Clean up the media stream audio track code

content/renderer/media/webrtc_audio_capturer.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_WEBRTC_AUDIO_CAPTURER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_
 
 #include <list>
 #include <string>
 
@@ skipping 87 matching lines @@
 
   const std::string& device_id() const { return device_info_.device.id; }
   int session_id() const { return device_info_.session_id; }
 
   // Stops recording audio. This method will empty its track lists since
   // stopping the capturer will implicitly invalidate all its tracks.
   // This method is exposed to the public because the MediaStreamAudioSource can
   // 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,
       media::AudioParameters params);
 
  protected:
   friend class base::RefCountedThreadSafe<WebRtcAudioCapturer>;
   ~WebRtcAudioCapturer() override;
 
  private:
@@ skipping 65 matching lines @@
   // Cached information of the device used by the capturer.
   const StreamDeviceInfo device_info_;
 
   // Stores latest microphone volume received in a CaptureData() callback.
   // Range is [0, 255].
   int volume_;
 
   // 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_;
 
   // Raw pointer to the MediaStreamAudioSource object that holds a reference
   // to this WebRtcAudioCapturer.
   // Since |audio_source_| is owned by a blink::WebMediaStreamSource object and
   // blink guarantees that the blink::WebMediaStreamSource outlives any
   // blink::WebMediaStreamTrack connected to the source, |audio_source_| is
   // guaranteed to exist as long as a WebRtcLocalAudioTrack is connected to this
   // WebRtcAudioCapturer.
   MediaStreamAudioSource* const audio_source_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_