Do this by allocating AVStream together with the data that is
currently in AVStreamInternal; or rather: Put AVStream at the
beginning of a new structure called FFStream (which encompasses
more than just the internal fields and is a proper context in its own
right, hence the name) and remove AVStreamInternal altogether.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Currently AVIOContext's private fields are all over AVIOContext.
This commit moves them into a new structure in avio_internal.h instead.
Said structure contains the public AVIOContext as its first element
in order to avoid having to allocate a separate AVIOContextInternal
which is costly for those use cases where one just wants to access
an already existing buffer via the AVIOContext-API.
For these cases ffio_init_context() can't fail and always returned zero,
which was typically not checked. Therefore it has been made to not
return anything.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Some encoders send GET_PARAMETER requests as a keep-alive mechanism.
If the client doesn't reply with an OK message, the encoder will close
the session. This was encountered with the impath i5110 encoder, when
the RTSP Keep-Alive checkbox is enabled under streaming settings.
Alternatively one may set the X-No-Keepalive: 1 header, but this is more
of a workaround. It's better practice to respond to an encoder's
keep-alive request, than disable the mechanism which may be manufacturer
specific.
Signed-off-by: Hayden Myers <hmyers@skylinenet.net>
Signed-off-by: Martin Storsjö <martin@martin.st>
Currently it is only checked that the rtp port does not exceed rtp_port_max.
Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
92c40ef882 added a listen_timeout option
for sdp. This allowed a user to set variable timeout which was
originally hard coded to 10 seconds.
The commit used the initial_timeout variable to store the value. But
this variable is shared with rtsp where it's used to infer a "listen"
mode. Thus, the timeout value could not be set in rtsp, and the default
value (initial_timeout = -1) would give 100ms timeout.
This was attempted to be fixed in c8101aabee,
which changed the meaning of initial_timeout = -1 to be an infinite
timeout. However, it did not address the issue that the timeout could
still not be set. Being able to set the timeout is useful because it
allows to automatically reconfigure from a udp to tcp connection in the
lower transport.
In this commit this is fixed by using the stimeout variable to
store the timeout value.
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
This is possible now that the next-API is gone.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Signed-off-by: James Almer <jamrial@gmail.com>
And forward it to the underlying UDP protocol.
Fixes ticket #7517.
Signed-off-by: Jiangjie Gao <gaojiangjie@live.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
rtsp.c uses a check of the form "if (CONFIG_RTSP_DEMUXER && ...) {}"
with the intent to make the code compilable even though the part guarded
by this check contains calls to functions that don't exist when the RTSP
demuxer is disabled. Yet even then compilers still need a declaration of
all the functions in the dead code block and error out if not (due to
our usage of -Werror=implicit-function-declaration) and no such
declaration exists for a static function in rtsp.c. Simply adding a
declaration leads to a "used but never defined" warning, therefore this
commit resorts to an #if.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Fixes#1941
Currently the media control uri is not correctly assigned when mpegts is
signalled in the media description.
The code checks whether at least one AVStream has been setup before
assigning to the media's uri. With mpegts the AVStreams are setup when
parsing packets and so the media's uri is skipped. This is fixed by
using rt->nb_rtsp_streams in the check which counts all medias in the
sdp.
Reviewed-by: Andriy Gelman <andriy.gelman@gmail.com>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
This can be used to receive the raw mpegts stream from a SAT>IP
server, by letting avformat handle the RTSP/RTP/UDP negotiation
and setup, but then simply passing the MP2T stream through
instead of demuxing it further.
For example, this command would demux/remux the mpegts stream:
SATIP_URL='satip://192.168.1.99:554/?src=1&freq=12188&pol=h&ro=0.35&msys=dvbs&mtype=qpsk&plts=off&sr=27500&fec=34&pids=0,17,18,167,136,47,71'
ffmpeg -i $SATIP_URL -map 0 -c copy -f mpegts -y remux.ts
Whereas this command will simply write out the raw stream, with
the original PAT/PMT/PIDs intact:
ffmpeg -rtsp_flags satip_raw -i $SATIP_URL -map 0 -c copy -f data -y raw.ts
Signed-off-by: Aman Karmani <aman@tmm1.net>
In sdp_read_header() some ff_network_close() calls were missed.
Also in rtp_read_header() update comment to explain why a single
call to ff_network_close() is enough to cover all cases even if
sdp_read_header() returns an error.
Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
In this error path ret still stores the number of bytes read in
ffurl_read().
Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
move comments for the size of SDP_MAX_SIZE here:
Some SDP lines, particularly for Realmedia or ASF RTSP streams,
contain long SDP lines containing complete ASF Headers (several
kB) or arrays of MDPR (RM stream descriptor) headers plus
"rulebooks" describing their properties. Therefore, the SDP line
buffer is large.
The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
in rtpdec_xiph.c.
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
Now the listen timeout is hardcoded(10s).
How to test(30s timeout):
./ffprobe -listen_timeout 30 -protocol_whitelist rtp,udp,file -i test.sdp
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
There is one general rtsp connection plus two connections per stream (rtp/rtcp).
Reviewed-by: Zhao Zhili <zhilizhao@tencent.com>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
1. Remove the assumption that the message method is TEARDOWN.
2. Don't ignore the error code of ff_rtsp_parse_streaming_commands.
Signed-off-by: Martin Storsjö <martin@martin.st>
In listen mode with UDP transport, once the sender has sent
the TEARDOWN and closed the connection, poll will indicate that
one can read from the connection (indicating that the socket has
reached EOF and should be closed by the receiver as well). In this
case, parse_rtsp_message won't try to parse the command (because
it's no longer in state STREAMING), but previously just returned
zero.
Prior to f6161fccf8, this caused
udp_read_packet to return zero, which is treated as EOF by
read_packet. But after that commit, udp_read_packet would continue
if parse_rtsp_message didn't return an explicit error code.
To keep the original behaviour from before that commit, more
explicitly return an error in parse_rtsp_message when in the wrong
state.
Fixes: #8840
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
In this example, the difference in length between the shortest and
longest string is three, so that not using pointers to strings saves
space even on 32bit systems.
Moreover, there is no need to use a sentinel here; it can be replaced
with FF_ARRAY_ELEMS.
Reviewed-by: Ross Nicholson <phunkyfish@gmail.com>
Reviewed-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Protocol options like buffer_size need to be passed to the
underlying transport implementation for udp multicasts as well.
Signed-off-by: Marton Balint <cus@passwd.hu>
This particular function is only required to return nonzero on
errors, but use the common AVERROR() pattern for consistency.
Signed-off-by: Martin Storsjö <martin@martin.st>
stimeout option is already used in tcp transport, since
http is based on tcp, pass the option to http for tunneling
case.
Reviewed-by: Steven Liu <lq@onvideo.cn>
Signed-off-by: Jun Li <junli1026@gmail.com>
stimeout option is already used in tcp transport, since
http is based on tcp, pass the option to http for tunneling
case.
Reviewed-by: Steven Liu <lq@onvideo.cn>
Signed-off-by: Jun Li <junli1026@gmail.com>
stimeout option is already used in tcp transport, since
http is based on tcp, pass the option to http for tunneling
case.
Reviewed-by: Steven Liu <lq@onvideo.cn>
Signed-off-by: Jun Li <junli1026@gmail.com>
Fix ticket #7297
The current setting for send-expect-100 option is either
enabled if applicable or forced enabled, no option to force
disable the header. This change is to expand the option setting
to provide more flexibility, which is useful for rstp case.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
