/* * 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_UNISTD_H #include <unistd.h> #endif #include <fcntl.h> #include <sys/ioctl.h> #include <sys/soundcard.h> #include "libavutil/internal.h" #include "libavutil/opt.h" #include "libavutil/time.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->url); 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 indev", .item_name = av_default_item_name, .option = options, .version = LIBAVUTIL_VERSION_INT, .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT, }; const 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, };