James Almer: > 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)
sub_packet_h has not been checked for being != 0 here and in the DEINT_ID_GENR codepath. >> + 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. Since when is a division better than casting to 64bits to perform a multiplication? - Andreas _______________________________________________ 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".