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Adaptive IIR filters offer a simple method for in-situ tracking of loudspeaker parameters. The convergence properties of standard adaptive IIR algorithms can, however, be sub-optimal, when applied to loudspeaker system identification using electrical impedance measurement. The normalized-LMS IIR algorithm described by Widrow and Stearns (1985) [ ] either converges slowly or creates excess residual error. This paper shows that, by varying the convergence parameter with the error signal, and by using the square-root of the gradient estimate, convergence time and residual error can be kept within useful limits. Examples based on music signals measured from actual loudspeakers are shown.
Author (s): Bright, Andrew;
Affiliation:
(See document for exact affiliation information.)
Publication Date:
2007-09-06
Session subject:
DSP for Loudspeakers
DOI:
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Bright, Andrew; 2007; Adaptive IIR Filters for Loudspeaker Parameter Tracking [PDF]; ; Paper 22; Available from: https://aes.org/publications/elibrary-page/?id=14218
Bright, Andrew; Adaptive IIR Filters for Loudspeaker Parameter Tracking [PDF]; ; Paper 22; 2007 Available: https://aes.org/publications/elibrary-page/?id=14218
@inproceedings{Bright2007adaptive,
title={{Adaptive IIR Filters for Loudspeaker Parameter Tracking}},
author={Bright, Andrew},
year={2007},
month={sep},
booktitle={Journal of the Audio Engineering Society},
publisher={Paper 22; AES Conference: 32nd International Conference; September 2007},
number={22},
organization={AES},
}
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