📄 tonal.c
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/**********************************************************************
* ISO MPEG Audio Subgroup Software Simulation Group (1996)
* ISO 13818-3 MPEG-2 Audio Encoder - Lower Sampling Frequency Extension
*
* $Id: tonal.c,v 1.1 1996/02/14 04:04:23 rowlands Exp $
*
* $Log: tonal.c,v $
* Revision 1.1 1996/02/14 04:04:23 rowlands
* Initial revision
*
* Received from Mike Coleman
**********************************************************************/
/**********************************************************************
* date programmers comment *
* 2/25/91 Douglas Wong start of version 1.1 records *
* 3/06/91 Douglas Wong rename: setup.h to endef.h *
* updated I_psycho_one and II_psycho_one*
* 3/11/91 W. J. Carter Added Douglas Wong's updates dated *
* 3/9/91 for I_Psycho_One() and for *
* II_Psycho_One(). *
* 5/10/91 W. Joseph Carter Ported to Macintosh and Unix. *
* Located and fixed numerous software *
* bugs and table data errors. *
* 6/11/91 Davis Pan corrected several bugs *
* based on comments from H. Fuchs *
* 01jul91 dpwe (Aware Inc.) Made pow() args float *
* Removed logical bug in I_tonal_label: *
* Sometimes *tone returned == STOP *
* 7/10/91 Earle Jennings no change necessary in port to MsDos *
* 11sep91 dpwe@aware.com Subtracted 90.3dB from II_f_f_t peaks *
* 10/1/91 Peter W. Farrett Updated II_Psycho_One(),I_Psycho_One()*
* to include comments. *
*11/29/91 Masahiro Iwadare Bug fix regarding POWERNORM *
* fixed several other miscellaneous bugs*
* 2/11/92 W. Joseph Carter Ported new code to Macintosh. Most *
* important fixes involved changing *
* 16-bit ints to long or unsigned in *
* bit alloc routines for quant of 65535 *
* and passing proper function args. *
* Removed "Other Joint Stereo" option *
* and made bitrate be total channel *
* bitrate, irrespective of the mode. *
* Fixed many small bugs & reorganized. *
* 2/12/92 Masahiro Iwadare Fixed some potential bugs in *
* Davis Pan subsampling() *
* 2/25/92 Masahiro Iwadare Fixed some more potential bugs *
* 6/24/92 Tan Ah Peng Modified window for FFT *
* (denominator N-1 to N) *
* Updated all critical band rate & *
* absolute threshold tables and critical*
* boundaries for use with Layer I & II *
* Corrected boundary limits for tonal *
* component computation *
* Placement of non-tonal component at *
* geometric mean of critical band *
* (previous placement method commented *
* out - can be used if desired) *
* 3/01/93 Mike Li Infinite looping fix in noise_label() *
* 3/19/93 Jens Spille fixed integer overflow problem in *
* psychoacoutic model 1 *
* 3/19/93 Giorgio Dimino modifications to better account for *
* tonal and non-tonal components *
* 5/28/93 Sriram Jayasimha "London" mod. to psychoacoustic model1*
* 8/05/93 Masahiro Iwadare noise_label modification "option" *
* 1/21/94 Seymore Shlien fixed another infinite looping problem*
* 7/12/95 Soeren H. Nielsen Changes for LSF, new tables *
**********************************************************************/
#include "common.h"
#include "encoder.h"
#define LONDON /* enable "LONDON" modification */
#define MAKE_SENSE /* enable "MAKE_SENSE" modification */
#define MI_OPTION /* enable "MI_OPTION" modification */
/**********************************************************************
*
* 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.
*
**********************************************************************/
int crit_band;
int FAR *cbound;
int sub_size;
void read_cbound(lay,freq) /* this function reads in critical */
int lay, freq; /* band boundaries */
{
int i,j,k;
FILE *fp;
char r[16], t[80];
strcpy(r, "2cb1");
r[0] = (char) lay + '0';
r[3] = (char) freq + '0';
if( !(fp = OpenTableFile(r)) ){ /* check boundary values */
printf("Please check %s boundary table\n",r);
exit(1);
}
fgets(t,80,fp); /* read input for critical bands */
sscanf(t,"%d\n",&crit_band);
cbound = (int FAR *) mem_alloc(sizeof(int) * crit_band, "cbound");
for(i=0;i<crit_band;i++){ /* continue to read input for */
fgets(t,80,fp); /* critical band boundaries */
sscanf(t,"%d %d\n",&j, &k);
if(i==j) cbound[j] = k;
else { /* error */
printf("Please check index %d in cbound table %s\n",i,r);
exit(1);
}
}
fclose(fp);
}
void read_freq_band(ltg,lay,freq) /* this function reads in */
int lay, freq; /* frequency bands and bark */
g_ptr FAR *ltg; /* values */
{
int i,j, k;
double b,c;
FILE *fp;
char r[16], t[80];
strcpy(r, "2th1");
r[0] = (char) lay + '0';
r[3] = (char) freq + '0';
if( !(fp = OpenTableFile(r)) ){ /* check freq. values */
printf("Please check frequency and cband table %s\n",r);
exit(1);
}
fgets(t,80,fp); /* read input for freq. subbands */
sscanf(t,"%d\n",&sub_size);
*ltg = (g_ptr FAR ) mem_alloc(sizeof(g_thres) * sub_size, "ltg");
(*ltg)[0].line = 0; /* initialize global masking threshold */
(*ltg)[0].bark = 0;
(*ltg)[0].hear = 0;
for(i=1;i<sub_size;i++){ /* continue to read freq. subband */
fgets(t,80,fp); /* and assign */
sscanf(t,"%d %d %lf %lf\n",&j, &k, &b, &c);
if(i == j){
(*ltg)[j].line = k;
(*ltg)[j].bark = b;
(*ltg)[j].hear = c;
}
else { /* error */
printf("Please check index %d in freq-cb table %s\n",i,r);
exit(1);
}
}
fclose(fp);
}
void make_map(power, ltg) /* this function calculates the */
mask FAR power[HAN_SIZE]; /* global masking threshold */
g_thres FAR *ltg;
{
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;
}
double add_db(a,b)
double a,b;
{
a = pow(10.0,a/10.0);
b = pow(10.0,b/10.0);
return 10 * log10(a+b);
}
/****************************************************************
*
* Fast Fourier transform of the input samples.
*
****************************************************************/
void II_f_f_t(sample, power) /* this function calculates an */
double FAR sample[FFT_SIZE]; /* FFT analysis for the freq. */
mask FAR power[HAN_SIZE]; /* domain */
{
int i,j,k,L,l=0;
int ip, le, le1;
double t_r, t_i, u_r, u_i;
static int M, MM1, init = 0, N;
double *x_r, *x_i, *energy;
static int *rev;
static double *w_r, *w_i;
x_r = (double *) mem_alloc(sizeof(DFFT), "x_r");
x_i = (double *) mem_alloc(sizeof(DFFT), "x_i");
energy = (double *) mem_alloc(sizeof(DFFT), "energy");
for(i=0;i<FFT_SIZE;i++) x_r[i] = x_i[i] = energy[i] = 0;
if(!init){
rev = (int *) mem_alloc(sizeof(IFFT), "rev");
w_r = (double *) mem_alloc(sizeof(D10), "w_r");
w_i = (double *) mem_alloc(sizeof(D10), "w_i");
M = 10;
MM1 = 9;
N = FFT_SIZE;
for(L=0;L<M;L++){
le = 1 << (M-L);
le1 = le >> 1;
w_r[L] = cos(PI/le1);
w_i[L] = -sin(PI/le1);
}
for(i=0;i<FFT_SIZE;rev[i] = l,i++) for(j=0,l=0;j<10;j++){
k=(i>>j) & 1;
l |= (k<<(9-j));
}
init = 1;
}
memcpy( (char *) x_r, (char *) sample, sizeof(double) * FFT_SIZE);
for(L=0;L<MM1;L++){
le = 1 << (M-L);
le1 = le >> 1;
u_r = 1;
u_i = 0;
for(j=0;j<le1;j++){
for(i=j;i<N;i+=le){
ip = i + le1;
t_r = x_r[i] + x_r[ip];
t_i = x_i[i] + x_i[ip];
x_r[ip] = x_r[i] - x_r[ip];
x_i[ip] = x_i[i] - x_i[ip];
x_r[i] = t_r;
x_i[i] = t_i;
t_r = x_r[ip];
x_r[ip] = x_r[ip] * u_r - x_i[ip] * u_i;
x_i[ip] = x_i[ip] * u_r + t_r * u_i;
}
t_r = u_r;
u_r = u_r * w_r[L] - u_i * w_i[L];
u_i = u_i * w_r[L] + t_r * w_i[L];
}
}
for(i=0;i<N;i+=2){
ip = i + 1;
t_r = x_r[i] + x_r[ip];
t_i = x_i[i] + x_i[ip];
x_r[ip] = x_r[i] - x_r[ip];
x_i[ip] = x_i[i] - x_i[ip];
x_r[i] = t_r;
x_i[i] = t_i;
energy[i] = x_r[i] * x_r[i] + x_i[i] * x_i[i];
}
for(i=0;i<FFT_SIZE;i++) if(i<rev[i]){
t_r = energy[i];
energy[i] = energy[rev[i]];
energy[rev[i]] = t_r;
}
for(i=0;i<HAN_SIZE;i++){ /* calculate power density spectrum */
if (energy[i] < 1E-20) energy[i] = 1E-20;
power[i].x = 10 * log10(energy[i]) + POWERNORM;
power[i].next = STOP;
power[i].type = FALSE;
}
mem_free((void **) &x_r);
mem_free((void **) &x_i);
mem_free((void **) &energy);
}
/****************************************************************
*
* Window the incoming audio signal.
*
****************************************************************/
void II_hann_win(sample) /* this function calculates a */
double FAR sample[FFT_SIZE]; /* Hann window for PCM (input) */
{ /* samples for a 1024-pt. FFT */
register int i;
register double sqrt_8_over_3;
static int init = 0;
static double FAR *window;
if(!init){ /* calculate window function for the Fourier transform */
window = (double FAR *) 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++) sample[i] *= window[i];
}
/*******************************************************************
*
* This function finds the maximum spectral component in each
* subband and return them to the encoder for time-domain threshold
* determination.
*
*******************************************************************/
#ifndef LONDON
void II_pick_max(power, spike)
double FAR spike[SBLIMIT];
mask FAR power[HAN_SIZE];
{
double max;
int i,j;
for(i=0;i<HAN_SIZE;spike[i>>4] = max, i+=16) /* calculate the */
for(j=0, max = DBMIN;j<16;j++) /* maximum spectral */
max = (max>power[i+j].x) ? max : power[i+j].x; /* component in each */
} /* subband from bound */
/* 4-16 */
#else
void II_pick_max(power, spike)
double FAR spike[SBLIMIT];
mask FAR power[HAN_SIZE];
{
double sum;
int i,j;
for(i=0;i<HAN_SIZE;spike[i>>4] = 10.0*log10(sum), i+=16)
/* calculate the */
for(j=0, sum = pow(10.0,0.1*DBMIN);j<16;j++) /* sum of spectral */
sum += pow(10.0,0.1*power[i+j].x); /* component in each */
} /* subband from bound */
/* 4-16 */
#endif
/****************************************************************
*
* This function labels the tonal component in the power
* spectrum.
*
****************************************************************/
void II_tonal_label(power, tone) /* this function extracts (tonal) */
mask FAR power[HAN_SIZE]; /* sinusoidals from the spectrum */
int *tone;
{
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;
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