then we can set the rtp read timeout instead of infinite timeout.
How to test(5s timeout):
./ffprobe -i rtp://192.168.1.67:1234?timeout=5000000
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
It avoids leaving dangling pointers behind in memory.
Also remove redundant checks for whether the URLContext to be closed is
already NULL.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Fixes a warning using musl:
In file included from libavformat/rtpproto.c:43:0:
/usr/local/musl/include/sys/poll.h:1:2: warning: #warning redirecting incorrect #include <sys/poll.h> to <poll.h>
Since all URLContexts have the same AVOptions, such AVOptions
will be applied on the outermost context only and removed from the
dict, while they probably make sense on all contexts.
This makes sure that rw_timeout gets propagated to the innermost
URLContext (to make sure it gets passed to the tcp protocol, when
opening a http connection for instance).
Alternatively, such matching options would be kept in the dict
and only removed after the ffurl_connect call.
Signed-off-by: Martin Storsjö <martin@martin.st>
We cannot play multiple multicast streams with the same port at the
same time. This is because both rtp and rtcp port are opened in
read-write mode, so they will not bind to the multicast address. Try
to make rtp port as read-only by default to solve this bug.
Signed-off-by: Zhao Zhili <wantlamy@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This commit also disables the async fate test, because it
used internal APIs in a non-kosher way, which no longer
exists.
* commit '2758cdedfb7ac61f8b5e4861f99218b6fd43491d':
lavf: reorganize URLProtocols
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Instead of a linked list constructed at av_register_all(), store them
in a constant array of pointers.
Since no registration is necessary now, this removes some global state
from lavf. This will also allow the urlprotocol layer caller to limit
the available protocols in a simple and flexible way in the following
commits.
* commit '8c9c5479c4ba729b4ba868ab541a90b2061a7c2f':
rtp: Add an option to set the send/receive buffer size
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '3c47e7c4350f73fc77d8e76f0dd6d2946b13c5cc':
rtp: Map the urloptions to AVOptions
Conflicts:
libavformat/rtpproto.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Only the upper 2 bits of the first byte are known to be
a fixed value.
The lower bits in the first byte of a RTP packet could be set
if the input is from another RTP packetizers than libavformat's,
but for RTCP packets, they would also be set when sending RTCP RR
packets, triggering false warnings about incorrect input format
to the protocol.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '29bc7bfba288ff8572ed967a8752a1dbde7b724b':
rtpproto: Write a warning if the input data written isn't RTP packetized
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Tell the user that the RTP muxer needs to be used to packetize
the data - using the RTP protocol on its own isn't enough.
Signed-off-by: Martin Storsjö <martin@martin.st>
By appending `?dscp=26` to the URL, IP packets will be classified as
AF31 (assured forwarding for multimedia flows with low probability of
loss). On congested network, this allows a user to assign priorities to
flows.
Signed-off-by: Vincent Bernat <vincent@bernat.im>
It appears this breaks build with MSVC
until someone who has MSVC setup has time to investigate and
workaround/fix this, its better to revert so that build is not broken
Thats even more so as the original commit only fixed a hypothetical issue
This reverts commit e587a428d7.
some video players on Android will not send udp hole punching messages if the rtcp port and rtp port are not two successive integers.
So, if the video player is behind NAT, it could not receive and rtp messages via udp
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '4b054a3400f728c54470ee6a1eefe1d82420f6a2':
rtpproto: Check the right feature detection macro
Merged-by: Michael Niedermayer <michaelni@gmx.at>
IPPROTO_IPV6 is unrelated here (it's only used in udp.c for
multicast sockopts), check for support for the sockaddr_in6
struct itself.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '6b58e11a8331690ec32e9869db89ae10c54614e9':
rtpproto: Add an option for writing return packets to the address of the last received packets
Merged-by: Michael Niedermayer <michaelni@gmx.at>
If we've received packets on the same socket before, the return
packets are sent to that address. If we've only received packets
on the other socket, try to guess the source port for the other
one assuming the basic +1/-1 logic.
Signed-off-by: Martin Storsjö <martin@martin.st>
Move the sources documentation up below the marker for deprecated
otpions. Also mention the new block parameter, that was added
in 749722209.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'ee37d5811caa8f4ad125a37fe6ce3f9e66cd72f2':
rtpproto: Allow specifying a separate rtcp port in ff_rtp_set_remote_url
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b7e6da988bfd5def40ccf3476eb8ce2f98a969a5':
rtpproto: Move rtpproto specific function declarations to a separate header
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '892b0be1dfbdeaf71235fb6c593286e4f5c7e4ec':
rtpproto: Simplify the rtp_read function by looping over the fds
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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>
