Added forced zero AEC output after call startup and echo path changes

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 62 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; }
 
+  // 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,
               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;
   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;
+  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_