webrtc_m130/webrtc/modules/audio_processing/agc/agc.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.
 */

#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_H_

#include <memory>

#include "webrtc/modules/audio_processing/vad/voice_activity_detector.h"
#include "webrtc/typedefs.h"

namespace webrtc {

class AudioFrame;
class LoudnessHistogram;

class Agc {
 public:
  Agc();
  virtual ~Agc();

  // Returns the proportion of samples in the buffer which are at full-scale
  // (and presumably clipped).
  virtual float AnalyzePreproc(const int16_t* audio, size_t length);
  // |audio| must be mono; in a multi-channel stream, provide the first (usually
  // left) channel.
  virtual int Process(const int16_t* audio, size_t length, int sample_rate_hz);

  // Retrieves the difference between the target RMS level and the current
  // signal RMS level in dB. Returns true if an update is available and false
  // otherwise, in which case |error| should be ignored and no action taken.
  virtual bool GetRmsErrorDb(int* error);
  virtual void Reset();

  virtual int set_target_level_dbfs(int level);
  virtual int target_level_dbfs() const { return target_level_dbfs_; }

  virtual float voice_probability() const {
    return vad_.last_voice_probability();
  }

 private:
  double target_level_loudness_;
  int target_level_dbfs_;
  std::unique_ptr<LoudnessHistogram> histogram_;
  std::unique_ptr<LoudnessHistogram> inactive_histogram_;
  VoiceActivityDetector vad_;
};

}  // namespace webrtc

#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_H_