New module for the adaptive level controlling functionality in 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.
  */
 
 #include "webrtc/modules/audio_processing/audio_processing_impl.h"
 
 #include <assert.h>
 #include <algorithm>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/platform_file.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/common_audio/audio_converter.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #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/level_controller/level_controller.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"
 #include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
+#include "webrtc/modules/audio_processing/level_controller/level_controller.h"
 #include "webrtc/modules/audio_processing/level_estimator_impl.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
@@ skipping 81 matching lines @@
   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::unique_ptr<Beamformer<float>> beamformer;
   std::unique_ptr<AgcManagerDirect> agc_manager;
+  std::unique_ptr<LevelController> level_controller;
 };
 
 AudioProcessing* AudioProcessing::Create() {
   Config config;
   return Create(config, nullptr);
 }
 
 AudioProcessing* AudioProcessing::Create(const Config& config) {
   return Create(config, nullptr);
 }
@@ skipping 23 matching lines @@
                  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<Intelligibility>().enabled,
+                         config.Get<LevelControl>().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_));
+
+    private_submodules_->level_controller.reset(new LevelController());
   }
 
   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_.
@@ skipping 104 matching lines @@
   InitializeEchoCanceller();
   InitializeEchoControlMobile();
   InitializeExperimentalAgc();
   InitializeTransient();
   InitializeBeamformer();
   InitializeIntelligibility();
   InitializeHighPassFilter();
   InitializeNoiseSuppression();
   InitializeLevelEstimator();
   InitializeVoiceDetection();
+  InitializeLevelController();
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_dump_.debug_file->is_open()) {
     int err = WriteInitMessage();
     if (err != kNoError) {
       return err;
     }
   }
 #endif
 
@@ skipping 66 matching lines @@
 
   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;
+    InitializeLevelController();
+  }
+
   if(capture_nonlocked_.intelligibility_enabled !=
      config.Get<Intelligibility>().enabled) {
     capture_nonlocked_.intelligibility_enabled =
         config.Get<Intelligibility>().enabled;
     InitializeIntelligibility();
   }
 
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
   if (capture_nonlocked_.beamformer_enabled !=
           config.Get<Beamforming>().enabled) {
@@ skipping 331 matching lines @@
             ? private_submodules_->agc_manager->voice_probability()
             : 1.f;
 
     public_submodules_->transient_suppressor->Suppress(
         ca->channels_f()[0], ca->num_frames(), ca->num_channels(),
         ca->split_bands_const_f(0)[kBand0To8kHz], ca->num_frames_per_band(),
         ca->keyboard_data(), ca->num_keyboard_frames(), voice_probability,
         capture_.key_pressed);
   }
 
+  if (capture_nonlocked_.level_controller_enabled) {
+    private_submodules_->level_controller->Process(ca);
+  }
+
   // The level estimator operates on the recombined data.
   public_submodules_->level_estimator->ProcessStream(ca);
 
   capture_.was_stream_delay_set = false;
   return kNoError;
 }
 
 int AudioProcessingImpl::AnalyzeReverseStream(const float* const* data,
                                               size_t samples_per_channel,
                                               int rev_sample_rate_hz,
@@ skipping 339 matching lines @@
   }
 
   // The capture data is otherwise unchanged.
   return false;
 }
 
 bool AudioProcessingImpl::output_copy_needed() 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_fwd_processed() || capture_.transient_suppressor_enabled);
+          is_fwd_processed() || capture_.transient_suppressor_enabled ||
+          capture_nonlocked_.level_controller_enabled);
 }
 
 bool AudioProcessingImpl::fwd_synthesis_needed() const {
   return (is_fwd_processed() &&
           is_multi_band(capture_nonlocked_.fwd_proc_format.sample_rate_hz()));
 }
 
 bool AudioProcessingImpl::fwd_analysis_needed() const {
   if (!is_fwd_processed() &&
       !public_submodules_->voice_detection->is_enabled() &&
@@ skipping 108 matching lines @@
   public_submodules_->echo_control_mobile->Initialize(
       proc_split_sample_rate_hz(),
       num_reverse_channels(),
       num_output_channels());
 }
 
 void AudioProcessingImpl::InitializeLevelEstimator() {
   public_submodules_->level_estimator->Initialize();
 }
 
+void AudioProcessingImpl::InitializeLevelController() {
+  private_submodules_->level_controller->Initialize(proc_sample_rate_hz(),
+                                                    num_proc_channels());
+}
+
 void AudioProcessingImpl::InitializeVoiceDetection() {
   public_submodules_->voice_detection->Initialize(proc_split_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 195 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