libwebrtc-audio-processing-1.so:audio/webrtc-audio-processing
RTP_USE= GSTREAMER1=srtp,gtk,pulse,opus,speex,vpx,x264,v4l2
RTP_VARS= ENABLED_PLUGINS+=rtp
RTP_CMAKE_BOOL= RTP_ENABLE_H264 RTP_ENABLE_VP9 RTP_ENABLE_H264 RTP_ENABLE_VAAPI
RTP_VARS_OFF= DISABLED_PLUGINS+=rtp
OMEMO_GH_TUPLE= signalapp:libsignal-protocol-c:v2.3.3:signalapp/plugins/signal-protocol/libsignal-protocol-c
OMEMO_LIB_DEPENDS= libqrencode.so:graphics/libqrencode