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FFmpeg/libavdevice/oss.c
Matt Jacobson b3e261bab3 avdevice/oss_dec: account for sample size when computing timestamp
Don't assume each sample is one byte in size. Doing so results in wrong and
occasionally non-monotonically-increasing timestamps.

Fix nearby cosmetic typo.

Signed-off-by: Marton Balint <cus@passwd.hu>
2022-06-19 23:01:20 +02:00

141 lines
3.9 KiB
C

/*
* Linux audio play and grab 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 <string.h>
#if HAVE_UNISTD_H
#include <unistd.h>
#endif
#include <fcntl.h>
#include <sys/ioctl.h>
#include <sys/soundcard.h>
#include "libavutil/log.h"
#include "avdevice.h"
#include "oss.h"
int ff_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");
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, av_err2str(AVERROR(errno)));
return AVERROR(EIO);
}
if (flip && *flip == '1') {
s->flip_left = 1;
}
/* non blocking mode */
if (!is_output) {
if (fcntl(audio_fd, F_SETFL, O_NONBLOCK) < 0) {
av_log(s1, AV_LOG_WARNING, "%s: Could not enable non block mode (%s)\n", audio_device, av_err2str(AVERROR(errno)));
}
}
s->frame_size = OSS_AUDIO_BLOCK_SIZE;
#define CHECK_IOCTL_ERROR(event) \
if (err < 0) { \
av_log(s1, AV_LOG_ERROR, #event ": %s\n", av_err2str(AVERROR(errno)));\
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. */
err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
if (err < 0) {
av_log(s1, AV_LOG_WARNING, "SNDCTL_DSP_GETFMTS: %s\n", av_err2str(AVERROR(errno)));
}
#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;
s->sample_size = 2;
break;
case AFMT_S16_BE:
s->codec_id = AV_CODEC_ID_PCM_S16BE;
s->sample_size = 2;
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_SETFMT)
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
}
int ff_oss_audio_close(OSSAudioData *s)
{
close(s->fd);
return 0;
}