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

content/renderer/media/webrtc_audio_device_impl.h

 // Copyright 2013 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_DEVICE_IMPL_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 
 #include <string>
 #include <vector>
 
@@ skipping 174 matching lines @@
 class WebRtcAudioRenderer;
 
 // TODO(xians): Move the following two interfaces to webrtc so that
 // libjingle can own references to the renderer and capturer.
 class WebRtcAudioRendererSource {
  public:
   // Callback to get the rendered data.
   virtual void RenderData(media::AudioBus* audio_bus,
                           int sample_rate,
                           int audio_delay_milliseconds,
+                          uint32_t skipped_frames,
                           base::TimeDelta* current_time) = 0;
 
   // Callback to notify the client that the renderer is going away.
   virtual void RemoveAudioRenderer(WebRtcAudioRenderer* renderer) = 0;
 
   // Callback to notify the client that the audio renderer thread stopped.
   // This function must be called only when that thread is actually stopped.
   // Otherwise a race may occur.
   virtual void AudioRendererThreadStopped() = 0;
 
  protected:
   virtual ~WebRtcAudioRendererSource() {}
 };
 
 // TODO(xians): Merge this interface with WebRtcAudioRendererSource.
 // The reason why we could not do it today is that WebRtcAudioRendererSource
 // gets the data by pulling, while the data is pushed into
 // WebRtcPlayoutDataSource::Sink.
 class WebRtcPlayoutDataSource {
  public:
   class Sink {
    public:
     // Callback to get the playout data.
     // Called on the render audio thread.
     virtual void OnPlayoutData(media::AudioBus* audio_bus,
                                int sample_rate,
-                               int audio_delay_milliseconds) = 0;
+                               int audio_delay_milliseconds,
+                               uint32_t skipped_frames) = 0;
 
     // Callback to notify the sink that the source has changed.
     // Called on the main render thread.
     virtual void OnPlayoutDataSourceChanged() = 0;
 
    protected:
     virtual ~Sink() {}
   };
 
   // Adds/Removes the sink of WebRtcAudioRendererSource to the ADM.
@@ skipping 103 matching lines @@
 
   // Make destructor private to ensure that we can only be deleted by Release().
   ~WebRtcAudioDeviceImpl() override;
 
   // WebRtcAudioRendererSource implementation.
 
   // Called on the AudioOutputDevice worker thread.
   void RenderData(media::AudioBus* audio_bus,
                   int sample_rate,
                   int audio_delay_milliseconds,
+                  uint32_t skipped_frames,
                   base::TimeDelta* current_time) override;
 
   // Called on the main render thread.
   void RemoveAudioRenderer(WebRtcAudioRenderer* renderer) override;
   void AudioRendererThreadStopped() override;
 
   // WebRtcPlayoutDataSource implementation.
   void AddPlayoutSink(WebRtcPlayoutDataSource::Sink* sink) override;
   void RemovePlayoutSink(WebRtcPlayoutDataSource::Sink* sink) override;
 
@@ skipping 48 matching lines @@
   // Buffer used for temporary storage during render callback.
   // It is only accessed by the audio render thread.
   std::vector<int16> render_buffer_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_