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

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 136 matching lines @@
 struct AudioProcessingImpl::ApmPrivateSubmodules {
   explicit ApmPrivateSubmodules(NonlinearBeamformer* beamformer)
       : beamformer(beamformer) {}
   // Accessed internally from capture or during initialization
   std::unique_ptr<NonlinearBeamformer> beamformer;
   std::unique_ptr<AgcManagerDirect> agc_manager;
   std::unique_ptr<LevelController> level_controller;
 };
 
 AudioProcessing* AudioProcessing::Create() {
-  Config config;
+  webrtc::Config config;
   return Create(config, nullptr);
 }
 
-AudioProcessing* AudioProcessing::Create(const Config& config) {
+AudioProcessing* AudioProcessing::Create(const webrtc::Config& config) {
   return Create(config, nullptr);
 }
 
-AudioProcessing* AudioProcessing::Create(const Config& config,
+AudioProcessing* AudioProcessing::Create(const webrtc::Config& config,
                                          NonlinearBeamformer* beamformer) {
   AudioProcessingImpl* apm = new AudioProcessingImpl(config, beamformer);
   if (apm->Initialize() != kNoError) {
     delete apm;
     apm = nullptr;
   }
 
   return apm;
 }
 
-AudioProcessingImpl::AudioProcessingImpl(const Config& config)
+AudioProcessingImpl::AudioProcessingImpl(const webrtc::Config& config)
     : AudioProcessingImpl(config, nullptr) {}
 
-AudioProcessingImpl::AudioProcessingImpl(const Config& config,
+AudioProcessingImpl::AudioProcessingImpl(const webrtc::Config& config,
                                          NonlinearBeamformer* beamformer)
     : public_submodules_(new ApmPublicSubmodules()),
       private_submodules_(new ApmPrivateSubmodules(beamformer)),
       constants_(config.Get<ExperimentalAgc>().startup_min_volume,
 #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
                config.Get<Beamforming>().array_geometry,
                config.Get<Beamforming>().target_direction),
       capture_nonlocked_(config.Get<Beamforming>().enabled,
-                         config.Get<Intelligibility>().enabled,
-                         config.Get<LevelControl>().enabled) {
+                         config.Get<Intelligibility>().enabled) {
   {
     rtc::CritScope cs_render(&crit_render_);
     rtc::CritScope cs_capture(&crit_capture_);
 
     public_submodules_->echo_cancellation.reset(
         new EchoCancellationImpl(&crit_render_, &crit_capture_));
     public_submodules_->echo_control_mobile.reset(
         new EchoControlMobileImpl(&crit_render_, &crit_capture_));
     public_submodules_->gain_control.reset(
         new GainControlImpl(&crit_capture_, &crit_capture_));
@@ skipping 216 matching lines @@
       capture_nonlocked_.fwd_proc_format.sample_rate_hz() == kSampleRate48kHz) {
     capture_nonlocked_.split_rate = kSampleRate16kHz;
   } else {
     capture_nonlocked_.split_rate =
         capture_nonlocked_.fwd_proc_format.sample_rate_hz();
   }
 
   return InitializeLocked();
 }
 
-void AudioProcessingImpl::SetExtraOptions(const Config& config) {
+void AudioProcessingImpl::ApplyConfig(const AudioProcessing::Config& config) {
+  AudioProcessing::Config config_to_use = config;
+
+  bool config_ok = LevelController::Validate(config_to_use.level_controller);
+  if (!config_ok) {
+    LOG(LS_ERROR) << "AudioProcessing module config error" << std::endl
+                  << "level_controller: "
+                  << LevelController::ToString(config_to_use.level_controller)
+                  << std::endl
+                  << "Reverting to default parameter set";
+    config_to_use.level_controller = AudioProcessing::Config::LevelController();
+  }
+
+  // Run in a single-threaded manner when applying the settings.
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
+
+  if (config.level_controller.enabled !=
+      capture_nonlocked_.level_controller_enabled) {
+    InitializeLevelController();
+    LOG(LS_INFO) << "Level controller activated: "
+                 << capture_nonlocked_.level_controller_enabled;
+    capture_nonlocked_.level_controller_enabled =
+        config.level_controller.enabled;
+  }
+}
+
+void AudioProcessingImpl::SetExtraOptions(const webrtc::Config& config) {
   // Run in a single-threaded manner when setting the extra options.
   rtc::CritScope cs_render(&crit_render_);
   rtc::CritScope cs_capture(&crit_capture_);
 
   public_submodules_->echo_cancellation->SetExtraOptions(config);
 
   if (capture_.transient_suppressor_enabled !=
       config.Get<ExperimentalNs>().enabled) {
     capture_.transient_suppressor_enabled =
         config.Get<ExperimentalNs>().enabled;
     InitializeTransient();
   }
 
-  if (capture_nonlocked_.level_controller_enabled !=
-      config.Get<LevelControl>().enabled) {
-    capture_nonlocked_.level_controller_enabled =
-        config.Get<LevelControl>().enabled;
-    LOG(LS_INFO) << "Level controller activated: "
-                 << config.Get<LevelControl>().enabled;
-
-    InitializeLevelController();
-  }
-
 #if WEBRTC_INTELLIGIBILITY_ENHANCER
   if(capture_nonlocked_.intelligibility_enabled !=
      config.Get<Intelligibility>().enabled) {
     capture_nonlocked_.intelligibility_enabled =
         config.Get<Intelligibility>().enabled;
     InitializeIntelligibility();
   }
 #endif
 
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
@@ skipping 1092 matching lines @@
       fwd_proc_format(kSampleRate16kHz),
       split_rate(kSampleRate16kHz) {}
 
 AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
 
 AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
 
 AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
 
 }  // namespace webrtc