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wav2ay.c
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wav2ay.c
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/***
latest official release at: https://github.com/EdouardBERGE/wav2ay
-----------------------------------------------------------------------------------------------------
This software is using MIT "expat" license
« Copyright © BERGE Edouard (roudoudou)
Permission is hereby granted, free of charge,to any person obtaining a copy of this software
and associated documentation/source files of RASM, to deal in the Software without restriction,
including without limitation the rights to use, copy, modify, merge, publish, distribute,
sublicense, and/or sell copies of the Software, and to permit persons to whom the Software is
furnished to do so, subject to the following conditions:
The above copyright notice and this permission notice shall be included in all copies or
substantial portions of the Software.
The Software is provided "as is", without warranty of any kind, express or implied,
including but not limited to the warranties of merchantability, fitness for a particular
purpose and noninfringement. In no event shall the authors or copyright holders be liable for
any claim, damages or other liability, whether in an action of contract, tort or otherwise,
arising from, out of or in connection with the software or the use or other dealings in the
Software. »
-----------------------------------------------------------------------------------------------------
***/
#ifdef _WIN32
#define OS_WIN 1
#endif
#ifdef _WIN64
#define OS_WIN 1
#endif
#ifdef OS_WIN
#define _USE_MATH_DEFINES
#include<io.h>
#include<fcntl.h>
#endif
#include<stdlib.h>
#include<string.h>
#include<stdio.h>
#include<errno.h>
#include<math.h>
#define MAXCHANNEL 3
enum
{
O32_LITTLE_ENDIAN = 0x03020100ul,
O32_BIG_ENDIAN = 0x00010203ul,
O32_PDP_ENDIAN = 0x01000302ul, /* DEC PDP-11 (aka ENDIAN_LITTLE_WORD) */
O32_HONEYWELL_ENDIAN = 0x02030001ul /* Honeywell 316 (aka ENDIAN_BIG_WORD) */
};
/*************************************************************************
CSV Export Func
*************************************************************************/
void export_csv(char *filename,double *data,int n)
{
char temp[1024];
FILE *f;
int i;
f=fopen(filename,"wb");
for (i=0;i<n;i++) {
sprintf(temp,"%.1lf\n",data[i]);
fputs(temp,f);
}
fclose(f);
}
/*************************************************************************
WAV Loading Func
*************************************************************************/
struct s_wav_header {
char ChunkID[4];
unsigned char ChunkSize[4];
char Format[4];
char SubChunk1ID[4];
unsigned char SubChunk1Size[4];
unsigned char AudioFormat[2];
unsigned char NumChannels[2];
unsigned char SampleRate[4];
unsigned char ByteRate[4];
unsigned char BlockAlign[2];
unsigned char BitsPerSample[2];
unsigned char SubChunk2ID[4];
unsigned char SubChunk2Size[4];
};
/*
* strange WAV header
*
00000000 52 49 46 46 b2 83 c8 00 57 41 56 45 66 6d 74 20 |RIFF....WAVEfmt |
00000010 12 00 00 00 01 00 01 00 22 56 00 00 44 ac 00 00 |........"V..D...|
00000020 02 00 10 00 00 00 66 61 63 74 04 00 00 00 c0 41 |......fact.....A|
00000030 64 00 64 61 74 61 80 83 c8 00 00 00 00 00 00 00 |d.data..........|
00000040 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 |................|
*/
void push_wav(char *outputfilename, short int *data, int n) {
struct s_wav_header wav_header={0};
unsigned int datasize,riffsize;
unsigned char chunksize[4];
char RIFF[5];
FILE *f;
#ifdef OS_WIN
int sr;
#endif
datasize=n*2;
memcpy(wav_header.ChunkID,"RIFF",4);
riffsize=datasize+16+8+datasize+8;
wav_header.ChunkSize[0]=riffsize&0xFF;
wav_header.ChunkSize[1]=(riffsize>>8)&0xFF;
wav_header.ChunkSize[2]=(riffsize>>16)&0xFF;
wav_header.ChunkSize[3]=(riffsize>>24)&0xFF;
memcpy(wav_header.Format,"WAVE",4);
memcpy(wav_header.SubChunk1ID,"fmt ",4);
wav_header.SubChunk1Size[0]=16;
wav_header.AudioFormat[0]=1; // integer
wav_header.NumChannels[0]=1;
wav_header.SampleRate[0]=0x44;
wav_header.SampleRate[1]=0xAC; // 44.1KHz
wav_header.ByteRate[0]=0x88;
wav_header.ByteRate[1]=0x58;
wav_header.ByteRate[2]=0x01; // 88.2Kbio
wav_header.BlockAlign[0]=2;
wav_header.BitsPerSample[0]=16;
memcpy(wav_header.SubChunk2ID,"data",4);
wav_header.SubChunk2Size[0]=datasize&0xFF;
wav_header.SubChunk2Size[1]=(datasize>>8)&0xFF;
wav_header.SubChunk2Size[2]=(datasize>>16)&0xFF;
wav_header.SubChunk2Size[3]=(datasize>>24)&0xFF;
// dataaaaa output
f=fopen(outputfilename,"wb");
#ifdef OS_WIN
sr=_setmode(_fileno(f), _O_BINARY );
if (sr==-1) {
fprintf(stderr,"wavoutput windows binary mode got problem!\n");
exit(1);
}
#endif
if (!f) {
fprintf(stderr,"wavoutput got problem!\n");
exit(1);
}
fwrite(&wav_header,1,sizeof(struct s_wav_header),f);
fwrite(data,1,datasize,f);
fclose(f);
}
double (*_internal_getsample)(unsigned char *data, int *idx);
double __internal_getsample8(unsigned char *data, int *idx) {
double v;
v=data[*idx]-128;*idx=*idx+1;return v;
}
double __internal_getsample16(unsigned char *data, int *idx) {
char *most;
double v;
most=(char*)data;
v=most[*idx+1];v+=data[*idx]/256.0;*idx=*idx+2;return v;
}
double __internal_getsample32(unsigned char *data, int *idx) {
float v,*peteher;
peteher=(float*)(&data[*idx]);
v=*peteher;
*idx=*idx+4;
return v*128.0;
}
double *load_wav(char *filename, int *n, double *acqui) {
struct s_wav_header *wav_header;
int controlsize,nbchannel,wFormat,frequency,bitspersample,nbsample;
unsigned char *subchunk;
unsigned char *data;
int subchunksize;
int filesize,idx;
int i,num;
FILE *f;
double *wav;
f=fopen(filename,"rb");
if (!f) {
fprintf(stderr,"file [%s] not found\n",filename);
return NULL;
}
fseek(f,0,SEEK_END);
filesize=ftell(f);
fseek(f,0,SEEK_SET);
if (filesize<sizeof(struct s_wav_header)) {
fprintf(stderr,"WAV import - this file is too small to be a valid WAV!\n");
return NULL;
}
data=(unsigned char *)malloc(filesize);
fread(data,1,filesize,f);
fclose(f);
wav_header=(struct s_wav_header *)data;
if (strncmp(wav_header->Format,"WAVE",4)) {
fprintf(stderr,"WAV import - unsupported audio sample type (format must be 'WAVE')\n");
free(data);
return NULL;
}
controlsize=wav_header->SubChunk1Size[0]+wav_header->SubChunk1Size[1]*256+wav_header->SubChunk1Size[2]*65536+wav_header->SubChunk1Size[3]*256*65536;
if (controlsize!=16) {
fprintf(stderr,"WAV import - invalid wav chunk size (subchunk1 control) => %d!=16\n",controlsize);
}
if (strncmp(wav_header->SubChunk1ID,"fmt",3)) {
fprintf(stderr,"WAV import - unsupported audio sample type (subchunk1id must be 'fmt')\n");
free(data);
return NULL;
}
if (controlsize==16) {
subchunk=(unsigned char *)&wav_header->SubChunk2ID;
} else {
subchunk=(unsigned char *)&wav_header->SubChunk2ID-16+controlsize;
}
// tant qu on n est pas sur le chunk data, on avance dans le "TIFF"
while (strncmp((char *)subchunk,"data",4)) {
subchunksize=8+subchunk[4]+subchunk[5]*256+subchunk[6]*65536+subchunk[7]*256*65536;
if (subchunksize>=filesize) {
fprintf(stderr,"WAV import - data subchunk not found\n");
free(data);
return NULL;
}
subchunk+=subchunksize;
}
subchunksize=subchunk[4]+subchunk[5]*256+subchunk[6]*65536+subchunk[7]*256*65536;
controlsize=subchunksize; // taille des samples
nbchannel=wav_header->NumChannels[0]+wav_header->NumChannels[1]*256;
if (nbchannel<1) {
fprintf(stderr,"WAV import - invalid number of audio channel\n");
free(data);
return NULL;
}
wFormat=wav_header->AudioFormat[0]+wav_header->AudioFormat[1]*256;
if (wFormat!=1 && wFormat!=3) {
fprintf(stderr,"WAV import - invalid or unsupported wFormatTag (%04X) ",wFormat);
switch (wFormat) {
case -2:fprintf(stderr,"Extensible Format (user defined) \n");
default:
case 0:fprintf(stderr," Unknown Format\n");
case 1:fprintf(stderr," PCM format (8 or 16 bit), Microsoft Corporation\n");
case 2:fprintf(stderr," AD PCM Format, Microsoft Corporation\n");
case 3:fprintf(stderr," IEEE PCM Float format (32 bit)\n");
case 48:fprintf(stderr,"AC2, Dolby Laboratories\n");
case 49:fprintf(stderr,"GSM 6.10, Microsoft Corporation\n");
case 50:fprintf(stderr,"MSN Audio, Microsoft Corporation\n");
case 80:fprintf(stderr,"MPEG format\n");
case 85:fprintf(stderr,"ISO/MPEG Layer3 Format\n");
case 146:fprintf(stderr,"AC3 Digital, Sonic Foundry\n");
case 255:fprintf(stderr,"Raw AAC\n");
case 352:fprintf(stderr,"Microsoft Corporation\n");
case 353:fprintf(stderr,"Windows Media Audio. This format is valid for versions 2 through 9\n");
case 354:fprintf(stderr,"Windows Media Audio 9 Professional\n");
case 355:fprintf(stderr,"Windows Media Audio 9 Lossless\n");
case 356:fprintf(stderr,"Windows Media SPDIF Digital Audio\n");
case 5632:fprintf(stderr,"ADTS AAC Audio\n");
case 5633:fprintf(stderr,"Raw AAC\n");
case 5634:fprintf(stderr,"MPEG-4 audio transport stream with a synchronization layer (LOAS) and a multiplex layer (LATM)\n");
case 5648:fprintf(stderr,"High-Efficiency Advanced Audio Coding (HE-AAC) stream\n");
}
free(data);
return NULL;
}
frequency=wav_header->SampleRate[0]+wav_header->SampleRate[1]*256+wav_header->SampleRate[2]*65536+wav_header->SampleRate[3]*256*65536;
bitspersample=wav_header->BitsPerSample[0]+wav_header->BitsPerSample[1]*256;
switch (bitspersample) {
case 8:_internal_getsample=__internal_getsample8;break;
case 16:_internal_getsample=__internal_getsample16;break;
case 32:_internal_getsample=__internal_getsample32;break;
default:fprintf(stderr,"unsupported bits per sample size %d (valid are PCM-8, PCM-16 and FLOAT-32)\n",bitspersample); free(data);return NULL;
}
nbsample=controlsize/nbchannel/(bitspersample/8);
if (controlsize+sizeof(struct s_wav_header)>filesize) {
fprintf(stderr,"WAV import - cannot read %d byte%s of audio whereas the file is %d bytes big!\n",controlsize,controlsize>1?"s":"",filesize);
free(data);
return NULL;
}
*n=nbsample;
fprintf(stderr,"nbchannel=%d | bitpersample=%d | Format=%s | frequency=%d\n",nbchannel,bitspersample,wFormat==1?"PCM":"IEEE Float",frequency);
*acqui=frequency;
wav=(double *)malloc(nbsample*sizeof(double));
idx=subchunk+8-data;
for (i=0;i<nbsample;i++) {
/* downmixing */
double accumulator=0.0;
for (num=0;num<nbchannel;num++) {
accumulator+=_internal_getsample(data,&idx);
}
wav[i]=accumulator/nbchannel;
}
return wav;
}
/*****************************************************
*
* Fourier stuff (basic transform)
*
*****************************************************/
/* slow but understandable Fourier transform */
void calcule_fourier(double *data, int n, double *dataout, int clean) {
double rel,img,inverse;
int i,j;
inverse=1.0/n;
// skip everything too low according to lowcut filter
for (i=0;i<clean;i++) dataout[i]=0.0;
for (;i<=n/2;i++) {
rel=img=0.0;
// cumul
for (j=0;j<n;j++) {
rel+=data[j]*cos((2*M_PI*j*i)/n);
img+=data[j]*sin((2*M_PI*j*i)/n);
}
// norme
rel*=inverse;
img*=inverse;
dataout[i]=2.0*sqrt(rel*rel+img*img);
}
}
/* partial Fourier trick to compute only peak neighborhood */
double calcule_fourier_precision(double *data, int n, int peak, double cutlow, double peakHz) {
double v[16384],rel,img,vmax,inverse;
int i,j,imax;
int lp,mp,ws,bornemin;
bornemin=(double)peak*cutlow/peakHz*16384/n;
if (bornemin<1) bornemin=1;
// [lp:mp] let us zoom on previous (small) fourier buffer
ws=16384/n+1;
lp=peak*16384/n-ws/2;
if (lp<bornemin) lp=bornemin; // may happend with strong bass
mp=peak*16384/n+ws/2;
if (mp>=8192) mp=8191;
inverse=1.0/n;
// do Fourier on that segment
for (i=lp;i<=mp;i++) {
rel=img=0.0;
for (j=0;j<n;j++) {
rel+=data[j]*cos((2*M_PI*j*i)/16384.0);
img+=data[j]*sin((2*M_PI*j*i)/16384.0);
}
// energy
rel*=inverse;
img*=inverse;
v[i]=2.0*sqrt(rel*rel+img*img);
}
// max is max
vmax=v[lp];imax=lp;
for (i=lp+1;i<=mp;i++) {
if (v[i]>vmax) {
vmax=v[i];
imax=i;
}
}
// precise peak
return imax*n/16384.0;
}
void passe_bande(double acqui,double basse,double haute,double *datain,double *dataout,int nb)
{
double *datatmp=NULL;
double A1,A2,B0,B1,B2;
double a,b;
int i;
/***/
double e,old_e,old_old_e;
double s,old_s,old_old_s;
if (nb<=0) return;
if (basse<0.0001) basse=0.0001;
if (haute<0.0001) haute=0.0001;
if (basse>acqui) basse=acqui;
if (haute>acqui) haute=acqui;
datatmp=(double *)malloc(sizeof(double)*nb);
if (!datatmp) return;
a=M_PI*basse/acqui;
b=M_PI*haute/acqui;
a=sin(a)/cos(a);
b=sin(b)/cos(b);
B0= -b / ((1 + a) * (1 + b));
B1= 0;
B2= b / ((1 + a) * (1 + b));
A1= ((1 + a) * (1 - b) + (1 - a) * (1 + b)) / ((1 + a) * (1 + b));
A2= -(1 - a) * (1 - b) / ((1 + a) * (1 + b));
/* init */
old_e=old_old_e=datain[0];
old_s=old_old_s=0;
/* a filter phases out signal */
for (i=0;i<nb;i++) {
e=datain[i];
s=e*B0+old_e*B1+old_old_e*B2+old_s*A1+old_old_s*A2;
datatmp[i]=s;
old_old_s=old_s;
old_s=s;
old_old_e=old_e;
old_e=e;
}
old_e=old_old_e=e;
old_s=old_old_s=0;
/* so we filter again reverse */
while (i>0) {
i--;
e=datatmp[i];
s=e*B0+old_e*B1+old_old_e*B2+old_s*A1+old_old_s*A2;
dataout[i]=s;
old_old_s=old_s;
old_s=s;
old_old_e=old_e;
old_e=e;
}
free(datatmp);
}
// AY period related to a frequency
struct s_ay_period {
double *fourier;
};
double * compute_AY(int ws, int replay, int clean, double cuthigh, double cutlow,int periode, double workingfreq) {
double *retfour;
double *minisignal;
double freq,target;
int i,j,k;
double vtic,tic,mtic;
freq=replay*ws; // echantillons/seconde
minisignal=(double *)malloc(ws*sizeof(double));
i=periode;
// generate AY signal for this period, windows size and replay freq
if (!i) target=workingfreq; else target=workingfreq/i;
// keep shanon compliant
if (target*2>freq) {
retfour=(double *)malloc(ws*sizeof(double));
memset(retfour,0,ws*sizeof(double));
free(minisignal);
return retfour;
}
tic=mtic=0.5*freq/target;
vtic=80.0;
for (j=0;j<ws;j++) {
minisignal[j]=vtic;
tic-=1.0;
if (tic<=0.0) {
vtic=-vtic;
tic+=mtic;
}
}
retfour=(double *)malloc(ws*sizeof(double));
calcule_fourier(minisignal,ws,retfour,clean);
free(minisignal);
return retfour;
}
int getpeak(double *fourier, int ws) {
double vmax;
int imax;
int j;
imax=0;
vmax=fourier[0];
for (j=1;j<=ws/2;j++) {
if (fourier[j]>vmax) {
vmax=fourier[j];
imax=j;
}
}
return imax;
}
int getvolume(double level) {
int volume;
if (level>=64) volume=15; else
if (level>=45) volume=14; else
if (level>=32) volume=13; else
if (level>=22) volume=12; else
if (level>=16) volume=11; else
if (level>=11) volume=10; else
if (level>=8) volume=9; else
if (level>=6) volume=8; else
if (level>=4) volume=7; else
if (level>=3.3) volume=6; else
if (level>=2) volume=5; else
if (level>=1.7) volume=4; else
if (level>=1) volume=3; else
if (level>=0.7) volume=2; else
volume=1;
return volume;
}
void do_sample(double *data,int n, double pw, double cutlow, double cuthigh, double acqui, double replay, double preamp, int info, double workingfreq, int nbchannel, double treshold, int dmalist, char *wavout_filename, int *channel_list, int cpclist, char *akifilename, int noiseblocker) {
double *fourier,*oldfourier;
double *newdata,subcoef;
double vmax,resolution,picfreq;
int nbwin,i,j,k,imax,ws,clean;
struct s_ay_period *ay=NULL;
short int *wavout;
int wavout_n=0;
// Amstrad registers
int psgperiod,psgvolume;
int nbpausedma;
int channel;
int AYprevperiod[MAXCHANNEL];
int AYprevvolume[MAXCHANNEL];
int AYperiod[MAXCHANNEL];
int AYvolume[MAXCHANNEL];
int nbchanges;
int makenoise=0;
int hadnoise=0;
FILE *akifile;
#ifdef OS_WIN
int sr;
#endif
if (pw<0.0 || pw>0.75) {
fprintf(stderr,"previous weight not in [0.0:0.75] interval. Default value is 0.25\n");
pw=0.25;
}
if (acqui<4000 || acqui>48000) {
fprintf(stderr,"wrong acquisition frequency. Default value is 15.6KHz\n");
acqui=15600.0;
}
if (cuthigh<cutlow || cutlow<0.0 || cuthigh>acqui || cutlow<replay) {
fprintf(stderr,"wrong band-pass filter frequencies. Default value are replay-2500Hz\n");
cuthigh=2500.0;
cutlow=replay;
}
if (cutlow<workingfreq/4095.0*1.3) cutlow=workingfreq/4095.0*1.3;
ws=acqui/replay; // 312 samples pour 15000Hz avec replay a 50Hz
if (ws<200) {
fprintf(stderr,"window size is small, results may be very innacurate\n");
}
if (ws<100) {
fprintf(stderr,"wont compute with such a small window, raise input frequency\n");
return;
}
// compléter le nombre de sample pour arriver pile sur un multiple de ws (window size) | default:312
nbwin=n/ws;
while (nbwin*ws<n) {
nbwin++;
}
if (nbwin*ws!=n) {
data=(double *)realloc(data,sizeof(double)*nbwin*ws);
for (i=n;i<nbwin*ws;i++) {
data[i]=data[i-1]*0.95; // histoire de terminer salement mais en evitant le poc => normalement le sample est ok de base!
}
n=nbwin*ws;
}
newdata=(double *)malloc(sizeof(double)*n);
// calculer la valeur du nettoyage a faire sur la transformee pour eviter les basses frequences mal filtrees
resolution=acqui/(double)ws;
clean=1+floor(replay/resolution+0.5);
// precalc des fouriers de l'AY sur la fenetre utilisee
ay=(struct s_ay_period *)malloc(4096*sizeof(struct s_ay_period));
memset(ay,0,4096*sizeof(struct s_ay_period));
// par defaut il faut filtrer en dessous de la frequence de replay et au dessus de 2500Hz a cause de la precision de restitution
fprintf(stderr,"band-pass %.1lf|%.1lf window size=%d preamp=%.1lf AY frequency=%.1lfMHz\n",cutlow,cuthigh,ws,preamp,workingfreq*16.0);
passe_bande(acqui,cutlow,cuthigh,data,newdata,n);
memcpy(data,newdata,n*sizeof(double));
// init temporal buffer
fourier=(double *)malloc(sizeof(double)*ws);
oldfourier=(double *)malloc(sizeof(double)*ws);
calcule_fourier(data,ws,oldfourier,clean);
fprintf(stderr,"%d windows for a %.1lfs sample\n",nbwin,n/acqui);
for (channel=0;channel<nbchannel;channel++) {
AYprevperiod[channel]=0xFFFF;
AYprevvolume[channel]=0;
}
if (!dmalist && !cpclist) {
printf("idx");
for (channel=0;channel<nbchannel;channel++) {
printf(";reg;value");
printf(";reg;value");
printf(";reg;value");
}
printf("\n");
}
if (wavout_filename) {
wavout=(short int *)malloc(nbwin*(sizeof(short int)*workingfreq*2.0/replay+1));
wavout_n=0;
}
if (preamp!=1.0) for (j=0;j<n;j++) data[j]*=preamp;
if (akifilename) {
akifile=fopen(akifilename,"wb");
if (!akifile) {
fprintf(stderr,"FATAL: cannot open %s for writing\n",akifilename);
exit(-66);
}
#ifdef OS_WIN
sr=_setmode(_fileno(akifile), _O_BINARY );
if (sr==-1) {
fprintf(stderr,"aki output binary mode got problem!\n");
exit(1);
}
#endif
fprintf(akifile,"<?xml version=\"1.0\" encoding=\"UTF-8\"?>\n"\
"<aks:instrument xmlns:aks=\"http://www.julien-nevo.com/ArkosTrackerInstrument\" "\
"xmlns:xsi=\"http://www.w3.org/2001/XMLSchema-instance\" xsi:schemaLocation=\"http://www.julien-nevo.com/ArkosTrackerInstrument.xsd\">\n"\
" <aks:fmInstrument version=\"1.0\">\n"\
" <aks:number>1</aks:number>\n"\
" <aks:title>%s</aks:title>\n"\
" <aks:speed>%d</aks:speed>\n"\
" <aks:isLooping>false</aks:isLooping>\n"\
" <aks:loopStartIndex>%d</aks:loopStartIndex>\n"\
" <aks:endIndex>%d</aks:endIndex>\n"\
" <aks:isRetrig>false</aks:isRetrig>\n"\
" <aks:colorArgb>4281370811</aks:colorArgb>\n",akifilename,replay==50?0:replay==25?1:2,nbwin-1,nbwin-1);
}
for (i=0;i<nbwin;i++) {
// calcul de fourier sur le segment
calcule_fourier(&data[ws*i],ws,fourier,clean);
for (channel=0;channel<nbchannel;channel++) {
/*
// mix avec le segment precedent si besoin
if (pw>0.0) {
for (j=0;j<ws/2;j++) {
fourier[j]=oldfourier[j]*pw+fourier[j]*(1.0-pw);
}
}
*/
// trouver le pic
imax=getpeak(fourier,ws);
// energie suffisante pour s'en soucier?
if (fourier[imax]>treshold) {
int volume;
int plow,phigh;
if (info) fprintf(stderr,"segment %3d pic en %.0lfHz (resolution=%.1lf) norme=%.1lf\n",i,imax*resolution,resolution,fourier[imax]);
AYvolume[channel]=getvolume(fourier[imax]);
k=imax;
//export_csv("fourier_ref.csv",fourier,ws/2+1);
// chercher la precision sur le pic
picfreq=calcule_fourier_precision(&data[ws*i],ws,imax,cutlow,imax*resolution)*resolution;
if (info) fprintf(stderr," => recherche de precision donne %.1lfHz\n",picfreq);
if (picfreq>2500 && !noiseblocker) makenoise++;
// appliquer la soustraction AY sur la transformee de Fourier pour la suite
imax=workingfreq/picfreq; // periode AY
if (imax>4095) imax=4095;
if (!ay[imax].fourier) ay[imax].fourier=compute_AY(ws,replay,clean,cuthigh,cutlow,imax,workingfreq);
subcoef=fourier[k]/ay[imax].fourier[k];
for (j=0;j<=ws/2;j++) fourier[j]-=ay[imax].fourier[j]*subcoef;
/* simple psycho acoustic model ???
double *tmpbuffer=NULL;
passe_bande(acqui,peak*0.8,peak*1.5,datain,tmpbuffer,nb);
// substract
*/
// produire les registres AY (imax=periode)
AYperiod[channel]=imax;
} else {
AYvolume[channel]=0;
/* volume a zero on ne touchera pas la periode! */
AYperiod[channel]=AYprevperiod[channel];
}
}
/*********************************
Channel optimisation
*********************************/
// d'abord un tri pour conserver une harmonie éventuelle sur le volume!
if (nbchannel>1) {
if (AYperiod[0]<AYperiod[1]) {
imax=AYperiod[1];
AYperiod[1]=AYperiod[0];
AYperiod[0]=imax;
imax=AYvolume[1];
AYvolume[1]=AYvolume[0];
AYvolume[0]=imax;
}
if (nbchannel>2) {
if (AYperiod[0]<AYperiod[2]) {
imax=AYperiod[2];
AYperiod[2]=AYperiod[0];
AYperiod[0]=imax;
imax=AYvolume[2];
AYvolume[2]=AYvolume[0];
AYvolume[0]=imax;
}
if (AYperiod[1]<AYperiod[2]) {
imax=AYperiod[2];
AYperiod[2]=AYperiod[1];
AYperiod[1]=imax;
imax=AYvolume[2];
AYvolume[2]=AYvolume[1];
AYvolume[1]=imax;
}
}
}
/* switch channel if period are exactly the same */
if (nbchannel>1) {
if (AYperiod[0]==AYprevperiod[1]) {
imax=AYperiod[1];
AYperiod[1]=AYperiod[0];
AYperiod[0]=imax;
imax=AYvolume[1];
AYvolume[1]=AYvolume[0];
AYvolume[0]=imax;
}
if (nbchannel>2) {
if (AYperiod[0]==AYprevperiod[2]) {
imax=AYperiod[2];
AYperiod[2]=AYperiod[0];
AYperiod[0]=imax;
imax=AYvolume[2];
AYvolume[2]=AYvolume[0];
AYvolume[0]=imax;
} else if (AYperiod[1]==AYprevperiod[2]) {
imax=AYperiod[2];
AYperiod[2]=AYperiod[1];
AYperiod[1]=imax;
imax=AYvolume[2];
AYvolume[2]=AYvolume[1];
AYvolume[1]=imax;
}
}
}
if (wavout_filename) {
int ref[16]={0,231,695,1158,2084,2779,4168,6716,8105,13200,18294,24315,32189,40757,52799,65535};
int pulse,maxp;
int acc;
int tic[MAXCHANNEL]={0};
int vchan[MAXCHANNEL];
maxp=workingfreq*2.0/replay;
for (channel=0;channel<nbchannel;channel++) {
vchan[channel]=ref[AYvolume[channel]]/2;
tic[channel]=AYperiod[channel];
}
for (pulse=0;pulse<maxp;pulse++) {
acc=0;
for (channel=0;channel<nbchannel;channel++) {
acc+=vchan[channel];
tic[channel]--;
if (!tic[channel]) {
tic[channel]=AYperiod[channel];
vchan[channel]=-vchan[channel];
}
}
wavout[wavout_n++]=acc/nbchannel;
}
}
/*********************************
Amstrad CPC list output
*********************************/
if (cpclist) {
int packed_exec=0;
int minilist[10];
int ilist=0;
int zebit=128;
if (!i) {
printf("defb %d ; nombre d'iterations\n",nbwin);
}
for (channel=0;channel<3;channel++) {
if (AYvolume[channel]!=AYprevvolume[channel]) {
packed_exec|=zebit;
minilist[ilist++]=AYvolume[channel];
}
zebit>>=1;
if ((AYperiod[channel]>>8)!=(AYprevperiod[channel]>>8)) {
packed_exec|=zebit;
minilist[ilist++]=AYperiod[channel]>>8;
}
zebit>>=1;
minilist[ilist++]=AYperiod[channel]&0xFF; // au minimum on enverra la frequence "basse"
}
if (hadnoise && !makenoise) {
packed_exec|=2;
minilist[ilist++]=56;
}
if (!hadnoise && makenoise) {
packed_exec|=2;
minilist[ilist++]=0; // bruit partout sinon on n'entend rien...
}
printf("defb #%02X",packed_exec);
if (packed_exec) {
for (channel=0;channel<ilist;channel++) printf(",#%02X",minilist[channel]);
printf(" ; %5d-%02d %5d-%02d %5d-%02d %s\n",AYperiod[0],AYvolume[0],AYperiod[1],AYvolume[1],AYperiod[2],AYvolume[2],packed_exec&2?"noise mod":"");
} else {
printf("\n");
}
hadnoise=makenoise;
makenoise=0;
for (channel=0;channel<nbchannel;channel++) {
AYprevperiod[channel]=AYperiod[channel];
AYprevvolume[channel]=AYvolume[channel];
}
} else
/*********************************
Amstrad Plus DMA output
*********************************/
if (dmalist) {
psgperiod=0;
psgvolume=8;
nbchanges=1;
for (channel=0;channel<nbchannel;channel++) {
if (AYvolume[channel]) {
if (AYperiod[channel]!=AYprevperiod[channel]) {
if ((AYperiod[channel]>>8)!=(AYprevperiod[channel]>>8)) {
printf("defb %d,%d : ",AYperiod[channel]>>8,psgperiod+1);
nbchanges++;
}
if ((AYperiod[channel]&0xFF)!=(AYprevperiod[channel]&0xFF)) {
printf("defb %d,%d : ",AYperiod[channel]&0xFF,psgperiod);
nbchanges++;
}
}
}
if (AYvolume[channel]!=AYprevvolume[channel]) {
printf("defb %d,%d : ",AYvolume[channel],psgvolume);
nbchanges++;
}
psgperiod+=2;
psgvolume+=1;
}
nbpausedma=312.0/replay*50.0-nbchanges;
// no control on this value
printf("defb %d,#30+%d\n",nbpausedma&0xFF,(nbpausedma>>8)&0xF); // pause 50Hz
for (channel=0;channel<nbchannel;channel++) {
AYprevperiod[channel]=AYperiod[channel];
AYprevvolume[channel]=AYvolume[channel];
}
} else {
/*********************************
CSV Export
*********************************/
if (!i) {
}
printf(";%d",i);
for (channel=0;channel<nbchannel;channel++) {
printf(";%d;%d",channel_list[channel] ,AYvolume[channel]);
printf(";%d;%d",0+channel*2,AYperiod[channel]&0xFF);
printf(";%d;%d",1+channel*2,AYperiod[channel]>>8);
}
if (hadnoise && !makenoise) {
printf(";7;0");
}
if (!hadnoise && makenoise) {
printf(";7;40");
}
if (makenoise) printf(" // noise");
hadnoise=makenoise;
makenoise=0;
printf("\n");
}
if (akifilename) {
if (AYperiod[0]>4095) AYperiod[0]=AYprevperiod[0]; // try to get previous value
if (AYperiod[0]>4095) AYperiod[0]=4095; // still out of bounds?
fprintf(akifile," <aks:fmInstrumentCell>\n"\
" <aks:link>softOnly</aks:link>\n"\
" <aks:volume>%d</aks:volume>\n"\
" <aks:noise>%d</aks:noise>\n"\
" <aks:softwarePeriod>%d</aks:softwarePeriod>\n"\
" <aks:softwareArpeggio>0</aks:softwareArpeggio>\n"\
" <aks:softwarePitch>0</aks:softwarePitch>\n"\
" <aks:ratio>1</aks:ratio>\n"\
" <aks:hardwareCurve>8</aks:hardwareCurve>\n"\
" <aks:hardwarePeriod>0</aks:hardwarePeriod>\n"\
" <aks:hardwareArpeggio>0</aks:hardwareArpeggio>\n"\
" <aks:hardwarePitch>0</aks:hardwarePitch>\n"\
" <aks:isRetrig>false</aks:isRetrig>\n"\
" </aks:fmInstrumentCell>\n",AYvolume[0],makenoise,AYperiod[0]);
}
}
if (akifilename) {
fprintf(akifile," </aks:fmInstrument>\n</aks:instrument>\n");
fclose(akifile);
if (nbwin>63) {
fprintf(stderr,"****************************************\n");
fprintf(stderr," Warning AKI instrument has more than\n");
fprintf(stderr," 64 pattern, Arkos Tracker 2 will crash\n");
fprintf(stderr,"\n");
fprintf(stderr," the file is saved anyway because it's\n");
fprintf(stderr," better to let YOU cut yourself data ;)\n");
fprintf(stderr,"****************************************\n");
}
}
if (wavout_filename) {
double acc,pos,step,aron,frac,integer_part;
int iidx,oidx=0;
step=2.0*workingfreq/44100.0;
for (pos=0;pos<wavout_n-step;pos+=step) {
frac=modf(pos,&integer_part);
iidx=integer_part;
acc=(1.0-frac)*wavout[iidx];
while (integer_part+1.0<pos+step) {
acc+=wavout[++iidx];
integer_part+=1.0;
}
frac=modf(pos,&integer_part);
acc+=frac*wavout[++iidx];
acc/=step;
wavout[oidx++]=acc;
}
push_wav(wavout_filename,wavout,oidx);
free(wavout);
}
}
void usage() {
printf("=========== conversion from WAV to AY registers ===========\n");
printf("usage: wav2ay.exe <wavfile> <options>\n");
printf("options:\n");
printf("-preamp <value> amplification | default 1.0\n");
printf("-tresh <value> minimal energy| default 0.25 is minimum (max usable approx 15)\n");
printf("-replay <value> frequency play| default 10\n");
printf("-high <value> highcut filter| default 4000\n");
printf("-low <value> lowcut filter | default is replay (min: 20Hz)\n");
// printf("-pw <value> sound inertia | default 0.0 max 0.75\n");