Update MediaStreamTrack::Stop to latest draft.

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 44 matching lines @@
   // same sink interface as MediaStreamAudioSink.
   void AddSink(PeerConnectionAudioSink* sink);
   void RemoveSink(PeerConnectionAudioSink* 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();
+  virtual void Stop() OVERRIDE;
 
   // Method called by the capturer to deliver the capture data.
   // Called on the capture audio thread.
   void Capture(const int16* audio_data,
                base::TimeDelta delay,
                int volume,
                bool key_pressed,
                bool need_audio_processing);
 
   // Method called by the capturer to set the audio parameters used by source
@@ skipping 42 matching lines @@
   // Used to calculate the signal level that shows in the UI.
   // Accessed on only the audio thread.
   scoped_ptr<MediaStreamAudioLevelCalculator> level_calculator_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_