move the APM to chrome.

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/debug/trace_event.h"
 #include "base/logging.h"
 #include "base/message_loop/message_loop_proxy.h"
 #include "base/metrics/histogram.h"
@@ skipping 41 matching lines @@
   }
 
   return audio_bus->frames();
 }
 
 void WebRtcLocalAudioRenderer::OnRenderError() {
   NOTIMPLEMENTED();
 }
 
 // content::WebRtcAudioCapturerSink implementation
-int WebRtcLocalAudioRenderer::CaptureData(const std::vector<int>& channels,
-                                          const int16* audio_data,
-                                          int sample_rate,
-                                          int number_of_channels,
-                                          int number_of_frames,
-                                          int audio_delay_milliseconds,
-                                          int current_volume,
-                                          bool need_audio_processing,
-                                          bool key_pressed) {
+void WebRtcLocalAudioRenderer::CaptureData(const std::vector<int>& channels,
+                                           const int16* audio_data,
+                                           int sample_rate,
+                                           int number_of_channels,
+                                           int number_of_frames) {
   TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::CaptureData");
   base::AutoLock auto_lock(thread_lock_);
   if (!playing_ || !volume_)
-    return 0;
+    return;
 
   // Push captured audio to FIFO so it can be read by a local sink.
   if (loopback_fifo_) {
     if (loopback_fifo_->frames() + number_of_frames <=
         loopback_fifo_->max_frames()) {
       scoped_ptr<media::AudioBus> audio_source = media::AudioBus::Create(
           number_of_channels, number_of_frames);
       audio_source->FromInterleaved(audio_data,
                                     audio_source->frames(),
                                     sizeof(audio_data[0]));
       loopback_fifo_->Push(audio_source.get());
 
       base::Time now = base::Time::Now();
       total_render_time_ += now - last_render_time_;
       last_render_time_ = now;
     } else {
       DVLOG(1) << "FIFO is full";
     }
   }
-
-  return 0;
 }
 
 void WebRtcLocalAudioRenderer::SetCaptureFormat(
     const media::AudioParameters& params) {
   audio_params_ = params;
 }
 
 // WebRtcLocalAudioRenderer::WebRtcLocalAudioRenderer implementation.
 WebRtcLocalAudioRenderer::WebRtcLocalAudioRenderer(
     WebRtcLocalAudioTrack* audio_track,
@@ skipping 175 matching lines @@
     return;
 
   sink_->Start();
   sink_started_ = true;
 
   UMA_HISTOGRAM_ENUMERATION("Media.LocalRendererSinkStates",
                             kSinkStarted, kSinkStatesMax);
 }
 
 }  // namespace content