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

content/renderer/media/media_stream_audio_processor.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_MEDIA_STREAM_AUDIO_PROCESSOR_H_
 #define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_
 
 #include "base/atomicops.h"
 #include "base/files/file.h"
 #include "base/gtest_prod_util.h"
@@ skipping 103 matching lines @@
 
  private:
   friend class MediaStreamAudioProcessorTest;
 
   FRIEND_TEST_ALL_PREFIXES(MediaStreamAudioProcessorTest,
                            GetAecDumpMessageFilter);
 
   // WebRtcPlayoutDataSource::Sink implementation.
   void OnPlayoutData(media::AudioBus* audio_bus,
                      int sample_rate,
-                     int audio_delay_milliseconds) override;
+                     int audio_delay_milliseconds,
+                     uint32_t skipped_frames) override;
   void OnPlayoutDataSourceChanged() override;
 
   // webrtc::AudioProcessorInterface implementation.
   // This method is called on the libjingle thread.
   void GetStats(AudioProcessorStats* stats) override;
 
   // Helper to initialize the WebRtc AudioProcessing.
   void InitializeAudioProcessingModule(
       const blink::WebMediaConstraints& constraints,
       const MediaStreamDevice::AudioDeviceParameters& input_params);
@@ skipping 71 matching lines @@
   // Object for logging echo information when the AEC is enabled. Accessible by
   // the libjingle thread through GetStats().
   scoped_ptr<EchoInformation> echo_information_;
 
   DISALLOW_COPY_AND_ASSIGN(MediaStreamAudioProcessor);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_