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

content/renderer/media/webrtc_audio_renderer.cc

Index: content/renderer/media/webrtc_audio_renderer.cc
diff --git a/content/renderer/media/webrtc_audio_renderer.cc b/content/renderer/media/webrtc_audio_renderer.cc
index 9714b237abb44442f927dc712cb4b43f9362bf0b..b47b4f90106432cc654c8bd5cc7f2b1c93d756e4 100644
--- a/content/renderer/media/webrtc_audio_renderer.cc
+++ b/content/renderer/media/webrtc_audio_renderer.cc
@@ -185,6 +185,7 @@ WebRtcAudioRenderer::WebRtcAudioRenderer(
       start_ref_count_(0),
       audio_delay_milliseconds_(0),
       fifo_delay_milliseconds_(0),
+      skipped_frames_(0),
       sink_params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                    media::CHANNEL_LAYOUT_STEREO,
                    0,
@@ -430,6 +431,7 @@ int WebRtcAudioRenderer::Render(media::AudioBus* audio_bus,
 
   DCHECK_LE(audio_delay_milliseconds, static_cast<uint32_t>(INT_MAX));
   audio_delay_milliseconds_ = static_cast<int>(audio_delay_milliseconds);
+  skipped_frames_ += frames_skipped;
 
   if (audio_fifo_)
     audio_fifo_->Consume(audio_bus, audio_bus->frames());
@@ -461,7 +463,9 @@ void WebRtcAudioRenderer::SourceCallback(
   // otherwise the data will be buffered up inside |source_|.
   source_->RenderData(audio_bus, sink_params_.sample_rate(),
                       output_delay_milliseconds,
+                      skipped_frames_,
                       &current_time_);
+  skipped_frames_ = 0;
 
   // Avoid filling up the audio bus if we are not playing; instead
   // return here and ensure that the returned value in Render() is 0.