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|Name: gstreamer-rtsp-server-devel||Distribution: SUSE Linux Enterprise 15 SP4|
|Version: 1.20.1||Vendor: openSUSE|
|Release: bp154.1.76||Build date: Thu May 12 03:20:59 2022|
|Group: Development/Languages/C and C++||Build host: obs-arm-10|
|Size: 1651438||Source RPM: gstreamer-rtsp-server-1.20.1-bp154.1.76.src.rpm|
|Summary: Development files for the GStreamer-based RTSP server library|
Development files for the GStreamer library for building an RTSP server.
* 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. * Wed Oct 03 2018 email@example.com - Update to version 1.14.4: + Bugfix release, please see .changes in gstreamer main package. * Wed Sep 26 2018 firstname.lastname@example.org - Update to version 1.14.3: + Bugfix release, please see .changes in gstreamer main package. * Tue Jul 24 2018 email@example.com - Update to version 1.14.2: + rtsp-media: - unref clock (if set) when finalizing. - add gst_rtsp_media_*_set_clock to docs. + media-factory: - unref old clock when setting new clock. - unref clock in finalize. + rtsp-onvif-media: - fix g-ir-scanner warnings. - export gst_rtsp_onvif_media_factory_requires_backchannel. + client: Strip transport parts as whitespaces could be around commas. + rtsp-stream: avoid pushing data on unlinked udpsrc pad during setup. + rtspclientsink: fix waiting for multiple streams. * Sat Jun 23 2018 firstname.lastname@example.org - Switch to meson build system: + Add meson, pkgconfig(glib-2.0),pkgconfig(gstreamer-app-1.0), pkgconfig(gstreamer-net-1.0), pkgconfig(gstreamer-rtp-1.0), pkgconfig(gstreamer-rtsp-1.0) and pkgconfig(gstreamer-sdp-1.0) BuildRequires. + Add meson macros, replacing autotools ones. + Pass disable_introspection=false, with-package-name='openSUSE GStreamer-rtsp-server package', with-package-origin='http://download.opensuse.org' and tests=false and examples=false to meson, ensure we build the features we want. Tests have always been disabled, be explicit about it, as they need a working network connection. + Drop pkgconfig(gstreamer-plugins-base-1.0) BuildRequires. + No longer rm la files, not needed when building with meson. * Fri Jun 22 2018 email@example.com - Drop gstreamer-plugins-good and pkgconfig(gstreamer-plugins-bad-1.0) BuildRequires: Only needed for unit tests and we do not build or run those tests. * Sun May 20 2018 firstname.lastname@example.org - Update to version 1.14.1: + GstPad: Fix race condition causing the same probe to be called multiple times + Fix occasional deadlocks on windows when outputting debug logging + Fix debug levels being applied in the wrong order + GIR annotation fixes for bindings + audiomixer, audioaggregator: fix some negotiation issues + gst-play-1.0: fix leaving stdin in non-blocking mode after exit + flvmux: wait for caps on all input pads before writing header even if source is live + flvmux: don't wake up the muxer unless there is data, fixes busy looping if there's no input data + flvmux: fix major leak of input buffers + rtspsrc, rtsp-server: revert to RTSP RFC handling of sendonly/recvonly attributes + rtpvrawpay: fix payloading with very large mtu sizes where everything fits into a single RTP packet + v4l2: Fix hard-coded enabled v4l2 probe on Linux/ARM + v4l2: Disable DMABuf for emulated formats when using libv4l2 + v4l2: Always set colorimetry in S_FMT + asfdemux: Set stream-format field for H264 streams and handle H.264 in bytestream format + x265enc: Fix tagging of keyframes on output buffers + ladspa: Fix critical during plugin load on Windows + decklink: Fix COM initialisation on Windows + h264parse: fix re-use across pipeline stop/restart + mpegtsmux: fix force-keyframe event handling and PCR/PMT changes that would confuse some players with generated HLS streams + adaptivedemux: Support period change in live playlist + rfbsrc: Fix support for applevncserver and support NULL pool in decide_allocation + jpegparse: Fix APP1 marker segment parsing + h265parse: Make caps writable before modifying them, fixes criticals + fakevideosink: request an extra buffer if enable-last-sample is enabled + wasapisrc: Don't provide a clock based on WASAPI's clock + wasapi: Only use audioclient3 when low-latency, as it might otherwise glitch with slow CPUs or VMs + wasapi: Don't derive device period from latency time, should make it more robust against glitches + audiolatency: Fix wave detection in buffers and avoid bogus pts values while starting + msdk: fix plugin load on implementations with only HW support + msdk: dec: set framerate to the driver only if provided, not in 0/1 case + msdk: Don't set extended coding options for JPEG encode + rtponviftimestamp: fix state change function init/reset causing races/crashes on shutdown + decklink: fix initialization failure in windows binary + ladspa: Fix critical warnings during plugin load on Windows and fix dependencies in meson build + gl: fix cross-compilation error with viv-fb + qmlglsink: make work with eglfs_kms + rtspclientsink: Don't deadlock in preroll on early close + rtspclientsink: Fix client ports for the RTCP backchannel + rtsp-server: Fix session timeout when streaming data to client over TCP + vaapiencode: h264: find best profile in those available, fixing negotiation errors + vaapi: remove custom GstGL context handling, use GstGL instead. Fixes GL Context sharing with WebkitGtk on wayland + gst-editing-services: various fixes + gst-python: bump pygobject req to 3.8; fix GstPad.set_query_function(); dist autogen.sh and configure.ac in tarball + g-i: pick up GstVideo-1.0.gir from local build directory in GstGL build + g-i: update constant values for bindings + avoid duplicate symbols in plugins across modules in static builds + ... and many, many more! * Tue Apr 17 2018 email@example.com - Update to version 1.14.0: + Highlights: - WebRTC support: real-time audio/video streaming to and from web browsers; - Experimental support for the next-gen royalty-free AV1 video codec - Video4Linux: encoding support, stable element names and faster device probing; - Support for the Secure Reliable Transport (SRT) video streaming protocol; - RTP Forward Error Correction (FEC) support (ULPFEC); - RTSP 2.0 support in rtspsrc and gst-rtsp-server; - ONVIF audio backchannel support in gst-rtsp-server and rtspsrc; - playbin3 gapless playback and pre-buffering support; - Tee, our stream splitter/duplication element, now does allocation query aggregation which is important for efficient data handling and zero-copy; - QuickTime muxer has a new prefill recording mode that allows file import in Adobe Premiere and FinalCut Pro while the file is still being written; - rtpjitterbuffer fast-start mode and timestamp offset adjustment smoothing; - souphttpsrc connection sharing, which allows for connection reuse, cookie sharing, etc; - nvdec: new plugin for hardware-accelerated video decoding using the NVIDIA NVDEC API; - Adaptive DASH trick play support; - ipcpipeline: new plugin that allows splitting a pipeline across multiple processes; - Major gobject-introspection annotation improvements for large parts of the library API; - GStreamer C# bindings have been revived and seen many updates and fixes; - The externally maintained GStreamer Rust bindings had many usability improvements and cover most of the API now. Coinciding with the 1.14 release, a new release with the 1.14 API additions is happening. + Updated translations. * Fri Mar 30 2018 firstname.lastname@example.org - Update to version 1.12.5: + Bugs fixed: bgo#789646, bgo#791743. - Drop upstream fixed patches: + gst-rtsp-server-add-annotations-and-API-guards.patch. + gst-rtsp-server-gst_rtsp_context_get_current.patch. + gst-rtsp-server-rtsp-client-add-type-annotations.patch. + gst-rtsp-server-Set-udpsink_out-ttl-mc-property.patch. * Mon Mar 26 2018 email@example.com - Drop pkgconfig(libcgroup) BuildRequires: libcgroup's functionality is largely deprecated by systemd and the two actually clash in some ways which cause bug reports. * Wed Feb 28 2018 firstname.lastname@example.org - Modernize spec-file by calling spec-cleaner * Mon Feb 12 2018 email@example.com - Add upstream bug fix patches: + gst-rtsp-server-rtsp-client-add-type-annotations.patch. + gst-rtsp-server-gst_rtsp_context_get_current.patch. + gst-rtsp-server-add-annotations-and-API-guards.patch. * Tue Jan 09 2018 firstname.lastname@example.org - Add gst-rtsp-server-Set-udpsink_out-ttl-mc-property.patch: rtsp: Set udpsink_out ttl-mc property on creation (bgo#791743). - Clean up spec, silence some rpmlint warnings. - Drop explicit libgstrtspserver-1_0-0 and typelib-1_0-GstRtspServer-1_0 Obsoletes and Provides: Not needed and only leads to a rpmlint warning. - Add gstreamer-rtsp-server-rpmlintrc: Filter out bogus warning about missing dependencies in devel package. * Mon Dec 11 2017 email@example.com - Update to version 1.12.4: + Bugs fixed: bgo#789646, bgo#769521. * Mon Sep 18 2017 firstname.lastname@example.org - Update to version 1.12.3: + Bugs fixed: bgo#784094, bgo#786457. * Fri Jul 14 2017 email@example.com - Update to version 1.12.2: + No changes, stable version bump only. * Wed Jun 21 2017 firstname.lastname@example.org - Update to version 1.12.1: + No changes, stable version bump only. * Wed May 10 2017 email@example.com - Update to version 1.12.0: + No changes, stable version bump only. - Changes from version 1.11.91: + gi: Fix some annotations and docstrings. + Automatic update of common submodule. - Changes from version 1.11.90: + examples: make test-launch pipeline shared by default as well. + gstreamer-rtsp-server: Add both srcdir and builddir to the include path. * Sat Feb 25 2017 firstname.lastname@example.org - Update to version 1.11.2: + Meson build fixes. + Minor changes and fixes. * Thu Feb 23 2017 email@example.com - Update to version 1.11.1: + Bugs fixed: bgo#758062, bgo#771830, bgo#774173, bgo#774640, bgo#776867, bgo#777037, bgo#774416. * Thu Feb 23 2017 firstname.lastname@example.org - Update to version 1.10.4: + Minor tweaks and fixes. * Mon Jan 30 2017 email@example.com - Update to version 1.10.3: + Bugs fixed: bgo#755329, bgo#776343, bgo#776345. * Sun Jan 01 2017 firstname.lastname@example.org - Summary updates. * Sat Dec 03 2016 email@example.com - Update to version 1.10.2: + Bugs fixed: bgo#765673, bgo#770239. * Sun Nov 27 2016 firstname.lastname@example.org - Update to version 1.10.1: + Meson update. - Changes from version 1.10.0: + Bugs fixed: bgo#771983, bgo#772478, bgo#773640. * Fri Aug 19 2016 email@example.com - Update to version 1.8.3 (boo#996937): + g-i: pass compiler env to g-ir-scanner. - Changes from version 1.8.2: + rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header. + rtsp-stream: - Fix crash on cleanup with shared media and multiple udpsrc. - Always bind to ANY when address is a multicast address and not only on Windows. - Rename package to gstreamer-rtsp-server. Align with the other gstreamer packages. Also obsolete and provide the previous ones to ease updates. * Wed Jun 15 2016 firstname.lastname@example.org - Update to version 1.8.1: + bgo#764744: Crashes when multiple udpsrc are created for each client on a shared media, misses tracking and cleanup. + bgo#766619: Space between ; and timeout= in session header is not RFC2326 compliant. * Thu Apr 21 2016 email@example.com - Update to version 1.8.1: + No changes, version bump only. * Sat Mar 26 2016 firstname.lastname@example.org - Update to version 1.8.0: + Hardware-accelerated zero-copy video decoding on Android + New video capture source for Android using the android.hardware.Camera API. + Windows Media reverse playback support (ASF/WMV/WMA). + New tracing system provides support for more sophisticated debugging tools. + New high-level GstPlayer playback convenience API. + Initial support for the new Vulkan API, see Matthew Waters' blog post for more details. + Improved Opus audio codec support: Support for more than two channels; MPEG-TS demuxer/muxer can now handle Opus; sample-accurate encoding/decoding/transmuxing with Ogg, Matroska, ISOBMFF (Quicktime/MP4), and MPEG-TS as container; new codec utility functions for Opus header and caps handling in pbutils library. The Opus encoder/decoder elements were also moved to gst-plugins-base (from -bad), and the opus RTP depayloader/payloader to -good. + GStreamer VAAPI module now released and maintained as part of the GStreamer project. + Asset proxy support in the GStreamer Editing Services. + Bugs fixed: bgo#740509. * Tue Dec 15 2015 email@example.com - Update to version 1.6.2: + rtsp-server: Change the logic so we don't pop a NULL context. * Sun Nov 01 2015 firstname.lastname@example.org - Update to version 1.6.1: + gst-rtsp-server: Retain reference to rtsp-media when preparing. + rtsp-stream: GstBin leak in udp-mcast case. - Changes from version 1.6.0: + For changelog, see mainpackage changes, everything is condensed there. - Drop grs-rtsp-fix-double-unlock-in_get_buffer_size.patch: Fixed upstream. * Wed Aug 05 2015 email@example.com - Add grs-rtsp-fix-double-unlock-in_get_buffer_size.patch: Fixes an abort when calling gst_rtsp_media_get_buffer_size() because of double g_mutex_unlock () usage (bgo#745434). * Fri Dec 26 2014 firstname.lastname@example.org - Update to version 1.4.5: + rtsp-stream: leaks srtp decoder when leaving rtpbin (bgo#739481). * Fri Nov 14 2014 email@example.com - Update to version 1.4.4: + rtsp-client: mikey memory leaks (bgo#739383). - Changes from version 1.4.3: + No changes. - Changes from version 1.4.2: + rtsp-media: Make sure that sequence numbers are monotonic after pause (bgo#736017). + rtsp-client: Protect saved clients watch with a mutex (bgo#735570).
/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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