HighPassFilter not a ProcessingComponent anymore (bit exact).

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 71 matching lines @@
         echo_control_mobile(nullptr),
         gain_control(nullptr),
         high_pass_filter(nullptr),
         level_estimator(nullptr),
         noise_suppression(nullptr),
         voice_detection(nullptr) {}
   // Accessed externally of APM without any lock acquired.
   EchoCancellationImpl* echo_cancellation;
   EchoControlMobileImpl* echo_control_mobile;
   GainControlImpl* gain_control;
-  HighPassFilterImpl* high_pass_filter;
+  rtc::scoped_ptr<HighPassFilterImpl> high_pass_filter;
   LevelEstimatorImpl* level_estimator;
   NoiseSuppressionImpl* noise_suppression;
   VoiceDetectionImpl* voice_detection;
   rtc::scoped_ptr<GainControlForNewAgc> gain_control_for_new_agc;
 
   // Accessed internally from both render and capture.
   rtc::scoped_ptr<TransientSuppressor> transient_suppressor;
   rtc::scoped_ptr<IntelligibilityEnhancer> intelligibility_enhancer;
 };
 
@@ skipping 133 matching lines @@
   {
     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_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 =
-        new HighPassFilterImpl(this, &crit_capture_);
+    public_submodules_->high_pass_filter.reset(
+        new HighPassFilterImpl(&crit_capture_));
     public_submodules_->level_estimator =
         new LevelEstimatorImpl(this, &crit_capture_);
     public_submodules_->noise_suppression =
         new NoiseSuppressionImpl(this, &crit_capture_);
     public_submodules_->voice_detection =
         new VoiceDetectionImpl(this, &crit_capture_);
     public_submodules_->gain_control_for_new_agc.reset(
         new GainControlForNewAgc(public_submodules_->gain_control));
 
     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);
     private_submodules_->component_list.push_back(
-        public_submodules_->high_pass_filter);
-    private_submodules_->component_list.push_back(
         public_submodules_->level_estimator);
     private_submodules_->component_list.push_back(
         public_submodules_->noise_suppression);
     private_submodules_->component_list.push_back(
         public_submodules_->voice_detection);
   }
 
   SetExtraOptions(config);
 }
 
@@ skipping 123 matching lines @@
   }
 
   InitializeExperimentalAgc();
 
   InitializeTransient();
 
   InitializeBeamformer();
 
   InitializeIntelligibility();
 
+  InitializeHighPassFilter();
+
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_dump_.debug_file->Open()) {
     int err = WriteInitMessage();
     if (err != kNoError) {
       return err;
     }
   }
 #endif
 
   return kNoError;
@@ skipping 341 matching lines @@
         ca->split_channels_f(kBand0To8kHz), capture_nonlocked_.split_rate,
         ca->num_channels());
   }
 
   if (constants_.beamformer_enabled) {
     private_submodules_->beamformer->ProcessChunk(*ca->split_data_f(),
                                                   ca->split_data_f());
     ca->set_num_channels(1);
   }
 
-  RETURN_ON_ERR(public_submodules_->high_pass_filter->ProcessCaptureAudio(ca));
+  public_submodules_->high_pass_filter->ProcessCaptureAudio(ca);
   RETURN_ON_ERR(public_submodules_->gain_control->AnalyzeCaptureAudio(ca));
   RETURN_ON_ERR(public_submodules_->noise_suppression->AnalyzeCaptureAudio(ca));
   RETURN_ON_ERR(public_submodules_->echo_cancellation->ProcessCaptureAudio(ca));
 
   if (public_submodules_->echo_control_mobile->is_enabled() &&
       public_submodules_->noise_suppression->is_enabled()) {
     ca->CopyLowPassToReference();
   }
   RETURN_ON_ERR(public_submodules_->noise_suppression->ProcessCaptureAudio(ca));
   RETURN_ON_ERR(
@@ skipping 359 matching lines @@
   // from the returned pointer.
   if (constants_.use_new_agc) {
     return public_submodules_->gain_control_for_new_agc.get();
   }
   return public_submodules_->gain_control;
 }
 
 HighPassFilter* AudioProcessingImpl::high_pass_filter() const {
   // Adding a lock here has no effect as it allows any access to the submodule
   // from the returned pointer.
-  return public_submodules_->high_pass_filter;
+  return public_submodules_->high_pass_filter.get();
 }
 
 LevelEstimator* AudioProcessingImpl::level_estimator() const {
   // Adding a lock here has no effect as it allows any access to the submodule
   // from the returned pointer.
   return public_submodules_->level_estimator;
 }
 
 NoiseSuppression* AudioProcessingImpl::noise_suppression() const {
   // Adding a lock here has no effect as it allows any access to the submodule
@@ skipping 125 matching lines @@
   if (constants_.intelligibility_enabled) {
     IntelligibilityEnhancer::Config config;
     config.sample_rate_hz = capture_nonlocked_.split_rate;
     config.num_capture_channels = capture_.capture_audio->num_channels();
     config.num_render_channels = render_.render_audio->num_channels();
     public_submodules_->intelligibility_enhancer.reset(
         new IntelligibilityEnhancer(config));
   }
 }
 
+void AudioProcessingImpl::InitializeHighPassFilter() {
+  int channels = num_output_channels();

[peah-webrtc, 2015/12/08 14:52:27]

Why are the number of output channels and sample rate cached in local variables
before doing the function call?

(and if there is a need to do that, I guess they should be const).

[the sun, 2015/12/08 15:08:05]

Done.

+  int sample_rate_hz = proc_sample_rate_hz();
+  public_submodules_->high_pass_filter->Initialize(channels, sample_rate_hz);
+}
+
 void AudioProcessingImpl::MaybeUpdateHistograms() {
   static const int kMinDiffDelayMs = 60;
 
   if (echo_cancellation()->is_enabled()) {
     // Activate delay_jumps_ counters if we know echo_cancellation is runnning.
     // If a stream has echo we know that the echo_cancellation is in process.
     if (capture_.stream_delay_jumps == -1 &&
         echo_cancellation()->stream_has_echo()) {
       capture_.stream_delay_jumps = 0;
     }
@@ skipping 164 matching lines @@
   debug_dump_.capture.event_msg->set_type(audioproc::Event::CONFIG);
   debug_dump_.capture.event_msg->mutable_config()->CopyFrom(config);
 
   RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
                                         &crit_debug_, &debug_dump_.capture));
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc