Use AVERROR_INVALIDDATA on invalid inputs, and AVERROR_EOF when no more
frames are available in an interleaved AVI.
Signed-off-by: Jindrich Makovicka <makovick@gmail.com>
Signed-off-by: Diego Biurrun <diego@biurrun.de>
The theoretical minimum for a (not totally well formed) RTCP packet
is 8 bytes, so we shouldn't require 12 bytes as minimum input.
Also return AVERROR_INVALIDDATA instead of 0 if something that is
not a proper packet is given.
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
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>
This is mostly useful for encryption together with the RTP muxer,
but could also be set up as IO towards the peer with the SDP
demuxer with custom IO.
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
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>
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>
Previously, for broken frames, we only returned the first partition
of the frame (we would append all the received packets to the packet
buffer, then set pkt->size to the size of the first partition, since
the rest of the frame could have lost data inbetween) - now instead
return the full buffered data we have, but don't append anything more
to the buffer after the lost packet discontinuity. Decoding the
truncated packet should hopefully get better quality than trimming out
everything after the first partition.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is required by RFC 3550 (section 6.5):
The list of items in each chunk MUST be terminated by one or more
null octets, the first of which is interpreted as an item type of
zero to denote the end of the list.
This was implicitly added as padding before, unless the host name
length matched up so no padding was added.
This makes wireshark parse the packets properly if other RTCP items
are appended to the same packet.
Signed-off-by: Martin Storsjö <martin@martin.st>
