The first four bytes of smacker audio are supposed to contain the number
of samples, so treat audio frames smaller than that as invalid.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
When reading a new frame, the Smacker demuxer seeks to the next frame
position where it excepts the next frame; then it (potentially) reads
the palette, the audio packets associated with the frame and finally the
actual video frame. It is only at the end that the frame counter as well
as the position where the next frame is expected get updated.
This has a downside: If e.g. invalid data is encountered when reading
the palette, the demuxer returns immediately (with an error) and if the
caller calls av_read_frame again, the demuxer seeks to the position where
it already was, reads the very same palette data again and therefore will
return an error again. If the caller calls av_read_frame repeatedly
(say, until a packet is received or until EOF), this meight become an
infinite loop.
This could also happen if e.g. the size of one of the audio frames was
invalid or if the frame size was gigantic.
This commit changes this by skipping a frame if it turns out to be
invalid or an error happens otherwise. This ensures that EOF will be
returned eventually in the above scenario.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The Smacker demuxer buffers audio packets before it outputs them, but it
increments the counter of buffered packets prematurely: If allocating
the audio buffer fails, an error (most likely AVERROR(ENOMEM)) is returned.
If the caller decides to call av_read_frame() again, the next call will
take the codepath for returning already buffered audio packets and it
will fail (because the buffer that ought to be allocated isn't) without
decrementing the number of supposedly buffered audio packets (it doesn't
matter whether there would be enough memory available in subsequent calls).
Depending on the caller's behaviour this is potentially an infinite loop.
This commit fixes this by only incrementing the number of buffered audio
packets after having successfully read them and unconditionally reducing
said number when outputting one of them. It also changes the semantics
of the curstream variable: It is now the number of currently buffered
audio packets whereas it used to be the index of the last audio stream
to be read. (Index refers to the index in the array of buffers, not to
the stream index.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Don't need to do double check by the description of the API.
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
Saves initialization of an HEVCDecoderConfigurationRecord when
the data is already in ISOBMFF-format or if it is plainly invalid.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Fix two cases of memleaks:
1. The leak of dv_demux
2. The leak of dv_fctx upon dv_demux allocate failure
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The code for demuxing DV audio predates the introduction of refcounted
packets and when the latter was added, changes to the former were
forgotten. This meant that when avpriv_dv_produce_packet initialized the
packet containing the AVBufferRef, the AVBufferRef as well as the
underlying AVBuffer leaked; the actual packet data didn't leak: They
were directly freed, but not via their AVBuffer's free function.
https://samples.ffmpeg.org/ffmpeg-bugs/trac/ticket4671/dir1.tar.bz2
contains samples for this (enable_drefs needs to be enabled for them).
Moreover, errors in avpriv_dv_produce_packet were ignored; this has been
changed, too.
Furthermore, in the hypothetical scenario that the track has a palette,
this would leak, too, so reorder the code so that the palette code
appears after the DV audio code.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Fixes: signed integer overflow: 8 * 9223372036854774783 cannot be represented in type 'long'
Fixes: 23381/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-4818340509122560
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The segments / url can be modified by the io read when reloading
This may be an alternative or additional fix for Ticket8673
as a further alternative the reload stuff could be disabled during
probing
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Also switch to using a pointer to access stream side data instead of
copying the stream's AVPacketSideData.
Reviewed-by: Limin Wang <lance.lmwang@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
When doing streamed output, with e.g. +dash, if the mfra box ended
up being larger than the AVIOContext write buffer, the (unchecked)
seeking back to update the box size would silently fail and produce
an invalid mfra box.
This is similar to how other boxes are written in fragmented mode.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Nothing written in avformat_write_trailer() for the submuxers will be
output anyway because the AVIOContexts used for actual output have been
closed before the call. Writing the trailer of the subcontext has probably
only been done in order to free the memory allocated by the submuxer.
And this job has been taken over by the deinit functions.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Fixes: signed integer overflow: 1394614304 * 2 cannot be represented in type 'int'
Fixes: 23491/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5697377020411904
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Up until now, ff_avc_decode_sps would parse a SPS and return some
properties from it in a freshly allocated structure. Yet said structure
is very small and completely internal to libavformat, so there is no
reason to use the heap for it. This commit therefore changes the
function to return an int and to modify a caller-provided structure.
This will also allow ff_avc_decode_sps to return better error codes in
the future.
It also fixes a memleak in mxfenc: If a packet contained multiple SPS,
only the SPS structure belonging to the last SPS would be freed, the
other ones would leak when the pointer is overwritten to point to the
new SPS structure. Of course, without allocations there are no leaks.
This is Coverity issue #1445194.
Furthermore, the SPS structure has been renamed from
H264SequenceParameterSet to H264SPS in order to avoid overlong lines.
Reviewed-by: Tomas Härdin <tjoppen@acc.umu.se>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
After parsing the end timestamp of a WebVTT cue block, the current code
skips everything after the start of the timestamp that is not a \t, ' '
or \n and treats what is next as the start of a WebVTT cue settings list.
Yet if there is no such list, but a single \r, this will skip a part of
the cue payload (namely everything until the first occurence of \t, ' '
or \n) and treat what has not been skipped as the beginning of the
WebVTT cue settings list that extends until the next \r or \n (or the
end).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Given that AV1 only has exactly one sequence header, it is unnecessary
to copy the content of said sequence header into an intermediate dynamic
buffer; instead the sequence header can be copied from where it is in
the input buffer.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Fixes: signed integer overflow: 1 - -9223372036854775808 cannot be represented in type 'long'
Fixes: 23490/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5133490093031424
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Given that av_get_packet returns a blank packet on error, the only
difference to the current approach (that uses intermediate AVPackets on
the stack) is that st->attached_pic will be properly initialized on error
(i.e. the timestamps are AV_NOPTS_VALUE) whereas right now st->attached_pic
is only zeroed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Document that it also sets the size in case the desired side data is
absent (if the pointer has been supplied).
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
If stream's bitrate is not specified:
- for static manifest: an average bitrate will be calculated and used,
- for dynamic manifest: first segment's bitrate will be calculated and used, as before,
for bandwidth setting in adaptation sets.
dash demuxer get the stream info from sub-stream, but missed side
data/disposition part, e,g, missed the DOVI side data when the
stream is Dolby Vision streams
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
hls demuxer get the stream info from sub-stream, but missed side
data/disposition part, e,g, missed the DOVI side data when the
stream is Dolby Vision streams.
Reviewed-by <liuqi05@kuaishou.com>
Signed-off-by: vacingfang <vacingfang@tencent.com>
