winamp/Src/vlb/audio_io_dsp.h
2024-09-24 14:54:57 +02:00

81 lines
2.5 KiB
C++

/* $Header: /cvs/root/winamp/vlb/audio_io_dsp.h,v 1.1 2009/04/28 20:21:08 audiodsp Exp $ */
/***************************************************************************\
*
* Copyright 2000-2002 Dolby Laboratories, Inc. All Rights
* Reserved. Do not copy. Do not distribute.
* Confidential information.
*
* (C) copyright Fraunhofer - IIS (1998)
* All Rights Reserved
*
* filename: audio_io_dsp.h
* project : MPEG-2 AAC Decoder
* contents/description: Audio DSP processing header file
*
\***************************************************************************/
#ifndef __AUDIO_IO_DSP_H__
#define __AUDIO_IO_DSP_H__
#define CLIPDLY 10
#define CLIPNORM (1.0/CLIPDLY)
#define DLYBUFSZ 128
#define PMAXF 32767.0f
#define PMAXFM1 (PMAXF-1)
#define NMAXF -32768.0f
#define MAXCHANNELS 2
#define NSAMPRATE 12
#define XLIMDELAY 5 // delay is 1/5 second before decay stage of clip delay return segment.
#define NSCALETAB 33
#define PEAKLIMIT (3.16*PMAXF) // Allows for 10 dB of smoothed limiting. Any overshoot greater
// than 10 dB is hard limited.
#define PEAKLIMITM1 (PEAKLIMIT-1)
#define INTERP_CORRECTION_FACT 0.013 // compensates for error between actual inverse and estimated inverse
// in scaletab table.
#define CX_LIMSCALE ((NSCALETAB-1)/(PEAKLIMIT-PMAXF))
typedef struct {
short index;
short delay;
short bufsize;
float *bufptr;
} DLY_VARS;
typedef struct {
short dcy_count;
float dcy_samp;
} SMOOTH_VARS;
class AudioIODSP{
private:
DLY_VARS **hsClipDly;
SMOOTH_VARS **hsSmoothVars;
int iNChannels;
bool bHardClip;
float *pfDlyBuf;
float fPrevDlyBuf[MAXCHANNELS][CLIPDLY];
float fPrevClipScale[MAXCHANNELS];
short initLimitCnt; // Amount of time to delay before decaying the limiter scaling value.
float dcyDelta; // Dcy_delta is the amount to attenuate the limiter scaling value
// on each call to CalcAtten().
unsigned int uiSampRate;
float AudioIODSP::CalcAtten(SMOOTH_VARS *smoothvars, float fMaxSamp, float thresh);
public:
short *pshOutBuf;
AudioIODSP(int _iNChannels, unsigned int _uiSampRate = 0);
~AudioIODSP();
void Float_to_Short_Clip(float **fInput, short *sOutput, int iLength, int grpIndex);
void Float_to_Int(float **fInput, int *iOutput, int iLength, int grpIndex,
int bitResolution);
void ApplyLimiter(float **fBuffer, int iLength, int grpIndex);
void SetSamplingRate(unsigned int _uiSampRate);
};
#endif /* __AUDIO_IO_DSP_H__ */