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-SP6:GA
-
Checkout Package
osc checkout SUSE:SLE-15-SP6:Update/webrtc-audio-processing && cd $_ - Create Badge
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Source Files
| Filename | Size | Changed |
|---|---|---|
| _service | 0000000658 658 Bytes | |
| baselibs.conf | 0000000058 58 Bytes | |
| big_endian_support.patch | 0000003470 3.39 KB | |
| big_endian_support_2.patch | 0000000905 905 Bytes | |
| fix-build.patch | 0000002664 2.6 KB | |
| fix-i586.patch | 0000006335 6.19 KB | |
| reduce-meson-dep.patch | 0000000483 483 Bytes | |
| webrtc-audio-processing-1.3.obscpio | 0004396556 4.19 MB | |
| webrtc-audio-processing.changes | 0000006969 6.81 KB | |
| webrtc-audio-processing.obsinfo | 0000000110 110 Bytes | |
| webrtc-audio-processing.spec | 0000007248 7.08 KB | |
| webrtc-ppc64.patch | 0000000883 883 Bytes | |
| webrtc-s390x.patch | 0000000560 560 Bytes |
Latest Revision
Samu Voutilainen (Smar)
committed
(revision 1)
osc copypac from project:openSUSE.org:SUSE:SLE-15-SP6:GA package:webrtc-audio-processing revision:2
Comments 0