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
Publication Date: May 1, 1980 Import into BibTeX
Permalink: https://www.aes.org/e-lib/browse.cfm?elib=3734

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