1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-11-26 19:01:44 +02:00
FFmpeg/libavformat/pvfdec.c
Andreas Cadhalpun 169c1cfa92 pvfdec: prevent overflow during block alignment calculation
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
2017-01-29 01:20:52 +01:00

78 lines
2.4 KiB
C

/*
* PVF demuxer
* Copyright (c) 2012 Paul B Mahol
*
* 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 "libavcodec/internal.h"
#include "avformat.h"
#include "internal.h"
#include "pcm.h"
static int pvf_probe(AVProbeData *p)
{
if (!memcmp(p->buf, "PVF1\n", 5))
return AVPROBE_SCORE_MAX;
return 0;
}
static int pvf_read_header(AVFormatContext *s)
{
char buffer[32];
AVStream *st;
int bps, channels, sample_rate;
avio_skip(s->pb, 5);
ff_get_line(s->pb, buffer, sizeof(buffer));
if (sscanf(buffer, "%d %d %d",
&channels,
&sample_rate,
&bps) != 3)
return AVERROR_INVALIDDATA;
if (channels <= 0 || channels > FF_SANE_NB_CHANNELS ||
bps <= 0 || bps > INT_MAX / FF_SANE_NB_CHANNELS || sample_rate <= 0)
return AVERROR_INVALIDDATA;
st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->channels = channels;
st->codecpar->sample_rate = sample_rate;
st->codecpar->codec_id = ff_get_pcm_codec_id(bps, 0, 1, 0xFFFF);
st->codecpar->bits_per_coded_sample = bps;
st->codecpar->block_align = bps * st->codecpar->channels / 8;
avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
return 0;
}
AVInputFormat ff_pvf_demuxer = {
.name = "pvf",
.long_name = NULL_IF_CONFIG_SMALL("PVF (Portable Voice Format)"),
.read_probe = pvf_probe,
.read_header = pvf_read_header,
.read_packet = ff_pcm_read_packet,
.read_seek = ff_pcm_read_seek,
.extensions = "pvf",
.flags = AVFMT_GENERIC_INDEX,
};