New module for the adaptive level controlling functionality in the audio processing module

webrtc/modules/audio_processing/test/audio_processing_simulator.h

 /*
  *  Copyright (c) 2016 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 38 matching lines @@
   rtc::Optional<bool> use_bf;
   rtc::Optional<bool> use_ie;
   rtc::Optional<bool> use_vad;
   rtc::Optional<bool> use_le;
   rtc::Optional<bool> use_all;
   rtc::Optional<int> aec_suppression_level;
   rtc::Optional<bool> use_delay_agnostic;
   rtc::Optional<bool> use_extended_filter;
   rtc::Optional<bool> use_drift_compensation;
   rtc::Optional<bool> use_aec3;
+  rtc::Optional<bool> use_lc;
   rtc::Optional<int> aecm_routing_mode;
   rtc::Optional<bool> use_aecm_comfort_noise;
   rtc::Optional<int> agc_mode;
   rtc::Optional<int> agc_target_level;
   rtc::Optional<bool> use_agc_limiter;
   rtc::Optional<int> agc_compression_gain;
   rtc::Optional<int> vad_likelihood;
   rtc::Optional<int> ns_level;
   rtc::Optional<bool> use_refined_adaptive_filter;
   bool report_performance = false;
@@ skipping 97 matching lines @@
   std::unique_ptr<ChannelBufferWavWriter> reverse_buffer_writer_;
   TickIntervalStats proc_time_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioProcessingSimulator);
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_