• DocumentCode
    2321553
  • Title

    An alternative formulation for low rank transform domain adaptive filtering

  • Author

    Hunter, Todd E. ; Linebarger, D.A.

  • Author_Institution
    Dept. of Electr. Eng., Texas Univ., Dallas, TX, USA
  • Volume
    1
  • fYear
    2000
  • fDate
    2000
  • Firstpage
    29
  • Abstract
    Linebarger et al. (see Proceedings of Thirty-First Asilomar Conference on Signals, Systems & Computers, Pacific Grove, California, USA, vol.1, p.123-27, 1997) introduced a new, optimal approach to low rank transform domain adaptive filtering is , using a least squares, matrix based framework. Further, Raghothaman (see Proceedings of 8th IEEE DSP Workshop, Utah, USA, 1998) provides a computationally efficient algorithm to solve the formulation of the problem proposed by Linebarger. In this paper, we examine an alternative method for applying an optimal low rank transform, within the framework derived by Linebarger, to convert an overdetermined, full rank system into a low rank system. In addition, we propose a computationally efficient algorithm for the implementation of our method, using the DCT as the unitary transformation. Finally, we evaluate the performance of our algorithm via simulation in an acoustic echo canceller application, and show that the performance of our method is superior to existing low rank methods, NLMS and affine projection
  • Keywords
    acoustic signal processing; adaptive filters; adaptive signal processing; discrete cosine transforms; echo suppression; filtering theory; least squares approximations; matrix algebra; DCT; NLMS; acoustic echo canceller application; affine projection; computationally efficient algorithm; least squares; low rank methods; low rank transform domain adaptive filtering; matrix based framework; optimal approach; optimal low rank transform; overdetermined full rank system; performance evaluation; simulation; unitary transformation; Acoustic applications; Adaptive filters; Discrete cosine transforms; Echo cancellers; Equations; Error correction; Finite impulse response filter; Least squares methods; Nonlinear filters; Signal processing algorithms;
  • fLanguage
    English
  • Publisher
    ieee
  • Conference_Titel
    Acoustics, Speech, and Signal Processing, 2000. ICASSP '00. Proceedings. 2000 IEEE International Conference on
  • Conference_Location
    Istanbul
  • ISSN
    1520-6149
  • Print_ISBN
    0-7803-6293-4
  • Type

    conf

  • DOI
    10.1109/ICASSP.2000.861852
  • Filename
    861852