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
-
Checkout Package
osc checkout SUSE:SLE-15-SP3:Update/webrtc-audio-processing && 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
osc copypac from project:openSUSE.org:SUSE:SLE-15:Update package:webrtc-audio-processing revision:14872084ee0c2fc5c49eb79ded759a24
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