ffmpeg/libavdevice/oss.c

244 lines
6.7 KiB
C

/*
* Linux audio play and grab interface
* Copyright (c) 2000, 2001 Fabrice Bellard
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "config.h"
#include <string.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include <sys/soundcard.h>
#include "libavutil/log.h"
#include "libavutil/opt.h"
#include "libavutil/time.h"
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
#include "libavformat/internal.h"
#define OSS_AUDIO_BLOCK_SIZE 4096
typedef struct OSSAudioData {
AVClass *class;
int fd;
int sample_rate;
int channels;
int frame_size; /* in bytes ! */
enum AVCodecID codec_id;
unsigned int flip_left : 1;
uint8_t buffer[OSS_AUDIO_BLOCK_SIZE];
int buffer_ptr;
} OSSAudioData;
static int oss_audio_open(AVFormatContext *s1, int is_output,
const char *audio_device)
{
OSSAudioData *s = s1->priv_data;
int audio_fd;
int tmp, err;
char *flip = getenv("AUDIO_FLIP_LEFT");
char errbuff[128];
if (is_output)
audio_fd = avpriv_open(audio_device, O_WRONLY);
else
audio_fd = avpriv_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 = OSS_AUDIO_BLOCK_SIZE;
#define CHECK_IOCTL_ERROR(event) \
if (err < 0) { \
av_strerror(AVERROR(errno), errbuff, sizeof(errbuff)); \
av_log(s1, AV_LOG_ERROR, #event ": %s\n", errbuff); \
goto fail; \
}
/* select format : favour native format
* We don't CHECK_IOCTL_ERROR here because even if failed OSS still may be
* usable. If OSS is not usable the SNDCTL_DSP_SETFMTS later is going to
* fail anyway. */
(void) 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 = AV_CODEC_ID_PCM_S16LE;
break;
case AFMT_S16_BE:
s->codec_id = AV_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);
CHECK_IOCTL_ERROR(SNDCTL_DSP_SETFMTS)
tmp = (s->channels == 2);
err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
CHECK_IOCTL_ERROR(SNDCTL_DSP_STEREO)
tmp = s->sample_rate;
err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
CHECK_IOCTL_ERROR(SNDCTL_DSP_SPEED)
s->sample_rate = tmp; /* store real sample rate */
s->fd = audio_fd;
return 0;
fail:
close(audio_fd);
return AVERROR(EIO);
#undef CHECK_IOCTL_ERROR
}
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 = oss_audio_open(s1, 0, s1->filename);
if (ret < 0) {
return AVERROR(EIO);
}
/* take real parameters */
st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->codec_id = s->codec_id;
st->codecpar->sample_rate = s->sample_rate;
st->codecpar->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_packet_unref(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;
close(s->fd);
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,
};
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,
};