On 4/16/2021 4:04 PM, Michael Niedermayer wrote:
On Thu, Apr 15, 2021 at 06:22:10PM -0300, James Almer wrote:
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?

The 2 if() execute different things, the 2nd requests a sample, the first
not. I think this suggestion would change when we request a sample

Why are we returning INVALIDDATA after requesting a sample, for that matter? If it's considered an invalid scenario, do we really need a sample?

In any case, if you don't want more files where "ast->coded_framesize * sub_packet_h != 2*ast->audio_framesize" would print a sample request, then maybe something like the following could be used instead?

diff --git a/libavformat/rmdec.c b/libavformat/rmdec.c
index fc3bff4859..10c1699a81 100644
--- a/libavformat/rmdec.c
+++ b/libavformat/rmdec.c
@@ -269,6 +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->audio_framesize > INT_MAX / sub_packet_h ||
                 ast->coded_framesize * sub_packet_h > (2 + (sub_packet_h & 1)) * 
ast->audio_framesize)
                 return AVERROR_INVALIDDATA;
             if (ast->coded_framesize * sub_packet_h != 2*ast->audio_framesize) 
{
@@ -278,12 +279,16 @@ static int rm_read_audio_stream_info(AVFormatContext *s, 
AVIOContext *pb,
             break;
         case DEINT_ID_GENR:
             if (ast->sub_packet_size <= 0 ||
+                ast->audio_framesize > INT_MAX / sub_packet_h ||
                 ast->sub_packet_size > ast->audio_framesize)
                 return AVERROR_INVALIDDATA;
             if (ast->audio_framesize % ast->sub_packet_size)
                 return AVERROR_INVALIDDATA;
             break;
         case DEINT_ID_SIPR:
+            if (ast->audio_framesize > INT_MAX / sub_packet_h)
+                return AVERROR_INVALIDDATA;
+            break;
         case DEINT_ID_INT0:
         case DEINT_ID_VBRS:
         case DEINT_ID_VBRF:
@@ -296,7 +301,6 @@ static int rm_read_audio_stream_info(AVFormatContext *s, 
AVIOContext *pb,
             ast->deint_id == DEINT_ID_GENR ||
             ast->deint_id == DEINT_ID_SIPR) {
             if (st->codecpar->block_align <= 0 ||
-                ast->audio_framesize * (uint64_t)sub_packet_h > 
(unsigned)INT_MAX ||
                 ast->audio_framesize * sub_packet_h < 
st->codecpar->block_align)
                 return AVERROR_INVALIDDATA;
             if (av_new_packet(&ast->pkt, ast->audio_framesize * sub_packet_h) 
< 0)

Same amount of checks for all three deint ids, and no integer casting to prevent overflows.
_______________________________________________
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".

Reply via email to