#include <unistd.h>
#include <fcntl.h>
#include <sys/ioctl.h>
-#include <sys/time.h>
-#include <sys/select.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 AUDIO_BLOCK_SIZE 4096
int sample_rate;
int channels;
int frame_size; /* in bytes ! */
- enum CodecID codec_id;
+ enum AVCodecID codec_id;
unsigned int flip_left : 1;
uint8_t buffer[AUDIO_BLOCK_SIZE];
int buffer_ptr;
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);
switch(tmp) {
case AFMT_S16_LE:
- s->codec_id = CODEC_ID_PCM_S16LE;
+ s->codec_id = AV_CODEC_ID_PCM_S16LE;
break;
case AFMT_S16_BE:
- s->codec_id = CODEC_ID_PCM_S16BE;
+ 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");
/* grab support */
-static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
+static int audio_read_header(AVFormatContext *s1)
{
AudioData *s = s1->priv_data;
AVStream *st;
int ret;
-#if FF_API_FORMAT_PARAMETERS
- if (ap->sample_rate > 0)
- s->sample_rate = ap->sample_rate;
- if (ap->channels > 0)
- s->channels = ap->channels;
-#endif
-
- st = av_new_stream(s1, 0);
+ st = avformat_new_stream(s1, NULL);
if (!st) {
return AVERROR(ENOMEM);
}
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 */
+ avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
return 0;
}
#if CONFIG_OSS_INDEV
static const AVOption options[] = {
- { "sample_rate", "", offsetof(AudioData, sample_rate), FF_OPT_TYPE_INT, {.dbl = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
- { "channels", "", offsetof(AudioData, channels), FF_OPT_TYPE_INT, {.dbl = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
+ { "sample_rate", "", offsetof(AudioData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
+ { "channels", "", offsetof(AudioData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
{ NULL },
};
};
AVInputFormat ff_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,
- .priv_class = &oss_demuxer_class,
+ .name = "oss",
+ .long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) capture"),
+ .priv_data_size = sizeof(AudioData),
+ .read_header = audio_read_header,
+ .read_packet = audio_read_packet,
+ .read_close = audio_read_close,
+ .flags = AVFMT_NOFILE,
+ .priv_class = &oss_demuxer_class,
};
#endif
#if CONFIG_OSS_OUTDEV
AVOutputFormat ff_oss_muxer = {
- "oss",
- NULL_IF_CONFIG_SMALL("Open Sound System playback"),
- "",
- "",
- sizeof(AudioData),
+ .name = "oss",
+ .long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) playback"),
+ .priv_data_size = 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 */
- AV_NE(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE),
- CODEC_ID_NONE,
- audio_write_header,
- audio_write_packet,
- audio_write_trailer,
- .flags = AVFMT_NOFILE,
+ .audio_codec = AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE),
+ .video_codec = AV_CODEC_ID_NONE,
+ .write_header = audio_write_header,
+ .write_packet = audio_write_packet,
+ .write_trailer = audio_write_trailer,
+ .flags = AVFMT_NOFILE,
};
#endif