Introduced new scheme for controlling the functionality inside 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 218 matching lines @@
     std::cout << " Forward: " << get_num_process_stream_calls() << std::endl;
     std::cout << " Reverse: " << get_num_reverse_process_stream_calls()
               << std::endl;
   }
 
   if (!settings_.discard_all_settings_in_aecdump) {
     if (settings_.use_verbose_logging) {
       std::cout << "Setting used in config:" << std::endl;
     }
     Config config;
+    AudioProcessing::Config apm_config;
 
     if (msg.has_aec_enabled() || settings_.use_aec) {
       bool enable = settings_.use_aec ? *settings_.use_aec : msg.aec_enabled();
       RTC_CHECK_EQ(AudioProcessing::kNoError,
                    ap_->echo_cancellation()->Enable(enable));
       if (settings_.use_verbose_logging) {
         std::cout << " aec_enabled: " << (enable ? "true" : "false")
                   << std::endl;
       }
     }
@@ skipping 186 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));
+      apm_config.level_controller.enabled = true;
     }
 
+    RTC_DCHECK(apm_config.Validate());

[the sun, 2016/09/01 13:48:01]

dd

[peah-webrtc, 2016/09/02 08:22:01]

Done.

+    RTC_CHECK(ap_->ApplyConfig(apm_config));
     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