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

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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 854 matching lines @@
     config.Set<webrtc::Intelligibility>(
         new webrtc::Intelligibility(*intelligibility_enhancer_));
   }
 
   if (options.level_control) {
     level_control_ = options.level_control;
   }
 
   LOG(LS_INFO) << "Level control: "
                << (!!level_control_ ? *level_control_ : -1);
+  webrtc::AudioProcessing::Config apm_config;
   if (level_control_) {
-    config.Set<webrtc::LevelControl>(new webrtc::LevelControl(*level_control_));
+    apm_config.level_controller.enabled = *level_control_;
   }
 
   // We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine
   // returns NULL on audio_processing().
   webrtc::AudioProcessing* audioproc = voe_wrapper_->base()->audio_processing();
   if (audioproc) {
     audioproc->SetExtraOptions(config);
+    RTC_DCHECK(apm_config.Validate());
+    RTC_DCHECK(audioproc->ApplyConfig(apm_config));

[hlundin-webrtc, 2016/09/06 08:56:03]

Using a DCHECK here means that ApplyConfig is only called in debug mode. Not
your intention, I suppose.

[peah-webrtc, 2016/09/07 05:42:59]

Ah, thanks!

Done.

   }
 
   if (options.recording_sample_rate) {
     LOG(LS_INFO) << "Recording sample rate is "
                  << *options.recording_sample_rate;
     if (adm()->SetRecordingSampleRate(*options.recording_sample_rate)) {
       LOG_RTCERR1(SetRecordingSampleRate, *options.recording_sample_rate);
     }
   }
 
@@ skipping 1769 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE