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Author SHA1 Message Date
Sebastian Dröge
2b9272c7eb fmp4mux: Move away from deprecated chrono function 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1503>
 2024-03-20 15:37:18 +02:00
Sebastian Dröge
cca3ebf520 rtp: Switch from chrono to time 
Which allows to simplify quite a bit of code and avoids us having to
handle some API deprecations.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1503>
 2024-03-20 15:05:39 +02:00
Sebastian Dröge
428f670753 version-helper: Use non-deprecated type alias from toml_edit 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1503>
 2024-03-19 18:16:42 +02:00
Sebastian Dröge
fadb7d0a26 deny: Add override for heck 0.4 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1503>
 2024-03-19 17:52:32 +02:00
Sebastian Dröge
2a88e29454 originalbufferstore: Update for VideoMetaTransform -> VideoMetaTransformScale rename 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1503>
 2024-03-19 17:51:41 +02:00
Sebastian Dröge
bfff0f7d87 Update Cargo.lock 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1503>
 2024-03-19 17:50:32 +02:00
Guillaume Desmottes
96337d5234 webrtc: allow resolution and framerate input changes 
Some changes do not require a WebRTC renegotiation so we can allow
those.

Fix #515

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1498>
 2024-03-18 14:52:01 +01:00
Tim-Philipp Müller
eb49459937 rtp: m2pt: add some unit tests 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1493>
 2024-03-16 10:07:37 +00:00
Tim-Philipp Müller
ce3960f37f rtp: Add MPEG-TS RTP payloader 
Pushes out pending TS packets on EOS.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1493>
 2024-03-16 10:07:37 +00:00
Tim-Philipp Müller
9f07ec35e6 rtp: Add MPEG-TS RTP depayloader 
Can handle different packet sizes, also see:
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1310

Has clock-rate=90000 as spec prescribes, see:
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/691

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1493>
 2024-03-16 10:07:37 +00:00
Mathieu Duponchelle
f4366f8b2e gstregex: add support for switches exposed by RegexBuilder 
The builder allows for instance for switching off case-sensitiveness for
the entire pattern, instead of having to do so inline with `(?i)`.

All the options exposed by the builder at
<https://docs.rs/regex/latest/regex/struct.RegexBuilder.html> can now be
passed as fields of invidual commands, snake-cased.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1497>
 2024-03-15 17:41:39 +00:00
Guillaume Desmottes
523a46b4f5 gtk4: scale texture position 
Fix regression in 0.12 introduced by 3423d05f77

Code from Ivan Molodetskikh suggested on Matrix.

Fix #519

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1499>
 2024-03-15 13:43:32 +01:00
Nirbheek Chauhan
6f8fc5f178 meson: Disable docs completely when the option is disabled 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1496>
 2024-03-14 15:30:17 +05:30
Guillaume Desmottes
8f997ea4e3 webrtc: janus: handle 'hangup' messages from Janus 
Fix error about this message not being handled:

{
   "janus": "hangup",
   "session_id": 4758817463851315,
   "sender": 4126342934227009,
   "reason": "Close PC"
}

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1481>
 2024-03-13 10:14:38 +00:00
Guillaume Desmottes
992f8d9a5d webrtc: janus: handle 'destroyed' messages from Janus 
Fix this error when the room is destroyed:

ERROR   webrtc-janusvr-signaller imp.rs:413:gstrswebrtc::janusvr_signaller:👿:Signaller::handle_msg:<GstJanusVRWebRTCSignallerU64@0x55b166a3fe40> Unknown message from server: {
   "janus": "event",
   "session_id": 6667171862739941,
   "sender": 1964690595468240,
   "plugindata": {
      "plugin": "janus.plugin.videoroom",
      "data": {
         "videoroom": "destroyed",
         "room": 8320333573294267
      }
   }
}

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1481>
 2024-03-13 10:14:38 +00:00
Guillaume Desmottes
9c6a39d692 webrtc: janus: handle (stopped-)talking events 
Expose those events using a signal.

Fix those errors when joining a Janus room configured with
'audiolevel_event: true'.

ERROR   webrtc-janusvr-signaller imp.rs:408:gstrswebrtc::janusvr_signaller:👿:Signaller::handle_msg:<GstJanusVRWebRTCSignaller@0x560cf2a55100> Unknown message from server: {
   "janus": "event",
   "session_id": 2384862538500481,
   "sender": 1867822625190966,
   "plugindata": {
      "plugin": "janus.plugin.videoroom",
      "data": {
         "videoroom": "talking",
         "room": 7564250471742314,
         "id": 6815475717947398,
         "mindex": 0,
         "mid": "0",
         "audio-level-dBov-avg": 37.939998626708984
      }
   }
}
ERROR   webrtc-janusvr-signaller imp.rs:408:gstrswebrtc::janusvr_signaller:👿:Signaller::handle_msg:<GstJanusVRWebRTCSignaller@0x560cf2a55100> Unknown message from server: {
   "janus": "event",
   "session_id": 2384862538500481,
   "sender": 1867822625190966,
   "plugindata": {
      "plugin": "janus.plugin.videoroom",
      "data": {
         "videoroom": "stopped-talking",
         "room": 7564250471742314,
         "id": 6815475717947398,
         "mindex": 0,
         "mid": "0",
         "audio-level-dBov-avg": 40.400001525878906
      }
   }
}

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1481>
 2024-03-13 10:14:38 +00:00
Guillaume Desmottes
b29a739fb2 uriplaylistbin: disable racy test 
https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/514

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1494>
 2024-03-12 16:57:22 +01:00
Guillaume Desmottes
1dea8f60a8 threadshare: disable racy tests 
https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/250

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1494>
 2024-03-12 16:54:21 +01:00
Guillaume Desmottes
2629719b4e livesync: disable racy tests 
https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/328
https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/357

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1494>
 2024-03-12 16:32:47 +01:00
Guillaume Desmottes
9e6e8c618e togglerecord: disable racy test_two_stream_close_open_nonlivein_liveout test 
See https://gitlab.freedesktop.org/gdesmott/gst-plugins-rs/-/jobs/56183085

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1494>
 2024-03-12 16:21:52 +01:00
François Laignel
995f64513d Update Cargo.lock to use latest gstreamer-rs 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1491>
 2024-03-11 14:42:36 +01:00
François Laignel
5b01e43a12 webrtc: update further to WebRTCSessionDescription sdp accessor changes 
See: https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/1406
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1491>
 2024-03-11 13:39:19 +01:00
Guillaume Desmottes
03abb5c681 spotify: document how to use with non Facebook accounts 
See discussion on #203.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1490>
 2024-03-11 09:46:40 +01:00
Zhao, Gang
7a46377627 rtp: tests: Simplify loop 
All buffers can be added in 100 outer loops. Add buffer less than 200 in the last (i = 99) loop.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1489>
 2024-03-10 16:47:30 +08:00
Olivier Crête
15e7a63e7b originalbuffer: Pair of elements to keep and restore original buffer 
The goal is to be able to get back the original buffer
after performing analysis on a transformed version. Then put the
various GstMeta back on the original buffer.

An example pipeline would be
.. ! originalbuffersave ! videoscale ! analysis ! originalbufferestore ! draw_overlay ! sink

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1428>
 2024-03-08 15:15:13 -05:00
Guillaume Desmottes
612f863ee9 webrtc: janusvrwebrtcsink: add 'use-string-ids' property 
Instead of exposing all ids properties as strings, we now have two
signaller implementations exposing those properties using their actual
type. This API is more natural and save the element and application
conversions when using numerical ids (Janus's default).

I also removed the 'joined-id' property as it's actually the same id as
'feed-id'. I think it would be better to have a 'janus-state' property or
something like that for applications willing to know when the room has
been joined.
This id is also no longer generated by the element by default, as Janus
will take care of generating one if not provided.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1486>
 2024-03-07 09:34:58 +01:00
Seungha Yang
237f22d131 sccparse: Ignore invalid timecode during seek as well 
sccparse holds last timecode in order to ignore invalid timecode
and fallback to the previous timecode. That should happen
when sccparse is handling seek event too. Otherwise single invalid
timecode before the target seek position will cause flow error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1485>
 2024-03-06 11:12:04 +00:00
Sebastian Dröge
2839e0078b rtp: Port RTP AV1 payloader/depayloader to new base classes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1472>
 2024-03-06 09:40:35 +00:00
Jordan Yelloz
0414f468c6 livekit_signaller: Added missing getter for excluded-producer-peer-ids 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1484>
 2024-03-04 10:08:11 -07:00
Jordan Yelloz
8b0731b5a2 webrtcsrc: Removed incorrect URIHandler from LiveKit source 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1484>
 2024-03-04 09:44:01 -07:00
Guillaume Desmottes
7d0397e1ad uriplaylistbin: re-enable all tests 
They now seem to work reliably. \o/

Fix #194

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1471>
 2024-03-04 12:00:13 +01:00
Guillaume Desmottes
f6476f1e8f uriplaylistbin: use vp9 in test media 
The Windows CI runner does not have a Theora decoder so those tests were
failing there.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1471>
 2024-03-04 12:00:13 +01:00
Guillaume Desmottes
cfebc32b82 uriplaylistbin: tests: use fakesink sync=true 
Tests is more reliable when using sync sink.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1471>
 2024-03-04 11:17:11 +01:00
Guillaume Desmottes
721b7e9c8c uriplaylistbin: rely on new uridecodebin3 gapless logic 
uridecodebin3 can now properly handle gapless switches so use that
instead of our own very complicated logic.

Fix #268
Fix #193

Depends on gst 1.23.90 as the plugin requires recent fixes to work properly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1471>
 2024-03-04 11:17:11 +01:00
Guillaume Desmottes
1e88971ec8 uriplaylistbin: pass valid URI in tests 
Fix critical raised by libsoup,
see https://gitlab.gnome.org/GNOME/libsoup/-/merge_requests/346

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1471>
 2024-03-04 11:06:19 +01:00
Sebastian Dröge
8a6bcb712f Remove empty line from the CHANGELOG.md that confuses the GitLab renderer 2024-03-01 16:46:21 +02:00
Jordan Yelloz
002dc36ab9 livekit_signaller: Improved shutdown behavior 
Without sending a Leave request to the server before disconnecting, the
disconnected client will appear present and stuck in the room for a little
while until the server removes it due to inactivity.

After this change, the disconnecting client will immediately leave the room.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1482>
 2024-02-29 08:21:13 -07:00
Sebastian Dröge
9c590f4223 Update Cargo.lock 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1483>
 2024-02-29 10:09:09 +00:00
Jordan Yelloz
f0b408d823 webrtcsrc: Removed flag setup from WhipServerSrc 
It's already done in the base class

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1461>
 2024-02-28 11:25:58 -07:00
Jordan Yelloz
17b2640237 webrtcsrc: Updated readme for LiveKit source 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1461>
 2024-02-28 11:25:58 -07:00
Jordan Yelloz
fa006b9fc9 webrtcsrc: Added LiveKit source element 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1461>
 2024-02-28 11:25:58 -07:00
Jordan Yelloz
96037fbcc5 webrtcsink: Updated livekitwebrtcsink for new signaller constructor 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1461>
 2024-02-28 11:25:58 -07:00
Jordan Yelloz
730b3459f1 livekit_signaller: Added dual-role support 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1461>
 2024-02-28 11:25:49 -07:00
Guillaume Desmottes
60bb72ddc3 webrtc: janus: add joined-id property to the signaller 
Fix #504

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1480>
 2024-02-28 15:05:11 +01:00
Guillaume Desmottes
eabf31e6d0 webrtc: janus: rename RoomId to JanusId 
Those weird ids are used in multiple places, not only for the room id,
so best to have a more generic name.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1480>
 2024-02-28 15:05:11 +01:00
Guillaume Desmottes
550018c917 webrtc: janus: room id not optional in 'joined' message 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1480>
 2024-02-28 14:16:46 +01:00
Guillaume Desmottes
0829898d73 webrtc: janus: remove 'audio' and 'video' from publish messages 
Those are deprecated and no longer used.

See https://janus.conf.meetecho.com/docs/videoroom and
https://github.com/meetecho/janus-gateway/blob/master/src/plugins/janus_videoroom.c#L9894

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1480>
 2024-02-28 13:39:04 +01:00
Guillaume Desmottes
ec17c58dee webrtc: janus: numerical room ids are u64 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1478>
 2024-02-28 11:56:44 +01:00
Yorick Smilda
563eff1193 Implement GstWebRTCAPI as class instead of global instance 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1373>
 2024-02-27 12:30:13 +00:00
Jordan Yelloz
594400a7f5 webrtcsrc: Made producer-peer-id optional 
It may be necessary for some signalling clients but the source element
doesn't need to depend on it.

Also, the value will fall back to the pad's MSID for the first argument
to the request-encoded-filter gobject signal when it isn't available
from the signalling client.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1477>
 2024-02-26 13:41:40 -07:00