Adaptive feedback cancellation in hearing aids using the IPLS algorithm

Randall S. Plate, Yongchao Wang, Zhi-Quan Luo, Chris Gao

Research output: Chapter in Book/Report/Conference proceedingConference contribution

3 Scopus citations

Abstract

Hearing aids suffer from the presence of a positive feedback loop between the output transducer and microphone. This feedback reduces both the stable gain achievable in the forward path as well as the sound quality of the output. Existing solutions to this problem perform feedback cancellation using adaptive filtering. The most common methods of adaptive filtering use the Least Mean Square (LMS) or Recursive Least Squares (RLS) algorithms, with the LMS being the most common choice for hearing aid applications. However, these approaches suffer from various drawbacks due to the unique challenges inherent in hearing aids. A new approach employing the Interior Point Least Squares (IPLS) algorithm demonstrates some distinct advantages over these other algorithms for use in the hearing aid schematic. Specifically, it is capable of achieving slightly faster convergence and, more significantly, maintains desired performance trade-offs with fixed parameter values despite changes in input signal power.

Original languageEnglish (US)
Title of host publication2010 IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP 2010 - Proceedings
Pages177-180
Number of pages4
DOIs
StatePublished - Nov 8 2010
Event2010 IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP 2010 - Dallas, TX, United States
Duration: Mar 14 2010Mar 19 2010

Publication series

NameICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings
ISSN (Print)1520-6149

Other

Other2010 IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP 2010
Country/TerritoryUnited States
CityDallas, TX
Period3/14/103/19/10

Keywords

  • Adaptive filtering
  • Hearing aid
  • IPLS

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