git: d21befbec688 - main - net-im/dino: Improve audio/video calling support
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Date: Fri, 04 Mar 2022 17:49:33 UTC
The branch main has been updated by ashish:
URL: https://cgit.FreeBSD.org/ports/commit/?id=d21befbec6881100a3a3258acea05d70019bdd3e
commit d21befbec6881100a3a3258acea05d70019bdd3e
Author: Ashish SHUKLA <ashish@FreeBSD.org>
AuthorDate: 2022-03-04 17:44:11 +0000
Commit: Ashish SHUKLA <ashish@FreeBSD.org>
CommitDate: 2022-03-04 17:49:13 +0000
net-im/dino: Improve audio/video calling support
Add couple more dependencies which were missing, but seem to be
necessary for audio/video calls. Many thanks to Goran for helping
test this.
PR: 262055
Reported by: Goran Mekić <meka at tilda dot center>
Tested by: Goran Mekić <meka at tilda dot center>
---
net-im/dino/Makefile | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/net-im/dino/Makefile b/net-im/dino/Makefile
index 5258c009d9de..5b5dc3955b44 100644
--- a/net-im/dino/Makefile
+++ b/net-im/dino/Makefile
@@ -1,7 +1,7 @@
PORTNAME= dino
DISTVERSIONPREFIX= v
DISTVERSION= 0.3.0
-PORTREVISION= 1
+PORTREVISION= 2
CATEGORIES= net-im
MAINTAINER= ashish@FreeBSD.org
@@ -47,7 +47,7 @@ ICE_VARS= ENABLED_PLUGINS+=ice
ICE_VARS_OFF= DISABLED_PLUGINS+=ice
RTP_LIB_DEPENDS= libgnutls.so:security/gnutls \
libwebrtc-audio-processing-1.so:audio/webrtc-audio-processing
-RTP_USE= GSTREAMER1=srtp
+RTP_USE= GSTREAMER1=srtp,gtk,pulse,opus,speex,vpx,x264,v4l2
RTP_VARS= ENABLED_PLUGINS+=rtp
RTP_VARS_OFF= DISABLED_PLUGINS+=rtp
OMEMO_GH_TUPLE= signalapp:libsignal-protocol-c:v2.3.3:signalapp/plugins/signal-protocol/libsignal-protocol-c