mirror of
https://github.com/xenia-project/FFmpeg.git
synced 2024-12-12 21:56:29 +00:00
140 lines
3.7 KiB
C
140 lines
3.7 KiB
C
/*
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* Linux audio play and grab interface
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* Copyright (c) 2000, 2001 Fabrice Bellard
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*
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* This file is part of Libav.
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*
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* Libav is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* Libav is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "config.h"
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#include <string.h>
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#if HAVE_SOUNDCARD_H
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#include <soundcard.h>
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#else
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#include <sys/soundcard.h>
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#endif
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#include <unistd.h>
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#include <fcntl.h>
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#include <sys/ioctl.h>
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#include "libavutil/log.h"
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#include "libavcodec/avcodec.h"
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#include "libavformat/avformat.h"
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#include "oss.h"
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int ff_oss_audio_open(AVFormatContext *s1, int is_output,
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const char *audio_device)
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{
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OSSAudioData *s = s1->priv_data;
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int audio_fd;
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int tmp, err;
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char *flip = getenv("AUDIO_FLIP_LEFT");
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char errbuff[128];
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if (is_output)
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audio_fd = avpriv_open(audio_device, O_WRONLY);
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else
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audio_fd = avpriv_open(audio_device, O_RDONLY);
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if (audio_fd < 0) {
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av_log(s1, AV_LOG_ERROR, "%s: %s\n", audio_device, strerror(errno));
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return AVERROR(EIO);
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}
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if (flip && *flip == '1') {
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s->flip_left = 1;
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}
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/* non blocking mode */
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if (!is_output)
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fcntl(audio_fd, F_SETFL, O_NONBLOCK);
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s->frame_size = OSS_AUDIO_BLOCK_SIZE;
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#define CHECK_IOCTL_ERROR(event) \
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if (err < 0) { \
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av_strerror(AVERROR(errno), errbuff, sizeof(errbuff)); \
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av_log(s1, AV_LOG_ERROR, #event ": %s\n", errbuff); \
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goto fail; \
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}
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/* select format : favour native format
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* We don't CHECK_IOCTL_ERROR here because even if failed OSS still may be
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* usable. If OSS is not usable the SNDCTL_DSP_SETFMTS later is going to
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* fail anyway. */
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(void) ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
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#if HAVE_BIGENDIAN
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if (tmp & AFMT_S16_BE) {
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tmp = AFMT_S16_BE;
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} else if (tmp & AFMT_S16_LE) {
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tmp = AFMT_S16_LE;
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} else {
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tmp = 0;
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}
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#else
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if (tmp & AFMT_S16_LE) {
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tmp = AFMT_S16_LE;
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} else if (tmp & AFMT_S16_BE) {
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tmp = AFMT_S16_BE;
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} else {
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tmp = 0;
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}
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#endif
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switch(tmp) {
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case AFMT_S16_LE:
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s->codec_id = AV_CODEC_ID_PCM_S16LE;
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break;
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case AFMT_S16_BE:
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s->codec_id = AV_CODEC_ID_PCM_S16BE;
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break;
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default:
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av_log(s1, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
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close(audio_fd);
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return AVERROR(EIO);
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}
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err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
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CHECK_IOCTL_ERROR(SNDCTL_DSP_SETFMTS)
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tmp = (s->channels == 2);
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err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
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CHECK_IOCTL_ERROR(SNDCTL_DSP_STEREO)
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tmp = s->sample_rate;
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err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
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CHECK_IOCTL_ERROR(SNDCTL_DSP_SPEED)
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s->sample_rate = tmp; /* store real sample rate */
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s->fd = audio_fd;
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return 0;
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fail:
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close(audio_fd);
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return AVERROR(EIO);
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#undef CHECK_IOCTL_ERROR
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}
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int ff_oss_audio_close(OSSAudioData *s)
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{
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close(s->fd);
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return 0;
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}
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