RepoMirrors
/
ffmpeg
mirror of https://git.ffmpeg.org/ffmpeg.git
1
0
Fork 0
Code Issues Projects Releases Wiki Activity

FATE: optionally write a WAVE header in audiogen

This commit is contained in:
Justin Ruggles 2012-04-17 10:12:38 -04:00
parent 0c0d1bce7c
commit 010943c6ce
1 changed files with 48 additions and 8 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

56
tests/audiogen.c
View File

@ -22,7 +22,9 @@
*/
#include <stdlib.h>
#include <stdint.h>
#include <stdio.h>
#include <string.h>
#define MAX_CHANNELS 8
@ -93,12 +95,45 @@ static int int_cos(int a)
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;
@ -107,10 +142,12 @@ int main(int argc, char **argv)
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"
"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);
}
@ -137,12 +174,15 @@ int main(int argc, char **argv)
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++)
put_sample(v);
put16(v);
a += (1000 * FRAC_ONE) / sample_rate;
}
@ -151,7 +191,7 @@ int main(int argc, char **argv)
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);
put16(v);
f = 100 + (((10000 - 100) * i) / sample_rate);
a += (f * FRAC_ONE) / sample_rate;
}
@ -160,14 +200,14 @@ int main(int argc, char **argv)
for (i = 0; i < sample_rate / 2; i++) {
v = myrnd(&seed, 20000) - 10000;
for (j = 0; j < nb_channels; j++)
put_sample(v);
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++)
put_sample(v);
put16(v);
}
/* 1 second of unrelated ramps for each channel */
@ -179,7 +219,7 @@ int main(int argc, char **argv)
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);
put16(v);
f = tabf1[j] + (((tabf2[j] - tabf1[j]) * i) / sample_rate);
taba[j] += (f * FRAC_ONE) / sample_rate;
}
@ -194,7 +234,7 @@ int main(int argc, char **argv)
if (j & 1)
amp = 10000 - amp;
v = (int_cos(a) * amp) >> FRAC_BITS;
put_sample(v);
put16(v);
a += (500 * FRAC_ONE) / sample_rate;
ampa += (2 * FRAC_ONE) / sample_rate;
}