FFmpeg/libavdevice/oss_audio.c
Stefano Sabatini 72415b2adb Define AVMediaType enum, and use it instead of enum CodecType, which
is deprecated and will be dropped at the next major bump.

Originally committed as revision 22735 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-30 23:30:55 +00:00

330 lines
7.9 KiB
C

/*
* Linux audio play and grab interface
* Copyright (c) 2000, 2001 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg 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 2.1 of the License, or (at your option) any later version.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "config.h"
#include <stdlib.h>
#include <stdio.h>
#include <stdint.h>
#include <string.h>
#include <errno.h>
#if HAVE_SOUNDCARD_H
#include <soundcard.h>
#else
#include <sys/soundcard.h>
#endif
#include <unistd.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include <sys/time.h>
#include <sys/select.h>
#include "libavutil/log.h"
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
#define AUDIO_BLOCK_SIZE 4096
typedef struct {
int fd;
int sample_rate;
int channels;
int frame_size; /* in bytes ! */
enum CodecID codec_id;
unsigned int flip_left : 1;
uint8_t buffer[AUDIO_BLOCK_SIZE];
int buffer_ptr;
} AudioData;
static int audio_open(AVFormatContext *s1, int is_output, const char *audio_device)
{
AudioData *s = s1->priv_data;
int audio_fd;
int tmp, err;
char *flip = getenv("AUDIO_FLIP_LEFT");
if (is_output)
audio_fd = open(audio_device, O_WRONLY);
else
audio_fd = open(audio_device, O_RDONLY);
if (audio_fd < 0) {
av_log(s1, AV_LOG_ERROR, "%s: %s\n", audio_device, strerror(errno));
return AVERROR(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);
#if HAVE_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:
av_log(s1, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
close(audio_fd);
return AVERROR(EIO);
}
err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
if (err < 0) {
av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SETFMT: %s\n", strerror(errno));
goto fail;
}
tmp = (s->channels == 2);
err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
if (err < 0) {
av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_STEREO: %s\n", strerror(errno));
goto fail;
}
tmp = s->sample_rate;
err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
if (err < 0) {
av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SPEED: %s\n", strerror(errno));
goto fail;
}
s->sample_rate = tmp; /* store real sample rate */
s->fd = audio_fd;
return 0;
fail:
close(audio_fd);
return AVERROR(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(s1, 1, s1->filename);
if (ret < 0) {
return AVERROR(EIO);
} else {
return 0;
}
}
static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
{
AudioData *s = s1->priv_data;
int len, ret;
int size= pkt->size;
uint8_t *buf= pkt->data;
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 AVERROR(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->sample_rate <= 0 || ap->channels <= 0)
return -1;
st = av_new_stream(s1, 0);
if (!st) {
return AVERROR(ENOMEM);
}
s->sample_rate = ap->sample_rate;
s->channels = ap->channels;
ret = audio_open(s1, 0, s1->filename);
if (ret < 0) {
return AVERROR(EIO);
}
/* take real parameters */
st->codec->codec_type = AVMEDIA_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(st, 64, 1, 1000000); /* 64 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 ((ret=av_new_packet(pkt, s->frame_size)) < 0)
return ret;
ret = read(s->fd, pkt->data, pkt->size);
if (ret <= 0){
av_free_packet(pkt);
pkt->size = 0;
if (ret<0) return AVERROR(errno);
else return AVERROR_EOF;
}
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;
}
/* subtract 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;
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;
}
#if CONFIG_OSS_INDEV
AVInputFormat oss_demuxer = {
"oss",
NULL_IF_CONFIG_SMALL("Open Sound System capture"),
sizeof(AudioData),
NULL,
audio_read_header,
audio_read_packet,
audio_read_close,
.flags = AVFMT_NOFILE,
};
#endif
#if CONFIG_OSS_OUTDEV
AVOutputFormat oss_muxer = {
"oss",
NULL_IF_CONFIG_SMALL("Open Sound System playback"),
"",
"",
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 */
#if HAVE_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,
};
#endif