Add AEC filter divergence metric to StatsCollector.

webrtc/api/mediastreaminterface.h

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 197 matching lines @@
 // Interface of the audio processor used by the audio track to collect
 // statistics.
 class AudioProcessorInterface : public rtc::RefCountInterface {
  public:
   struct AudioProcessorStats {
     AudioProcessorStats() : typing_noise_detected(false),
                             echo_return_loss(0),
                             echo_return_loss_enhancement(0),
                             echo_delay_median_ms(0),
                             aec_quality_min(0.0),
-                            echo_delay_std_ms(0) {}
+                            echo_delay_std_ms(0),
+                            aec_divergent_filter_fraction(0.0) {}
     ~AudioProcessorStats() {}
 
     bool typing_noise_detected;
     int echo_return_loss;
     int echo_return_loss_enhancement;
     int echo_delay_median_ms;
     float aec_quality_min;
     int echo_delay_std_ms;
+    float aec_divergent_filter_fraction;
   };
 
   // Get audio processor statistics.
   virtual void GetStats(AudioProcessorStats* stats) = 0;
 
  protected:
   virtual ~AudioProcessorInterface() {}
 };
 
 class AudioTrackInterface : public MediaStreamTrackInterface {
@@ skipping 43 matching lines @@
   virtual bool RemoveTrack(AudioTrackInterface* track) = 0;
   virtual bool RemoveTrack(VideoTrackInterface* track) = 0;
 
  protected:
   virtual ~MediaStreamInterface() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_MEDIASTREAMINTERFACE_H_