Revert of Added functionality for specifying the initial signal level to use for the gain estimation

webrtc/modules/audio_processing/level_controller/level_controller.h

 /*
  *  Copyright (c) 2016 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 20 matching lines @@
 
 class LevelController {
  public:
   LevelController();
   ~LevelController();
 
   void Initialize(int sample_rate_hz);
   void Process(AudioBuffer* audio);
   float GetLastGain() { return last_gain_; }
 
-  // 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].
-  void SetInitialLevel(float level);
-
  private:
   class Metrics {
    public:
     Metrics() { Initialize(AudioProcessing::kSampleRate48kHz); }
     void Initialize(int sample_rate_hz);
     void Update(float peak_level, float noise_level, float gain);
 
    private:
     void Reset();
 
@@ skipping 13 matching lines @@
   SignalClassifier signal_classifier_;
   NoiseLevelEstimator noise_level_estimator_;
   PeakLevelEstimator peak_level_estimator_;
   SaturatingGainEstimator saturating_gain_estimator_;
   Metrics metrics_;
   rtc::Optional<int> sample_rate_hz_;
   static int instance_count_;
   float dc_level_[2];
   float dc_forgetting_factor_;
   float last_gain_;
-  bool gain_jumpstart_ = false;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(LevelController);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_