encode.c

ISO mp3 sources (distribution 10) Layer 1/2/3, C Source, 512 k Sources of the Mpeg 1,2 layer 1,2

double FAR filter[SBLIMIT][64];
{
   register int i,k;
 
   for (i=0; i<32; i++)
      for (k=0; k<64; k++) {
          if ((filter[i][k] = 1e9*cos((double)((2*i+1)*(16-k)*PI64))) >= 0)
             modf(filter[i][k]+0.5, &filter[i][k]);
          else
             modf(filter[i][k]-0.5, &filter[i][k]);
          filter[i][k] *= 1e-9;
   }
}

/************************************************************************
*
* filter_subband()
*
* PURPOSE:  Calculates the analysis filter bank coefficients
*
* SEMANTICS:
*      The windowed samples #z# is filtered by the digital filter matrix #m#
* to produce the subband samples #s#. This done by first selectively
* picking out values from the windowed samples, and then multiplying
* them by the filter matrix, producing 32 subband samples.
*
************************************************************************/
 
void filter_subband(z,s)
double FAR z[HAN_SIZE], s[SBLIMIT];
{
   double y[64];
   int i,j;
static char init = 0;
   typedef double MM[SBLIMIT][64];
static MM FAR *m;
#ifdef MS_DOS
   long    SIZE_OF_MM;
   SIZE_OF_MM      = SBLIMIT*64;
   SIZE_OF_MM      *= 8;
   if (!init) {
       m = (MM FAR *) mem_alloc(SIZE_OF_MM, "filter");
       create_ana_filter(*m);
       init = 1;
   }
#else
   if (!init) {
       m = (MM FAR *) mem_alloc(sizeof(MM), "filter");
       create_ana_filter(*m);
       init = 1;
   }
#endif
   for (i=0;i<64;i++) for (j=0, y[i] = 0;j<8;j++) y[i] += z[i+64*j];
   for (i=0;i<SBLIMIT;i++)
       for (j=0, s[i]= 0;j<64;j++) s[i] += (*m)[i][j] * y[j];
}

/************************************************************************
* encode_info()
*
* PURPOSE:  Puts the syncword and header information on the output
* bitstream.
*
************************************************************************/
 
void encode_info(fr_ps,bs)
frame_params *fr_ps;
Bit_stream_struc *bs;
{
        layer *info = fr_ps->header;
 
        putbits(bs,0xfff,12);                    /* syncword 12 bits */
        put1bit(bs,info->version);               /* ID        1 bit  */
        putbits(bs,4-info->lay,2);               /* layer     2 bits */
        put1bit(bs,!info->error_protection);     /* bit set => no err prot */
        putbits(bs,info->bitrate_index,4);
        putbits(bs,info->sampling_frequency,2);
        put1bit(bs,info->padding);
        put1bit(bs,info->extension);             /* private_bit */
        putbits(bs,info->mode,2);
        putbits(bs,info->mode_ext,2);
        put1bit(bs,info->copyright);
        put1bit(bs,info->original);
        putbits(bs,info->emphasis,2);
}
 
/************************************************************************
*
* mod()
*
* PURPOSE:  Returns the absolute value of its argument
*
************************************************************************/
 
double mod(a)
double a;
{
    return (a > 0) ? a : -a;
}
 
/************************************************************************
*
* I_combine_LR    (Layer I)
* II_combine_LR   (Layer II)
*
* PURPOSE:Combines left and right channels into a mono channel
*
* SEMANTICS:  The average of left and right subband samples is put into
* #joint_sample#
*
* Layer I and II differ in frame length and # subbands used
*
************************************************************************/
 
void I_combine_LR(sb_sample, joint_sample)
double FAR sb_sample[2][3][SCALE_BLOCK][SBLIMIT];
double FAR joint_sample[3][SCALE_BLOCK][SBLIMIT];
{   /* make a filtered mono for joint stereo */
    int sb, smp;
 
   for(sb = 0; sb<SBLIMIT; ++sb)
      for(smp = 0; smp<SCALE_BLOCK; ++smp)
        joint_sample[0][smp][sb] = .5 *
                    (sb_sample[0][0][smp][sb] + sb_sample[1][0][smp][sb]);
}
 
void II_combine_LR(sb_sample, joint_sample, sblimit)
double FAR sb_sample[2][3][SCALE_BLOCK][SBLIMIT];
double FAR joint_sample[3][SCALE_BLOCK][SBLIMIT];
int sblimit;
{  /* make a filtered mono for joint stereo */
   int sb, smp, sufr;
 
   for(sb = 0; sb<sblimit; ++sb)
      for(smp = 0; smp<SCALE_BLOCK; ++smp)
         for(sufr = 0; sufr<3; ++sufr)
            joint_sample[sufr][smp][sb] = .5 * (sb_sample[0][sufr][smp][sb]
                                           + sb_sample[1][sufr][smp][sb]);
}
 
/************************************************************************
*
* I_scale_factor_calc     (Layer I)
* II_scale_factor_calc    (Layer II)
*
* PURPOSE:For each subband, calculate the scale factor for each set
* of the 12 subband samples
*
* SEMANTICS:  Pick the scalefactor #multiple[]# just larger than the
* absolute value of the peak subband sample of 12 samples,
* and store the corresponding scalefactor index in #scalar#.
*
* Layer II has three sets of 12-subband samples for a given
* subband.
*
************************************************************************/
 
void I_scale_factor_calc(sb_sample,scalar,stereo)
double FAR sb_sample[][3][SCALE_BLOCK][SBLIMIT];
unsigned int scalar[][3][SBLIMIT];
int stereo;
{
   int i,j, k;
   double s[SBLIMIT];
 
   for (k=0;k<stereo;k++) {
     for (i=0;i<SBLIMIT;i++)
       for (j=1, s[i] = mod(sb_sample[k][0][0][i]);j<SCALE_BLOCK;j++)
         if (mod(sb_sample[k][0][j][i]) > s[i])
            s[i] = mod(sb_sample[k][0][j][i]);
 
     for (i=0;i<SBLIMIT;i++)
       for (j=SCALE_RANGE-2,scalar[k][0][i]=0;j>=0;j--) /* $A 6/16/92 */
         if (s[i] <= multiple[j]) {
            scalar[k][0][i] = j;
            break;
         }
   }
}

/******************************** Layer II ******************************/
 
void II_scale_factor_calc(sb_sample,scalar,stereo,sblimit)
double FAR sb_sample[][3][SCALE_BLOCK][SBLIMIT];
unsigned int scalar[][3][SBLIMIT];
int stereo,sblimit;
{
  int i,j, k,t;
  double s[SBLIMIT];
 
  for (k=0;k<stereo;k++) for (t=0;t<3;t++) {
    for (i=0;i<sblimit;i++)
      for (j=1, s[i] = mod(sb_sample[k][t][0][i]);j<SCALE_BLOCK;j++)
        if (mod(sb_sample[k][t][j][i]) > s[i])
             s[i] = mod(sb_sample[k][t][j][i]);
 
  for (i=0;i<sblimit;i++)
    for (j=SCALE_RANGE-2,scalar[k][t][i]=0;j>=0;j--)    /* $A 6/16/92 */
      if (s[i] <= multiple[j]) {
         scalar[k][t][i] = j;
         break;
      }
      for (i=sblimit;i<SBLIMIT;i++) scalar[k][t][i] = SCALE_RANGE-1;
    }
}

/************************************************************************
*
* pick_scale  (Layer II)
*
* PURPOSE:For each subband, puts the smallest scalefactor of the 3
* associated with a frame into #max_sc#.  This is used
* used by Psychoacoustic Model I.
* (I would recommend changin max_sc to min_sc)
*
************************************************************************/
 
void pick_scale(scalar, fr_ps, max_sc)
unsigned int scalar[2][3][SBLIMIT];
frame_params *fr_ps;
double FAR max_sc[2][SBLIMIT];
{
  int i,j,k,max;
  int stereo  = fr_ps->stereo;
  int sblimit = fr_ps->sblimit;
 
  for (k=0;k<stereo;k++)
    for (i=0;i<sblimit;max_sc[k][i] = multiple[max],i++)
      for (j=1, max = scalar[k][0][i];j<3;j++)
         if (max > scalar[k][j][i]) max = scalar[k][j][i];
  for (i=sblimit;i<SBLIMIT;i++) max_sc[0][i] = max_sc[1][i] = 1E-20;
}

/************************************************************************
*
* put_scale   (Layer I)
*
* PURPOSE:Sets #max_sc# to the scalefactor index in #scalar.
* This is used by Psychoacoustic Model I
*
************************************************************************/
 
void put_scale(scalar, fr_ps, max_sc)
unsigned int scalar[2][3][SBLIMIT];
frame_params *fr_ps;
double FAR max_sc[2][SBLIMIT];
{
   int i,j,k, max;
   int stereo  = fr_ps->stereo;
   int sblimit = fr_ps->sblimit;
 
   for (k=0;k<stereo;k++) for (i=0;i<SBLIMIT;i++)
        max_sc[k][i] = multiple[scalar[k][0][i]];
}
 
/************************************************************************
*
* II_transmission_pattern (Layer II only)
*
* PURPOSE:For a given subband, determines whether to send 1, 2, or
* all 3 of the scalefactors, and fills in the scalefactor
* select information accordingly
*
* SEMANTICS:  The subbands and channels are classified based on how much
* the scalefactors changes over its three values (corresponding
* to the 3 sets of 12 samples per subband).  The classification
* will send 1 or 2 scalefactors instead of three if the scalefactors
* do not change much.  The scalefactor select information,
* #scfsi#, is filled in accordingly.
*
************************************************************************/
 
void II_transmission_pattern(scalar, scfsi, fr_ps)
unsigned int scalar[2][3][SBLIMIT];
unsigned int scfsi[2][SBLIMIT];
frame_params *fr_ps;
{
   int stereo  = fr_ps->stereo;
   int sblimit = fr_ps->sblimit;
   int dscf[2];
   int class[2],i,j,k;
static int pattern[5][5] = {0x123, 0x122, 0x122, 0x133, 0x123,
                            0x113, 0x111, 0x111, 0x444, 0x113,
                            0x111, 0x111, 0x111, 0x333, 0x113,
                            0x222, 0x222, 0x222, 0x333, 0x123,
                            0x123, 0x122, 0x122, 0x133, 0x123};
 
   for (k=0;k<stereo;k++)
     for (i=0;i<sblimit;i++) {
       dscf[0] =  (scalar[k][0][i]-scalar[k][1][i]);
       dscf[1] =  (scalar[k][1][i]-scalar[k][2][i]);
       for (j=0;j<2;j++) {
         if (dscf[j]<=-3) class[j] = 0;
         else if (dscf[j] > -3 && dscf[j] <0) class[j] = 1;
              else if (dscf[j] == 0) class[j] = 2;
                   else if (dscf[j] > 0 && dscf[j] < 3) class[j] = 3;
                        else class[j] = 4;
       }
       switch (pattern[class[0]][class[1]]) {
         case 0x123 :    scfsi[k][i] = 0;
                         break;
         case 0x122 :    scfsi[k][i] = 3;
                         scalar[k][2][i] = scalar[k][1][i];
                         break;
         case 0x133 :    scfsi[k][i] = 3;
                         scalar[k][1][i] = scalar[k][2][i];
                         break;
         case 0x113 :    scfsi[k][i] = 1;
                         scalar[k][1][i] = scalar[k][0][i];
                         break;
         case 0x111 :    scfsi[k][i] = 2;
                         scalar[k][1][i] = scalar[k][2][i] = scalar[k][0][i];
                         break;
         case 0x222 :    scfsi[k][i] = 2;
                         scalar[k][0][i] = scalar[k][2][i] = scalar[k][1][i];
                         break;
         case 0x333 :    scfsi[k][i] = 2;
                         scalar[k][0][i] = scalar[k][1][i] = scalar[k][2][i];
                         break;
         case 0x444 :    scfsi[k][i] = 2;
                         if (scalar[k][0][i] > scalar[k][2][i])
                              scalar[k][0][i] = scalar[k][2][i];
                         scalar[k][1][i] = scalar[k][2][i] = scalar[k][0][i];
      }
   }
}
 
/************************************************************************
*
* I_encode_scale  (Layer I)
* II_encode_scale (Layer II)
*
* PURPOSE:The encoded scalar factor information is arranged and
* queued into the output fifo to be transmitted.
*
* For Layer II, the three scale factors associated with
* a given subband and channel are transmitted in accordance
* with the scfsi, which is transmitted first.
*
************************************************************************/
 
void I_encode_scale(scalar, bit_alloc, fr_ps, bs)
unsigned int scalar[2][3][SBLIMIT];
unsigned int bit_alloc[2][SBLIMIT];
frame_params *fr_ps;
Bit_stream_struc *bs;
{
   int stereo  = fr_ps->stereo;
   int sblimit = fr_ps->sblimit;
   int i,j;
 
   for (i=0;i<SBLIMIT;i++) for (j=0;j<stereo;j++)
      if (bit_alloc[j][i]) putbits(bs,scalar[j][0][i],6);
}
 
/***************************** Layer II  ********************************/
 
void II_encode_scale(bit_alloc, scfsi, scalar, fr_ps, bs)
unsigned int bit_alloc[2][SBLIMIT], scfsi[2][SBLIMIT];
unsigned int scalar[2][3][SBLIMIT];
frame_params *fr_ps;
Bit_stream_struc *bs;
{
   int stereo  = fr_ps->stereo;
   int sblimit = fr_ps->sblimit;