Forward the number of skipped frames by the OS in audio playout.

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 <string>
 #include <vector>
 
@@ skipping 79 matching lines @@
   // Called on the AudioInputDevice worker thread.
   void OnData(const media::AudioBus& audio_bus,
               base::TimeTicks estimated_capture_time) override;
 
   // Called on the AudioInputDevice worker thread.
   void OnSetFormat(const media::AudioParameters& params) override;
 
   // media::AudioRendererSink::RenderCallback implementation.
   // Render() is called on the AudioOutputDevice thread and OnRenderError()
   // on the IO thread.
-  int Render(media::AudioBus* audio_bus, int audio_delay_milliseconds) override;
+  int Render(media::AudioBus* audio_bus, int audio_delay_milliseconds,
+             uint32_t frames_skipped) override;
   void OnRenderError() override;
 
   // Initializes and starts the |sink_| if
   //  we have received valid |source_params_| &&
   //  |playing_| has been set to true &&
   //  |volume_| is not zero.
   void MaybeStartSink();
 
   // Sets new |source_params_| and then re-initializes and restarts |sink_|.
   void ReconfigureSink(const media::AudioParameters& params);
@@ skipping 58 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_