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

content/renderer/media/webrtc_local_audio_track.h

 // Copyright 2013 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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_
 
 #include <list>
 #include <string>
 
@@ skipping 26 matching lines @@
 
   // Add a sink to the track. This function will trigger a SetCaptureFormat()
   // call on the |sink|.
   // Called on the main render thread.
   void AddSink(WebRtcAudioCapturerSink* sink);
 
   // Remove a sink from the track.
   // Called on the main render thread.
   void RemoveSink(WebRtcAudioCapturerSink* sink);
 
+  // Starts the local audio track. Called on the main render thread and
+  // should be called only once when audio track is created.
+  void Start();
+
+  // Stops the local audio track. Called on the main render thread and
+  // should be called only once when audio track going away.
+  void Stop();
+
  protected:
   WebRtcLocalAudioTrack(const std::string& label,
                         const scoped_refptr<WebRtcAudioCapturer>& capturer,
                         webrtc::AudioSourceInterface* stream_source);
   virtual ~WebRtcLocalAudioTrack();
 
  private:
   typedef std::list<scoped_refptr<WebRtcAudioCapturerSinkOwner> > SinkList;
 
   // content::WebRtcAudioCapturerSink implementation.
@@ skipping 32 matching lines @@
 
   // Protects |params_| and |sinks_|.
   mutable base::Lock lock_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_