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Adaptive Blind Multichannel System Identification

  • Chapter
Speech Dereverberation

Part of the book series: Signals and Commmunication Technology ((SCT))

Abstract

The use of adaptive algorithms for blind system identification in speech dereverberation was proposed recently. This chapter reviews adaptive multichannel system identification using minimization of the cross-relation error. One of the algorithms that adopt this approach is the Normalized Multichannel Frequency Domain Least Mean Square (NMCFLMS) algorithm. We show that, in the presence of additive noise, the coefficients of the adaptive filter employing NMCFLMS converge initially toward the true acoustic impulse responses after which they then misconverge. We provide a technique to address this misconvergence problem in NMCFLMS. This is achieved by reformulating the minimization problem into one involving a constraint. As will be shown, this constrained minimization problem requires knowledge of the direct-path components of the acoustic impulse responses and one of the main contributions of this work is to illustrate how these direct-path components can be estimated under practical conditions.We will then illustrate how these estimates can be incorporated into the proposed extended NMCFLMS (ext-NMCFLMS) algorithm so as to address the problem of misconvergence. The simulation results presented showthe noise robustness of the proposed algorithm for both white Gaussian noise and speech inputs. In addition, we illustrate how errors due to the estimation of the direct-paths affect the performance of the proposed algorithm.

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Khong, A., Naylor, P. (2010). Adaptive Blind Multichannel System Identification. In: Naylor, P., Gaubitch, N. (eds) Speech Dereverberation. Signals and Commmunication Technology. Springer, London. https://doi.org/10.1007/978-1-84996-056-4_6

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  • DOI: https://doi.org/10.1007/978-1-84996-056-4_6

  • Publisher Name: Springer, London

  • Print ISBN: 978-1-84996-055-7

  • Online ISBN: 978-1-84996-056-4

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