Added the ability to adjust the AEC3 performance for large rooms

webrtc/modules/audio_processing/aec3/aec_state.cc

 /*
  *  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 60 matching lines @@
   return rtc::Optional<size_t>();
 }
 
 constexpr int kEchoPathChangeCounterInitial = kNumBlocksPerSecond / 5;
 constexpr int kEchoPathChangeCounterMax = 2 * kNumBlocksPerSecond;
 
 }  // namespace
 
 int AecState::instance_count_ = 0;
 
-AecState::AecState()
+AecState::AecState(float echo_decay)
     : data_dumper_(
           new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
-      echo_path_change_counter_(kEchoPathChangeCounterInitial) {}
+      echo_path_change_counter_(kEchoPathChangeCounterInitial),
+      echo_decay_factor_(echo_decay) {}
 
 AecState::~AecState() = default;
 
 void AecState::HandleEchoPathChange(
     const EchoPathVariability& echo_path_variability) {
   if (echo_path_variability.AudioPathChanged()) {
     blocks_since_last_saturation_ = 0;
     usable_linear_estimate_ = false;
     echo_leakage_detected_ = false;
     capture_signal_saturation_ = false;
@@ skipping 79 matching lines @@
   // After an amount of active render samples for which an echo should have been
   // detected in the capture signal if the ERL was not infinite, flag that a
   // headset is used.
   headset_detected_ =
       !external_delay_ && !filter_delay_ &&
       (!render_received_ ||
        blocks_with_filter_adaptation_ >= kEchoPathChangeConvergenceBlocks);
 }
 
 }  // namespace webrtc