Abstract
A new speech coding and multiplexing scheme matched to the asynchronous transfer mode is described. A block coding technique that is based on a variable-rate coding algorithm that makes the most of the burstiness of voice information is employed. The main feature of the scheme is considerable bit reduction, which is attained by a fairly simple algorithm. It is demonstrated that the proposed algorithm exhibits better quality than that of a 32 kb/s ADPCM at a mean bit rate of less than 13 kb/s. The effect of statistical multiplexing is verified by means of simulation employing long conversational speech samples. Methods for constructing variable- and fixed-length frames (units of information multiplexed and transferred in the network) are proposed. The proposed coding algorithm is shown to be applicable to both variable- and fixed-length frame strategies.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">></ETX>
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