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1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. | 1 // Copyright (c) 2012 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 "content/renderer/media/webrtc_audio_renderer.h" | 5 #include "content/renderer/media/webrtc_audio_renderer.h" |
6 | 6 |
7 #include <utility> | 7 #include <utility> |
8 | 8 |
9 #include "base/logging.h" | 9 #include "base/logging.h" |
10 #include "base/metrics/histogram.h" | 10 #include "base/metrics/histogram.h" |
11 #include "base/strings/string_util.h" | 11 #include "base/strings/string_util.h" |
12 #include "base/strings/stringprintf.h" | 12 #include "base/strings/stringprintf.h" |
13 #include "build/build_config.h" | 13 #include "build/build_config.h" |
14 #include "content/renderer/media/audio_device_factory.h" | |
15 #include "content/renderer/media/media_stream_audio_track.h" | 14 #include "content/renderer/media/media_stream_audio_track.h" |
16 #include "content/renderer/media/media_stream_dispatcher.h" | 15 #include "content/renderer/media/media_stream_dispatcher.h" |
17 #include "content/renderer/media/media_stream_track.h" | 16 #include "content/renderer/media/media_stream_track.h" |
| 17 #include "content/renderer/media/restartable_audio_output_device_factory.h" |
18 #include "content/renderer/media/webrtc_audio_device_impl.h" | 18 #include "content/renderer/media/webrtc_audio_device_impl.h" |
19 #include "content/renderer/media/webrtc_logging.h" | 19 #include "content/renderer/media/webrtc_logging.h" |
20 #include "content/renderer/render_frame_impl.h" | 20 #include "content/renderer/render_frame_impl.h" |
21 #include "media/audio/audio_output_device.h" | |
22 #include "media/audio/audio_parameters.h" | 21 #include "media/audio/audio_parameters.h" |
23 #include "media/audio/sample_rates.h" | 22 #include "media/audio/sample_rates.h" |
| 23 #include "media/base/restartable_audio_output_device.h" |
24 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" | 24 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" |
25 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" | 25 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" |
26 #include "third_party/libjingle/source/talk/media/base/audiorenderer.h" | 26 #include "third_party/libjingle/source/talk/media/base/audiorenderer.h" |
27 | 27 |
28 #if defined(OS_WIN) | 28 #if defined(OS_WIN) |
29 #include "base/win/windows_version.h" | 29 #include "base/win/windows_version.h" |
30 #include "media/audio/win/core_audio_util_win.h" | 30 #include "media/audio/win/core_audio_util_win.h" |
31 #endif | 31 #endif |
32 | 32 |
33 namespace content { | 33 namespace content { |
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214 DCHECK(thread_checker_.CalledOnValidThread()); | 214 DCHECK(thread_checker_.CalledOnValidThread()); |
215 DCHECK(source); | 215 DCHECK(source); |
216 DCHECK(!sink_.get()); | 216 DCHECK(!sink_.get()); |
217 DCHECK_GE(session_id_, 0); | 217 DCHECK_GE(session_id_, 0); |
218 { | 218 { |
219 base::AutoLock auto_lock(lock_); | 219 base::AutoLock auto_lock(lock_); |
220 DCHECK_EQ(state_, UNINITIALIZED); | 220 DCHECK_EQ(state_, UNINITIALIZED); |
221 DCHECK(!source_); | 221 DCHECK(!source_); |
222 } | 222 } |
223 | 223 |
224 sink_ = | 224 sink_ = RestartableAudioOutputDeviceFactory::NewOutputDevice( |
225 AudioDeviceFactory::NewOutputDevice(source_render_frame_id_, session_id_, | 225 RestartableAudioOutputDeviceFactory::kSourceWebRTC, |
226 output_device_id_, security_origin_); | 226 source_render_frame_id_, session_id_, output_device_id_, |
| 227 security_origin_); |
| 228 |
227 if (sink_->GetDeviceStatus() != media::OUTPUT_DEVICE_STATUS_OK) | 229 if (sink_->GetDeviceStatus() != media::OUTPUT_DEVICE_STATUS_OK) |
228 return false; | 230 return false; |
229 | 231 |
230 PrepareSink(); | 232 PrepareSink(); |
231 { | 233 { |
232 // No need to reassert the preconditions because the other thread accessing | 234 // No need to reassert the preconditions because the other thread accessing |
233 // the fields (checked by |audio_renderer_thread_checker_|) only reads them. | 235 // the fields (checked by |audio_renderer_thread_checker_|) only reads them. |
234 base::AutoLock auto_lock(lock_); | 236 base::AutoLock auto_lock(lock_); |
235 source_ = source; | 237 source_ = source; |
236 | 238 |
237 // User must call Play() before any audio can be heard. | 239 // User must call Play() before any audio can be heard. |
238 state_ = PAUSED; | 240 state_ = PAUSED; |
239 } | 241 } |
240 sink_->Start(); | 242 sink_->Start(); |
| 243 sink_->Play(); // RestartableAudioOutputDevice does not play on start. |
241 | 244 |
242 return true; | 245 return true; |
243 } | 246 } |
244 | 247 |
245 scoped_refptr<MediaStreamAudioRenderer> | 248 scoped_refptr<MediaStreamAudioRenderer> |
246 WebRtcAudioRenderer::CreateSharedAudioRendererProxy( | 249 WebRtcAudioRenderer::CreateSharedAudioRendererProxy( |
247 const blink::WebMediaStream& media_stream) { | 250 const blink::WebMediaStream& media_stream) { |
248 content::SharedAudioRenderer::OnPlayStateChanged on_play_state_changed = | 251 content::SharedAudioRenderer::OnPlayStateChanged on_play_state_changed = |
249 base::Bind(&WebRtcAudioRenderer::OnPlayStateChanged, this); | 252 base::Bind(&WebRtcAudioRenderer::OnPlayStateChanged, this); |
250 return new SharedAudioRenderer(this, media_stream, on_play_state_changed); | 253 return new SharedAudioRenderer(this, media_stream, on_play_state_changed); |
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373 const media::SwitchOutputDeviceCB& callback) { | 376 const media::SwitchOutputDeviceCB& callback) { |
374 DVLOG(1) << "WebRtcAudioRenderer::SwitchOutputDevice()"; | 377 DVLOG(1) << "WebRtcAudioRenderer::SwitchOutputDevice()"; |
375 DCHECK(thread_checker_.CalledOnValidThread()); | 378 DCHECK(thread_checker_.CalledOnValidThread()); |
376 DCHECK_GE(session_id_, 0); | 379 DCHECK_GE(session_id_, 0); |
377 { | 380 { |
378 base::AutoLock auto_lock(lock_); | 381 base::AutoLock auto_lock(lock_); |
379 DCHECK(source_); | 382 DCHECK(source_); |
380 DCHECK_NE(state_, UNINITIALIZED); | 383 DCHECK_NE(state_, UNINITIALIZED); |
381 } | 384 } |
382 | 385 |
383 scoped_refptr<media::AudioOutputDevice> new_sink = | 386 scoped_refptr<media::RestartableAudioOutputDevice> new_sink = |
384 AudioDeviceFactory::NewOutputDevice(source_render_frame_id_, session_id_, | 387 RestartableAudioOutputDeviceFactory::NewOutputDevice( |
385 device_id, security_origin); | 388 RestartableAudioOutputDeviceFactory::kSourceWebRTC, |
| 389 source_render_frame_id_, session_id_, device_id, security_origin); |
386 if (new_sink->GetDeviceStatus() != media::OUTPUT_DEVICE_STATUS_OK) { | 390 if (new_sink->GetDeviceStatus() != media::OUTPUT_DEVICE_STATUS_OK) { |
387 callback.Run(new_sink->GetDeviceStatus()); | 391 callback.Run(new_sink->GetDeviceStatus()); |
388 return; | 392 return; |
389 } | 393 } |
390 | 394 |
391 // Make sure to stop the sink while _not_ holding the lock since the Render() | 395 // Make sure to stop the sink while _not_ holding the lock since the Render() |
392 // callback may currently be executing and trying to grab the lock while we're | 396 // callback may currently be executing and trying to grab the lock while we're |
393 // stopping the thread on which it runs. | 397 // stopping the thread on which it runs. |
394 sink_->Stop(); | 398 sink_->Stop(); |
395 audio_renderer_thread_checker_.DetachFromThread(); | 399 audio_renderer_thread_checker_.DetachFromThread(); |
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663 base::Bind(&WebRtcAudioRenderer::SourceCallback, | 667 base::Bind(&WebRtcAudioRenderer::SourceCallback, |
664 base::Unretained(this)))); | 668 base::Unretained(this)))); |
665 } | 669 } |
666 sink_params_ = new_sink_params; | 670 sink_params_ = new_sink_params; |
667 } | 671 } |
668 | 672 |
669 sink_->Initialize(new_sink_params, this); | 673 sink_->Initialize(new_sink_params, this); |
670 } | 674 } |
671 | 675 |
672 } // namespace content | 676 } // namespace content |
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