Compensate for the compression gain in the IntelligibilityEnhancer

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 688 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 (constants_.intelligibility_enabled) {
     RTC_DCHECK(public_submodules_->noise_suppression->is_enabled());
+    RTC_DCHECK(public_submodules_->gain_control->is_enabled());
     public_submodules_->intelligibility_enhancer->SetCaptureNoiseEstimate(
-        public_submodules_->noise_suppression->NoiseEstimate());
+        public_submodules_->noise_suppression->NoiseEstimate(),
+        public_submodules_->gain_control->compression_gain_db());
   }
 
   // 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(
@@ skipping 738 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