mirror of
https://git.ffmpeg.org/ffmpeg.git
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de6d9b6404
Originally committed as revision 5 to svn://svn.ffmpeg.org/ffmpeg/trunk
180 lines
3.8 KiB
C
180 lines
3.8 KiB
C
/*
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* Linux audio play and grab interface
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* Copyright (c) 2000, 2001 Gerard Lantau.
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program 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
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
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*/
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#include <stdlib.h>
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#include <stdio.h>
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#include <string.h>
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#include <linux/soundcard.h>
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#include <unistd.h>
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#include <fcntl.h>
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#include <sys/ioctl.h>
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#include <sys/mman.h>
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#include <errno.h>
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#include <sys/time.h>
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#include "avformat.h"
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const char *audio_device = "/dev/dsp";
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typedef struct {
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int fd;
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int rate;
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int channels;
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} AudioData;
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#define AUDIO_BLOCK_SIZE 4096
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/* audio read support */
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static int audio_read(URLContext *h, UINT8 *buf, int size)
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{
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AudioData *s = h->priv_data;
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int ret;
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ret = read(s->fd, buf, size);
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if (ret < 0)
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return -errno;
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else
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return ret;
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}
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static int audio_write(URLContext *h, UINT8 *buf, int size)
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{
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AudioData *s = h->priv_data;
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int ret;
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ret = write(s->fd, buf, size);
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if (ret < 0)
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return -errno;
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else
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return ret;
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}
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static int audio_get_format(URLContext *h, URLFormat *f)
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{
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AudioData *s = h->priv_data;
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strcpy(f->format_name, "pcm");
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f->sample_rate = s->rate;
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f->channels = s->channels;
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return 0;
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}
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/* URI syntax: 'audio:[rate[,channels]]'
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default: rate=44100, channels=2
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*/
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static int audio_open(URLContext *h, const char *uri, int flags)
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{
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AudioData *s;
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const char *p;
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int freq, channels, audio_fd;
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int tmp, err;
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h->is_streamed = 1;
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h->packet_size = AUDIO_BLOCK_SIZE;
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s = malloc(sizeof(AudioData));
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if (!s)
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return -ENOMEM;
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h->priv_data = s;
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/* extract parameters */
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p = uri;
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strstart(p, "audio:", &p);
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freq = strtol(p, (char **)&p, 0);
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if (freq <= 0)
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freq = 44100;
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if (*p == ',')
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p++;
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channels = strtol(p, (char **)&p, 0);
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if (channels <= 0)
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channels = 2;
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s->rate = freq;
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s->channels = channels;
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/* open linux audio device */
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if (flags & URL_WRONLY)
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audio_fd = open(audio_device,O_WRONLY);
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else
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audio_fd = open(audio_device,O_RDONLY);
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if (audio_fd < 0) {
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perror(audio_device);
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return -EIO;
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}
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/* non blocking mode */
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fcntl(audio_fd, F_SETFL, O_NONBLOCK);
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#if 0
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tmp=(NB_FRAGMENTS << 16) | FRAGMENT_BITS;
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err=ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp);
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if (err < 0) {
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perror("SNDCTL_DSP_SETFRAGMENT");
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}
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#endif
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tmp=AFMT_S16_LE;
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err=ioctl(audio_fd,SNDCTL_DSP_SETFMT,&tmp);
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if (err < 0) {
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perror("SNDCTL_DSP_SETFMT");
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goto fail;
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}
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tmp= (channels == 2);
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err=ioctl(audio_fd,SNDCTL_DSP_STEREO,&tmp);
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if (err < 0) {
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perror("SNDCTL_DSP_STEREO");
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goto fail;
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}
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tmp = freq;
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err=ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
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if (err < 0) {
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perror("SNDCTL_DSP_SPEED");
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goto fail;
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}
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s->rate = tmp;
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s->fd = audio_fd;
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return 0;
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fail:
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close(audio_fd);
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free(s);
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return -EIO;
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}
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static int audio_close(URLContext *h)
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{
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AudioData *s = h->priv_data;
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close(s->fd);
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free(s);
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return 0;
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}
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URLProtocol audio_protocol = {
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"audio",
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audio_open,
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audio_read,
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audio_write,
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NULL, /* seek */
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audio_close,
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audio_get_format,
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};
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