Opus codec runtime library The Opus codec is designed for interactive speech and audio transmission over the Internet. It is designed by the IETF Codec Working Group and incorporates technology from Skype's SILK codec and Xiph.Org's CELT codec. . It is intended to suit a wide range of interactive audio applications, including Voice over IP, videoconferencing, in-game chat, and even remote live music performances. It can scale from low bit-rate narrowband speech to very high quality stereo music. The current features are: . Bit-rates from 6 kb/s 510 kb/s Sampling rates from 8 to 48 kHz Frame sizes from 2.5 ms to 60 ms Support for both constant bit-rate (CBR) and variable bit-rate (VBR) Audio bandwidth from narrowband to full-band Support for speech and music Support for mono and stereo Support for up to 255 channels (multistream frames) Dynamically adjustable bitrate, audio bandwidth, and frame size Good loss robustness and packet loss concealment (PLC) Floating point and fixed-point implementation . This package provides the Opus runtime library.
Version History
Packages are not hosted on our server, they could be protected or unavailable.
