• DocumentCode
    2329184
  • Title

    Proportionate Frequency Domain Adaptive Algorithms for Blind Channel Identification

  • Author

    Ahmad, Rehan ; Khong, Andy W H ; Naylor, Patrick A.

  • Author_Institution
    Imperial Coll. London
  • Volume
    5
  • fYear
    2006
  • fDate
    14-19 May 2006
  • Abstract
    We present fast-converging adaptive blind channel identification algorithms for acoustic room impulse responses. These new algorithms exploit the fast-convergence of the improved proportionate normalized least-mean-square (IPNLMS) algorithm and address the problem of delay inherent in frequency domain algorithms by employing the multi-delay filter (MDF) structure. Simulation results for both speech and white Gaussian noise show that the proposed algorithms outperform current frequency domain blind channel estimation algorithms
  • Keywords
    acoustic signal processing; channel estimation; least mean squares methods; speech processing; acoustic room impulse responses; adaptive blind channel identification algorithms; blind channel identification; frequency domain algorithms; improved proportionate normalized least-mean-square algorithm; multi-delay filter structure; proportionate frequency domain adaptive algorithms; speech noise; white Gaussian noise; Adaptive algorithm; Additive noise; Convergence; Delay; Educational institutions; Filters; Frequency domain analysis; Higher order statistics; Signal processing algorithms; Speech enhancement;
  • fLanguage
    English
  • Publisher
    ieee
  • Conference_Titel
    Acoustics, Speech and Signal Processing, 2006. ICASSP 2006 Proceedings. 2006 IEEE International Conference on
  • Conference_Location
    Toulouse
  • ISSN
    1520-6149
  • Print_ISBN
    1-4244-0469-X
  • Type

    conf

  • DOI
    10.1109/ICASSP.2006.1661204
  • Filename
    1661204