The codec was developed by David Grant Rowe, with support and cooperation of other researchers (e.g., Jean-Marc Valin from Opus).
[citation needed] The speech codec uses 16-bit PCM sampled audio, and outputs packed digital bytes.
[3] The source code is released under the terms of version 2.1 of the GNU Lesser General Public License (LGPL).
The software is developed on Linux and a port for Microsoft Windows created with Cygwin is offered in addition to an Apple MacOS version.
[6] Internally, parametric audio coding algorithms operate on 10 ms PCM frames using a model of the human voice.
Spoken audio is recreated by modelling speech as a sum of harmonically related sine waves with independent amplitudes called Line spectral pairs, or LSP, on top of a determined fundamental frequency of the speaker's voice (pitch).
The (quantised) pitch and the amplitude (energy) of the harmonics are encoded, and with the LSP's are exchanged across a channel in a digital format.
There was an FM-to-Codec2 digital voice repeater in earth orbit on amateur radio CubeSat LilacSat-1 (call sign ON02CN, QB50 constellation), which was launched and subsequently deployed from the International Space Station in 2017.
[14][15] The underlying sinusoidal modelling goes back to developments by Robert J. McAulay and Thomas F. Quatieri (MIT Lincoln labs) from the mid-1980s.
In January 2012, at linux.conf.au, Jean-Marc Valin helped improve the quantization of line spectral pairs, which Rowe is less familiar with.