Implements the GetSignalLevel and GetStats interface for the local audio track.

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 "content/renderer/media/webrtc_local_audio_source_provider.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.
@@ skipping 26 matching lines @@
 
   // Starts the local audio track. Called on the main render thread and
   // should be called only once when audio track is created.
   void Start();
 
   // Stops the local audio track. Called on the main render thread and
   // should be called only once when audio track going away.
   void Stop();
 
   // Method called by the capturer to deliver the capture data.
-  // Call on the capture audio thread.
+  // Called on the capture audio thread.
   void Capture(const int16* audio_data,
                base::TimeDelta delay,
                int volume,
                bool key_pressed,
                bool need_audio_processing);
 
   // Method called by the capturer to set the audio parameters used by source
   // of the capture data..
-  // Call on the capture audio thread.
+  // 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(
+      const scoped_refptr<MediaStreamAudioProcessor>& processor);
+
   blink::WebAudioSourceProvider* audio_source_provider() const {
     return source_provider_.get();
   }
 
  private:
   typedef TaggedList<MediaStreamAudioTrackSink> SinkList;
 
   // All usage of libjingle is through this adapter. The adapter holds
   // a reference on this object, but not vice versa.
   WebRtcLocalAudioTrackAdapter* adapter_;
@@ skipping 30 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_