Mercurial > libavformat.hg
view daud.c @ 5178:092a527cabc8 libavformat
Extend check for integer overflow for malloc argument to take into account
also the addition of "sound_buffers" not only the multiplication.
author | reimar |
---|---|
date | Mon, 14 Sep 2009 17:15:18 +0000 |
parents | c3102b189cb6 |
children | 536e5527c1e0 |
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/* * 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