Added the ability to adjust the AEC3 performance for large rooms

webrtc/modules/audio_processing/include/audio_processing.h

 /*
  *  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 250 matching lines @@
     struct HighPassFilter {
       bool enabled = false;
     } high_pass_filter;
 
     // Enables the next generation AEC functionality. This feature replaces the
     // standard methods for echo removal in the AEC.
     // The functionality is not yet activated in the code and turning this on
     // does not yet have the desired behavior.
     struct EchoCanceller3 {
       bool enabled = false;
+      float echo_decay = 0.f;
     } echo_canceller3;
 
     // Enables the next generation AGC functionality. This feature replaces the
     // standard methods of gain control in the previous AGC.
     // The functionality is not yet activated in the code and turning this on
     // does not yet have the desired behavior.
     struct GainController2 {
       bool enabled = false;
     } gain_controller2;
   };
@@ skipping 817 matching lines @@
   // This does not impact the size of frames passed to |ProcessStream()|.
   virtual int set_frame_size_ms(int size) = 0;
   virtual int frame_size_ms() const = 0;
 
  protected:
   virtual ~VoiceDetection() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_