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 language | English |
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Pages | 389-393 |
Number of pages | 5 |
DOIs | |
Publication status | Published - 1998 |
Event | 32nd Asilomar Conference on Signals, Systems, and Computers - Pacific Grove, CA, United States Duration: 1 Nov 1998 → 4 Nov 1998 |
Conference
Conference | 32nd Asilomar Conference on Signals, Systems, and Computers |
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Country/Territory | United States |
City | Pacific Grove, CA |
Period | 1/11/98 → 4/11/98 |
Keywords
- acoustic distortion
- adaptive algorithm
- signal synthesis
- adaptive signal processing