Compensate for the compression gain in the IntelligibilityEnhancer

webrtc/modules/audio_processing/gain_control_impl.h

 /*
  *  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 33 matching lines @@
   Mode mode() const override;
 
   // Checks whether the module is enabled. Must only be
   // called from the render side of APM as otherwise
   // deadlocks may occur.
   bool is_enabled_render_side_query() const;
 
   // Reads render side data that has been queued on the render call.
   void ReadQueuedRenderData();
 
+  int compression_gain_db() const override;
+
  private:
   class GainController;
 
   // GainControl implementation.
   int Enable(bool enable) override;
   int set_stream_analog_level(int level) override;
   int set_mode(Mode mode) override;
   int set_target_level_dbfs(int level) override;
   int target_level_dbfs() const override;
   int set_compression_gain_db(int gain) override;
-  int compression_gain_db() const override;
   int enable_limiter(bool enable) override;
   int set_analog_level_limits(int minimum, int maximum) override;
   int analog_level_minimum() const override;
   int analog_level_maximum() const override;
   bool stream_is_saturated() const override;
 
   void AllocateRenderQueue();
   int Configure();
 
   rtc::CriticalSection* const crit_render_ ACQUIRED_BEFORE(crit_capture_);
@@ skipping 24 matching lines @@
   std::vector<std::unique_ptr<GainController>> gain_controllers_;
 
   rtc::Optional<size_t> num_proc_channels_ GUARDED_BY(crit_capture_);
   rtc::Optional<int> sample_rate_hz_ GUARDED_BY(crit_capture_);
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainControlImpl);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_