Audio Over IP Desktop Application using C/C++ OPUS Codec

Develop a simple POINT TO POINT full-duplex AUDIO desktop application for internet telephony. Preferred languague is C/C++ due to Codec, but will take recommendations. Adjustment of audio sample rates, stereo, mono, etc,... can be made through either an intuitive GUI on the application or a text based ini file. The application will require a simple way of selecting and connecting (point-to-point) with a remote machine over the internet and establishing a full-duplex audio connection using the OPUS Codec. Ability to select which local audio source (such as microphone, system audio, or other) will need to be incorporated as well. ### **Opus description:** This is a C/C++ codec designed for interactive speech and audio transmission over the Internet It is a collaboration of developers from Skype, XIPH, Juniper, Octasic and GN. Opus incorporates technology from Skype's SILK codec and [login to view URL]'s CELT codec. **Here are some key features of "Opus":** ? 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 This application should use the new open source audio codec known as OPUS. [login to view URL] ## Deliverables Utilize Open Source Audio Codec known as OPUS. This codec was developed in C/C++ More information available upon request.