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Speech Signal Processor Using Comb Filter, Time Compression and Expansion of Speech

An improved speech signal processor uses shift registers capable of control for frequency transformation of the input signal. A pair of complementary comb filters are provided for filtering out those frequencies which would produce disruptive discontinuities due to sampling and subsequent reassembling their samples. This method provides a simple and economical solution to the problems of improving qualities of time compressed and expanded speeches.

Authors: Hirata, Yoshimutsu; Ueki, Masaaki; Hayashibara, Tohru
Affiliation: Waseda University, Tokyo, Japan
AES Convention:66 (May 1980) Paper Number:1641 Import into BibTeX

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