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|Name: gstreamer-rtsp-server-devel||Distribution: openSUSE:Factory:zSystems|
|Version: 1.20.2||Vendor: obs://build.opensuse.org/openSUSE:Factory:zSystems|
|Release: 1.1||Build date: Sun May 15 03:58:03 2022|
|Group: Development/Languages/C and C++||Build host: s390zl29|
|Size: 1633999||Source RPM: gstreamer-rtsp-server-1.20.2-1.1.src.rpm|
|Summary: Development files for the GStreamer-based RTSP server library|
Development files for the GStreamer library for building an RTSP server.
* Mon May 09 2022 Antonio Larrosa <firstname.lastname@example.org> - Update to version 1.20.2: + rtspclientsink: fix possible shutdown deadlock in collect_streams() + Minor spelling fixes * Wed Apr 06 2022 Antonio Larrosa <email@example.com> - Remove BuildRequires: hotdoc and disable the doc generation. It's really not used at all. * Fri Mar 18 2022 Antonio Larrosa <firstname.lastname@example.org> - Update to version 1.20.1: + Fix race in rtsp-client when tunneling over HTTP * Wed Feb 09 2022 Bjørn Lie <email@example.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 <firstname.lastname@example.org> - 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 <email@example.com> - 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 <firstname.lastname@example.org> - 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 <email@example.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 <firstname.lastname@example.org> - 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 <email@example.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 <firstname.lastname@example.org> - 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 <email@example.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 <firstname.lastname@example.org> - 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 <email@example.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 <firstname.lastname@example.org> - Update to version 1.16.1: + See main gstreamer package for changelog. * Tue Jun 25 2019 Bjørn Lie <email@example.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 <firstname.lastname@example.org> - 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/lib64/gstreamer-1.0/libgstrtspclientsink.so /usr/lib64/libgstrtspserver-1.0.so /usr/lib64/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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