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

webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc

 /*
  *  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 423 matching lines @@
 
     if (settings_.use_refined_adaptive_filter) {
       config.Set<RefinedAdaptiveFilter>(
           new RefinedAdaptiveFilter(*settings_.use_refined_adaptive_filter));
     }
 
     if (settings_.use_aec3) {
       config.Set<EchoCanceller3>(new EchoCanceller3(*settings_.use_aec3));
     }
 
+    if (settings_.use_lc) {
+      config.Set<LevelControl>(new LevelControl(true));
+    }
+
     ap_->SetExtraOptions(config);
   }
 }
 
 void AecDumpBasedSimulator::HandleMessage(const webrtc::audioproc::Init& msg) {
   RTC_CHECK(msg.has_sample_rate());
   RTC_CHECK(msg.has_num_input_channels());
   RTC_CHECK(msg.has_num_reverse_channels());
   RTC_CHECK(msg.has_reverse_sample_rate());
 
@@ skipping 53 matching lines @@
 }
 
 void AecDumpBasedSimulator::HandleMessage(
     const webrtc::audioproc::ReverseStream& msg) {
   PrepareReverseProcessStreamCall(msg);
   ProcessReverseStream(interface_used_ == InterfaceType::kFixedInterface);
 }
 
 }  // namespace test
 }  // namespace webrtc