Adds fuzzer for the residual echo detector.

webrtc/modules/audio_processing/residual_echo_detector.cc

 /*
  *  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.
  */
 
@@ skipping 86 matching lines @@
     const size_t read_index =
         (kLookbackFrames + next_insertion_index_ - delay) % kLookbackFrames;
     RTC_DCHECK_LT(read_index, render_power_.size());
     covariances_[delay].Update(capture_power, capture_mean,
                                capture_std_deviation, render_power_[read_index],
                                render_power_mean_[read_index],
                                render_power_std_dev_[read_index]);
     echo_likelihood_ = std::max(
         echo_likelihood_, covariances_[delay].normalized_cross_correlation());
   }
+  RTC_DCHECK_LT(echo_likelihood_, 1.1f);
   reliability_ = (1.0f - kAlpha) * reliability_ + kAlpha * 1.0f;
   echo_likelihood_ *= reliability_;
   // This is a temporary fix to prevent echo likelihood values > 1.0.
   // TODO(ivoc): Find the root cause of this issue and fix it.
   echo_likelihood_ = std::min(echo_likelihood_, 1.0f);
   int echo_percentage = static_cast<int>(echo_likelihood_ * 100);
   RTC_HISTOGRAM_COUNTS("WebRTC.Audio.ResidualEchoDetector.EchoLikelihood",
                        echo_percentage, 0, 100, 100 /* number of bins */);
 
   // Update the buffer of recent likelihood values.
@@ skipping 26 matching lines @@
   RTC_DCHECK_GE(160, audio->num_frames_per_band());
 
   packed_buffer->clear();
   packed_buffer->insert(packed_buffer->end(),
                         audio->split_bands_const_f(0)[kBand0To8kHz],
                         (audio->split_bands_const_f(0)[kBand0To8kHz] +
                          audio->num_frames_per_band()));
 }
 
 }  // namespace webrtc