Merge requests · GStreamer · GitLab

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bus: do not leak messages if there is no async handler

gstreamer!6124 · created Feb 15, 2024 by Xavier Claessens 1.23.90
Core Memory
- Merged
- 4
updated Feb 16, 2024
Build 1.22 branch again after 1.22.10

cerbero!1369 · created Feb 15, 2024 by Tim-Philipp Müller 1.22
No milestone
updated Feb 15, 2024
Backport of "urisourcebin: Don't acquire STATE_LOCK if shutting down" into 1.22

gstreamer!6123 · created Feb 15, 2024 by Tim-Philipp Müller 1.22.11 1.22
- Merged
- 1
updated Feb 16, 2024
Add elements to convert relation metas to/from ONVIF metas

gst-plugins-rs!1464 · created Feb 15, 2024 by Benjamin Gaignard 0.14.0
Meta
- Merged
- 102
- 1
- Approved
updated Oct 23, 2024
Implement RTP Reference Picture Selection Indication

gstreamer!6122 · created Feb 15, 2024 by Benjamin Gaignard
Enhancement RTP VP8
- 8
updated Oct 22, 2024
urisourcebin: Don't acquire STATE_LOCK if shutting down

gstreamer!6121 · created Feb 15, 2024 by Edward Hervey 1.23.2
1.22 Backported into 1.22
- Merged
- 1
updated Feb 15, 2024
libgstplay: Use playbin3 by default

gstreamer!6120 · created Feb 15, 2024 by Edward Hervey 1.23.2
- Merged
- Approved
updated Feb 20, 2024
webrtcbin: RFC5576 - early CNAME support

gstreamer!6119 · created Feb 14, 2024 by François Laignel 1.25.1
WebRTC
- Merged
- 61
- Approved
updated Mar 08, 2024
video: Fix NV12_16L32S video frame size

gstreamer!6118 · created Feb 14, 2024 by Nicolas Dufresne 1.23.90
1.22 Backported into 1.22
- Merged
- 2
updated Feb 17, 2024
aja: Replace global semaphore with per-device flock()

gstreamer!6117 · created Feb 14, 2024 by Xavier Claessens 1.23.2
- Merged
- 7
updated Feb 15, 2024
Revert "rtpvp8pay: Use GstBitReader instead of dboolhuff implementation from libvpx"

gstreamer!6116 · created Feb 14, 2024 by Sebastian Dröge 1.23.2
- Merged
- 3
updated Feb 14, 2024
gst-play: Default to using playbin3

gstreamer!6115 · created Feb 14, 2024 by Edward Hervey 1.23.2
Merge in 24h
- Merged
- 1
updated Feb 14, 2024
Draft: v4l2: videodec: DMA_DRM caps support PoC

gstreamer!6114 · created Feb 14, 2024 by Robert Mader
- Closed
- 9
updated Oct 10, 2024
webrtc: add precise synchronization example

gst-plugins-rs!1463 · created Feb 14, 2024 by François Laignel 0.13.0
WebRTC
- Merged
- 80
- Approved
updated May 29, 2024
rtpgccbwe: Don't reset PTS/DTS to `None`

gst-plugins-rs!1462 · created Feb 14, 2024 by Sebastian Dröge 0.13.0
updated May 29, 2024
Update translations

gstreamer!6113 · created Feb 14, 2024 by Tim-Philipp Müller 1.23.2
updated Feb 14, 2024
Enable Rust plugins on Android

cerbero!1368 · created Feb 13, 2024 by amyspark 1.23.2
Android Rust
- Merged
- 1
updated Feb 14, 2024
webrtc: Add LiveKit source element

gst-plugins-rs!1461 · created Feb 13, 2024 by Jordan Yelloz 0.13.0
- Merged
- 16
- Approved
updated May 29, 2024
webrtcbin, rtpbin: check before setting properties on jitterbuffer

gstreamer!6112 · created Feb 13, 2024 by Mathieu Duponchelle 1.23.2
- Merged
- 2
- Approved
updated Feb 14, 2024
Back to development after 1.22.10

gstreamer!6111 · created Feb 13, 2024 by Tim-Philipp Müller 1.22
No milestone
updated Feb 13, 2024