Add AEC filter divergence metric to StatsCollector.

webrtc/api/statscollector.cc

 /*
  *  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 919 matching lines @@
   auto audio_processor(track->GetAudioProcessor());
 
   if (audio_processor.get()) {
     AudioProcessorInterface::AudioProcessorStats stats;
     audio_processor->GetStats(&stats);
 
     SetAudioProcessingStats(
         report, stats.typing_noise_detected, stats.echo_return_loss,
         stats.echo_return_loss_enhancement, stats.echo_delay_median_ms,
         stats.aec_quality_min, stats.echo_delay_std_ms);
+
+    report->AddFloat(StatsReport::kStatsValueNameAecDivergentFilterFraction,
+                     stats.aec_divergent_filter_fraction);
   }
 }
 
 bool StatsCollector::GetTrackIdBySsrc(uint32_t ssrc,
                                       std::string* track_id,
                                       StatsReport::Direction direction) {
   RTC_DCHECK(pc_->session()->signaling_thread()->IsCurrent());
   if (direction == StatsReport::kSend) {
     if (!pc_->session()->GetLocalTrackIdBySsrc(ssrc, track_id)) {
       LOG(LS_WARNING) << "The SSRC " << ssrc
@@ skipping 21 matching lines @@
     StatsReport* report = entry.second;
     report->set_timestamp(stats_gathering_started_);
   }
 }
 
 void StatsCollector::ClearUpdateStatsCacheForTest() {
   stats_gathering_started_ = 0;
 }
 
 }  // namespace webrtc