ffmpeg/libav/audio.c

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/*
* Linux audio play and grab interface
* Copyright (c) 2000, 2001 Gerard Lantau.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <linux/soundcard.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include <sys/mman.h>
#include <errno.h>
#include <sys/time.h>
#include "avformat.h"
const char *audio_device = "/dev/dsp";
typedef struct {
int fd;
int rate;
int channels;
} AudioData;
#define AUDIO_BLOCK_SIZE 4096
/* audio read support */
static int audio_read(URLContext *h, UINT8 *buf, int size)
{
AudioData *s = h->priv_data;
int ret;
ret = read(s->fd, buf, size);
if (ret < 0)
return -errno;
else
return ret;
}
static int audio_write(URLContext *h, UINT8 *buf, int size)
{
AudioData *s = h->priv_data;
int ret;
ret = write(s->fd, buf, size);
if (ret < 0)
return -errno;
else
return ret;
}
static int audio_get_format(URLContext *h, URLFormat *f)
{
AudioData *s = h->priv_data;
strcpy(f->format_name, "pcm");
f->sample_rate = s->rate;
f->channels = s->channels;
return 0;
}
/* URI syntax: 'audio:[rate[,channels]]'
default: rate=44100, channels=2
*/
static int audio_open(URLContext *h, const char *uri, int flags)
{
AudioData *s;
const char *p;
int freq, channels, audio_fd;
int tmp, err;
h->is_streamed = 1;
h->packet_size = AUDIO_BLOCK_SIZE;
s = malloc(sizeof(AudioData));
if (!s)
return -ENOMEM;
h->priv_data = s;
/* extract parameters */
p = uri;
strstart(p, "audio:", &p);
freq = strtol(p, (char **)&p, 0);
if (freq <= 0)
freq = 44100;
if (*p == ',')
p++;
channels = strtol(p, (char **)&p, 0);
if (channels <= 0)
channels = 2;
s->rate = freq;
s->channels = channels;
/* open linux audio device */
if (flags & URL_WRONLY)
audio_fd = open(audio_device,O_WRONLY);
else
audio_fd = open(audio_device,O_RDONLY);
if (audio_fd < 0) {
perror(audio_device);
return -EIO;
}
/* non blocking mode */
fcntl(audio_fd, F_SETFL, O_NONBLOCK);
#if 0
tmp=(NB_FRAGMENTS << 16) | FRAGMENT_BITS;
err=ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp);
if (err < 0) {
perror("SNDCTL_DSP_SETFRAGMENT");
}
#endif
tmp=AFMT_S16_LE;
err=ioctl(audio_fd,SNDCTL_DSP_SETFMT,&tmp);
if (err < 0) {
perror("SNDCTL_DSP_SETFMT");
goto fail;
}
tmp= (channels == 2);
err=ioctl(audio_fd,SNDCTL_DSP_STEREO,&tmp);
if (err < 0) {
perror("SNDCTL_DSP_STEREO");
goto fail;
}
tmp = freq;
err=ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
if (err < 0) {
perror("SNDCTL_DSP_SPEED");
goto fail;
}
s->rate = tmp;
s->fd = audio_fd;
return 0;
fail:
close(audio_fd);
free(s);
return -EIO;
}
static int audio_close(URLContext *h)
{
AudioData *s = h->priv_data;
close(s->fd);
free(s);
return 0;
}
URLProtocol audio_protocol = {
"audio",
audio_open,
audio_read,
audio_write,
NULL, /* seek */
audio_close,
audio_get_format,
};