view mp3lib/decod386.c @ 31685:31b6397e3b28

Another try at fixing swscale on win64, as per r31153. Don't change paramater passing, but instead use casts. Shouldn't affect asm output on anything other than win64. libswscale should work on win64 now. The rest of ffmpeg still isn't win64 compatible due to the issue of xmm clobbers, but swscale doesn't use any SSE. Patch by Anton Mitrofanov <BugMaster AT narod DOT ru>.

/*
 * Modified for use with MPlayer, for details see the changelog at
 * http://svn.mplayerhq.hu/mplayer/trunk/
 * $Id$
 */

/*
 * 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 dependend. Maybe
 * the decode.c version for 'normal' processor may be faster
 * even for Intel processors.
 */


#include "config.h"

#if 0
 /* old WRITE_SAMPLE */
   /* is portable */
#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 */

/*
 * should be the same as the "old WRITE_SAMPLE" macro above, but uses
 * some tricks to avoid double->int conversions and floating point compares.
 *
 * Here's how it works:
 * ((((65536.0 * 65536.0 * 16)+(65536.0 * 0.5))* 65536.0)) is
 * 0x0010000080000000LL in hex.  It computes 0x0010000080000000LL + sum
 * as a double IEEE fp value and extracts the low-order 32-bits from the
 * IEEE fp representation stored in memory.  The 2^56 bit in the constant
 * is intended to force the bits of "sum" into the least significant bits
 * of the double mantissa.  After an integer substraction of 0x80000000
 * we have the original double value "sum" converted to an 32-bit int value.
 *
 * (Is that really faster than the clean and simple old version of the macro?)
 */

/*
 * On a SPARC cpu, we fetch the low-order 32-bit from the second 32-bit
 * word of the double fp value stored in memory.  On an x86 cpu, we fetch it
 * from the first 32-bit word.
 * I'm not sure if the HAVE_BIGENDIAN feature test covers all possible memory
 * layouts of double floating point values an all cpu architectures.  If
 * it doesn't work for you, just enable the "old WRITE_SAMPLE" macro.
 */
#if HAVE_BIGENDIAN
#define MANTISSA_OFFSET 1
#else
#define MANTISSA_OFFSET 0
#endif

   /* sizeof(int) == 4 */
#define WRITE_SAMPLE(samples,sum,clip) { \
  union { double dtemp; int itemp[2]; } u; int v; \
  u.dtemp = ((((65536.0 * 65536.0 * 16)+(65536.0 * 0.5))* 65536.0)) + (sum);\
  v = u.itemp[MANTISSA_OFFSET] - 0x80000000; \
  if( v > 32767) { *(samples) = 0x7fff; (clip)++; } \
  else if( v < -32768) { *(samples) = -0x8000; (clip)++; } \
  else { *(samples) = v; } \
}
#endif


/*
#define WRITE_SAMPLE(samples,sum,clip) { \
  double dtemp; int v;                    \
  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; } \
}
*/

static int synth_1to1(real *bandPtr,int channel,unsigned char *out,int *pnt);

static 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 synth_func_t synth_func;

#if HAVE_ALTIVEC
#define dct64_base(a,b,c) if(gCpuCaps.hasAltiVec) dct64_altivec(a,b,c); else dct64(a,b,c)
#else /* HAVE_ALTIVEC */
#define dct64_base(a,b,c) dct64(a,b,c)
#endif /* HAVE_ALTIVEC */

static int synth_1to1(real *bandPtr,int channel,unsigned char *out,int *pnt)
{
  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;

  *pnt += 128;

/* optimized for x86 */
#if ARCH_X86
  if ( synth_func )
   {
//    printf("Calling %p, bandPtr=%p channel=%d samples=%p\n",synth_func,bandPtr,channel,samples);
    // FIXME: synth_func() may destroy EBP, don't rely on stack contents!!!
    return (*synth_func)( bandPtr,channel,samples);
   }
#endif
  if(!channel) {     /* channel=0 */
    bo--;
    bo &= 0xf;
    buf = buffs[0];
  }
  else {
    samples++;
    buf = buffs[1];
  }

  if(bo & 0x1) {
    b0 = buf[0];
    bo1 = bo;
    dct64_base(buf[1]+((bo+1)&0xf),buf[0]+bo,bandPtr);
  }
  else {
    b0 = buf[1];
    bo1 = bo+1;
    dct64_base(buf[0]+bo,buf[1]+bo+1,bandPtr);
  }

  {
    register int j;
    real *window = mp3lib_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);
    }
  }

  return clip;

}

#ifdef CONFIG_FAKE_MONO
static int synth_1to1_l(real *bandPtr,int channel,unsigned char *out,int *pnt)
{
  int i,ret;

  ret = synth_1to1(bandPtr,channel,out,pnt);
  out = out + *pnt - 128;

  for(i=0;i<32;i++) {
    ((short *)out)[1] = ((short *)out)[0];
    out+=4;
  }

  return ret;
}

static int synth_1to1_r(real *bandPtr,int channel,unsigned char *out,int *pnt)
{
  int i,ret;

  ret = synth_1to1(bandPtr,channel,out,pnt);
  out = out + *pnt - 128;

  for(i=0;i<32;i++) {
    ((short *)out)[0] = ((short *)out)[1];
    out+=4;
  }

  return ret;
}
#endif