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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
3 // found in the LICENSE file.
4 
5 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
6 
7 #include "base/logging.h"
8 #include "content/renderer/media/media_stream_audio_processor.h"
9 #include "content/renderer/media/webrtc/webrtc_audio_sink_adapter.h"
10 #include "content/renderer/media/webrtc_local_audio_track.h"
11 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
12 
13 namespace content {
14 
15 static const char kAudioTrackKind[] = "audio";
16 
17 scoped_refptr<WebRtcLocalAudioTrackAdapter>
Create(const std::string & label,webrtc::AudioSourceInterface * track_source)18 WebRtcLocalAudioTrackAdapter::Create(
19     const std::string& label,
20     webrtc::AudioSourceInterface* track_source) {
21   rtc::RefCountedObject<WebRtcLocalAudioTrackAdapter>* adapter =
22       new rtc::RefCountedObject<WebRtcLocalAudioTrackAdapter>(
23           label, track_source);
24   return adapter;
25 }
26 
WebRtcLocalAudioTrackAdapter(const std::string & label,webrtc::AudioSourceInterface * track_source)27 WebRtcLocalAudioTrackAdapter::WebRtcLocalAudioTrackAdapter(
28     const std::string& label,
29     webrtc::AudioSourceInterface* track_source)
30     : webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(label),
31       owner_(NULL),
32       track_source_(track_source),
33       signal_level_(0) {
34 }
35 
~WebRtcLocalAudioTrackAdapter()36 WebRtcLocalAudioTrackAdapter::~WebRtcLocalAudioTrackAdapter() {
37 }
38 
Initialize(WebRtcLocalAudioTrack * owner)39 void WebRtcLocalAudioTrackAdapter::Initialize(WebRtcLocalAudioTrack* owner) {
40   DCHECK(!owner_);
41   DCHECK(owner);
42   owner_ = owner;
43 }
44 
SetAudioProcessor(const scoped_refptr<MediaStreamAudioProcessor> & processor)45 void WebRtcLocalAudioTrackAdapter::SetAudioProcessor(
46     const scoped_refptr<MediaStreamAudioProcessor>& processor) {
47   base::AutoLock auto_lock(lock_);
48   audio_processor_ = processor;
49 }
50 
kind() const51 std::string WebRtcLocalAudioTrackAdapter::kind() const {
52   return kAudioTrackKind;
53 }
54 
AddSink(webrtc::AudioTrackSinkInterface * sink)55 void WebRtcLocalAudioTrackAdapter::AddSink(
56     webrtc::AudioTrackSinkInterface* sink) {
57   DCHECK(sink);
58 #ifndef NDEBUG
59   // Verify that |sink| has not been added.
60   for (ScopedVector<WebRtcAudioSinkAdapter>::const_iterator it =
61            sink_adapters_.begin();
62        it != sink_adapters_.end(); ++it) {
63     DCHECK(!(*it)->IsEqual(sink));
64   }
65 #endif
66 
67   scoped_ptr<WebRtcAudioSinkAdapter> adapter(
68       new WebRtcAudioSinkAdapter(sink));
69   owner_->AddSink(adapter.get());
70   sink_adapters_.push_back(adapter.release());
71 }
72 
RemoveSink(webrtc::AudioTrackSinkInterface * sink)73 void WebRtcLocalAudioTrackAdapter::RemoveSink(
74     webrtc::AudioTrackSinkInterface* sink) {
75   DCHECK(sink);
76   for (ScopedVector<WebRtcAudioSinkAdapter>::iterator it =
77            sink_adapters_.begin();
78        it != sink_adapters_.end(); ++it) {
79     if ((*it)->IsEqual(sink)) {
80       owner_->RemoveSink(*it);
81       sink_adapters_.erase(it);
82       return;
83     }
84   }
85 }
86 
GetSignalLevel(int * level)87 bool WebRtcLocalAudioTrackAdapter::GetSignalLevel(int* level) {
88   base::AutoLock auto_lock(lock_);
89   // It is required to provide the signal level after audio processing. In
90   // case the audio processing is not enabled for the track, we return
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.
93   // http://crbug/365672 .
94   if (!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled())
95     return false;
96 
97   *level = signal_level_;
98   return true;
99 }
100 
101 rtc::scoped_refptr<webrtc::AudioProcessorInterface>
GetAudioProcessor()102 WebRtcLocalAudioTrackAdapter::GetAudioProcessor() {
103   base::AutoLock auto_lock(lock_);
104   return audio_processor_.get();
105 }
106 
VoeChannels() const107 std::vector<int> WebRtcLocalAudioTrackAdapter::VoeChannels() const {
108   base::AutoLock auto_lock(lock_);
109   return voe_channels_;
110 }
111 
SetSignalLevel(int signal_level)112 void WebRtcLocalAudioTrackAdapter::SetSignalLevel(int signal_level) {
113   base::AutoLock auto_lock(lock_);
114   signal_level_ = signal_level;
115 }
116 
AddChannel(int channel_id)117 void WebRtcLocalAudioTrackAdapter::AddChannel(int channel_id) {
118   DVLOG(1) << "WebRtcLocalAudioTrack::AddChannel(channel_id="
119            << channel_id << ")";
120   base::AutoLock auto_lock(lock_);
121   if (std::find(voe_channels_.begin(), voe_channels_.end(), channel_id) !=
122       voe_channels_.end()) {
123     // We need to handle the case when the same channel is connected to the
124     // track more than once.
125     return;
126   }
127 
128   voe_channels_.push_back(channel_id);
129 }
130 
RemoveChannel(int channel_id)131 void WebRtcLocalAudioTrackAdapter::RemoveChannel(int channel_id) {
132   DVLOG(1) << "WebRtcLocalAudioTrack::RemoveChannel(channel_id="
133            << channel_id << ")";
134   base::AutoLock auto_lock(lock_);
135   std::vector<int>::iterator iter =
136       std::find(voe_channels_.begin(), voe_channels_.end(), channel_id);
137   DCHECK(iter != voe_channels_.end());
138   voe_channels_.erase(iter);
139 }
140 
GetSource() const141 webrtc::AudioSourceInterface* WebRtcLocalAudioTrackAdapter::GetSource() const {
142   return track_source_;
143 }
144 
GetRenderer()145 cricket::AudioRenderer* WebRtcLocalAudioTrackAdapter::GetRenderer() {
146   // When the audio track processing is enabled, return a NULL so that capture
147   // data goes through Libjingle LocalAudioTrackHandler::LocalAudioSinkAdapter
148   // ==> WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer ==> WebRTC.
149   // When the audio track processing is disabled, WebRtcLocalAudioTrackAdapter
150   // is used to pass the channel ids to WebRtcAudioDeviceImpl, the data flow
151   // becomes WebRtcAudioDeviceImpl ==> WebRTC.
152   // TODO(xians): Only return NULL after the APM in WebRTC is deprecated.
153   // See See http://crbug/365672 for details.
154   return MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled()?
155       NULL : this;
156 }
157 
158 }  // namespace content
159