start/stop the source of the capturer when 1st audiotrack/last audiotrack is added/removed

content/renderer/media/webrtc_local_audio_track.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/webrtc_local_audio_track.h"
 
 #include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_audio_capturer_sink_owner.h"
 
 namespace content {
@@ skipping 11 matching lines @@
 }
 
 WebRtcLocalAudioTrack::WebRtcLocalAudioTrack(
     const std::string& label,
     const scoped_refptr<WebRtcAudioCapturer>& capturer,
     webrtc::AudioSourceInterface* track_source)
     : webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(label),
       capturer_(capturer),
       track_source_(track_source) {
   DCHECK(capturer.get());
-  capturer_->AddSink(this);
   DVLOG(1) << "WebRtcLocalAudioTrack::WebRtcLocalAudioTrack()";
 }
 
 WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack() {
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(sinks_.empty());
-  capturer_->RemoveSink(this);
   DVLOG(1) << "WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack()";
+
+  // Users might not call Stop() on the track.
+  if (capturer_)
+    Stop();
 }
 
 // Content::WebRtcAudioCapturerSink implementation.
 void WebRtcLocalAudioTrack::CaptureData(const int16* audio_data,
                                         int number_of_channels,
                                         int number_of_frames,
                                         int audio_delay_milliseconds,
                                         double volume) {
   SinkList sinks;
   {
@@ skipping 66 matching lines @@
       WebRtcAudioCapturerSinkOwner::WrapsSink(sink));
   if (it != sinks_.end()) {
     // Clear the delegate to ensure that no more capture callbacks will
     // be sent to this sink. Also avoids a possible crash which can happen
     // if this method is called while capturing is active.
     (*it)->Reset();
     sinks_.erase(it);
   }
 }
 
+void WebRtcLocalAudioTrack::Start() {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  DVLOG(1) << "WebRtcLocalAudioTrack::Start()";
+  if (capturer_)
+    capturer_->AddSink(this);
+}
+
+void WebRtcLocalAudioTrack::Stop() {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  DVLOG(1) << "WebRtcLocalAudioTrack::Stop()";
+  if (capturer_) {
+    capturer_->RemoveSink(this);
+    capturer_ = NULL;
+  }
+}
+
 }  // namespace content