MediaStream audio: Refactor 3 separate "glue" implementations into one.

content/renderer/media/webrtc_audio_device_impl.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_audio_device_impl.h"
 
-#include "base/bind.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
 #include "base/win/windows_version.h"
-#include "content/renderer/media/media_stream_audio_processor.h"
-#include "content/renderer/media/webrtc_audio_capturer.h"
+#include "content/renderer/media/webrtc/processed_local_audio_source.h"
 #include "content/renderer/media/webrtc_audio_renderer.h"
-#include "content/renderer/render_thread_impl.h"
 #include "media/audio/sample_rates.h"
+#include "media/base/audio_bus.h"
 #include "media/base/audio_parameters.h"
 
 using media::AudioParameters;
 using media::ChannelLayout;
 
 namespace content {
 
 WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl()
     : ref_count_(0),
       audio_transport_callback_(NULL),
@@ skipping 327 matching lines @@
   DCHECK(worker_thread_checker_.CalledOnValidThread());
   base::AutoLock auto_lock(lock_);
   *delay_ms = static_cast<uint16_t>(output_delay_ms_);
   return 0;
 }
 
 int32_t WebRtcAudioDeviceImpl::RecordingDelay(uint16_t* delay_ms) const {
   DCHECK(signaling_thread_checker_.CalledOnValidThread());
 
   // There is no way to report a correct delay value to WebRTC since there
-  // might be multiple WebRtcAudioCapturer instances.
+  // might be multiple ProcessedLocalAudioSource instances.
   NOTREACHED();
   return -1;
 }
 
 int32_t WebRtcAudioDeviceImpl::RecordingSampleRate(
     uint32_t* sample_rate) const {
   DCHECK(signaling_thread_checker_.CalledOnValidThread());
   // We use the default capturer as the recording sample rate.
   base::AutoLock auto_lock(lock_);
   if (capturers_.empty())
@@ skipping 40 matching lines @@
   audio_renderer_thread_checker_.DetachFromThread();
 
   // We acquire |lock_| again and assert our precondition, since we are
   // accessing the internal state again.
   base::AutoLock auto_lock(lock_);
   DCHECK(!renderer_.get());
   renderer_ = renderer;
   return true;
 }
 
-void WebRtcAudioDeviceImpl::AddAudioCapturer(WebRtcAudioCapturer* capturer) {
+void WebRtcAudioDeviceImpl::AddAudioCapturer(
+    ProcessedLocalAudioSource* capturer) {
   DCHECK(main_thread_checker_.CalledOnValidThread());
   DVLOG(1) << "WebRtcAudioDeviceImpl::AddAudioCapturer()";
   DCHECK(capturer);
   DCHECK(!capturer->device_info().device.id.empty());
 
   base::AutoLock auto_lock(lock_);
   DCHECK(std::find(capturers_.begin(), capturers_.end(), capturer) ==
       capturers_.end());
   capturers_.push_back(capturer);
 }
 
-void WebRtcAudioDeviceImpl::RemoveAudioCapturer(WebRtcAudioCapturer* capturer) {
+void WebRtcAudioDeviceImpl::RemoveAudioCapturer(
+    ProcessedLocalAudioSource* capturer) {
   DCHECK(main_thread_checker_.CalledOnValidThread());
   DVLOG(1) << "WebRtcAudioDeviceImpl::RemoveAudioCapturer()";
   DCHECK(capturer);
   base::AutoLock auto_lock(lock_);
   capturers_.remove(capturer);
 }
 
 void WebRtcAudioDeviceImpl::AddPlayoutSink(
     WebRtcPlayoutDataSource::Sink* sink) {
   DCHECK(main_thread_checker_.CalledOnValidThread());
@@ skipping 33 matching lines @@
 
   *session_id = device_info.session_id;
   *output_sample_rate = device_info.device.matched_output.sample_rate;
   *output_frames_per_buffer =
       device_info.device.matched_output.frames_per_buffer;
 
   return true;
 }
 
 }  // namespace content