Clean up the media stream audio track code

content/renderer/media/webrtc_local_audio_track.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_LOCAL_AUDIO_TRACK_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_
 
 #include <list>
 #include <string>
 
 #include "base/memory/ref_counted.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
 #include "content/renderer/media/media_stream_track.h"
 #include "content/renderer/media/tagged_list.h"
-#include "content/renderer/media/webrtc_audio_device_impl.h"
+#include "media/audio/audio_parameters.h"
 
 namespace content {
 
 class MediaStreamAudioLevelCalculator;
 class MediaStreamAudioProcessor;
 class MediaStreamAudioSink;
 class MediaStreamAudioSinkOwner;
 class MediaStreamAudioTrackSink;
-class PeerConnectionAudioSink;
 class WebAudioCapturerSource;
 class WebRtcAudioCapturer;
 class WebRtcLocalAudioTrackAdapter;
 
 // A WebRtcLocalAudioTrack instance contains the implementations of
 // MediaStreamTrackExtraData.
 // When an instance is created, it will register itself as a track to the
 // WebRtcAudioCapturer to get the captured data, and forward the data to
 // its |sinks_|. The data flow can be stopped by disabling the audio track.
 class CONTENT_EXPORT WebRtcLocalAudioTrack
     : NON_EXPORTED_BASE(public MediaStreamTrack) {
  public:
   WebRtcLocalAudioTrack(WebRtcLocalAudioTrackAdapter* adapter,
                         const scoped_refptr<WebRtcAudioCapturer>& capturer,
                         WebAudioCapturerSource* webaudio_source);
 
   virtual ~WebRtcLocalAudioTrack();
 
   // Add a sink to the track. This function will trigger a OnSetFormat()
   // call on the |sink|.
   // Called on the main render thread.
   void AddSink(MediaStreamAudioSink* sink);
 
   // Remove a sink from the track.
   // Called on the main render thread.
   void RemoveSink(MediaStreamAudioSink* sink);
 
-  // Add/remove PeerConnection sink to/from the track.
-  // TODO(xians): Remove these two methods after PeerConnection can use the
-  // same sink interface as MediaStreamAudioSink.
-  void AddSink(PeerConnectionAudioSink* sink);
-  void RemoveSink(PeerConnectionAudioSink* sink);
-
   // Starts the local audio track. Called on the main render thread and
   // should be called only once when audio track is created.
   void Start();
 
   // Overrides for MediaStreamTrack.
 
   void SetEnabled(bool enabled) override;
 
   // Stops the local audio track. Called on the main render thread and
   // should be called only once when audio track going away.
   void Stop() override;
 
   webrtc::AudioTrackInterface* GetAudioAdapter() override;
 
   // Method called by the capturer to deliver the capture data.
   // Called on the capture audio thread.
-  void Capture(const int16* audio_data,
-               base::TimeDelta delay,
-               int volume,
-               bool key_pressed,
-               bool need_audio_processing,
-               bool force_report_nonzero_energy);
+  void Capture(const int16* audio_data, bool force_report_nonzero_energy);
 
   // Method called by the capturer to set the audio parameters used by source
   // of the capture data..
   // Called on the capture audio thread.
   void OnSetFormat(const media::AudioParameters& params);
 
   // Method called by the capturer to set the processor that applies signal
   // processing on the data of the track.
   // Called on the capture audio thread.
   void SetAudioProcessor(
@@ skipping 36 matching lines @@
   // Used to calculate the signal level that shows in the UI.
   // Accessed on only the audio thread.
   scoped_ptr<MediaStreamAudioLevelCalculator> level_calculator_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_