Deactivated the intelligibility enhancement functionality by default

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 12 matching lines @@
 #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/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.
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
 #else
 #include "webrtc/modules/audio_processing/debug.pb.h"
 #endif
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
+// Check to verify that the define for the intelligibility enhancer is properly
+// set.
+#if !defined(WEBRTC_INTELLIGIBILITY_ENHANCER) || \
+    (WEBRTC_INTELLIGIBILITY_ENHANCER != 0 &&     \
+     WEBRTC_INTELLIGIBILITY_ENHANCER != 1)
+#error "Set WEBRTC_INTELLIGIBILITY_ENHANCER to either 0 or 1"
+#endif
+
 #define RETURN_ON_ERR(expr) \
   do {                      \
     int err = (expr);       \
     if (err != kNoError) {  \
       return err;           \
     }                       \
   } while (0)
 
 namespace webrtc {
 
@@ skipping 54 matching lines @@
   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<LevelController> level_controller;
 };
@@ skipping 176 matching lines @@
                       capture_nonlocked_.fwd_proc_format.num_frames(),
                       fwd_audio_buffer_channels,
                       formats_.api_format.output_stream().num_frames()));
 
   InitializeGainController();
   InitializeEchoCanceller();
   InitializeEchoControlMobile();
   InitializeExperimentalAgc();
   InitializeTransient();
   InitializeBeamformer();
+#if WEBRTC_INTELLIGIBILITY_ENHANCER
   InitializeIntelligibility();
+#endif
   InitializeHighPassFilter();
   InitializeNoiseSuppression();
   InitializeLevelEstimator();
   InitializeVoiceDetection();
   InitializeLevelController();
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_dump_.debug_file->is_open()) {
     int err = WriteInitMessage();
     if (err != kNoError) {
@@ skipping 81 matching lines @@
   if (capture_nonlocked_.level_controller_enabled !=
       config.Get<LevelControl>().enabled) {
     capture_nonlocked_.level_controller_enabled =
         config.Get<LevelControl>().enabled;
     LOG(LS_INFO) << "Level controller activated: "
                  << config.Get<LevelControl>().enabled;
 
     InitializeLevelController();
   }
 
+#if WEBRTC_INTELLIGIBILITY_ENHANCER
   if(capture_nonlocked_.intelligibility_enabled !=
      config.Get<Intelligibility>().enabled) {
     capture_nonlocked_.intelligibility_enabled =
         config.Get<Intelligibility>().enabled;
     InitializeIntelligibility();
   }
+#endif
 
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
   if (capture_nonlocked_.beamformer_enabled !=
           config.Get<Beamforming>().enabled) {
     capture_nonlocked_.beamformer_enabled = config.Get<Beamforming>().enabled;
     if (config.Get<Beamforming>().array_geometry.size() > 1) {
       capture_.array_geometry = config.Get<Beamforming>().array_geometry;
     }
     capture_.target_direction = config.Get<Beamforming>().target_direction;
     InitializeBeamformer();
@@ skipping 276 matching lines @@
   }
 
   RETURN_ON_ERR(public_submodules_->echo_cancellation->ProcessCaptureAudio(
       ca, stream_delay_ms()));
 
   if (public_submodules_->echo_control_mobile->is_enabled() &&
       public_submodules_->noise_suppression->is_enabled()) {
     ca->CopyLowPassToReference();
   }
   public_submodules_->noise_suppression->ProcessCaptureAudio(ca);
+#if WEBRTC_INTELLIGIBILITY_ENHANCER
   if (capture_nonlocked_.intelligibility_enabled) {
     RTC_DCHECK(public_submodules_->noise_suppression->is_enabled());
     int gain_db = public_submodules_->gain_control->is_enabled() ?
                   public_submodules_->gain_control->compression_gain_db() :
                   0;
     float gain = std::pow(10.f, gain_db / 20.f);
     gain *= capture_nonlocked_.level_controller_enabled ?
             private_submodules_->level_controller->GetLastGain() :
             1.f;
     public_submodules_->intelligibility_enhancer->SetCaptureNoiseEstimate(
         public_submodules_->noise_suppression->NoiseEstimate(), gain);
   }
+#endif
 
   // Ensure that the stream delay was set before the call to the
   // AECM ProcessCaptureAudio function.
   if (public_submodules_->echo_control_mobile->is_enabled() &&
       !was_stream_delay_set()) {
     return AudioProcessing::kStreamParameterNotSetError;
   }
 
   RETURN_ON_ERR(public_submodules_->echo_control_mobile->ProcessCaptureAudio(
       ca, stream_delay_ms()));
@@ skipping 179 matching lines @@
   }
   return kNoError;
 }
 
 int AudioProcessingImpl::ProcessReverseStreamLocked() {
   AudioBuffer* ra = render_.render_audio.get();  // For brevity.
   if (rev_analysis_needed()) {
     ra->SplitIntoFrequencyBands();
   }
 
+#if WEBRTC_INTELLIGIBILITY_ENHANCER
   if (capture_nonlocked_.intelligibility_enabled) {
     public_submodules_->intelligibility_enhancer->ProcessRenderAudio(
         ra->split_channels_f(kBand0To8kHz), capture_nonlocked_.split_rate,
         ra->num_channels());
   }
+#endif
 
   RETURN_ON_ERR(public_submodules_->echo_cancellation->ProcessRenderAudio(ra));
   RETURN_ON_ERR(
       public_submodules_->echo_control_mobile->ProcessRenderAudio(ra));
   if (!constants_.use_experimental_agc) {
     RETURN_ON_ERR(public_submodules_->gain_control->ProcessRenderAudio(ra));
   }
 
   if (rev_synthesis_needed()) {
     ra->MergeFrequencyBands();
@@ skipping 211 matching lines @@
   } else if (is_multi_band(
                  capture_nonlocked_.fwd_proc_format.sample_rate_hz())) {
     // Something besides public_submodules_->level_estimator is enabled, and we
     // have super-wb.
     return true;
   }
   return false;
 }
 
 bool AudioProcessingImpl::is_rev_processed() const {
+#if WEBRTC_INTELLIGIBILITY_ENHANCER
   return capture_nonlocked_.intelligibility_enabled;
+#else
+  return false;
+#endif
 }
 
 bool AudioProcessingImpl::rev_synthesis_needed() const {
   return (is_rev_processed() &&
           is_multi_band(formats_.rev_proc_format.sample_rate_hz()));
 }
 
 bool AudioProcessingImpl::rev_analysis_needed() const {
   return is_multi_band(formats_.rev_proc_format.sample_rate_hz()) &&
          (is_rev_processed() ||
@@ skipping 44 matching lines @@
     if (!private_submodules_->beamformer) {
       private_submodules_->beamformer.reset(new NonlinearBeamformer(
           capture_.array_geometry, 1u, capture_.target_direction));
     }
     private_submodules_->beamformer->Initialize(kChunkSizeMs,
                                                 capture_nonlocked_.split_rate);
   }
 }
 
 void AudioProcessingImpl::InitializeIntelligibility() {
+#if WEBRTC_INTELLIGIBILITY_ENHANCER
   if (capture_nonlocked_.intelligibility_enabled) {
     public_submodules_->intelligibility_enhancer.reset(
         new IntelligibilityEnhancer(capture_nonlocked_.split_rate,
                                     render_.render_audio->num_channels(),
                                     NoiseSuppressionImpl::num_noise_bins()));
   }
+#endif
 }
 
 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());
@@ skipping 236 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