Merge requests · GStreamer

v4l2bufferpool: actually queue back the empty buffer flagged LAST

gstreamer!7386 · created Aug 20, 2024 by Randy Li
1.24 Needs backport Video4Linux
- Closed
- 6
updated Sep 09, 2024
Draft: CI: Update fedora base image to f40 6 of 16 checklist items completed

gstreamer!5525 · created Oct 22, 2023 by Jordan Petridіs
CI
- Closed
- 54
updated Sep 05, 2024
Draft: webrtcbin & rtpsource: Fix stats for sendonly streams

gstreamer!3984 · created Feb 16, 2023 by Nirbheek Chauhan
RTP WebRTC
- Closed
- 3
updated Sep 03, 2024
webrtcbin: Prevent crash when attempting to set answer on invalid SDP

gstreamer!7438 · created Sep 02, 2024 by Philippe Normand 1.24
WebRTC
- Closed
- 2
updated Sep 02, 2024
utils: Deprecate gst_util_ceil_log2()

gstreamer!7411 · created Aug 26, 2024 by Sebastian Dröge
- Closed
- 18
updated Aug 30, 2024
rtpmanager: save the report block statistics in each RTPSource

gst-plugins-good!401 · created Nov 21, 2019 by Fede Claramonte master
Good Plugins
- Closed
- 8
updated Aug 29, 2024
rtmp2: Add llnw auth support to rtmp client

gstreamer!7335 · created Aug 12, 2024 by Daniel Pendse
RTMP
- Closed
- 43
updated Aug 26, 2024
config: Fix config variants override with first option (two passes)

cerbero!1537 · created Aug 04, 2024 by Maxim P. DEMENTIEV
- Closed
- 40
updated Aug 21, 2024
ci: Remove RUST_IMAGE_FULL env var

gstreamer-rs!1501 · created Aug 20, 2024 by Jordan Petridіs
CI
- Closed
- 1
updated Aug 20, 2024
audio: add speechmatics transcriber

gst-plugins-rs!1724 · created Aug 16, 2024 by Mathieu Duponchelle
- Closed
- 1
updated Aug 16, 2024
webrtcbin: Accept SDP without a=mid lines

gstreamer!3759 · created Jan 19, 2023 by Arun Raghavan
WebRTC
- Closed
- 4
updated Aug 15, 2024
webrtcbin: Fix default value when SDP direction is unspecified

gstreamer!3735 · created Jan 16, 2023 by Arun Raghavan
WebRTC
- Closed
- 8
updated Aug 15, 2024
Draft: kmssink: Add 'asahi' to the list of driver name

gstreamer!6176 · created Feb 22, 2024 by Daniel Morin
- Closed
- 2
updated Aug 12, 2024
tests: test-filter: Avoid gcc 14 warning about uninitialized variable

gstreamer!6662 · created Apr 16, 2024 by Jordan Petridіs
Tests
- Closed
- 1
updated Aug 12, 2024
rtpsession: save the report block statistics in each RTPSource

gst-plugins-good!541 · created Mar 26, 2020 by Miguel París Díaz master
Good Plugins
- Closed
- 1
updated Aug 12, 2024
{audio,video}aggregator: Add has_internal_buffer vfunc

gst-plugins-base!158 · created Mar 15, 2019 by Seungha Yang master
Base Plugins
- Closed
- 3
updated Aug 09, 2024
aggregator: Add signal pad-drained to notify actual drained status of pad

gstreamer!112 · created Mar 15, 2019 by Seungha Yang master
Core
- Closed
- 14
updated Aug 09, 2024
appsink: Drop last buffer reference on NULL state

gstreamer!7332 · created Aug 08, 2024 by Xavier Claessens
- Closed
- 3
updated Aug 08, 2024
appsink: Drop last buffer reference on NULL state

gst-plugins-base!787 · created Aug 12, 2020 by Xavier Claessens master
Base Move to Monorepo Plugins
- Closed
- 25
updated Aug 08, 2024
gst-play: Restrict the maximum allowed volume value to 2.0

gst-plugins-base!190 · created Mar 31, 2019 by Seungha Yang master
Base Move to Monorepo Plugins Tools
- Closed
- 4
updated Aug 08, 2024

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