Turn audio ducking on by default on Windows again.

content/renderer/media/webrtc_local_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_LOCAL_AUDIO_RENDERER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_
 
 #include <vector>
 
 #include "base/callback.h"
@@ skipping 117 matching lines @@
 
   // Stores last time a render callback was received. The time difference
   // between a new time stamp and this value can be used to derive the
   // total render time.
   base::TimeTicks last_render_time_;
 
   // Keeps track of total time audio has been rendered.
   base::TimeDelta total_render_time_;
 
   // The audio parameters of the capture source.
+  // Must only be touched on the main thread.

[no longer working on chromium, 2014/07/08 12:38:55]

thanks.

   media::AudioParameters source_params_;
 
   // The audio parameters used by the sink.
+  // Must only be touched on the main thread.
   media::AudioParameters sink_params_;
 
   // Set when playing, cleared when paused.
   bool playing_;
 
   // Protects |loopback_fifo_|, |playing_| and |sink_|.
   mutable base::Lock thread_lock_;
 
   // The preferred buffer size provided via the ctor.
   const int frames_per_buffer_;
@@ skipping 10 matching lines @@
 
   // Used to DCHECK that some methods are called on the capture audio thread.
   base::ThreadChecker capture_thread_checker_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioRenderer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_