/*
- * Generates a synthetic stereo sound
- * NOTE: No floats are used to guarantee a bit exact output.
+ * Generate a synthetic stereo sound.
+ * NOTE: No floats are used to guarantee bitexact output.
*
* Copyright (c) 2002 Fabrice Bellard
*
*/
#include <stdlib.h>
+#include <stdint.h>
#include <stdio.h>
+#include <string.h>
#define MAX_CHANNELS 8
/* 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,
+ 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)
neg = 0;
if (a > (FRAC_ONE / 4)) {
neg = -1;
- a = (FRAC_ONE / 2) - a;
+ a = (FRAC_ONE / 2) - a;
}
p = cos_table + (a >> CSHIFT);
/* linear interpolation */
FILE *outfile;
-static void put_sample(int v)
+static void put16(int16_t v)
{
- fputc(v & 0xff, outfile);
+ 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;
int taba[MAX_CHANNELS];
int sample_rate = 44100;
int nb_channels = 2;
+ char *ext;
- if (argc < 2 || argc > 4) {
- printf("usage: %s file [<sample rate> [<channels>]]\n"
+ 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 > 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++) {
+ 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);
+ 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 */
+ /* 1 second of varying frequency between 100 and 10000 Hz */
a = 0;
- for(i=0;i<1 * sample_rate;i++) {
+ 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);
+ 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++) {
+ for (i = 0; i < sample_rate / 2; i++) {
v = myrnd(&seed, 20000) - 10000;
- for(j=0;j<nb_channels;j++)
- put_sample(v);
+ 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++) {
+ for (i = 0; i < sample_rate / 2; i++) {
v = myrnd(&seed, 65535) - 32768;
- for(j=0;j<nb_channels;j++)
- put_sample(v);
+ 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;
+ 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++) {
+ 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);
+ 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;
+ a = 0;
ampa = 0;
- for(i=0;i<2 * sample_rate;i++) {
- for(j=0;j<nb_channels;j++) {
+ 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;
+ put16(v);
+ a += (500 * FRAC_ONE) / sample_rate;
ampa += (2 * FRAC_ONE) / sample_rate;
}
}
projects / ffmpeg / blobdiff