Commit graph

1706 commits

Author SHA1 Message Date
Sebastian Dröge
d06f3af0be Revert "rtsp-stream: avoid duplicating the first multicast client"
This reverts commit 3357094440.

Commits where accidentially squashed together
2018-08-14 14:25:53 +03:00
Sebastian Dröge
443c2b73e5 Revert "Add new API for setting/getting maximum multicast ttl value"
This reverts commit 7f0ae77e40.

Commits where accidentially squashed together
2018-08-14 14:25:42 +03:00
Sebastian Dröge
17335e9906 Revert "rtsp-stream: Don't require address pool in the transport specific case"
This reverts commit a9db3e7f09.

Commits where accidentially squashed together
2018-08-14 14:25:37 +03:00
Sebastian Dröge
29ae15f6f1 Revert "stream: Choose the maximum ttl value provided by multicast clients"
This reverts commit 499e437e50.

Commits where accidentially squashed together
2018-08-14 14:25:14 +03:00
Sebastian Dröge
98b480fae9 examples: Fix indentation 2018-08-14 14:10:56 +03:00
Patricia Muscalu
499e437e50 stream: Choose the maximum ttl value provided by multicast clients
The maximum ttl value provided so far by the multicast clients
will be chosen and reported in the response to the current
client request.

https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-08-14 14:10:41 +03:00
Patricia Muscalu
a9db3e7f09 rtsp-stream: Don't require address pool in the transport specific case
If "transport.client-settings" parameter is set to true, the client is
allowed to specify destination, ports and ttl.
There is no need for pre-configured address pool.

https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-08-14 14:10:23 +03:00
Patricia Muscalu
7f0ae77e40 Add new API for setting/getting maximum multicast ttl value
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-08-14 14:10:20 +03:00
Mathieu Duponchelle
3357094440 rtsp-stream: avoid duplicating the first multicast client
In dcb4533fed , we made it so
clients were dynamically added and removed to the multicast
udp sinks, as such we should no longer add a first client in
set_multicast_socket_for_udpsink

https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-08-14 14:10:02 +03:00
Thibault Saunier
20dc49749c rtsp-server: Add gstreamer-base gir dir in autotools 2018-08-06 15:33:04 -04:00
Mathieu Duponchelle
12f8abb549 rtsp-client: always allocate both IPV4 and IPV6 sockets
multiudpsink does not support setting the socket* properties
after it has started, which meant that rtsp-server could no
longer serve on both IPV4 and IPV6 sockets since the patches
from https://bugzilla.gnome.org/show_bug.cgi?id=757488 were
merged.

When first connecting an IPV6 client then an IPV4 client,
multiudpsink fell back to using the IPV6 socket.

When first connecting an IPV4 client, then an IPV6 client,
multiudpsink errored out, released the IPV4 socket, then
crashed when trying to send a message on NULL nevertheless,
that is however a separate issue.

This could probably be fixed by handling the setting of
sockets in multiudpsink after it has started, that will
however be a much more significant effort.

For now, this commit simply partially reverts the behaviour
of rtsp-stream: it will continue to only create the udpsinks
when needed, as was the case since the patches were merged,
it will however when creating them, always allocate both
sockets and set them on the sink before it starts, as was
the case prior to the patches.

Transport configuration will only error out if the allocation
of UDP sockets fails for the actual client's family, this
also downgrades the GST_ERRORs in alloc_ports_one_family
to GST_WARNINGs, as failing to allocate is no longer
necessarily fatal.

https://bugzilla.gnome.org/show_bug.cgi?id=796875
2018-08-01 20:42:34 +02:00
Nirbheek Chauhan
3baf94ed55 meson: Convert common options to feature options
These are necessary for gst-build to set options correctly. The
remaining automagic option is cgroup support in examples.

https://bugzilla.gnome.org/show_bug.cgi?id=795107
2018-07-27 18:48:46 +05:30
Sebastian Dröge
37e75cb8ea rtsp-stream: Slightly simplify locking 2018-07-23 18:03:51 +03:00
David Svensson Fors
12169f1e84 Limit queued TCP data messages to one per stream
Before, the watch backlog size in GstRTSPClient was changed
dynamically between unlimited and a fixed size, trying to avoid both
unlimited memory usage and deadlocks while waiting for place in the
queue. (Some of the deadlocks were described in a long comment in
handle_request().)

In the previous commit, we changed to a fixed backlog size of 100.
This is possible, because we now handle RTP/RTCP data messages differently
from RTSP request/response messages.

The data messages are messages tunneled over TCP. We allow at most one
queued data message per stream in GstRTSPClient at a time, and
successfully sent data messages are acked by sending a "message-sent"
callback from the GstStreamTransport. Until that ack comes, the
GstRTSPStream does not call pull_sample() on its appsink, and
therefore the streaming thread in the pipeline will not be blocked
inside GstRTSPClient, waiting for a place in the queue.

pull_sample() is called when we have both an ack and a "new-sample"
signal from the appsink. Then, we know there is a buffer to write.

RTSP request/response messages are not acked in the same way as data
messages. The rest of the 100 places in the queue are used for
them. If the queue becomes full of request/response messages, we
return an error and close the connection to the client.

Change-Id: I275310bc90a219ceb2473c098261acc78be84c97
2018-07-23 17:45:00 +03:00
David Svensson Fors
287345f6ac rtsp-client: Use fixed backlog size
Change to using a fixed backlog size WATCH_BACKLOG_SIZE.

Preparation for the next commit, which changes to a different way of
avoiding both deadlocks and unlimited memory usage with the watch
backlog.
2018-07-23 17:44:15 +03:00
Carlos Rafael Giani
12c2dd6e1c rtsp-media: unref clock (if set) when finalizing
https://bugzilla.gnome.org/show_bug.cgi?id=796814
2018-07-16 23:56:09 +01:00
Carlos Rafael Giani
e99471a56b rtsp-media: add gst_rtsp_media_*_set_clock to docs
https://bugzilla.gnome.org/show_bug.cgi?id=796814
2018-07-16 23:55:37 +01:00
Tim-Philipp Müller
7f7a210b84 media-factory: unref old clock when setting new clock
https://bugzilla.gnome.org/show_bug.cgi?id=796724
2018-07-12 19:02:40 +01:00
Brendan Shanks
f304096994 media-factory: unref clock in finalize
https://bugzilla.gnome.org/show_bug.cgi?id=796724
2018-07-12 19:02:40 +01:00
Tim-Philipp Müller
1cd6c0340e rtsp-onvif-media: fix g-ir-scanner warnings 2018-07-12 19:02:40 +01:00
Tim-Philipp Müller
318525570f .gitignore: add another example binary 2018-07-10 23:56:23 +01:00
Tim-Philipp Müller
c2e3aa49f8 meson: add new test-appsrc2 example to meson build 2018-07-10 23:55:20 +01:00
Tim-Philipp Müller
35e284a946 examples: fix build of new test-appsrc2 example
Need to link against libgstapp-1.0.
2018-07-10 23:53:41 +01:00
Jan Schmidt
b6c3960f11 examples: Add test-appsrc2
Add an example of feeding both audio and video into an RTSP
pipeline via appsrc.
2018-07-11 01:25:51 +10:00
Louis-Francis Ratté-Boulianne
604240f7eb client: Strip transport parts as whitespaces could be around commas
https://bugzilla.gnome.org/show_bug.cgi?id=758428
2018-07-06 16:13:33 -04:00
Göran Jönsson
c1fab570d8 rtsp-stream: avoid pushing data on unlinked udpsrc pad during setup
Fix race when setting up source elements.

Since we set the source element(s) to PLAYING state before hooking
them up to the downstream funnel, it's possible for the source element
to receive packets before we actually get to linking it to the funnel,
in which case buffers would be pushed out on an unlinked pad, causing
it to error out and stop receiving more data.

We fix this by blocking the source's srcpad until we have linked it.

https://bugzilla.gnome.org/show_bug.cgi?id=796160
2018-06-27 12:25:45 +02:00
Ognyan Tonchev
f110016ac6 rtsp-stream: Fix mismatch between allowed and configured protocols
https://bugzilla.gnome.org/show_bug.cgi?id=796679
2018-06-26 15:41:07 +02:00
Ulf Olsson
4d25e04bd7 rtsp-stream: Emit a signal when the SRTP decoder is created
https://bugzilla.gnome.org/show_bug.cgi?id=778080
2018-06-26 15:38:33 +02:00
Patricia Muscalu
4007050335 rtsp-stream: Don't require presence of sinks in _get_*_socket()
Transport specific sink elements are added to the pipeline
in PLAY request and sockets are already created in SETUP so
it's actually wrong to require the presence of sinks in
_get_*_socket() functions.

https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-06-26 14:01:02 +02:00
Patricia Muscalu
dcb4533fed rtsp-stream: Update transport for multicast clients as well
If a multicast client requests different transport settings
than the existing one make sure that this new transport
configuruation is propagated to the multicast udp sink.

https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-06-26 11:08:45 +02:00
Patricia Muscalu
1a38de2b17 rtsp-stream: Set the multicast TTL parameter on multicast udp sinks
And not on unicast udp sinks

https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-06-26 10:59:25 +02:00
Tim-Philipp Müller
2eb4d1b810 Update for g_type_class_add_private() deprecation in recent GLib 2018-06-24 12:48:11 +02:00
Tim-Philipp Müller
e82ba1e52f Fix indentation 2018-06-24 12:45:49 +02:00
Jan Schmidt
e0f31b4ca4 examples: Add test-video-disconnect example
Simple example which cuts off all clients 10 seconds
after the first one connects.
2018-06-22 23:17:53 +10:00
Mathieu Duponchelle
5ede2a5c5c rtsp-auth: Add support for parsing .htdigest files
Passwords are usually not stored in clear text, but instead
stored already hashed in a .htdigest file.

Add support for parsing such files, add API to allow setting
a custom realm in RTSPAuth, and update the digest example.

https://bugzilla.gnome.org/show_bug.cgi?id=796637
2018-06-21 15:47:39 +02:00
Matthew Waters
b5a61f488d rtspclientsink: fix waiting for multiple streams
We were previously only ever waiting for a single stream to notify it's
blocked status through GstRTSPStreamBlocking.  Actually count streams to
wait for.

Fixes rtspclientsink sending SDP's without out some of the input
streams.

https://bugzilla.gnome.org/show_bug.cgi?id=796624
2018-06-21 20:56:46 +10:00
Mathieu Duponchelle
ea67b932b3 docs: add missing auth methods 2018-06-20 04:30:04 +02:00
Mathieu Duponchelle
3b70c68e6e rtsp-stream: only create funnel if it didn't exist already.
This precented using multiple protocols for the same stream.

https://bugzilla.gnome.org/show_bug.cgi?id=796634
2018-06-20 01:36:57 +02:00
Mathieu Duponchelle
36cc79c5e2 meson: build auth-digest example 2018-06-20 01:35:47 +02:00
Patricia Muscalu
768fb5695c Get payloader stats only for the sending streams
Get/set payloader properties only for streams that actually
contain a payloader element.

https://bugzilla.gnome.org/show_bug.cgi?id=796523
2018-06-13 10:13:12 +03:00
Edward Hervey
89e6ee73b1 Makefile: Don't hardcode libtool for g-i build
Similar to the other commits in core/base/bad
2018-05-18 14:54:46 +02:00
Johan Bjäreholt
913eae2e7e rtsp-onvif-media-factory: export gst_rtsp_onvif_media_factory_requires_backchannel
https://bugzilla.gnome.org/show_bug.cgi?id=796229
2018-05-18 08:57:28 +01:00
Jan Schmidt
b3a4df7ab8 rtspclientsink: Don't deadlock in preroll on early close
If the connection is closed very early, the flushing
marker might not get set and rtspclientsink can get
deadlocked waiting for preroll forever.

https://bugzilla.gnome.org/show_bug.cgi?id=786961
2018-05-09 04:09:02 +10:00
Nirbheek Chauhan
208258472c meson: Update option names to omit disable_ and with- prefixes
Also yield common options to the outer project (gst-build in our case)
so that they don't have to be set manually.
2018-05-05 20:31:43 +05:30
Tim-Philipp Müller
858a9d2054 meson: use -Wl,-Bsymbolic-functions where supported
Just like the autotools build.
2018-04-25 11:00:32 +01:00
Tim-Philipp Müller
90d006141a configure: check for -good and -bad plugins only in uninstalled setup
Avoids confusing configure messages looking or a -good .pc file
that doesn't exist.

Also use plugindir variables that common macros set while at it.

https://bugzilla.gnome.org/show_bug.cgi?id=795466
2018-04-22 20:10:15 +01:00
Joakim Johansson
808b49cbfc rtsp-client: Fix session timeout
When streaming data over TCP then is not the keep-alive
functionality working.

The reason is that the function do_send_data have changed
to boolean but the code is still checking the received result
from send_func with GST_RTSP_OK.

The result is that a successful send_func will always lead to
that do_send_data is returning false and the keep-alive will
not be updated.

https://bugzilla.gnome.org/show_bug.cgi?id=795321
2018-04-20 10:13:53 +03:00
Mathieu Duponchelle
bfc35ae1ae Implement support for ULP Forward Error Correction
In this initial commit, interface is only exposed for RECORD,
further work will be needed in rtspsrc to support this for
PLAY.

https://bugzilla.gnome.org/show_bug.cgi?id=794911
2018-04-19 18:25:31 +02:00
Sebastian Dröge
9f5d3ee7a8 Revert "rtsp-server: Switch around sendonly/recvonly attributes"
This reverts commit 3d275b1345.

While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of
the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say
the opposite, just like the ONVIF standard.

Let's follow those RFCs as we're doing RTSP here, and add a property at
a later time if needed to switch to the SDP RFC behaviour.

https://bugzilla.gnome.org/show_bug.cgi?id=793964
2018-04-17 17:50:05 +03:00
Tim-Philipp Müller
d49edc5582 Automatic update of common submodule
From 3fa2c9e to ed78bee
2018-04-16 10:53:52 +01:00