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

 // 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.
 
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 
 #include "base/bind.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
 #include "base/win/windows_version.h"
@@ skipping 46 matching lines @@
   int ret = base::subtle::Barrier_AtomicIncrement(&ref_count_, -1);
   if (ret == 0) {
     delete this;
   }
   return ret;
 }
 
 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());
   {
     base::AutoLock auto_lock(lock_);
     if (!playing_) {
       // Force silence to AudioBus after stopping playout in case
       // there is lingering audio data in AudioBus.
       audio_bus->Zero();
       return;
     }
@@ skipping 36 matching lines @@
   // De-interleave each channel and convert to 32-bit floating-point
   // with nominal range -1.0 -> +1.0 to match the callback format.
   audio_bus->FromInterleaved(&render_buffer_[0],
                              audio_bus->frames(),
                              bytes_per_sample);
 
   // Pass the render data to the playout sinks.
   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);
   }
 }
 
 void WebRtcAudioDeviceImpl::RemoveAudioRenderer(WebRtcAudioRenderer* renderer) {
   DCHECK(main_thread_checker_.CalledOnValidThread());
   base::AutoLock auto_lock(lock_);
   DCHECK_EQ(renderer, renderer_.get());
   // Notify the playout sink of the change.
   for (PlayoutDataSinkList::const_iterator it = playout_sinks_.begin();
        it != playout_sinks_.end(); ++it) {
@@ skipping 375 matching lines @@
   // If there is no capturer or there are more than one open capture devices,
   // return false.
   if (capturers_.size() != 1)
     return false;
 
   return capturers_.back()->GetPairedOutputParameters(
       session_id, output_sample_rate, output_frames_per_buffer);
 }
 
 }  // namespace content