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Author SHA1 Message Date
Sebastian Dröge
7b4665c793 Fix some new clippy 1.84 warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2032>
 2025-01-10 10:08:38 +02:00
Sebastian Dröge
38e8134edd Update to itertools 0.14 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2018>
 2025-01-01 12:46:07 +02:00
Ruben Gonzalez
ebfa0fb890 deps: update itertools to 0.13 
same used in gstreamer-rs

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2002>
 2024-12-20 16:23:52 +00:00
Thibault Saunier
1e3eef253b webrtcsrc: Add a 'connect-to-first-producer' property 
This is an helper property which allows to avoid requiring to know
peer IDs, which is very useful during development.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/386
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1996>
 2024-12-19 14:32:16 +00:00
Sebastian Dröge
7d4ddc7eb9 webrtc: Specify to use playbin3 instead of playbin 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1995>
 2024-12-18 07:31:17 +00:00
Sebastian Dröge
248b7ac059 webrtcsink: Configure custom host/port on the signaller when running signalling server internally 
Otherwise it just tries connecting to the default URL, which doesn't
work if either the host or the port are changed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1994>
 2024-12-17 16:22:41 +02:00
Mathieu Duponchelle
8886cceaf0 webrtcsink: add nvh265enc support 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1980>
 2024-12-11 08:07:15 +00:00
Taruntej Kanakamalla
c9a0731e61 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/1970>
 2024-12-05 17:28:09 +05:30
Guillaume Desmottes
45519a7d85 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/1929>
 2024-12-02 15:38:24 +00:00
Guillaume Desmottes
867c2b78b6 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/1929>
 2024-12-02 15:38:24 +00:00
Sebastian Dröge
6ee745edee Update for GLib signal accumulator API changes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1954>
 2024-11-30 15:10:06 +02:00
Sebastian Dröge
6aeb3f2af2 Fix / silence various new Rust 1.83 clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1954>
 2024-11-30 14:57:24 +02:00
Mathieu Duponchelle
4720b575b6 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/1955>
 2024-11-29 18:52:41 +01:00
Ruben Gonzalez
f646504fce webrtcsink: add openh264enc support 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1948>
 2024-11-29 13:44:11 +00:00
Sebastian Dröge
f4d2bd1a5d 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/1949>
 2024-11-28 21:14:43 +00:00
Xavier Claessens
e5f3ab4053 webrtcsink: Ignore more fields in caps change 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1838>
 2024-11-26 15:49:21 -05:00
Diego Nieto
362216f40b net/webrtc: add whipclient example 
Add a simple example producing both audio and video to make it
work with the whipserver example

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1938>
 2024-11-25 20:29:43 +01:00
Diego Nieto
0135aea9e4 net/webrtc: whipserver example 
extend the example to support both audio and video conversions

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1938>
 2024-11-25 20:29:43 +01:00
Sebastian Dröge
347bee16d4 Update for GLib signal API changes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1936>
 2024-11-22 15:52:41 +02:00
François Laignel
a8146f333f all: use builder conditional setters where applicable 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1926>
 2024-11-21 12:57:16 +00:00
François Laignel
4262a8aafe all: update due to new has_property signature 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1926>
 2024-11-21 12:57:16 +00:00
Matthew Waters
25bb2a12f1 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/1931>
 2024-11-21 17:15:44 +11:00
Jerome Colle
f88c88ddb3 webrtcsink: set rtpgccbwe min bitrate 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1896>
 2024-11-07 18:00:12 +00:00
Sebastian Dröge
ef39046e18 Update to thiserror 2 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1911>
 2024-11-06 11:02:41 +02:00
Xavier Claessens
372c44655a 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/1848>
 2024-10-30 12:36:01 +00:00
Chris Bainbridge
5010ee872d webrtc: Fix Python custom signaller receiving SDP offer 
The GstWebRTC API web interface defaults to receiving an SDP offer and
generating an answer, but this can be overridden by entering "offer
options" before clicking to open the remote stream. The Python
webrtcsink-custom-signaller.py example failed in this mode as it was
coded to only generate an offer and receive an answer. Fix this by
implementing support for receiving an offer and sending an answer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1883>
 2024-10-28 11:23:32 +00:00
Chris Bainbridge
e30d80c71e webrtc: README: add webrtcsink-custom-signaller.py 
Document the Python webrtcsink custom signaller example.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1888>
 2024-10-28 10:19:25 +00:00
Sebastian Dröge
4abc5c7a48 Be stricter with Impl-trait bounds to enforce type hierarchies 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1871>
 2024-10-22 13:43:12 +00:00
Sebastian Dröge
7e59c3f0fd Remove once_cell dependency 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1868>
 2024-10-21 17:53:18 +00:00
Sebastian Dröge
0e3d019e24 aws: Don't unnecessarily clone AWS behaviour version 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1864>
 2024-10-20 19:53:15 +00:00
Sebastian Dröge
00a4398aee aws: Allow a deprecated BehaviourVersion for now 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1864>
 2024-10-20 19:53:15 +00:00
Sebastian Dröge
b43a778a8e Fix a couple of type hierarchy bugs 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1864>
 2024-10-20 19:53:15 +00:00
Sebastian Dröge
54bc7a898e webrtc: Silence two new Rust 1.82 clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1860>
 2024-10-17 21:38:10 +00:00
Mathieu Duponchelle
959463ff65 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/1852>
 2024-10-17 12:29:53 +00:00
Mathieu Duponchelle
ef06421a25 webrtcsrc: make updated transceiver retrieval backward compatible 
In 1.24 and before transceivers for remote sendonly medias are only
created at answer time. If that is the case, we can add the transceiver
ourself, it will get associated when creating the answer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1853>
 2024-10-16 14:48:20 +00:00
Mathieu Duponchelle
82d0eaf438 webrtcsrc: fix debug message on offer created 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1853>
 2024-10-16 14:48:20 +00:00
Mathieu Duponchelle
3d257b4819 webrtcsink: improve debut message when start session failed 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1853>
 2024-10-16 14:48:20 +00:00
Chris Bainbridge
785209cc7f 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/1854>
 2024-10-16 15:45:50 +02:00
Mathieu Duponchelle
5f0ca7acde webrtcsink: fix custom_signaller hanging 
Since 6a23ae168f, the chain function
of webrtcsink adds a custom meta on input buffers.

That custom meta was registered only by the class_init of the subclasses
of BaseWebRTCSink, but the custom signaller example uses
BaseWebRTCSink::with_signaller() directly.

Fix by registering the meta in BaseWebRTCSink::class_init()

Fixes: #610
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1854>
 2024-10-16 15:25:09 +02:00
Mathieu Duponchelle
5e49f1d10e webrtcsrc: address non-compliant transceiver creation 
Instead of adding transceivers explicitly then setting the remote
description, expecting the manually added transceivers to get picked
up, we pass a promise to set-remote-description-set, and set the
relevant properties on the automatically created transceivers at that
point.

We then call create-answer and proceed as before.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/596
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1829>
 2024-10-14 11:19:38 +00:00
Guillaume Desmottes
027eead86d 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/1505>
 2024-10-10 10:59:50 -04:00
Guillaume Desmottes
d8b9a7a486 webrtc: janus: fix typo in doc 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1505>
 2024-10-10 10:57:02 -04:00
Mathieu Duponchelle
b3ace3678b webrtcsink: fix naming of error dot files for discovery pipelines 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1843>
 2024-10-03 14:35:45 +00:00
Guillaume Desmottes
d9e8f4054c 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/1830>
 2024-09-30 14:40:48 +02:00
Sebastian Dröge
dcb072ee23 webrtc: livekit: Set connection earlier during setup 
Otherwise it's not available yet when handling the initial participants
that are already in the session when joining.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1794>
 2024-09-30 13:04:24 +03:00
Sebastian Dröge
cd2b641321 livekitwebrtcsrc: Add API for disabling/enabling a track 
A disabled track is still negotiated but no data is sent for it
temporarily until it is enabled again.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1794>
 2024-09-30 13:04:24 +03:00
Sebastian Dröge
27dc76826e livekitwebrtcsrc: Add pad properties for various LiveKit participant / track metadata 
The content of the TrackInfo and ParticipantInfo structs is exposed as
gst::Structure.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1794>
 2024-09-30 13:04:24 +03:00
Mathieu Duponchelle
87c6719e1d webrtcsink: add define-encoder-bitrates signal 
When congestion control is used for a session with multiple encoders,
the default implementation simply divides the overall bitrate equally
between encoders.

This is not always desirable, and this patch exposes a new signal
that users can register to, with two arguments:

* The overall bitrate to allocate
* A structure with an encoder.stream_name -> bitrate mapping

Handlers should return a similar structure with a custom mapping.

An example is also provided.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1792>
 2024-09-25 15:19:44 +00:00
François Laignel
f532d523b2 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/1810>
 2024-09-25 09:30:48 +00:00
Mathieu Duponchelle
5c66d8c107 webrtcsrc: ensure source pad has msid when added 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1800>
 2024-09-24 14:50:30 +00:00