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- /*
- * Linux audio play 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 <stdint.h>
- #if HAVE_SOUNDCARD_H
- #include <soundcard.h>
- #else
- #include <sys/soundcard.h>
- #endif
- #if HAVE_UNISTD_H
- #include <unistd.h>
- #endif
- #include <fcntl.h>
- #include <sys/ioctl.h>
- #include "libavutil/internal.h"
- #include "libavutil/opt.h"
- #include "libavutil/time.h"
- #include "libavcodec/avcodec.h"
- #include "avdevice.h"
- #include "libavformat/internal.h"
- #include "oss.h"
- static int audio_read_header(AVFormatContext *s1)
- {
- OSSAudioData *s = s1->priv_data;
- AVStream *st;
- int ret;
- st = avformat_new_stream(s1, NULL);
- if (!st) {
- return AVERROR(ENOMEM);
- }
- ret = ff_oss_audio_open(s1, 0, s1->filename);
- if (ret < 0) {
- return AVERROR(EIO);
- }
- /* take real parameters */
- st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
- st->codecpar->codec_id = s->codec_id;
- st->codecpar->sample_rate = s->sample_rate;
- st->codecpar->channels = s->channels;
- avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
- return 0;
- }
- static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
- {
- OSSAudioData *s = s1->priv_data;
- int ret, bdelay;
- int64_t cur_time;
- struct audio_buf_info abufi;
- if ((ret=av_new_packet(pkt, s->frame_size)) < 0)
- return ret;
- ret = read(s->fd, pkt->data, pkt->size);
- if (ret <= 0){
- av_packet_unref(pkt);
- pkt->size = 0;
- if (ret<0) return AVERROR(errno);
- else return AVERROR_EOF;
- }
- pkt->size = ret;
- /* compute pts of the start of the packet */
- cur_time = av_gettime();
- bdelay = ret;
- if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
- bdelay += abufi.bytes;
- }
- /* subtract time represented by the number of bytes in the audio fifo */
- cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
- /* convert to wanted units */
- pkt->pts = cur_time;
- if (s->flip_left && s->channels == 2) {
- int i;
- short *p = (short *) pkt->data;
- for (i = 0; i < ret; i += 4) {
- *p = ~*p;
- p += 2;
- }
- }
- return 0;
- }
- static int audio_read_close(AVFormatContext *s1)
- {
- OSSAudioData *s = s1->priv_data;
- ff_oss_audio_close(s);
- return 0;
- }
- static const AVOption options[] = {
- { "sample_rate", "", offsetof(OSSAudioData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
- { "channels", "", offsetof(OSSAudioData, 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,
- .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
- };
- AVInputFormat ff_oss_demuxer = {
- .name = "oss",
- .long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) capture"),
- .priv_data_size = sizeof(OSSAudioData),
- .read_header = audio_read_header,
- .read_packet = audio_read_packet,
- .read_close = audio_read_close,
- .flags = AVFMT_NOFILE,
- .priv_class = &oss_demuxer_class,
- };
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