| 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_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/base/audio_bus.h" | |
| 16 #include "media/base/audio_capturer_source.h" | |
| 17 #include "media/base/audio_parameters.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 std::unique_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 void SetVolume(double volume) final {} | |
| 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 std::unique_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 MockConstraintFactory constraint_factory; | |
| 160 blink_source_.initialize("dummy", blink::WebMediaStreamSource::TypeAudio, | |
| 161 "dummy", | |
| 162 false /* remote */); | |
| 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 { | |
| 169 std::unique_ptr<WebRtcAudioCapturer> capturer = | |
| 170 WebRtcAudioCapturer::CreateCapturer( | |
| 171 -1, device, constraint_factory.CreateWebMediaConstraints(), | |
| 172 nullptr, audio_source); | |
| 173 capturer_ = capturer.get(); | |
| 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()); | |
| 179 EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); | |
| 180 EXPECT_CALL(*capturer_source_.get(), OnStart()); | |
| 181 capturer_->SetCapturerSource(capturer_source_, params_); | |
| 182 } | |
| 183 | |
| 184 void TearDown() override { | |
| 185 blink_source_.reset(); | |
| 186 blink::WebHeap::collectAllGarbageForTesting(); | |
| 187 } | |
| 188 | |
| 189 media::AudioParameters params_; | |
| 190 blink::WebMediaStreamSource blink_source_; | |
| 191 WebRtcAudioCapturer* capturer_; // Owned by |blink_source_|. | |
| 192 scoped_refptr<MockCapturerSource> capturer_source_; | |
| 193 }; | |
| 194 | |
| 195 // 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 | |
| 197 // get data callback when the track is connected to the capturer but not when | |
| 198 // the track is disconnected from the capturer. | |
| 199 TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) { | |
| 200 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( | |
| 201 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
| 202 std::unique_ptr<WebRtcLocalAudioTrack> track( | |
| 203 new WebRtcLocalAudioTrack(adapter.get())); | |
| 204 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()); | |
| 210 | |
| 211 std::unique_ptr<MockMediaStreamAudioSink> sink( | |
| 212 new MockMediaStreamAudioSink()); | |
| 213 base::WaitableEvent event(false, false); | |
| 214 EXPECT_CALL(*sink, FormatIsSet()); | |
| 215 EXPECT_CALL(*sink, | |
| 216 CaptureData()).Times(AtLeast(1)) | |
| 217 .WillRepeatedly(SignalEvent(&event)); | |
| 218 track->AddSink(sink.get()); | |
| 219 EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); | |
| 220 track->RemoveSink(sink.get()); | |
| 221 | |
| 222 EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return()); | |
| 223 capturer_->Stop(); | |
| 224 } | |
| 225 | |
| 226 // The same setup as ConnectAndDisconnectOneSink, but enable and disable the | |
| 227 // audio track on the fly. When the audio track is disabled, there is no data | |
| 228 // callback to the sink; when the audio track is enabled, there comes data | |
| 229 // callback. | |
| 230 // TODO(xians): Enable this test after resolving the racing issue that TSAN | |
| 231 // reports on MediaStreamTrack::enabled(); | |
| 232 TEST_F(WebRtcLocalAudioTrackTest, DISABLED_DisableEnableAudioTrack) { | |
| 233 EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); | |
| 234 EXPECT_CALL(*capturer_source_.get(), OnStart()); | |
| 235 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( | |
| 236 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
| 237 std::unique_ptr<WebRtcLocalAudioTrack> track( | |
| 238 new WebRtcLocalAudioTrack(adapter.get())); | |
| 239 track->Start( | |
| 240 base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo, | |
| 241 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(), | |
| 242 track.get())); | |
| 243 capturer_->AddTrack(track.get()); | |
| 244 EXPECT_TRUE(track->GetAudioAdapter()->enabled()); | |
| 245 EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(false)); | |
| 246 std::unique_ptr<MockMediaStreamAudioSink> sink( | |
| 247 new MockMediaStreamAudioSink()); | |
| 248 const media::AudioParameters params = capturer_->GetInputFormat(); | |
| 249 base::WaitableEvent event(false, false); | |
| 250 EXPECT_CALL(*sink, FormatIsSet()).Times(1); | |
| 251 EXPECT_CALL(*sink, CaptureData()).Times(0); | |
| 252 EXPECT_EQ(sink->audio_params().frames_per_buffer(), | |
| 253 params.sample_rate() / 100); | |
| 254 track->AddSink(sink.get()); | |
| 255 EXPECT_FALSE(event.TimedWait(TestTimeouts::tiny_timeout())); | |
| 256 | |
| 257 event.Reset(); | |
| 258 EXPECT_CALL(*sink, CaptureData()).Times(AtLeast(1)) | |
| 259 .WillRepeatedly(SignalEvent(&event)); | |
| 260 EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(true)); | |
| 261 EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); | |
| 262 track->RemoveSink(sink.get()); | |
| 263 | |
| 264 EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return()); | |
| 265 capturer_->Stop(); | |
| 266 track.reset(); | |
| 267 } | |
| 268 | |
| 269 // Create multiple audio tracks and enable/disable them, verify that the audio | |
| 270 // callbacks appear/disappear. | |
| 271 // Flaky due to a data race, see http://crbug.com/295418 | |
| 272 TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) { | |
| 273 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1( | |
| 274 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
| 275 std::unique_ptr<WebRtcLocalAudioTrack> track_1( | |
| 276 new WebRtcLocalAudioTrack(adapter_1.get())); | |
| 277 track_1->Start( | |
| 278 base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo, | |
| 279 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(), | |
| 280 track_1.get())); | |
| 281 capturer_->AddTrack(track_1.get()); | |
| 282 EXPECT_TRUE(track_1->GetAudioAdapter()->enabled()); | |
| 283 std::unique_ptr<MockMediaStreamAudioSink> sink_1( | |
| 284 new MockMediaStreamAudioSink()); | |
| 285 const media::AudioParameters params = capturer_->GetInputFormat(); | |
| 286 base::WaitableEvent event_1(false, false); | |
| 287 EXPECT_CALL(*sink_1, FormatIsSet()).WillOnce(Return()); | |
| 288 EXPECT_CALL(*sink_1, CaptureData()).Times(AtLeast(1)) | |
| 289 .WillRepeatedly(SignalEvent(&event_1)); | |
| 290 EXPECT_EQ(sink_1->audio_params().frames_per_buffer(), | |
| 291 params.sample_rate() / 100); | |
| 292 track_1->AddSink(sink_1.get()); | |
| 293 EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout())); | |
| 294 | |
| 295 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2( | |
| 296 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
| 297 std::unique_ptr<WebRtcLocalAudioTrack> track_2( | |
| 298 new WebRtcLocalAudioTrack(adapter_2.get())); | |
| 299 track_2->Start( | |
| 300 base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo, | |
| 301 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(), | |
| 302 track_2.get())); | |
| 303 capturer_->AddTrack(track_2.get()); | |
| 304 EXPECT_TRUE(track_2->GetAudioAdapter()->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 std::unique_ptr<MockMediaStreamAudioSink> sink_2( | |
| 311 new MockMediaStreamAudioSink()); | |
| 312 EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(Return()); | |
| 313 EXPECT_CALL(*sink_1, CaptureData()).Times(AtLeast(1)) | |
| 314 .WillRepeatedly(SignalEvent(&event_1)); | |
| 315 EXPECT_EQ(sink_1->audio_params().frames_per_buffer(), | |
| 316 params.sample_rate() / 100); | |
| 317 EXPECT_CALL(*sink_2, CaptureData()).Times(AtLeast(1)) | |
| 318 .WillRepeatedly(SignalEvent(&event_2)); | |
| 319 EXPECT_EQ(sink_2->audio_params().frames_per_buffer(), | |
| 320 params.sample_rate() / 100); | |
| 321 track_2->AddSink(sink_2.get()); | |
| 322 EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout())); | |
| 323 EXPECT_TRUE(event_2.TimedWait(TestTimeouts::tiny_timeout())); | |
| 324 | |
| 325 track_1->RemoveSink(sink_1.get()); | |
| 326 track_1->Stop(); | |
| 327 track_1.reset(); | |
| 328 | |
| 329 EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return()); | |
| 330 track_2->RemoveSink(sink_2.get()); | |
| 331 track_2->Stop(); | |
| 332 track_2.reset(); | |
| 333 } | |
| 334 | |
| 335 | |
| 336 // Start one track and verify the capturer is correctly starting its source. | |
| 337 // And it should be fine to not to call Stop() explicitly. | |
| 338 TEST_F(WebRtcLocalAudioTrackTest, StartOneAudioTrack) { | |
| 339 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( | |
| 340 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
| 341 std::unique_ptr<WebRtcLocalAudioTrack> track( | |
| 342 new WebRtcLocalAudioTrack(adapter.get())); | |
| 343 track->Start( | |
| 344 base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo, | |
| 345 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(), | |
| 346 track.get())); | |
| 347 capturer_->AddTrack(track.get()); | |
| 348 | |
| 349 // When the track goes away, it will automatically stop the | |
| 350 // |capturer_source_|. | |
| 351 EXPECT_CALL(*capturer_source_.get(), OnStop()); | |
| 352 track.reset(); | |
| 353 } | |
| 354 | |
| 355 // Start two tracks and verify the capturer is correctly starting its source. | |
| 356 // When the last track connected to the capturer is stopped, the source is | |
| 357 // stopped. | |
| 358 TEST_F(WebRtcLocalAudioTrackTest, StartTwoAudioTracks) { | |
| 359 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter1( | |
| 360 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
| 361 std::unique_ptr<WebRtcLocalAudioTrack> track1( | |
| 362 new WebRtcLocalAudioTrack(adapter1.get())); | |
| 363 track1->Start( | |
| 364 base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo, | |
| 365 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(), | |
| 366 track1.get())); | |
| 367 capturer_->AddTrack(track1.get()); | |
| 368 | |
| 369 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter2( | |
| 370 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
| 371 std::unique_ptr<WebRtcLocalAudioTrack> track2( | |
| 372 new WebRtcLocalAudioTrack(adapter2.get())); | |
| 373 track2->Start( | |
| 374 base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo, | |
| 375 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(), | |
| 376 track2.get())); | |
| 377 capturer_->AddTrack(track2.get()); | |
| 378 | |
| 379 track1->Stop(); | |
| 380 // When the last track is stopped, it will automatically stop the | |
| 381 // |capturer_source_|. | |
| 382 EXPECT_CALL(*capturer_source_.get(), OnStop()); | |
| 383 track2->Stop(); | |
| 384 } | |
| 385 | |
| 386 // Start/Stop tracks and verify the capturer is correctly starting/stopping | |
| 387 // its source. | |
| 388 TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) { | |
| 389 base::WaitableEvent event(false, false); | |
| 390 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1( | |
| 391 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
| 392 std::unique_ptr<WebRtcLocalAudioTrack> track_1( | |
| 393 new WebRtcLocalAudioTrack(adapter_1.get())); | |
| 394 track_1->Start( | |
| 395 base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo, | |
| 396 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(), | |
| 397 track_1.get())); | |
| 398 capturer_->AddTrack(track_1.get()); | |
| 399 | |
| 400 // Verify the data flow by connecting the sink to |track_1|. | |
| 401 std::unique_ptr<MockMediaStreamAudioSink> sink( | |
| 402 new MockMediaStreamAudioSink()); | |
| 403 event.Reset(); | |
| 404 EXPECT_CALL(*sink, FormatIsSet()).WillOnce(SignalEvent(&event)); | |
| 405 EXPECT_CALL(*sink, CaptureData()) | |
| 406 .Times(AnyNumber()).WillRepeatedly(Return()); | |
| 407 track_1->AddSink(sink.get()); | |
| 408 EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); | |
| 409 | |
| 410 // Start the second audio track will not start the |capturer_source_| | |
| 411 // since it has been started. | |
| 412 EXPECT_CALL(*capturer_source_.get(), OnStart()).Times(0); | |
| 413 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2( | |
| 414 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
| 415 std::unique_ptr<WebRtcLocalAudioTrack> track_2( | |
| 416 new WebRtcLocalAudioTrack(adapter_2.get())); | |
| 417 track_2->Start( | |
| 418 base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo, | |
| 419 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(), | |
| 420 track_2.get())); | |
| 421 capturer_->AddTrack(track_2.get()); | |
| 422 | |
| 423 // Stop the capturer will clear up the track lists in the capturer. | |
| 424 EXPECT_CALL(*capturer_source_.get(), OnStop()); | |
| 425 capturer_->Stop(); | |
| 426 | |
| 427 // Adding a new track to the capturer. | |
| 428 track_2->AddSink(sink.get()); | |
| 429 EXPECT_CALL(*sink, FormatIsSet()).Times(0); | |
| 430 | |
| 431 // Stop the capturer again will not trigger stopping the source of the | |
| 432 // capturer again.. | |
| 433 event.Reset(); | |
| 434 EXPECT_CALL(*capturer_source_.get(), OnStop()).Times(0); | |
| 435 capturer_->Stop(); | |
| 436 } | |
| 437 | |
| 438 // Create a new capturer with new source, connect it to a new audio track. | |
| 439 #if defined(THREAD_SANITIZER) | |
| 440 // Fails under TSan, see https://crbug.com/576634. | |
| 441 #define MAYBE_ConnectTracksToDifferentCapturers \ | |
| 442 DISABLED_ConnectTracksToDifferentCapturers | |
| 443 #else | |
| 444 #define MAYBE_ConnectTracksToDifferentCapturers \ | |
| 445 ConnectTracksToDifferentCapturers | |
| 446 #endif | |
| 447 TEST_F(WebRtcLocalAudioTrackTest, MAYBE_ConnectTracksToDifferentCapturers) { | |
| 448 // Setup the first audio track and start it. | |
| 449 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1( | |
| 450 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
| 451 std::unique_ptr<WebRtcLocalAudioTrack> track_1( | |
| 452 new WebRtcLocalAudioTrack(adapter_1.get())); | |
| 453 track_1->Start( | |
| 454 base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo, | |
| 455 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(), | |
| 456 track_1.get())); | |
| 457 capturer_->AddTrack(track_1.get()); | |
| 458 | |
| 459 // Verify the data flow by connecting the |sink_1| to |track_1|. | |
| 460 std::unique_ptr<MockMediaStreamAudioSink> sink_1( | |
| 461 new MockMediaStreamAudioSink()); | |
| 462 EXPECT_CALL(*sink_1.get(), CaptureData()) | |
| 463 .Times(AnyNumber()).WillRepeatedly(Return()); | |
| 464 EXPECT_CALL(*sink_1.get(), FormatIsSet()).Times(AnyNumber()); | |
| 465 track_1->AddSink(sink_1.get()); | |
| 466 | |
| 467 // Create a new capturer with new source with different audio format. | |
| 468 MockConstraintFactory constraint_factory; | |
| 469 StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE, | |
| 470 std::string(), std::string()); | |
| 471 std::unique_ptr<WebRtcAudioCapturer> new_capturer( | |
| 472 WebRtcAudioCapturer::CreateCapturer( | |
| 473 -1, device, constraint_factory.CreateWebMediaConstraints(), NULL, | |
| 474 NULL)); | |
| 475 scoped_refptr<MockCapturerSource> new_source( | |
| 476 new MockCapturerSource(new_capturer.get())); | |
| 477 EXPECT_CALL(*new_source.get(), OnInitialize(_, new_capturer.get(), -1)); | |
| 478 EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(true)); | |
| 479 EXPECT_CALL(*new_source.get(), OnStart()); | |
| 480 | |
| 481 media::AudioParameters new_param( | |
| 482 media::AudioParameters::AUDIO_PCM_LOW_LATENCY, | |
| 483 media::CHANNEL_LAYOUT_MONO, 44100, 16, 441); | |
| 484 new_capturer->SetCapturerSource(new_source, new_param); | |
| 485 | |
| 486 // Setup the second audio track, connect it to the new capturer and start it. | |
| 487 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2( | |
| 488 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
| 489 std::unique_ptr<WebRtcLocalAudioTrack> track_2( | |
| 490 new WebRtcLocalAudioTrack(adapter_2.get())); | |
| 491 track_2->Start( | |
| 492 base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo, | |
| 493 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(), | |
| 494 track_2.get())); | |
| 495 new_capturer->AddTrack(track_2.get()); | |
| 496 | |
| 497 // Verify the data flow by connecting the |sink_2| to |track_2|. | |
| 498 std::unique_ptr<MockMediaStreamAudioSink> sink_2( | |
| 499 new MockMediaStreamAudioSink()); | |
| 500 base::WaitableEvent event(false, false); | |
| 501 EXPECT_CALL(*sink_2, CaptureData()) | |
| 502 .Times(AnyNumber()).WillRepeatedly(Return()); | |
| 503 EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(SignalEvent(&event)); | |
| 504 track_2->AddSink(sink_2.get()); | |
| 505 EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); | |
| 506 | |
| 507 // Stopping the new source will stop the second track. | |
| 508 event.Reset(); | |
| 509 EXPECT_CALL(*new_source.get(), OnStop()) | |
| 510 .Times(1).WillOnce(SignalEvent(&event)); | |
| 511 new_capturer->Stop(); | |
| 512 EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); | |
| 513 | |
| 514 // Stop the capturer of the first audio track. | |
| 515 EXPECT_CALL(*capturer_source_.get(), OnStop()); | |
| 516 capturer_->Stop(); | |
| 517 } | |
| 518 | |
| 519 // Make sure a audio track can deliver packets with a buffer size smaller than | |
| 520 // 10ms when it is not connected with a peer connection. | |
| 521 TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) { | |
| 522 // Setup a capturer which works with a buffer size smaller than 10ms. | |
| 523 media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, | |
| 524 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128); | |
| 525 | |
| 526 // Create a capturer with new source which works with the format above. | |
| 527 MockConstraintFactory factory; | |
| 528 factory.DisableDefaultAudioConstraints(); | |
| 529 std::unique_ptr<WebRtcAudioCapturer> capturer( | |
| 530 WebRtcAudioCapturer::CreateCapturer( | |
| 531 -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", "", | |
| 532 params.sample_rate(), params.channel_layout(), | |
| 533 params.frames_per_buffer()), | |
| 534 factory.CreateWebMediaConstraints(), NULL, NULL)); | |
| 535 scoped_refptr<MockCapturerSource> source( | |
| 536 new MockCapturerSource(capturer.get())); | |
| 537 EXPECT_CALL(*source.get(), OnInitialize(_, capturer.get(), -1)); | |
| 538 EXPECT_CALL(*source.get(), SetAutomaticGainControl(true)); | |
| 539 EXPECT_CALL(*source.get(), OnStart()); | |
| 540 capturer->SetCapturerSource(source, params); | |
| 541 | |
| 542 // Setup a audio track, connect it to the capturer and start it. | |
| 543 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( | |
| 544 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
| 545 std::unique_ptr<WebRtcLocalAudioTrack> track( | |
| 546 new WebRtcLocalAudioTrack(adapter.get())); | |
| 547 track->Start( | |
| 548 base::Bind(&MediaStreamAudioSource::StopAudioDeliveryTo, | |
| 549 MediaStreamAudioSource::From(blink_source_)->GetWeakPtr(), | |
| 550 track.get())); | |
| 551 capturer->AddTrack(track.get()); | |
| 552 | |
| 553 // Verify the data flow by connecting the |sink| to |track|. | |
| 554 std::unique_ptr<MockMediaStreamAudioSink> sink( | |
| 555 new MockMediaStreamAudioSink()); | |
| 556 base::WaitableEvent event(false, false); | |
| 557 EXPECT_CALL(*sink, FormatIsSet()).Times(1); | |
| 558 // Verify the sinks are getting the packets with an expecting buffer size. | |
| 559 #if defined(OS_ANDROID) | |
| 560 const int expected_buffer_size = params.sample_rate() / 100; | |
| 561 #else | |
| 562 const int expected_buffer_size = params.frames_per_buffer(); | |
| 563 #endif | |
| 564 EXPECT_CALL(*sink, CaptureData()) | |
| 565 .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event)); | |
| 566 track->AddSink(sink.get()); | |
| 567 EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout())); | |
| 568 EXPECT_EQ(expected_buffer_size, sink->audio_params().frames_per_buffer()); | |
| 569 | |
| 570 // Stopping the new source will stop the second track. | |
| 571 EXPECT_CALL(*source.get(), OnStop()).Times(1); | |
| 572 capturer->Stop(); | |
| 573 | |
| 574 // Even though this test don't use |capturer_source_| it will be stopped | |
| 575 // during teardown of the test harness. | |
| 576 EXPECT_CALL(*capturer_source_.get(), OnStop()); | |
| 577 } | |
| 578 | |
| 579 } // namespace content | |
| OLD | NEW |
|---|
Chromium Code Reviews
Issue 1834323002:
MediaStream audio: Refactor 3 separate "glue" implementations into one. (Closed)
Base URL: https://chromium.googlesource.com/chromium/src.git@master