Wavelet-based adaptive filtering

  • Authors:
  • Milos Doroslovacki;Hong Fan

  • Affiliations:
  • Department of ECE, University of Cincinnati, Cincinnati, OH;Department of ECE, University of Cincinnati, Cincinnati, OH

  • Venue:
  • ICASSP'93 Proceedings of the 1993 IEEE international conference on Acoustics, speech, and signal processing: digital speech processing - Volume III
  • Year:
  • 1993

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Abstract

Theoretical and experimental analysis and description of wavelet-based filtering are given in the case of stationary desired signal. The impulse responses of adaptive filter and unknown system producing the desired signal are represented by discrete-time wavelet series. We have found the coefficients that minimize the mean square error and pointed out the time-frequency localized structure of modeling error. An LMS adaptive filtering algorithm is derived. Its transform domain interpretation is shown, as well as possibilities for faster convergence and better numerical properties. We have noticed the better modeling of desired signals in the time-frequency plane, faster convergence, and smaller error than in the case of FIR filters.