Commit graph

1072 commits

Author SHA1 Message Date
Branko Subasic
421ac85150 rtsp-media: allow specifying rate when seeking
Add new function gst_rtsp_media_seek_full_with_rate() which allows the
caller to specify the rate for the seek. Also added functions in
rtsp-stream and rtsp-media for retreiving current rate and applied rate.

https://bugzilla.gnome.org/show_bug.cgi?id=754575
2019-06-04 14:32:51 +02:00
Thibault Saunier
f4b20b010d doc: Fix some docstrings 2019-05-13 17:00:00 -04:00
Thibault Saunier
5b039416db docs: Port to hotdoc 2019-05-13 11:38:39 -04:00
Sebastian Dröge
abf6be1d7a rtsp-server: Fix various Since markers 2019-04-23 15:09:34 +03:00
Sebastian Dröge
8d3bef4c1e rtsp-server: Add various Since: 1.14 markers 2019-04-23 15:01:32 +03:00
Sebastian Dröge
802a648723 rtsp-server: Add various missing Since: 1.16 markers 2019-04-23 14:56:42 +03:00
Kristofer Bjorkstrom
b578628dc1 rtsp-client: Handle Content-Length limitation
Add functionality to limit the Content-Length.
API addition, Enhancement.

Define an appropriate request size limit and reject requests
exceeding the limit with response status 413 Request Entity Too Large

Related to !182
2019-04-22 09:17:13 +00:00
Göran Jönsson
3cfe88632f rtsp_server: Free thread pool before clean transport cache
If not waiting for free thread pool before clean transport caches, there
can be a crash if a thread is executing in transport list loop in
function send_tcp_message.

Also add a check if priv->send_pool in on_message_sent to avoid that a
new thread is pushed during wait of free thread pool. This is possible
since when waiting for free thread pool mutex have to be unlocked.
2019-04-11 08:02:52 +02:00
Ulf Olsson
d09b7c8e4f rtsp-stream: Add support for GCM (RFC 7714)
Follow-up to !198
2019-04-10 08:43:29 +00:00
Erlend Eriksen
a9d3bcc03e session pool: fix missing klass-> in klass->create_session 2019-03-28 00:27:37 +01:00
Göran Jönsson
1fd49d36d1 rtsp-media: Handle set state when preparing.
Handle the situation when  a call to gst_rtsp_media_set_state is done
when media status is preparing.

Also add unit test for this scenario.

The unit test simulate on a media level when two clients share a (live)
media.
Both clients have done SETUP and got responses. Now client 1 is doing
play and client 2 is just closing the connection.

Then without patch there are a problem when
client1 is calling gst_rtsp_media_unsuspend in handle_play_request.
And client2 is doing closing connection we can end up in a call
to gst_rtsp_media_set_state when
priv->status == GST_RTSP_MEDIA_STATUS_PREPARING and all the logic for
shut down media is jumped over .

With this patch and this scenario we wait until
priv->status == GST_RTSP_MEDIA_STATUS_PREPARED and then continue to
execute after that and now we will execute the logic for
shut down media.
2019-03-20 12:26:50 +01:00
Göran Jönsson
7e01dfd151 rtsp-media: Fix multicast use case with common media
Use case
client 1: SETUP
client 1: PLAY
client 2: SETUP
client 1: TEARDOWN
client 2: PLAY
client 2: TEARDOWN
2019-02-19 12:12:34 +01:00
Göran Jönsson
afb27f91cf rtsp-server: remove recursive behavior
Introduce a threadpool to send rtp and rtcp to avoid recursive behavior.
2019-02-02 10:42:33 +00:00
Sebastian Dröge
4be7424de5 rtsp-client: Only allow to set either a send_func or send_messages_func but not both
And route all messages through the send_func if no send_messages_func
was provided.

We otherwise break backwards compatibility.
2019-01-30 14:40:09 +02:00
Sebastian Dröge
c372643e1e rtsp-client: Add support for sending buffer lists directly
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2019-01-30 14:40:09 +02:00
Sebastian Dröge
d708f9736b rtsp-server: Add support for buffer lists
This adds new functions for passing buffer lists through the different
layers without breaking API/ABI, and enables the appsink to actually
provide buffer lists.

This should already reduce CPU usage and potentially context switches a
bit by passing a whole buffer list from the appsink instead of
individual buffers. As a next step it would be necessary to
  a) Add support for a vector of data for the GstRTSPMessage body
  b) Add support for sending multiple messages at once to the
    GstRTSPWatch and let it be handled internally
  c) Adding API to GOutputStream that works like writev()

Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2019-01-30 14:39:50 +02:00
Benjamin Berg
621e140a8e client: Fix crash in close handler
The close handler could trigger a crash because it invalidated the
watch_context while still leaving a source attached to it which would be
cleaned up at a later point.
2019-01-29 18:34:08 +00:00
Edward Hervey
a48711fa9d rtsp-stream: Use cached address when allocating sockets
If an address/port was previously decided upon (ex: multicast in the
SDP), then use that instead of re-creating another one

Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/57
2019-01-29 14:42:35 +01:00
Lars Wiréen
ae32203cb0 rtsp-media: Fix race codition in finish_unprepare
The previous fix for race condition around finish_unprepare where the
function could be called twice assumed that the status wouldn't change
during execution of the function. This assumption is incorrect as the
state may change, for example if an error message arrives from the
pipeline bus.

Instead a flag keeping track on whether the finish_unprepare function
is currently executing is introduced and checked.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/59
2019-01-25 12:44:23 +00:00
Patricia Muscalu
3be1b9bba8 Add source elements to the pipeline before activation
In plug_src we changed the element state before adding it to
the owner container. This prevented the pipeline from intercepting
a GST_STREAM_STATUS_TYPE_CREATE message from the pad in order
to assign a custom task pool.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/53
2018-12-06 08:59:04 +00:00
Patricia Muscalu
4370f3a901 rtsp-media: Update priv->blocked when linked streams are unblocked.
Media is considered to be blocked when all streams that belong to
that media are blocked.
This patch solves the problem of inconsistent updates of
priv->blocked that are not synchronized with the media state.
2018-11-19 11:35:26 +00:00
Patricia Muscalu
146b3da174 rtsp-media: Don't block streams before seeking
Before the seek operation is performed on media, it's required that
its pipeline is prepared <=> the pipeline is in the PAUSED state.
At this stage, all transport parts (transport sinks) have been successfully
added to the pipeline and there is no need for blocking the streams.
2018-11-19 11:35:26 +00:00
Linus Svensson
185385924d rtsp-stream: Use seqnum-offset for rtpinfo
The sequence number in the rtpinfo is supposed to be the first RTP
sequence number. The "seqnum" property on a payloader is supposed to be
the number from the last processed RTP packet. The sequence number for
payloaders that inherit gstrtpbasepayload will not be correct in case of
buffer lists. In order to fix the seqnum property on the payloaders
gst-rtsp-server must get the sequence number for rtpinfo elsewhere and
"seqnum-offset" from the "stats" property contains the value of the
very first RTP packet in a stream. The server will, however, try to look
at the last simple in the sink element and only use properties on the
payloader in case there no sink elements yet, and by looking at the last
sample of the sink gives the server full control of which RTP packet it
looks at. If the payloader does not have the "stats" property, "seqnum"
is still used since "seqnum-offset" is only present in as part of
"stats" and this is still an issue not solved with this patch.

Needed for gst-plugins-base!17
2018-11-14 12:29:58 +00:00
Linus Svensson
1c4d3b36fb rtsp-stream: Plug memory leak
Attaching a GSource to a context will increase the refcount. The idle
source will never be free'd since the initial reference is never
dropped.
2018-11-14 12:29:58 +00:00
Mathieu Duponchelle
f43df76ee7 meson: add new onvif types 2018-11-01 14:20:36 +01:00
Sebastian Dröge
961de06be2 Add ONVIF subclass headers to the installed headers in meson.build too 2018-11-01 12:49:51 +02:00
Sebastian Dröge
5392cbba8a rtsp-server: Declare GstRTSPServer struct before anything else
It's needed by all kinds of other headers, including the ones that are
required for defining the GstRTSPServer struct itself and its API.
2018-11-01 11:29:01 +02:00
Sebastian Dröge
d525000f6a Mark all ONVIF-specific subclasses as Since 1.14 2018-11-01 10:23:02 +02:00
Sebastian Dröge
84a7f459a1 Include ONVIF types from single-include rtsp-server.h
... by actually making it a single-include header and moving everything
related to the GstRTSPServer type to rtsp-server-object.h instead.
Otherwise there are too many circular includes.

https://bugzilla.gnome.org/show_bug.cgi?id=797361
2018-11-01 10:18:22 +02:00
Göran Jönsson
7cfd59820a rtsp-stream: use idle source in on_message_sent
When the underlying layers are running on_message_sent, this sometimes
causes the underlying layer to send more data, which will cause the
underlying layer to run callback on_message_sent again. This can go on
and on.

To break this chain, we introduce an idle source that takes care of
sending data if there are more to send when running callback

https://bugzilla.gnome.org/show_bug.cgi?id=797289
2018-10-23 08:18:52 +01:00
Edward Hervey
ebafccb65a rtsp-client: Remove timeout GSource on cleanup
Avoids ending up with races where a timeout would still be around
*after* a client was gone. This could happen rather easily in
RTSP-over-HTTP mode on a local connection, where each RTSP message
would be sent as a different HTTP connection with the same tunnelid.

If not properly removed, that timeout would then try to free again
a client (and its contents).
2018-10-22 09:35:54 +02:00
Tim-Philipp Müller
22ced50da5 autotools: fix distcheck 2018-10-04 14:31:59 +01:00
Ognyan Tonchev
7b5e232a9e onvif: encapsulate onvif part into a bin
...and thus do not let onvif affect pipelines latency

https://bugzilla.gnome.org/show_bug.cgi?id=797174
2018-10-03 13:26:36 +03:00
Patricia Muscalu
c394de2348 New property for socket binding to mcast addresses
By default the multicast sockets are bound to INADDR_ANY,
as it's not allowed to bind sockets to multicast addresses
in Windows. This default behaviour can be changed by setting
bind-mcast-address property on the media-factory object.

https://bugzilla.gnome.org/show_bug.cgi?id=797059
2018-09-28 13:27:48 +03:00
Tim-Philipp Müller
62d4c0b179 libs: fix API export/import and 'inconsistent linkage' on MSVC
Export rtsp-server library API in headers when we're building the
library itself, otherwise import the API from the headers.

This fixes linker warnings on Windows when building with MSVC.

Fix up some missing config.h includes when building the lib which
is needed to get the export api define from config.h

https://bugzilla.gnome.org/show_bug.cgi?id=797185
2018-09-24 09:36:21 +01:00
Edward Hervey
ff51e9a68d rtsp-media-factory: Add missing break statements
This resulted in warnings/assertions whenever one accessed the
max-mcast-ttl property.

CID #1439515
CID #1439523
2018-09-19 14:31:56 +02:00
Nirbheek Chauhan
8a7f5cfe5f meson: Maintain macOS ABI through dylib versioning
Requires Meson 0.48, but the feature will be ignored on older versions
so it's safe to add it without bumping the requirement.

Documentation:
https://github.com/mesonbuild/meson/blob/master/docs/markdown/Reference-manual.md#shared_library
2018-08-31 14:42:48 +05:30
David Svensson Fors
a2e182c3b4 rtsp-client: Avoid reuse of channel numbers for interleaved
If a (strange) client would reuse interleaved channel numbers in
multiple SETUP requests, we should not accept them. The channel
numbers are used for looking up stream transports in the
priv->transports hash table, and transports disappear from the table
if channel numbers are reused.

RFC 7826 (RTSP 2.0), Section 18.54, clarifies that it is OK for the
server to change the channel numbers suggested by the client.

https://bugzilla.gnome.org/show_bug.cgi?id=796988
2018-08-29 14:46:01 +03:00
Sebastian Dröge
bd76c2f9c5 Fix indentation again 2018-08-14 14:31:55 +03:00
Patricia Muscalu
cbe6ae3c48 stream: Added a list of multicast client addresses
When media is shared, the same media stream can be sent
to multiple multicast groups. Currently, there is no API
to retrieve multicast addresses from the stream.
When calling gst_rtsp_stream_get_multicast_address() function,
only the first multicast address is returned.
With this patch, each multicast destination requested in SETUP
will be stored in an internal list (call to
gst_rtsp_stream_add_multicast_client_address()).
The list of multicast groups requested by the clients can be
retrieved by calling gst_rtsp_stream_get_multicast_client_addresses().
There still exist some problems with the current implementation
in the multicast case:
1) The receiving part is currently only configured with
regard to the first multicast client (see
https://bugzilla.gnome.org/show_bug.cgi?id=796917).
2) Secondly, of security reasons, some constraints should be
put on the requested multicast destinations (see
https://bugzilla.gnome.org/show_bug.cgi?id=796916).

Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea

https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-08-14 14:31:42 +03:00
Patricia Muscalu
4c6cecf5d6 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.

Change-Id: I5408646e3b5a0a224d907ae215bdea60c4f1905f

https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-08-14 14:31:42 +03:00
Patricia Muscalu
048e24a7c6 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.

Change-Id: I6ae578fb5164d78e8ec1e2ee82dc4eaacd0912d1

https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-08-14 14:31:42 +03:00
Patricia Muscalu
308480e762 client: Don't reserve multicast address in the client setting case
When two multicast clients request specific transport
configurations, and "transport.client-settings" parameter is
set to true, it's wrong to actually require that these two
clients request the same multicast group.
Removed test_client_multicast_invalid_transport_specific test
cases as they wrongly require that the requested destination
address is supposed to be present in the address pool, also in
the case when "transport.client-settings" parameter is set to true.

Change-Id: I4580182ef35996caf644686d6139f72ec599c9fa

https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-08-14 14:31:41 +03:00
Patricia Muscalu
a7bb684e9b Add new API for setting/getting maximum multicast ttl value
Change-Id: I5ef4758188c14785e17fb8fbf42a3dc0cb054233

https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-08-14 14:31:41 +03:00
Mathieu Duponchelle
c414158022 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 14:31:41 +03:00
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
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