Mercurial > libavformat.hg
view daud.c @ 5149:4a53fcd622ea libavformat
Check for seek failures in avi_load_index, otherwise if the index offset
is invalid (e.g. truncated file) we might end up reading the whole file
since trying to seek beyond the end of file does not set EOF.
author | reimar |
---|---|
date | Wed, 26 Aug 2009 08:38:44 +0000 |
parents | c3102b189cb6 |
children | 536e5527c1e0 |
line wrap: on
line source
/* * D-Cinema audio demuxer * Copyright (c) 2005 Reimar Döffinger * * 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 "avformat.h" static int daud_header(AVFormatContext *s, AVFormatParameters *ap) { AVStream *st = av_new_stream(s, 0); if (!st) return AVERROR(ENOMEM); st->codec->codec_type = CODEC_TYPE_AUDIO; st->codec->codec_id = CODEC_ID_PCM_S24DAUD; st->codec->codec_tag = MKTAG('d', 'a', 'u', 'd'); st->codec->channels = 6; st->codec->sample_rate = 96000; st->codec->bit_rate = 3 * 6 * 96000 * 8; st->codec->block_align = 3 * 6; st->codec->bits_per_coded_sample = 24; return 0; } static int daud_packet(AVFormatContext *s, AVPacket *pkt) { ByteIOContext *pb = s->pb; int ret, size; if (url_feof(pb)) return AVERROR(EIO); size = get_be16(pb); get_be16(pb); // unknown ret = av_get_packet(pb, pkt, size); pkt->stream_index = 0; return ret; } static int daud_write_header(struct AVFormatContext *s) { AVCodecContext *codec = s->streams[0]->codec; if (codec->channels!=6 || codec->sample_rate!=96000) return -1; return 0; } static int daud_write_packet(struct AVFormatContext *s, AVPacket *pkt) { put_be16(s->pb, pkt->size); put_be16(s->pb, 0x8010); // unknown put_buffer(s->pb, pkt->data, pkt->size); put_flush_packet(s->pb); return 0; } #if CONFIG_DAUD_DEMUXER AVInputFormat daud_demuxer = { "daud", NULL_IF_CONFIG_SMALL("D-Cinema audio format"), 0, NULL, daud_header, daud_packet, NULL, NULL, .extensions = "302", }; #endif #if CONFIG_DAUD_MUXER AVOutputFormat daud_muxer = { "daud", NULL_IF_CONFIG_SMALL("D-Cinema audio format"), NULL, "302", 0, CODEC_ID_PCM_S24DAUD, CODEC_ID_NONE, daud_write_header, daud_write_packet, .flags= AVFMT_NOTIMESTAMPS, }; #endif