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

webrtc/modules/audio_processing/residual_echo_detector.h

 /*
  *  Copyright (c) 2016 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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_
 
 #include <vector>
 
 #include "webrtc/base/array_view.h"
 #include "webrtc/modules/audio_processing/echo_detector/circular_buffer.h"
 #include "webrtc/modules/audio_processing/echo_detector/mean_variance_estimator.h"
+#include "webrtc/modules/audio_processing/echo_detector/moving_max.h"
 #include "webrtc/modules/audio_processing/echo_detector/normalized_covariance_estimator.h"
 
 namespace webrtc {
 
 class AudioBuffer;
 class EchoDetector;
 
 class ResidualEchoDetector {
  public:
   ResidualEchoDetector();
@@ skipping 10 matching lines @@
 
   // This function is for testing purposes only.
   void SetReliabilityForTest(float value) { reliability_ = value; }
 
   static void PackRenderAudioBuffer(AudioBuffer* audio,
                                     std::vector<float>* packed_buffer);
 
   // This function should be called while holding the capture lock.
   float echo_likelihood() const { return echo_likelihood_; }
 
+  float echo_likelihood_recent_max() const {
+    return recent_likelihood_max_.max();
+  }
+
  private:
   // Keep track if the |Process| function has been previously called.
   bool first_process_call_ = true;
   // Buffer for storing the power of incoming farend buffers. This is needed for
   // cases where calls to BufferFarend and Process are jittery.
   CircularBuffer render_buffer_;
   // Count how long ago it was that the size of |render_buffer_| was zero. This
   // value is also reset to zero when clock drift is detected and a value from
   // the renderbuffer is discarded, even though the buffer is not actually zero
   // at that point. This is done to avoid repeatedly removing elements in this
@@ skipping 10 matching lines @@
   // Index where next element should be inserted in all of the above circular
   // buffers.
   size_t next_insertion_index_ = 0;
 
   MeanVarianceEstimator render_statistics_;
   MeanVarianceEstimator capture_statistics_;
   // Current echo likelihood.
   float echo_likelihood_ = 0.f;
   // Reliability of the current likelihood.
   float reliability_ = 0.f;
+  MovingMax recent_likelihood_max_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_