Feed the WebRTC APM with empty far-end frames for the number of frames skipped by OS.

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 <map>
 #include <string>
 #include <vector>
@@ skipping 222 matching lines @@
   // the client buffer size is not the same as the output device buffer size.
   scoped_ptr<media::AudioPullFifo> audio_fifo_;
 
   // Contains the accumulated delay estimate which is provided to the WebRTC
   // AEC.
   int audio_delay_milliseconds_;
 
   // Delay due to the FIFO in milliseconds.
   int fifo_delay_milliseconds_;
 
+  // Contains the accumulated number of skipped frames which is provided to the
+  // WebRTC AEC. Accessed on audio render thread.
+  uint32_t skipped_frames_;
+
   base::TimeDelta current_time_;
 
   // Saved volume and playing state of the root renderer.
   PlayingState playing_state_;
 
   // Audio params used by the sink of the renderer.
   media::AudioParameters sink_params_;
 
   // The preferred device id of the output device or empty for the default
   // output device. Can change as a result of a SetSinkId() call.
@@ skipping 10 matching lines @@
   // Used for triggering new UMA histogram. Counts number of render
   // callbacks modulo |kNumCallbacksBetweenRenderTimeHistograms|.
   int render_callback_count_;
 
   DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_