Removed the inheritance from ProcessingComponent for EchoCancellerImpl.

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 67 matching lines @@
 
 // Throughout webrtc, it's assumed that success is represented by zero.
 static_assert(AudioProcessing::kNoError == 0, "kNoError must be zero");
 
 struct AudioProcessingImpl::ApmPublicSubmodules {
   ApmPublicSubmodules()
       : echo_cancellation(nullptr),
         echo_control_mobile(nullptr),
         gain_control(nullptr) {}
   // Accessed externally of APM without any lock acquired.
-  EchoCancellationImpl* echo_cancellation;
+  std::unique_ptr<EchoCancellationImpl> echo_cancellation;
   EchoControlMobileImpl* echo_control_mobile;
   GainControlImpl* gain_control;
   std::unique_ptr<HighPassFilterImpl> high_pass_filter;
   std::unique_ptr<LevelEstimatorImpl> level_estimator;
   std::unique_ptr<NoiseSuppressionImpl> noise_suppression;
   std::unique_ptr<VoiceDetectionImpl> voice_detection;
   std::unique_ptr<GainControlForExperimentalAgc>
       gain_control_for_experimental_agc;
 
   // Accessed internally from both render and capture.
@@ skipping 62 matching lines @@
       capture_(config.Get<ExperimentalNs>().enabled,
 #endif
                config.Get<Beamforming>().array_geometry,
                config.Get<Beamforming>().target_direction),
       capture_nonlocked_(config.Get<Beamforming>().enabled)
 {
   {
     rtc::CritScope cs_render(&crit_render_);
     rtc::CritScope cs_capture(&crit_capture_);
 
-    public_submodules_->echo_cancellation =
-        new EchoCancellationImpl(this, &crit_render_, &crit_capture_);
+    public_submodules_->echo_cancellation.reset(
+        new EchoCancellationImpl(this, &crit_render_, &crit_capture_));
     public_submodules_->echo_control_mobile =
         new EchoControlMobileImpl(this, &crit_render_, &crit_capture_);
     public_submodules_->gain_control =
         new GainControlImpl(this, &crit_capture_, &crit_capture_);
     public_submodules_->high_pass_filter.reset(
         new HighPassFilterImpl(&crit_capture_));
     public_submodules_->level_estimator.reset(
         new LevelEstimatorImpl(&crit_capture_));
     public_submodules_->noise_suppression.reset(
         new NoiseSuppressionImpl(&crit_capture_));
     public_submodules_->voice_detection.reset(
         new VoiceDetectionImpl(&crit_capture_));
     public_submodules_->gain_control_for_experimental_agc.reset(
         new GainControlForExperimentalAgc(public_submodules_->gain_control,
                                           &crit_capture_));
-
-    private_submodules_->component_list.push_back(
-        public_submodules_->echo_cancellation);
     private_submodules_->component_list.push_back(
         public_submodules_->echo_control_mobile);
     private_submodules_->component_list.push_back(
         public_submodules_->gain_control);
   }
 
   SetExtraOptions(config);
 }
 
 AudioProcessingImpl::~AudioProcessingImpl() {
@@ skipping 114 matching lines @@
                       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;
     }
   }
 
+  InitializeEchoCanceller();
   InitializeExperimentalAgc();
   InitializeTransient();
   InitializeBeamformer();
   InitializeIntelligibility();
   InitializeHighPassFilter();
   InitializeNoiseSuppression();
   InitializeLevelEstimator();
   InitializeVoiceDetection();
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
@@ skipping 81 matching lines @@
 }
 
 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();
   }
 
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
   if (capture_nonlocked_.beamformer_enabled !=
           config.Get<Beamforming>().enabled) {
@@ skipping 644 matching lines @@
   }
   return kNoError;
 #else
   return kUnsupportedFunctionError;
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 }
 
 EchoCancellation* AudioProcessingImpl::echo_cancellation() const {
   // Adding a lock here has no effect as it allows any access to the submodule
   // from the returned pointer.
-  return public_submodules_->echo_cancellation;
+  return public_submodules_->echo_cancellation.get();
 }
 
 EchoControlMobile* AudioProcessingImpl::echo_control_mobile() const {
   // Adding a lock here has no effect as it allows any access to the submodule
   // from the returned pointer.
   return public_submodules_->echo_control_mobile;
 }
 
 GainControl* AudioProcessingImpl::gain_control() const {
   // Adding a lock here has no effect as it allows any access to the submodule
@@ skipping 37 matching lines @@
     return true;
   }
 
   // All of the private submodules modify the data.
   for (auto item : private_submodules_->component_list) {
     if (item->is_component_enabled()) {
       return true;
     }
   }
 
+  if (public_submodules_->echo_cancellation->is_enabled()) {

[the sun, 2016/03/03 13:13:05]

I think you can put this in the first conditional in this function, with the hpf
and ns equivalents. Or make things consistent by having a separate condition for
each sub component.

[peah-webrtc, 2016/03/03 13:56:16]

Done.

+    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 84 matching lines @@
 void AudioProcessingImpl::InitializeHighPassFilter() {
   public_submodules_->high_pass_filter->Initialize(num_proc_channels(),
                                                    proc_sample_rate_hz());
 }
 
 void AudioProcessingImpl::InitializeNoiseSuppression() {
   public_submodules_->noise_suppression->Initialize(num_proc_channels(),
                                                     proc_sample_rate_hz());
 }
 
+void AudioProcessingImpl::InitializeEchoCanceller() {
+  public_submodules_->echo_cancellation->Initialize();
+}
+
 void AudioProcessingImpl::InitializeLevelEstimator() {
   public_submodules_->level_estimator->Initialize();
 }
 
 void AudioProcessingImpl::InitializeVoiceDetection() {
   public_submodules_->voice_detection->Initialize(proc_split_sample_rate_hz());
 }
 
 void AudioProcessingImpl::MaybeUpdateHistograms() {
   static const int kMinDiffDelayMs = 60;
@@ skipping 188 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