Analysis of the adaptive filter algorithm for feedback-type active noise control

被引：0

作者：

Sakai, H ^{[1
]}

Miyagi, S ^{[1
]}

机构：

[1] Kyoto Univ, Grad Sch Informat, Dept Syst Sci, Sakyo Ku, Kyoto 6068501, Japan

来源：

2001 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, VOLS I-VI, PROCEEDINGS: VOL I: SPEECH PROCESSING 1; VOL II: SPEECH PROCESSING 2 IND TECHNOL TRACK DESIGN & IMPLEMENTATION OF SIGNAL PROCESSING SYSTEMS NEURALNETWORKS FOR SIGNAL PROCESSING; VOL III: IMAGE & MULTIDIMENSIONAL SIGNAL PROCESSING MULTIMEDIA SIGNAL PROCESSING | 2001年

关键词：

D O I：

暂无

中图分类号：

O42 [声学];

学科分类号：

070206 ; 082403 ;

摘要：

The feedback-type active noise control (ANC) system uses only one microphone to provide necessary signals for adjusting the adaptive filter. Due to the complicated nature of the whole adaptive filter structure there have been no theoretical results about its convergence properties. In this paper, first a stationary point of the adaptive filter using the filtered-X LMS algorithm is obtained by the averaging method combined with the frequency domain technique. Then the local convergence condition is derived. This is a counterpart of the well-known 90degrees condition for the feedforward-type ANC. Finally, the convergence condition is explicitly given for a simple example and its validity is shown by some simulations.

引用

页码：3841 / 3844

页数：4

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