commit: d1aa3f915ebd1b16d2c8f134d094557194411420
Author: Igor V. Kovalenko <igor.v.kovalenko <AT> gmail <DOT> com>
AuthorDate: Fri Aug 26 13:00:41 2022 +0000
Commit: Sam James <sam <AT> gentoo <DOT> org>
CommitDate: Fri Aug 26 13:10:06 2022 +0000
URL: https://gitweb.gentoo.org/repo/gentoo.git/commit/?id=d1aa3f91
media-plugins/gst-plugins-webrtc: add 1.20.3
Signed-off-by: Sam James <sam <AT> gentoo.org>
Signed-off-by: Igor V. Kovalenko <igor.v.kovalenko <AT> gmail.com>
media-plugins/gst-plugins-webrtc/Manifest | 1 +
.../gst-plugins-webrtc-1.20.3.ebuild | 42 ++++++++++++++++++++++
2 files changed, 43 insertions(+)
diff --git a/media-plugins/gst-plugins-webrtc/Manifest
b/media-plugins/gst-plugins-webrtc/Manifest
index 8afe0c32cc8d..fdff776ab51a 100644
--- a/media-plugins/gst-plugins-webrtc/Manifest
+++ b/media-plugins/gst-plugins-webrtc/Manifest
@@ -1 +1,2 @@
DIST gst-plugins-bad-1.20.2.tar.xz 6216208 BLAKE2B
bbbe77a1255991a2f96696996fb0c99f14f6d63fef455feb1ce90ae518bb9f80fd61bcfb223c20407b8d6240faaa93478495f8e9fda16fab36a311d167e88e25
SHA512
3f98973dc07ead745418e0a30f9f6b5c8d328e3d126f54d92c10ab5da04271768a5c5dffc36ea24ccf8fb516b1e3733be9fb18dc0db419dea4d37d17018f8a70
+DIST gst-plugins-bad-1.20.3.tar.xz 6222824 BLAKE2B
01aae59adbe76b8e50a49fb8bb8037e6f3aa93cbc2b658aab05ebbf30f8d1aef98c1981712caa39e3c9d08f1e0c9d76f2f874f7d2fdd994b3a0735b2809eafdf
SHA512
cfcf126eabff550455decd7054a269b73489708c10a6b6090dddb5fde29bfba07ed330c339927ff170e025fa3a08d2ffb822322dc3798679366207a54132c71b
diff --git a/media-plugins/gst-plugins-webrtc/gst-plugins-webrtc-1.20.3.ebuild
b/media-plugins/gst-plugins-webrtc/gst-plugins-webrtc-1.20.3.ebuild
new file mode 100644
index 000000000000..c3f6d58eef3a
--- /dev/null
+++ b/media-plugins/gst-plugins-webrtc/gst-plugins-webrtc-1.20.3.ebuild
@@ -0,0 +1,42 @@
+# Copyright 1999-2022 Gentoo Authors
+# Distributed under the terms of the GNU General Public License v2
+
+EAPI=7
+GST_ORG_MODULE=gst-plugins-bad
+
+inherit gstreamer-meson
+
+DESCRIPTION="WebRTC plugins for GStreamer"
+KEYWORDS="~amd64"
+
+# == ext/webrtc/meson.build
+# dev-libs/glib (eclass): gio_dep
+# net-libs/libnice: libnice_dep
+# media-libs/gst-plugins-base: gstbase_dep, gstsdp_dep, gstapp_dep, gstrtp_dep
+# media-plugins/gst-plugins-sctp: gstsctp_dep
+# == ext/webrtcdsp/meson.build
+# media-libs/gst-plugins-base: gstbase_dep, gstaudio_dep
+# media-libs/gst-plugins-bad: gstbadaudio_dep
+# media-libs/webrtc-audio-processing: webrtc_dep
+# (android): gnustl_dep
+# == gst-libs/gst/webrtc/meson.build
+# media-libs/gst-plugins-base: gstbase_dep, gstsdp_dep
+RDEPEND="
+ >=media-libs/gst-plugins-base-${PV}:1.0[${MULTILIB_USEDEP}]
+ >=media-libs/gst-plugins-bad-${PV}:1.0[${MULTILIB_USEDEP}]
+ >=media-plugins/gst-plugins-sctp-${PV}:1.0[${MULTILIB_USEDEP}]
+ >=media-libs/webrtc-audio-processing-0.2:0[${MULTILIB_USEDEP}]
+ <media-libs/webrtc-audio-processing-0.4:0
+ >=net-libs/libnice-0.1.17[${MULTILIB_USEDEP}]
+"
+DEPEND="${RDEPEND}"
+
+GST_PLUGINS_ENABLED="webrtc webrtcdsp"
+GST_PLUGINS_BUILD_DIR="webrtc webrtcdsp"
+
+src_prepare() {
+ default
+ gstreamer_system_package gstwebrtc_dep:gstreamer-webrtc
+ gstreamer_system_package gstsctp_dep:gstreamer-sctp
+ gstreamer_system_package gstbadaudio_dep:gstreamer-bad-audio
+}