Real-Time Communication Library for Web Browsers
http://www.freedesktop.org/software/pulseaudio/webrtc-audio-processing/
WebRTC is an open source project that enables web browsers with Real-Time
Communications (RTC) capabilities via simple Javascript APIs. The WebRTC
components have been optimized to best serve this purpose.
WebRTC implements the W3C's proposal for video conferencing on the web.
- Developed at multimedia:libs
- Sources inherited from project openSUSE:Factory
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5
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osc -A https://api.opensuse.org checkout home:seife:Factory/webrtc-audio-processing && cd $_
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Source Files
Filename | Size | Changed |
---|---|---|
baselibs.conf | 0000000028 28 Bytes | |
big_endian_support.patch | 0000003613 3.53 KB | |
big_endian_support_2.patch | 0000000905 905 Bytes | |
webrtc-audio-processing-0.3.tar.xz | 0000688096 672 KB | |
webrtc-audio-processing.changes | 0000004261 4.16 KB | |
webrtc-audio-processing.spec | 0000004865 4.75 KB | |
webrtc-ppc64.patch | 0000000539 539 Bytes | |
webrtc-s390x.patch | 0000000434 434 Bytes |
Revision 10 (latest revision is 15)
Dominique Leuenberger (dimstar_suse)
accepted
request 451393
from
Olaf Hering (olh)
(revision 10)
- Add baselibs.conf for gstreamer-plugins-bad-32bit (forwarded request 451390 from olh)
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