This only takes care of decrypting incoming packets; the outgoing
RTCP packets are not encrypted. This is enough for some use cases,
and signalling crypto keys for use with outgoing RTCP packets
doesn't fit as simply into the API. If the SDP demuxer is hooked
up with custom IO, the return packets can be encrypted e.g. via the
SRTP protocol.
If the SRTP keys aren't available within the SDP, the decryption
can be handled externally as well (when using custom IO).
Signed-off-by: Martin Storsjö <martin@martin.st>
This supports the AES_CM_128_HMAC_SHA1_80 and
AES_CM_128_HMAC_SHA1_32 cipher suites (from RFC 4568) at the
moment. The main missing features are replay protection (which can be
added later without changing the internal API), and the F8 and null
ciphers.
Signed-off-by: Martin Storsjö <martin@martin.st>
The function is a callback that is called by ff_gen_search with
a constant stream index.
Avoid a false positive on older gcc version.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This makes the behaviour defined when they wrap around. The value
assigned to expected_prior was a uint32_t already.
Signed-off-by: Martin Storsjö <martin@martin.st>
Without this, we'd signal a huge loss rate (due to unsigned
wraparound) if we had received one packet more than expected (that
is, one seq number sent twice). The code has a check for lost_interval
<= 0, but that doesn't do what was intended as long as the variable is
unsigned.
Signed-off-by: Martin Storsjö <martin@martin.st>
The code below the comment does not at all relate to statistics,
and even if moved to the right place, the comment adds little
value.
Signed-off-by: Martin Storsjö <martin@martin.st>
"analyzeduration" is not used to detect the input duration, but to
specify the max probe data duration. Fix option description and related
doc entry accordingly.
* commit 'abae27ed3acd0a7c54f11760c5be2d2653c4edf8':
rtpdec: Fix the calculation of expected number of packets
fate: vp3: Fix fate-vp3-coeff-level64 test dependencies
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Previously, we always signalled a zero time since the last RTCP
SR, which is dubious.
The code also suggested that this would be the difference in
RTP NTP time units (32.32 fixed point), while it actually is
in in 1/65536 second units. (RFC 3550 section 6.4.1)
Signed-off-by: Martin Storsjö <martin@martin.st>
This brings back some code that was added originally in 4a6cc061
but never was used, and was removed as unused in 4cc843fa. The
code is updated to actually work and is tested to return sane
values.
Signed-off-by: Martin Storsjö <martin@martin.st>
The base_seq variable is set to first_seq - 1 (in
rtp_init_sequence), so no + 1 is needed here.
This avoids reporting 1 lost packet from the start.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '54cb096ee4558b3bfc28c2fcd6418ce82dc39fe1':
rtsp: Remove an outdated comment
rtsp: Remove references to weirdly named variables in other files
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'c44784c9bb9d0ddf5d39d0dfa640816a57b8f457':
rtp: Rename a static variable to normal naming conventions
rtp: Cosmetic cleanup
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The question can be answered: No, we do not know the initial sequence
number from the SDP. In certain cases, it can be known from the
RTP-Info response header in RTSP though. (In that case, we use it as
timestamp origin, but not for rtp receiver statistics.)
Signed-off-by: Martin Storsjö <martin@martin.st>
It is unclear what the bug exactly was and if it ever was fixed,
and we don't even support decoding via faad any longer. The
comment has been present since d0deedcb in 2006.
Signed-off-by: Martin Storsjö <martin@martin.st>
One of them is renamed now, but mentioning it by name serves
no purpose here. The other table mentioned ceased to exist
under that name in 4934884a1 in 2006.
Signed-off-by: Martin Storsjö <martin@martin.st>
Fixes division by zero
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Reviewed-by: Matthieu Bouron <matthieu.bouron@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '7b8c5b263bc680eff5710bee5994de39d47fc15e':
vc1dec: prevent a crash due missing pred_flag parameter
matroska: Fix use after free
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtpdec_vp8: Don't trim too much data from broken frames
rtpdec_vp8: Simplify code by using an existing helper function
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'ed79093222ceb42f0c3a39095a69af0b32be5450':
rtpdec: Add a terminating null byte at the end of the SDES/CNAME
yuv4mpeg: do not use deprecated functions
oggdec: fix faulty cleanup prototype
idcin: return 0 from idcin_read_packet() on success.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '7040e479a1530b2eda4b89a182d5eb50a77bd907':
idcin: allow seeking back to the first packet
idcin: set AV_PKT_FLAG_KEY for video packets with a palette
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'ccc0ffb1ba3fc1adb05a9f56dfc26131e61db3fb':
idcin: set start_time and packet duration instead of manually tracking pts.
idcin: set channel_layout
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '12c2530b1d87fa94f81ea97df575b77c825e6f4f':
idcin: fix check for presence of an audio stream
idcin: validate header parameters
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'c88d245c9866e48cb8a238b7564964c1fcf3315f':
au: use ff_raw_write_packet()
au: set stream start time and packet durations
Conflicts:
libavformat/au.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'af68a2baae6761044cbed95575e8bcfebf55c6f1':
au: use %u when printing id and channels since they are unsigned
au: validate sample rate
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'c837b38dd33a11c3810e988a60193a858eb4f58c':
au: move skipping of unused data to before parameter validation
au: do not arbitrarily limit channel count
Conflicts:
libavformat/au.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
