Echo canceller 3 improvements for setups with headsets.

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 23 matching lines @@
   AecState();
   ~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.
-  bool ActiveRender() const { return active_render_blocks_ > 200; }
+  // TODO(peah): Deprecate this in an upcoming CL.
+  bool ActiveRender() const { return blocks_with_filter_adaptation_ > 200; }
 
   // Returns the ERLE.
   const std::array<float, kFftLengthBy2Plus1>& Erle() const {
     return erle_estimator_.Erle();
   }
 
   // Returns the ERL.
   const std::array<float, kFftLengthBy2Plus1>& Erl() const {
     return erl_estimator_.Erl();
   }
@@ skipping 37 matching lines @@
               const std::array<float, kFftLengthBy2Plus1>& Y2,
               rtc::ArrayView<const float> x,
               bool echo_leakage_detected);
 
  private:
   static int instance_count_;
   std::unique_ptr<ApmDataDumper> data_dumper_;
   ErlEstimator erl_estimator_;
   ErleEstimator erle_estimator_;
   int echo_path_change_counter_;
-  size_t active_render_blocks_ = 0;
+  size_t blocks_with_filter_adaptation_ = 0;
   bool usable_linear_estimate_ = false;
   bool echo_leakage_detected_ = false;
   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);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_