[FFmpeg-devel] [PATCH] fix parsing of broken mp3 streams

[Zdenek Kabelac]

2009/4/21 Michael Niedermayer <michaelni at gmx.at>:
> On Tue, Apr 21, 2009 at 11:14:16AM +0200, Zdenek Kabelac wrote:
>> 2009/4/21 Michael Niedermayer <michaelni at gmx.at>:
>> > On Tue, Apr 21, 2009 at 01:01:04AM +0200, Zdenek Kabelac wrote:
>> >> 2009/4/20 Michael Niedermayer <michaelni at gmx.at>:
>> >> > On Mon, Apr 20, 2009 at 09:37:25PM +0200, Zdenek Kabelac wrote:
>> >> >> 2009/4/19 Michael Niedermayer <michaelni at gmx.at>:
>> >> >> > On Sun, Apr 19, 2009 at 11:18:06PM +0200, Zdenek Kabelac wrote:
>> >> >> >> Hi
>> >> >> >>
>> >> >> >> Here is a small patch that fixes of running out-of-buffer in parsing
>> >> >> >> broken mp3 data stream.
>> >> >> >> This solution is rather a hotfix - better solution would be to check
>> >> >> >> at least one or two next mp3
>> >> >> >> frames in sequence whether they are part of the same audio stream or
>> >> >> >> some random junk
>> >> >> >> which has 0xfffx header inside. With this patch ugly noise could be
>> >> >> >> sometimes noticed.
>> >> >> >>
>> >> >> >> Also questionable is whether it should return -1 if no header is found
>> >> >> >> or rather return skipped
>> >> >> >> bytes and out_size = 0 - as then usually such packet is rescaned
>> >> >> >> multiple times with
>> >> >> >> one-byte step forward...
>> >> >> >>
>> >> >> >> Zdenek
>> >> >> >>
>> >> >> >> - Fix buffer overrun
>> >> >> >> - Properly return parsed bytes together with skipped bytes
>> >> >> >
>> >> >> > please provide a sample so we can confirm the bugfix, the patch
>> >> >> > looks mostly correct though
>> >> >> >
>> >> >>
>> >> >> I've upload just one mp3 dumped stream upload.ffmpeg.org as
>> >> >> junk_at_mp3stream ?directory - together with short text and two patch
>> >> >
>> >> >> - I'm attaching patch for api-example.c ?to easily compare results.
>> >> >
>> >> > i dont care what a modified tool does
>> >> > is there a problem that is reproduceable with ffmpeg or ffplay that
>> >> > your patch fixes?
>> >>
>> >> Patch is fixing mp3 decoder to skip only broken junk inside passed
>> >> data ?while decoding as much mp3 frames as possible and avoid buffer
>> >> over reading - don't ask me which tools are muxing avi streams with
>> >> junk in packets - obviously it some kind of re-synchronization from
>> >> splinting huge avi streams into small chunks....
>> >>
>> >> You could check for your self is to compare the result of extracted
>> >> wav size via api-example and then do
>> >> the same with ffmpeg -i junk.mp3 ?o.wav - you might observe small
>> >> difference 4027436 != 4018220
>> >> To do my homework and complete the list: mplayer -ao pcm:file=wav
>> >> junk.mp3 - creates 4022830 - but IMHO it decodes some broken packets
>> >> at the begining)
>> >>
>> >> (btw the patch for api-example should be probably commited into svn as well...)
>> >> Usually such badly muxed sample streams are way to small to notice
>> >> significant desynchronization.
>> >
>> > your original patch looked fine but after that you just talk nonsense
>> > apiexample is a example for codecs, not containers, mp3 must be passed
>> > through a demuxer and parser.
>>
>> I knew it would be hard - anyway I'll try once again - please check my
>> original patch
>> and see the mpegaudiodec.c code then please answer me following question:
>>
>> - What will stop parser from checking given buffer for mp3 header tag
>> after the buffer size
>> ?i.e. pass there zero memory area ?- I think decoder shouldn't run
>> behind the given buffer
>> even in the case it contains obviously wrong data - i.e. non-mp3 in this case.
>> (user would have to put false mp3 header after the passed buffer to
>> stop the parser)
>>
>> - If the mp3 packet is found within some offset from the beginning why
>> it should return
>> the size of parsed packed without the skipped bytes from the start of buffer.
>> (so next parsing will again start in the middle of previous mp3 packet)
>>
>> - Explain how the libavformat/mp3.c:mp3_read_packet() solve the problem?
>> (speaking of MP3_PACKET_SIZE - theoretical mythical max size of mp3
>> chunk is 1440)
>
> Iam not disputing that the original patch possibly fixes a issue, i
> am asking if you have a test case so we can test it.
>
> either
> A. the patch has no effect at all on ffmpeg & ffplay

I think I've already shown that we could get a different amount of WAV
samples from particular mp3 audio stream - we might have a discussion
which number is correct - but IMHO ffmpeg tool  should always try to
get as much as possible original samples from data stream - but I
could be alone...

> B. there is a file for which behavior changes

The fact that it's not running out of memory bounds when the mp3
header could not be found in the given buffer is probably because
usually lots of other mp3 frames are lying nearby in memory so it will
effective stop - and there are not too many heavily broken stream.

So to answer your questions
A - currently my patch does not influence those tools as they discard
whole data chunk if the error is found.
B - artificial file could be probably created which will show problem
from scanning data past the buffer - and generate coredump - though
it's not probably so simple to ensure memory layout that no mp3 header
will not be found past the allocated header.

I assume ffmpeg is not leaving simple buffer scanning bugs inside just
because there is no real file in the world that shows generate a
segfault?

Zdenek