Index index by Group index by Distribution index by Vendor index by creation date index by Name Mirrors Help Search

gstreamer-rtsp-server-devel-1.16.2-lp152.2.2 RPM for x86_64

From OpenSuSE Leap 15.2 for x86_64

Name: gstreamer-rtsp-server-devel Distribution: openSUSE Leap 15.2
Version: 1.16.2 Vendor: openSUSE
Release: lp152.2.2 Build date: Sat May 16 19:56:42 2020
Group: Development/Languages/C and C++ Build host: lamb69
Size: 1469259 Source RPM: gstreamer-rtsp-server-1.16.2-lp152.2.2.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.

Provides

Requires

License

LGPL-2.0-or-later

Changelog

* 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.
* Wed Oct 03 2018 bjorn.lie@gmail.com
  - Update to version 1.14.4:
    + Bugfix release, please see .changes in gstreamer main package.
* Wed Sep 26 2018 bjorn.lie@gmail.com
  - Update to version 1.14.3:
    + Bugfix release, please see .changes in gstreamer main package.
* Tue Jul 24 2018 bjorn.lie@gmail.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 bjorn.lie@gmail.com
  - 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 bjorn.lie@gmail.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 bjorn.lie@gmail.com
  - 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 bjorn.lie@gmail.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 bjorn.lie@gmail.com
  - 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 dimstar@opensuse.org
  - 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 dimstar@opensuse.org
  - Modernize spec-file by calling spec-cleaner
* Mon Feb 12 2018 bjorn.lie@gmail.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 zaitor@opensuse.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 zaitor@opensuse.org
  - Update to version 1.12.4:
    + Bugs fixed: bgo#789646, bgo#769521.
* Mon Sep 18 2017 zaitor@opensuse.org
  - Update to version 1.12.3:
    + Bugs fixed: bgo#784094, bgo#786457.
* Fri Jul 14 2017 zaitor@opensuse.org
  - Update to version 1.12.2:
    + No changes, stable version bump only.
* Wed Jun 21 2017 zaitor@opensuse.org
  - Update to version 1.12.1:
    + No changes, stable version bump only.
* Wed May 10 2017 zaitor@opensuse.org
  - 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 zaitor@opensuse.org
  - Update to version 1.11.2:
    + Meson build fixes.
    + Minor changes and fixes.
* Thu Feb 23 2017 zaitor@opensuse.org
  - 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 zaitor@opensuse.org
  - Update to version 1.10.4:
    + Minor tweaks and fixes.
* Mon Jan 30 2017 zaitor@opensuse.org
  - Update to version 1.10.3:
    + Bugs fixed: bgo#755329, bgo#776343, bgo#776345.
* Sun Jan 01 2017 jengelh@inai.de
  - Summary updates.
* Sat Dec 03 2016 zaitor@opensuse.org
  - Update to version 1.10.2:
    + Bugs fixed: bgo#765673, bgo#770239.
* Sun Nov 27 2016 zaitor@opensuse.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 zaitor@opensuse.org
  - 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 zaitor@opensuse.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 zaitor@opensuse.org
  - Update to version 1.8.1:
    + No changes, version bump only.
* Sat Mar 26 2016 zaitor@opensuse.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 zaitor@opensuse.org
  - Update to version 1.6.2:
    + rtsp-server: Change the logic so we don't pop a NULL context.
* Sun Nov 01 2015 zaitor@opensuse.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 zaitor@opensuse.org
  - 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 zaitor@opensuse.org
  - Update to version 1.4.5:
    + rtsp-stream: leaks srtp decoder when leaving rtpbin
      (bgo#739481).
* Fri Nov 14 2014 zaitor@opensuse.org
  - 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).

Files

/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


Generated by rpm2html 1.8.1

Fabrice Bellet, Sat Mar 9 11:51:54 2024