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ffmpeg/libavformat/audio.c

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/*
* Linux audio play and grab interface
* Copyright (c) 2000, 2001 Fabrice Bellard.
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include "avformat.h"
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <sys/soundcard.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include <sys/mman.h>
#include <sys/time.h>
#define AUDIO_BLOCK_SIZE 4096
typedef struct {
int fd;
int sample_rate;
int channels;
int frame_size; /* in bytes ! */
int codec_id;
int flip_left : 1;
UINT8 buffer[AUDIO_BLOCK_SIZE];
int buffer_ptr;
} AudioData;
static int audio_open(AudioData *s, int is_output, const char *audio_device)
{
int audio_fd;
int tmp, err;
char *flip = getenv("AUDIO_FLIP_LEFT");
/* open linux audio device */
if (!audio_device)
audio_device = "/dev/dsp";
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;
}
if (flip && *flip == '1') {
s->flip_left = 1;
}
/* non blocking mode */
if (!is_output)
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;
}
if (tmp)
s->channels = 2;
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 = s1->priv_data;
AVStream *st;
int ret;
st = s1->streams[0];
s->sample_rate = st->codec.sample_rate;
s->channels = st->codec.channels;
ret = audio_open(s, 1, NULL);
if (ret < 0) {
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);
return 0;
}
/* grab support */
static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
{
AudioData *s = s1->priv_data;
AVStream *st;
int ret;
if (!ap || ap->sample_rate <= 0 || ap->channels <= 0)
return -1;
st = av_new_stream(s1, 0);
if (!st) {
return -ENOMEM;
}
s->sample_rate = ap->sample_rate;
s->channels = ap->channels;
ret = audio_open(s, 0, ap->device);
if (ret < 0) {
av_free(st);
return -EIO;
}
/* 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;
av_set_pts_info(s1, 48, 1, 1000000); /* 48 bits pts in us */
return 0;
}
static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
{
AudioData *s = s1->priv_data;
int ret, bdelay;
int64_t cur_time;
struct audio_buf_info abufi;
if (av_new_packet(pkt, s->frame_size) < 0)
return -EIO;
for(;;) {
struct timeval tv;
fd_set fds;
tv.tv_sec = 0;
tv.tv_usec = 30 * 1000; /* 30 msecs -- a bit shorter than 1 frame at 30fps */
FD_ZERO(&fds);
FD_SET(s->fd, &fds);
/* This will block until data is available or we get a timeout */
(void) select(s->fd + 1, &fds, 0, 0, &tv);
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;
/* compute pts of the start of the packet */
cur_time = av_gettime();
bdelay = ret;
if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
bdelay += abufi.bytes;
}
/* substract time represented by the number of bytes in the audio fifo */
cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
/* convert to wanted units */
pkt->pts = cur_time & ((1LL << 48) - 1);
if (s->flip_left && s->channels == 2) {
int i;
short *p = (short *) pkt->data;
for (i = 0; i < ret; i += 4) {
*p = ~*p;
p += 2;
}
}
return 0;
}
static int audio_read_close(AVFormatContext *s1)
{
AudioData *s = s1->priv_data;
audio_close(s);
return 0;
}
static AVInputFormat audio_in_format = {
"audio_device",
"audio grab and output",
sizeof(AudioData),
NULL,
audio_read_header,
audio_read_packet,
audio_read_close,
.flags = AVFMT_NOFILE,
};
static AVOutputFormat audio_out_format = {
"audio_device",
"audio grab and output",
"",
"",
sizeof(AudioData),
/* 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,
.flags = AVFMT_NOFILE,
};
int audio_init(void)
{
av_register_input_format(&audio_in_format);
av_register_output_format(&audio_out_format);
return 0;
}
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