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>

diff --git a/libavformat/dcstr.c b/libavformat/dcstr.c
index 9041a8653953b6eadee3814555fe6d0ece3ca680..9ddccc7f2d48c021564251f74a9b8ca12fb86cef 100644 (file)
--- 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);