| Index | index by Group | index by Distribution | index by Vendor | index by creation date | index by Name | Mirrors | Help | Search |
| Name: gstreamer-rtsp-server-devel | Distribution: openSUSE Tumbleweed |
| Version: 1.24.9 | Vendor: openSUSE |
| Release: 1.1 | Build date: Tue Nov 5 10:56:58 2024 |
| Group: Development/Languages/C and C++ | Build host: reproducible |
| Size: 1131051 | Source RPM: gstreamer-rtsp-server-1.24.9-1.1.src.rpm |
| Packager: http://bugs.opensuse.org | |
| Url: https://gstreamer.freedesktop.org | |
| Summary: Development files for the GStreamer-based RTSP server library | |
Development files for the GStreamer library for building an RTSP server.
LGPL-2.0-or-later
* Tue Nov 05 2024 Bjørn Lie <bjorn.lie@gmail.com>
- Update to version 1.24.9:
+ rtsp-server: Remove pointless assertions that can happen if
client provides invalid rates (security fix)
* Mon Sep 23 2024 Bjørn Lie <bjorn.lie@gmail.com>
- Update to version 1.24.8:
+ No changes, stable version bump only.
* Fri Aug 23 2024 Bjørn Lie <bjorn.lie@gmail.com>
- Update to version 1.24.7:
+ No changes, stable version bump only.
* Wed Aug 14 2024 Bjørn Lie <bjorn.lie@gmail.com>
- Update to version 1.24.6:
+ Highlighted bugfixes:
- Fix compatibility with FFmpeg 7.0.
- qmlglsink: Fix failure to display content on recent Android
devices.
- adaptivedemux: Fix handling of closed caption streams.
- cuda: Fix runtime compiler loading with old CUDA tookit.
- decodebin3 stream selection handling fixes.
- d3d11compositor, d3d12compositor: Fix transparent background
mode with YUV output.
- d3d12converter: Make gamma remap work as intended.
- h264decoder: Update output frame duration for interlaced
video when second field frame is discarded.
- macOS audio device provider now listens to audio devices
being added/removed at runtime.
- Rust plugins: audioloudnorm, s3hlssink, gtk4paintablesink,
livesync and webrtcsink fixes.
- videoaggregator: preserve features in non-alpha caps for
subclasses with non-system memory sink caps.
- vtenc: Fix redistribute latency spam.
- v4l2: fixes for complex video formats.
- va: Fix strides when importing DMABUFs, dmabuf handle leaks,
and blocklist unmaintained Intel i965 driver for encoding.
- waylandsink: Fix surface cropping for rotated streams.
- webrtcdsp: Enable multi_channel processing to fix handling of
stereo streams.
- Various bug fixes, memory leak fixes, and other stability and
reliability improvements.
* Fri Jun 28 2024 Bjørn Lie <bjorn.lie@gmail.com>
- Update to version 1.24.5:
+ Highlighted bugfixes:
- webrtcsink: Support for AV1 via nvav1enc, av1enc or rav1enc
encoders
- AV1 RTP payloader/depayloader fixes to work correctly with
Chrome and Pion WebRTC
- av1parse, av1dec error handling/robustness improvements
- av1enc: Handle force-keyunit events properly for WebRTC
- decodebin3: selection and collection handling improvements
- hlsdemux2: Various fixes for discontinuities, variant
switching, playlist updates
- qml6glsink: fix RGB format support
- rtspsrc: more control URL handling fixes
- v4l2src: Interpret V4L2 report of sync loss as video signal
loss
- d3d12 encoder, memory and videosink fixes
- vtdec: more robust error handling, fix regression
- ndi: support for NDI SDK v6
- Various bug fixes, memory leak fixes, and other stability and
reliability improvements
- Please see https://gstreamer.freedesktop.org/releases/1.24/ for
changes between 1.24.0 and this version and even more in-depth
info.
* Tue Mar 05 2024 Antonio Larrosa <alarrosa@suse.com>
- Update to version 1.24.0:
* Highlights
- New Discourse forum and Matrix chat space
- New Analytics and Machine Learning abstractions and elements
- Playbin3 and decodebin3 are now stable and the default in
gst-play-1.0, GstPlay/GstPlayer
- The va plugin is now preferred over gst-vaapi and has higher
ranks
- GstMeta serialization/deserialization and other GstMeta
improvements
- New GstMeta for SMPTE ST-291M HANC/VANC Ancillary Data
- New unixfd plugin for efficient 1:N inter-process
communication on Linux
- cudaipc source and sink for zero-copy CUDA memory sharing
between processes
- New intersink and intersrc elements for 1:N pipeline
decoupling within the same process
- Qt5 + Qt6 QML integration improvements including qml6glsrc,
qml6glmixer, qml6gloverlay, and qml6d3d11sink elements
- DRM Modifier Support for dmabufs on Linux
- OpenGL, Vulkan and CUDA integration enhancements
- Vulkan H.264 and H.265 video decoders
- RTP stack improvements including new RFC7273 modes and more
correct header extension handling in depayloaders
- WebRTC improvements such as support for ICE consent
freshness, and a new webrtcsrc element to complement
webrtcsink
- WebRTC signallers and webrtcsink implementations for LiveKit
and AWS Kinesis Video Streams
- WHIP server source and client sink, and a WHEP source
- Precision Time Protocol (PTP) clock support for Windows and
other additions
- Low-Latency HLS (LL-HLS) support and many other HLS and DASH
enhancements
- New W3C Media Source Extensions library
- Countless closed caption handling improvements including new
cea608mux and cea608tocea708 elements
- Translation support for awstranscriber
- Bayer 10/12/14/16-bit depth support
- MPEG-TS support for asynchronous KLV demuxing and segment
seeking, plus various new muxer features
- Capture source and sink for AJA capture and playout cards
- SVT-AV1 and VA-API AV1 encoders, stateless AV1 video decoder
- New uvcsink element for exporting streams as UVC camera
- DirectWrite text rendering plugin for windows
- Direct3D12-based video decoding, conversion, composition, and
rendering
- AMD Advanced Media Framework AV1 + H.265 video encoders with
10-bit and HDR support
- AVX/AVX2 support and NEON support on macOS on Apple ARM64
CPUs via new liborc
- GStreamer C# bindings have been updated
- Rust bindings improvements and many new and improved Rust
plugins
- Rust plugins now shipped in packages for all major platforms
including Android and iOS
- Lots of new plugins, features, performance improvements and
bug fixes
* For more detailed information on this update, please see
https://gstreamer.freedesktop.org/releases/1.24/
- Remove patch reduce-required-meson.patch since meson 1.1 is
really required now.
* Thu Feb 01 2024 Antonio Larrosa <alarrosa@suse.com>
- Update to version 1.22.9:
+ No changes, stable bump only.
- Rebase reduce-required-meson.patch.
* Thu Jan 04 2024 Antonio Larrosa <alarrosa@suse.com>
- Update to version 1.22.8:
+ No changes, stable bump only.
- Rebase reduce-required-meson.patch.
* Wed Nov 15 2023 Antonio Larrosa <alarrosa@suse.com>
- Update to version 1.22.7:
+ rtspclientsink: Don't leak previous server_ip
- Rebase reduce-required-meson.patch.
* Fri Sep 22 2023 Bjørn Lie <bjorn.lie@gmail.com>
- Update to version 1.22.6:
+ No changes, stable bump only.
- Rebase reduce-required-meson.patch.
* Tue Jul 25 2023 Bjørn Lie <bjorn.lie@gmail.com>
- Update to version 1.22.5:
+ No changes
- Rebase reduce-required-meson.patch.
* Mon Jun 26 2023 Bjørn Lie <bjorn.lie@gmail.com>
- Update to version 1.22.4:
+ No changes.
- Rebase reduce-required-meson.patch.
* Wed May 24 2023 Bjørn Lie <bjorn.lie@gmail.com>
- Update to version 1.22.3:
+ No changes.
- Rebase patch.
* Wed Apr 12 2023 Bjørn Lie <bjorn.lie@gmail.com>
- Update to version 1.22.2:
+ rtsp-server: fix deadlock on shutdown with non-live pipeline if
media isn't playing/prerolled yet and eos-shutdown is enabled
for the media
- Rebase patch.
* Thu Mar 09 2023 Bjørn Lie <bjorn.lie@gmail.com>
- Update to version 1.22.1:
+ No changes.
- Rebase patch with quilt.
* Wed Mar 01 2023 Antonio Larrosa <alarrosa@suse.com>
- Add patch to reduce the required meson version to 0.61.0 since
that's what we have in SLE 15:
* reduce-required-meson.patch
* Wed Jan 25 2023 Bjørn Lie <bjorn.lie@gmail.com>
- Update to version 1.22.0:
+ Please see changes in gstreamer main package, major version
bump.
* Fri Dec 23 2022 Bjørn Lie <bjorn.lie@gmail.com>
- Update to version 1.20.5:
+ rtsp-server: Free client if no connection could be created
* Sat Oct 22 2022 Bjørn Lie <bjorn.lie@gmail.com>
- Update to version 1.20.4:
+ gst-rtsp-server: Fix pushing backlog to client.
+ rtsp-server: stream: Don't loop forever if binding to the
multicast address fails.
* Wed Jun 22 2022 Aaron Stern <ukbeast89@protonmail.com>
- Update to version 1.20.3:
+ No changes.
* Mon May 09 2022 Antonio Larrosa <alarrosa@suse.com>
- Update to version 1.20.2:
+ rtspclientsink: fix possible shutdown deadlock in
collect_streams()
+ Minor spelling fixes
* Wed Apr 06 2022 Antonio Larrosa <alarrosa@suse.com>
- Remove BuildRequires: hotdoc and disable the doc generation.
It's really not used at all.
* Fri Mar 18 2022 Antonio Larrosa <alarrosa@suse.com>
- Update to version 1.20.1:
+ Fix race in rtsp-client when tunneling over HTTP
* Wed Feb 09 2022 Bjørn Lie <bjorn.lie@gmail.com>
- Update to version 1.20.0:
+ GstRTSPMediaFactory gained API to disable RTCP
(gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp"
property). Previously RTCP was always allowed for all RTSP
medias. With this change it is possible to disable RTCP
completely, irrespective of whether the client wants to do RTCP
or not.
+ Make a mount point of / work correctly. While not allowed by
the RTSP 2 spec, the RTSP 1 spec is silent on this and it is
used in the wild. It is now possible to use / as a mount path
in gst-rtsp-server, e.g. rtsp://example.com/ would work with
this now. Note that query/fragment parts of the URI are not
necessarily correctly handled, and behaviour will differ
between various client/server implementations; so use it if you
must but don't bug us if it doesn't work with third party
clients as you'd hoped.
+ multithreading fixes (races, refcounting issues, deadlocks).
+ ONVIF audio backchannel fixes.
+ ONVIF trick mode optimisations.
+ rtspclientsink: new "update-sdp" signal that allows updating
the SDP before sending it to the server via ANNOUNCE. This can
be used to add additional metadata to the SDP, for example. The
order and number of medias must not be changed, however.
* Fri Feb 04 2022 Bjørn Lie <bjorn.lie@gmail.com>
- Update to version 1.18.6:
+ rtsp-stream: fix get_rates raciness
+ rtsp-media: Only unprepare a media if it was not already
unpreparing anyway
+ rtsp-media: Unprepare suspended medias too
+ rtsp-client: make sure sessmedia will not get freed while used
+ rtsp-media: Also mark receive-only (RECORD) medias as prepared
when unsuspending
+ rtsp-session: Don't unref medias twice if it is removed inside
+ examples: Fix leak in appsrc2 example
- Drop service, use source url, upstream changes in git.
* Thu Jan 20 2022 Dominique Leuenberger <dimstar@opensuse.org>
- Fix parameters passed to meson: with meson 60, the parameters are
strictly checked, which helps in identifying those wrong
parameters.
* Wed Sep 15 2021 Bjørn Lie <bjorn.lie@gmail.com>
- Update to version 1.18.5:
+ rtsp-media:
- Ensure the bus watch is removed during unprepare
- Add one more case to seek avoidance
- Improve skipping trickmode seek
+ Fix a few memory leaks
* Wed Mar 31 2021 Antonio Larrosa <alarrosa@suse.com>
- Update to version 1.18.4:
+ rtspclientsink: fix deadlock on shutdown if no data has been
received yet
+ rtspclientsink: fix leaks in unit tests
+ rtsp-stream: avoid deadlock in send_func
+ rtsp-client: cleanup transports during TEARDOWN
* Sat Jan 16 2021 Bjørn Lie <bjorn.lie@gmail.com>
- Update to version 1.18.3:
+ rtsp-media: Only count senders when counting blocked streams
+ rtsp-client: Only unref client watch context on finalize, to
avoid deadlock
/usr/include/gstreamer-1.0/gst/rtsp-server /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-address-pool.h /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-auth.h /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-client.h /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-context.h /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-media-factory-uri.h /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-media-factory.h /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-media.h /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-mount-points.h /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-onvif-client.h /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-onvif-media-factory.h /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-onvif-media.h /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-onvif-server.h /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-params.h /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-permissions.h /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-sdp.h /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-server-object.h /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-server-prelude.h /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-server.h /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-session-media.h /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-session-pool.h /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-session.h /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-stream-transport.h /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-stream.h /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-thread-pool.h /usr/include/gstreamer-1.0/gst/rtsp-server/rtsp-token.h /usr/lib/gstreamer-1.0/libgstrtspclientsink.so /usr/lib/libgstrtspserver-1.0.so /usr/lib/pkgconfig/gstreamer-rtsp-server-1.0.pc /usr/share/doc/packages/gstreamer-rtsp-server-devel /usr/share/doc/packages/gstreamer-rtsp-server-devel/ChangeLog /usr/share/doc/packages/gstreamer-rtsp-server-devel/README /usr/share/gir-1.0/GstRtspServer-1.0.gir
Generated by rpm2html 1.8.1
Fabrice Bellet, Sat Nov 16 01:18:36 2024