Pulling AEC divergent filter fraction.

webrtc/modules/audio_processing/include/audio_processing.h

 /*
  *  Copyright (c) 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 654 matching lines @@
     AudioProcessing::Statistic residual_echo_return_loss;
 
     // ERL = 10log_10(P_far / P_echo)
     AudioProcessing::Statistic echo_return_loss;
 
     // ERLE = 10log_10(P_echo / P_out)
     AudioProcessing::Statistic echo_return_loss_enhancement;
 
     // (Pre non-linear processing suppression) A_NLP = 10log_10(P_echo / P_a)
     AudioProcessing::Statistic a_nlp;
+
+    // Fraction of filter being divergent, based on a 0.5-second, non-overlapped

[peah-webrtc, 2016/04/07 11:25:40]

I think a more proper description would be
// Fraction of the call where the filter is divergent,...

as it is now, at least to me it sounds as if  a certain part of the filter is
divergent.

[minyue-webrtc, 2016/04/07 11:40:55]

Good point, I also think it is a bit vague. But "of the call" might be a bit
misleading since we use "time windows". How about "Fraction of time that filter
is divergent, in a 0.5-second, non-overlapped aggregation window"

+    // aggregation window.
+    float divergent_filter_fraction;
   };
 
   // TODO(ajm): discuss the metrics update period.
   virtual int GetMetrics(Metrics* metrics) = 0;
 
   // Enables computation and logging of delay values. Statistics are obtained
   // through |GetDelayMetrics()|.
   virtual int enable_delay_logging(bool enable) = 0;
   virtual bool is_delay_logging_enabled() const = 0;
 
@@ skipping 279 matching lines @@
   // This does not impact the size of frames passed to |ProcessStream()|.
   virtual int set_frame_size_ms(int size) = 0;
   virtual int frame_size_ms() const = 0;
 
  protected:
   virtual ~VoiceDetection() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_