/* * 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 <stdint.h> #include <stdio.h> #include <string.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 put16(int16_t v) { fputc( v & 0xff, outfile); fputc((v >> 8) & 0xff, outfile); } static void put32(uint32_t v) { fputc( v & 0xff, outfile); fputc((v >> 8) & 0xff, outfile); fputc((v >> 16) & 0xff, outfile); fputc((v >> 24) & 0xff, outfile); } #define HEADER_SIZE 46 #define FMT_SIZE 18 #define SAMPLE_SIZE 2 #define WFORMAT_PCM 0x0001 static void put_wav_header(int sample_rate, int channels, int nb_samples) { int block_align = SAMPLE_SIZE * channels; int data_size = block_align * nb_samples; fputs("RIFF", outfile); put32(HEADER_SIZE + data_size); fputs("WAVEfmt ", outfile); put32(FMT_SIZE); put16(WFORMAT_PCM); put16(channels); put32(sample_rate); put32(block_align * sample_rate); put16(block_align); put16(SAMPLE_SIZE * 8); put16(0); fputs("data", outfile); put32(data_size); } 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; char *ext; if (argc < 2 || argc > 5) { printf("usage: %s file [<sample rate> [<channels>] [<random seed>]]\n" "generate a test raw 16 bit audio stream\n" "If the file extension is .wav a WAVE header will be added.\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; } } if (argc > 4) seed = atoi(argv[4]); outfile = fopen(argv[1], "wb"); if (!outfile) { perror(argv[1]); return 1; } if ((ext = strrchr(argv[1], '.')) != NULL && !strcmp(ext, ".wav")) put_wav_header(sample_rate, nb_channels, 6 * sample_rate); /* 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++) put16(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++) put16(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++) put16(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++) put16(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; put16(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; put16(v); a += (500 * FRAC_ONE) / sample_rate; ampa += (2 * FRAC_ONE) / sample_rate; } } fclose(outfile); return 0; }