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/*
* Linux audio play 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 <stdint.h>
#if HAVE_SOUNDCARD_H
#include <soundcard.h>
#else
#include <sys/soundcard.h>
#endif
#if HAVE_UNISTD_H
#include <unistd.h>
#endif
#include <fcntl.h>
#include <sys/ioctl.h>
#include "libavutil/internal.h"
#include "libavutil/opt.h"
#include "libavutil/time.h"
#include "libavcodec/avcodec.h"
#include "avdevice.h"
#include "libavformat/internal.h"
#include "oss_audio.h"
static int audio_read_header(AVFormatContext *s1)
{
OSSAudioData *s = s1->priv_data;
AVStream *st;
int ret;
st = avformat_new_stream(s1, NULL);
if (!st) {
return AVERROR(ENOMEM);
}
ret = ff_oss_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;
avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
return 0;
}
static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
{
OSSAudioData *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)
{
OSSAudioData *s = s1->priv_data;
ff_oss_audio_close(s);
return 0;
}
static const AVOption options[] = {
{ "sample_rate", "", offsetof(OSSAudioData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
{ "channels", "", offsetof(OSSAudioData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
{ NULL },
};
static const AVClass oss_demuxer_class = {
.class_name = "OSS demuxer",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
.category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
};
AVInputFormat ff_oss_demuxer = {
.name = "oss",
.long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) capture"),
.priv_data_size = sizeof(OSSAudioData),
.read_header = audio_read_header,
.read_packet = audio_read_packet,
.read_close = audio_read_close,
.flags = AVFMT_NOFILE,
.priv_class = &oss_demuxer_class,
};
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