Further utilizing the AEC3 config struct for constants

webrtc/modules/audio_processing/aec3/matched_filter.h

 /*
  *  Copyright (c) 2017 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 65 matching lines @@
     float accuracy = 0.f;
     bool reliable = false;
     size_t lag = 0;
     bool updated = false;
   };
 
   MatchedFilter(ApmDataDumper* data_dumper,
                 Aec3Optimization optimization,
                 size_t window_size_sub_blocks,
                 int num_matched_filters,
-                size_t alignment_shift_sub_blocks);
+                size_t alignment_shift_sub_blocks,
+                float excitation_limit);
 
   ~MatchedFilter();
 
   // Updates the correlation with the values in the capture buffer.
   void Update(const DownsampledRenderBuffer& render_buffer,
               const std::array<float, kSubBlockSize>& capture);
 
   // Resets the matched filter.
   void Reset();
 
   // Returns the current lag estimates.
   rtc::ArrayView<const MatchedFilter::LagEstimate> GetLagEstimates() const {
     return lag_estimates_;
   }
 
   // Returns the number of lag estimates produced using the shifted signals.
   size_t NumLagEstimates() const { return filters_.size(); }
 
  private:
   ApmDataDumper* const data_dumper_;
   const Aec3Optimization optimization_;
   const size_t filter_intra_lag_shift_;
   std::vector<std::vector<float>> filters_;
   std::vector<LagEstimate> lag_estimates_;
+  const float excitation_limit_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MatchedFilter);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_MATCHED_FILTER_H_