polyphase.c

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/**************************************************************************************
 * Fixed-point MP3 decoder
 * Jon Recker (jrecker@real.com), Ken Cooke (kenc@real.com)
 * June 2003
 *
 * polyphase.c - final stage of subband transform (polyphase synthesis filter)
 *
 * This is the C reference version using __int64
 * Look in the appropriate subdirectories for optimized asm implementations 
 *   (e.g. arm/asmpoly.s)
 **************************************************************************************/
 
#include "coder.h"
#include "assembly.h"
 
/* input to Polyphase = Q(DQ_FRACBITS_OUT-2), gain 2 bits in convolution
 *  we also have the implicit bias of 2^15 to add back, so net fraction bits = 
 *    DQ_FRACBITS_OUT - 2 - 2 - 15
 *  (see comment on Dequantize() for more info)
 */
#define DEF_NFRACBITS (DQ_FRACBITS_OUT - 2 - 2 - 15)  
#define CSHIFT  12  /* coefficients have 12 leading sign bits for early-terminating mulitplies */
 
static __inline short ClipToShort(int x, int fracBits)
{
  int sign;
  
  /* assumes you've already rounded (x += (1 << (fracBits-1))) */
  x >>= fracBits;
  
  /* Ken's trick: clips to [-32768, 32767] */
  sign = x >> 31;
  if (sign != (x >> 15))
    x = sign ^ ((1 << 15) - 1);
 
  return (short)x;
}
 
#define MC0M(x) { \
  c1 = *coef;   coef++;   c2 = *coef;   coef++; \
  vLo = *(vb1+(x));     vHi = *(vb1+(23-(x))); \
  sum1L = MADD64(sum1L, vLo,  c1);  sum1L = MADD64(sum1L, vHi, -c2); \
}
 
#define MC1M(x) { \
  c1 = *coef;   coef++; \
  vLo = *(vb1+(x)); \
  sum1L = MADD64(sum1L, vLo,  c1); \
}
 
#define MC2M(x) { \
    c1 = *coef;   coef++;   c2 = *coef;   coef++; \
    vLo = *(vb1+(x)); vHi = *(vb1+(23-(x))); \
    sum1L = MADD64(sum1L, vLo,  c1);  sum2L = MADD64(sum2L, vLo,  c2); \
    sum1L = MADD64(sum1L, vHi, -c2);  sum2L = MADD64(sum2L, vHi,  c1); \
}
 
/**************************************************************************************
 * Function:    PolyphaseMono
 *
 * Description: filter one subband and produce 32 output PCM samples for one channel
 *
 * Inputs:      pointer to PCM output buffer
 *              number of "extra shifts" (vbuf format = Q(DQ_FRACBITS_OUT-2))
 *              pointer to start of vbuf (preserved from last call)
 *              start of filter coefficient table (in proper, shuffled order)
 *              no minimum number of guard bits is required for input vbuf 
 *                (see additional scaling comments below)
 *
 * Outputs:     32 samples of one channel of decoded PCM data, (i.e. Q16.0)
 *
 * Return:      none
 *
 * TODO:        add 32-bit version for platforms where 64-bit mul-acc is not supported
 *                (note max filter gain - see polyCoef[] comments)
 **************************************************************************************/
void PolyphaseMono(short *pcm, int *vbuf, const int *coefBase)
{ 
  int i;
  const int *coef;
  int *vb1;
  int vLo, vHi, c1, c2;
  Word64 sum1L, sum2L, rndVal;
 
  rndVal = (Word64)( 1 << (DEF_NFRACBITS - 1 + (32 - CSHIFT)) );
 
  /* special case, output sample 0 */
  coef = coefBase;
  vb1 = vbuf;
  sum1L = rndVal;
 
  MC0M(0)
  MC0M(1)
  MC0M(2)
  MC0M(3)
  MC0M(4)
  MC0M(5)
  MC0M(6)
  MC0M(7)
 
  *(pcm + 0) = ClipToShort((int)SAR64(sum1L, (32-CSHIFT)), DEF_NFRACBITS);
 
  /* special case, output sample 16 */
  coef = coefBase + 256;
  vb1 = vbuf + 64*16;
  sum1L = rndVal;
 
  MC1M(0)
  MC1M(1)
  MC1M(2)
  MC1M(3)
  MC1M(4)
  MC1M(5)
  MC1M(6)
  MC1M(7)
 
  *(pcm + 16) = ClipToShort((int)SAR64(sum1L, (32-CSHIFT)), DEF_NFRACBITS);
 
  /* main convolution loop: sum1L = samples 1, 2, 3, ... 15   sum2L = samples 31, 30, ... 17 */
  coef = coefBase + 16;
  vb1 = vbuf + 64;
  pcm++;
 
  /* right now, the compiler creates bad asm from this... */
  for (i = 15; i > 0; i--) {
    sum1L = sum2L = rndVal;
 
    MC2M(0)
    MC2M(1)
    MC2M(2)
    MC2M(3)
    MC2M(4)
    MC2M(5)
    MC2M(6)
    MC2M(7)
 
    vb1 += 64;
    *(pcm)       = ClipToShort((int)SAR64(sum1L, (32-CSHIFT)), DEF_NFRACBITS);
    *(pcm + 2*i) = ClipToShort((int)SAR64(sum2L, (32-CSHIFT)), DEF_NFRACBITS);
    pcm++;
  }
}
 
#define MC0S(x) { \
  c1 = *coef;   coef++;   c2 = *coef;   coef++; \
  vLo = *(vb1+(x));   vHi = *(vb1+(23-(x))); \
  sum1L = MADD64(sum1L, vLo,  c1);  sum1L = MADD64(sum1L, vHi, -c2); \
  vLo = *(vb1+32+(x));  vHi = *(vb1+32+(23-(x))); \
  sum1R = MADD64(sum1R, vLo,  c1);  sum1R = MADD64(sum1R, vHi, -c2); \
}
 
#define MC1S(x) { \
  c1 = *coef;   coef++; \
  vLo = *(vb1+(x)); \
  sum1L = MADD64(sum1L, vLo,  c1); \
  vLo = *(vb1+32+(x)); \
  sum1R = MADD64(sum1R, vLo,  c1); \
}
 
#define MC2S(x) { \
    c1 = *coef;   coef++;   c2 = *coef;   coef++; \
    vLo = *(vb1+(x)); vHi = *(vb1+(23-(x))); \
    sum1L = MADD64(sum1L, vLo,  c1);  sum2L = MADD64(sum2L, vLo,  c2); \
    sum1L = MADD64(sum1L, vHi, -c2);  sum2L = MADD64(sum2L, vHi,  c1); \
    vLo = *(vb1+32+(x));  vHi = *(vb1+32+(23-(x))); \
    sum1R = MADD64(sum1R, vLo,  c1);  sum2R = MADD64(sum2R, vLo,  c2); \
    sum1R = MADD64(sum1R, vHi, -c2);  sum2R = MADD64(sum2R, vHi,  c1); \
}
 
/**************************************************************************************
 * Function:    PolyphaseStereo
 *
 * Description: filter one subband and produce 32 output PCM samples for each channel
 *
 * Inputs:      pointer to PCM output buffer
 *              number of "extra shifts" (vbuf format = Q(DQ_FRACBITS_OUT-2))
 *              pointer to start of vbuf (preserved from last call)
 *              start of filter coefficient table (in proper, shuffled order)
 *              no minimum number of guard bits is required for input vbuf 
 *                (see additional scaling comments below)
 *
 * Outputs:     32 samples of two channels of decoded PCM data, (i.e. Q16.0)
 *
 * Return:      none
 *
 * Notes:       interleaves PCM samples LRLRLR...
 *
 * TODO:        add 32-bit version for platforms where 64-bit mul-acc is not supported
 **************************************************************************************/
void PolyphaseStereo(short *pcm, int *vbuf, const int *coefBase)
{
  int i;
  const int *coef;
  int *vb1;
  int vLo, vHi, c1, c2;
  Word64 sum1L, sum2L, sum1R, sum2R, rndVal;
 
  rndVal = (Word64)( 1 << (DEF_NFRACBITS - 1 + (32 - CSHIFT)) );
 
  /* special case, output sample 0 */
  coef = coefBase;
  vb1 = vbuf;
  sum1L = sum1R = rndVal;
 
  MC0S(0)
  MC0S(1)
  MC0S(2)
  MC0S(3)
  MC0S(4)
  MC0S(5)
  MC0S(6)
  MC0S(7)
 
  *(pcm + 0) = ClipToShort((int)SAR64(sum1L, (32-CSHIFT)), DEF_NFRACBITS);
  *(pcm + 1) = ClipToShort((int)SAR64(sum1R, (32-CSHIFT)), DEF_NFRACBITS);
 
  /* special case, output sample 16 */
  coef = coefBase + 256;
  vb1 = vbuf + 64*16;
  sum1L = sum1R = rndVal;
 
  MC1S(0)
  MC1S(1)
  MC1S(2)
  MC1S(3)
  MC1S(4)
  MC1S(5)
  MC1S(6)
  MC1S(7)
 
  *(pcm + 2*16 + 0) = ClipToShort((int)SAR64(sum1L, (32-CSHIFT)), DEF_NFRACBITS);
  *(pcm + 2*16 + 1) = ClipToShort((int)SAR64(sum1R, (32-CSHIFT)), DEF_NFRACBITS);
 
  /* main convolution loop: sum1L = samples 1, 2, 3, ... 15   sum2L = samples 31, 30, ... 17 */
  coef = coefBase + 16;
  vb1 = vbuf + 64;
  pcm += 2;
 
  /* right now, the compiler creates bad asm from this... */
  for (i = 15; i > 0; i--) {
    sum1L = sum2L = rndVal;
    sum1R = sum2R = rndVal;
 
    MC2S(0)
    MC2S(1)
    MC2S(2)
    MC2S(3)
    MC2S(4)
    MC2S(5)
    MC2S(6)
    MC2S(7)
 
    vb1 += 64;
    *(pcm + 0)         = ClipToShort((int)SAR64(sum1L, (32-CSHIFT)), DEF_NFRACBITS);
    *(pcm + 1)         = ClipToShort((int)SAR64(sum1R, (32-CSHIFT)), DEF_NFRACBITS);
    *(pcm + 2*2*i + 0) = ClipToShort((int)SAR64(sum2L, (32-CSHIFT)), DEF_NFRACBITS);
    *(pcm + 2*2*i + 1) = ClipToShort((int)SAR64(sum2R, (32-CSHIFT)), DEF_NFRACBITS);
    pcm += 2;
  }
}