Introduced new scheme for controlling the functionality inside the audio processing module

webrtc/modules/audio_processing/test/audio_processing_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 200 matching lines @@
 }
 
 void AudioProcessingSimulator::DestroyAudioProcessor() {
   if (settings_.aec_dump_output_filename) {
     RTC_CHECK_EQ(AudioProcessing::kNoError, ap_->StopDebugRecording());
   }
 }
 
 void AudioProcessingSimulator::CreateAudioProcessor() {
   Config config;
+  AudioProcessing::Config apm_config;
   if (settings_.use_bf && *settings_.use_bf) {
     config.Set<Beamforming>(new Beamforming(
         true, ParseArrayGeometry(*settings_.microphone_positions),
         SphericalPointf(DegreesToRadians(settings_.target_angle_degrees), 0.f,
                         1.f)));
   }
   if (settings_.use_ts) {
     config.Set<ExperimentalNs>(new ExperimentalNs(*settings_.use_ts));
   }
   if (settings_.use_ie) {
     config.Set<Intelligibility>(new Intelligibility(*settings_.use_ie));
   }
   if (settings_.use_aec3) {
     config.Set<EchoCanceller3>(new EchoCanceller3(*settings_.use_aec3));
   }
   if (settings_.use_lc) {
-    config.Set<LevelControl>(new LevelControl(true));
+    apm_config.level_controller.enabled = *settings_.use_lc;
   }
   if (settings_.use_refined_adaptive_filter) {
     config.Set<RefinedAdaptiveFilter>(
         new RefinedAdaptiveFilter(*settings_.use_refined_adaptive_filter));
   }
   config.Set<ExtendedFilter>(new ExtendedFilter(
       !settings_.use_extended_filter || *settings_.use_extended_filter));
   config.Set<DelayAgnostic>(new DelayAgnostic(!settings_.use_delay_agnostic ||
                                               *settings_.use_delay_agnostic));
 
   ap_.reset(AudioProcessing::Create(config));
+  RTC_CHECK(ap_);
+
+  ap_->ApplyConfig(apm_config);
 
   if (settings_.use_aec) {
     RTC_CHECK_EQ(AudioProcessing::kNoError,
                  ap_->echo_cancellation()->Enable(*settings_.use_aec));
   }
   if (settings_.use_aecm) {
     RTC_CHECK_EQ(AudioProcessing::kNoError,
                  ap_->echo_control_mobile()->Enable(*settings_.use_aecm));
   }
   if (settings_.use_agc) {
@@ skipping 80 matching lines @@
     size_t kMaxFilenameSize = AudioProcessing::kMaxFilenameSize;
     RTC_CHECK_LE(settings_.aec_dump_output_filename->size(), kMaxFilenameSize);
     RTC_CHECK_EQ(AudioProcessing::kNoError,
                  ap_->StartDebugRecording(
                      settings_.aec_dump_output_filename->c_str(), -1));
   }
 }
 
 }  // namespace test
 }  // namespace webrtc