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