gstreamer-rtsp-server-1.22.0-bp156.2.6

- Add patch to reduce the required meson version to 0.61.0 since
  that's what we have in SLE 15:
  * reduce-required-meson.patch

- Update to version 1.22.0:
  + Please see changes in gstreamer main package, major version
    bump.

- Update to version 1.20.5:
  + rtsp-server: Free client if no connection could be created

- Update to version 1.20.4:
  + gst-rtsp-server: Fix pushing backlog to client.
  + rtsp-server: stream: Don't loop forever if binding to the
    multicast address fails.

- Update to version 1.20.3:
  + No changes.

- Update to version 1.20.2:
  + rtspclientsink: fix possible shutdown deadlock in
    collect_streams()
  + Minor spelling fixes

- Remove BuildRequires: hotdoc and disable the doc generation.
  It's really not used at all.

- Update to version 1.20.1:
  + Fix race in rtsp-client when tunneling over HTTP

- 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.

- 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.

- Fix parameters passed to meson: with meson 60, the parameters are
  strictly checked, which helps in identifying those wrong
  parameters.

- 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

- 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

- 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

- 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.

- 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

- 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

- 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.

- 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

- Update to version 1.16.1:
  + See main gstreamer package for changelog.

- 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.

- 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.

- Update to version 1.14.4:
  + Bugfix release, please see .changes in gstreamer main package.

- Update to version 1.14.3:
  + Bugfix release, please see .changes in gstreamer main package.

- 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.

- 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.

- 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.

- 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!

- 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.

- Add gst-rtsp-Fix-NULL-pointer.patch: rtsp-auth: Fix NULL pointer
  dereference when handling an invalid basic Authorization header
  This fixes CVE-2020-6095 and TALOS-2020-1018 (boo#1168026).

- 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.

- Drop pkgconfig(libcgroup) BuildRequires: libcgroup's
  functionality is largely deprecated by systemd and the two
  actually clash in some ways which cause bug reports.

- Modernize spec-file by calling spec-cleaner

- 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.

- 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.

- Update to version 1.12.4:
  + Bugs fixed: bgo#789646, bgo#769521.

- Update to version 1.12.3:
  + Bugs fixed: bgo#784094, bgo#786457.

- Update to version 1.12.2:
  + No changes, stable version bump only.

- Update to version 1.12.1:
  + No changes, stable version bump only.

- 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.

- Update to version 1.11.2:
  + Meson build fixes.
  + Minor changes and fixes.

- Update to version 1.11.1:
  + Bugs fixed: bgo#758062, bgo#771830, bgo#774173, bgo#774640,
    bgo#776867, bgo#777037, bgo#774416.

- Update to version 1.10.4:
  + Minor tweaks and fixes.

- Update to version 1.10.3:
  + Bugs fixed: bgo#755329, bgo#776343, bgo#776345.

- Summary updates.

- Update to version 1.10.2:
  + Bugs fixed: bgo#765673, bgo#770239.

- Update to version 1.10.1:
  + Meson update.
- Changes from version 1.10.0:
  + Bugs fixed: bgo#771983, bgo#772478, bgo#773640.

- 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.

- 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.

- Update to version 1.8.1:
  + No changes, version bump only.