AGC: Route clipping parameter from webrtc::Config to AGC

webrtc/modules/audio_processing/audio_processing_impl.cc

 /*
  *  Copyright (c) 2012 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 284 matching lines @@
 
 AudioProcessingImpl::AudioProcessingImpl(const webrtc::Config& config)
     : AudioProcessingImpl(config, nullptr) {}
 
 AudioProcessingImpl::AudioProcessingImpl(const webrtc::Config& config,
                                          NonlinearBeamformer* beamformer)
     : high_pass_filter_impl_(new HighPassFilterImpl(this)),
       public_submodules_(new ApmPublicSubmodules()),
       private_submodules_(new ApmPrivateSubmodules(beamformer)),
       constants_(config.Get<ExperimentalAgc>().startup_min_volume,
+                 config.Get<ExperimentalAgc>().clipped_level_min,
 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
                  false),
 #else
                  config.Get<ExperimentalAgc>().enabled),
 #endif
 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
       capture_(false,
 #else
       capture_(config.Get<ExperimentalNs>().enabled,
 #endif
@@ skipping 166 matching lines @@
       proc_split_sample_rate_hz(), num_reverse_channels(),
       num_output_channels());
 
   public_submodules_->gain_control->Initialize(num_proc_channels(),
                                                proc_sample_rate_hz());
   if (constants_.use_experimental_agc) {
     if (!private_submodules_->agc_manager.get()) {
       private_submodules_->agc_manager.reset(new AgcManagerDirect(
           public_submodules_->gain_control.get(),
           public_submodules_->gain_control_for_experimental_agc.get(),
-          constants_.agc_startup_min_volume));
+          constants_.agc_startup_min_volume, constants_.agc_clipped_level_min));
     }
     private_submodules_->agc_manager->Initialize();
     private_submodules_->agc_manager->SetCaptureMuted(
         capture_.output_will_be_muted);
     public_submodules_->gain_control_for_experimental_agc->Initialize();
   }
   InitializeTransient();
   InitializeBeamformer();
 #if WEBRTC_INTELLIGIBILITY_ENHANCER
   InitializeIntelligibility();
@@ skipping 1344 matching lines @@
   config.set_intelligibility_enhancer_enabled(
       capture_nonlocked_.intelligibility_enabled);
 
   std::string experiments_description =
       public_submodules_->echo_cancellation->GetExperimentsDescription();
   // TODO(peah): Add semicolon-separated concatenations of experiment
   // descriptions for other submodules.
   if (capture_nonlocked_.level_controller_enabled) {
     experiments_description += "LevelController;";
   }
+  if (constants_.agc_clipped_level_min != kClippedLevelMin) {
+    experiments_description += "AgcClippingLevelExperiment;";
+  }
   config.set_experiments_description(experiments_description);
 
   std::string serialized_config = config.SerializeAsString();
   if (!forced &&
       debug_dump_.capture.last_serialized_config == serialized_config) {
     return kNoError;
   }
 
   debug_dump_.capture.last_serialized_config = serialized_config;
 
@@ skipping 25 matching lines @@
       capture_processing_format(kSampleRate16kHz),
       split_rate(kSampleRate16kHz) {}
 
 AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
 
 AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
 
 AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
 
 }  // namespace webrtc