IETF Opus Interactive Audio Codec
Overview
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.
Technology
The Opus codec is designed to handle 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:
- Sampling rates from 8 to 48 kHz
- Bit-rates from 6 kb/s 510 kb/s (*)
- 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 (*)
- Frame sizes from 2.5 ms to 60 ms (*)
- Good loss robustness and packet loss concealment (PLC)
- Floating point and fixed-point implementation
(*) can be changed dynamically during a stream with no signalling required
To start, you might read the latest Internet Draft or check out the prototype code:
$ git clone git://git.opus-codec.org/opus.git