diff options
Diffstat (limited to 'mpg123_artsplugin/mpg123/decode_i386.c')
| -rw-r--r-- | mpg123_artsplugin/mpg123/decode_i386.c | 257 |
1 files changed, 257 insertions, 0 deletions
diff --git a/mpg123_artsplugin/mpg123/decode_i386.c b/mpg123_artsplugin/mpg123/decode_i386.c new file mode 100644 index 00000000..d39795f8 --- /dev/null +++ b/mpg123_artsplugin/mpg123/decode_i386.c @@ -0,0 +1,257 @@ +/* + * Mpeg Layer-1,2,3 audio decoder + * ------------------------------ + * copyright (c) 1995,1996,1997 by Michael Hipp, All rights reserved. + * See also 'README' + * + * slighlty optimized for machines without autoincrement/decrement. + * The performance is highly compiler dependent. 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 "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 */ +#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 + +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; +} + + +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; +} + +int synth_1to1(real *bandPtr,int channel,unsigned char *out,int *pnt) +{ +#ifndef PENTIUM_OPT + static real buffs[2][2][0x110]; + static const int step = 2; + static int bo = 1; + short *samples = (short *) (out + *pnt); + + real *b0,(*buf)[0x110]; + int clip = 0; + int bo1; +#endif + +#ifndef NO_EQUALIZER + if(param.enable_equalizer) + do_equalizer(bandPtr,channel); +#endif + +#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 +} |