| OLD | NEW |
| 1 // Copyright 2014 The Chromium Authors. All rights reserved. | 1 // Copyright 2014 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/webrtc_local_audio_track_adapter.h" | 5 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" |
| 6 | 6 |
| 7 #include "base/location.h" | 7 #include "base/location.h" |
| 8 #include "base/logging.h" | 8 #include "base/logging.h" |
| 9 #include "content/renderer/media/media_stream_audio_processor.h" | 9 #include "content/renderer/media/media_stream_audio_processor.h" |
| 10 #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h" | 10 #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h" |
| 11 #include "content/renderer/media/webrtc/webrtc_audio_sink_adapter.h" | 11 #include "content/renderer/media/webrtc/webrtc_audio_sink_adapter.h" |
| 12 #include "content/renderer/media/webrtc_local_audio_track.h" | 12 #include "content/renderer/media/webrtc_local_audio_track.h" |
| 13 #include "content/renderer/render_thread_impl.h" | 13 #include "content/renderer/render_thread_impl.h" |
| 14 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" | 14 #include "third_party/webrtc/api/mediastreaminterface.h" |
| 15 | 15 |
| 16 namespace content { | 16 namespace content { |
| 17 | 17 |
| 18 static const char kAudioTrackKind[] = "audio"; | 18 static const char kAudioTrackKind[] = "audio"; |
| 19 | 19 |
| 20 scoped_refptr<WebRtcLocalAudioTrackAdapter> | 20 scoped_refptr<WebRtcLocalAudioTrackAdapter> |
| 21 WebRtcLocalAudioTrackAdapter::Create( | 21 WebRtcLocalAudioTrackAdapter::Create( |
| 22 const std::string& label, | 22 const std::string& label, |
| 23 webrtc::AudioSourceInterface* track_source) { | 23 webrtc::AudioSourceInterface* track_source) { |
| 24 // TODO(tommi): Change this so that the signaling thread is one of the | 24 // TODO(tommi): Change this so that the signaling thread is one of the |
| (...skipping 120 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 145 base::AutoLock auto_lock(lock_); | 145 base::AutoLock auto_lock(lock_); |
| 146 signal_level_ = signal_level; | 146 signal_level_ = signal_level; |
| 147 } | 147 } |
| 148 | 148 |
| 149 webrtc::AudioSourceInterface* WebRtcLocalAudioTrackAdapter::GetSource() const { | 149 webrtc::AudioSourceInterface* WebRtcLocalAudioTrackAdapter::GetSource() const { |
| 150 DCHECK(signaling_thread_checker_.CalledOnValidThread()); | 150 DCHECK(signaling_thread_checker_.CalledOnValidThread()); |
| 151 return track_source_; | 151 return track_source_; |
| 152 } | 152 } |
| 153 | 153 |
| 154 } // namespace content | 154 } // namespace content |
| OLD | NEW |