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