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-rw-r--r--libtoolame-dab/psycho_1.c601
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diff --git a/libtoolame-dab/psycho_1.c b/libtoolame-dab/psycho_1.c
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+++ b/libtoolame-dab/psycho_1.c
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+#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");
+}
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