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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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