Parsing should probably be enabled for all codecs, at least for headers,
but e.g. the AAC parser produces 1-byte packets of zero padding with it,
so I'm just enabling it for EAC3 for the moment.
Removed the unnecessary calls to ff_format_io_close
this patch introduced in dashenc_delete_file.
dashenc_delete_file functions open a
new HTTP connection regardless of the http_persistent value,
So change their behaviour to keep http connections open
if http_persistent is set.
Signed-off-by: Basel Sayeh <basel.sayeh@hotmail.com>
Removed the unnecessary calls to ff_format_io_close
this patch introduced in hls_delete_file.
hls_delete_file functions open a new HTTP connection
regardless of the http_persistent value,
So change their behaviour to keep http connections open
if http_persistent is set
Signed-off-by: Basel Sayeh <basel.sayeh@hotmail.com>
The HEIF specification permits specifying the looping behavior of
animated sequences by using the EditList (elst) box. The track
duration will be set to the total duration of all the loops (or
infinite) and the duration of a single loop will be set in the edit
list box.
The default behavior is to loop infinitely.
Compliance verification:
* This was added in libavif recently [1] and the files produced by
ffmpeg after this change have EditList boxes similar to the ones
produced by libavif (and avifdec is able to parse the loop count as
intended).
* ComplianceWarden is ok with the produced files.
* Chrome is able to play back the produced files.
[1] 4d2776a3af
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
Allow specifying the movie_timescale options to AVIF ouptut.
This also makes sure that when movie_timescale is not specified,
the default value of 1000 is used instead of 0. Animated AVIF
files which don't specify the movie_timescale will have the
correct duration written in the track and movie headers after this
change (instead of writing 0).
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
It is a URL rewriter for IPFS gateways, not an actual implementation of
IPFS, and naming it as such was both incorrect and misleading.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Advanced edit list support is entirely broken for fragmented MP4s,
currently. mov_fix_index is never run in mov_build_index, since
in fragmented MP4s the stco, stsz, stts, and stsc boxes have zero
entries, with the index being filled in as each fragment's trun
box is seen.
The result of this is that the skip samples is never set properly,
since half the code thinks it doesn't need to, as advanced_editlist
is enabled, but as mov_fix_index is never called, it doesnt get set.
This means that any edits for e.g. priming are not properly applied
as skip samples side data.
This also means remuxing to fragmented MP4 from progressive MP4 with
lavf will quietly drop the edit list, currently.
Example:
$ ffmpeg -loglevel quiet -advanced_editlist 1 -i non_fragmented.mp4 -f md5 -
MD5=d02d929f8eb4edef624758a298d5f7c6
$ ffmpeg -loglevel quiet -advanced_editlist 0 -i non_fragmented.mp4 -f md5 -
MD5=d02d929f8eb4edef624758a298d5f7c6
$ ffmpeg -loglevel quiet -advanced_editlist 1 -i fragmented.mp4 -f md5 -
MD5=e38b110f586fa886ff94e0ca98a95d59 <-- wrong, extra samples are output instead of being skipped
$ ffmpeg -loglevel quiet -advanced_editlist 0 -i fragmented.mp4 -f md5 -
MD5=d02d929f8eb4edef624758a298d5f7c6
We cannot call mov_fix_index after reading a trun box
since mov_fix_index seems to assume it is only called once, on a
fully complete index, an multiple calls to it don't seem like
they'd work, so the "best" option seems to be disabling advanced
edit list support entirely for the time being, as it is broken
for these types of files.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Commit 18f24527eb accidentally made side data only packets be handled like a
flush request. Fix this regression by effectively ignoring them as was the
original intention.
Signed-off-by: James Almer <jamrial@gmail.com>
Fixes: signed integer overflow: 0 - -9223372036854775808 cannot be represented in type 'long int'
Fixes: fate-cover-art-aiff-id3v2-remux, fate-cover-art-mp3-id3v2-remux and fate-mov-cover-image
under ubsan.
Signed-off-by: James Almer <jamrial@gmail.com>
New option can be used to avoid creating very short segments with inputs
whose GOP size is variable or unharmonic with segment_time.
Only effective with segment_time.
Fixes: signed integer overflow: 48000 * 223587 cannot be represented in type 'int'
Fixes: 54513/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5817594836025344
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Tomas Härdin <git@haerdin.se>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Some encoders, like flac, can send side data only packets at the end.
Eventually, said extradata update should ideally be used to update the header
when writting to seekable output, but for now, ignore them.
Should fix the undefined behavior of passing NULL to memcpy().
Signed-off-by: James Almer <jamrial@gmail.com>
Fixes: OOM testcase
Fixes: 51527/clusterfuzz-testcase-minimized-ffmpeg_dem_LAF_fuzzer-5453663505612800
OOM can still happen after this as an arbitrary sized block is allocated and read
this would require a redesign or some limit on the sample rate.
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The old warning is no longer applicable in the inner block after
c5b20cfe19.
Reviewed-by: Zhao Zhili <quinkblack@foxmail.com>
Signed-off-by: Gyan Doshi <ffmpeg@gyani.pro>
Currently, several components select atsc_a53, despite
not using anything from it themselves. They only select
it because parsing SEI messages adds an indirect dependency.
But using direct dependencies is more natural, so add
dedicated subsystems for them.
It already allows to remove a superfluous dependency of
the HEVC QSV encoder on hevc_sei and atsc_a53.
Adding new subsystems only becomes effective after a reconfiguration.
In order to force this, some needed headers (which are only included
implicitly before this commit) were included explicitly in
libavformat/allformats.c.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Fixes some MP4F files which have duration in mdhd set to UINT_MAX instead of zero.
Signed-off-by: Sasi Inguva <isasi@google.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Fixes: Timeout (read mostly the same data repeatly)
Fixes: 52457/clusterfuzz-testcase-minimized-ffmpeg_dem_ALP_fuzzer-6610706313379840
Fixes: 53098/clusterfuzz-testcase-minimized-ffmpeg_dem_SOL_fuzzer-6481382981632000
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
VP6 alpha in EA format is a second VP6 encoded video stream where only the Y
component is used and is interpreted as the alpha channel of the first VP6
stream. The alpha VP6 stream is muxed separately from the main VP6 stream, has
its own stream headers and packet headers. In theory the two streams might not
even have the same resolution (although most likely that is not something that
is seen or supported in the wild), but the format is capable of doing it.
Merged VP6 alpha (also known as the VP6A codec) means that a packet of the
video stream contains the corresponding packet of both VP6 substreams like
this:
{OffsetOfAlpha, DataPacket, AlphaDataPacket}
So data and alpha data of a frame is merged to a single packet, this is how VP6
video with alpha is muxed in FLV and SWF.
The first approach is more like how the demuxer sees data in the EA format,
unfortunately it is different to what the FLV or SWF format expects, so -
having no better place for it in the framework - I decided to do an optional
format conversion in the EA demuxer.
Signed-off-by: Marton Balint <cus@passwd.hu>
Profile can be derived from values codecpar pixel format only with software
formats. For hardware formats, we're forced to parse a frame header to get
the required information.
Signed-off-by: James Almer <jamrial@gmail.com>
Fixes: signed integer overflow: -2147483648 * 100000 cannot be represented in type 'int'
Fixes: 52060/clusterfuzz-testcase-minimized-ffmpeg_dem_MP3_fuzzer-5131616708329472
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Paul B Mahol
rcombs
Basel Sayeh
Vignesh Venkatasubramanian
Zhao Zhili
Derek Buitenhuis
James Almer
Gyan Doshi
Michael Niedermayer
Marvin Scholz
Andreas Rheinhardt
Sasi Inguva
Marton Balint