[FFmpeg-devel] [PATCH 3/4] avformat/rmdec: Use 64bit for intermediate for DEINT_ID_INT4

Andreas Rheinhardt andreas.rheinhardt at outlook.com
Sat Apr 17 02:45:38 EEST 2021 

James Almer:
> On 4/16/2021 7:45 PM, James Almer wrote:
>> On 4/16/2021 7:24 PM, Andreas Rheinhardt wrote:
>>> 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 <michael at 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.
>>
>> Ah, good catch. This also means av_new_packet() is potentially being
>> called with 0 as size for these two codepaths.
>>
>>>
>>>>> +                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?
>>
>> This is done in plenty of places across the codebase to catch the same
>> kind of overflows. Does it make any measurable difference even worth
>> mentioning, especially considering this is read in the header?
>>
>> All these casts make the code really ugly and harder to read.
>> Especially things like (unsigned)INT_MAX. So if there are cleaner
>> solutions, they should be used if possible.
>> Code needs to not only work, but also be maintainable.
> 
> Another option is to just change the type of the RMStream fields, like so:
> 
>> diff --git a/libavformat/rmdec.c b/libavformat/rmdec.c
>> index fc3bff4859..304984d2b0 100644
>> --- a/libavformat/rmdec.c
>> +++ b/libavformat/rmdec.c
>> @@ -50,8 +50,8 @@ struct RMStream {
>>      /// Audio descrambling matrix parameters
>>      int64_t audiotimestamp; ///< Audio packet timestamp
>>      int sub_packet_cnt; // Subpacket counter, used while reading
>> -    int sub_packet_size, sub_packet_h, coded_framesize; ///<
>> Descrambling parameters from container
>> -    int audio_framesize; /// Audio frame size from container
>> +    unsigned sub_packet_size, sub_packet_h, coded_framesize; ///<
>> Descrambling parameters from container
>> +    unsigned audio_framesize; /// Audio frame size from container
>>      int sub_packet_lengths[16]; /// Length of each subpacket
>>      int32_t deint_id;  ///< deinterleaver used in audio stream
>>  };
>> @@ -277,7 +277,7 @@ static int
>> rm_read_audio_stream_info(AVFormatContext *s, AVIOContext *pb,
>>              }
>>              break;
>>          case DEINT_ID_GENR:
>> -            if (ast->sub_packet_size <= 0 ||
>> +            if (!ast->sub_packet_size ||
>>                  ast->sub_packet_size > ast->audio_framesize)
>>                  return AVERROR_INVALIDDATA;
>>              if (ast->audio_framesize % ast->sub_packet_size)
>> @@ -296,7 +296,7 @@ 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 > 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)
> 
> ast->audio_framesize and sub_packet_h are never bigger than INT16_MAX,
> so unless I'm missing something, this should be enough.

In the multiplication ast->coded_framesize * sub_packet_h the first is
read via av_rb32(). Your patch will indeed eliminate the undefined
behaviour (because unsigned), but it might be that the check will now
not trigger when it should trigger because only the lower 32bits are
compared.

- Andreas

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