Leaking this private structure opens up the possibility that it may
be re-used when parsing later packets in the stream. This is
problematic if the later packets are not the same codec type (e.g.
private allocated during Vorbis parsing, but later packets are Opus
and the private is assumed to be the oggopus_private type in
opus_header()).
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Larger values would imply file durations of astronomic proportions and cause
overflows
Fixes integer overflow
Fixes: usan_int64_overflow
Found-by: Thomas Guilbert <tguilbert@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Replace av_copy_packet and deprecated av_dup_packet by
creating reference using av_packet_ref.
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Jan Sebechlebsky <sebechlebskyjan@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
TeeSlave.bsfs is array of pointers to AVBitStreamFilterContext,
so element size should be really size of a pointer, not size
of TeeSlave structure.
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Jan Sebechlebsky <sebechlebskyjan@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
Adds per slave option 'onfail' to the tee muxer allowing an output to
fail, so other slave outputs can continue.
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Jan Sebechlebsky <sebechlebskyjan@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
In open_slave failure can happen before bsfs array is initialized,
close_slave must check that bsfs is not NULL before accessing
tee_slave->bsfs[i] element.
Slave muxer expects write_trailer to be called if it's
write_header suceeded (so resources allocated in write_header
are freed). Therefore if failure happens after successfull
write_header call, we must ensure that write_trailer of
that particular slave is called.
Some cleanups are made by Marton Balint.
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Jan Sebechlebsky <sebechlebskyjan@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
Until now, the decoding API was restricted to outputting 0 or 1 frames
per input packet. It also enforces a somewhat rigid dataflow in general.
This new API seeks to relax these restrictions by decoupling input and
output. Instead of doing a single call on each decode step, which may
consume the packet and may produce output, the new API requires the user
to send input first, and then ask for output.
For now, there are no codecs supporting this API. The API can work with
codecs using the old API, and most code added here is to make them
interoperate. The reverse is not possible, although for audio it might.
From Libav commit 05f66706d1.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This codepath isn't quite as bad as it used to sound, if fragments
are cut automatically at video packets.
Signed-off-by: Martin Storsjö <martin@martin.st>
Chris Cunningham
Michael Niedermayer
Jan Sebechlebsky
compn
Piotr Bandurski
James Almer
foo86
smallishzulu
Thomas Volkert
Carl Eugen Hoyos
Paul B Mahol
Peter Ross
Александр Слободенюк
Nicolas George
Derek Buitenhuis
wm4
Martin Storsjö
Luca Barbato
Anton Khirnov