Deactivated the intelligibility enhancement functionality by default

webrtc/modules/audio_processing/audio_processing_impl.cc

Index: webrtc/modules/audio_processing/audio_processing_impl.cc
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index 3b3a9518b1723497b6e352b6f729e26e096ae2d1..69a5df40d39320e34aa82c3154a1fd045c5f0757 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -30,7 +30,9 @@
 #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 == 1)

[the sun, 2016/08/26 12:08:56]

#if WEBRTC_INTELLIGIBILITY_ENHANCER
should be enough

[peah-webrtc, 2016/08/26 13:09:43]

Done, + in the other places in this file.

 #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"
@@ -50,6 +52,14 @@
 #endif
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
+// Check to verify that the define for the intelligibility enhancer is properly
+// set.
+#if !defined(WEBRTC_INTELLIGIBILITY_ENHANCER) || \

[the sun, 2016/08/26 12:08:56]

Don't think you need this

[peah-webrtc, 2016/08/26 13:09:43]

You mean because this is a source file and not a header file?

The purpose is to enforce this to be explicitly set everywhere, so that it by
mistake is not specified in the build system. 

E.g., omitting specifying it properly in the build system should not cause a
silent error by causing the functionality to be turned off.

+    (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);       \
@@ -124,7 +134,9 @@ struct AudioProcessingImpl::ApmPublicSubmodules {
 
   // Accessed internally from both render and capture.
   std::unique_ptr<TransientSuppressor> transient_suppressor;
+#if (WEBRTC_INTELLIGIBILITY_ENHANCER == 1)
   std::unique_ptr<IntelligibilityEnhancer> intelligibility_enhancer;
+#endif
 };
 
 struct AudioProcessingImpl::ApmPrivateSubmodules {
@@ -321,7 +333,9 @@ int AudioProcessingImpl::InitializeLocked() {
   InitializeExperimentalAgc();
   InitializeTransient();
   InitializeBeamformer();
+#if (WEBRTC_INTELLIGIBILITY_ENHANCER == 1)
   InitializeIntelligibility();
+#endif
   InitializeHighPassFilter();
   InitializeNoiseSuppression();
   InitializeLevelEstimator();
@@ -423,12 +437,14 @@ void AudioProcessingImpl::SetExtraOptions(const Config& config) {
     InitializeLevelController();
   }
 
+#if (WEBRTC_INTELLIGIBILITY_ENHANCER == 1)
   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 !=
@@ -725,6 +741,7 @@ int AudioProcessingImpl::ProcessStreamLocked() {
     ca->CopyLowPassToReference();
   }
   public_submodules_->noise_suppression->ProcessCaptureAudio(ca);
+#if (WEBRTC_INTELLIGIBILITY_ENHANCER == 1)
   if (capture_nonlocked_.intelligibility_enabled) {
     RTC_DCHECK(public_submodules_->noise_suppression->is_enabled());
     int gain_db = public_submodules_->gain_control->is_enabled() ?
@@ -737,6 +754,7 @@ int AudioProcessingImpl::ProcessStreamLocked() {
     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.
@@ -936,11 +954,13 @@ int AudioProcessingImpl::ProcessReverseStreamLocked() {
     ra->SplitIntoFrequencyBands();
   }
 
+#if (WEBRTC_INTELLIGIBILITY_ENHANCER == 1)
   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(
@@ -1172,7 +1192,11 @@ bool AudioProcessingImpl::fwd_analysis_needed() const {
 }
 
 bool AudioProcessingImpl::is_rev_processed() const {
+#if (WEBRTC_INTELLIGIBILITY_ENHANCER == 1)
   return capture_nonlocked_.intelligibility_enabled;
+#else
+  return false;
+#endif
 }
 
 bool AudioProcessingImpl::rev_synthesis_needed() const {
@@ -1237,12 +1261,14 @@ void AudioProcessingImpl::InitializeBeamformer() {
 }
 
 void AudioProcessingImpl::InitializeIntelligibility() {
+#if (WEBRTC_INTELLIGIBILITY_ENHANCER == 1)
   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() {