Opus

Previously ported / uploaded by kas1e

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Opus Interactive Audio Codec


Overview
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Opus is a totally open, royalty-free, highly versatile audio codec. Opus is
unmatched for interactive speech and music transmission over the Internet, but
is also intended for storage and streaming applications. It is standardized by
the Internet Engineering Task Force (IETF) as RFC 6716 which incorporated
technology from Skype’s SILK codec and Xiph.Org’s CELT codec.


Technology
==========

Opus can handle a wide range of audio applications, including Voice over IP,
videoconferencing, in-game chat, and even remote live music performances. It can
scale from low bitrate narrowband speech to very high quality stereo music.
Supported features are:

    Bitrates from 6 kb/s to 510 kb/s
    Sampling rates from 8 kHz (narrowband) to 48 kHz (fullband)
    Frame sizes from 2.5 ms to 60 ms
    Support for both constant bitrate (CBR) and variable bitrate (VBR)
    Audio bandwidth from narrowband to fullband
    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

You can read the full specification, including the reference implementation, in
RFC 6716. An up-to-date implementation of the Opus standard is also available
from the downloads page.