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/*
* 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 MODULES_AUDIO_PROCESSING_AGC_AGC_H_
#define MODULES_AUDIO_PROCESSING_AGC_AGC_H_
#include <memory>
#include "modules/audio_processing/vad/voice_activity_detector.h"
namespace webrtc {
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 void 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;
virtual float voice_probability() const;
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 // MODULES_AUDIO_PROCESSING_AGC_AGC_H_
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