Forward the number of skipped frames by the OS in audio playout.

content/renderer/media/webrtc_audio_renderer.cc

 // Copyright (c) 2012 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_renderer.h"
 
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
 #include "base/strings/stringprintf.h"
@@ skipping 399 matching lines @@
 
 media::OutputDeviceStatus WebRtcAudioRenderer::GetDeviceStatus() {
   DCHECK(thread_checker_.CalledOnValidThread());
   if (!sink_.get())
     return media::OUTPUT_DEVICE_STATUS_ERROR_INTERNAL;
 
   return sink_->GetDeviceStatus();
 }
 
 int WebRtcAudioRenderer::Render(media::AudioBus* audio_bus,
-                                int audio_delay_milliseconds) {
+                                uint32_t audio_delay_milliseconds,
+                                uint32_t frames_skipped) {
   DCHECK(audio_renderer_thread_checker_.CalledOnValidThread());
   base::AutoLock auto_lock(lock_);
   if (!source_)
     return 0;
 
   DVLOG(2) << "WebRtcAudioRenderer::Render()";
   DVLOG(2) << "audio_delay_milliseconds: " << audio_delay_milliseconds;
 
-  audio_delay_milliseconds_ = audio_delay_milliseconds;
+  DCHECK_LE(audio_delay_milliseconds, static_cast<uint32_t>(INT_MAX));
+  audio_delay_milliseconds_ = static_cast<int>(audio_delay_milliseconds);
 
   if (audio_fifo_)
     audio_fifo_->Consume(audio_bus, audio_bus->frames());
   else
     SourceCallback(0, audio_bus);
 
   return (state_ == PLAYING) ? audio_bus->frames() : 0;
 }
 
 void WebRtcAudioRenderer::OnRenderError() {
@@ skipping 208 matching lines @@
     sink_params_ = new_sink_params;
     fifo_delay_milliseconds_ = new_fifo_delay_milliseconds;
     if (new_audio_fifo.get())
       audio_fifo_ = new_audio_fifo.Pass();
   }
 
   sink_->Initialize(new_sink_params, this);
 }
 
 }  // namespace content