Reland of added functionality for specifying the initial signal level to use for the gain estimation

webrtc/modules/audio_processing/include/audio_processing.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 234 matching lines @@
   // introduced, it is prone to change.
   // TODO(peah): Remove this comment once the new config scheme is fully rolled
   // out.
   //
   // The parameters and behavior of the audio processing module are controlled
   // by changing the default values in the AudioProcessing::Config struct.
   // The config is applied by passing the struct to the ApplyConfig method.
   struct Config {
     struct LevelController {
       bool enabled = false;
+
+      // Sets the initial peak level to use inside the level controller in order
+      // to compute the signal gain. The unit for the peak level is dBFS and
+      // the allowed range is [-100, 0].
+      float initial_peak_level_dbfs = -6.0206f;
     } level_controller;
   };
 
   // TODO(mgraczyk): Remove once all methods that use ChannelLayout are gone.
   enum ChannelLayout {
     kMono,
     // Left, right.
     kStereo,
     // Mono, keyboard, and mic.
     kMonoAndKeyboard,
@@ skipping 730 matching lines @@
   // This does not impact the size of frames passed to |ProcessStream()|.
   virtual int set_frame_size_ms(int size) = 0;
   virtual int frame_size_ms() const = 0;
 
  protected:
   virtual ~VoiceDetection() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_