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authorMartin Storsjö <martin@martin.st>2013-09-16 21:27:49 +0300
committerLuca Barbato <lu_zero@gentoo.org>2013-10-04 03:26:27 +0200
commit00516b5491fbd99e4057f21eae231fc02cc596e3 (patch)
tree64ef13ad2213ce40dcd56bba47c294cdd7ed3c26
parent3706c229920115f022876ef07b16217dce504db6 (diff)
downloadffmpeg-00516b5491fbd99e4057f21eae231fc02cc596e3.tar.gz
xmv: Add more sanity checks for parameters read from the bitstream
Since the number of channels is multiplied by 36 and assigned to to a uint16_t, make sure this calculation didn't overflow. (In certain cases the calculation could overflow leaving the truncated block_align at 0, leading to divisions by zero later.) Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org Signed-off-by: Martin Storsjö <martin@martin.st> (cherry picked from commit d4c2a3740fb95f952a87ba320d2bf31f126bdf68) Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
-rw-r--r--libavformat/xmv.c7
1 files changed, 5 insertions, 2 deletions
diff --git a/libavformat/xmv.c b/libavformat/xmv.c
index d491dec8f6..201dc3ac72 100644
--- a/libavformat/xmv.c
+++ b/libavformat/xmv.c
@@ -43,6 +43,8 @@
XMV_AUDIO_ADPCM51_FRONTCENTERLOW | \
XMV_AUDIO_ADPCM51_REARLEFTRIGHT)
+#define XMV_BLOCK_ALIGN_SIZE 36
+
typedef struct XMVAudioTrack {
uint16_t compression;
uint16_t channels;
@@ -207,7 +209,7 @@ static int xmv_read_header(AVFormatContext *s)
track->bit_rate = track->bits_per_sample *
track->sample_rate *
track->channels;
- track->block_align = 36 * track->channels;
+ track->block_align = XMV_BLOCK_ALIGN_SIZE * track->channels;
track->block_samples = 64;
track->codec_id = ff_wav_codec_get_id(track->compression,
track->bits_per_sample);
@@ -224,7 +226,8 @@ static int xmv_read_header(AVFormatContext *s)
av_log(s, AV_LOG_WARNING, "Unsupported 5.1 ADPCM audio stream "
"(0x%04X)\n", track->flags);
- if (!track->channels || !track->sample_rate) {
+ if (!track->channels || !track->sample_rate ||
+ track->channels >= UINT16_MAX / XMV_BLOCK_ALIGN_SIZE) {
av_log(s, AV_LOG_ERROR, "Invalid parameters for audio track %d.\n",
audio_track);
ret = AVERROR_INVALIDDATA;