Clean up the media stream audio track code

content/renderer/media/webrtc_audio_device_impl.h

Index: content/renderer/media/webrtc_audio_device_impl.h
diff --git a/content/renderer/media/webrtc_audio_device_impl.h b/content/renderer/media/webrtc_audio_device_impl.h
index debb4b71b2e630c39d0c68675169519d146f4e0a..e235320889ce909c7a273100fef3235e319731e4 100644
--- a/content/renderer/media/webrtc_audio_device_impl.h
+++ b/content/renderer/media/webrtc_audio_device_impl.h
@@ -199,39 +199,6 @@ class WebRtcAudioRendererSource {
   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
@@ -268,8 +235,7 @@ class WebRtcPlayoutDataSource {
 // 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:
@@ -363,22 +329,6 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
   // 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.
@@ -445,9 +395,6 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
   // 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);
 };