Added the ability to adjust the AEC3 performance for large rooms

webrtc/modules/audio_processing/aec3/aec_state.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 13 matching lines @@
 #include "webrtc/modules/audio_processing/aec3/erle_estimator.h"
 #include "webrtc/modules/audio_processing/aec3/render_buffer.h"
 
 namespace webrtc {
 
 class ApmDataDumper;
 
 // Handles the state and the conditions for the echo removal functionality.
 class AecState {
  public:
-  AecState();
+  explicit AecState(float echo_decay);
   ~AecState();
 
   // Returns whether the linear filter estimate is usable.
   bool UsableLinearEstimate() const { return usable_linear_estimate_; }
 
   // Returns whether there has been echo leakage detected.
   bool EchoLeakageDetected() const { return echo_leakage_detected_; }
 
   // Returns whether the render signal is currently active.
   // TODO(peah): Deprecate this in an upcoming CL.
@@ skipping 27 matching lines @@
   }
 
   // Returns whether a probable headset setup has been detected.
   bool HeadsetDetected() const { return headset_detected_; }
 
   // Takes appropriate action at an echo path change.
   void HandleEchoPathChange(const EchoPathVariability& echo_path_variability);
 
   // Returns the decay factor for the echo reverberation.
   // TODO(peah): Make this adaptive.
-  float ReverbDecayFactor() const { return 0.f; }
+  float ReverbDecayFactor() const { return echo_decay_factor_; }
 
   // Returns whether the echo suppression gain should be forced to zero.
   bool ForcedZeroGain() const { return force_zero_gain_; }
 
   // Updates the aec state.
   void Update(const std::vector<std::array<float, kFftLengthBy2Plus1>>&
                   adaptive_filter_frequency_response,
               const rtc::Optional<size_t>& external_delay_samples,
               const RenderBuffer& render_buffer,
               const std::array<float, kFftLengthBy2Plus1>& E2_main,
@@ skipping 13 matching lines @@
   bool capture_signal_saturation_ = false;
   bool echo_saturation_ = false;
   bool headset_detected_ = false;
   float previous_max_sample_ = 0.f;
   bool force_zero_gain_ = false;
   bool render_received_ = false;
   size_t force_zero_gain_counter_ = 0;
   rtc::Optional<size_t> filter_delay_;
   rtc::Optional<size_t> external_delay_;
   size_t blocks_since_last_saturation_ = 1000;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(AecState);
+  const float echo_decay_factor_;
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AecState);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_