summaryrefslogtreecommitdiff
path: root/libavformat/matroskadec.c
diff options
context:
space:
mode:
authorAndreas Rheinhardt <andreas.rheinhardt@gmail.com>2020-04-22 00:15:54 +0200
committerAndreas Rheinhardt <andreas.rheinhardt@gmail.com>2020-05-01 07:27:36 +0200
commit4f5c6c1b0ec2407dfd42fcfa3441067de1962a54 (patch)
treedaab26d918d23996f3fd839d137170cb0bed9c06 /libavformat/matroskadec.c
parentc91e3690d9c6667123b116c9fd3becf5f4f4530e (diff)
avformat/matroskadec: Fix buffer overflow when demuxing RealAudio 28.8
RealAudio 28.8 (like other RealAudio codecs) uses a special demuxing mode in which the data of the existing Matroska Blocks is not simply forwarded as-is. Instead data from several Blocks is recombined together to output several packets. The parameters governing this process are parsed from the CodecPrivate: Coded framesize (cfs), frame size (w) and sub_packet_h (h). During demuxing, h/2 pieces of data of size cfs each are read from every Matroska (Simple)Block and put at offset m * 2 * w + n * cfs of a buffer of size h * w, where m ranges from 0 to h/2 - 1 for each Block while n is initially zero and incremented after a Block has been parsed until it is h, at which poin the assembled packets are output and n reset. The highest offset is given by (h/2 - 1) * 2 * w + (h - 1) * cfs + cfs while the destination buffer's size is given by h * w. For even h, this leads to a buffer overflow (and potential segfault) if h * cfs > 2 * w; for odd h, the condition is h * cfs > 3 * w. This commit adds a check to rule this out. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Diffstat (limited to 'libavformat/matroskadec.c')
-rw-r--r--libavformat/matroskadec.c3
1 files changed, 3 insertions, 0 deletions
diff --git a/libavformat/matroskadec.c b/libavformat/matroskadec.c
index 844f96cd52..951695b5b5 100644
--- a/libavformat/matroskadec.c
+++ b/libavformat/matroskadec.c
@@ -2612,6 +2612,9 @@ static int matroska_parse_tracks(AVFormatContext *s)
return AVERROR_INVALIDDATA;
if (codec_id == AV_CODEC_ID_RA_288) {
+ if ((int64_t)track->audio.sub_packet_h * track->audio.coded_framesize
+ > (2 + (track->audio.sub_packet_h & 1)) * track->audio.frame_size)
+ return AVERROR_INVALIDDATA;
st->codecpar->block_align = track->audio.coded_framesize;
track->codec_priv.size = 0;
} else {