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Side by Side Diff: content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc

Issue 429113002: Webrtc deps roll. (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: Use webrtc version 6825 and rebase and switch back to original workspace. Created 6 years, 4 months ago
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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),
(...skipping 58 matching lines...) Expand 10 before | Expand all | Expand 10 after
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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