Include toolame-dab as library

1 files changed, 601 insertions, 0 deletions

diff --git a/libtoolame-dab/psycho_1.c b/libtoolame-dab/psycho_1.c
new file mode 100644
index 0000000..004813c
--- /dev/null
+++ b/libtoolame-dab/psycho_1.c
@@ -0,0 +1,601 @@
+#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;
+  }
+}
+
+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");
+}