Abstract
We have combined the standard front-end filter bank with a feature extraction process to produce a model that requires dramatically less hardware (or software). This new model, based on the frequency-sampled ratio spectrum, can be interpreted as a small set of constant-Q filters whose center frequencies adapt to locations of high signal energy. The resulting feature vectors are shown to outperform several competing techniques for phoneme recognition. Results from fabricated CMOS analog VLSI circuits illustrate a hardware efficient method to sample the ratio spectrum.
Recommended Citation
J. G. Harris and S. J. Lim, "An Analog Front-end Speech Processor Using The Ratio Spectrum," Proceedings IEEE International Symposium on Circuits and Systems, vol. 3, pp. I - 327, Institute of Electrical and Electronics Engineers, Jan 2000.
The definitive version is available at https://doi.org/10.1109/ISCAS.2000.856063
Department(s)
Electrical and Computer Engineering
International Standard Serial Number (ISSN)
0271-4310
Document Type
Article - Conference proceedings
Document Version
Citation
File Type
text
Language(s)
English
Rights
© 2025 Institute of Electrical and Electronics Engineers, All rights reserved.
Publication Date
01 Jan 2000
