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authorMichael Niedermayer <michael@niedermayer.cc>2021-03-03 10:49:03 +0100
committerMichael Niedermayer <michael@niedermayer.cc>2021-03-03 16:54:20 +0100
commitfdcb966f4a3c6f872891b8dd554e3652b9e02d4f (patch)
tree7a3cf1e3f04bcb91f28ece03d2486237478e9bdc
parent6a3f345c1ab78428b970775a6632362b8515dd04 (diff)
downloadffmpeg-fdcb966f4a3c6f872891b8dd554e3652b9e02d4f.tar.gz
avformat/dcstr: Check sample rate
Fixes: signed integer overflow: -1300248894420254720 * 16 cannot be represented in type 'long' Fixes: 30879/clusterfuzz-testcase-minimized-ffmpeg_dem_DCSTR_fuzzer-5094464215449600 Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg Reviewed-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
-rw-r--r--libavformat/dcstr.c2
1 files changed, 2 insertions, 0 deletions
diff --git a/libavformat/dcstr.c b/libavformat/dcstr.c
index 9041a86539..9ddccc7f2d 100644
--- a/libavformat/dcstr.c
+++ b/libavformat/dcstr.c
@@ -43,6 +43,8 @@ static int dcstr_read_header(AVFormatContext *s)
st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->channels = avio_rl32(s->pb);
st->codecpar->sample_rate = avio_rl32(s->pb);
+ if (st->codecpar->sample_rate <= 0)
+ return AVERROR_INVALIDDATA;
codec = avio_rl32(s->pb);
align = avio_rl32(s->pb);
avio_skip(s->pb, 4);