===> The following configuration options are available for baresip-4.12.0:
AAC=on: AAC audio codec
ALSA=on: ALSA audio architecture support
AMR=off: AMR-NB and AMR-WB audio codecs
AV1=on: AV1 Video Codec
AVCODEC=on: Video codecs using libavcodec
AVFILTER=on: Video filter using libavfilter
AVFORMAT=on: libavformat media-source
CODEC2=on: Codec2 low-bitrate speech codec
DBUS=on: D-Bus IPC system support
DOCS=on: Build and/or install documentation
EVDEV=on: evdev input support
EXAMPLES=on: Build and/or install examples
GSTREAMER=on: Multimedia support via GStreamer
GTK3=on: GTK 3 GUI toolkit support
JACK=off: JACK audio server support
MQTT=off: Message Queue Telemetry Transport (MQTT) client
OPUS=on: Opus audio codec support
PIPEWIRE=off: PipeWire multimedia server support
PLC=on: Packet Loss Concealment
PORTAUDIO=off: PortAudio library support
PULSEAUDIO=off: PulseAudio sound server support
SDL=on: Simple Direct Media Layer support
SNAPSHOT=on: Snapshot video module
SNDFILE=on: Audio conversion support via libsndfile
SNDIO=off: Sndio audio support
SWSCALE=on: Video filter for scaling and pixel conversion
V4L=on: Video 4 Linux support
VPX=on: VP8/VP9 video codec support
WEBRTC_AEC=on: WebRTC Acoustic Echo Cancellation
X11=on: X11 (graphics) support
====> Options available for the group EXTRAMODULES
AULOOP=off: Application module for testing audio drivers
AUTOTEST=off: Autotest module
EBUACIP=off: EBU Audio Contribution over IP Profile
INTERCOM=off: Intercom module
KAOPTIONS=off: Keepalive via SIP OPTIONS module
PARCALL=off: Parallel call module
QUALIFY=off: Qualify peer module
REDIRECT=off: Redirect incoming calls module
VIDLOOP=off: Video-loop module for testing
====> G.722 audio codec: you can only select none or one of them
G722=on: G.722 audio codec
G722L=off: G.722 audio codec via libg722
===> Use 'make config' to modify these settings
audio/baresip: Fix typos in option helpers
I did two typos in commit 724c2486e632fc451c0ca70a372e9fd409fe7f54, fix
them.
PR: 286347
Approved by: Herbert J. Skuhra <herbert@gojira.at>
audio/baresip: Modernize port
Use option helpers instead of many if-statements.
The MODULES_LIST is handled by OPT_VARS option helpers. The make's
variable modifier sets the semicolon separator in CMAKE_ARGS.
PR: 286024
Approved by: Herbert J. Skuhra <herbert@gojira.at> (maintainer)
audio/baresip: remove audio/rem dependency
audio/rem is now merged into audio/re [1]
[1] https://github.com/baresip/re/pull/683
Reported by: Herbert J. Skuhra <herbert@gojira.at> (by email)
Remove WWW entries moved into port Makefiles
Commit b7f05445c00f has added WWW entries to port Makefiles based on
WWW: lines in pkg-descr files.
This commit removes the WWW: lines of moved-over URLs from these
pkg-descr files.
Approved by: portmgr (tcberner)
Add WWW entries to port Makefiles
It has been common practice to have one or more URLs at the end of the
ports' pkg-descr files, one per line and prefixed with "WWW:". These
URLs should point at a project website or other relevant resources.
Access to these URLs required processing of the pkg-descr files, and
they have often become stale over time. If more than one such URL was
present in a pkg-descr file, only the first one was tarnsfered into
the port INDEX, but for many ports only the last line did contain the
port specific URL to further information.
There have been several proposals to make a project URL available as
a macro in the ports' Makefiles, over time.
(Only the first 15 lines of the commit message are shown above )
Add an option to enable PulseAudio
The option is disabled by default, so the package is not modified, no
PORTREVISION bump needed.
PR: 234225
Approved by: crees (maintainer)
Mark net/libsrtp deprecated, superseded by libsrtp2.
audio/baresip: remove SRTP option (had been marked BROKEN).
www/qt5-webengine: remove unused dependency on libsrtp.
When there is a do-install target, do not use a post-install target, do
everything at once. Sometime, rename post-install into a options helper
target.
I did not fix ports that were such a mess that I could not figure out
what they really wanted to do. I also did not change ports that had
some version of an auto-plist code in post-install, for the same reason.
With hat: portmgr
Sponsored by: Absolight
Update portaudio to v19/Remove portaudio2 [1]
Chase portaudio change
Add patches from debian for games/cultivation
Add patches from upsteam for audio/rezound
Mark py-fastaudio as broken
Approved by: maintainer
Replace USES=libtool:oldver with USES=libtool or USES=libtool:keepla in
the 32 ports that still use it. Bump PORTREVISION on their dependent
ports except the ones that depend on these:
audio/libogg
audio/libvorbis
devel/pcre
ftp/curl
graphics/jpeg
graphics/libart_lgpl
graphics/tiff
textproc/expat2
textproc/libxslt
In these cases the same trick as in the recent gettext update is used.
(Only the first 15 lines of the commit message are shown above )
Convert all :U to :tu and :L to :tl
Since FreeBSD 8.4 and FreeBSD 9.1 make(1) do support :tu and :tl as a
replacement for :U and :L (which has been marked as deprecated)
bmake which is the default on FreeBSD 10+ only support by default
:tu/:tl a hack has been added at the time to support :U and :L to ease
migration. This hack is now not necessary anymore
Note that this makes the ports tree incompatible with make(1) from
FreeBSD 8.3 or earlier
With hat: portmgr
Update Baresip to 0.4.8, and the supporting re and rem libraries
to their latest versions
Several new modules, however some will no longer work after changes to the
core code; a best effort has been made to investigate and mark such broken
modules.
Convert audio from USE_GMAKE to USES=gmake
While here:
- Trim headers
- Convert some USE_GNOME=pkgconfig to USES=pkgconfig
- Add some missing pkgconf dependencies
- Convert some USE_GNOME=gnomehack to USES=pathfix
Various spelling corrections
PR: ports/175331
Submitted by: Christoph Mallon
Approved by: No objections within three weeks from any maintainer
While here, style and duplicate phrase fixes in bsdcflow pkg-descr
Submitted by: mi
Introducing baresip 0.4.2, along with the same versions of libre and librem.
re and rem must be updated first, since baresip will not compile against older
versions.
[
Fix linking of Speex modules
PR: ports/174925 (based on)
Submitted by: Anish Mistry <amistry@unitedware.com>
]
baresip is a small, modular telephony client that works
on multiple platforms and supports several audio and video
input/output methods.
WWW: http://www.creytiv.com/baresip.html
Submitted by: luigi (original port)
