Clean up the media stream audio track code

content/renderer/media/webaudio_capturer_source.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/webaudio_capturer_source.h"
 
 #include "base/logging.h"
 #include "base/time/time.h"
-#include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
 
 using media::AudioBus;
 using media::AudioFifo;
 using media::AudioParameters;
 using media::ChannelLayout;
 using media::CHANNEL_LAYOUT_MONO;
 using media::CHANNEL_LAYOUT_STEREO;
 
 static const int kMaxNumberOfBuffersInFifo = 5;
 
 namespace content {
 
 WebAudioCapturerSource::WebAudioCapturerSource()
     : track_(NULL),
-      capturer_(NULL),
       audio_format_changed_(false) {
 }
 
 WebAudioCapturerSource::~WebAudioCapturerSource() {
 }
 
 void WebAudioCapturerSource::setFormat(
     size_t number_of_channels, float sample_rate) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DVLOG(1) << "WebAudioCapturerSource::setFormat(sample_rate="
@@ skipping 18 matching lines @@
 
   wrapper_bus_ = AudioBus::CreateWrapper(params_.channels());
   capture_bus_ = AudioBus::Create(params_);
   audio_data_.reset(
       new int16[params_.frames_per_buffer() * params_.channels()]);
   fifo_.reset(new AudioFifo(
       params_.channels(),
       kMaxNumberOfBuffersInFifo * params_.frames_per_buffer()));
 }
 
-void WebAudioCapturerSource::Start(
-    WebRtcLocalAudioTrack* track, WebRtcAudioCapturer* capturer) {
+void WebAudioCapturerSource::Start(WebRtcLocalAudioTrack* track) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(track);
   base::AutoLock auto_lock(lock_);
   track_ = track;
-  capturer_ = capturer;
 }
 
 void WebAudioCapturerSource::Stop() {
   DCHECK(thread_checker_.CalledOnValidThread());
   base::AutoLock auto_lock(lock_);
   track_ = NULL;
-  capturer_ = NULL;
 }
 
 void WebAudioCapturerSource::consumeAudio(
     const blink::WebVector<const float*>& audio_data,
     size_t number_of_frames) {
   base::AutoLock auto_lock(lock_);
   if (!track_)
     return;
 
   // Update the downstream client if the audio format has been changed.
@@ skipping 13 matching lines @@
 
   // Handle mismatch between WebAudio buffer-size and WebRTC.
   int available = fifo_->max_frames() - fifo_->frames();
   if (available < static_cast<int>(number_of_frames)) {
     NOTREACHED() << "WebAudioCapturerSource::Consume() : FIFO overrun.";
     return;
   }
 
   fifo_->Push(wrapper_bus_.get());
   int capture_frames = params_.frames_per_buffer();
-  base::TimeDelta delay;
-  int volume = 0;
-  bool key_pressed = false;
-  if (capturer_) {
-    capturer_->GetAudioProcessingParams(&delay, &volume, &key_pressed);
-  }
-
-  // Turn off audio processing if the delay value is 0, since in such case,
-  // it indicates the data is not from microphone.
-  // TODO(xians): remove the flag when supporting one APM per audio track.
-  // See crbug/264611 for details.
-  bool need_audio_processing = (delay.InMilliseconds() != 0);
   while (fifo_->frames() >= capture_frames) {
     fifo_->Consume(capture_bus_.get(), 0, capture_frames);
     // TODO(xians): Avoid this interleave/deinterleave operation.
     capture_bus_->ToInterleaved(capture_bus_->frames(),
                                 params_.bits_per_sample() / 8,
                                 audio_data_.get());
-    track_->Capture(audio_data_.get(), delay, volume, key_pressed,
-                    need_audio_processing, false);
+    track_->Capture(audio_data_.get(), false);
   }
 }
 
 }  // namespace content