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/*
* Generate a synthetic stereo sound.
* NOTE: No floats are used to guarantee bitexact output.
*
* Copyright (c) 2002 Fabrice Bellard
*
* This file is part of Libav.
*
* Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdlib.h>
#include <stdio.h>
#define MAX_CHANNELS 8
static unsigned int myrnd(unsigned int *seed_ptr, int n)
{
unsigned int seed, val;
seed = *seed_ptr;
seed = (seed * 314159) + 1;
if (n == 256) {
val = seed >> 24;
} else {
val = seed % n;
}
*seed_ptr = seed;
return val;
}
#define FRAC_BITS 16
#define FRAC_ONE (1 << FRAC_BITS)
#define COS_TABLE_BITS 7
/* integer cosinus */
static const unsigned short cos_table[(1 << COS_TABLE_BITS) + 2] = {
0x8000, 0x7ffe, 0x7ff6, 0x7fea, 0x7fd9, 0x7fc2, 0x7fa7, 0x7f87,
0x7f62, 0x7f38, 0x7f0a, 0x7ed6, 0x7e9d, 0x7e60, 0x7e1e, 0x7dd6,
0x7d8a, 0x7d3a, 0x7ce4, 0x7c89, 0x7c2a, 0x7bc6, 0x7b5d, 0x7aef,
0x7a7d, 0x7a06, 0x798a, 0x790a, 0x7885, 0x77fb, 0x776c, 0x76d9,
0x7642, 0x75a6, 0x7505, 0x7460, 0x73b6, 0x7308, 0x7255, 0x719e,
0x70e3, 0x7023, 0x6f5f, 0x6e97, 0x6dca, 0x6cf9, 0x6c24, 0x6b4b,
0x6a6e, 0x698c, 0x68a7, 0x67bd, 0x66d0, 0x65de, 0x64e9, 0x63ef,
0x62f2, 0x61f1, 0x60ec, 0x5fe4, 0x5ed7, 0x5dc8, 0x5cb4, 0x5b9d,
0x5a82, 0x5964, 0x5843, 0x571e, 0x55f6, 0x54ca, 0x539b, 0x5269,
0x5134, 0x4ffb, 0x4ec0, 0x4d81, 0x4c40, 0x4afb, 0x49b4, 0x486a,
0x471d, 0x45cd, 0x447b, 0x4326, 0x41ce, 0x4074, 0x3f17, 0x3db8,
0x3c57, 0x3af3, 0x398d, 0x3825, 0x36ba, 0x354e, 0x33df, 0x326e,
0x30fc, 0x2f87, 0x2e11, 0x2c99, 0x2b1f, 0x29a4, 0x2827, 0x26a8,
0x2528, 0x23a7, 0x2224, 0x209f, 0x1f1a, 0x1d93, 0x1c0c, 0x1a83,
0x18f9, 0x176e, 0x15e2, 0x1455, 0x12c8, 0x113a, 0x0fab, 0x0e1c,
0x0c8c, 0x0afb, 0x096b, 0x07d9, 0x0648, 0x04b6, 0x0324, 0x0192,
0x0000, 0x0000,
};
#define CSHIFT (FRAC_BITS - COS_TABLE_BITS - 2)
static int int_cos(int a)
{
int neg, v, f;
const unsigned short *p;
a = a & (FRAC_ONE - 1); /* modulo 2 * pi */
if (a >= (FRAC_ONE / 2))
a = FRAC_ONE - a;
neg = 0;
if (a > (FRAC_ONE / 4)) {
neg = -1;
a = (FRAC_ONE / 2) - a;
}
p = cos_table + (a >> CSHIFT);
/* linear interpolation */
f = a & ((1 << CSHIFT) - 1);
v = p[0] + (((p[1] - p[0]) * f + (1 << (CSHIFT - 1))) >> CSHIFT);
v = (v ^ neg) - neg;
v = v << (FRAC_BITS - 15);
return v;
}
FILE *outfile;
static void put_sample(int v)
{
fputc(v & 0xff, outfile);
fputc((v >> 8) & 0xff, outfile);
}
int main(int argc, char **argv)
{
int i, a, v, j, f, amp, ampa;
unsigned int seed = 1;
int tabf1[MAX_CHANNELS], tabf2[MAX_CHANNELS];
int taba[MAX_CHANNELS];
int sample_rate = 44100;
int nb_channels = 2;
if (argc < 2 || argc > 4) {
printf("usage: %s file [<sample rate> [<channels>]]\n"
"generate a test raw 16 bit audio stream\n"
"default: 44100 Hz stereo\n", argv[0]);
exit(1);
}
if (argc > 2) {
sample_rate = atoi(argv[2]);
if (sample_rate <= 0) {
fprintf(stderr, "invalid sample rate: %d\n", sample_rate);
return 1;
}
}
if (argc > 3) {
nb_channels = atoi(argv[3]);
if (nb_channels < 1 || nb_channels > MAX_CHANNELS) {
fprintf(stderr, "invalid number of channels: %d\n", nb_channels);
return 1;
}
}
outfile = fopen(argv[1], "wb");
if (!outfile) {
perror(argv[1]);
return 1;
}
/* 1 second of single freq sinus at 1000 Hz */
a = 0;
for (i = 0; i < 1 * sample_rate; i++) {
v = (int_cos(a) * 10000) >> FRAC_BITS;
for (j = 0; j < nb_channels; j++)
put_sample(v);
a += (1000 * FRAC_ONE) / sample_rate;
}
/* 1 second of varing frequency between 100 and 10000 Hz */
a = 0;
for (i = 0; i < 1 * sample_rate; i++) {
v = (int_cos(a) * 10000) >> FRAC_BITS;
for (j = 0; j < nb_channels; j++)
put_sample(v);
f = 100 + (((10000 - 100) * i) / sample_rate);
a += (f * FRAC_ONE) / sample_rate;
}
/* 0.5 second of low amplitude white noise */
for (i = 0; i < sample_rate / 2; i++) {
v = myrnd(&seed, 20000) - 10000;
for (j = 0; j < nb_channels; j++)
put_sample(v);
}
/* 0.5 second of high amplitude white noise */
for (i = 0; i < sample_rate / 2; i++) {
v = myrnd(&seed, 65535) - 32768;
for (j = 0; j < nb_channels; j++)
put_sample(v);
}
/* 1 second of unrelated ramps for each channel */
for (j = 0; j < nb_channels; j++) {
taba[j] = 0;
tabf1[j] = 100 + myrnd(&seed, 5000);
tabf2[j] = 100 + myrnd(&seed, 5000);
}
for (i = 0; i < 1 * sample_rate; i++) {
for (j = 0; j < nb_channels; j++) {
v = (int_cos(taba[j]) * 10000) >> FRAC_BITS;
put_sample(v);
f = tabf1[j] + (((tabf2[j] - tabf1[j]) * i) / sample_rate);
taba[j] += (f * FRAC_ONE) / sample_rate;
}
}
/* 2 seconds of 500 Hz with varying volume */
a = 0;
ampa = 0;
for (i = 0; i < 2 * sample_rate; i++) {
for (j = 0; j < nb_channels; j++) {
amp = ((FRAC_ONE + int_cos(ampa)) * 5000) >> FRAC_BITS;
if (j & 1)
amp = 10000 - amp;
v = (int_cos(a) * amp) >> FRAC_BITS;
put_sample(v);
a += (500 * FRAC_ONE) / sample_rate;
ampa += (2 * FRAC_ONE) / sample_rate;
}
}
fclose(outfile);
return 0;
}
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