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kolibrios/programs/media/ac97snd/mpg/decode_i386.c
Sergey Semyonov (Serge) 268f6d98b7 new folders layout
git-svn-id: svn://kolibrios.org@576 a494cfbc-eb01-0410-851d-a64ba20cac60
2007-07-09 16:31:02 +00:00

270 lines
6.0 KiB
C

/*
decode_i386.c: decode for i386 (really faster?)
copyright 1995-2006 by the mpg123 project - free software under the terms of the LGPL 2.1
see COPYING and AUTHORS files in distribution or http://mpg123.de
initially written by Michael Hipp
slighlty optimized for machines without autoincrement/decrement.
The performance is highly compiler dependend. Maybe
the decode.c version for 'normal' processor may be faster
even for Intel processors.
*/
//#include <stdlib.h>
#include <math.h>
//#include <string.h>
//#include "config.h"
#include "mpg123.h"
#if 0
/* old WRITE_SAMPLE */
#define WRITE_SAMPLE(samples,sum,clip) \
if( (sum) > 32767.0) { *(samples) = 0x7fff; (clip)++; } \
else if( (sum) < -32768.0) { *(samples) = -0x8000; (clip)++; } \
else { *(samples) = sum; }
#else
/* new WRITE_SAMPLE */
/* keep in mind that we are on known little-endian i386 here and special tricks are allowed... */
#define WRITE_SAMPLE(samples,sum,clip) { \
double dtemp; int v; /* sizeof(int) == 4 */ \
dtemp = ((((65536.0 * 65536.0 * 16)+(65536.0 * 0.5))* 65536.0)) + (sum); \
v = ((*(int *)&dtemp) - 0x80000000); \
if( v > 32767) { *(samples) = 0x7fff; (clip)++; } \
else if( v < -32768) { *(samples) = -0x8000; (clip)++; } \
else { *(samples) = v; } \
}
#endif
#if 0
int synth_1to1_8bit(real *bandPtr,int channel,unsigned char *samples,int *pnt)
{
short samples_tmp[64];
short *tmp1 = samples_tmp + channel;
int i,ret;
int pnt1 = 0;
ret = synth_1to1(bandPtr,channel,(unsigned char *)samples_tmp,&pnt1);
samples += channel + *pnt;
for(i=0;i<32;i++) {
*samples = conv16to8[*tmp1>>AUSHIFT];
samples += 2;
tmp1 += 2;
}
*pnt += 64;
return ret;
}
int synth_1to1_8bit_mono(real *bandPtr,unsigned char *samples,int *pnt)
{
short samples_tmp[64];
short *tmp1 = samples_tmp;
int i,ret;
int pnt1 = 0;
ret = synth_1to1(bandPtr,0,(unsigned char *)samples_tmp,&pnt1);
samples += *pnt;
for(i=0;i<32;i++) {
*samples++ = conv16to8[*tmp1>>AUSHIFT];
tmp1+=2;
}
*pnt += 32;
return ret;
}
int synth_1to1_8bit_mono2stereo(real *bandPtr,unsigned char *samples,int *pnt)
{
short samples_tmp[64];
short *tmp1 = samples_tmp;
int i,ret;
int pnt1 = 0;
ret = synth_1to1(bandPtr,0,(unsigned char *)samples_tmp,&pnt1);
samples += *pnt;
for(i=0;i<32;i++) {
*samples++ = conv16to8[*tmp1>>AUSHIFT];
*samples++ = conv16to8[*tmp1>>AUSHIFT];
tmp1 += 2;
}
*pnt += 64;
return ret;
}
int synth_1to1_mono(real *bandPtr,unsigned char *samples,int *pnt)
{
short samples_tmp[64];
short *tmp1 = samples_tmp;
int i,ret;
int pnt1 = 0;
ret = synth_1to1(bandPtr,0,(unsigned char *) samples_tmp,&pnt1);
samples += *pnt;
for(i=0;i<32;i++) {
*( (short *) samples) = *tmp1;
samples += 2;
tmp1 += 2;
}
*pnt += 64;
return ret;
}
#endif
int synth_1to1_mono2stereo(real *bandPtr,unsigned char *samples,int *pnt)
{
int i,ret;
ret = synth_1to1(bandPtr,0,samples,pnt);
samples = samples + *pnt - 128;
for(i=0;i<32;i++) {
((short *)samples)[1] = ((short *)samples)[0];
samples+=4;
}
return ret;
}
static real buffs[2][2][0x110];
static const int step = 2;
static int bo = 1;
void init_dct()
{
bo = 1;
memset(buffs,0, sizeof(buffs));
};
int synth_1to1(real *bandPtr,int channel,unsigned char *out,int *pnt)
{
#ifndef PENTIUM_OPT
short *samples = (short *) (out + *pnt);
real *b0,(*buf)[0x110];
int clip = 0;
int bo1;
#endif
// if(have_eq_settings)
// do_equalizer(bandPtr,channel);
#ifndef PENTIUM_OPT
if(!channel) {
bo--;
bo &= 0xf;
buf = buffs[0];
}
else {
samples++;
buf = buffs[1];
}
if(bo & 0x1) {
b0 = buf[0];
bo1 = bo;
dct64(buf[1]+((bo+1)&0xf),buf[0]+bo,bandPtr);
}
else {
b0 = buf[1];
bo1 = bo+1;
dct64(buf[0]+bo,buf[1]+bo+1,bandPtr);
}
{
register int j;
real *window = decwin + 16 - bo1;
for (j=16;j;j--,b0+=0x10,window+=0x20,samples+=step)
{
real sum;
sum = window[0x0] * b0[0x0];
sum -= window[0x1] * b0[0x1];
sum += window[0x2] * b0[0x2];
sum -= window[0x3] * b0[0x3];
sum += window[0x4] * b0[0x4];
sum -= window[0x5] * b0[0x5];
sum += window[0x6] * b0[0x6];
sum -= window[0x7] * b0[0x7];
sum += window[0x8] * b0[0x8];
sum -= window[0x9] * b0[0x9];
sum += window[0xA] * b0[0xA];
sum -= window[0xB] * b0[0xB];
sum += window[0xC] * b0[0xC];
sum -= window[0xD] * b0[0xD];
sum += window[0xE] * b0[0xE];
sum -= window[0xF] * b0[0xF];
WRITE_SAMPLE(samples,sum,clip);
}
{
real sum;
sum = window[0x0] * b0[0x0];
sum += window[0x2] * b0[0x2];
sum += window[0x4] * b0[0x4];
sum += window[0x6] * b0[0x6];
sum += window[0x8] * b0[0x8];
sum += window[0xA] * b0[0xA];
sum += window[0xC] * b0[0xC];
sum += window[0xE] * b0[0xE];
WRITE_SAMPLE(samples,sum,clip);
b0-=0x10,window-=0x20,samples+=step;
}
window += bo1<<1;
for (j=15;j;j--,b0-=0x10,window-=0x20,samples+=step)
{
real sum;
sum = -window[-0x1] * b0[0x0];
sum -= window[-0x2] * b0[0x1];
sum -= window[-0x3] * b0[0x2];
sum -= window[-0x4] * b0[0x3];
sum -= window[-0x5] * b0[0x4];
sum -= window[-0x6] * b0[0x5];
sum -= window[-0x7] * b0[0x6];
sum -= window[-0x8] * b0[0x7];
sum -= window[-0x9] * b0[0x8];
sum -= window[-0xA] * b0[0x9];
sum -= window[-0xB] * b0[0xA];
sum -= window[-0xC] * b0[0xB];
sum -= window[-0xD] * b0[0xC];
sum -= window[-0xE] * b0[0xD];
sum -= window[-0xF] * b0[0xE];
sum -= window[-0x0] * b0[0xF];
WRITE_SAMPLE(samples,sum,clip);
}
}
*pnt += 128;
return clip;
#elif defined(USE_MMX)
{
static short buffs[2][2][0x110];
static int bo = 1;
short *samples = (short *) (out + *pnt);
synth_1to1_MMX(bandPtr, channel, samples, (short *) buffs, &bo);
*pnt += 128;
return 0;
}
#else
{
int ret;
ret = synth_1to1_pent(bandPtr,channel,out+*pnt);
*pnt += 128;
return ret;
}
#endif
}