Clean up the media stream audio track code

content/renderer/media/webaudio_capturer_source.h

 // 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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBAUDIO_CAPTURER_SOURCE_H_
 #define CONTENT_RENDERER_MEDIA_WEBAUDIO_CAPTURER_SOURCE_H_
 
 #include "base/memory/ref_counted.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_capturer_source.h"
 #include "media/base/audio_fifo.h"
 #include "third_party/WebKit/public/platform/WebAudioDestinationConsumer.h"
 #include "third_party/WebKit/public/platform/WebVector.h"
 
 namespace content {
 
-class WebRtcAudioCapturer;
 class WebRtcLocalAudioTrack;
 
 // WebAudioCapturerSource is the missing link between
 // WebAudio's MediaStreamAudioDestinationNode and WebRtcLocalAudioTrack.
 //
 // 1. WebKit calls the setFormat() method setting up the basic stream format
 //    (channels, and sample-rate).
 // 2. consumeAudio() is called periodically by WebKit which dispatches the
 //    audio stream to the WebRtcLocalAudioTrack::Capture() method.
 class WebAudioCapturerSource
     :  public base::RefCountedThreadSafe<WebAudioCapturerSource>,
        public blink::WebAudioDestinationConsumer {
  public:
   WebAudioCapturerSource();
 
   // WebAudioDestinationConsumer implementation.
   // setFormat() is called early on, so that we can configure the audio track.
   virtual void setFormat(size_t number_of_channels, float sample_rate) override;
   // MediaStreamAudioDestinationNode periodically calls consumeAudio().
   // Called on the WebAudio audio thread.
   virtual void consumeAudio(const blink::WebVector<const float*>& audio_data,
       size_t number_of_frames) override;
 
   // Called when the WebAudioCapturerSource is hooking to a media audio track.
   // |track| is the sink of the data flow. |source_provider| is the source of
   // the data flow where stream information like delay, volume, key_pressed,
   // is stored.
-  void Start(WebRtcLocalAudioTrack* track, WebRtcAudioCapturer* capturer);
+  void Start(WebRtcLocalAudioTrack* track);
 
   // Called when the media audio track is stopping.
   void Stop();
 
  protected:
   friend class base::RefCountedThreadSafe<WebAudioCapturerSource>;
   virtual ~WebAudioCapturerSource();
 
  private:
   // Used to DCHECK that some methods are called on the correct thread.
   base::ThreadChecker thread_checker_;
 
   // The audio track this WebAudioCapturerSource is feeding data to.
   // WebRtcLocalAudioTrack is reference counted, and owning this object.
   // To avoid circular reference, a raw pointer is kept here.
   WebRtcLocalAudioTrack* track_;
 
-  // A raw pointer to the capturer to get audio processing params like
-  // delay, volume, key_pressed information.
-  // This |capturer_| is guaranteed to outlive this object.
-  WebRtcAudioCapturer* capturer_;
-
   media::AudioParameters params_;
 
   // Flag to help notify the |track_| when the audio format has changed.
   bool audio_format_changed_;
 
   // Wraps data coming from HandleCapture().
   scoped_ptr<media::AudioBus> wrapper_bus_;
 
   // Bus for reading from FIFO and calling the CaptureCallback.
   scoped_ptr<media::AudioBus> capture_bus_;
 
   // Handles mismatch between WebAudio buffer size and WebRTC.
   scoped_ptr<media::AudioFifo> fifo_;
 
   // Buffer to pass audio data to WebRtc.
   scoped_ptr<int16[]> audio_data_;
 
   // Synchronizes HandleCapture() with AudioCapturerSource calls.
   base::Lock lock_;
   bool started_;
 
   DISALLOW_COPY_AND_ASSIGN(WebAudioCapturerSource);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBAUDIO_CAPTURER_SOURCE_H_