Compensate for the compression gain in the IntelligibilityEnhancer

webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc

 /*
  *  Copyright (c) 2014 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 91 matching lines @@
   size_t window_size = static_cast<size_t>(1) << RealFourier::FftOrder(freqs_);
   std::vector<float> kbd_window(window_size);
   WindowGenerator::KaiserBesselDerived(kKbdAlpha, window_size,
                                        kbd_window.data());
   render_mangler_.reset(new LappedTransform(
       num_render_channels_, num_render_channels_, chunk_length_,
       kbd_window.data(), window_size, window_size / 2, this));
 }
 
 void IntelligibilityEnhancer::SetCaptureNoiseEstimate(
-    std::vector<float> noise) {
+    std::vector<float> noise, int gain_db) {
   RTC_DCHECK_EQ(noise.size(), num_noise_bins_);
+  const float gain = std::pow(10.f, gain_db / 20.f);
+  for (auto& bin : noise) {
+    bin *= gain;
+  }
   // Disregarding return value since buffer overflow is acceptable, because it
   // is not critical to get each noise estimate.
   if (noise_estimation_queue_.Insert(&noise)) {
   };
 }
 
 void IntelligibilityEnhancer::ProcessRenderAudio(float* const* audio,
                                                  int sample_rate_hz,
                                                  size_t num_channels) {
   RTC_CHECK_EQ(sample_rate_hz_, sample_rate_hz);
@@ skipping 214 matching lines @@
   vad_.ProcessChunk(audio_s16_.data(), chunk_length_, sample_rate_hz_);
   if (vad_.last_voice_probability() > kVoiceProbabilityThreshold) {
     chunks_since_voice_ = 0;
   } else if (chunks_since_voice_ < kSpeechOffsetDelay) {
     ++chunks_since_voice_;
   }
   return chunks_since_voice_ < kSpeechOffsetDelay;
 }
 
 }  // namespace webrtc