move the APM to chrome.

content/renderer/media/webrtc_local_audio_source_provider.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_local_audio_source_provider.h"
 
 #include "base/logging.h"
 #include "content/renderer/render_thread_impl.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_fifo.h"
@@ skipping 50 matching lines @@
   // the converter.
   audio_converter_.reset(
       new media::AudioConverter(source_params, sink_params_, false));
   audio_converter_->AddInput(this);
   fifo_.reset(new media::AudioFifo(
       source_params.channels(),
       kMaxNumberOfBuffers * source_params.frames_per_buffer()));
 }
 
 void WebRtcLocalAudioSourceProvider::DeliverData(
-    media::AudioBus* audio_source,
-    int audio_delay_milliseconds,
-    int volume,
-    bool key_pressed) {
+    const int16* data,
+    int sample_rate,
+    int number_of_channels,
+    int number_of_frames) {
   base::AutoLock auto_lock(lock_);
   if (!is_enabled_)
     return;
 
   DCHECK(fifo_.get());
-
+  // TODO(xians): FIXME.
+/*
   if (fifo_->frames() + audio_source->frames() <= fifo_->max_frames()) {
     fifo_->Push(audio_source);
   } else {
     // This can happen if the data in FIFO is too slowed to be consumed or
     // WebAudio stops consuming data.
     DLOG(WARNING) << "Local source provicer FIFO is full" << fifo_->frames();
   }
 
   // Cache the values for GetAudioProcessingParams().
   last_fill_ = base::TimeTicks::Now();
   audio_delay_ms_ = audio_delay_milliseconds;
   volume_ = volume;
-  key_pressed_ = key_pressed;
+  key_pressed_ = key_pressed;*/
 }
 
 void WebRtcLocalAudioSourceProvider::GetAudioProcessingParams(
     int* delay_ms, int* volume, bool* key_pressed) {
   int elapsed_ms = 0;
   if (!last_fill_.is_null()) {
     elapsed_ms = static_cast<int>(
         (base::TimeTicks::Now() - last_fill_).InMilliseconds());
   }
   *delay_ms = audio_delay_ms_ + elapsed_ms + static_cast<int>(
@@ skipping 42 matching lines @@
   return 1.0;
 }
 
 void WebRtcLocalAudioSourceProvider::SetSinkParamsForTesting(
     const media::AudioParameters& sink_params) {
   DCHECK(thread_checker_.CalledOnValidThread());
   sink_params_ = sink_params;
 }
 
 }  // namespace content