Clean up the media stream audio track code

content/renderer/media/webrtc_audio_capturer.h

Index: content/renderer/media/webrtc_audio_capturer.h
diff --git a/content/renderer/media/webrtc_audio_capturer.h b/content/renderer/media/webrtc_audio_capturer.h
index d98db0f106cdcaae2a017056d8a7caedeb98a021..b8aae22c4f094ff5594ef61d648a210566790efd 100644
--- a/content/renderer/media/webrtc_audio_capturer.h
+++ b/content/renderer/media/webrtc_audio_capturer.h
@@ -105,12 +105,6 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   // call Stop()
   void Stop();
 
-  // Called by the WebAudioCapturerSource to get the audio processing params.
-  // This function is triggered by provideInput() on the WebAudio audio thread,
-  // TODO(xians): Remove after moving APM from WebRtc to Chrome.
-  void GetAudioProcessingParams(base::TimeDelta* delay, int* volume,
-                                bool* key_pressed);
-
   // Used by the unittests to inject their own source to the capturer.
   void SetCapturerSourceForTesting(
       const scoped_refptr<media::AudioCapturerSource>& source,
@@ -196,13 +190,6 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   // Flag which affects the buffer size used by the capturer.
   bool peer_connection_mode_;
 
-  // Cache value for the audio processing params.
-  base::TimeDelta audio_delay_;
-  bool key_pressed_;
-
-  // Flag to help deciding if the data needs audio processing.
-  bool need_audio_processing_;
-
   // Raw pointer to the WebRtcAudioDeviceImpl, which is valid for the lifetime
   // of RenderThread.
   WebRtcAudioDeviceImpl* audio_device_;