Clean up the media stream audio track code

content/renderer/media/webrtc_audio_device_impl.h

 // 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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 
 #include <string>
 #include <vector>
 
@@ skipping 181 matching lines @@
                           int audio_delay_milliseconds,
                           base::TimeDelta* current_time) = 0;
 
   // Callback to notify the client that the renderer is going away.
   virtual void RemoveAudioRenderer(WebRtcAudioRenderer* renderer) = 0;
 
  protected:
   virtual ~WebRtcAudioRendererSource() {}
 };
 
-class PeerConnectionAudioSink {
- public:
-  // Callback to deliver the captured interleaved data.
-  // |channels| contains a vector of WebRtc VoE channels.
-  // |audio_data| is the pointer to the audio data.
-  // |sample_rate| is the sample frequency of audio data.
-  // |number_of_channels| is the number of channels reflecting the order of
-  // surround sound channels.
-  // |audio_delay_milliseconds| is recording delay value.
-  // |current_volume| is current microphone volume, in range of |0, 255].
-  // |need_audio_processing| indicates if the audio needs WebRtc AEC/NS/AGC
-  // audio processing.
-  // The return value is the new microphone volume, in the range of |0, 255].
-  // When the volume does not need to be updated, it returns 0.
-  virtual int OnData(const int16* audio_data,
-                     int sample_rate,
-                     int number_of_channels,
-                     int number_of_frames,
-                     const std::vector<int>& channels,
-                     int audio_delay_milliseconds,
-                     int current_volume,
-                     bool need_audio_processing,
-                     bool key_pressed) = 0;
-
-  // Set the format for the capture audio parameters.
-  // This is called when the capture format has changed, and it must be called
-  // on the same thread as calling CaptureData().
-  virtual void OnSetFormat(const media::AudioParameters& params) = 0;
-
- protected:
- virtual ~PeerConnectionAudioSink() {}
-};
-
 // TODO(xians): Merge this interface with WebRtcAudioRendererSource.
 // The reason why we could not do it today is that WebRtcAudioRendererSource
 // gets the data by pulling, while the data is pushed into
 // WebRtcPlayoutDataSource::Sink.
 class WebRtcPlayoutDataSource {
  public:
   class Sink {
    public:
     // Callback to get the playout data.
     // Called on the render audio thread.
@@ skipping 16 matching lines @@
   virtual void RemovePlayoutSink(Sink* sink) = 0;
 
  protected:
   virtual ~WebRtcPlayoutDataSource() {}
 };
 
 // Note that this class inherits from webrtc::AudioDeviceModule but due to
 // the high number of non-implemented methods, we move the cruft over to the
 // WebRtcAudioDeviceNotImpl.
 class CONTENT_EXPORT WebRtcAudioDeviceImpl
-    : NON_EXPORTED_BASE(public PeerConnectionAudioSink),
-      NON_EXPORTED_BASE(public WebRtcAudioDeviceNotImpl),
+    : NON_EXPORTED_BASE(public WebRtcAudioDeviceNotImpl),
       NON_EXPORTED_BASE(public WebRtcAudioRendererSource),
       NON_EXPORTED_BASE(public WebRtcPlayoutDataSource) {
  public:
   // The maximum volume value WebRtc uses.
   static const int kMaxVolumeLevel = 255;
 
   // Instances of this object are created on the main render thread.
   WebRtcAudioDeviceImpl();
 
   // webrtc::RefCountedModule implementation.
@@ skipping 71 matching lines @@
   }
 
  private:
   typedef std::list<scoped_refptr<WebRtcAudioCapturer> > CapturerList;
   typedef std::list<WebRtcPlayoutDataSource::Sink*> PlayoutDataSinkList;
   class RenderBuffer;
 
   // Make destructor private to ensure that we can only be deleted by Release().
   ~WebRtcAudioDeviceImpl() override;
 
-  // PeerConnectionAudioSink implementation.
-
-  // Called on the AudioInputDevice worker thread.
-  int OnData(const int16* audio_data,
-             int sample_rate,
-             int number_of_channels,
-             int number_of_frames,
-             const std::vector<int>& channels,
-             int audio_delay_milliseconds,
-             int current_volume,
-             bool need_audio_processing,
-             bool key_pressed) override;
-
-  // Called on the AudioInputDevice worker thread.
-  void OnSetFormat(const media::AudioParameters& params) override;
-
   // WebRtcAudioRendererSource implementation.
 
   // Called on the AudioOutputDevice worker thread.
   void RenderData(media::AudioBus* audio_bus,
                   int sample_rate,
                   int audio_delay_milliseconds,
                   base::TimeDelta* current_time) override;
 
   // Called on the main render thread.
   void RemoveAudioRenderer(WebRtcAudioRenderer* renderer) override;
@@ skipping 43 matching lines @@
   bool recording_;
 
   // Stores latest microphone volume received in a CaptureData() callback.
   // Range is [0, 255].
   uint32_t microphone_volume_;
 
   // Buffer used for temporary storage during render callback.
   // It is only accessed by the audio render thread.
   std::vector<int16> render_buffer_;
 
-  // Flag to tell if audio processing is enabled in MediaStreamAudioProcessor.
-  const bool is_audio_track_processing_enabled_;
-
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_