Opus audio codec (RFC 6716): API and operations manual
| 1.3.1

Opus

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.

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. Its main features are:

- Sampling rates from 8 to 48 kHz
- Bit-rates from 6 kb/s to 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
- Support for multichannel (up to 255 channels)
- Frame sizes from 2.5 ms to 60 ms
- Good loss robustness and packet loss concealment (PLC)
- Floating point and fixed-point implementation

Documentation sections:

- Opus Encoder
- Opus Decoder
- Repacketizer
- Opus Multistream API
- Opus library information functions
- Opus Custom

For more information visit the Opus Website. |