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 "avformat.h"
#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 <sys/time.h>
const char *audio_device = "/dev/dsp";
#define AUDIO_BLOCK_SIZE 4096
typedef struct {
int fd;
int sample_rate;
int channels;
int frame_size; /* in bytes ! */
int codec_id;
UINT8 buffer[AUDIO_BLOCK_SIZE];
int buffer_ptr;
} AudioData;
static int audio_open(AudioData *s, int is_output)
{
int audio_fd;
int tmp, err;
/* open linux audio device */
if (is_output)
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);
s->frame_size = AUDIO_BLOCK_SIZE;
#if 0
tmp = (NB_FRAGMENTS << 16) | FRAGMENT_BITS;
err = ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp);
if (err < 0) {
perror("SNDCTL_DSP_SETFRAGMENT");
}
#endif
/* select format : favour native format */
err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
#ifdef WORDS_BIGENDIAN
if (tmp & AFMT_S16_BE) {
tmp = AFMT_S16_BE;
} else if (tmp & AFMT_S16_LE) {
tmp = AFMT_S16_LE;
} else {
tmp = 0;
}
#else
if (tmp & AFMT_S16_LE) {
tmp = AFMT_S16_LE;
} else if (tmp & AFMT_S16_BE) {
tmp = AFMT_S16_BE;
} else {
tmp = 0;
}
#endif
switch(tmp) {
case AFMT_S16_LE:
s->codec_id = CODEC_ID_PCM_S16LE;
break;
case AFMT_S16_BE:
s->codec_id = CODEC_ID_PCM_S16BE;
break;
default:
fprintf(stderr, "Soundcard does not support 16 bit sample format\n");
close(audio_fd);
return -EIO;
}
err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
if (err < 0) {
perror("SNDCTL_DSP_SETFMT");
goto fail;
}
tmp = (s->channels == 2);
err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
if (err < 0) {
perror("SNDCTL_DSP_STEREO");
goto fail;
}
tmp = s->sample_rate;
err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
if (err < 0) {
perror("SNDCTL_DSP_SPEED");
goto fail;
}
s->sample_rate = tmp; /* store real sample rate */
s->fd = audio_fd;
return 0;
fail:
close(audio_fd);
return -EIO;
}
static int audio_close(AudioData *s)
{
close(s->fd);
return 0;
}
/* sound output support */
static int audio_write_header(AVFormatContext *s1)
{
AudioData *s;
AVStream *st;
int ret;
s = av_mallocz(sizeof(AudioData));
if (!s)
return -ENOMEM;
s1->priv_data = s;
st = s1->streams[0];
s->sample_rate = st->codec.sample_rate;
s->channels = st->codec.channels;
ret = audio_open(s, 1);
if (ret < 0) {
free(s);
return -EIO;
} else {
return 0;
}
}
static int audio_write_packet(AVFormatContext *s1, int stream_index,
UINT8 *buf, int size, int force_pts)
{
AudioData *s = s1->priv_data;
int len, ret;
while (size > 0) {
len = AUDIO_BLOCK_SIZE - s->buffer_ptr;
if (len > size)
len = size;
memcpy(s->buffer + s->buffer_ptr, buf, len);
s->buffer_ptr += len;
if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
for(;;) {
ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
if (ret != 0)
break;
if (ret < 0 && (errno != EAGAIN && errno != EINTR))
return -EIO;
}
s->buffer_ptr = 0;
}
buf += len;
size -= len;
}
return 0;
}
static int audio_write_trailer(AVFormatContext *s1)
{
AudioData *s = s1->priv_data;
audio_close(s);
free(s);
return 0;
}
/* grab support */
static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
{
AudioData *s;
AVStream *st;
int ret;
if (!ap || ap->sample_rate <= 0 || ap->channels <= 0)
return -1;
s = av_mallocz(sizeof(AudioData));
if (!s)
return -ENOMEM;
st = av_mallocz(sizeof(AVStream));
if (!st) {
free(s);
return -ENOMEM;
}
s1->priv_data = s;
s1->nb_streams = 1;
s1->streams[0] = st;
s->sample_rate = ap->sample_rate;
s->channels = ap->channels;
ret = audio_open(s, 0);
if (ret < 0) {
free(st);
free(s);
return -EIO;
} else {
/* take real parameters */
st->codec.codec_type = CODEC_TYPE_AUDIO;
st->codec.codec_id = s->codec_id;
st->codec.sample_rate = s->sample_rate;
st->codec.channels = s->channels;
return 0;
}
}
static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
{
AudioData *s = s1->priv_data;
int ret;
if (av_new_packet(pkt, s->frame_size) < 0)
return -EIO;
for(;;) {
ret = read(s->fd, pkt->data, pkt->size);
if (ret > 0)
break;
if (ret == -1 && (errno == EAGAIN || errno == EINTR)) {
av_free_packet(pkt);
pkt->size = 0;
return 0;
}
if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) {
av_free_packet(pkt);
return -EIO;
}
}
pkt->size = ret;
return 0;
}
static int audio_read_close(AVFormatContext *s1)
{
AudioData *s = s1->priv_data;
audio_close(s);
free(s);
return 0;
}
AVFormat audio_device_format = {
"audio_device",
"audio grab and output",
"",
"",
/* XXX: we make the assumption that the soundcard accepts this format */
/* XXX: find better solution with "preinit" method, needed also in
other formats */
#ifdef WORDS_BIGENDIAN
CODEC_ID_PCM_S16BE,
#else
CODEC_ID_PCM_S16LE,
#endif
CODEC_ID_NONE,
audio_write_header,
audio_write_packet,
audio_write_trailer,
audio_read_header,
audio_read_packet,
audio_read_close,
NULL,
AVFMT_NOFILE,
};