/* 
 * 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);
  }
  
  {
    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
}
