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

content/renderer/media/webrtc_local_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_local_audio_renderer.h"
 
 #include "base/location.h"
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/synchronization/lock.h"
@@ skipping 15 matching lines @@
 enum LocalRendererSinkStates {
   kSinkStarted = 0,
   kSinkNeverStarted,
   kSinkStatesMax  // Must always be last!
 };
 
 }  // namespace
 
 // media::AudioRendererSink::RenderCallback implementation
 int WebRtcLocalAudioRenderer::Render(
-    media::AudioBus* audio_bus, uint32_t audio_delay_milliseconds,
-    uint32_t frames_skipped) {
+    media::AudioBus* audio_bus, int audio_delay_milliseconds) {
   TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::Render");
   base::AutoLock auto_lock(thread_lock_);
 
   if (!playing_ || !volume_ || !audio_shifter_) {
     audio_bus->Zero();
     return 0;
   }
 
   audio_shifter_->Pull(
       audio_bus,
@@ skipping 297 matching lines @@
       AudioDeviceFactory::NewOutputDevice(source_render_frame_id_, session_id_,
                                           output_device_id_, security_origin_);
   int frames_per_buffer = sink_->GetOutputParameters().frames_per_buffer();
   sink_params_ = source_params_;
   sink_params_.set_frames_per_buffer(WebRtcAudioRenderer::GetOptimalBufferSize(
       source_params_.sample_rate(), frames_per_buffer));
   MaybeStartSink();
 }
 
 }  // namespace content