Free video and audio codec developer, XIPH.org, has released Audio codec opus 1.4.0, which promises high-quality coding and minimal delay for both streaming sound and compressed voice in VOIP applications with a limited throughput. The team implemented the encoder and decoder under the BSD license, and the full specifications for the OPUS format are publicly available and approved as an internet standard (RFC 6716).
OPUS was developed by combining the top technologies from Xiph.org codec celt and Skype Silk. The development of OPUS was participated in by Xiph.org, Mozilla, Octasic, Broadadcom, and Google, among others which provide patents for unlimited use without paying licensing deductions. Any applications and products using OPUS do not require any additional approval for intellectual rights-related to OPUS and licenses to patents. There are no restrictions on creating third-party implementations. However, granted rights are responded to with actions should OPUS technology be used in patent proceedings against any user.
Previously, OPUS was rated as the best codec that uses the 64kbit Bitrate available, overtaking competitors such as Apple He- AAC, Nero He-Aac, Vorbis, and AAC LC. Firefox browser, Gstreamer framework, and the FFMPEG package are some of the products that support OPUS from the box.
OPUS has several key features such as support for permanent (CBR) and alternating (VBR) bitrates, and stereo and mono. It also supports voice and music and has the ability to restore the sound flow in case of loss of personnel (PLC). Additionally, it supports up to 255 channels (multi-flow personnel) and the availability of implementations using arithmetic with floating and fixed commas.
The newly-released OPUS 1.4 has undergone optimization of its coding parameters to improve the subjective indicators of sound quality when the feat is turned on.
For more information, visit the OPUS 1.4 release page on Github.