한국통신학회논문지 (The Journal of Korean Institute of Communications and Information Sciences)

  • 제29권5C호
  • /
  • Pages.673-682
  • /
  • 2004
  • /
  • 1226-4717(pISSN)
  • /
  • 2287-3880(eISSN)
한국통신학회 (The Korean Institute of Commucations and Information Sciences)
권호 표지

적응 빔포밍 기법을 적용한 보청기 시스템의 성능 향상에 관한 연구

Improvement for Hearing Aids System Using Adaptive Beam-forming Algorithm

  • 이채욱 (대구대학교 정보통신공학과) ;
  • 오신범 (대구대학교 정보통신공학과)
  • 발행 : 2004.05.01
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⟨ 이전 논문 다음 논문 ⟩

초록

적응 빔포밍(beam-forming)기법은 디지털보청기 시스템에서 노이즈를 제거하기 위한 적절한 방식이다. 적응 빔포밍 기법은 디지털 신호처리프로세서의 발전과 더불어 최근에 주목을 받고 있는 방식으로 신호처리 과정에서 일반적으로 LMS(Least Mean Square)알고리즘을 사용하여 웨이트 벡터를 업데이트 시킨다. 본 논문에서는 신호의 환경에 따라 적응상수가 변화하는 가변스텝사이즈 알고리즘을 적용한 고속 웨이블렛 기반 적응알고리즘를 제안한다. 제안한 알고리즘을 적응 빔포머 방식인 디지털보청기 시스템에 적용하여 기존 시간영역 알고리즘과 그 특성을 비교하여 그 결과 제안한 알고리즘이 디지털보청기 시스템에 적합함을 입증한다. 그리고 제안한 알고리즘은 계산량에서도 시스템의 환경변화에도 안정하게 수렴한다는 장점이 있는 것을 보인다.

The adaptive beam-forming is promising approach for noise reduction in hearing aids. This approach has come in the focus of interest only recently, because of the availability of new and powerful digital signal processors. The adaptation U using usually a Least Mean Squares algorithm, updates the weight vector. In this Paper, we propose a fast wavelet based adaptive algorithm using variable step size algorithm which varies adaptive constant by the change of signal environment. We compared the performance of the proposed algorithm with the known adaptive algorithm using computer simulation of multi channel adaptive bemformer in hearing aids. As the result the proposed algorithm is suitable for adaptive signal processing area using hearing aids and has advantages reducing computational complexity. And we show the beam-forming system using proposed algorithm converges stably in a sudden change of system environment.

키워드

참고문헌

  1. Acoust.., Speech, and Signal Proc.., ICASSP v.6 Multimicrophone noise cancellation for improvement of hearing aid performance Shields,P.W.;Campbell,D.R.
  2. Engineering in Medicine and Biology Society, 1998. Proceedings of the 20th Annual International Conference of the IEEE v.6 Acombined fixed/adaptive beamforming noise-reduction system for hearing aids Kompis.M;Feuz.P;Valentini.G;Pelizzone.M
  3. Acoustics, Speech, and Signal Processing, ICASSP-89 Speech enhancement for hearing aids using adaptive beamformers Farassopulos,A
  4. Proc. ICASSP-93 Waelet Transform Domain LMS Algorithm Hosur;A.H.Tewfik
  5. Proceedings of the 1999 IEEE International Conference on Acoustics, Speech, and Signal Processing v.4 On the convergence properties of multidelay frequency domain adaptive filter Junghsi Lee;Sheng Chieh Chang
  6. IEEE Trams. Signal Proc.. v.39 Short-length FIR Filters and their use on fast nonrecursive filtering Z.J.Mou P.Dauhamel https://doi.org/10.1109/78.136539
  7. IEEE Trans. Acost..,Speech, Signal Proc. v.ASSP-36 Running FIR and IIR Filtering Using Multirate Filter Banks M.Vetterli
  8. IEEE Trans. Information Theory v.38 no.2 Fast Algorithms for Discrete and Continuous Wavelet Transforms O.Rioul;P.Duhamel
  9. IEEE Trans. Signal Processing v.41 Analysis and Implementation of Variable Step Size Adaptive Algorithms Joseph B.Evans;Ping Xue;B.Liu https://doi.org/10.1109/78.229885
  10. IEEE Trans. Acoustic, Speech, Signal Processing v.ASSP-35 Optimality in the choice of convergence factor for gradient based adaptive algorithm F.F.Yassa
  11. IEEE Trans. Signal Proc.. v.49 A new class of gradient adaptive step-size LMS algorithms Wee Peng Ang;Farhang Boroujeny,B https://doi.org/10.1109/78.912925