The Speex Codec Manual Version 1.2 Beta 3 (2007)

 

 

The Speex codec (http://www.speex.org/) exists because there is a need for a speech codec that is open-source and free from software patent royalties. These are essential conditions for being usable in any open-source software. In essence, Speex is to speech what Vorbis is to audio/music. Unlike many other speech codecs, Speex is not designed for mobile phones but rather for packet networks and voice over IP (VoIP) applications. File-based compression is of course also supported. The Speex codec is designed to be very flexible and support a wide range of speech quality and bit-rate. Support for very good quality speech also means that Speex can encode wideband speech (16 kHz sampling rate) in addition to narrowband speech (telephone quality, 8 kHz sampling rate). Designing for VoIP instead of mobile phones means that Speex is robust to lost packets, but not to corrupted ones. This is based on the assumption that in VoIP, packets either arrive unaltered or don’t arrive at all. Because Speex is targeted at a wide range of devices, it has modest (adjustable) complexity and a small memory footprint. All the design goals led to the choice of CELP as the encoding technique. One of the main reasons is that CELP has long proved that it could work reliably and scale well to both low bit-rates (e.g. DoD CELP @ 4.8 kbps) and high bit-rates (e.g. G.728 @ 16 kbps). [via]
http://speex.org/docs/manual/speex-manual.pdf...

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