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| Name: gstreamer-rtsp-server-devel | Distribution: openSUSE Tumbleweed |
| Version: 1.20.2 | Vendor: openSUSE |
| Release: 1.1 | Build date: Sat May 14 23:29:13 2022 |
| Group: Development/Languages/C and C++ | Build host: lamb09 |
| Size: 1641641 | Source RPM: gstreamer-rtsp-server-1.20.2-1.1.src.rpm |
| Packager: https://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
* 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
* Thu Dec 10 2020 Bjørn Lie <bjorn.lie@gmail.com>
- Update to version 1.18.2:
+ stream: collect a clock_rate when blocking
+ media:
- Ignore GstRTSPStreamBlocking from incomplete streams, to
prevent cases with prerolling when the inactive stream
prerolls first and the server proceeds without waiting for
the active stream. When there are no complete streams (during
DESCRIBE), we will listen to all streams.
- Use guint64 for setting the size-time property on rtpstorage,
fixes potential crashes or memory corruption.
- Get rates only on sender streams, fixing issue with ONVIF
audio backchannel streams
- Plug memory leak
- Fix the _service file and spec to really use the tarball
generated by service.
* Wed Oct 28 2020 Antonio Larrosa <alarrosa@suse.com>
- Update to 1.18.1:
+ Highlighted bugfixes in 1.18.1
- important security fixes
- bug fixes and memory leak fixes
- various stability and reliability improvements
+ gst-rtsp-server changes:
- rtsp-stream: collect rtp info when blocking
- rtsp-media: set a 0 storage size for TCP receivers
- rtsp-stream: preroll on gap events
- rtsp-media: do not unblock on unsuspend
* Thu Sep 17 2020 Antonio Larrosa <alarrosa@suse.com>
- Update to 1.18.0:
+ Highlights:
- GstTranscoder: new high level API for applications to
transcode media files from one format to another
- High Dynamic Range (HDR) video information representation
and signalling enhancements
- Instant playback rate change support
- Active Format Description (AFD) and Bar Data support
- RTSP server and client implementations gained ONVIF trick
modes support
- Hardware-accelerated video decoding on Windows via
DXVA2/Direct3D11
- Microsoft Media Foundation plugin for video capture and
hardware-accelerated video encoding on Windows
- qmlgloverlay: New overlay element that renders a QtQuick
scene over the top of an input video stream
- imagesequencesrc: New element to easily create a video
stream from a sequence of jpeg or png images
- dashsink: New sink to produce DASH content
- dvbsubenc: New DVB Subtitle encoder element
- MPEG-TS muxing now also supports TV broadcast compliant
muxing with constant bitrate muxing and SCTE-35 support
- rtmp2: New RTMP client source and sink element from-scratch
implementation
- svthevcenc: New SVT-HEVC-based H.265 video encoder
- vaapioverlay: New compositor element using VA-API
- rtpmanager gained support for Google's Transport-Wide
Congestion Control (twcc) RTP extension
- splitmuxsink and splitmuxsrc gained support for auxiliary
video streams
- webrtcbin now contains some initial support for
renegotiation involving stream addition and removal
- RTP support was enhanced with new RTP source and sink
elements to easily set up RTP streaming via rtp:// URIs
- avtp: New Audio Video Transport Protocol (AVTP) plugin for
Time-Sensitive Applications
- Support for the Video Services Forum's Reliable Internet
Stream Transport (RIST) TR-06-1 Simple Profile
- Universal Windows Platform (UWP) support
- rpicamsrc: New element for capturing from the Raspberry Pi
camera
- RTSP Server TCP interleaved backpressure handling
improvements as well as support for Scale/Speed headers
- GStreamer Editing Services gained support for nested
timelines, per-clip speed rate control and the OpenTimelineIO
format.
- Autotools build system has been removed in favour of Meson
- Drop patches already included upstream:
* gst-rtsp-Fix-NULL-pointer.patch
* gst-rtsp-fix-token-leak.patch
* gst-rtsp-replace-G_TYPE_INSTANCE_GET_PRIVATE.patch
* Sun Apr 12 2020 Bjørn Lie <bjorn.lie@gmail.com>
- Fix boo#1168026, CVE-2020-6095 and TALOS-2020-1018:
+ Add gst-rtsp-Fix-NULL-pointer.patch: rtsp-auth: Fix NULL
pointer dereference when handling an invalid basic
Authorization header.
- Add upstream bug fix patches:
+ Add gst-rtsp-fix-token-leak.patch: rtsp-auth: Fix default token
leak.
+ Add gst-rtsp-replace-G_TYPE_INSTANCE_GET_PRIVATE.patch:
rtsp-latency-bin: replace G_TYPE_INSTANCE_GET_PRIVATE as it's
been deprecated.
* Wed Dec 04 2019 Bjørn Lie <bjorn.lie@gmail.com>
- Update to version 1.16.2:
+ rtsp-media: Use lock in gst_rtsp_media_is_receive_only
+ rtsp-client:
- RTP Info when completed_sender
- Fix location uri-format by getting uri directly from context
instead
* Tue Sep 24 2019 Bjørn Lie <bjorn.lie@gmail.com>
- Update to version 1.16.1:
+ See main gstreamer package for changelog.
* Tue Jun 25 2019 Bjørn Lie <bjorn.lie@gmail.com>
- Update to version 1.16.0:
+ Highlights:
- GStreamer WebRTC stack gained support for data channels for
peer-to-peer communication based on SCTP, BUNDLE support,
as well as support for multiple TURN servers.
- AV1 video codec support for Matroska and QuickTime/MP4
containers and more configuration options and supported
input formats for the AOMedia AV1 encoder
- Support for Closed Captions and other Ancillary Data in video
- Support for planar (non-interleaved) raw audio
- GstVideoAggregator, compositor and OpenGL mixer elements are
now in -base
- New alternate fields interlace mode where each buffer carries
a single field
- WebM and Matroska ContentEncryption support in the Matroska
demuxer
- new WebKit WPE-based web browser source element
- Video4Linux: HEVC encoding and decoding, JPEG encoding, and
improved dmabuf import/export
- Hardware-accelerated Nvidia video decoder gained support for
VP8/VP9 decoding, whilst the encoder gained support for
H.265/HEVC encoding.
- Many improvements to the Intel Media SDK based
hardware-accelerated video decoder and encoder plugin
(msdk): dmabuf import/export for zero-copy integration with
other components; VP9 decoding; 10-bit HEVC encoding; video
post-processing (vpp) support including deinterlacing; and
the video decoder now handles dynamic resolution changes.
- The ASS/SSA subtitle overlay renderer can now handle multiple
subtitles that overlap in time and will show them on screen
simultaneously
- The Meson build is now feature-complete (*) and it is now the
recommended build system on all platforms. The Autotools
build is scheduled to be removed in the next cycle.
- The GStreamer Rust bindings and Rust plugins module are now
officially part of upstream GStreamer.
- The GStreamer Editing Services gained a gesdemux element
that allows directly playing back serialized edit list with
playbin or (uri)decodebin
- Many performance improvements.
- Updated options passed to meson following upstream changes.
* Fri May 31 2019 Bjørn Lie <bjorn.lie@gmail.com>
- Update to version 1.14.5:
+ rtsp-client: Fix crash in close handler and remove timeout
GSource on cleanup.
+ rtsp-media:
- Handle set state when preparing.
- Fix race condition in finish_unprepare.
+ rtsp-stream:
- Use cached address when allocating sockets.
- Use seqnum-offset for rtpinfo.
- Add source elements to the pipeline before activation for
stream-status create message.
/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
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