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Author SHA1 Message Date
Matthew Waters b51eeb8ea1 Merge branch 'rtp2' into 'main' 
Draft: rtp: new rtpbin2 element

See merge request gstreamer/gst-plugins-rs!1426
 2024-03-19 08:09:49 +00:00
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 0c033f65ce rtspsrc2: Update rtpbin2 support to use rtprecv and rtpsend 
USE_RTPBIN2 is now USE_RTP2 because there is no "rtpbin2" now.
 2024-03-14 17:54:59 +05:30
Matthew Waters e3458bb58b rtpbin2: split send and receive halves into separate elements 
There is now two elements, rtpsend and rtprecv that represent the two
halves of a rtpsession.  This avoids the potential pipeline loop if two
peers are sending/receiving data towards each other.  The two halves can
be connected by setting the rtp-id property on each element to the same
value and they will behave like a combined rtpbin-like element.
 2024-03-14 23:22:43 +11:00
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
Matthew Waters 7b67c87dda rtpbin2: expose session signals for new/bye ssrc 2024-03-14 19:20:00 +11:00
Matthew Waters 36cf3f6aae rtpbin2/config: add stats to session GObject 2024-03-14 19:20:00 +11:00
Matthew Waters 7e79d91580 rtpbin2/config: add a new-ssrc signal 2024-03-14 19:20:00 +11:00
Matthew Waters 041fbebe70 rtpbin2: implement a session configuration object 
Currently only contains pt-map
 2024-03-14 19:20:00 +11:00
Matthew Waters 9146d3dfb5 jitterbuffer: handle flush-start/stop 2024-03-14 19:20:00 +11:00
Matthew Waters 2c70a313e7 jitterbuffer: remove mpsc channel for every packet 
It is very slow.
 2024-03-14 19:20:00 +11:00
Mathieu Duponchelle 9e50a31dbc jitterbuffer: implement support for serialized events / queries 2024-03-14 19:20:00 +11:00
Mathieu Duponchelle 93666313a5 rtpbin2: implement and use synchronization context 
Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
Co-Authored-By: Matthew Waters <matthew@centricular.com>
 2024-03-14 19:20:00 +11:00
Mathieu Duponchelle 509b615ba4 rtpbin2: implement jitterbuffer 
The jitterbuffer implements both reordering and duplicate packet
handling.

Co-Authored-By: Sebastian Dröge <sebastian@centricular.com>
Co-Authored-By: Matthew Waters <matthew@centricular.com>
 2024-03-14 19:20:00 +11:00
Sebastian Dröge 52ea10c03a rtpbin2: Add support for receiving rtcp-mux packets 2024-03-14 19:20:00 +11:00
Sebastian Dröge 726e6b6e49 rtpbin2: Implement support for reduced size RTCP (RFC 5506) 2024-03-14 19:20:00 +11:00
Sebastian Dröge 9817145910 rtpbin2: Add support for sending NACK/PLI and FIR 
Co-Authored-By: Matthew Waters <matthew@centricular.com>
 2024-03-14 19:20:00 +11:00
Sebastian Dröge 9a9a30bf10 rtpbin2: Add handling for receiving NACK/PLI and FIR 
Co-Authored-By: Matthew Waters <matthew@centricular.com>
Co-Authored-By: Mathieu Duponchelle <mathieu@centricular.com>
 2024-03-14 19:20:00 +11:00
Matthew Waters 1c04618f8d rtpbin2: add support for RFC 4585 (RTP/AVPF) 
Implements the timing rules for RTP/AVPF

Co-Authored-By: Sebastian Dröge <sebastian@centricular.com>
 2024-03-14 19:20:00 +11:00
Matthew Waters 513c4bb9e1 rtp: Initial rtpbin2 element 
Can receive and recevie one or more RTP sessions containing multiple
pt/ssrc combinations.

Demultiplexing happens internally instead of relying on separate
elements.

Co-Authored-By: François Laignel <francois@centricular.com>
Co-Authored-By: Mathieu Duponchelle <mathieu@centricular.com>
Co-Authored-By: Sebastian Dröge <sebastian@centricular.com>
 2024-03-14 19:19:58 +11:00
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