Removed the ProcessingComponent class

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 15 matching lines @@
 #include "webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h"
 #include "webrtc/modules/audio_processing/common.h"
 #include "webrtc/modules/audio_processing/echo_cancellation_impl.h"
 #include "webrtc/modules/audio_processing/echo_control_mobile_impl.h"
 #include "webrtc/modules/audio_processing/gain_control_for_experimental_agc.h"
 #include "webrtc/modules/audio_processing/gain_control_impl.h"
 #include "webrtc/modules/audio_processing/high_pass_filter_impl.h"
 #include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
 #include "webrtc/modules/audio_processing/level_estimator_impl.h"
 #include "webrtc/modules/audio_processing/noise_suppression_impl.h"
-#include "webrtc/modules/audio_processing/processing_component.h"
 #include "webrtc/modules/audio_processing/transient/transient_suppressor.h"
 #include "webrtc/modules/audio_processing/voice_detection_impl.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
 // Files generated at build-time by the protobuf compiler.
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
@@ skipping 50 matching lines @@
 
   // Accessed internally from both render and capture.
   std::unique_ptr<TransientSuppressor> transient_suppressor;
   std::unique_ptr<IntelligibilityEnhancer> intelligibility_enhancer;
 };
 
 struct AudioProcessingImpl::ApmPrivateSubmodules {
   explicit ApmPrivateSubmodules(Beamformer<float>* beamformer)
       : beamformer(beamformer) {}
   // Accessed internally from capture or during initialization
-  std::list<ProcessingComponent*> component_list;
   std::unique_ptr<Beamformer<float>> beamformer;
   std::unique_ptr<AgcManagerDirect> agc_manager;
 };
 
 const int AudioProcessing::kNativeSampleRatesHz[] = {
     AudioProcessing::kSampleRate8kHz,
     AudioProcessing::kSampleRate16kHz,
     AudioProcessing::kSampleRate32kHz,
     AudioProcessing::kSampleRate48kHz};
 const size_t AudioProcessing::kNumNativeSampleRates =
@@ skipping 71 matching lines @@
 
   SetExtraOptions(config);
 }
 
 AudioProcessingImpl::~AudioProcessingImpl() {
   // Depends on gain_control_ and
   // public_submodules_->gain_control_for_experimental_agc.
   private_submodules_->agc_manager.reset();
   // Depends on gain_control_.
   public_submodules_->gain_control_for_experimental_agc.reset();
-  while (!private_submodules_->component_list.empty()) {
-    ProcessingComponent* component =
-        private_submodules_->component_list.front();
-    component->Destroy();
-    delete component;
-    private_submodules_->component_list.pop_front();
-  }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_dump_.debug_file->Open()) {
     debug_dump_.debug_file->CloseFile();
   }
 #endif
 }
 
 int AudioProcessingImpl::Initialize() {
   // Run in a single-threaded manner during initialization.
@@ skipping 84 matching lines @@
     render_.render_audio.reset(nullptr);
     render_.render_converter.reset(nullptr);
   }
   capture_.capture_audio.reset(
       new AudioBuffer(formats_.api_format.input_stream().num_frames(),
                       formats_.api_format.input_stream().num_channels(),
                       capture_nonlocked_.fwd_proc_format.num_frames(),
                       fwd_audio_buffer_channels,
                       formats_.api_format.output_stream().num_frames()));
 
-  // Initialize all components.
-  for (auto item : private_submodules_->component_list) {
-    int err = item->Initialize();
-    if (err != kNoError) {
-      return err;
-    }
-  }
-
   InitializeGainController();
   InitializeEchoCanceller();
   InitializeEchoControlMobile();
   InitializeExperimentalAgc();
   InitializeTransient();
   InitializeBeamformer();
   InitializeIntelligibility();
   InitializeHighPassFilter();
   InitializeNoiseSuppression();
   InitializeLevelEstimator();
@@ skipping 80 matching lines @@
         capture_nonlocked_.fwd_proc_format.sample_rate_hz();
   }
 
   return InitializeLocked();
 }
 
 void AudioProcessingImpl::SetExtraOptions(const 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_);
-  for (auto item : private_submodules_->component_list) {
-    item->SetExtraOptions(config);
-  }
 
   public_submodules_->echo_cancellation->SetExtraOptions(config);
 
   if (capture_.transient_suppressor_enabled !=
       config.Get<ExperimentalNs>().enabled) {
     capture_.transient_suppressor_enabled =
         config.Get<ExperimentalNs>().enabled;
     InitializeTransient();
   }
 
@@ skipping 701 matching lines @@
   // modify the data.
   if (capture_nonlocked_.beamformer_enabled ||
       public_submodules_->high_pass_filter->is_enabled() ||
       public_submodules_->noise_suppression->is_enabled() ||
       public_submodules_->echo_cancellation->is_enabled() ||
       public_submodules_->echo_control_mobile->is_enabled() ||
       public_submodules_->gain_control->is_enabled()) {
     return true;
   }
 
-  // All of the private submodules modify the data.
-  for (auto item : private_submodules_->component_list) {
-    if (item->is_component_enabled()) {
-      return true;
-    }
-  }
-
   // The capture data is otherwise unchanged.
   return false;
 }
 
 bool AudioProcessingImpl::output_copy_needed(bool is_data_processed) const {
   // Check if we've upmixed or downmixed the audio.
   return ((formats_.api_format.output_stream().num_channels() !=
            formats_.api_format.input_stream().num_channels()) ||
           is_data_processed || capture_.transient_suppressor_enabled);
 }
@@ skipping 306 matching lines @@
   debug_dump_.capture.event_msg->mutable_config()->CopyFrom(config);
 
   RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
                                         &debug_dump_.num_bytes_left_for_log_,
                                         &crit_debug_, &debug_dump_.capture));
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc