Some New Methods for Digital Encoding of Voice Signals and for Voice Code Translation

Abstract

Recently developed techniques are described for improving the speech quality of voice signals that are first digitally encoded, placed in random access storage, and on demand are then translated into normal speech in an audio response unit under the control of a host processor. The development is an extension and modification of the channel vocoder principle. Speech quality is enhanced by hardware and software features for treatment of unvoiced components of the coded speech signal in particular by separating harmonics from the excitation function digital signal before smoothing. A new program of bit selection is used to assure that the aggregation function digital signal carries maximum information. In addition, an efficient method of storage assignment is shown for the excitation function and the aggregate function registers in the voice code translator.