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

content/renderer/media/webrtc_local_audio_track.cc

Index: content/renderer/media/webrtc_local_audio_track.cc
diff --git a/content/renderer/media/webrtc_local_audio_track.cc b/content/renderer/media/webrtc_local_audio_track.cc
index 0aa63658c516256a90b33696437143d775199f2b..dc54c00450899d837384c0db94020e4fd3ffcbc5 100644
--- a/content/renderer/media/webrtc_local_audio_track.cc
+++ b/content/renderer/media/webrtc_local_audio_track.cc
@@ -29,15 +29,17 @@ WebRtcLocalAudioTrack::WebRtcLocalAudioTrack(
       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.
@@ -124,4 +126,20 @@ void WebRtcLocalAudioTrack::RemoveSink(
   }
 }
 
+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