Adaptive inverse filtering of room acoustics

Equalization techniques for high order, multichannel, FIR systems are important for dereverberation of speech observed in reverberation using multiple microphones. In this case the multichannel system represents the room impulse responses (RIRs). The existence of near-common zeros in multichannel RIRs can slow down the convergence rate of adaptive inverse filtering algorithms. In this paper, the effect of common and near-common zeros on both the closed-form and the adaptive inverse filtering algorithms is studied. An adaptive shortening algorithm of room acoustics is presented based on this study.

[1]  Xiang Lin,et al.  Algorithms for identifying clusters of near-common zeros in multichannel blind system identification and equalization , 2008, 2008 IEEE International Conference on Acoustics, Speech and Signal Processing.

[2]  Naofal Al-Dhahir,et al.  FIR channel-shortening equalizers for MIMO ISI channels , 2001, IEEE Trans. Commun..

[3]  B.L. Evans,et al.  Unification and evaluation of equalization structures and design algorithms for discrete multitone modulation systems , 2005, IEEE Transactions on Signal Processing.

[4]  S. Haykin,et al.  Adaptive Filter Theory , 1986 .

[5]  Hareo Hamada,et al.  Inverse filter design and equalization zones in multichannel sound reproduction , 1995, IEEE Trans. Speech Audio Process..

[6]  Patrick A. Naylor,et al.  EVALUATION OF SPEECH DEREVERBERATION ALGORITHMS USING THE MARDY DATABASE , 2006 .

[7]  Philip A. Nelson,et al.  Algorithm for multichannel LMS adaptive filtering , 1985 .

[8]  Richard K. Martin,et al.  Low-complexity MIMO blind, adaptive channel shortening , 2005, IEEE Trans. Signal Process..

[9]  Patrick A. Naylor,et al.  Equalization of Multichannel Acoustic Systems in Oversampled Subbands , 2009, IEEE Transactions on Audio, Speech, and Language Processing.

[10]  Masato Miyoshi,et al.  Inverse filtering of room acoustics , 1988, IEEE Trans. Acoust. Speech Signal Process..

[11]  Simon Haykin,et al.  Adaptive Filter Theory 4th Edition , 2002 .

[12]  Mohammed Nafie,et al.  Time-domain equalizer training for ADSL , 1997, Proceedings of ICC'97 - International Conference on Communications.

[13]  L. Tong,et al.  Multichannel blind identification: from subspace to maximum likelihood methods , 1998, Proc. IEEE.

[14]  Charles E. Rohrs,et al.  Impulse response shortening for discrete multitone transceivers , 1996, IEEE Trans. Commun..

[15]  S. Barnett,et al.  Degrees of greatest common divisors of invariant factors of two regular polynomial matrices , 1969, Mathematical Proceedings of the Cambridge Philosophical Society.

[16]  William A. Sethares,et al.  A blind adaptive TEQ for multicarrier systems , 2002, IEEE Signal Processing Letters.

[17]  Stephen J. Elliott,et al.  A multiple error LMS algorithm and its application to the active control of sound and vibration , 1987, IEEE Trans. Acoust. Speech Signal Process..

[18]  Patrick A. Naylor,et al.  Speech Dereverberation , 2010 .