Added handling of 'agc_compression_gain' flag in audioproc_f.

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 267 matching lines @@
   }
 
   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));
+  config.Set<ExperimentalAgc>(new ExperimentalAgc(
+      !settings_.use_experimental_agc || *settings_.use_experimental_agc));
   if (settings_.use_ed) {
     apm_config.residual_echo_detector.enabled = *settings_.use_ed;
   }
 
   ap_.reset(AudioProcessing::Create(config));
   RTC_CHECK(ap_);
 
   ap_->ApplyConfig(apm_config);
 
   if (settings_.use_aec) {
@@ skipping 22 matching lines @@
   }
   if (settings_.use_agc_limiter) {
     RTC_CHECK_EQ(AudioProcessing::kNoError, ap_->gain_control()->enable_limiter(
                                                 *settings_.use_agc_limiter));
   }
   if (settings_.agc_target_level) {
     RTC_CHECK_EQ(AudioProcessing::kNoError,
                  ap_->gain_control()->set_target_level_dbfs(
                      *settings_.agc_target_level));
   }
-
+  if (settings_.agc_compression_gain) {
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 ap_->gain_control()->set_compression_gain_db(
+                     *settings_.agc_compression_gain));
+  }
   if (settings_.agc_mode) {
     RTC_CHECK_EQ(
         AudioProcessing::kNoError,
         ap_->gain_control()->set_mode(
             static_cast<webrtc::GainControl::Mode>(*settings_.agc_mode)));
   }
 
   if (settings_.use_drift_compensation) {
     RTC_CHECK_EQ(AudioProcessing::kNoError,
                  ap_->echo_cancellation()->enable_drift_compensation(
@@ skipping 41 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