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Opus - VideoLAN Wiki
https://wiki.videolan.org/Opus/
Opus is a general-purpose high-quality low-latency lossy audio codec created by the Xiph.org Foundation. The Opus developers claim it makes other lossy audio codecs obsolete (including Vorbis and MP3) and recommend it over Speex as a speech codec. Source code. modules/codec/opus.c
Category:Opus Audio Codec - Wikimedia Commons
https://commons.wikimedia.org/wiki/Category:Opus_Audio_Codec
Media in category "Opus Audio Codec" The following 29 files are in this category, out of 29 total. JMValin presenting Opus at LCA2012 edit.jpg 596 × 734; 99 KB
OpusFAQ - XiphWiki - wiki.xiph.org
https://wiki.xiph.org/OpusFAQ
Opus is a totally open, royalty-free, highly versatile audio codec. It is primarily designed for interactive speech and music transmission over the Internet, but is also applicable to storage and streaming applications. It incorporates technology from Skype's SILKcodec and Xiph.Org's CELTcodec.
Comparison of audio coding formats - Wikipedia
https://en.wikipedia.org/wiki/Comparison_of_audio_coding_formats
Opus Audio Codec — Video Conferencing Blog
https://trueconf.com/blog/wiki/opus-codec
Opus is a multipurpose audio codec that combines the balance of high-quality audio signal compression with low delay rates. It was developed in 2012 by the IETF working group. Its flexibility lies in adapting to changes in channel’s bandwidth capacity and support of any kind of audio encoding.
Opus Codec - Opus Interactive Audio Codec
Opus is a totally open, royalty-free, highly versatile audio codec. speech and music transmission over the Internet, but is also intended for storage and streaming It is standardized by the Internet Engineering Task Force (IETF) as RFC 6716which incorporated technology from Skype’s SILK codec and Xiph.Org’s CELT codec. Technology
Opus Interactive Audio Codec — Wikipédia
https://fr.wikipedia.org/wiki/Opus_Interactive_Audio_Codec
opus est un format ouvert de compression audio avec pertes, libre de redevances et normalisé par l' internet engineering task force (ietf), conçu pour encoder efficacement la voix et plus largement l'audio dans un format unique, tout en ayant une latence suffisamment faible pour la communication en temps réel et une complexité suffisamment faible …
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