* commit 'a2a991b2ddf951454ffceb7bcedc9db93e26c610':
srtp: Improve the minimum encryption buffer size check
srtp: Add support for a few DTLS-SRTP related crypto suites
Conflicts:
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'f53490cc0c809975f8238d5a9edbd26f83bd2f84':
rtpdec/srtp: Handle CSRC fields being present
rtpdec: Check the return value from av_new_packet
ac3dec: fix non-optimal dithering of zero bit mantissas
Conflicts:
libavcodec/ac3dec.c
libavformat/rtpdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'c6f1dc8e4cd967ae056698eafb891a08003c211c':
rtpdec: Move setting the parsing flags to the actual depacketizers
rtpdec: Split handling of mpeg12 audio/video to a separate depacketizer
Conflicts:
libavformat/rtpdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'a717f9904227d7979473bad40c50eb40af41d01d':
mpegts: Share the cleanup code between the demuxer and lavf-internal parser functions
rtpdec_mpeg4: Return one AAC AU per AVPacket
ppc: Include string.h for memset
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This clarifies where the limit number comes from, and only
requires exactly as much padding space as will be needed.
Signed-off-by: Martin Storsjö <martin@martin.st>
The main difference to the existing suites from RFC 4568 is
that the version with a 32 bit HMAC still uses 80 bit HMAC
for RTCP packets.
Signed-off-by: Martin Storsjö <martin@martin.st>
Several compilers such as clang/icc/pathscale will optimize the check
pos + size < pos (assuming size > 0) into false, since signed integer
overflow is undefined behavior in C. This breaks overflow checking.
Use a safe precondition check instead.
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The lavf-internal parser functions are used when receiving
mpegts over RTP. This fixes memory leaks in this setup.
The normal mpegts demuxer close function was updated in ec7d0d2e in
2004 to fix leaks, but the parsing function used for RTP wasn't
updated and has been leaking ever since.
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes the returned data valid to stream copy into other
containers as well, not only for decoding straight away.
Signed-off-by: Martin Storsjö <martin@martin.st>
Makes ff_id3v2_read reset stream position at the end of ID3 data if the
header size is not matched (caused by an EOF for example).
Current behaviour (without the patch):
filesize = 400
id3 data size = 399
file offset after ff_id3v2_read is 400 instead of 399
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Write the packet unaltered if found.
Fixes ticket #1917
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The code did not account properly for packets that where added to
the end of the packet list. Also flags for such packets where not
set correctly leading to incorrect chunked interleaving.
Reported-by: bcoudurier
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The exact packing of Opus inside Matroska is not finalized.
Use A_OPUS/EXPERIMENTAL as codec name, like mkvtoolnix.
The A_OPUS name stays to let ffmpeg open files it has produced
until now, but newly produced file use the EXPERIMENTAL version.
Once the spec is stabilized it will be possible to consider
options to ensure compatibility with these files.
a small value was rounded to 0 and then treated special as if
chunked_duration was 0. This led to a inconsistency that further led
to wrong interleaving
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
lavc: Move vector_fmul_window to AVFloatDSPContext
rtpdec_mpeg4: Check the remaining amount of data before reading
Conflicts:
libavcodec/dsputil.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '977d4a3b8a2dbc2fb5e747c7072485016c9cdfaa':
rtpdec_mpeg4: Check the return value from malloc
srtp: Mark a few variables as uninitialized
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '0eecafc948b74c247ebbc59f18f508db5d590d0b':
configure: Make the new srtp protocol depend on the rtp protocol
lavf: Add a fate test for the SRTP functions
lavu: Add a fate test for the HMAC API
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Attempting to re-parse the headers at demuxer level is a
pandora box the way its done currently.
This allows full reconfiguration of vorbis streams
Fixes Ticket2117
Fixes Ticket2121
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
