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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 WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_H_
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class AudioFrame;
class AgcAudioProc;
class Histogram;
class PitchBasedVad;
class Resampler;
class StandaloneVad;
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, int length);
// |audio| must be mono; in a multi-channel stream, provide the first (usually
// left) channel.
virtual int Process(const int16_t* audio, int 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 void EnableStandaloneVad(bool enable);
virtual bool standalone_vad_enabled() const {
return standalone_vad_enabled_;
}
virtual double voice_probability() const { return last_voice_probability_; }
private:
double target_level_loudness_;
double last_voice_probability_;
int target_level_dbfs_;
bool standalone_vad_enabled_;
scoped_ptr<Histogram> histogram_;
scoped_ptr<Histogram> inactive_histogram_;
scoped_ptr<AgcAudioProc> audio_processing_;
scoped_ptr<PitchBasedVad> pitch_based_vad_;
scoped_ptr<StandaloneVad> standalone_vad_;
scoped_ptr<Resampler> resampler_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_H_
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