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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/logging.h" | 7 #include "base/logging.h" |
8 #include "content/renderer/media/media_stream_audio_processor.h" | 8 #include "content/renderer/media/media_stream_audio_processor.h" |
9 #include "content/renderer/media/webrtc/webrtc_audio_sink_adapter.h" | 9 #include "content/renderer/media/webrtc/webrtc_audio_sink_adapter.h" |
10 #include "content/renderer/media/webrtc_local_audio_track.h" | 10 #include "content/renderer/media/webrtc_local_audio_track.h" |
11 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" | 11 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" |
12 | 12 |
13 namespace content { | 13 namespace content { |
14 | 14 |
15 static const char kAudioTrackKind[] = "audio"; | 15 static const char kAudioTrackKind[] = "audio"; |
16 | 16 |
17 scoped_refptr<WebRtcLocalAudioTrackAdapter> | 17 scoped_refptr<WebRtcLocalAudioTrackAdapter> |
18 WebRtcLocalAudioTrackAdapter::Create( | 18 WebRtcLocalAudioTrackAdapter::Create( |
19 const std::string& label, | 19 const std::string& label, |
20 webrtc::AudioSourceInterface* track_source) { | 20 webrtc::AudioSourceInterface* track_source) { |
21 talk_base::RefCountedObject<WebRtcLocalAudioTrackAdapter>* adapter = | 21 rtc::RefCountedObject<WebRtcLocalAudioTrackAdapter>* adapter = |
22 new talk_base::RefCountedObject<WebRtcLocalAudioTrackAdapter>( | 22 new rtc::RefCountedObject<WebRtcLocalAudioTrackAdapter>( |
23 label, track_source); | 23 label, track_source); |
24 return adapter; | 24 return adapter; |
25 } | 25 } |
26 | 26 |
27 WebRtcLocalAudioTrackAdapter::WebRtcLocalAudioTrackAdapter( | 27 WebRtcLocalAudioTrackAdapter::WebRtcLocalAudioTrackAdapter( |
28 const std::string& label, | 28 const std::string& label, |
29 webrtc::AudioSourceInterface* track_source) | 29 webrtc::AudioSourceInterface* track_source) |
30 : webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(label), | 30 : webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(label), |
31 owner_(NULL), | 31 owner_(NULL), |
32 track_source_(track_source), | 32 track_source_(track_source), |
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91 // false here in order not to overwrite the value from WebRTC. | 91 // false here in order not to overwrite the value from WebRTC. |
92 // TODO(xians): Remove this after we turn on the APM in Chrome by default. | 92 // TODO(xians): Remove this after we turn on the APM in Chrome by default. |
93 // http://crbug/365672 . | 93 // http://crbug/365672 . |
94 if (!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled()) | 94 if (!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled()) |
95 return false; | 95 return false; |
96 | 96 |
97 *level = signal_level_; | 97 *level = signal_level_; |
98 return true; | 98 return true; |
99 } | 99 } |
100 | 100 |
101 talk_base::scoped_refptr<webrtc::AudioProcessorInterface> | 101 rtc::scoped_refptr<webrtc::AudioProcessorInterface> |
102 WebRtcLocalAudioTrackAdapter::GetAudioProcessor() { | 102 WebRtcLocalAudioTrackAdapter::GetAudioProcessor() { |
103 base::AutoLock auto_lock(lock_); | 103 base::AutoLock auto_lock(lock_); |
104 return audio_processor_.get(); | 104 return audio_processor_.get(); |
105 } | 105 } |
106 | 106 |
107 std::vector<int> WebRtcLocalAudioTrackAdapter::VoeChannels() const { | 107 std::vector<int> WebRtcLocalAudioTrackAdapter::VoeChannels() const { |
108 base::AutoLock auto_lock(lock_); | 108 base::AutoLock auto_lock(lock_); |
109 return voe_channels_; | 109 return voe_channels_; |
110 } | 110 } |
111 | 111 |
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149 // When the audio track processing is disabled, WebRtcLocalAudioTrackAdapter | 149 // When the audio track processing is disabled, WebRtcLocalAudioTrackAdapter |
150 // is used to pass the channel ids to WebRtcAudioDeviceImpl, the data flow | 150 // is used to pass the channel ids to WebRtcAudioDeviceImpl, the data flow |
151 // becomes WebRtcAudioDeviceImpl ==> WebRTC. | 151 // becomes WebRtcAudioDeviceImpl ==> WebRTC. |
152 // TODO(xians): Only return NULL after the APM in WebRTC is deprecated. | 152 // TODO(xians): Only return NULL after the APM in WebRTC is deprecated. |
153 // See See http://crbug/365672 for details. | 153 // See See http://crbug/365672 for details. |
154 return MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled()? | 154 return MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled()? |
155 NULL : this; | 155 NULL : this; |
156 } | 156 } |
157 | 157 |
158 } // namespace content | 158 } // namespace content |
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