An adaptive inverse digital filter for formant analysis of speech
Document Type
Conference Proceeding
Date of Original Version
1-1-1976
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.
Publication Title, e.g., Journal
ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings
Volume
1976-April
Citation/Publisher Attribution
Jackson, Leland B., and John Bertrand. "An adaptive inverse digital filter for formant analysis of speech." ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings 1976-April, (1976): 84-86. doi: 10.1109/ICASSP.1976.1170070.