Accepting request 1299095 from multimedia:libs

OBS-URL: https://build.opensuse.org/request/show/1299095
OBS-URL: https://build.opensuse.org/package/show/openSUSE:Factory/gstreamer-plugins-rs?expand=0&rev=38

_service

@@ -3,8 +3,8 @@
   <service name="obs_scm" mode="manual">
     <param name="url">https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs.git</param>
     <param name="filename">gst-plugins-rs</param>
-    <param name="versionformat">1.26.2</param>
-    <param name="revision">refs/tags/gstreamer-1.26.2</param>
+    <param name="versionformat">1.26.5</param>
+    <param name="revision">refs/tags/gstreamer-1.26.5</param>
     <param name="scm">git</param>
   </service>
   <service name="set_version" mode="manual"/>
@@ -17,5 +17,6 @@
 	  <param name="srcdir">gst-plugins-rs</param>
 	  <param name="compression">zst</param>
 	  <param name="update">false</param>
+	  <param name="i-accept-the-risk">RUSTSEC-2023-0071</param>
   </service>
 </services>

gst-plugins-rs-1.26.5.obscpio

@@ -0,0 +1,3 @@
+version https://git-lfs.github.com/spec/v1
+oid sha256:f7f9f29bb03c201e91f7cae7cdcc42c34f9d1e1f9579be97f10d224e167769fa
+size 14103565

gst-plugins-rs.obsinfo

@@ -1,4 +1,4 @@
 name: gst-plugins-rs
-version: 1.26.2
-mtime: 1748422381
-commit: 0826007d970a473475b6bf993229ebcde173fdba
+version: 1.26.5
+mtime: 1754571656
+commit: 4114bcad01f99c372431fac07dcdfec731a8abec

gstreamer-plugins-rs.changes

@@ -1,3 +1,127 @@
+-------------------------------------------------------------------
+Tue Aug 12 07:14:39 UTC 2025 - Bjørn Lie <bjorn.lie@gmail.com>
+
+- Update to version 1.26.5:
+  + awstranscriber2, awstranslate: Handle multiple stream-start
+    event
+  + ceaX08overlay: support ANY caps features, allowing e.g.
+    memory:GLMemory if downstream supports the overlay composition
+    meta
+  + hlsmultivariantsink: Fix master playlist version
+  + rtprecv: Drop state lock before chaining RTCP packets from the
+    RTP chain function
+  + Add rtpbin2 examples
+  + rtpmp4apay2: fix payload size prefix
+  + rtp: threadshare: fix some property ranges
+  + mpegtslivesrc: Remove leftover debug message
+  + ts-audiotestsrc fixes
+  + threadshare: fix flush for ts-queue ts-proxy & ts-intersrc
+  + threadshare: fix regression in ts-proxysrc
+  + threadshare: improvements to some elements
+  + threadshare: Enable windows Win32_Networking feature
+  + threadshare: queue & proxy: fix race condition stopping
+  + threadshare: Also enable windows Win32_Networking_WinSock
+    feature
+  + tracers: pipeline-snapshot: reduce WebSocket connection log
+    level
+  + tracers: queue-levels: add support for threadshare DataQueue
+    related elements
+  + tracers: Update to etherparse 0.19
+  + transcriberbin: Fix handling of upstream latency query
+  + webrtcsink: Move videorate before videoconvert and videoscale
+    to avoid processing frames that would be dropped
+  + Fix various new clippy 1.89 warnings
+
+-------------------------------------------------------------------
+Thu Jul 24 07:32:32 UTC 2025 - Bjørn Lie <bjorn.lie@gmail.com>
+
+- Update to version 1.26.4:
+  + aws: s3hlssink: Write to S3 on OutputStream flush
+  + cea708mux: fix clipping function
+  + dav1ddec: Use video decoder base class latency reporting API
+  + elevenlabssynthesizer: fix running time checks
+  + gopbuffer: Push GOPs in order of time on EOS
+  + gtk4: Improve color-state fallbacks for unknown values
+  + gtk4: Add YCbCr memory texture formats
+  + gtk4: Promote set_caps debug log to info
+  + hlssink3: Fix a comment typo
+  + hlssink3: Use closed fragment location in playlist generation
+  + livekit: add room-timeout
+  + mccparse: Convert "U" to the correct byte representation
+  + mp4mux: add TAI timestamp element and muxing
+  + threadshare: add a ts-rtpdtmfsrc element
+  + rtp: Update to rtcp-types 0.2
+  + rtpsend: Don't configure a zero min RTCP interval for senders
+  + rtpbin2: Fix handling of unknown PTs and don't warn about
+    incomplete RTP caps to allow for bundling
+  + rtpbin2: Improve rtcp-mux support
+  + rtpbin2: fix race condition on serialized Queries
+  + rtpbin2: sync: fix race condition
+  + rtprecv optimize src pad scheduling
+  + rtprecv: fix SSRC collision event sent in wrong direction
+  + skia: Add harfbuzz, freetype and fontconfig as dependencies in
+    the meson build
+  + tttocea{6,7}08: Disallow pango markup from input caps
+  + ts-intersrc: handle dynamic inter-ctx changes
+  + threadshare: src elements: don't pause the task in downward
+    state transitions
+  + webrtc: sink: avoid recursive locking of the session
+  + webrtcsink: fix deadlock on error setting remote description
+  + webrtcsink: add mitigation modes parameter and signal
+  + webrtc: fix Safari addIceCandidate crash
+  + webrtc-api: Set default bundle policy to max-bundle
+  + WHIP client: emit shutdown after DELETE request
+  + Fix various new clippy 1.88 warnings
+  + Update dependencies
+
+-------------------------------------------------------------------
+Tue Jul  1 20:32:32 UTC 2025 - Bjørn Lie <bjorn.lie@gmail.com>
+
+- Update to version 1.26.3:
+  + Add new speech synthesis element around ElevenLabs API
+  + cea708mux: fix another WouldOverflow case
+  + cea708mux: support configuring a limit to how much data will be
+    pending.
+  + cea708overlay: also reset the output size on flush stop
+  + gcc: handle out of order packets
+  + fmp4mux: Fix panic on late GOP
+  + livekit: expose a connection state property
+  + mp4mux: add taic box
+  + mp4mux: test the trak structure
+  + pcap_writer: Make target-property and pad-path properties
+    writable again
+  + skia: Don't build skia plugin by default for now
+  + threadshare: cleanups & usability improvements
+  + threadshare: sync runtime with latest async-io
+  + threadshare: fix kqueue reactor
+  + threadshare: Update to getifaddrs 0.2
+  + threadshare: add new thread-sharing inter elements
+  + threadshare: add a ts-rtpdtmfsrc element
+  + transcriberbin: fix naming of subtitle pads
+  + tttocea708: don't panic if a new service would overflow
+  + webrtc: android: Update Gradle and migrate to
+    FindGStreamerMobile
+  + webrtc: add new examples for stream selection over data channel
+  + webrtcsrc: the webrtcbin get-transceiver index is not
+    mlineindex
+  + webrtcsrc: send CustomUpstream events over control channel ..
+  + webrtcsink: Don't require encoder element for pre-encoded
+    streams
+  + webrtcsink: Don't reject caps events if the codec_data changes
+  + whip: server: pick session-id from the endpoint if specified
+  + cargo: add config file to force
+    CARGO_NET_GIT_FETCH_WITH_CLI=true
+  + Cargo.lock, deny: Update dependencies and log duplicated
+    targo-lexicon
+  + Update windows-sys dependency from ">=0.52, <=0.59" to ">=0.52,
+    <=0.60"
+  + deny: Add override for windows-sys 0.59
+  + deny: Update lints
+  + cargo_wrapper: Fix backslashes being parsed as escape codes on
+    Windows
+  + Fixes for Clock: non-optional return types
+  + Rename relationmeta plugin to analytics
+
 -------------------------------------------------------------------
 Sun Jun  1 07:36:19 UTC 2025 - Bjørn Lie <bjorn.lie@gmail.com>
 

gstreamer-plugins-rs.spec

@@ -1,7 +1,7 @@
 #
 # spec file for package gstreamer-plugins-rs
 #
-# Copyright (c) 2025 SUSE LLC
+# Copyright (c) 2025 SUSE LLC and contributors
 #
 # All modifications and additions to the file contributed by third parties
 # remain the property of their copyright owners, unless otherwise agreed
@@ -27,7 +27,7 @@
 %dnl #%%endif
 
 Name:           gstreamer-plugins-rs
-Version:        1.26.2
+Version:        1.26.5
 Release:        0
 Summary:        GStreamer Streaming-Media Framework Plug-Ins
 License:        LGPL-2.1-or-later
@@ -125,6 +125,7 @@ cp %{SOURCE4} %{buildroot}%{_datadir}/appdata/
 # Disable csound for now, bring issue upstream
 #%%{_libdir}/gstreamer-%%{gst_branch}/libgstcsound.so
 %{_libdir}/gstreamer-%{gst_branch}/libgstdav1d.so
+%{_libdir}/gstreamer-%{gst_branch}/libgstelevenlabs.so
 %{_libdir}/gstreamer-%{gst_branch}/libgstfallbackswitch.so
 %{_libdir}/gstreamer-%{gst_branch}/libgstffv1.so
 %{_libdir}/gstreamer-%{gst_branch}/libgstfmp4.so
@@ -145,6 +146,7 @@ cp %{SOURCE4} %{buildroot}%{_datadir}/appdata/
 %{_libdir}/gstreamer-%{gst_branch}/libgstrav1e.so
 %{_libdir}/gstreamer-%{gst_branch}/libgstregex.so
 %{_libdir}/gstreamer-%{gst_branch}/libgstreqwest.so
+%{_libdir}/gstreamer-%{gst_branch}/libgstrsanalytics.so
 %{_libdir}/gstreamer-%{gst_branch}/libgstrsaudiofx.so
 %{_libdir}/gstreamer-%{gst_branch}/libgstrsclosedcaption.so
 %{_libdir}/gstreamer-%{gst_branch}/libgstrsfile.so
@@ -167,7 +169,6 @@ cp %{SOURCE4} %{buildroot}%{_datadir}/appdata/
 %{_libdir}/gstreamer-%{gst_branch}/libgsturiplaylistbin.so
 %{_libdir}/gstreamer-%{gst_branch}/libgstwebrtchttp.so
 %{_libdir}/gstreamer-%{gst_branch}/libgsthlsmultivariantsink.so
-%{_libdir}/gstreamer-%{gst_branch}/libgstrsrelationmeta.so
 %{_libdir}/gstreamer-%{gst_branch}/libgstspeechmatics.so
 %{_libdir}/gstreamer-%{gst_branch}/libgststreamgrouper.so
 %dir %{_datadir}/appdata