Drop WebRTC audio data if OS has skipped frames.

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 185 matching lines @@
                           PlayingState* state);
 
   // Called whenever the Play/Pause state changes of any of the renderers
   // or if the volume of any of them is changed.
   // Here we update the shared Play state and apply volume scaling to all audio
   // sources associated with the |media_stream| based on the collective volume
   // of playing renderers.
   void OnPlayStateChanged(const blink::WebMediaStream& media_stream,
                           PlayingState* state);
 
-  // Updates |sink_params_|, |audio_fifo_| and |fifo_delay_milliseconds_| based
-  // on |sink_|, and initializes |sink_|.
+  // Updates |sink_params_| and |audio_fifo_| based on |sink_|, and initializes
+  // |sink_|.
   void PrepareSink();
 
   // The RenderFrame in which the audio is rendered into |sink_|.
   const int source_render_frame_id_;
   const int session_id_;
 
   const scoped_refptr<base::SingleThreadTaskRunner> signaling_thread_;
 
   // The sink (destination) for rendered audio.
   scoped_refptr<media::AudioOutputDevice> sink_;
@@ skipping 16 matching lines @@
   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.
   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_;
-
   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_