gstreamer-plugins-rs-1.26.2-160000.2.2

- Update to version 1.26.2:
  + awstranslate: improve control over accumulator behavior
  + awstranslate: output buffer lists
  + cea608tott: make test text less shocking by having more cues as
    context
  + dav1ddec: Directly decode into downstream allocated buffers if
    possible
  + deny: Allow webpki-root-certs license
  + fmp4mux: Add support for AC-3 / EAC-3
  + fmp4mux: Use earliest PTS for the base media decode time (tfdt)
  + fmp4mux: Fix handling of negative DTS in composition time
    offset
  + fmp4mux: Write lmsg as compatible brand into the last fragment
  + mp4mux: add extra brands
  + mp4: avoid dumping test output into build directory
  + mp4: migrate to mp4-atom to check muxing
  + mp4: test the trak structure
  + gtk4: Update and adapt to texture builder API changes
  + gtk4: Initial colorimetry support
  + gtk4: Update default GTK4 target version to 4.10
  + rtp: Update to bitstream-io 4.0
  + skia: Implement a video compositor using skia
  + webrtc: addressing a few deadlocks
  + webrtc: Support for producer sessions targeted at a given
    consumer
  + webrtc: add new JanusVR source element
  + webrtc: janus: clean up and refactoring
  + webrtcsink: Use seq number instead of Uuid for discovery
  + webrtc: Make older peers less likely to crash when webrtcsrc is
    used
  + Fix or silence various new clippy warnings
  + Update Cargo.lock to fix duplicated target-lexicon
- Pass skia=disabled to meson setup, not yet available in openSUSE.

- Update to version 1.26.1:
  + Bump MSRV to 1.83
  + Allow any windows-sys version >= 0.52 and <= 0.59
  + aws/polly: add GstScaletempoTargetDurationMeta to output
    buffers
  + awstranslate: improve message posted on bus
  + cdg: typefind: Division by zero fix
  + cea708mux: Improve support for overflowing input captions
  + colordetect: Change to videofilter base class
  + dav1ddec:
  - Drain decoder on caps changes if necessary
  - Only update unknown parts of the upstream colorimetry and not
    all of it
  - Support RGB encoded AV1
  - Use downstream buffer pool for copying if video meta is not
    supported
  - Use max-frame-delay value from the decoder instead of
    calculating it
  - Use max-frame-delay value from the decoder instead of
    calculating it
  + doc: Update to latest way of generating hotdoc config files
  + Fix gtk4 compile
  + Fix various clippy 1.86 warnings and update gstreamer-rs /
    gtk-rs dependencies
  + fmp4mux:
  - Add a couple of minor new features
  - Add manual-split mode that is triggered by serialized
    downstream events
  - Add send-force-keyunit property
  - Fix latency configuration for properties set during
    construction
  - Improve split-at-running-time handling
  - Write a v0 tfdt box if the decode time is small enough
  + fmp4mux/mp4mux: Handle the case of multiple tags per taglist
    correctly
  + gstwebrtc-api: Add TypeScript type definitions, build ESM for
    broader compatibility, improve JSDocs
  + hlsmultivariantsink: Allow users to specify playlist and
    segment location
  + hlssink3: Add Support for NTP timestamp from buffer
  + livesync:
  - Notify in/out/drop/duplicate properties on change
  - Only notify drop/duplicate properties
  + meson: Require gst 1.18 features for dav1d
  + mp4mux: Don't write composition time offsets if they're all
    zero
  + mp4mux, fmp4mux: Use correct timescales for edit lists
  + mpegtslivesrc:
  - Increase threshold for PCR <-> PTS DISCONT
  - Use a separate mutex for the properties
  + mux: use smaller number of samples for testing
  + net/aws: punctuation-related improvements to our
    span_tokenize_items function
  + pcap_writer: Mark target-factory and pad-path props as
    construct-only
  + speechmatics: Handle multiple stream-start event
  + tracers: buffer-lateness: don't panic on add overflow + reduce
    graph legend entry font size a bit
  + tracers: Update to etherparse 0.17
  + transcriberbin: make auto passthrough work when transcriber is
    a bin
  + ts-jitterbuffer: improve scheduling of lost events
  + tttocea708: fix origin-row handling for roll-up in CEA-708
  + Update dependencies
  + Update gtk-rs / gstreamer-rs dependencies and update for API
    changes
  + Update to bitstream-io 3
  + uriplaylistbin: skip cache test when offline
  + webrtc: Port to reqwest 0.12
  + webrtcsink: Fix compatibility with audio level header extension
- Pass vvdec=disabled to meson setup, libvvdec is currently not
  available in openSUSE.
- Disable aws plugin for all arches via bcond, fails to build.

- Update to version 0.13.5:
  + Fixed:
  - cdg: Fix typefind errors on specific file sizes.
  - cea608overlay:
    . Ensure lines are rendered in order.
    . Clear output on each switch.
  - cea608overlay / cea708overlay: Fix field lookup for S334-1A
    captions.
  - cea608tocea708: Fix S334-1A field flag usage.
  - closedcaption:
    . Fix rollup mode not always using the correct base row
    . Only increase dtvcc packet sequence if there are services.
  - fmp4mux:
    . Fix state cleanup on flush.
    . Handle language/orientation tags as per-stream tags.
  - hlssink3: Write playlist atomically.
  - inter: Don't leak hashmap objects.
  - mpegtslivesrc:
    . Handle zero-byte adaptation fields correctly.
    . Consider initial calibration of the clock.
    . Ignore NIT programs from the PAT.
  - onvifmetadatacombiner: Unset PTS/DTS of metadata.
  - rtpbasepay / rtpbasedepay: Only forward buffers after a
    segment event.
  - rtpac3depay2: Fix handling of non-fragmented payloads.
  - togglerecord: Drop locks before sending queries to avoid
    deadlocks.
  - tttocea708: Don't reset service writer for every incoming
    caption.
  - whipserversrc: Handle concurrent POSTs.
  + Added:
  - mpegtslivesrc: Take adaptation field discontinuity flag into
    account.
  - uriplaylistbin: Add caching support
  + Changed: - Updated various dependencies.

- Update to version 0.13.4:
  + Fixed:
  - cea608overlay: Fix rendering when roll-up base row is at the
    top.
  - cea708mux:
    . Handle CEA608 data correctly and output padding by default.
    . Clear leftover pending codes correctly.
  - cea708overlay:
    . Produce better CEA608 layouts.
    . Fix background/foreground types and enable black background
    by default.
    . Clear correctly on caption timeout.
  - mpegtslivesrc: Various fixes related to stream
    discontinuities.
  - tttocea708: Fix various conformance issues.
  - togglerecord: Fix various deadlocks and simplify mutexes.
  - webrtcsink:
    . Fix various deadlocks.
    . Set caps-change-mode=delayed on encoder capsfilter.
    . Ignore more fields on caps changes.
  + Added:
  - awss3putobjectsink: Add next-file support.
  - tracers: Add signal to force writing log file to queue-levels
    and buffer-lateness tracers.
  - webrtc: Handle some more Janus events.
  - webrtcsink: Add support for openh264enc and nvh265enc.
  - webrtcsrc: Add connect-to-first-producer property.

- Update to version 0.13.3:
  + Fixed:
  - gtk4paintablesink:
    . Don't check for a GL context when filtering dmabuf caps.
    . Use a correctly typed None value when retrieving paintable
    property fails.
  - mpegtslivesrc: Parse PAT/PMT to lock to a single program/PCR
    in case multiple are in the stream.
  - rtp: Fix reference timestamp meta de-duplication in
    depayloaders.
  - quinn: Specify a default crypto provider to avoid conflicts.
  - transcriberbin: Fix linking of user-provided transcriber.
  - webrtcsink:
    . Allow pixel-aspect-ratio changes.
    . Fix naming of error dot files of discovery pipelines.
    . Fix session not in place errors.
  - webrtc: janus: Do not block in end_session().
  + Added:
  - awstranscriber: Post warning message with details when items
    are too late.
  - transcriberbin: Support both latency and translate-latency
    properties.
  - webrtc: janus: Add janus-state property.
  + Changed: gtk4paintablesink: Deprecated "wayland" feature and
    call it "waylandegl" as it has nothing to do with generic
    Wayland support.

- Update to version 0.13.2:
  + Fixed:
  - cea608overlay: Avoid overflow when deciding which lines to
    retain.
  - cea708mux:
    . Actually push gap events downstream.
    . Stop with EOS once all pads are EOS.
    . Fix off-by-one when deciding if a buffer belongs to this or
    the next frame.
  - mpegtslivesrc: Various timestamp tracking fixes.
  - onvifmetadatapay: Set output caps earlier.
  - transcriberbin: Fix passthrough state change.
  - webrtcsink: Fix setting of RFC7273 attributes in the SDP.
  + Added:
  - dav1ddec: Add properties for film grain synthesis and in-loop
    filters.
  - mpegtslivesrc: Handle PCR discontinuities.
  - rtpav1depay: Add wait-for-keyframe and request-keyframe
    properties.
  - webrtcsrc: Expose msid property on source pads.
  + Changed: spotify: Reduce dependencies.

- Update to version 0.13.1:
  + Fixed:
  - Various new clippy warnings.
  - awstranscriber: Fix sanity check in transcribe loop.
  - gtk4paintablesink: Move dmabuf cfg to the correct bracket
    level.
  - mpegtslivesrc: Handle PCR discontinuities as errors.
  - ndisrc: Calculate timestamps for metadata buffers too.
  - rtpbasepay: Various fixes to payloader base class.
  - transcriberbin: Fix gst-inspect with missing elements.
  - webrtcsink:
    . Fix segment format mismatch when using a remote offer.
    . Fix various assertions when finalizing.
  - webrtcsrc:
    . Don't hold the state lock while removing sessions.
    . Make sure to always call end_session() without state lock.
  - whepsrc: Fix incorrect default caps.
  + Changed:
  - gtk4paintablesink: Enable
    gtk::GraphicsOffload::black-background when building with GTK
    4.16 or newer.
  - gstwebrtc-api: Always include index file in dist for
    convenience.
  - rtpbasepay: Negotiate SSRC/PT with downstream via caps for
    backwards compatibility.
  - hlssink3: Use more accurate fragment duration from
    splitmuxsink if available.
  + Added:
  - gtk4paintablesink:
    . Add window-width and window-height properties.
    . Add custom widget for automatically updating window size.
  - fmp4mux / mp4mux: Add image orientation tag support.
  - webrtcsink: Add nvv4l2av1enc support.
  - cmafmux: Add Opus support.

- Update to version 0.13.0:
  * Added
  - rtp: New RTP payloader and depayloader base classes, in
    addition to new payloader and depayloaders for: PCMA, PCMU,
    AC-3, AV1 (ported to the new base classes), MPEG-TS, VP8,
    VP9, MP4A, MP4G, JPEG, Opus, KLV.
  - originalbuffer: New pair of elements that allows to save a
    buffer, perform transformations on it and then restore the
    original buffer but keeping any new analytics and other
    metadata on it.
  - gopbuffer: New element for buffering an entire
    group-of-pictures.
  - tttocea708: New element for converting timed text to CEA-708
    closed captions.
  - cea708mux: New element for muxing multiple CEA-708 services
    together.
  - transcriberbin: Add support for generating CEA-708 closed
    captions and CEA-608-in-708.
  - cea708overlay: New overlay element for CEA-708 and CEA-608
    closed captions.
  - dav1ddec: Signal colorimetry in the caps.
  - webrtc: Add support for RFC7273 clock signalling and
    synchronization to webrtcsrc and webrtcsink.
  - tracers: Add a new pad push durations tracer.
  - transcriberbin: Add support for a secondary audio stream.
  - quinn: New plugin with a QUIC source and sink element.
  - rtpgccbwe: New mode based on linear regression instead of a
    kalman filter.
  - rtp: New rtpsend and rtprecv elements that provide a new
    implementation of the rtpbin element with a separate send and
    receive side.
  - rtpsrc2: Add support for new rtpsend / rtprecv elements
    instead of rtpbin.
  - webrtcsrc: Add multi-producer support.
  - livesync: Add sync property for enabling/disabling syncing of
    the output buffers to the clock.
  - mpegtslivesrc: New element for receiving an MPEG-TS stream,
    e.g. over SRT or UDP, and exposing the remote PCR clock as a
    local GStreamer clock.
  - gtk4paintablesink: Add support for rotations / flipping.
  - gtk4paintablesink: Add support for RGBx formats in non-GL
    mode.
  * Fixed
  - livesync: Queue up to latency buffers instead of requiring a
    queue of the same size in front of livesync.
  - livesync: Synchronize the first buffer to the clock too.
  - livesync: Use correct duration for deciding whether a filler
    has to be inserted or not.
  - audioloudnorm: Fix possible off-by-one in the limiter when
    handling the very last buffer.
  - webrtcsink: Fix property types for rav1enc.
  * Changed
  - sccparse, mccparse: Port from nom to winnow.
  - uriplaylistbin: Rely on uridecodebin3 gapless logic instead
    of re-implementing it.
  - webrtc: Refactor of JavaScript API.
  - janusvrwebrtcsink: New use-string-ids property to distinguish
    between integer and string room IDs, instead of always using
    strings and guessing what the server expects.
  - janusvrwebrtcsink: Handle more events and expose some via
    signals.
  - dav1ddec: Require dav1d 1.3.0.
  - closedcaption: Drop libcaption C code and switch to a pure
    Rust implementation.
- Update to version 0.12.7:
  * Fixed
  - aws, spotifyaudiosrc, reqwesthttpsrc, webrtchttp: Fix race
    condition when unlocking
  - rtp: Allow any payload type for the AV1 RTP
    payloader/depayloader
  - rtp: Various fixes to the AV1 RTP payloader/depayloader to
    work correctly with Chrome and Pion
  - meson: Various fixes to the meson-based build system around
    cargo
  - webrtcsink: Use correct property names for configuring
    av1enc
  - webrtcsink: Avoid lock poisoning when setting encoder
    properties
  * Added
  - ndi: Support for NDI SDK v6
  - webrtcsink: Support for AV1 via nvav1enc, av1enc or rav1enc
  * Changed
  - Update to async-tungstenite 0.26
- Update to version 0.12.6:
  * Fixed
  - Various Rust 1.78 clippy warnings.
  - gtk4paintablesink: Fix plugin description.
  * Added
  - fmp4mux / mp4mux: Add support for adding AV1 header OBUs into
    the MP4 headers.
  - fmp4mux / mp4mux: Take track language from the tags if
    provided.
  - gtk4paintablesink: Add GST_GTK4_WINDOW_FULLSCREEN environment
    variable to create a fullscreen window for debugging
    purposes.
  - gtk4paintablesink: Also create a window automatically when
    called from gst-play-1.0.
  - webrtc: Add support for insecure TLS connections.
  - webrtcsink: Add VP9 parser after the encoder.
  * Changed
  - webrtcsink: Improve error when no discovery pipeline runs.
  - rtpgccbwe: Improve debug output in various places.
- Update to version 0.12.5:
  * Fixed
  - hrtfrender: Use a bitmask instead of an int in the caps for
    the channel-mask.
  - rtpgccbwe: Don't log an error when pushing a buffer list
    fails while stopping.
  - webrtcsink: Don't panic in bitrate handling with unsupported
    encoders.
  - webrtcsink: Don't panic if unsupported input caps are used.
  - webrtcsrc: Allow a None producer-id in request-encoded-filter
    signal.
  * Added
  - aws: New property to support path-style addressing.
  - fmp4mux / mp4mux: Support FLAC instead (f)MP4.
  - gtk4: Support directly importing dmabufs with GTK 4.14.
  - gtk4: Add force-aspect-ratio property similar to other video
    sinks.
- Update to version 0.12.4:
  * Fixed
  - aws: Use fixed behaviour version to ensure that updates to
    the AWS SDK don't change any defaults configurations in
    unexpected ways.
  - onvifmetadataparse: Fix possible deadlock on shutdown.
  - webrtcsink: Set perfect-timestamp=true on audio encoders to
    work around bugs in Chrome's audio decoders.
  - Various clippy warnings.
  * Changed
  - reqwest: Update to reqwest 0.12.
  - webrtchttp: Update to reqwest 0.12.
- Update to version 0.12.3:
  * Fixed
  - gtk4paintablesink: Fix scaling of texture position.
  - janusvrwebrtcsink: Handle 64 bit numerical room ids.
  - janusvrwebrtcsink: Don't include deprecated audio/video
    fields in publish messages.
  - janusvrwebrtcsink: Handle various other messages to avoid
    printing errors.
  - livekitwebrtc: Fix shutdown behaviour.
  - rtpgccbwe: Don't forward buffer lists with buffers from
    different SSRCs to avoid breaking assumptions in rtpsession.
  - sccparse: Ignore invalid timecodes during seeking.
  - webrtcsink: Don't try parsing audio caps as video caps.
  * Changed
  - webrtc: Allow resolution and framerate changes.
  - webrtcsrc: Make producer-peer-id optional.
  * Added
  - livekitwebrtcsrc: Add new LiveKit source element.
  - regex: Add support for configuring regex behaviour.
  - spotifyaudiosrc: Document how to use with non-Facebook
    accounts.
  - webrtcsrc: Add do-retransmission property.

- Update to version 0.12.2:
  * Fixed
  - rtpgccbwe: Don't reset PTS/DTS to None as otherwise
    rtpsession won't be able to generate valid RTCP.
  - webrtcsink: Fix usage with 1.22.
  * Added
  - janusvrwebrtcsink: Add secret-key property.
  - janusvrwebrtcsink: Allow for string room ids and add
    string-ids property.
  - textwrap: Don't split on all whitespaces, especially not on
    non-breaking whitespace.
- Update to version 0.12.1:
  * Added
  - gtk4: Create a window for testing purposes when running in
    gst-launch-1.0 or if GST_GTK4_WINDOW=1 is set.
  - webrtcsink: Add msid property.
- Update to version 0.12.0:
  * Changed
  - ndi: ndisrc passes received data downstream without an
    additional copy, if possible.
  - webrtc: Cleanups to webrtcsrc/sink default signalling
    protocol, JavaScript implementation and server
    implementation.
  - webrtc: whipwebrtcsink is renamed to whipclientsink and
    deprecate old whipsink.
  * Fixed
  - gtk4: Fix Windows build when using EGL.
  - gtk4: Fix ARGB pre-multiplication with GTK 4.14. This
    requires building with the gtk_v4_10 or even better gtk_v4_14
    feature.
  - gtk4: Fix segfault if GTK3 is used in the same process.
  - gtk4: Always draw background behind the video frame and not
    only when borders have to be added to avoid glitches.
  - livekitwebrtcsink: Add high-quality layer for video streams.
  - webrtc: Fix potential hang and fd leak in signalling server.
  - webrtc: Fix closing of WebSockets.
  - webrtchttp: Allow setting None for audio/video caps for WHEP.
  * Added
  - New awss3putobjectsink that works similar to awss3sink but
    with a different upload strategy.
  - New hlscmafsink element for writing HLS streams with
    CMAF/ISOBMFF fragments.
  - New inter plugin with intersink / intersrc elements that
    allow to connect different pipelines in the same process.
  - New janusvrwebrtcsink element for the Janus VideoRoom API.
  - New rtspsrc2 element.
  - New whipserversrc element.
  - gtk4: New background-color property for setting the color of
    the background of the frame and the borders, if any.
  - gtk4: New scale-filter property for defining how to scale the
    frames.
  - livesync: Add support for image formats.
  - ndi: Closed Caption support in ndisrc / ndisink.
  - textwrap: Add support for gaps.
  - tracers: Optionally only show late buffers in buffer-lateness
    tracer.
  - webrtc: Add support for custom headers.
  - webrtcsink: New payloader-setup signal to configure payloader
    elements.
  - webrtcsrc: Support for navigation events.

- Increase memory to 12G for s390x builds in _constraints
  (bsc#1218563).