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

Index: content/renderer/media/media_stream_audio_processor.h
diff --git a/content/renderer/media/media_stream_audio_processor.h b/content/renderer/media/media_stream_audio_processor.h
index f5d5781f3a68d2e79b35d35da98be764b5947c88..295e51354627d8c4981babf2100e8623f4b2a1a1 100644
--- a/content/renderer/media/media_stream_audio_processor.h
+++ b/content/renderer/media/media_stream_audio_processor.h
@@ -121,7 +121,8 @@ class CONTENT_EXPORT MediaStreamAudioProcessor :
   // 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.
@@ -207,6 +208,10 @@ class CONTENT_EXPORT MediaStreamAudioProcessor :
   // the libjingle thread through GetStats().
   scoped_ptr<EchoInformation> echo_information_;
 
+  // Stores the number of skipped output frames. The APM only accepts 10 ms
+  // chunks of data, so we ned to store it until it reaches that amount.
+  uint32_t skipped_output_frames_;
+
   DISALLOW_COPY_AND_ASSIGN(MediaStreamAudioProcessor);
 };