Abstract
An adaptive inverse digital filter has been developed for formant analysis of speech using the LMS adaptive algorithm of Widrow and Hoff. The inverse filter is implemented in cascade form, as opposed to the traditional direct-form implementation of adaptive filters, which simplifies both the algorithm and the utilization of its output. The simplicity of the filter and the adaptive algorithm makes this an attractive technique for real-time hardware realization. Variations and improvements of the basic algorithm are discussed.
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