Opus codec library development files

 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 library headers and development files.