As long as there is enough contiguous data in the avio buffer,
just return a pointer to it instead of copying it to the caller
provided buffer.
Signed-off-by: Martin Storsjö <martin@martin.st>
A separate rtcp port can already be set when opening the rtp
protocol normally, but when doing port setup as in RTSP (where
we first need to open the local ports and pass them to the peer,
and only then receive the remote peer port numbers), we didn't
check the same url parameter as in the normal open routine.
Signed-off-by: Martin Storsjö <martin@martin.st>
I doubt that anyone ever would try to send a 1 byte packet
via the RTP protocol, but check just in case - it shouldn't
crash at least.
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows the chained demuxer (or more precisely, the lavf
utility code) to better fill in timestamps on packets from
these, especially for cases where one stream is a raw ADTS
stream.
Signed-off-by: Martin Storsjö <martin@martin.st>
Add support for domain names, for multiple source addresses,
for exclusions, and for session level specification of addresses.
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows us to explicitly fail if the caller tried to set
both inclusions and exclusions at the same time.
Signed-off-by: Martin Storsjö <martin@martin.st>
Previously this only allowed literal IP addresses. When these
are conveyed in a SDP file as in RFC4570, host names are allowed
as well.
Signed-off-by: Martin Storsjö <martin@martin.st>
Also parse segment durations as floating point, which is allowed
since HLS version 3.
This is based on a patch by Zhang Rui.
Signed-off-by: Martin Storsjö <martin@martin.st>
When first_timestamp was stored as-is, its actual time base
wasn't known later in the seek function.
Additionally, the logic (from 795d9594cf) for scaling it
based on stream_index is flawed - stream_index in the seek
function only specifies which stream the seek timestamp refers
to, but obviously doesn't say anything about which stream
first_timestamp belongs to.
In the cases where stream_index was >= 0 and all streams had the
same time base, this didn't matter in practice.
Seeking taking first_timestamp into account is problematic
when one variant is mpegts (with real timestamps) and one variant
is raw ADTS (with timestamps only being accumulated packet
duration), where the variants start at totally different timestamps.
Signed-off-by: Martin Storsjö <martin@martin.st>
Without the information, an application may choose audio from one
variant and video from another variant, which leads to fetching two
variants from the network. This enables av_find_best_stream() to find
matching audio and video streams, so that only one variant is fetched.
Signed-off-by: Martin Storsjö <martin@martin.st>
Also adjust the streams timestamps according to their start
timestamp when comparing. This helps getting correctly interleaved
packets if one stream lacks timestamps (such as a plain ADTS
stream when the other variants are full mpegts) when the others
have timestamps that don't start from zero.
This probably doesn't work properly if such a stream is
temporarily disabled (via the discard flags) and then reenabled,
and such streams are hard to correctly sync against the other
streams as well - but this works better than before at least.
The segment number restriction makes sure all variants advance
roughly at the same pace as well.
Signed-off-by: Martin Storsjö <martin@martin.st>
If passing the end of one segment while initializing the
chained demuxer, the parent demuxer's streams aren't set up
yet, so we can't recheck the discard flags.
Signed-off-by: Martin Storsjö <martin@martin.st>
This serves as a safeguard; normally we want to use the dts
comparison to interleave packets from all active variants. If that
dts comparison for some reason doesn't work as intended, make sure
that all packets in all variants for a certain sequence number have
been returned before moving on to the next one.
Signed-off-by: Martin Storsjö <martin@martin.st>
Incomplete crypted files would lead to a read after buffer boundary
otherwise.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Derived from VLC's http module.
Original authors:
Antoine Cellerier <dionoea@videolan.org>
Sébastien Escudier <sebastien-devel@celeos.eu>
Rémi Duraffort <ivoire@videolan.org>
Rémi Denis-Courmont <remi@remlab.net>
Francois Cartegnie <fcvlcdev@free.fr>
Normally, http servers shouldn't send this to us since we
don't advertise it with an Accept-Encoding header, but some
servers still do it anyway.
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes sure that values that are left-shifted by this constant
end up casted to 64 bit before shifting, avoiding overflow if the
value ends up larger than 2 GB.
Signed-off-by: Martin Storsjö <martin@martin.st>
This supports inclusion of one single IP address for now,
at the media level. Specifying the filter at the session level
(instead of at the media level), multiple source addresses,
exclusion, or using FQDNs instead of plain IP addresses is not
supported (yet at least).
Signed-off-by: Martin Storsjö <martin@martin.st>
If another peer is sending unicast packets to the same port that
we are listening on, those packets can end up being received despite
using source specific multicast. For those cases, manually check the
source address of received packets against the intended source address.
This only handles the case when the source list is one single IP
address for now, which probably is the most common case.
Based on a patch by Ed Torbett.
Signed-off-by: Martin Storsjö <martin@martin.st>
Blocking/exclusion is not supported yet.
The rtp protocol parameter takes the same form as the existing
sources parameter for the udp protocol.
Signed-off-by: Martin Storsjö <martin@martin.st>
Ben Avison
Diego Biurrun
Alexandra Khirnova
Gavriloaie Eugen-Andrei
Martin Storsjö
Michael Niedermayer
Ed Torbett
LYF
Luca Barbato
Zhang Rui
Hendrik Schreiber