRtcLocalAudioTrack(adapter1.get())); | 334 new WebRtcLocalAudioTrack(adapter1.get(), capturer_, NULL)); |
359 track1->Start( | 335 track1->Start(); |
360 base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo, | |
361 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(), | |
362 track1.get())); | |
363 capturer_->AddTrack(track1.get()); | |
364 | 336 |
365 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter2( | 337 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter2( |
366 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | 338 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
367 scoped_ptr<WebRtcLocalAudioTrack> track2( | 339 scoped_ptr<WebRtcLocalAudioTrack> track2( |
368 new WebRtcLocalAudioTrack(adapter2.get())); | 340 new WebRtcLocalAudioTrack(adapter2.get(), capturer_, NULL)); |
369 track2->Start( | 341 track2->Start(); |
370 base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo, | |
371 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(), | |
372 track2.get())); | |
373 capturer_->AddTrack(track2.get()); | |
374 | 342 |
375 track1->Stop(); | 343 track1->Stop(); |
376 // When the last track is stopped, it will automatically stop the | 344 // When the last track is stopped, it will automatically stop the |
377 // |capturer_source_|. | 345 // |capturer_source_|. |
378 EXPECT_CALL(*capturer_source_.get(), OnStop()); | 346 EXPECT_CALL(*capturer_source_.get(), OnStop()); |
379 track2->Stop(); | 347 track2->Stop(); |
380 } | 348 } |
381 | 349 |
382 // Start/Stop tracks and verify the capturer is correctly starting/stopping | 350 // Start/Stop tracks and verify the capturer is correctly starting/stopping |
383 // its source. | 351 // its source. |
384 TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) { | 352 TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) { |
385 base::WaitableEvent event(false, false); | 353 base::WaitableEvent event(false, false); |
386 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1( | 354 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1( |
387 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | 355 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
388 scoped_ptr<WebRtcLocalAudioTrack> track_1( | 356 scoped_ptr<WebRtcLocalAudioTrack> track_1( |
389 new WebRtcLocalAudioTrack(adapter_1.get())); | 357 new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL)); |
390 track_1->Start( | 358 track_1->Start(); |
391 base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo, | |
392 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(), | |
393 track_1.get())); | |
394 capturer_->AddTrack(track_1.get()); | |
395 | 359 |
396 // Verify the data flow by connecting the sink to |track_1|. | 360 // Verify the data flow by connecting the sink to |track_1|. |
397 scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink()); | 361 scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink()); |
398 event.Reset(); | 362 event.Reset(); |
399 EXPECT_CALL(*sink, FormatIsSet()).WillOnce(SignalEvent(&event)); | 363 EXPECT_CALL(*sink, FormatIsSet()).WillOnce(SignalEvent(&event)); |
400 EXPECT_CALL(*sink, CaptureData()) | 364 EXPECT_CALL(*sink, CaptureData()) |
401 .Times(AnyNumber()).WillRepeatedly(Return()); | 365 .Times(AnyNumber()).WillRepeatedly(Return()); |
402 track_1->AddSink(sink.get()); | 366 track_1->AddSink(sink.get()); |
403 EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); | 367 EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); |
404 | 368 |
405 // Start the second audio track will not start the |capturer_source_| | 369 // Start the second audio track will not start the |capturer_source_| |
406 // since it has been started. | 370 // since it has been started. |
407 EXPECT_CALL(*capturer_source_.get(), OnStart()).Times(0); | 371 EXPECT_CALL(*capturer_source_.get(), OnStart()).Times(0); |
408 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2( | 372 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2( |
409 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | 373 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
410 scoped_ptr<WebRtcLocalAudioTrack> track_2( | 374 scoped_ptr<WebRtcLocalAudioTrack> track_2( |
411 new WebRtcLocalAudioTrack(adapter_2.get())); | 375 new WebRtcLocalAudioTrack(adapter_2.get(), capturer_, NULL)); |
412 track_2->Start( | 376 track_2->Start(); |
413 base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo, | |
414 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(), | |
415 track_2.get())); | |
416 capturer_->AddTrack(track_2.get()); | |
417 | 377 |
418 // Stop the capturer will clear up the track lists in the capturer. | 378 // Stop the capturer will clear up the track lists in the capturer. |
419 EXPECT_CALL(*capturer_source_.get(), OnStop()); | 379 EXPECT_CALL(*capturer_source_.get(), OnStop()); |
420 capturer_->Stop(); | 380 capturer_->Stop(); |
421 | 381 |
422 // Adding a new track to the capturer. | 382 // Adding a new track to the capturer. |
423 track_2->AddSink(sink.get()); | 383 track_2->AddSink(sink.get()); |
424 EXPECT_CALL(*sink, FormatIsSet()).Times(0); | 384 EXPECT_CALL(*sink, FormatIsSet()).Times(0); |
425 | 385 |
426 // Stop the capturer again will not trigger stopping the source of the | 386 // Stop the capturer again will not trigger stopping the source of the |
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437 DISABLED_ConnectTracksToDifferentCapturers | 397 DISABLED_ConnectTracksToDifferentCapturers |
438 #else | 398 #else |
439 #define MAYBE_ConnectTracksToDifferentCapturers \ | 399 #define MAYBE_ConnectTracksToDifferentCapturers \ |
440 ConnectTracksToDifferentCapturers | 400 ConnectTracksToDifferentCapturers |
441 #endif | 401 #endif |
442 TEST_F(WebRtcLocalAudioTrackTest, MAYBE_ConnectTracksToDifferentCapturers) { | 402 TEST_F(WebRtcLocalAudioTrackTest, MAYBE_ConnectTracksToDifferentCapturers) { |
443 // Setup the first audio track and start it. | 403 // Setup the first audio track and start it. |
444 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1( | 404 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1( |
445 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | 405 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
446 scoped_ptr<WebRtcLocalAudioTrack> track_1( | 406 scoped_ptr<WebRtcLocalAudioTrack> track_1( |
447 new WebRtcLocalAudioTrack(adapter_1.get())); | 407 new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL)); |
448 track_1->Start( | 408 track_1->Start(); |
449 base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo, | |
450 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(), | |
451 track_1.get())); | |
452 capturer_->AddTrack(track_1.get()); | |
453 | 409 |
454 // Verify the data flow by connecting the |sink_1| to |track_1|. | 410 // Verify the data flow by connecting the |sink_1| to |track_1|. |
455 scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink()); | 411 scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink()); |
456 EXPECT_CALL(*sink_1.get(), CaptureData()) | 412 EXPECT_CALL(*sink_1.get(), CaptureData()) |
457 .Times(AnyNumber()).WillRepeatedly(Return()); | 413 .Times(AnyNumber()).WillRepeatedly(Return()); |
458 EXPECT_CALL(*sink_1.get(), FormatIsSet()).Times(AnyNumber()); | 414 EXPECT_CALL(*sink_1.get(), FormatIsSet()).Times(AnyNumber()); |
459 track_1->AddSink(sink_1.get()); | 415 track_1->AddSink(sink_1.get()); |
460 | 416 |
461 // Create a new capturer with new source with different audio format. | 417 // Create a new capturer with new source with different audio format. |
462 MockConstraintFactory constraint_factory; | 418 MockConstraintFactory constraint_factory; |
463 StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE, | 419 StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE, |
464 std::string(), std::string()); | 420 std::string(), std::string()); |
465 scoped_ptr<WebRtcAudioCapturer> new_capturer( | 421 scoped_refptr<WebRtcAudioCapturer> new_capturer( |
466 WebRtcAudioCapturer::CreateCapturer( | 422 WebRtcAudioCapturer::CreateCapturer( |
467 -1, device, constraint_factory.CreateWebMediaConstraints(), NULL, | 423 -1, device, constraint_factory.CreateWebMediaConstraints(), NULL, |
468 NULL)); | 424 NULL)); |
469 scoped_refptr<MockCapturerSource> new_source( | 425 scoped_refptr<MockCapturerSource> new_source( |
470 new MockCapturerSource(new_capturer.get())); | 426 new MockCapturerSource(new_capturer.get())); |
471 EXPECT_CALL(*new_source.get(), OnInitialize(_, new_capturer.get(), -1)); | 427 EXPECT_CALL(*new_source.get(), OnInitialize(_, new_capturer.get(), -1)); |
472 EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(true)); | 428 EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(true)); |
473 EXPECT_CALL(*new_source.get(), OnStart()); | 429 EXPECT_CALL(*new_source.get(), OnStart()); |
474 | 430 |
475 media::AudioParameters new_param( | 431 media::AudioParameters new_param( |
476 media::AudioParameters::AUDIO_PCM_LOW_LATENCY, | 432 media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
477 media::CHANNEL_LAYOUT_MONO, 44100, 16, 441); | 433 media::CHANNEL_LAYOUT_MONO, 44100, 16, 441); |
478 new_capturer->SetCapturerSource(new_source, new_param); | 434 new_capturer->SetCapturerSource(new_source, new_param); |
479 | 435 |
480 // Setup the second audio track, connect it to the new capturer and start it. | 436 // Setup the second audio track, connect it to the new capturer and start it. |
481 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2( | 437 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2( |
482 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | 438 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
483 scoped_ptr<WebRtcLocalAudioTrack> track_2( | 439 scoped_ptr<WebRtcLocalAudioTrack> track_2( |
484 new WebRtcLocalAudioTrack(adapter_2.get())); | 440 new WebRtcLocalAudioTrack(adapter_2.get(), new_capturer, NULL)); |
485 track_2->Start( | 441 track_2->Start(); |
486 base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo, | |
487 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(), | |
488 track_2.get())); | |
489 new_capturer->AddTrack(track_2.get()); | |
490 | 442 |
491 // Verify the data flow by connecting the |sink_2| to |track_2|. | 443 // Verify the data flow by connecting the |sink_2| to |track_2|. |
492 scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink()); | 444 scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink()); |
493 base::WaitableEvent event(false, false); | 445 base::WaitableEvent event(false, false); |
494 EXPECT_CALL(*sink_2, CaptureData()) | 446 EXPECT_CALL(*sink_2, CaptureData()) |
495 .Times(AnyNumber()).WillRepeatedly(Return()); | 447 .Times(AnyNumber()).WillRepeatedly(Return()); |
496 EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(SignalEvent(&event)); | 448 EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(SignalEvent(&event)); |
497 track_2->AddSink(sink_2.get()); | 449 track_2->AddSink(sink_2.get()); |
498 EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); | 450 EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); |
499 | 451 |
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512 // Make sure a audio track can deliver packets with a buffer size smaller than | 464 // Make sure a audio track can deliver packets with a buffer size smaller than |
513 // 10ms when it is not connected with a peer connection. | 465 // 10ms when it is not connected with a peer connection. |
514 TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) { | 466 TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) { |
515 // Setup a capturer which works with a buffer size smaller than 10ms. | 467 // Setup a capturer which works with a buffer size smaller than 10ms. |
516 media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, | 468 media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
517 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128); | 469 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128); |
518 | 470 |
519 // Create a capturer with new source which works with the format above. | 471 // Create a capturer with new source which works with the format above. |
520 MockConstraintFactory factory; | 472 MockConstraintFactory factory; |
521 factory.DisableDefaultAudioConstraints(); | 473 factory.DisableDefaultAudioConstraints(); |
522 scoped_ptr<WebRtcAudioCapturer> capturer(WebRtcAudioCapturer::CreateCapturer( | 474 scoped_refptr<WebRtcAudioCapturer> capturer( |
523 -1, | 475 WebRtcAudioCapturer::CreateCapturer( |
524 StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", "", params.sample_rate(), | 476 -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", "", |
525 params.channel_layout(), params.frames_per_buffer()), | 477 params.sample_rate(), params.channel_layout(), |
526 factory.CreateWebMediaConstraints(), NULL, NULL)); | 478 params.frames_per_buffer()), |
| 479 factory.CreateWebMediaConstraints(), NULL, NULL)); |
527 scoped_refptr<MockCapturerSource> source( | 480 scoped_refptr<MockCapturerSource> source( |
528 new MockCapturerSource(capturer.get())); | 481 new MockCapturerSource(capturer.get())); |
529 EXPECT_CALL(*source.get(), OnInitialize(_, capturer.get(), -1)); | 482 EXPECT_CALL(*source.get(), OnInitialize(_, capturer.get(), -1)); |
530 EXPECT_CALL(*source.get(), SetAutomaticGainControl(true)); | 483 EXPECT_CALL(*source.get(), SetAutomaticGainControl(true)); |
531 EXPECT_CALL(*source.get(), OnStart()); | 484 EXPECT_CALL(*source.get(), OnStart()); |
532 capturer->SetCapturerSource(source, params); | 485 capturer->SetCapturerSource(source, params); |
533 | 486 |
534 // Setup a audio track, connect it to the capturer and start it. | 487 // Setup a audio track, connect it to the capturer and start it. |
535 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( | 488 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
536 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | 489 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
537 scoped_ptr<WebRtcLocalAudioTrack> track( | 490 scoped_ptr<WebRtcLocalAudioTrack> track( |
538 new WebRtcLocalAudioTrack(adapter.get())); | 491 new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL)); |
539 track->Start( | 492 track->Start(); |
540 base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo, | |
541 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(), | |
542 track.get())); | |
543 capturer->AddTrack(track.get()); | |
544 | 493 |
545 // Verify the data flow by connecting the |sink| to |track|. | 494 // Verify the data flow by connecting the |sink| to |track|. |
546 scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink()); | 495 scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink()); |
547 base::WaitableEvent event(false, false); | 496 base::WaitableEvent event(false, false); |
548 EXPECT_CALL(*sink, FormatIsSet()).Times(1); | 497 EXPECT_CALL(*sink, FormatIsSet()).Times(1); |
549 // Verify the sinks are getting the packets with an expecting buffer size. | 498 // Verify the sinks are getting the packets with an expecting buffer size. |
550 #if defined(OS_ANDROID) | 499 #if defined(OS_ANDROID) |
551 const int expected_buffer_size = params.sample_rate() / 100; | 500 const int expected_buffer_size = params.sample_rate() / 100; |
552 #else | 501 #else |
553 const int expected_buffer_size = params.frames_per_buffer(); | 502 const int expected_buffer_size = params.frames_per_buffer(); |
554 #endif | 503 #endif |
555 EXPECT_CALL(*sink, CaptureData()) | 504 EXPECT_CALL(*sink, CaptureData()) |
556 .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event)); | 505 .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event)); |
557 track->AddSink(sink.get()); | 506 track->AddSink(sink.get()); |
558 EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); | 507 EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); |
559 EXPECT_EQ(expected_buffer_size, sink->audio_params().frames_per_buffer()); | 508 EXPECT_EQ(expected_buffer_size, sink->audio_params().frames_per_buffer()); |
560 | 509 |
561 // Stopping the new source will stop the second track. | 510 // Stopping the new source will stop the second track. |
562 EXPECT_CALL(*source.get(), OnStop()).Times(1); | 511 EXPECT_CALL(*source.get(), OnStop()).Times(1); |
563 capturer->Stop(); | 512 capturer->Stop(); |
564 | 513 |
565 // Even though this test don't use |capturer_source_| it will be stopped | 514 // Even though this test don't use |capturer_source_| it will be stopped |
566 // during teardown of the test harness. | 515 // during teardown of the test harness. |
567 EXPECT_CALL(*capturer_source_.get(), OnStop()); | 516 EXPECT_CALL(*capturer_source_.get(), OnStop()); |
568 } | 517 } |
569 | 518 |
570 } // namespace content | 519 } // namespace content |
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