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

content/renderer/media/media_stream_audio_processor_options.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_MEDIA_STREAM_AUDIO_PROCESSOR_OPTIONS_H_
 #define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_OPTIONS_H_
 
 #include <string>
 
+#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
+
 namespace blink {
 class WebMediaConstraints;
 }
 
 namespace webrtc {
 
 class AudioFrame;
 class AudioProcessing;
 class MediaConstraintsInterface;
 class TypingDetection;
@@ skipping 45 matching lines @@
 void EnableExperimentalEchoCancellation(AudioProcessing* audio_processing);
 
 // Starts the echo cancellation dump in |audio_processing|.
 void StartAecDump(AudioProcessing* audio_processing);
 
 // Stops the echo cancellation dump in |audio_processing|.
 void StopAecDump(AudioProcessing* audio_processing);
 
 void EnableAutomaticGainControl(AudioProcessing* audio_processing);
 
+void GetAecStats(AudioProcessing* audio_processing,
+                 webrtc::AudioProcessorInterface::AudioProcessorStats* stats);
+
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_OPTIONS_H_