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Institute for Communication
Systems and Data Processing
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An Adaptive Filterbank Equalizer for Speech Enhancement

Author:
Peter Vary
Journal:
Signal Processing
Volume:
86
Number:
6
Date:
June 2006
Pages:
1206–1214
Note:
Special Issue on Applied Speech and Audio Processing (dedicated to Prof. Hänsler)
Language:
English

Abstract

A digital filter structure is proposed for fixed or adaptive spectral modification of speech and audio signals, which is derived from a reference filter bank. Main applications are audio equalization and noise reduction. The signal is filtered in the time domain, while the filter coefficients are calculated in the (short-term) frequency domain. In comparison to the usual short-term spectral-domain processing, the proposed structure exhibits distinct advantages with respect to signal delay, prototype-filter design, and complexity. The algorithm allows for uniform, as well as for non-uniform spectral resolution.