Compensate for the compression gain in the IntelligibilityEnhancer

webrtc/modules/audio_processing/gain_control_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 256 matching lines @@
       analog_capture_level_ += gain_controller->get_capture_level();
     }
 
     analog_capture_level_ /= (*num_proc_channels_);
   }
 
   was_analog_level_set_ = false;
   return AudioProcessing::kNoError;
 }
 
+int GainControlImpl::compression_gain_db() const {
+  rtc::CritScope cs(crit_capture_);
+  return compression_gain_db_;
+}
+
 // TODO(ajm): ensure this is called under kAdaptiveAnalog.
 int GainControlImpl::set_stream_analog_level(int level) {
   rtc::CritScope cs(crit_capture_);
 
   was_analog_level_set_ = true;
   if (level < minimum_capture_level_ || level > maximum_capture_level_) {
     return AudioProcessing::kBadParameterError;
   }
   analog_capture_level_ = level;
 
@@ skipping 119 matching lines @@
   if (gain < 0 || gain > 90) {
     return AudioProcessing::kBadParameterError;
   }
   {
     rtc::CritScope cs(crit_capture_);
     compression_gain_db_ = gain;
   }
   return Configure();
 }
 
-int GainControlImpl::compression_gain_db() const {
-  rtc::CritScope cs(crit_capture_);
-  return compression_gain_db_;
-}
-
 int GainControlImpl::enable_limiter(bool enable) {
   {
     rtc::CritScope cs(crit_capture_);
     limiter_enabled_ = enable;
   }
   return Configure();
 }
 
 bool GainControlImpl::is_limiter_enabled() const {
   rtc::CritScope cs(crit_capture_);
@@ skipping 67 matching lines @@
   for (auto& gain_controller : gain_controllers_) {
     const int handle_error =
         WebRtcAgc_set_config(gain_controller->state(), config);
     if (handle_error != AudioProcessing::kNoError) {
       error = handle_error;
     }
   }
   return error;
 }
 }  // namespace webrtc