Introduced the new parameter setting scheme for activating the high-pass filter in APM

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 10 matching lines @@
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
 #include "webrtc/modules/audio_processing/agc/agc_manager_direct.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #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"
 #if WEBRTC_INTELLIGIBILITY_ENHANCER
 #include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
 #endif
 #include "webrtc/modules/audio_processing/level_controller/level_controller.h"
 #include "webrtc/modules/audio_processing/level_estimator_impl.h"
+#include "webrtc/modules/audio_processing/low_cut_filter.h"
 #include "webrtc/modules/audio_processing/noise_suppression_impl.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.
@@ skipping 72 matching lines @@
   return uppermost_native_rate;
 }
 
 // Maximum length that a frame of samples can have.
 static const size_t kMaxAllowedValuesOfSamplesPerFrame = 160;
 // Maximum number of frames to buffer in the render queue.
 // TODO(peah): Decrease this once we properly handle hugely unbalanced
 // reverse and forward call numbers.
 static const size_t kMaxNumFramesToBuffer = 100;
 
+class HighPassFilterImpl : public HighPassFilter {
+ public:
+  explicit HighPassFilterImpl(AudioProcessingImpl* apm) : apm_(apm) {}
+  ~HighPassFilterImpl() override = default;
+
+  // HighPassFilter implementation.
+  int Enable(bool enable) override {
+    apm_->MutateConfig([enable](AudioProcessing::Config* config) {
+      config->high_pass_filter.enabled = enable;
+    });
+
+    return AudioProcessing::kNoError;
+  }
+
+  bool is_enabled() const override {
+    return apm_->GetConfig().high_pass_filter.enabled;
+  }
+
+ private:
+  AudioProcessingImpl* apm_;
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(HighPassFilterImpl);
+};
+
 }  // namespace
 
 // Throughout webrtc, it's assumed that success is represented by zero.
 static_assert(AudioProcessing::kNoError == 0, "kNoError must be zero");
 
 AudioProcessingImpl::ApmSubmoduleStates::ApmSubmoduleStates() {}
 
 bool AudioProcessingImpl::ApmSubmoduleStates::Update(
-    bool high_pass_filter_enabled,
+    bool low_cut_filter_enabled,
     bool echo_canceller_enabled,
     bool mobile_echo_controller_enabled,
     bool noise_suppressor_enabled,
     bool intelligibility_enhancer_enabled,
     bool beamformer_enabled,
     bool adaptive_gain_controller_enabled,
     bool level_controller_enabled,
     bool voice_activity_detector_enabled,
     bool level_estimator_enabled,
     bool transient_suppressor_enabled) {
   bool changed = false;
-  changed |= (high_pass_filter_enabled != high_pass_filter_enabled_);
+  changed |= (low_cut_filter_enabled != low_cut_filter_enabled_);
   changed |= (echo_canceller_enabled != echo_canceller_enabled_);
   changed |=
       (mobile_echo_controller_enabled != mobile_echo_controller_enabled_);
   changed |= (noise_suppressor_enabled != noise_suppressor_enabled_);
   changed |=
       (intelligibility_enhancer_enabled != intelligibility_enhancer_enabled_);
   changed |= (beamformer_enabled != beamformer_enabled_);
   changed |=
       (adaptive_gain_controller_enabled != adaptive_gain_controller_enabled_);
   changed |= (level_controller_enabled != level_controller_enabled_);
   changed |= (level_estimator_enabled != level_estimator_enabled_);
   changed |=
       (voice_activity_detector_enabled != voice_activity_detector_enabled_);
   changed |= (transient_suppressor_enabled != transient_suppressor_enabled_);
   if (changed) {
-    high_pass_filter_enabled_ = high_pass_filter_enabled;
+    low_cut_filter_enabled_ = low_cut_filter_enabled;
     echo_canceller_enabled_ = echo_canceller_enabled;
     mobile_echo_controller_enabled_ = mobile_echo_controller_enabled;
     noise_suppressor_enabled_ = noise_suppressor_enabled;
     intelligibility_enhancer_enabled_ = intelligibility_enhancer_enabled;
     beamformer_enabled_ = beamformer_enabled;
     adaptive_gain_controller_enabled_ = adaptive_gain_controller_enabled;
     level_controller_enabled_ = level_controller_enabled;
     level_estimator_enabled_ = level_estimator_enabled;
     voice_activity_detector_enabled_ = voice_activity_detector_enabled;
     transient_suppressor_enabled_ = transient_suppressor_enabled;
   }
 
   changed |= first_update_;
   first_update_ = false;
   return changed;
 }
 
 bool AudioProcessingImpl::ApmSubmoduleStates::CaptureMultiBandSubModulesActive()
     const {
 #if WEBRTC_INTELLIGIBILITY_ENHANCER
   return CaptureMultiBandProcessingActive() ||
          intelligibility_enhancer_enabled_ || voice_activity_detector_enabled_;
 #else
   return CaptureMultiBandProcessingActive() || voice_activity_detector_enabled_;
 #endif
 }
 
 bool AudioProcessingImpl::ApmSubmoduleStates::CaptureMultiBandProcessingActive()
     const {
-  return high_pass_filter_enabled_ || echo_canceller_enabled_ ||
+  return low_cut_filter_enabled_ || echo_canceller_enabled_ ||
          mobile_echo_controller_enabled_ || noise_suppressor_enabled_ ||
          beamformer_enabled_ || adaptive_gain_controller_enabled_;
 }
 
 bool AudioProcessingImpl::ApmSubmoduleStates::RenderMultiBandSubModulesActive()
     const {
   return RenderMultiBandProcessingActive() || echo_canceller_enabled_ ||
          mobile_echo_controller_enabled_ || adaptive_gain_controller_enabled_;
 }
 
 bool AudioProcessingImpl::ApmSubmoduleStates::RenderMultiBandProcessingActive()
     const {
 #if WEBRTC_INTELLIGIBILITY_ENHANCER
   return intelligibility_enhancer_enabled_;
 #else
   return false;
 #endif
 }
 
 struct AudioProcessingImpl::ApmPublicSubmodules {
   ApmPublicSubmodules() {}
   // Accessed externally of APM without any lock acquired.
   std::unique_ptr<EchoCancellationImpl> echo_cancellation;
   std::unique_ptr<EchoControlMobileImpl> echo_control_mobile;
   std::unique_ptr<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.
   std::unique_ptr<TransientSuppressor> transient_suppressor;
 #if WEBRTC_INTELLIGIBILITY_ENHANCER
   std::unique_ptr<IntelligibilityEnhancer> intelligibility_enhancer;
 #endif
 };
 
 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<LowCutFilter> low_cut_filter;
   std::unique_ptr<LevelController> level_controller;
 };
 
 AudioProcessing* AudioProcessing::Create() {
   webrtc::Config config;
   return Create(config, nullptr);
 }
 
 AudioProcessing* AudioProcessing::Create(const webrtc::Config& config) {
   return Create(config, nullptr);
 }
 
 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 webrtc::Config& config)
     : AudioProcessingImpl(config, nullptr) {}
 
 AudioProcessingImpl::AudioProcessingImpl(const webrtc::Config& config,
                                          NonlinearBeamformer* beamformer)
-    : public_submodules_(new ApmPublicSubmodules()),
+    : high_pass_filter_impl_(new HighPassFilterImpl(this)),
+      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) {
   {
     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_));
-    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.get(), &crit_capture_));
 
@@ skipping 148 matching lines @@
     private_submodules_->agc_manager->Initialize();
     private_submodules_->agc_manager->SetCaptureMuted(
         capture_.output_will_be_muted);
     public_submodules_->gain_control_for_experimental_agc->Initialize();
   }
   InitializeTransient();
   InitializeBeamformer();
 #if WEBRTC_INTELLIGIBILITY_ENHANCER
   InitializeIntelligibility();
 #endif
-  public_submodules_->high_pass_filter->Initialize(num_proc_channels(),
-                                                   proc_sample_rate_hz());
+  InitializeLowCutFilter();
   public_submodules_->noise_suppression->Initialize(num_proc_channels(),
                                                     proc_sample_rate_hz());
   public_submodules_->voice_detection->Initialize(proc_split_sample_rate_hz());
   public_submodules_->level_estimator->Initialize();
   InitializeLevelController();
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_dump_.debug_file->is_open()) {
     int err = WriteInitMessage();
     if (err != kNoError) {
@@ skipping 101 matching lines @@
     capture_nonlocked_.level_controller_enabled =
         config_.level_controller.enabled;
     // TODO(peah): Remove the conditional initialization to always initialize
     // the level controller regardless of whether it is enabled or not.
     InitializeLevelController();
   }
   LOG(LS_INFO) << "Level controller activated: "
                << capture_nonlocked_.level_controller_enabled;
 
   private_submodules_->level_controller->ApplyConfig(config_.level_controller);
+
+  InitializeLowCutFilter();
+
+  LOG(LS_INFO) << "Highpass filter activated: "
+               << config_.high_pass_filter.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 !=
@@ skipping 428 matching lines @@
     capture_buffer->SplitIntoFrequencyBands();
   }
 
   if (capture_nonlocked_.beamformer_enabled) {
     private_submodules_->beamformer->AnalyzeChunk(
         *capture_buffer->split_data_f());
     // Discards all channels by the leftmost one.
     capture_buffer->set_num_channels(1);
   }
 
-  public_submodules_->high_pass_filter->ProcessCaptureAudio(capture_buffer);
+  if (private_submodules_->low_cut_filter) {
+    private_submodules_->low_cut_filter->Process(capture_buffer);
+  }
   RETURN_ON_ERR(
       public_submodules_->gain_control->AnalyzeCaptureAudio(capture_buffer));
   public_submodules_->noise_suppression->AnalyzeCaptureAudio(capture_buffer);
 
   // Ensure that the stream delay was set before the call to the
   // AEC ProcessCaptureAudio function.
   if (public_submodules_->echo_cancellation->is_enabled() &&
       !was_stream_delay_set()) {
     return AudioProcessing::kStreamParameterNotSetError;
   }
@@ skipping 379 matching lines @@
 }
 
 GainControl* AudioProcessingImpl::gain_control() const {
   if (constants_.use_experimental_agc) {
     return public_submodules_->gain_control_for_experimental_agc.get();
   }
   return public_submodules_->gain_control.get();
 }
 
 HighPassFilter* AudioProcessingImpl::high_pass_filter() const {
-  return public_submodules_->high_pass_filter.get();
+  return high_pass_filter_impl_.get();
 }
 
 LevelEstimator* AudioProcessingImpl::level_estimator() const {
   return public_submodules_->level_estimator.get();
 }
 
 NoiseSuppression* AudioProcessingImpl::noise_suppression() const {
   return public_submodules_->noise_suppression.get();
 }
 
 VoiceDetection* AudioProcessingImpl::voice_detection() const {
   return public_submodules_->voice_detection.get();
 }
 
+void AudioProcessingImpl::MutateConfig(
+    std::function<void(AudioProcessing::Config*)> mutator) {
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
+  mutator(&config_);
+  ApplyConfig(config_);
+}
+
+AudioProcessing::Config AudioProcessingImpl::GetConfig() const {
+  rtc::CritScope cs_render(&crit_render_);
+  rtc::CritScope cs_capture(&crit_capture_);
+  return config_;
+}
+
 bool AudioProcessingImpl::UpdateActiveSubmoduleStates() {
   return submodule_states_.Update(
-      public_submodules_->high_pass_filter->is_enabled(),
+      config_.high_pass_filter.enabled,
       public_submodules_->echo_cancellation->is_enabled(),
       public_submodules_->echo_control_mobile->is_enabled(),
       public_submodules_->noise_suppression->is_enabled(),
       capture_nonlocked_.intelligibility_enabled,
       capture_nonlocked_.beamformer_enabled,
       public_submodules_->gain_control->is_enabled(),
       capture_nonlocked_.level_controller_enabled,
       public_submodules_->voice_detection->is_enabled(),
       public_submodules_->level_estimator->is_enabled(),
       capture_.transient_suppressor_enabled);
@@ skipping 27 matching lines @@
   if (capture_nonlocked_.intelligibility_enabled) {
     public_submodules_->intelligibility_enhancer.reset(
         new IntelligibilityEnhancer(capture_nonlocked_.split_rate,
                                     render_.render_audio->num_channels(),
                                     render_.render_audio->num_bands(),
                                     NoiseSuppressionImpl::num_noise_bins()));
   }
 #endif
 }
 
+void AudioProcessingImpl::InitializeLowCutFilter() {
+  if (config_.high_pass_filter.enabled) {
+    private_submodules_->low_cut_filter.reset(
+        new LowCutFilter(num_proc_channels(), proc_sample_rate_hz()));
+  } else {
+    private_submodules_->low_cut_filter.reset();
+  }
+}
+
 void AudioProcessingImpl::InitializeLevelController() {
   private_submodules_->level_controller->Initialize(proc_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.
@@ skipping 159 matching lines @@
   config.set_aecm_routing_mode(static_cast<int>(
       public_submodules_->echo_control_mobile->routing_mode()));
 
   config.set_agc_enabled(public_submodules_->gain_control->is_enabled());
   config.set_agc_mode(
       static_cast<int>(public_submodules_->gain_control->mode()));
   config.set_agc_limiter_enabled(
       public_submodules_->gain_control->is_limiter_enabled());
   config.set_noise_robust_agc_enabled(constants_.use_experimental_agc);
 
-  config.set_hpf_enabled(public_submodules_->high_pass_filter->is_enabled());
+  config.set_hpf_enabled(config_.high_pass_filter.enabled);
 
   config.set_ns_enabled(public_submodules_->noise_suppression->is_enabled());
   config.set_ns_level(
       static_cast<int>(public_submodules_->noise_suppression->level()));
 
   config.set_transient_suppression_enabled(
       capture_.transient_suppressor_enabled);
   config.set_intelligibility_enhancer_enabled(
       capture_nonlocked_.intelligibility_enabled);
 
@@ skipping 42 matching lines @@
       capture_processing_format(kSampleRate16kHz),
       split_rate(kSampleRate16kHz) {}
 
 AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
 
 AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
 
 AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
 
 }  // namespace webrtc