We discuss the use of a variable bit rate ADPCM system for speech coding. We present an AIIPCM system, using Arithmetic Coding and Kalman Filtering techniques, that can code signals a t very low bit rates, where standard fixed Lit rate ADPCM systems fail. The proposed system, at an average bit r a t e of 16 kbps, produces a reconstructed speech signal that by informal listening t.ests !ias been judged t o be of significantly higher quality than the LD-C'ELP 16 kbps speech coding standard We believe ihat the output quality is comparable t o that of LD-CELP a t an average rate of around 12 kbps, and the quality a t 8 kbps is very promising. Other advantages of the scheme are a reduction in complrxity over the LD-CELP standard, and an increase i n design flexibility.
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