Added a new echo likelihood stat that reports the maximum value from a previous time period.

webrtc/modules/audio_processing/audio_processing_impl.cc

 /*
  *  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 1567 matching lines @@
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // We just return if recording hasn't started.
   debug_dump_.debug_file->CloseFile();
   return kNoError;
 #else
   return kUnsupportedFunctionError;
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 }
 
+AudioProcessing::AudioProcessingStatistics::AudioProcessingStatistics() {
+  residual_echo_return_loss.Set(-100.0f, -100.0f, -100.0f, -100.0f);
+  echo_return_loss.Set(-100.0f, -100.0f, -100.0f, -100.0f);
+  echo_return_loss_enhancement.Set(-100.0f, -100.0f, -100.0f, -100.0f);
+  a_nlp.Set(-100.0f, -100.0f, -100.0f, -100.0f);
+}
+
+AudioProcessing::AudioProcessingStatistics::AudioProcessingStatistics(
+    const AudioProcessingStatistics& other) = default;
+
+AudioProcessing::AudioProcessingStatistics::~AudioProcessingStatistics() =
+    default;
+
 // TODO(ivoc): Remove this when GetStatistics() becomes pure virtual.
 AudioProcessing::AudioProcessingStatistics AudioProcessing::GetStatistics()
     const {
   return AudioProcessingStatistics();
 }
 
 AudioProcessing::AudioProcessingStatistics AudioProcessingImpl::GetStatistics()
     const {
   AudioProcessingStatistics stats;
   EchoCancellation::Metrics metrics;
   int success = public_submodules_->echo_cancellation->GetMetrics(&metrics);
   if (success == Error::kNoError) {
     stats.a_nlp.Set(metrics.a_nlp);
     stats.divergent_filter_fraction = metrics.divergent_filter_fraction;
     stats.echo_return_loss.Set(metrics.echo_return_loss);
     stats.echo_return_loss_enhancement.Set(
         metrics.echo_return_loss_enhancement);
     stats.residual_echo_return_loss.Set(metrics.residual_echo_return_loss);
   }
   stats.residual_echo_likelihood =
       private_submodules_->residual_echo_detector->echo_likelihood();
+  stats.residual_echo_likelihood_recent_max =
+      private_submodules_->residual_echo_detector->echo_likelihood_recent_max();
   public_submodules_->echo_cancellation->GetDelayMetrics(
       &stats.delay_median, &stats.delay_standard_deviation,
       &stats.fraction_poor_delays);
   return stats;
 }
 
 EchoCancellation* AudioProcessingImpl::echo_cancellation() const {
   return public_submodules_->echo_cancellation.get();
 }
 
@@ skipping 352 matching lines @@
       capture_processing_format(kSampleRate16kHz),
       split_rate(kSampleRate16kHz) {}
 
 AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
 
 AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
 
 AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
 
 }  // namespace webrtc