Add support for Opus Interactive Audio Codec

[Premislaus]

​http://www.opus-codec.org/

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:

    Bit-rates from 6 kb/s to 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

You can read the specification in the latest Internet Draft. 

[diver]

Opus will be supported by ​FFmpeg.