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D. Freitas, and A. Marques, "A Novel IIR Equalizer for Non-Minimum Phase Loudspeaker Systems," Paper 6953, (2006 October.). doi: D. Freitas, and A. Marques, "A Novel IIR Equalizer for Non-Minimum Phase Loudspeaker Systems," Paper 6953, (2006 October.). doi:
Abstract: A novel approach for the equalization of non-minimum phase loudspeaker systems based on the design of an IIR inverse filter is presented. This IIR inverse filter is designed in time domain by minimization of the least squares error function that results of using the typical “Output Error” configuration in the inverse modeling of non-minimum phase systems, with an adjustable delay. Due to the nonlinear nature of the error function, iterative optimization methods for nonlinear least squares problems were applied, namely the Levenberg-Marquardt method. This approach allows the design of inverse filter based equalization solutions with lower computational requirements, lower equalization error and lower delay of the equalized loudspeaker system than the most used one, the FIR based inverse filter. The advantages of this new approach are demonstrated with its application for the equalization of two loudspeaker systems. The results of the objective evaluation of this application are outlined, presented and discussed regarding time and frequency domain equalization errors and the delay of the equalized loudspeaker.

@article{freitas2006a,
author={freitas, diamantino and marques, avelino},
journal={journal of the audio engineering society},
title={a novel iir equalizer for non-minimum phase loudspeaker systems},
year={2006},
volume={},
number={},
pages={},
doi={},
month={october},} @article{freitas2006a,
author={freitas, diamantino and marques, avelino},
journal={journal of the audio engineering society},
title={a novel iir equalizer for non-minimum phase loudspeaker systems},
year={2006},
volume={},
number={},
pages={},
doi={},
month={october},
abstract={a novel approach for the equalization of non-minimum phase loudspeaker systems based on the design of an iir inverse filter is presented. this iir inverse filter is designed in time domain by minimization of the least squares error function that results of using the typical “output error” configuration in the inverse modeling of non-minimum phase systems, with an adjustable delay. due to the nonlinear nature of the error function, iterative optimization methods for nonlinear least squares problems were applied, namely the levenberg-marquardt method. this approach allows the design of inverse filter based equalization solutions with lower computational requirements, lower equalization error and lower delay of the equalized loudspeaker system than the most used one, the fir based inverse filter. the advantages of this new approach are demonstrated with its application for the equalization of two loudspeaker systems. the results of the objective evaluation of this application are outlined, presented and discussed regarding time and frequency domain equalization errors and the delay of the equalized loudspeaker.},}

TY - paper
TI - A Novel IIR Equalizer for Non-Minimum Phase Loudspeaker Systems
SP - EP -
AU - Freitas, Diamantino
AU - Marques, Avelino
PY - 2006
JO - Journal of the Audio Engineering Society
IS -
VO -
VL -
Y1 - October 2006 TY - paper
TI - A Novel IIR Equalizer for Non-Minimum Phase Loudspeaker Systems
SP - EP -
AU - Freitas, Diamantino
AU - Marques, Avelino
PY - 2006
JO - Journal of the Audio Engineering Society
IS -
VO -
VL -
Y1 - October 2006
AB - A novel approach for the equalization of non-minimum phase loudspeaker systems based on the design of an IIR inverse filter is presented. This IIR inverse filter is designed in time domain by minimization of the least squares error function that results of using the typical “Output Error” configuration in the inverse modeling of non-minimum phase systems, with an adjustable delay. Due to the nonlinear nature of the error function, iterative optimization methods for nonlinear least squares problems were applied, namely the Levenberg-Marquardt method. This approach allows the design of inverse filter based equalization solutions with lower computational requirements, lower equalization error and lower delay of the equalized loudspeaker system than the most used one, the FIR based inverse filter. The advantages of this new approach are demonstrated with its application for the equalization of two loudspeaker systems. The results of the objective evaluation of this application are outlined, presented and discussed regarding time and frequency domain equalization errors and the delay of the equalized loudspeaker.