* Linux audio play and grab interface
* Copyright (c) 2000, 2001 Fabrice Bellard.
*
- * This library is free software; you can redistribute it and/or
+ * 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 of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
*
- * This library is distributed in the hope that it will be useful,
+ * 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 this library; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avformat.h"
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
+#ifdef HAVE_SOUNDCARD_H
+#include <soundcard.h>
+#else
#include <sys/soundcard.h>
+#endif
#include <unistd.h>
#include <fcntl.h>
#include <sys/ioctl.h>
int tmp, err;
char *flip = getenv("AUDIO_FLIP_LEFT");
- /* open linux audio device */
- if (!audio_device)
- audio_device = "/dev/dsp";
-
if (is_output)
audio_fd = open(audio_device, O_WRONLY);
else
audio_fd = open(audio_device, O_RDONLY);
if (audio_fd < 0) {
perror(audio_device);
- return -EIO;
+ return AVERROR(EIO);
}
if (flip && *flip == '1') {
/* select format : favour native format */
err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
-
+
#ifdef WORDS_BIGENDIAN
if (tmp & AFMT_S16_BE) {
tmp = AFMT_S16_BE;
s->codec_id = CODEC_ID_PCM_S16BE;
break;
default:
- fprintf(stderr, "Soundcard does not support 16 bit sample format\n");
+ av_log(NULL, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
close(audio_fd);
- return -EIO;
+ return AVERROR(EIO);
}
err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
if (err < 0) {
perror("SNDCTL_DSP_SETFMT");
goto fail;
}
-
+
tmp = (s->channels == 2);
err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
if (err < 0) {
}
if (tmp)
s->channels = 2;
-
+
tmp = s->sample_rate;
err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
if (err < 0) {
return 0;
fail:
close(audio_fd);
- return -EIO;
+ return AVERROR(EIO);
}
static int audio_close(AudioData *s)
int ret;
st = s1->streams[0];
- s->sample_rate = st->codec.sample_rate;
- s->channels = st->codec.channels;
- ret = audio_open(s, 1, NULL);
+ s->sample_rate = st->codec->sample_rate;
+ s->channels = st->codec->channels;
+ ret = audio_open(s, 1, s1->filename);
if (ret < 0) {
- return -EIO;
+ return AVERROR(EIO);
} else {
return 0;
}
}
-static int audio_write_packet(AVFormatContext *s1, int stream_index,
- const uint8_t *buf, int size, int64_t pts)
+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 (ret > 0)
break;
if (ret < 0 && (errno != EAGAIN && errno != EINTR))
- return -EIO;
+ return AVERROR(EIO);
}
s->buffer_ptr = 0;
}
AVStream *st;
int ret;
- if (!ap || ap->sample_rate <= 0 || ap->channels <= 0)
+ if (ap->sample_rate <= 0 || ap->channels <= 0)
return -1;
st = av_new_stream(s1, 0);
if (!st) {
- return -ENOMEM;
+ return AVERROR(ENOMEM);
}
s->sample_rate = ap->sample_rate;
s->channels = ap->channels;
- ret = audio_open(s, 0, ap->device);
+ ret = audio_open(s, 0, s1->filename);
if (ret < 0) {
av_free(st);
- return -EIO;
+ return AVERROR(EIO);
}
/* take real parameters */
- st->codec.codec_type = CODEC_TYPE_AUDIO;
- st->codec.codec_id = s->codec_id;
- st->codec.sample_rate = s->sample_rate;
- st->codec.channels = s->channels;
+ st->codec->codec_type = CODEC_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(s1, 48, 1, 1000000); /* 48 bits pts in us */
+ av_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
return 0;
}
int ret, bdelay;
int64_t cur_time;
struct audio_buf_info abufi;
-
+
if (av_new_packet(pkt, s->frame_size) < 0)
- return -EIO;
+ return AVERROR(EIO);
for(;;) {
struct timeval tv;
fd_set fds;
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 -EIO;
+ return AVERROR(EIO);
}
}
pkt->size = ret;
cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
/* convert to wanted units */
- pkt->pts = cur_time & ((1LL << 48) - 1);
+ pkt->pts = cur_time;
if (s->flip_left && s->channels == 2) {
int i;
return 0;
}
-static AVInputFormat audio_in_format = {
- "audio_device",
+#ifdef CONFIG_OSS_DEMUXER
+AVInputFormat oss_demuxer = {
+ "oss",
"audio grab and output",
sizeof(AudioData),
NULL,
audio_read_close,
.flags = AVFMT_NOFILE,
};
+#endif
-static AVOutputFormat audio_out_format = {
- "audio_device",
+#ifdef CONFIG_OSS_MUXER
+AVOutputFormat oss_muxer = {
+ "oss",
"audio grab and output",
"",
"",
audio_write_trailer,
.flags = AVFMT_NOFILE,
};
-
-int audio_init(void)
-{
- av_register_input_format(&audio_in_format);
- av_register_output_format(&audio_out_format);
- return 0;
-}
+#endif
projects / ffmpeg / blobdiff