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Author SHA1 Message Date
Taruntej Kanakamalla
6cf39d4a00 webrtc: use the nick to set enum type properties on openh264enc 
The properties `rate-control` and `complexity` are of enum types and passing
a gint value is resulting in a panic. So pass the corresponding nick of the enum
value instead

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1971>
 2024-12-05 17:17:51 +00:00
Guillaume Desmottes
625eeccf25 webrtc: janus: handle slowlink event 
Fix this warning:

webrtc-janusvr-signaller imp.rs:426:gstrswebrtc::janusvr_signaller:👿:Signaller::handle_msg:<GstJanusVRWebRTCSignallerU64@0x7f317009b4d0> Unknown message from server: {
   "janus": "slowlink",
   "session_id": 980554280060589,
   "sender": 5867141593320621,
   "mid": "video0",
   "media": "video",
   "uplink": false,
   "lost": 15
}

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1967>
 2024-12-04 13:05:26 +00:00
Guillaume Desmottes
8434126a58 webrtc: janus: handle slow_link videoroom event 
Fix this warning:

webrtc-janusvr-signaller imp.rs:426:gstrswebrtc::janusvr_signaller:👿:Signaller::handle_msg:<GstJanusVRWebRTCSignallerU64@0x7f317009b4d0> Unknown message from server: {
   "janus": "event",
   "session_id": 980554280060589,
   "sender": 5867141593320621,
   "plugindata": {
      "plugin": "janus.plugin.videoroom",
      "data": {
         "videoroom": "slow_link",
         "current-bitrate": 0
      }
   }
}

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1967>
 2024-12-04 13:05:26 +00:00
Mathieu Duponchelle
f8178fbd21 webrtscink: fix deadlock when answering 
Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/637
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1959>
 2024-12-02 16:49:18 +00:00
Ruben Gonzalez
0821c39640 webrtcsink: add openh264enc support 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1957>
 2024-11-29 23:08:45 +00:00
Sebastian Dröge
3cccede612 webrtcsink: Set caps-change-mode=delayed on encoder capsfilter 
Otherwise when changing the target caps (e.g. for reducing quality)
there is a race condition between buffers between the converter elements
and renegotiation.

For example, videoconvertscale might've output a 1920x1080 buffer, then
the capsfilter is configured to 1280x720, the buffer arrives in
videorate, videorate notices that renegotiation is pending, tries to
renegotiate and ends up with EMPTY caps because it can only change the
framerate but not the resolution.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1951>
 2024-11-29 12:39:20 +00:00
Matthew Waters
0537734aee webrtcsink: don't block the tokio runtime while holding state lock in unprepare() 
It is possible that in unprepare(), waiting for a task to complete while
holding the state lock, that task may be waiting to acquire the state lock and
result in a deadlock.

This is quick to reproduce when starting and stopping webrtcsink in very quick
succession.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1932>
 2024-11-28 16:44:05 +11:00
Xavier Claessens
52f48d7279 webrtcsink: Ignore more fields in caps change 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1940>
 2024-11-27 12:18:33 +00:00
Sebastian Dröge
be2b4952d2 Update to async-tungstenite 0.28 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1905>
 2024-11-02 12:34:43 +02:00
Sebastian Dröge
1afe165fce Update async-tungstenite dependencies 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1905>
 2024-11-02 12:34:11 +02:00
Xavier Claessens
634ec19e24 janusvr_signaller: Do not block in end_session() 
Only stop() is allowed to block, wait there.

Fixes #603

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1903>
 2024-11-01 15:11:17 +00:00
Sebastian Dröge
07f7730632 aws: Allow a deprecated BehaviourVersion for now 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1873>
 2024-10-22 20:21:55 +00:00
Sebastian Dröge
347b5987bd Fix a couple of type hierarchy bugs 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1873>
 2024-10-22 20:21:55 +00:00
Sebastian Dröge
6c0bfd3ffc webrtc: Silence two new Rust 1.82 clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1874>
 2024-10-22 19:29:28 +00:00
Mathieu Duponchelle
2790fb41b2 webrtcsink: fix session not in place errors 
The InPlace/Taken logic was introduced to avoid using an extra lock
around the session, but it places expectations that are not always
obvious to meet around when a session is expected to be taken or not.

Any code that expects to have access to the sessions at all times thus
needs either extra logic in the session wrapper, or to maintain the
state of the session outside of the session (eg mids).

This commit removes the logic, and wraps sessions in Arc<Mutex>>.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1859>
 2024-10-18 10:53:16 +00:00
Chris Bainbridge
f5b90ba261 custom-signaller: add missing manual-sdp-munging property 
All signallers must now implement this property

Fixes #611

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1858>
 2024-10-18 09:33:42 +00:00
Guillaume Desmottes
57234522ec webrtc: janus: add 'janus-state' property to the sink 
This property can be used by applications to track the state of the
signaller, especially to know when the stream is up.

Fix #510

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1849>
 2024-10-10 16:56:25 -04:00
Guillaume Desmottes
6cc9945d4e webrtc: janus: fix typo in doc 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1849>
 2024-10-10 16:43:23 -04:00
Mathieu Duponchelle
5958e342c7 webrtcsink: fix naming of error dot files for discovery pipelines 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1844>
 2024-10-03 16:45:42 +01:00
Guillaume Desmottes
87697609a2 webrtc: allow PAR change in webrtcsink input caps 
We are already allowing resolution changes which can lead to change in
pixel-aspect-ratio.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1833>
 2024-10-01 17:42:21 +01:00
Mathieu Duponchelle
8eedd0ac6d webrtcsrc: ensure source pad has msid when added 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1812>
 2024-09-26 06:33:44 +00:00
Mathieu Duponchelle
41f75378df webrtcsrc: fix default msid property value 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1812>
 2024-09-26 06:33:44 +00:00
François Laignel
8f542f084c webrtcsink: fix RFC7273 attributes 
RFC7273 related attributes are set in the SDP offer by passing them via the
transceiver `codec-preferences` signal. These attributes are intended to be set
at the media level so they must be prefixed by `a-` in the `Caps` argument to
the signal. Otherwise they end up under `a=fmtp`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1811>
 2024-09-25 11:25:51 +01:00
Mathieu Duponchelle
9331824479 webrtcsrc: expose MSID property on source pad 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1796>
 2024-09-21 00:27:57 +02:00
Arun Raghavan
b1ea6d2e65 webrtc: Fix whipclientsink name in README 
The element name was changed, but the documentation wasn't updated to
match.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1765>
 2024-09-04 13:48:06 +01:00
Mathieu Duponchelle
88a6b6d428 net/webrtc: Add missing npm command to README 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/589

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1747>
 2024-08-23 10:30:18 +00:00
Mathieu Duponchelle
4559b6a9f9 webrtcsink: fix segment format mismatch with remote offer 
webrtcsink was starting the negotiation process on Ready and concurrently
moving the consumer pipeline to Playing, but when answering the remote
description was set so fast that input streams were connected (and the time
format set on appsrc) before the state change to Paused had completed.

This meant gst_base_src_start was happening after that and setting the format
back to bytes, the time segment that was next coming in then caused:

basesrc gstbasesrc.c:4255:gst_base_src_push_segment:<video_0> segment format mismatched, ignore

And the consumer pipeline errored out.

The same issue existed in theory when webrtcsink was creating the offer,
but was much harder to trigger as it required that the remote answer
came in before the state change to Paused had completed.

This commit fixes the issue by simply waiting for the state to have
changed to Paused before negotiating.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1738>
 2024-08-23 09:30:03 +00:00
Sebastian Dröge
85151a6e4f Fix various 1.80 clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1740>
 2024-08-23 08:59:20 +03:00
Jerome Colle
1abe0fd5fe webrtcsink: add nvv4l2av1enc support 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1743>
 2024-08-22 09:59:22 +00:00
Dave Lucia
a3d6308579 net/webrtc: Fix turn-servers nick: user -> use 
Noticed this typo

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1713>
 2024-08-14 09:34:59 +00:00
Sebastian Dröge
b673de4e07 webrtcsrc: Make sure to always call end_session() without the state lock 
This was already done in another place for the same reason: preventing a
deadlock. It's probably not correct as hinted by the FIXME comment but
better than deadlocking at least.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1715>
 2024-08-14 08:17:48 +00:00
Mathieu Duponchelle
c87ddd43e4 webrtcsink: fix assertions when finalizing 
Dumping the pipeline on state changes from an async bus handler was
triggering criticals.

Instead, dump from the sync handler.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1714>
 2024-08-13 18:07:13 +01:00
Sebastian Dröge
c712b36082 webrtcsrc: Don't hold the state lock while removing sessions 
Removing a session can drop its bin and during release of the bin its
pads are removed, but the pad-removed handler is also taking the state
lock.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1700>
 2024-08-07 12:00:57 +00:00
Loïc Le Page
4bef63ba26 gstwebrtc-api: always include index file in dist for convenience 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1675>
 2024-07-18 23:38:56 +00:00
Sebastian Dröge
06fb47d197 webrtc: Add version to gst-plugin-webrtc-protocol dependency 2024-07-16 18:59:19 +03:00
François Laignel
b50f76223c webrtcsink: fix property types for rav1enc 
Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/572
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1667>
 2024-07-16 12:47:33 +03:00
Sebastian Dröge
a8ccfe49d9 webrtc: Require livekit-protocol < 0.3.4 due to uncoordinated breaking changes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1663>
 2024-07-11 20:00:24 +03:00
Sebastian Dröge
98b28d69ce Update for new debug log macro syntax 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1658>
 2024-07-08 11:25:23 +03:00
leonardo salvatore
f303992e0c webrtcsink: initial support for vpuenc_h264 encoder for imx8mp, default values set to cover a common streaming scenario 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1639>
 2024-07-01 07:34:04 +00:00
Sebastian Dröge
47d62b6d78 Update for new clone/closure macro syntax 
Also fix various weak/strong references in the webrtc plugin, and make
sure to pass the object to debug log functions in every place.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1625>
 2024-06-21 11:54:58 +03:00
Sebastian Dröge
9b323a6519 Use Option::is_some_and(...) instead of Option::map_or(false, ...) 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1630>
 2024-06-19 13:03:37 +00:00
Sebastian Dröge
23d998a1db Slightly improve code making use of element factories retrieved from an element 
We can use `is_some_and(...)` instead of `map_or(false, ...)`.

Also in a few places the factory was retrieved multiple times, one time
with unwrapping and another time with handling the `None` case
correctly. Instead of unwrapping, move code to handle the `None` case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1630>
 2024-06-19 13:03:37 +00:00
Sebastian Dröge
ba70bb1154 deny: Add override for older tungstenite 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1603>
 2024-06-06 10:34:12 +00:00
Sebastian Dröge
85c38107cf webrtc: Update to async-tungstenite 0.26 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1603>
 2024-06-06 10:34:12 +00:00
Francisco Javier Velázquez-García
8fc652f208 webrtcsink: Refactor value retrieval to avoid lock poisoning 
When setting an incorrect property name in settings,
start_stream_discovery_if_needed would panic because it attempts to
unwrap a poisoned lock for settings.

This refactor avoids that situation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1594>
 2024-05-31 08:10:23 +00:00
Francisco Javier Velázquez-García
568e8533fa webrtcsink: Fix typo in property name for av1enc 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1594>
 2024-05-31 08:10:23 +00:00
Arun Raghavan
04e9e5284c webrtc: signaller: A couple of minor doc fixups 
The expectation is `Returns:`, not `Return:`

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1525>
 2024-05-30 22:16:46 +03:00
Arun Raghavan
1c54c77840 webrtcsink: Add a mechanism for SDP munging 
Unfortunately, server implementations might have odd SDP-related quirks,
so let's allow clients a way to work around these oddities themselves.
For now, this means that a client can fix up the H.264 profile-level-id
as required by Twitch (whose media pipeline is more permissive than the
WHIP implementation).

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/516
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1525>
 2024-05-30 22:16:46 +03:00
Taruntej Kanakamalla
83f76280f5 net/webrtc: Example for whipserver 
rudimentary sample to test multiple WHIP client connections

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
 2024-05-29 21:03:27 +00:00
Taruntej Kanakamalla
712d4757c3 net/webrtc/whip_signaller: multiple client support in the server 
- generate a new session id for every new client
use the session id in the resource url

- remove the producer-peer-id property in the WhipServer signaler as it
is redundant to have producer id in a session having only one producer

- read the 'producer-peer-id' property on the signaller conditionally
if it exists else use the session id as producer id

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
 2024-05-29 21:03:27 +00:00