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.rheinha...@gmail.com> --- libavformat/matroskadec.c | 3 +++ 1 file changed, 3 insertions(+) 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 { -- 2.20.1 _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe".