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|Name: gstreamer-rtsp-server-devel||Distribution: openSUSE Leap 15.2|
|Version: 1.16.2||Vendor: openSUSE|
|Release: lp152.2.1||Build date: Fri Apr 17 14:14:46 2020|
|Group: Development/Languages/C and C++||Build host: obs-arm-9|
|Size: 1493715||Source RPM: gstreamer-rtsp-server-1.16.2-lp152.2.1.src.rpm|
|Summary: Development files for the GStreamer-based RTSP server library|
Development files for the GStreamer library for building an RTSP server.
* 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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