Subversion Repositories Kolibri OS

/*

	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 

#include 

//#include 

//#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

}