Fix data race in WebRtcAudioRenderer.

content/renderer/media/webrtc_audio_renderer.h

 // 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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
 
 #include <stdint.h>
 
 #include <map>
@@ skipping 195 matching lines @@
   scoped_refptr<media::AudioRendererSink> sink_;
 
   // The media stream that holds the audio tracks that this renderer renders.
   const blink::WebMediaStream media_stream_;
 
   // Audio data source from the browser process.
   WebRtcAudioRendererSource* source_;
 
   // Protects access to |state_|, |source_|, |audio_fifo_|,
   // |audio_delay_milliseconds_|, |fifo_delay_milliseconds_|, |current_time_|,
-  // |sink_params_| and |render_callback_count_|
+  // |sink_params_|, |render_callback_count_| and |max_render_time_|.
   mutable base::Lock lock_;
 
   // Ref count for the MediaPlayers which are playing audio.
   int play_ref_count_;
 
   // Ref count for the MediaPlayers which have called Start() but not Stop().
   int start_ref_count_;
 
   // Used to buffer data between the client and the output device in cases where
   // the client buffer size is not the same as the output device buffer size.
@@ skipping 26 matching lines @@
   // Stores the maximum time spent waiting for render data from the source. Used
   // for logging UMA data. Logged and reset when Stop() is called.
   base::TimeDelta max_render_time_;
 
   DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_