forked from KolibriOS/kolibrios
249 lines
7.3 KiB
C
249 lines
7.3 KiB
C
/*
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* Generate a synthetic stereo sound.
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* NOTE: No floats are used to guarantee bitexact output.
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*
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* Copyright (c) 2002 Fabrice Bellard
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <stdlib.h>
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#include <stdint.h>
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#include <stdio.h>
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#include <string.h>
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#define MAX_CHANNELS 8
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static unsigned int myrnd(unsigned int *seed_ptr, int n)
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{
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unsigned int seed, val;
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seed = *seed_ptr;
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seed = (seed * 314159) + 1;
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if (n == 256) {
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val = seed >> 24;
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} else {
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val = seed % n;
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}
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*seed_ptr = seed;
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return val;
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}
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#define FRAC_BITS 16
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#define FRAC_ONE (1 << FRAC_BITS)
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#define COS_TABLE_BITS 7
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/* integer cosine */
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static const unsigned short cos_table[(1 << COS_TABLE_BITS) + 2] = {
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0x8000, 0x7ffe, 0x7ff6, 0x7fea, 0x7fd9, 0x7fc2, 0x7fa7, 0x7f87,
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0x7f62, 0x7f38, 0x7f0a, 0x7ed6, 0x7e9d, 0x7e60, 0x7e1e, 0x7dd6,
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0x7d8a, 0x7d3a, 0x7ce4, 0x7c89, 0x7c2a, 0x7bc6, 0x7b5d, 0x7aef,
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0x7a7d, 0x7a06, 0x798a, 0x790a, 0x7885, 0x77fb, 0x776c, 0x76d9,
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0x7642, 0x75a6, 0x7505, 0x7460, 0x73b6, 0x7308, 0x7255, 0x719e,
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0x70e3, 0x7023, 0x6f5f, 0x6e97, 0x6dca, 0x6cf9, 0x6c24, 0x6b4b,
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0x6a6e, 0x698c, 0x68a7, 0x67bd, 0x66d0, 0x65de, 0x64e9, 0x63ef,
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0x62f2, 0x61f1, 0x60ec, 0x5fe4, 0x5ed7, 0x5dc8, 0x5cb4, 0x5b9d,
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0x5a82, 0x5964, 0x5843, 0x571e, 0x55f6, 0x54ca, 0x539b, 0x5269,
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0x5134, 0x4ffb, 0x4ec0, 0x4d81, 0x4c40, 0x4afb, 0x49b4, 0x486a,
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0x471d, 0x45cd, 0x447b, 0x4326, 0x41ce, 0x4074, 0x3f17, 0x3db8,
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0x3c57, 0x3af3, 0x398d, 0x3825, 0x36ba, 0x354e, 0x33df, 0x326e,
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0x30fc, 0x2f87, 0x2e11, 0x2c99, 0x2b1f, 0x29a4, 0x2827, 0x26a8,
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0x2528, 0x23a7, 0x2224, 0x209f, 0x1f1a, 0x1d93, 0x1c0c, 0x1a83,
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0x18f9, 0x176e, 0x15e2, 0x1455, 0x12c8, 0x113a, 0x0fab, 0x0e1c,
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0x0c8c, 0x0afb, 0x096b, 0x07d9, 0x0648, 0x04b6, 0x0324, 0x0192,
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0x0000, 0x0000,
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};
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#define CSHIFT (FRAC_BITS - COS_TABLE_BITS - 2)
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static int int_cos(int a)
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{
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int neg, v, f;
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const unsigned short *p;
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a = a & (FRAC_ONE - 1); /* modulo 2 * pi */
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if (a >= (FRAC_ONE / 2))
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a = FRAC_ONE - a;
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neg = 0;
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if (a > (FRAC_ONE / 4)) {
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neg = -1;
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a = (FRAC_ONE / 2) - a;
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}
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p = cos_table + (a >> CSHIFT);
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/* linear interpolation */
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f = a & ((1 << CSHIFT) - 1);
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v = p[0] + (((p[1] - p[0]) * f + (1 << (CSHIFT - 1))) >> CSHIFT);
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v = (v ^ neg) - neg;
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v = v << (FRAC_BITS - 15);
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return v;
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}
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FILE *outfile;
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static void put16(int16_t v)
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{
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fputc( v & 0xff, outfile);
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fputc((v >> 8) & 0xff, outfile);
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}
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static void put32(uint32_t v)
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{
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fputc( v & 0xff, outfile);
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fputc((v >> 8) & 0xff, outfile);
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fputc((v >> 16) & 0xff, outfile);
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fputc((v >> 24) & 0xff, outfile);
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}
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#define HEADER_SIZE 46
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#define FMT_SIZE 18
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#define SAMPLE_SIZE 2
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#define WFORMAT_PCM 0x0001
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static void put_wav_header(int sample_rate, int channels, int nb_samples)
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{
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int block_align = SAMPLE_SIZE * channels;
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int data_size = block_align * nb_samples;
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fputs("RIFF", outfile);
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put32(HEADER_SIZE + data_size);
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fputs("WAVEfmt ", outfile);
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put32(FMT_SIZE);
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put16(WFORMAT_PCM);
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put16(channels);
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put32(sample_rate);
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put32(block_align * sample_rate);
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put16(block_align);
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put16(SAMPLE_SIZE * 8);
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put16(0);
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fputs("data", outfile);
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put32(data_size);
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}
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int main(int argc, char **argv)
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{
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int i, a, v, j, f, amp, ampa;
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unsigned int seed = 1;
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int tabf1[MAX_CHANNELS], tabf2[MAX_CHANNELS];
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int taba[MAX_CHANNELS];
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int sample_rate = 44100;
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int nb_channels = 2;
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char *ext;
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if (argc < 2 || argc > 5) {
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printf("usage: %s file [<sample rate> [<channels>] [<random seed>]]\n"
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"generate a test raw 16 bit audio stream\n"
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"If the file extension is .wav a WAVE header will be added.\n"
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"default: 44100 Hz stereo\n", argv[0]);
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exit(1);
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}
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if (argc > 2) {
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sample_rate = atoi(argv[2]);
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if (sample_rate <= 0) {
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fprintf(stderr, "invalid sample rate: %d\n", sample_rate);
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return 1;
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}
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}
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if (argc > 3) {
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nb_channels = atoi(argv[3]);
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if (nb_channels < 1 || nb_channels > MAX_CHANNELS) {
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fprintf(stderr, "invalid number of channels: %d\n", nb_channels);
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return 1;
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}
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}
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if (argc > 4)
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seed = atoi(argv[4]);
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outfile = fopen(argv[1], "wb");
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if (!outfile) {
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perror(argv[1]);
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return 1;
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}
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if ((ext = strrchr(argv[1], '.')) != NULL && !strcmp(ext, ".wav"))
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put_wav_header(sample_rate, nb_channels, 6 * sample_rate);
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/* 1 second of single freq sine at 1000 Hz */
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a = 0;
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for (i = 0; i < 1 * sample_rate; i++) {
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v = (int_cos(a) * 10000) >> FRAC_BITS;
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for (j = 0; j < nb_channels; j++)
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put16(v);
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a += (1000 * FRAC_ONE) / sample_rate;
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}
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/* 1 second of varying frequency between 100 and 10000 Hz */
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a = 0;
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for (i = 0; i < 1 * sample_rate; i++) {
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v = (int_cos(a) * 10000) >> FRAC_BITS;
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for (j = 0; j < nb_channels; j++)
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put16(v);
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f = 100 + (((10000 - 100) * i) / sample_rate);
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a += (f * FRAC_ONE) / sample_rate;
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}
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/* 0.5 second of low amplitude white noise */
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for (i = 0; i < sample_rate / 2; i++) {
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v = myrnd(&seed, 20000) - 10000;
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for (j = 0; j < nb_channels; j++)
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put16(v);
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}
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/* 0.5 second of high amplitude white noise */
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for (i = 0; i < sample_rate / 2; i++) {
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v = myrnd(&seed, 65535) - 32768;
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for (j = 0; j < nb_channels; j++)
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put16(v);
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}
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/* 1 second of unrelated ramps for each channel */
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for (j = 0; j < nb_channels; j++) {
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taba[j] = 0;
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tabf1[j] = 100 + myrnd(&seed, 5000);
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tabf2[j] = 100 + myrnd(&seed, 5000);
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}
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for (i = 0; i < 1 * sample_rate; i++) {
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for (j = 0; j < nb_channels; j++) {
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v = (int_cos(taba[j]) * 10000) >> FRAC_BITS;
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put16(v);
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f = tabf1[j] + (((tabf2[j] - tabf1[j]) * i) / sample_rate);
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taba[j] += (f * FRAC_ONE) / sample_rate;
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}
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}
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/* 2 seconds of 500 Hz with varying volume */
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a = 0;
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ampa = 0;
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for (i = 0; i < 2 * sample_rate; i++) {
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for (j = 0; j < nb_channels; j++) {
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amp = ((FRAC_ONE + int_cos(ampa)) * 5000) >> FRAC_BITS;
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if (j & 1)
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amp = 10000 - amp;
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v = (int_cos(a) * amp) >> FRAC_BITS;
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put16(v);
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a += (500 * FRAC_ONE) / sample_rate;
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ampa += (2 * FRAC_ONE) / sample_rate;
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}
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}
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fclose(outfile);
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return 0;
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}
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