Adaptive equalization in oversampled subbands

Stephan Weiss, Saul Dooley, Robert Stewart, Nandi Asoke

Research output: Contribution to conferencePaper

2 Citations (Scopus)
32 Downloads (Pure)

Abstract

The potential presence of fractional delays, non-minimum phase parts, and a colouring of the channel output can require adaptive equalizers to adapt very long filters, which can have slow convergence for LMS-type adaptive algorithms. These problems can be addressed by a subband approach to reduce computational complexity and improve convergence speed. We discuss, why amongst other possibilities of subband processing the oversampled approach is particularly appealing to significantly reduce computational complexity and improve convergence speed. Simulation results for typical systems found in acoustics and communication channels are presented and highlight the benefit of our method.
Original languageEnglish
Pages389-393
Number of pages5
DOIs
Publication statusPublished - 1998
Event32nd Asilomar Conference on Signals, Systems, and Computers - Pacific Grove, CA, United Kingdom
Duration: 1 Nov 19984 Nov 1998

Conference

Conference32nd Asilomar Conference on Signals, Systems, and Computers
CountryUnited Kingdom
CityPacific Grove, CA
Period1/11/984/11/98

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Keywords

  • acoustic distortion
  • adaptive algorithm
  • signal synthesis
  • adaptive signal processing

Cite this

Weiss, S., Dooley, S., Stewart, R., & Asoke, N. (1998). Adaptive equalization in oversampled subbands. 389-393. Paper presented at 32nd Asilomar Conference on Signals, Systems, and Computers, Pacific Grove, CA, United Kingdom. https://doi.org/10.1109/ACSSC.1998.750892