Add algorithm for Residual Echo Detector.

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 <memory>
 #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/normalized_covariance_estimator.h"
 
 namespace webrtc {
 
 class AudioBuffer;
 class EchoDetector;
 
 class ResidualEchoDetector {
  public:
   ResidualEchoDetector();
   ~ResidualEchoDetector();
 
   // This function should be called while holding the render lock.
-  void AnalyzeRenderAudio(rtc::ArrayView<const float> render_audio) const;
+  void AnalyzeRenderAudio(rtc::ArrayView<const float> render_audio);
 
   // This function should be called while holding the capture lock.
   void AnalyzeCaptureAudio(rtc::ArrayView<const float> capture_audio);
 
   // This function should be called while holding the capture lock.
   void Initialize();
 
   static void PackRenderAudioBuffer(AudioBuffer* audio,
                                     std::vector<float>* packed_buffer);
 
   // This function should be called while holding the capture lock.
-  float get_echo_likelihood() const;
+  float echo_likelihood() const { return echo_likelihood_; }
+
+ 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
+  // situation.
+  size_t frames_since_zero_buffer_size_ = 0;
+
+  // Circular buffers containing delayed versions of the power, mean and
+  // standard deviation, for calculating the delayed covariance values.
+  std::vector<float> render_power_;
+  std::vector<float> render_power_mean_;
+  std::vector<float> render_power_std_dev_;
+  // Covariance estimates for different delay values.
+  std::vector<NormalizedCovarianceEstimator> covariances_;
+  // 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;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_