MediaStreamVideo*/VideoTrackAdapter and RTCVideoRenderer (small) cleanup

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 90 matching lines @@
 
   // The source of the audio track which is used by WebAudio, which provides
   // data to the audio track when hooking up with WebAudio.
   scoped_refptr<WebAudioCapturerSource> webaudio_source_;
 
   // A tagged list of sinks that the audio data is fed to. Tags
   // indicate tracks that need to be notified that the audio format
   // has changed.
   SinkList sinks_;
 
-  // Used to DCHECK that some methods are called on the main render thread.
-  base::ThreadChecker main_render_thread_checker_;
   // Tests that methods are called on libjingle's signaling thread.
   base::ThreadChecker signal_thread_checker_;
 
   // Used to DCHECK that some methods are called on the capture audio thread.
   base::ThreadChecker capture_thread_checker_;
 
   // Protects |params_| and |sinks_|.
   mutable base::Lock lock_;
 
   // Audio parameters of the audio capture stream.
   // Accessed on only the audio capture thread.
   media::AudioParameters audio_parameters_;
 
   // 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_