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@@ -3,19 +3,22 @@
<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">0.13.5</param>
<param name="revision">refs/tags/0.13.5</param>
<param name="versionformat">0.11.3</param>
<param name="revision">refs/tags/0.11.3+fixup</param>
<param name="scm">git</param>
</service>
<service name="set_version" mode="manual"/>
<service name="tar" mode="buildtime"/>
<service name="recompress" mode="buildtime">
<param name="file">*.tar</param>
<param name="compression">zst</param>
<param name="file">*.tar</param>
<param name="compression">xz</param>
</service>
<service name="cargo_vendor" mode="manual">
<param name="srcdir">gst-plugins-rs</param>
<param name="compression">zst</param>
<param name="update">true</param>
<param name="srcdir">gst-plugins-rs</param>
<param name="update">true</param>
<!-- <param name="i-accept-the-risk">RUSTSEC-2023-0065</param>-->
</service>
<service name="cargo_audit" mode="manual">
<param name="srcdir">gst-plugins-rs</param>
</service>
</services>

29
cargo_config Normal file
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@@ -0,0 +1,29 @@
[source.crates-io]
replace-with = "vendored-sources"
[source."git+https://github.com/gtk-rs/gtk-rs-core?branch=0.18"]
git = "https://github.com/gtk-rs/gtk-rs-core"
branch = "0.18"
replace-with = "vendored-sources"
[source."git+https://github.com/gtk-rs/gtk4-rs?branch=0.7"]
git = "https://github.com/gtk-rs/gtk4-rs"
branch = "0.7"
replace-with = "vendored-sources"
[source."git+https://github.com/rust-av/ffv1.git?rev=2afb025a327173ce891954c052e804d0f880368a"]
git = "https://github.com/rust-av/ffv1.git"
rev = "2afb025a327173ce891954c052e804d0f880368a"
replace-with = "vendored-sources"
[source."git+https://github.com/rust-av/flavors"]
git = "https://github.com/rust-av/flavors"
replace-with = "vendored-sources"
[source."git+https://gitlab.freedesktop.org/gstreamer/gstreamer-rs?branch=0.21"]
git = "https://gitlab.freedesktop.org/gstreamer/gstreamer-rs"
branch = "0.21"
replace-with = "vendored-sources"
[source.vendored-sources]
directory = "vendor"

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gst-plugins-rs-0.11.3.obscpio (Stored with Git LFS) Normal file

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gst-plugins-rs-0.13.5.obscpio (Stored with Git LFS)

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@@ -1,4 +1,4 @@
name: gst-plugins-rs
version: 0.13.5
mtime: 1741090040
commit: 49e4a1438dd8ff1c3c7be07e48c95a02b20a0707
version: 0.11.3
mtime: 1704366323
commit: 5bba2f783632bb27687662017d703af2fb2fd4f1

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@@ -1,392 +1,3 @@
-------------------------------------------------------------------
Wed Mar 12 14:06:37 UTC 2025 - Antonio Larrosa <alarrosa@suse.com>
- 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.
-------------------------------------------------------------------
Thu Jan 2 21:55:22 UTC 2025 - Bjørn Lie <bjorn.lie@gmail.com>
- 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.
-------------------------------------------------------------------
Mon Dec 9 11:12:56 UTC 2024 - Bjørn Lie <bjorn.lie@gmail.com>
- 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.
-------------------------------------------------------------------
Wed Oct 16 18:58:43 UTC 2024 - Bjørn Lie <bjorn.lie@gmail.com>
- 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.
-------------------------------------------------------------------
Tue Sep 10 12:56:17 UTC 2024 - Bjørn Lie <bjorn.lie@gmail.com>
- 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.
-------------------------------------------------------------------
Mon Jul 22 11:44:43 UTC 2024 - Antonio Larrosa <alarrosa@suse.com>
- 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.
-------------------------------------------------------------------
Thu Feb 29 11:52:01 UTC 2024 - Antonio Larrosa <alarrosa@suse.com>
- 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.
-------------------------------------------------------------------
Mon Jan 29 10:22:55 UTC 2024 - Antonio Larrosa <alarrosa@suse.com>

View File

@@ -1,7 +1,7 @@
#
# spec file for package gstreamer-plugins-rs
#
# Copyright (c) 2025 SUSE LLC
# Copyright (c) 2024 SUSE LLC
#
# All modifications and additions to the file contributed by third parties
# remain the property of their copyright owners, unless otherwise agreed
@@ -19,6 +19,8 @@
%global _lto_cflags %{_lto_cflags} -ffat-lto-objects
%define _name gst-plugins-rs
%define gst_branch 1.0
# Disable csound for now, bring issue upstream
#%%global __requires_exclude pkgconfig\\(csound\\)
%ifarch s390 s390x ppc ppc64
%bcond_with aws
@@ -26,30 +28,38 @@
%bcond_without aws
%endif
%if %{?pkg_vcmp:%{pkg_vcmp dav1d-devel >= 1.3.0}}%{!?pkg_vcmp:0}
%define has_dav1d_1_3_0 1
%endif
Name: gstreamer-plugins-rs
Version: 0.13.5
Version: 0.11.3
Release: 0
Summary: GStreamer Streaming-Media Framework Plug-Ins
License: LGPL-2.1-or-later
Group: Productivity/Multimedia/Other
URL: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs
Source: %{_name}-%{version}.tar.zst
Source: %{_name}-%{version}.tar.xz
Source2: vendor.tar.zst
Source3: cargo_config
Source4: gstreamer-plugins-rs.appdata.xml
Source5: vendor-for-dav1d-1.3.0.tar.zst
BuildRequires: cargo-c >= 0.9.21
BuildRequires: cargo-packaging >= 1.2.0+3
BuildRequires: clang
BuildRequires: git
# Disable csound for now, bring issue upstream
#BuildRequires: csound-devel
BuildRequires: llvm
BuildRequires: git
BuildRequires: meson >= 0.60
BuildRequires: nasm
BuildRequires: pkgconfig
BuildRequires: python3-tomli
BuildRequires: zstd
BuildRequires: pkgconfig(cairo) >= 1.10.0
BuildRequires: pkgconfig(dav1d) >= 1.3.0
BuildRequires: pkgconfig(dav1d)
BuildRequires: pkgconfig(gstreamer-1.0)
BuildRequires: pkgconfig(gstreamer-base-1.0)
BuildRequires: pkgconfig(gstreamer-plugins-base-1.0)
@@ -62,7 +72,7 @@ BuildRequires: pkgconfig(pango)
Requires: gstreamer
Requires: gstreamer-plugins-base
Enhances: gstreamer
ExclusiveArch: %{rust_tier1_arches}
ExcludeArch: ppc ppc64 ppc64le s390 %ix86 %arm
%description
GStreamer is a streaming media framework based on graphs of filters
@@ -78,6 +88,7 @@ This package provides various plugins written in Rust.
Summary: GStreamer Streaming-Media Framework Plug-Ins development files
Group: Development/Libraries/Other
Requires: %{name} = %{version}
#Requires: csound-devel
%description devel
GStreamer is a streaming media framework based on graphs of filters
@@ -91,9 +102,31 @@ This package contains the pkgconfig development files for the rust
plugins.
%prep
%if 0%{?has_dav1d_1_3_0}
%autosetup -n %{_name}-%{version} -a5 -p1
sed -ie 's/^dav1d = "[0-9\.]*"/dav1d = "0.10"/' video/dav1d/Cargo.toml
%else
%autosetup -n %{_name}-%{version} -a2 -p1
%endif
%if %{?suse_version} < 1600
sed -ie "s/meson_version : '>= 1.1'/meson_version : '>= 0.61.4'/" meson.build
sed -ie "s/\.enable_if.*//" meson.build
sed -ie "s/find_program('cargo-cbuild', version:'>=0.9.21'/find_program('cargo-cbuild', version:'>=0.9.15'/" meson.build
%endif
mkdir -p .cargo
cp %{SOURCE3} .cargo/config
sed -i -e 's/version = "8"/version = "9"/' vendor/livekit-api/Cargo.toml
sed -i -e "s/ab6a42a4752e822c794421fa4b939e7e9690e85541c5e0ae28a34f17fe8fd170/2c69748813bcb4e4f3d06343e05fb9f43a8ae623fbdbb340847fd536f1974aa9/" vendor/livekit-api/.cargo-checksum.json
%build
# Disable csound for now, bring issue upstream
#export CSOUND_LIB_DIR=%%{_libdir}
export RUSTFLAGS="%{build_rustflags}"
%meson \
--default-library=shared \
-Ddoc=disabled \
@@ -107,6 +140,7 @@ plugins.
%meson_build
%install
export RUSTFLAGS="%{build_rustflags}"
%meson_install
mkdir -p %{buildroot}%{_datadir}/appdata
cp %{SOURCE4} %{buildroot}%{_datadir}/appdata/
@@ -127,7 +161,6 @@ cp %{SOURCE4} %{buildroot}%{_datadir}/appdata/
%{_libdir}/gstreamer-%{gst_branch}/libgstffv1.so
%{_libdir}/gstreamer-%{gst_branch}/libgstfmp4.so
%{_libdir}/gstreamer-%{gst_branch}/libgstgif.so
%{_libdir}/gstreamer-%{gst_branch}/libgstgopbuffer.so
%{_libdir}/gstreamer-%{gst_branch}/libgstgtk4.so
%{_libdir}/gstreamer-%{gst_branch}/libgsthlssink3.so
%{_libdir}/gstreamer-%{gst_branch}/libgsthsv.so
@@ -135,10 +168,7 @@ cp %{SOURCE4} %{buildroot}%{_datadir}/appdata/
%{_libdir}/gstreamer-%{gst_branch}/libgstlewton.so
%{_libdir}/gstreamer-%{gst_branch}/libgstlivesync.so
%{_libdir}/gstreamer-%{gst_branch}/libgstmp4.so
%{_libdir}/gstreamer-%{gst_branch}/libgstmpegtslive.so
%{_libdir}/gstreamer-%{gst_branch}/libgstndi.so
%{_libdir}/gstreamer-%{gst_branch}/libgstoriginalbuffer.so
%{_libdir}/gstreamer-%{gst_branch}/libgstquinn.so
%{_libdir}/gstreamer-%{gst_branch}/libgstraptorq.so
%{_libdir}/gstreamer-%{gst_branch}/libgstrav1e.so
%{_libdir}/gstreamer-%{gst_branch}/libgstregex.so
@@ -147,11 +177,9 @@ cp %{SOURCE4} %{buildroot}%{_datadir}/appdata/
%{_libdir}/gstreamer-%{gst_branch}/libgstrsclosedcaption.so
%{_libdir}/gstreamer-%{gst_branch}/libgstrsfile.so
%{_libdir}/gstreamer-%{gst_branch}/libgstrsflv.so
%{_libdir}/gstreamer-%{gst_branch}/libgstrsinter.so
%{_libdir}/gstreamer-%{gst_branch}/libgstrsonvif.so
%{_libdir}/gstreamer-%{gst_branch}/libgstrspng.so
%{_libdir}/gstreamer-%{gst_branch}/libgstrsrtp.so
%{_libdir}/gstreamer-%{gst_branch}/libgstrsrtsp.so
%{_libdir}/gstreamer-%{gst_branch}/libgstrstracers.so
%{_libdir}/gstreamer-%{gst_branch}/libgstrsvideofx.so
%{_libdir}/gstreamer-%{gst_branch}/libgstrswebp.so

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