Codage algébrique de la parole : prédiction linéaire à excitation par code ternaire

This article presents new speech coding methods for real time application (telephone, videophone) or offline applications (storage). Speech quality is in the classical telephone range, with a 4 kHz bandwidth and a sampling at 8 kHz. An elementary approach leads to a 16 kbit/s codec and a 24 kbit/s codec, using integer codebooks and fast computations. The speech quality of the two codecs has been measured in comparison with more complex ones and in realistic conditions, with noisy telecommunication channels. The elementary approach is completed by a synthetic model, with a systematic generalization of the algorithms (e.g. for a generalized vselp). Some methods for channel protection, which are already known by the speech coding researchers, are summed up in the Appendix. A change of representation for low density codes (less than 1 bit/sample) is proposed.