AGC2 as a new APM sub-module operating with hard-coded gain.

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 252 matching lines @@
   }
   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) {
     apm_config.echo_canceller3.enabled = *settings_.use_aec3;
   }
+  if (settings_.use_agc2) {
+    apm_config.gain_controller2.enabled = *settings_.use_agc2;
+  }
   if (settings_.use_lc) {
     apm_config.level_controller.enabled = *settings_.use_lc;
   }
   if (settings_.use_hpf) {
     apm_config.high_pass_filter.enabled = *settings_.use_hpf;
   }
 
   if (settings_.use_refined_adaptive_filter) {
     config.Set<RefinedAdaptiveFilter>(
         new RefinedAdaptiveFilter(*settings_.use_refined_adaptive_filter));
@@ skipping 105 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