Real-Time Communication Library for Web Browsers
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.
- Sources inherited from project SUSE:SLE-15:Update
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osc -A https://api.opensuse.org checkout openSUSE:Step:15-SP2/webrtc-audio-processing.30992 && cd $_ - Create Badge
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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 |
Latest Revision
Daniel Mach (dmach)
committed
(revision 1)
Release from SUSE:Maintenance:30992 / webrtc-audio-processing.SUSE_SLE-15_Update
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