mirrors/gst-plugins-rs
mirrors/gst-plugins-rs
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs.git synced 2024-12-23 10:30:40 +00:00
Code Issues Projects Releases Wiki Activity
3164 commits 14 branches 101 tags 30 MiB
Commit graph

3164 commits

This branch
This branch
All branches
Author SHA1 Message Date
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
Sebastian Dröge
47ba068966 Update CHANGELOG.md for 0.12.2 2024-02-26 14:58:58 +02:00
Sebastian Dröge
5df7c01cb5 closedcaption: Port from nom to winnow 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1475>
 2024-02-26 14:00:08 +02:00
Xavier Claessens
f7ffa13543 janusvr: Add string-ids property 
It forces usage of strings even if it can be parsed into an integer.
This allows joining room `"133"` in a server configured with string
room ids.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1466>
 2024-02-26 11:10:00 +00:00
Xavier Claessens
23955d2dbb janusvr: Room IDs can be strings 
Sponsored-by: Netflix Inc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1466>
 2024-02-26 11:10:00 +00:00
Sebastian Dröge
340d65d7a4 Update Cargo.lock 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1474>
 2024-02-26 11:14:01 +02:00
Sebastian Dröge
b9195ed309 fmp4mux: Update to dash-mpd 0.15 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1474>
 2024-02-26 11:14:01 +02:00
Sebastian Dröge
fc1c017fc6 Update CHANGELOG.md for 0.12.1 2024-02-23 19:19:17 +02:00
Sebastian Dröge
f563f8334b rtp: Add PCMU/PCMA RTP payloader / depayloader elements 
These come with new generic RTP payloader, RTP raw-ish audio payloader
and RTP depayloader base classes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1424>
 2024-02-23 14:43:45 +02:00
Xavier Claessens
e09f9e9540 meson: Fix error when default_library=both 
Skip duplicated plugin_name when we have both the static and shared
plugin in the plugins list.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1470>
 2024-02-22 12:31:23 -05:00
Maksym Khomenko
da21dc853d webrtcsink: extensions: separate API and signal checks 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1469>
 2024-02-20 19:29:46 +02:00
Maksym Khomenko
2228f882d8 webrtcsink: apply rustfmt 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1469>
 2024-02-20 19:29:28 +02:00
Mathieu Duponchelle
8f3a6171ac textwrap: don't split on all whitespaces .. 
but only on ASCII whitespaces, as we want to honor non-breaking
whitespaces (\u{a0})

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1468>
 2024-02-16 19:38:38 +01:00
Xavier Claessens
2572afbf15 janusvr: Add secret-key property 
Every API calls have an optional "apisecret" argument.

Sponsored-by: Netflix Inc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1465>
 2024-02-16 14:04:59 +00:00
Sebastian Dröge
0faac3b875 deny: Add winnow 0.5 override 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1467>
 2024-02-16 14:27:29 +02:00
Sebastian Dröge
cb0cc764ba Update Cargo.lock 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1467>
 2024-02-16 14:26:44 +02:00
Sebastian Dröge
45f55423fb Remove Cargo.lock from .gitignore 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1467>
 2024-02-16 14:25:54 +02:00
Sebastian Dröge
8ef12a72e8 rtpgccbwe: Don't reset PTS/DTS to None 
The element is usually placed before `rtpsession`, and `rtpsession`
needs the PTS/DTS for correctly determining the running time. The
running time is then used to produce correct RTCP SR, and to potentially
update an NTP-64 RTP header extension if existing on the packets.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/496

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1462>
 2024-02-14 08:05:54 +00:00
Sebastian Dröge
05884de50c textwrap: Remove unnecessary to_string() in debug output of a string 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1458>
 2024-02-12 19:09:06 +02:00
Jordan Yelloz
67b7cf9764 webrtcsink: Added sinkpad with "msid" property 
This forwards to the webrtcbin sinkpad's msid when specified.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1442>
 2024-02-12 15:04:44 +00:00
Sebastian Dröge
9827106961 Update Cargo.lock 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1455>
 2024-02-11 11:55:37 +02:00