Commit graph

41 commits

Author SHA1 Message Date
Matthew Waters
7b526775b4 rtsp-server-media-factory: silence some maybe-unused warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7875>
2024-11-18 12:10:57 +11:00
Edward Hervey
38c07a2d01 all: Fix closure annotations
This was misused almost everywhere.

See
https://gi.readthedocs.io/en/latest/annotations/giannotations.html#support-for-gobject-closures

and: https://www.bassi.io/articles/2023/02/20/bindable-api-2023/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7725>
2024-10-25 08:58:29 +00:00
Sebastian Dröge
f3e707c71c rtsp-server: Remove pointless assertions that can happen if client provides invalid rates
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3731
Fixes CVE-2024-44331

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7731>
2024-10-25 06:41:20 +00:00
Sebastian Dröge
0ef396359c gst: Move GstQueueArray as GstVecDeque to core
And change lengths and indices from guint to gsize for a more correct type.

Also deprecate GstQueueArray and implement it in terms of GstVecDeque.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6779>
2024-05-06 18:25:42 +00:00
Mikhail Rudenko
05ef1bbc06 rtsp-stream: clear sockets when leaving bin
Since commit 4d86f994, when setting an RTSP media both shared and
reusable, streaming cannot be restarted after the first time all the
clients disconnect. That happens because the sockets (unlike
addresses) of GstRTSPStream are not cleared in
gst_rtsp_stream_leave_bin, and on restart sockets and addresses are
not allocated in gst_rtsp_stream_allocate_udp_sockets, and then the
check in create_sender_part fails. Fix this by clearing sockets in
gst_rtsp_stream_leave_bin.

Fixes 

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6325>
2024-03-11 18:22:38 +03:00
Patricia Muscalu
bd4a9fde89 rtsp-server: Unprepare media that is in error state
Without this patch a prepared media that entered an error state
remains unprepared.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5621>
2023-11-08 14:39:01 +00:00
Jacob Johnsson
e49a9df621 rtsp-server: Only unblock live streams when complete
When media consists of multiple streams we should only unblock the
complete streams.

Fixes 

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4334>
2023-10-02 16:22:33 +00:00
Jacob Johnsson
eb0272e210 rtsp-server: Add new ensure-keyunit-on-start property
While the suspend modes NONE and PAUSED provided a low startup latency
for connecting clients they did not ensure that streams started on
fresh data.

With this property we can maintain the low startup latency of those
suspend modes while also ensuring that a stream starts on a key unit.
Furthermore, by modifying the value of a new property,
ensure-keyunit-on-start-timeout, it is possible to accept a keyunit of
a certain age but discard it if too much time has passed and instead
force a new keyunit.

Fixes 

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4334>
2023-10-02 16:22:33 +00:00
Sebastian Dröge
55c961a4dc rtsp-server: media-factory: Make sure a shared media is actually still usable
Previously it was possible that a shared media was just in the process
of being unprepared because the last client disappeared, while another
client retrieved it from the cache and then tried to use it. Unless the
media was reusable this would've then failed unnecessarily.

To avoid this it is necessary to lock the media directly in
gst_rtsp_media_factory_construct() and return a locked media. After
locking the cached media it is necessary to check if the media was ever
unprepared or is actually reusable and based on that either reuse it or
create a new media.

This minimally changes the gst_rtsp_media_factory_construct() API to
always return a locked media, and adds a new
gst_rtsp_media_can_be_shared() function to check if a media can actually
be shared in practice.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4606>
2023-05-19 11:09:48 +00:00
Thibault Saunier
b14e675a27 gir: Checkout all .gir files and check that they are updated on the CI
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3010>
2023-04-22 09:32:32 -04:00
Sebastian Dröge
5b178caadf rtsp-server: media: First set state to PLAYING again temporarily, then send EOS
Sending the EOS event while the pipeline is PAUSED can deadlock on the
stream lock if a sink is currently blocked because of pre-rolling.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4340>
2023-04-05 08:06:50 +00:00
Tim-Philipp Müller
2abf3e363d gst-rtsp-server: re-indent with GNU indent 2.2.12
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4182>
2023-03-17 03:18:54 +00:00
Tim-Philipp Müller
f5977dae15 rtsp-server: drop use of GSlice allocator
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3784>
2023-02-03 17:48:09 +00:00
Sebastian Dröge
366893e9ac Fix various warnings from gobject-introspection
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3261>
2022-10-25 09:45:25 +03:00
Sebastian Dröge
502eddfc36 rtsp-server: Add/fix various annotations
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>
2022-10-18 13:51:16 +03:00
Edward Hervey
ae8a5e110c rtsp-client: Remove duplicate documentation
Confuses the documentation builder, since it's documented twice it complains
about a missing "Since:" marker whereas it's present in the documentation
comment further down

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3180>
2022-10-14 08:54:17 +02:00
Linus Svensson
f5451f7ff2 rtsp-server: Free client if no connection could be created
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3164>
2022-10-12 11:09:41 +00:00
Peter Stensson
11982bcaba rtsp-server: Add since marker for adjust_error_code
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3157>
2022-10-12 08:08:27 +00:00
Peter Stensson
ec605e7b52 rtsp-server: Add support for adjusting request response on pipeline errors
The idea is to give the application the possibility to adjust the error
code when responding to a request. For that purpose the pipeline's bus
messages are emitted to subscribers through a signal handle-message.
The subscribers can then check those messages for errors and adjust
the response error code by overriding the virtual method
adjust_error_code().

Fixes 

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2972>
2022-10-11 07:42:28 +02:00
Chris Wiggins
5666debd5f rtsp-server: context: Add method to set the RTSPToken on some RTSPContext
Fixes .

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2979>
2022-09-13 10:42:52 +03:00
Bruce Liang
657cc3e6d6 gst-rtsp-server: Fix pushing backlog to client
Check back pressure of a stream transport before popping buffer from its backlog.

If the stream transport is not experiencing back pressure, the buffer can be popped from backlog and pushed to client.

Fixes:#1298

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2936>
2022-09-02 16:04:06 +00:00
Sebastian Dröge
57a6e48ed1 rtsp-server: stream: Don't loop forever if binding to the multicast address fails
The address/port is pre-defined by the caller of the function, so
retrying is only going to loop forever.

Ideally the multicast address should be checked after allocating but
this doesn't happen currently, so it's better to error out cleanly then
to loop forever trying the same address.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2975>
2022-09-02 14:28:26 +00:00
Thibault Saunier
bc9c1e3956 meson: Namespace the plugins_doc_dep/libraries variables
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2970>
2022-09-01 21:17:35 +00:00
Bruce Liang
ebd8bd8f13 rtsp-client: Fix url for generating key in media factory
The mount point at / can be accessed by both the URL forms rtsp://<IP>:<PORT> and rtsp://<IP>:<PORT>/.
To make media factory generating the same key for both the URL forms, the url sent to gst_rtsp_media_factory_construct() needs to be normalized first.
This commit creates a new GstRTSPUrl as the normalized url to send to gst_rtsp_media_factory_construct().

Fixes:https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1297

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2681>
2022-07-12 10:01:35 +00:00
Patricia Muscalu
6c3445a83f rtsp-media: Correct logic on GstRTSPStreamBlocking message reception
We must take into account the receiving streams as well when calculating
the expected number of the received GstRTSPStreamBlocking messages.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2429>
2022-05-20 07:37:05 +00:00
Sebastian Dröge
a91b1c64a1 rtsp-server: Add RFC5576 Source-specific media attribute to the SDP media for signalling the CNAME
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
2022-04-20 14:40:25 +00:00
Marc Leeman
5926da85ba gst-rtsp-server: minor spelling fixes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2170>
2022-04-13 14:38:52 +02:00
Matthew Waters
67e364b34d rtsp-stream: remove unused variable:
Fixes:

../gst/rtsp-server/rtsp-stream.c:2670:9: error: variable 'n_messages' set but not used [-Werror,-Wunused-but-set-variable]
  guint n_messages = 0;
        ^

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2046>
2022-03-28 10:30:23 +00:00
Vivienne Watermeier
8cb5d9f49e documentation: improve misleading wording
The documentation for several gst_*_writable_structure functions stated
that they would never return NULL, without making clear that the passed
object is required to be writable. This changes the wording in those
cases to make that requirement more clear.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1784>
2022-03-17 11:56:37 +00:00
Branko Subasic
41d436e56e gst-rtsp-server: fix race in rtsp-client
When tunneling over HTTP, if connection on the second channel happens
before the control timer is created we may trigger an assert in
rtsp_ctrl_timeout_remove(). Avoid that by taking the priv->lock before
attaching the client thread to the context.

Fixes 

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1867>
2022-03-07 09:15:11 +00:00
Fabrice Fontaine
e637aae629 rtsp-server: add gst_dep to gst_rtsp_server_deps
Add gst_dep to gst_rtsp_server_deps, in the context of buildroot, this
will avoid the following build failure, because the correct girdir
location will be retrieved from gstreamer-1.0.pc:

/home/giuliobenetti/autobuild/run/instance-3/output-1/host/riscv32-buildroot-linux-gnu/sysroot/usr/bin/g-ir-compiler gst/rtsp-server/GstRtspServer-1.0.gir --output gst/rtsp-server/GstRtspServer-1.0.typelib --includedir=/usr/share/gir-1.0
Could not find GIR file 'Gst-1.0.gir'; check XDG_DATA_DIRS or use --includedir
error parsing file gst/rtsp-server/GstRtspServer-1.0.gir: Failed to parse included gir Gst-1.0
If the above error message is about missing .so libraries, then setting up GIR_EXTRA_LIBS_PATH in the .mk file should help.
Typically like this: PKG_MAKE_ENV += GIR_EXTRA_LIBS_PATH="$(@D)/.libs"

Fixes:
 - http://autobuild.buildroot.org/results/04af6b22cfa0cffb6a3109a3b32b27137ad2e0b0

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1460>
2021-12-20 13:08:33 +00:00
Mathieu Duponchelle
79f11eb778 rtsp-stream: fix get_rates raciness
Prior to this patch, we considered that a stream was blocking
whenever a pad probe was triggered for either the RTP pad or
the RTCP pad.

This led to situations where we subsequently unblocked and expected
to find a segment on the RTP pad, which was racy.

Instead, we now only consider that the stream is blocking when
the pad probe for the RTP pad has triggered with a blockable object
(buffer, buffer list, gap event).

The RTCP pad is simply blocked without affecting the state of the
stream otherwise.

Fixes 

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1452>
2021-12-16 22:18:12 +00:00
Tim-Philipp Müller
3603d94080 rtsp-server: define G_LOG_DOMAIN
Fixes 

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1009>
2021-10-19 00:12:25 +00:00
Thibault Saunier
6e79932ad9 meson: List libraries and their corresponding gir definition
Introduces a `libraries` variable that contains all libraries in a
list with the following format:

``` meson
libraries = [
    [pkg_name, {
        'lib': library_object
        'gir': [ {full gir definition in a dict } ]
    ],
    ....
]
```

It therefore refactors the way we build the gir so that we can reuse the
same information to build them against 'gstreamer-full' in gst-build
when linking statically

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
2021-10-15 19:27:30 -03:00
Thibault Saunier
e2dd28a753 meson: Mark files as files()
Making it more robust and future proof

And fix issues that it creates

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1093>
2021-10-15 19:27:30 -03:00
Sebastian Dröge
14d636b224 rtsp-media: Unprepare suspended medias too
Previously suspended medias immediately reached the UNPREPARED state
without going through the media's unprepare() vfunc. This didn't allow
the media subclass to do any additional cleanup, and for example the
shutdown-eos property of GstRTSPMedia was ignored.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1090>
2021-10-07 11:16:39 +00:00
Sebastian Dröge
e9d551b45c rtsp-media: Only unprepare a media if it was not already unpreparing anyway
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1083>
2021-10-07 05:28:19 +00:00
Ognyan Tonchev
7ba665995f rtsp-client: make sure sessmedia will not get freed while used
handle_*_request() functions were all retrieving the session media from
the session by calling gst_rtsp_session_get_media () which is a transfer-none
call. If a session timeout happens at that time, the session media may get freed
making the pointer invalid..

Fixes 

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1053>
2021-10-06 19:42:43 +00:00
Sebastian Dröge
55222db66e rtsp-media: Also mark receive-only (RECORD) medias as prepared when unsuspending
Previously the status was only changed for other medias.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1058>
2021-10-05 16:40:07 +00:00
Sebastian Dröge
7aa88364ac rtsp-session: Don't unref medias twice if it is removed inside gst_rtsp_session_filter() while the mutex is shortly released
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/757

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1004>
2021-10-01 21:15:44 +00:00
Thibault Saunier
a43d7eaef4 Move files from gst-rtsp-server into the "subprojects/gst-rtsp-server/" subdir 2021-09-24 16:15:21 -03:00