* Linux audio play and grab interface
* Copyright (c) 2000, 2001 Fabrice Bellard
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * 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.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * 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 FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#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
typedef struct {
+ AVClass *class;
int fd;
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);
-#ifdef WORDS_BIGENDIAN
+#if HAVE_BIGENDIAN
if (tmp & AFMT_S16_BE) {
tmp = AFMT_S16_BE;
} else if (tmp & AFMT_S16_LE) {
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");
uint8_t *buf= pkt->data;
while (size > 0) {
- len = AUDIO_BLOCK_SIZE - s->buffer_ptr;
- if (len > size)
- len = size;
+ len = FFMIN(AUDIO_BLOCK_SIZE - s->buffer_ptr, size);
memcpy(s->buffer + s->buffer_ptr, buf, len);
s->buffer_ptr += len;
if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
/* 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 (ap->sample_rate <= 0 || ap->channels <= 0)
- return -1;
-
- st = av_new_stream(s1, 0);
+ st = avformat_new_stream(s1, NULL);
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) {
- av_free(st);
return AVERROR(EIO);
}
/* take real parameters */
- st->codec->codec_type = CODEC_TYPE_AUDIO;
+ 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 */
+ avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
return 0;
}
int64_t cur_time;
struct audio_buf_info abufi;
- if (av_new_packet(pkt, s->frame_size) < 0)
- return AVERROR(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;
- pkt->pts = av_gettime();
- return 0;
- }
- if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) {
- av_free_packet(pkt);
- return AVERROR(EIO);
- }
+ 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;
return 0;
}
-#if CONFIG_OSS_DEMUXER
-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,
+#if CONFIG_OSS_INDEV
+static const AVOption options[] = {
+ { "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 },
+};
+
+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(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_MUXER
-AVOutputFormat oss_muxer = {
- "oss",
- NULL_IF_CONFIG_SMALL("Open Sound System playback"),
- "",
- "",
- sizeof(AudioData),
+#if CONFIG_OSS_OUTDEV
+AVOutputFormat ff_oss_muxer = {
+ .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 */
-#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,
+ .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
projects / ffmpeg / blobdiff