On 4/15/2021 5:44 PM, Michael Niedermayer wrote:
Fixes: runtime error: signed integer overflow: 65312 * 65535 cannot be
represented in type 'int'
Fixes:
32832/clusterfuzz-testcase-minimized-ffmpeg_dem_RM_fuzzer-4817710040088576
Found-by: continuous fuzzing process
https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <mich...@niedermayer.cc>
---
libavformat/rmdec.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/libavformat/rmdec.c b/libavformat/rmdec.c
index fc3bff4859..af032ed90a 100644
--- a/libavformat/rmdec.c
+++ b/libavformat/rmdec.c
@@ -269,9 +269,9 @@ static int rm_read_audio_stream_info(AVFormatContext *s,
AVIOContext *pb,
case DEINT_ID_INT4:
if (ast->coded_framesize > ast->audio_framesize ||
sub_packet_h <= 1 ||
- ast->coded_framesize * sub_packet_h > (2 + (sub_packet_h & 1)) *
ast->audio_framesize)
+ ast->coded_framesize * (uint64_t)sub_packet_h > (2 + (sub_packet_h
& 1)) * ast->audio_framesize)
This check seems superfluous with the one below right after it.
ast->coded_framesize * sub_packet_h must be equal to 2 *
ast->audio_framesize. It can be removed.
return AVERROR_INVALIDDATA;
- if (ast->coded_framesize * sub_packet_h != 2*ast->audio_framesize)
{
+ if (ast->coded_framesize * (uint64_t)sub_packet_h !=
2*ast->audio_framesize) {
avpriv_request_sample(s, "mismatching interleaver
parameters");
return AVERROR_INVALIDDATA;
}
How about something like
diff --git a/libavformat/rmdec.c b/libavformat/rmdec.c
index fc3bff4859..09880ee3fe 100644
--- a/libavformat/rmdec.c
+++ b/libavformat/rmdec.c
@@ -269,7 +269,7 @@ static int rm_read_audio_stream_info(AVFormatContext *s,
AVIOContext *pb,
case DEINT_ID_INT4:
if (ast->coded_framesize > ast->audio_framesize ||
sub_packet_h <= 1 ||
- ast->coded_framesize * sub_packet_h > (2 + (sub_packet_h & 1)) *
ast->audio_framesize)
+ ast->audio_framesize > INT_MAX / sub_packet_h)
return AVERROR_INVALIDDATA;
if (ast->coded_framesize * sub_packet_h != 2*ast->audio_framesize)
{
avpriv_request_sample(s, "mismatching interleaver parameters");
Instead?
We already know that ast->coded_framesize is not bigger than
ast->audio_framesize, and with this change we'll also know that
ast->audio_framesize * sub_packet_h can't overflow, so neither will
ast->coded_framesize * sub_packet_h.
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