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| author | Raghav Gururajan <raghavgururajan@disroot.org> | 2020-04-01 23:04:34 -0400 |
|---|---|---|
| committer | Danny Milosavljevic <dannym@scratchpost.org> | 2020-04-06 01:05:43 +0200 |
| commit | fe8f1c0b194de2fa7aa5265a4a0424e82bd11af1 (patch) | |
| tree | 5372186fbeddd5441bef2059102fe23342799613 | |
| parent | f689d3d7d9ba3fa25b4839bdd1e4b98905f4869c (diff) | |
| download | guix-fe8f1c0b194de2fa7aa5265a4a0424e82bd11af1.tar.gz | |
gnu: Add mswebrtc.
* gnu/packages/linphone.scm (mswebrtc): New variable. Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
| -rw-r--r-- | gnu/packages/linphone.scm | 28 |
1 files changed, 28 insertions, 0 deletions
diff --git a/gnu/packages/linphone.scm b/gnu/packages/linphone.scm index 75c3476675..68a4e3e140 100644 --- a/gnu/packages/linphone.scm +++ b/gnu/packages/linphone.scm @@ -504,3 +504,31 @@ mediastreamer2 based on the openh264 library.") AMR codec. It is based on the Skype's SILK implementation.") (home-page "https://gitlab.linphone.org/BC/public/mssilk") (license license:gpl2+))) + +(define-public mswebrtc + (package + (name "mswebrtc") + (version "1.1.1") + (source + (origin + (method url-fetch) + (uri + (string-append "https://www.linphone.org/releases/sources/plugins/" + name "/" name "-" version ".tar.gz")) + (sha256 + (base32 "1wj28hl9myhshqmn64xg0jf07aw75gmnilb5rff6rcbdxim87mqr")))) + (build-system cmake-build-system) + (arguments + `(#:tests? #f ; No test target + #:configure-flags + (list + "-DENABLE_STATIC=NO"))) + (inputs + `(("bctoolbox" ,bctoolbox) + ("mediastreamer2" ,mediastreamer2) + ("ortp" ,ortp))) + (synopsis "Media Streamer WebRTC Codec") + (description "MSWebRTC is a plugin of MediaStreamer, adding support for +WebRTC codec. It includes features from WebRTC, such as, iSAC and AECM.") + (home-page "https://gitlab.linphone.org/BC/public/mswebrtc") + (license license:gpl2+))) |