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

content/renderer/media/webrtc_audio_device_impl.cc

Index: content/renderer/media/webrtc_audio_device_impl.cc
diff --git a/content/renderer/media/webrtc_audio_device_impl.cc b/content/renderer/media/webrtc_audio_device_impl.cc
index 8931f75e40aa51a73d4b5078e8b26e3083873fd4..de5bc7dd20d6acaf250a55e28918ba4741dbaffc 100644
--- a/content/renderer/media/webrtc_audio_device_impl.cc
+++ b/content/renderer/media/webrtc_audio_device_impl.cc
@@ -64,6 +64,7 @@ int32_t WebRtcAudioDeviceImpl::Release() const {
 void WebRtcAudioDeviceImpl::RenderData(media::AudioBus* audio_bus,
                                        int sample_rate,
                                        int audio_delay_milliseconds,
+                                       uint32_t skipped_frames,
                                        base::TimeDelta* current_time) {
   DCHECK(audio_renderer_thread_checker_.CalledOnValidThread());
   {
@@ -120,7 +121,8 @@ void WebRtcAudioDeviceImpl::RenderData(media::AudioBus* audio_bus,
   base::AutoLock auto_lock(lock_);
   for (PlayoutDataSinkList::const_iterator it = playout_sinks_.begin();
        it != playout_sinks_.end(); ++it) {
-    (*it)->OnPlayoutData(audio_bus, sample_rate, audio_delay_milliseconds);
+    (*it)->OnPlayoutData(audio_bus, sample_rate, audio_delay_milliseconds,
+                         skipped_frames);
   }
 }