mirror of
https://git.ffmpeg.org/ffmpeg.git
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143 lines
3.9 KiB
C
143 lines
3.9 KiB
C
/*
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* Linux audio play interface
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* Copyright (c) 2000, 2001 Fabrice Bellard
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg 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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* FFmpeg 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 FFmpeg; 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 <stdint.h>
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#if HAVE_UNISTD_H
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#include <unistd.h>
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#endif
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#include <fcntl.h>
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#include <sys/ioctl.h>
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#include <sys/soundcard.h>
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#include "libavutil/internal.h"
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#include "libavutil/opt.h"
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#include "libavutil/time.h"
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#include "avdevice.h"
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#include "libavformat/internal.h"
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#include "oss.h"
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static int audio_read_header(AVFormatContext *s1)
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{
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OSSAudioData *s = s1->priv_data;
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AVStream *st;
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int ret;
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st = avformat_new_stream(s1, NULL);
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if (!st) {
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return AVERROR(ENOMEM);
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}
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ret = ff_oss_audio_open(s1, 0, s1->url);
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if (ret < 0) {
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return AVERROR(EIO);
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}
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/* take real parameters */
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st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codecpar->codec_id = s->codec_id;
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st->codecpar->sample_rate = s->sample_rate;
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st->codecpar->ch_layout.nb_channels = s->channels;
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avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
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return 0;
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}
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static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
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{
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OSSAudioData *s = s1->priv_data;
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int ret, bdelay;
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int64_t cur_time;
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struct audio_buf_info abufi;
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if ((ret=av_new_packet(pkt, s->frame_size)) < 0)
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return ret;
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ret = read(s->fd, pkt->data, pkt->size);
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if (ret <= 0){
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av_packet_unref(pkt);
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pkt->size = 0;
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if (ret<0) return AVERROR(errno);
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else return AVERROR_EOF;
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}
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pkt->size = ret;
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/* compute pts of the start of the packet */
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cur_time = av_gettime();
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bdelay = ret;
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if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
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bdelay += abufi.bytes;
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}
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/* subtract time represented by the number of bytes in the audio fifo */
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cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
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/* convert to wanted units */
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pkt->pts = cur_time;
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if (s->flip_left && s->channels == 2) {
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int i;
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short *p = (short *) pkt->data;
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for (i = 0; i < ret; i += 4) {
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*p = ~*p;
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p += 2;
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}
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}
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return 0;
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}
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static int audio_read_close(AVFormatContext *s1)
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{
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OSSAudioData *s = s1->priv_data;
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ff_oss_audio_close(s);
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return 0;
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}
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static const AVOption options[] = {
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{ "sample_rate", "", offsetof(OSSAudioData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
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{ "channels", "", offsetof(OSSAudioData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
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{ NULL },
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};
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static const AVClass oss_demuxer_class = {
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.class_name = "OSS indev",
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.item_name = av_default_item_name,
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.option = options,
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.version = LIBAVUTIL_VERSION_INT,
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.category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
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};
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const AVInputFormat ff_oss_demuxer = {
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.name = "oss",
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.long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) capture"),
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.priv_data_size = sizeof(OSSAudioData),
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.read_header = audio_read_header,
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.read_packet = audio_read_packet,
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.read_close = audio_read_close,
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.flags = AVFMT_NOFILE,
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.priv_class = &oss_demuxer_class,
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};
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