Revert 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 392 matching lines @@
   //     audio hardware and t_pull is the time the same frame is passed to
   //     ProcessStream().
   virtual int set_stream_delay_ms(int delay) = 0;
   virtual int stream_delay_ms() const = 0;
   virtual bool was_stream_delay_set() const = 0;
 
   // Call to signal that a key press occurred (true) or did not occur (false)
   // with this chunk of audio.
   virtual void set_stream_key_pressed(bool key_pressed) = 0;
 
-  // 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].
-  virtual void SetLevelControllerInitialLevel(float level) = 0;
-
   // Sets a delay |offset| in ms to add to the values passed in through
   // set_stream_delay_ms(). May be positive or negative.
   //
   // Note that this could cause an otherwise valid value passed to
   // set_stream_delay_ms() to return an error.
   virtual void set_delay_offset_ms(int offset) = 0;
   virtual int delay_offset_ms() const = 0;
 
   // Starts recording debugging information to a file specified by |filename|,
   // a NULL-terminated string. If there is an ongoing recording, the old file
@@ skipping 551 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_