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
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