Commit graph

168 commits

Author SHA1 Message Date
Sebastian Dröge
4ec17b1975 rtsp-stream: Set multicast TTL on the multicast sockets
And not if we do unicast UDP.

https://bugzilla.gnome.org/show_bug.cgi?id=791743
2017-12-19 11:34:37 +02:00
Sebastian Dröge
4d86f99449 rtsp-stream: Decide based on the sockets, not the addresses if we already allocated a socket
In the multicast case (as in test-multicast, not test-multicast2), the
address could be allocated/reserved (and thus set) already without
allocating the actual socket. We need to allocate the socket here still
instead of just claiming that it was already allocated.

See https://bugzilla.gnome.org/show_bug.cgi?id=791743#c2
2017-12-19 11:16:51 +02:00
Edward Hervey
64a46d47ba rtsp-server: Minor doc fixes
Mostly for g-i
2017-12-07 16:08:50 +01:00
Patricia Muscalu
caa3f1caac rtsp-stream: Do not reset 'blocking' if stream is already blocked
https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-27 07:58:42 +01:00
Edward Hervey
9514f2d354 rtsp-media: Enable seeking query before pipeline is complete
SDP are now provided *before* the pipeline is fully complete. In order
to know whether a media is seekable or not therefore requires asking
the invididual streams.

API: gst_rtsp_stream_seekable

https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-25 07:53:11 +01:00
Edward Hervey
4d98bc5e55 Run gst-indent 2017-11-21 09:53:08 +01:00
Sebastian Dröge
d51f8abe56 rtsp-stream: Only update the RTP udpsink if it actually exists
For send-only streams it does not exist, but the RTCP udpsink might.
2017-11-15 19:56:26 +02:00
Patricia Muscalu
efdb795c86 rtsp-media: seek on media pipelines that are complete
Make sure that a seek is performed on pipelines that
contain at least one sink element.

Change-Id: Icf398e10add3191d104b1289de612412da326819

https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-11-15 19:56:26 +02:00
Patricia Muscalu
a7732a68e8 Dynamically reconfigure pipeline in PLAY based on transports
The initial pipeline does not contain specific transport
elements. The receiver and the sender parts are added
after PLAY.
If the media is shared, the streams are dynamically
reconfigured after each PLAY.

https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-11-15 19:56:15 +02:00
Patricia Muscalu
930a602e17 rtsp-stream: obtain stream position from pad
If no sinks have been added yet, obtain the current and
the stop position of the stream from the send_src pad.

Change-Id: Iacd4ab4bdc69f6b49370d06012880ce48a7d595a

https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-11-15 17:46:00 +02:00
Patricia Muscalu
51d670f73b rtsp-stream: add functions to get rtp and rtcp multicast sockets
Change-Id: Iddfe6e0bd250cb0159096d5eba9e4202d22b56db

https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-11-15 17:46:00 +02:00
Patricia Muscalu
c9605cc5e1 stream: set async=sync=false only for RTCP appsink
Change-Id: I929a218a9adf4759f61322b6f2063aacc5595f90

https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-11-15 17:46:00 +02:00
Thibault Saunier
8b38aa9c46 stream: Use stream duration as stream-stop if segment was not configured with a stop
Allowing client to know stream duration when no seeking happened.

https://bugzilla.gnome.org/show_bug.cgi?id=783435
2017-10-05 12:07:13 -03:00
Julien Isorce
d72284bdf8 rtsp-stream: fix connection delay due to wrong assumption on last-sample
Commit 852cc09f54 assumed that
multiudpsink's last-sample always comes from the payloader. Which
is wrong if auxiliary streams are multiplexed in the same stream.

So check the buffer's ssrc against the caps'ssrc before to use its
seqnum. If not the same ssrc just use the payloader as done prior
the commit above or when there is no last-sample yet.

https://bugzilla.gnome.org/show_bug.cgi?id=784094
2017-06-29 14:52:09 +01:00
Thibault Saunier
b56930704f gi: Fix some annotations and docstrings 2017-04-13 14:20:10 -03:00
Patricia Muscalu
fb7833245d rtsp-stream: corrected if-statement in _get_server_port()
This bug was accidentally introduced while fixing a segfault
in _get_server_port() function.

https://bugzilla.gnome.org/show_bug.cgi?id=776345
2017-01-10 10:38:13 +00:00
Patricia Muscalu
f47e6ab9f6 rtsp-stream: fixed segmenation fault in _get_server_port()
Calling function gst_rtsp_stream_get_server_port() results in
segmenation fault in the RTP/RTSP/TCP case.
Port that the server will use to receive RTCP makes only
sense in the UDP case, however the function should handle
the TCP case in a nicer way.

https://bugzilla.gnome.org/show_bug.cgi?id=776345
2017-01-09 15:27:40 +02:00
Patricia Muscalu
42f270e7f2 rtsp-stream: Fixed TCP transport case
Make sure that the appsink element is actually added to
the bin before trying to link it with the elements in it.

https://bugzilla.gnome.org/show_bug.cgi?id=776343
2016-12-22 14:21:54 +02:00
Matthew Waters
b38eb8e99e stream: block the output of rtpbin instead of the source pipeline
85c52e194b introduced a more correct
detection of the srtp rollover counter to add to the SDP.

Unfortunately, it was incomplete for live pipelines where the logic
blocks the source bin before creating the SDP and thus would never have
the necessary informaiton to create a correct SDP with srtp encryption.

Move the pad blocks to rtpbin's output pads instead so that the
necessary information can be created before we need the information for
the SDP.

https://bugzilla.gnome.org/show_bug.cgi?id=770239
2016-11-23 23:08:16 +11:00
Göran Jönsson
335d279a96 rtsp-stream: Set close-socket FALSE on UDP src:es
With this RTSP server can use the sockets independent on the udpsrc
state.
When the udp src is finalized it will unref socket and when g_socket
is finalized the socket will be closed.

https://bugzilla.gnome.org/show_bug.cgi?id=765673
2016-11-22 13:59:30 +02:00
Xavier Claessens
c0f24fea83 stream: Fix randomly missing streams from SDP with dynamic elements
When using dynamic elements, gst_rtsp_stream_join_bin() is called from
"pad-added" signal. In that case priv->srcpad could already have its caps,
and they'll be sent to priv->send_src[0] pad. That means that when it
connects "notify::caps" signal, that pad could already have received its
caps and the signal won't be emitted anymore.

In that case priv->caps stay to NULL and when building the SDP that stream
gets ignored. Leading to missing video or audio when playing in client side.

https://bugzilla.gnome.org/show_bug.cgi?id=772478
2016-10-06 19:05:36 +03:00
Ian Jamison
34389831cb rtsp-server: Hint that set_multicast_iface expects the name of the interface
To prevent any possibly confusion with IPs or anything else.

https://bugzilla.gnome.org/show_bug.cgi?id=771530
2016-09-18 10:00:29 -04:00
Sebastian Dröge
74c8a9f4cf rtsp-stream: Remove unused _locked() variant of a function
It was added during refactoring.
2016-09-07 18:44:34 +03:00
Xavier Claessens
e882fe9e06 stream: cosmetic cleanup
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-07 18:40:57 +03:00
Xavier Claessens
f5f350645a stream: Compare IP addresses case insensitive in more places
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-07 18:40:57 +03:00
Xavier Claessens
f90ab92547 stream: Fix leaked joined_bin
There is no need to keep a strong ref on it, and _leave_bin() was
setting it to NULL before calling g_clear_object() so it was leaked.

https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-07 18:40:57 +03:00
Sebastian Dröge
d33eca6156 rtsp-stream: Compare IP address strings case insensitive
Otherwise IPv6 addresses might fail this comparision.
2016-09-06 19:15:23 +03:00
Sebastian Dröge
e5a49efa7f rtsp-stream: Bind multicast sockets to ANY as before
https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
2016-09-06 19:10:21 +03:00
Sebastian Dröge
ca855abae1 rtsp-stream: Always create multicast UDP elements if the protocol flag is set
Adding them later will cause deadlocks due to
1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
2) adding the multicast sink
3) waiting for it to get data to preroll again

3) never happens because the queues after the tee are full.
2016-09-05 18:09:22 +03:00
Sebastian Dröge
be4b9718e3 rtsp-stream: Fix up various multicast related issues 2016-09-05 16:32:57 +03:00
Xavier Claessens
8495c47a9d stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
This is basically reverting changes introduced in commit f62a9a7,
because it was introducing various regressions:

- It introduces a leak of udpsrc elements that got wrongly fixed by adding
  an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
  ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
- If a mcast client connects, it creates a new socket in SETUP to try to respect
  the destination/port given by the client in the transport, and overrides the
  socket already set on the udpsink element. That means that if we already had a
  client connected, the source address on the udp packets it receives suddenly
  changes.
- If a 2nd mcast client connects, the destination/port in its transport is
  ignored but its transport wasn't updated.

What this patch does:

- Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
- Always have a tee+queue when udp is enabled. This could be optimized
  again in a later patch, but is more complicated. If no unicast clients
  connects then those elements are useless, this could be also optimized
  in a later patch.
- When mcast transport is added, it creates a new set of udpsrc/udpsink,
  seperated from those for unicast clients. Since we already support only
  one mcast address, we also create only one set of elements.

https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-05 13:36:17 +03:00
Xavier Claessens
aa0e60445d stream: factor our plug_src function
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-05 13:26:08 +03:00
Xavier Claessens
47a3956b48 stream: factor out plug_sink function
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-05 13:26:02 +03:00
Xavier Claessens
a44f198ffc stream: small documentation clarification
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-05 13:25:57 +03:00
Xavier Claessens
82a618c2e6 stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-05 13:25:51 +03:00
Xavier Claessens
55a1df5724 stream: Keep a ref on joined bin
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-05 13:25:39 +03:00
Xavier Claessens
3ff4529a92 stream: code cleanup
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-05 13:24:06 +03:00
Xavier Claessens
2b223af792 stream: small fix in error code path
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-05 13:24:01 +03:00
Xavier Claessens
07f17c2cce Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
This partly reverts commit cba045e1b1,
but keeps unit tests.

https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-05 13:23:53 +03:00
Aleix Conchillo Flaqué
85c52e194b sdp: add rollover counters for all sender SSRC
We add different crypto sessions in MIKEY, one for each sender
SSRC. Currently, all of them will have the same security policy, 0.

The rollover counters are obtained from the srtpenc element using the
"stats" property.

https://bugzilla.gnome.org/show_bug.cgi?id=730539
2016-06-14 11:14:48 +02:00
Jake Foytik
fe5f8077c1 rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
- Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
 - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
 - Create unit test for shared media.

https://bugzilla.gnome.org/show_bug.cgi?id=764744
2016-04-29 11:49:14 +03:00
Sebastian Dröge
aa9a2443a1 rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
For IPv6 addresses, binding to a multicast group does not work on Linux
either. Always bind to ANY and then later join the multicast group.

https://bugzilla.gnome.org/show_bug.cgi?id=764679
2016-04-29 11:48:57 +03:00
Sebastian Dröge
9fab555cc5 rtsp-server: Implement clock signalling according to RFC7273
For NTP and PTP clocks we signal the actual clock that is used and signal
the direct media clock offset.

For all other clocks we at least signal that it's the local sender clock.

This allows receivers to know which clock was used to generate the media and
its RTP timestamps. Receivers can then implement network synchronization,
either absolute or at least relative by getting the sender clock rate directly
via NTP/PTP instead of estimating it from RTP timestamps and packet receive
times.

https://bugzilla.gnome.org/show_bug.cgi?id=760005
2016-04-03 11:22:31 +03:00
Sebastian Dröge
69d04f3838 rtsp-media: Add support for setting the multicast interface
https://bugzilla.gnome.org/show_bug.cgi?id=763000
2016-03-25 12:52:12 +02:00
Sebastian Dröge
8e72e69eec rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
This would get us NO_PREROLL in the bin again and break seeking.
Thanks to Carlos Rafael Giani for helping to debug this!

https://bugzilla.gnome.org/show_bug.cgi?id=740509
2016-03-16 23:36:30 +02:00
Sebastian Dröge
8b68edd138 rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
Without this, RECORD pipelines are broken because
a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
added later. Previously it was there earlier and due to NO_PREROLL caused the
pipeline to preroll immediately
b) the udpsrc for the pipeline is added later and never set to PLAYING state,
as the corresponding code previously was only for PLAY pipelines.

https://bugzilla.gnome.org/show_bug.cgi?id=763281
2016-03-10 19:47:13 +02:00
Jan Schmidt
4a6f63ad03 rtsp-stream: Fix typo in the docstring
gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
2016-03-11 01:23:15 +11:00
Sebastian Dröge
206d2ded09 rtsp-stream: Disable multicast loopback for all our sockets
On Windows this is a receiver-side setting, on Linux a sender-side setting. As
we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
loopback setting on the socket... while udpsink does which unfortunately has
no effect here on Windows but on Linux.

https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-03-05 10:53:15 +02:00
Sebastian Dröge
9794822549 rtsp-stream: Only bind multicast sockets to ANY on Windows
On Linux it is still needed to bind to the multicast address
to filter out random other packets, while on Windows binding
to multicast addresses just fails.
2016-03-04 13:51:12 +02:00
Sebastian Dröge
a7ced98346 rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
Otherwise we fail to allocate UDP ports if the pool only contains multicast
addresses, which is something that used to work before. For unicast addresses
if the pool contains none, we just allocate them as if there is no pool at
all.

https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-03-03 10:43:13 +02:00