r1053: Add Russian translation.

[cinelerra_cv.git] / toolame-02l / psycho_1.c

#include <stdio.h>
#include <math.h>
#include "common.h"
#include "encoder.h"
#include "mem.h"
#include "fft.h"
#include "psycho_1.h"
#include "psycho_1_priv.h"

#define DBTAB 1000
double dbtable[DBTAB];

/**********************************************************************

        This module implements the psychoacoustic model I for the
 MPEG encoder layer II. It uses simplified tonal and noise masking
 threshold analysis to generate SMR for the encoder bit allocation
 routine.

**********************************************************************/

void psycho_1 (short buffer[2][1152], double scale[2][SBLIMIT],
               double ltmin[2][SBLIMIT], frame_info * frame)
{
  frame_header *header = frame->header;
  int nch = frame->nch;
  int sblimit = frame->sblimit;
  int k, i, tone = 0, noise = 0;
  static char init = 0;
  static int off[2] = { 256, 256 };
  double sample[FFT_SIZE];
  double spike[2][SBLIMIT];
  static D1408 *fft_buf;
  static mask_ptr power;
  static g_ptr ltg;
  FLOAT energy[FFT_SIZE];

  /* call functions for critical boundaries, freq. */
  if (!init) {                  /* bands, bark values, and mapping */
    fft_buf = (D1408 *) mem_alloc ((long) sizeof (D1408) * 2, "fft_buf");
    power = (mask_ptr) mem_alloc (sizeof (mask) * HAN_SIZE, "power");
    if (header->version == MPEG_AUDIO_ID) {
      psycho_1_read_cbound (header->lay, header->sampling_frequency);
      psycho_1_read_freq_band (&ltg, header->lay, header->sampling_frequency);
    } else {
      psycho_1_read_cbound (header->lay, header->sampling_frequency + 4);
      psycho_1_read_freq_band (&ltg, header->lay, header->sampling_frequency + 4);
    }
    psycho_1_make_map (power, ltg);
    for (i = 0; i < 1408; i++)
      fft_buf[0][i] = fft_buf[1][i] = 0;

    psycho_1_init_add_db ();            /* create the add_db table */

    init = 1;
  }
  for (k = 0; k < nch; k++) {
    /* check pcm input for 3 blocks of 384 samples */
    /* sami's speedup, added in 02j
       saves about 4% overall during an encode */
    int ok = off[k] % 1408;
    for (i = 0; i < 1152; i++) {
      fft_buf[k][ok++] = (double) buffer[k][i] / SCALE;
      if (ok >= 1408)
        ok = 0;
    }
    ok = (off[k] + 1216) % 1408;
    for (i = 0; i < FFT_SIZE; i++) {
      sample[i] = fft_buf[k][ok++];
      if (ok >= 1408)
        ok = 0;
    }
    off[k] += 1152;
    off[k] %= 1408;

    psycho_1_hann_fft_pickmax (sample, power, &spike[k][0], energy);
    psycho_1_tonal_label (power, &tone);
    psycho_1_noise_label (power, &noise, ltg, energy);
    //psycho_1_dump(power, &tone, &noise) ;
    psycho_1_subsampling (power, ltg, &tone, &noise);
    psycho_1_threshold (power, ltg, &tone, &noise,
               bitrate[header->version][header->bitrate_index] / nch);
    psycho_1_minimum_mask (ltg, &ltmin[k][0], sblimit);
    psycho_1_smr (&ltmin[k][0], &spike[k][0], &scale[k][0], sblimit);
  }

}


int crit_band;
int *cbound;
int sub_size;

void psycho_1_read_cbound (int lay, int freq)   
/* this function reads in critical  band boundaries */
{

#include "critband.h"
  //static const int FirstCriticalBand[7][27] = {...

  int i, k;

  if ((lay < 1) || (lay > 2)) {
    printf ("Internal error (read_cbound())\n");
    return;
  }
  if ((freq < 0) || (freq > 6) || (freq == 3)) {
    printf ("Internal error (read_cbound())\n");
    return;
  }

  crit_band = SecondCriticalBand[freq][0];
  cbound = (int *) mem_alloc (sizeof (int) * crit_band, "cbound");
  for (i = 0; i < crit_band; i++) {
    k = SecondCriticalBand[freq][i + 1];
    if (k != 0) {
      cbound[i] = k;
    } else {
      printf ("Internal error (read_cbound())\n");
      return;
    }
  }
}

void psycho_1_read_freq_band (ltg, lay, freq)   /* this function reads in   */
     int lay, freq;             /* frequency bands and bark */
     g_ptr *ltg;                /* values                   */
{

#include "freqtable.h"

  int i, k;

  if ((freq < 0) || (freq > 6) || (freq == 3)) {
    printf ("Internal error (read_freq_band())\n");
    return;
  }

  /* read input for freq. subbands */

  sub_size = SecondFreqEntries[freq] + 1;
  *ltg = (g_ptr) mem_alloc (sizeof (g_thres) * sub_size, "ltg");
  (*ltg)[0].line = 0;           /* initialize global masking threshold */
  (*ltg)[0].bark = 0.0;
  (*ltg)[0].hear = 0.0;
  for (i = 1; i < sub_size; i++) {
    k = SecondFreqSubband[freq][i - 1].line;
    if (k != 0) {
      (*ltg)[i].line = k;
      (*ltg)[i].bark = SecondFreqSubband[freq][i - 1].bark;
      (*ltg)[i].hear = SecondFreqSubband[freq][i - 1].hear;
    } else {
      printf ("Internal error (read_freq_band())\n");
        return;
    }
  }
}


void psycho_1_make_map (mask power[HAN_SIZE], g_thres * ltg)
/* this function calculates the global masking threshold */
{
  int i, j;

  for (i = 1; i < sub_size; i++)
    for (j = ltg[i - 1].line; j <= ltg[i].line; j++)
      power[j].map = i;
}

void psycho_1_init_add_db (void)
{
  int i;
  double x;
  for (i = 0; i < DBTAB; i++) {
    x = (double) i / 10.0;
    dbtable[i] = 10 * log10 (1 + pow (10.0, x / 10.0)) - x;
  }
}

INLINE double add_db (double a, double b)
{
  /* MFC - if the difference between a and b is large (>99), then just return the
     largest one. (about 10% of the time)
     - For differences between 0 and 99, return the largest value, but add
     in a pre-calculated difference value.  
     - the value 99 was chosen arbitarily.
     - maximum (a-b) i've seen is 572 */
  FLOAT fdiff;
  int idiff;
  fdiff = (10.0 * (a - b));

  if (fdiff > 990.0) {
    return a;
  }
  if (fdiff < -990.0) {
    return (b);
  }

  idiff = (int) fdiff;
  if (idiff >= 0) {
    return (a + dbtable[idiff]);
  }

  return (b + dbtable[-idiff]);
}

/****************************************************************
*       Window the samples then, 
*        Fast Fourier transform of the input samples.
* 
*   (  call the FHT-based fft() in fft.c )
*    
*
****************************************************************/
void psycho_1_hann_fft_pickmax (double sample[FFT_SIZE], mask power[HAN_SIZE],
                       double spike[SBLIMIT], FLOAT energy[FFT_SIZE])
{
  FLOAT x_real[FFT_SIZE];
  register int i, j;
  register double sqrt_8_over_3;
  static int init = 0;
  static double *window;
  double sum;

  if (!init) {                  /* calculate window function for the Fourier transform */
    window = (double *) mem_alloc (sizeof (DFFT), "window");
    sqrt_8_over_3 = pow (8.0 / 3.0, 0.5);
    for (i = 0; i < FFT_SIZE; i++) {
      /* Hann window formula */
      window[i] =
        sqrt_8_over_3 * 0.5 * (1 -
                               cos (2.0 * PI * i / (FFT_SIZE))) / FFT_SIZE;
    }
    init = 1;
  }
  for (i = 0; i < FFT_SIZE; i++)
    x_real[i] = (FLOAT) (sample[i] * window[i]);

  psycho_1_fft (x_real, energy, FFT_SIZE);

  for (i = 0; i < HAN_SIZE; i++) {      /* calculate power density spectrum */
    if (energy[i] < 1E-20)
      power[i].x = -200.0 + POWERNORM;
    else
      power[i].x = 10 * log10 (energy[i]) + POWERNORM;
    power[i].next = STOP;
    power[i].type = FALSE;
  }

  /* Calculate the sum of spectral component in each subband from bound 4-16 */

#define CF 1073741824           /* pow(10, 0.1*POWERNORM) */
#define DBM  1E-20              /* pow(10.0, 0.1*DBMIN */
  for (i = 0; i < HAN_SIZE; spike[i >> 4] = 10.0 * log10 (sum), i += 16) {
    for (j = 0, sum = DBM; j < 16; j++)
      sum += CF * energy[i + j];
  }
}

/****************************************************************
*
*        This function labels the tonal component in the power
* spectrum.
*
****************************************************************/

void psycho_1_tonal_label (mask power[HAN_SIZE], int *tone)
/* this function extracts (tonal)  sinusoidals from the spectrum  */
{
  int i, j, last = LAST, first, run, last_but_one = LAST;       /* dpwe */
  double max;

  *tone = LAST;
  for (i = 2; i < HAN_SIZE - 12; i++) {
    if (power[i].x > power[i - 1].x && power[i].x >= power[i + 1].x) {
      power[i].type = TONE;
      power[i].next = LAST;
      if (last != LAST)
        power[last].next = i;
      else
        first = *tone = i;
      last = i;
    }
  }
  last = LAST;
  first = *tone;
  *tone = LAST;
  while ((first != LAST) && (first != STOP)) {  /* the conditions for the tonal          */
    if (first < 3 || first > 500)
      run = 0;                  /* otherwise k+/-j will be out of bounds */
    else if (first < 63)
      run = 2;                  /* components in layer II, which         */
    else if (first < 127)
      run = 3;                  /* are the boundaries for calc.          */
    else if (first < 255)
      run = 6;                  /* the tonal components                  */
    else
      run = 12;
    max = power[first].x - 7;   /* after calculation of tonal   */
    for (j = 2; j <= run; j++)  /* components, set to local max */
      if (max < power[first - j].x || max < power[first + j].x) {
        power[first].type = FALSE;
        break;
      }
    if (power[first].type == TONE) {    /* extract tonal components */
      int help = first;
      if (*tone == LAST)
        *tone = first;
      while ((power[help].next != LAST) && (power[help].next - first) <= run)
        help = power[help].next;
      help = power[help].next;
      power[first].next = help;
      if ((first - last) <= run) {
        if (last_but_one != LAST)
          power[last_but_one].next = first;
      }
      if (first > 1 && first < 500) {   /* calculate the sum of the */
        double tmp;             /* powers of the components */
        tmp = add_db (power[first - 1].x, power[first + 1].x);
        power[first].x = add_db (power[first].x, tmp);
      }
      for (j = 1; j <= run; j++) {
        power[first - j].x = power[first + j].x = DBMIN;
        power[first - j].next = power[first + j].next = STOP;
        power[first - j].type = power[first + j].type = FALSE;
      }
      last_but_one = last;
      last = first;
      first = power[first].next;
    } else {
      int ll;
      if (last == LAST);        /* *tone = power[first].next; dpwe */
      else
        power[last].next = power[first].next;
      ll = first;
      first = power[first].next;
      power[ll].next = STOP;
    }
  }
}

/****************************************************************
*
*        This function groups all the remaining non-tonal
* spectral lines into critical band where they are replaced by
* one single line.
*
****************************************************************/

void psycho_1_noise_label (mask * power, int *noise, g_thres * ltg,
                  FLOAT energy[FFT_SIZE])
{
  int i, j, centre, last = LAST;
  double index, weight, sum;
  /* calculate the remaining spectral */
  for (i = 0; i < crit_band - 1; i++) { /* lines for non-tonal components   */
    for (j = cbound[i], weight = 0.0, sum = DBMIN; j < cbound[i + 1]; j++) {
      if (power[j].type != TONE) {
        if (power[j].x != DBMIN) {
          sum = add_db (power[j].x, sum);
          /* Weight is used in finding the geometric mean of the noise energy within a subband */
          weight += CF * energy[j] * (double) (j - cbound[i]) / (double) (cbound[i + 1] - cbound[i]);   /* correction */
          power[j].x = DBMIN;
        }
      }                         /*  check to see if the spectral line is low dB, and if  */
    }                           /* so replace the center of the critical band, which is */
    /* the center freq. of the noise component              */

    if (sum <= DBMIN)
      centre = (cbound[i + 1] + cbound[i]) / 2;
    else {
      /* fprintf(stderr, "%i [%f %f] -", count++,weight/pow(10.0,0.1*sum), weight*pow(10.0,-0.1*sum)); */
      index = weight * pow (10.0, -0.1 * sum);
      centre =
        cbound[i] + (int) (index * (double) (cbound[i + 1] - cbound[i]));
    }


    /* locate next non-tonal component until finished; */
    /* add to list of non-tonal components             */

    /* Masahiro Iwadare's fix for infinite looping problem? */
    if (power[centre].type == TONE) {
      if (power[centre + 1].type == TONE) {
        centre++;
      } else
        centre--;
    }

    if (last == LAST)
      *noise = centre;
    else {
      power[centre].next = LAST;
      power[last].next = centre;
    }
    power[centre].x = sum;
    power[centre].type = NOISE;
    last = centre;
  }
}

/****************************************************************
*
*        This function reduces the number of noise and tonal
* component for further threshold analysis.
*
****************************************************************/

void psycho_1_subsampling (mask power[HAN_SIZE], g_thres * ltg, int *tone, int *noise)
{
  int i, old;

  i = *tone;
  old = STOP;                   /* calculate tonal components for */

  while ((i != LAST) && (i != STOP))
  {                             /* reduction of spectral lines    */
    if (power[i].x < ltg[power[i].map].hear) {
      power[i].type = FALSE;
      power[i].x = DBMIN;
      if (old == STOP)
        *tone = power[i].next;
      else
        power[old].next = power[i].next;
    } else
      old = i;
    i = power[i].next;
  }
  i = *noise;
  old = STOP;                   /* calculate non-tonal components for */
  while ((i != LAST) && (i != STOP)) {  /* reduction of spectral lines        */
    if (power[i].x < ltg[power[i].map].hear) {
      power[i].type = FALSE;
      power[i].x = DBMIN;
      if (old == STOP)
        *noise = power[i].next;
      else
        power[old].next = power[i].next;
    } else
      old = i;
    i = power[i].next;
  }
  i = *tone;
  old = STOP;
  while ((i != LAST) && (i != STOP))
  {                             /* if more than one */
    if (power[i].next == LAST)
      break;                    /* tonal component  */
    if (ltg[power[power[i].next].map].bark -    /* is less than .5  */
        ltg[power[i].map].bark < 0.5) { /* bark, take the   */
      if (power[power[i].next].x > power[i].x) {        /* maximum          */
        if (old == STOP)
          *tone = power[i].next;
        else
          power[old].next = power[i].next;
        power[i].type = FALSE;
        power[i].x = DBMIN;
        i = power[i].next;
      } else {
        power[power[i].next].type = FALSE;
        power[power[i].next].x = DBMIN;
        power[i].next = power[power[i].next].next;
        old = i;
      }
    } else {
      old = i;
      i = power[i].next;
    }
  }
}

/****************************************************************
*
*        This function calculates the individual threshold and
* sum with the quiet threshold to find the global threshold.
*
****************************************************************/

/* mainly just changed the way range checking was done MFC Nov 1999 */
void psycho_1_threshold (mask power[HAN_SIZE], g_thres * ltg, int *tone, int *noise,
                int bit_rate)
{
  int k, t;
  double dz, tmps, vf;

  for (k = 1; k < sub_size; k++) {
    ltg[k].x = DBMIN;
    t = *tone;                  /* calculate individual masking threshold for */
    while ((t != LAST) && (t != STOP))
    {                           /* components in order to find the global     */
      dz = ltg[k].bark - ltg[power[t].map].bark;        /* distance of bark value */
      if (dz >= -3.0 && dz < 8.0) {
        tmps = -1.525 - 0.275 * ltg[power[t].map].bark - 4.5 + power[t].x;
        /* masking function for lower & upper slopes */
        if (dz < -1)
          vf = 17 * (dz + 1) - (0.4 * power[t].x + 6);
        else if (dz < 0)
          vf = (0.4 * power[t].x + 6) * dz;
        else if (dz < 1)
          vf = (-17 * dz);
        else
          vf = -(dz - 1) * (17 - 0.15 * power[t].x) - 17;
        ltg[k].x = add_db (ltg[k].x, tmps + vf);
      }
      t = power[t].next;
    }

    t = *noise;                 /* calculate individual masking threshold  */
    while ((t != LAST) && (t != STOP)) {        /* for non-tonal components to find LTG    */
      dz = ltg[k].bark - ltg[power[t].map].bark;        /* distance of bark value */
      if (dz >= -3.0 && dz < 8.0) {
        tmps = -1.525 - 0.175 * ltg[power[t].map].bark - 0.5 + power[t].x;
        /* masking function for lower & upper slopes */
        if (dz < -1)
          vf = 17 * (dz + 1) - (0.4 * power[t].x + 6);
        else if (dz < 0)
          vf = (0.4 * power[t].x + 6) * dz;
        else if (dz < 1)
          vf = (-17 * dz);
        else
          vf = -(dz - 1) * (17 - 0.15 * power[t].x) - 17;
        ltg[k].x = add_db (ltg[k].x, tmps + vf);
      }
      t = power[t].next;
    }
    if (bit_rate < 96)
      ltg[k].x = add_db (ltg[k].hear, ltg[k].x);
    else
      ltg[k].x = add_db (ltg[k].hear - 12.0, ltg[k].x);
  }

}

/****************************************************************
*
*        This function finds the minimum masking threshold and
* return the value to the encoder.
*
****************************************************************/

void psycho_1_minimum_mask (g_thres * ltg, double ltmin[SBLIMIT], int sblimit)
{
  double min;
  int i, j;

  j = 1;
  for (i = 0; i < sblimit; i++)
    if (j >= sub_size - 1)      /* check subband limit, and       */
      ltmin[i] = ltg[sub_size - 1].hear;        /* calculate the minimum masking  */
    else {                      /* level of LTMIN for each subband */
      min = ltg[j].x;
      while (ltg[j].line >> 4 == i && j < sub_size) {
        if (min > ltg[j].x)
          min = ltg[j].x;
        j++;
      }
      ltmin[i] = min;
    }
}

/*****************************************************************
*
*        This procedure is called in musicin to pick out the
* smaller of the scalefactor or threshold.
*
*****************************************************************/

void psycho_1_smr (double ltmin[SBLIMIT], double spike[SBLIMIT], double scale[SBLIMIT],
          int sblimit)
{
  int i;
  double max;

  for (i = 0; i < sblimit; i++) {       /* determine the signal   */
    max = 20 * log10 (scale[i] * 32768) - 10;   /* level for each subband */
    if (spike[i] > max)
      max = spike[i];           /* for the maximum scale  */
    max -= ltmin[i];            /* factors                */
    ltmin[i] = max;
  }
}

void psycho_1_dump(mask power[HAN_SIZE], int *tone, int *noise) {
  int t;

  fprintf(stdout,"1 Ton: ");
  t=*tone;
  while (t!=LAST && t!=STOP) {
    fprintf(stdout,"[%i] %3.0f ",t, power[t].x);
    t = power[t].next;
  }
  fprintf(stdout,"\n");  
  
  fprintf(stdout,"1 Nos: ");
  t=*noise;
  while (t!=LAST && t!=STOP) {
    fprintf(stdout,"[%i] %3.0f ",t, power[t].x);
    t = power[t].next;
  }
  fprintf(stdout,"\n");
}