1:00, SPTM-L1.1
SPEECH DEREVERBERATION VIA MAXIMUM-KURTOSIS SUBBAND ADAPTIVE FILTERING
B. GILLESPIE, H. MALVAR, D. FLORENCIO
This paper presents an efficient algorithm for high-quality speech capture in applications such as hands-free teleconferencing or voice recording by personal computers. We process the micro-phone signals by a subband adaptive filtering structure using a modulated complex lapped transform (MCLT), in which the subband filters are adapted to maximize the kurtosis of the linear prediction (LP) residual of the reconstructed speech. In this way, we attain good solutions to the problem of blind speech derever-beration. Experimental results with actual data, as well as with artificially difficult reverberant situations, show very good per-formance, both in terms of a significant reduction of the per-ceived reverberation, as well as improvement in spectral fidelity.

1:20, SPTM-L1.2
FILTERED GRADIENT ALGORITHMS APPLIED TO A SUBBAND ADAPTIVE FILTER STRUCTURE
J. APOLINARIO JR., R. ALVES, P. DINIZ, M. SWAMY
Adaptive filtering techniques in subbands have been recently developed for a number of applications including acoustic echo cancellation and wideband active noise control. In such applications, hundred of taps are required resulting in high computational complexity and low convergence rate when using LMS based algorithms. For fullband systems, new algorithms which try to overcome these drawbacks have been investigated. A class of these algorithms employing variants of the filtered gradient adaptive (FGA) algorithm has been successfully developed. In this paper, we apply these techniques to a recently proposed subband adaptive filter structure in order to improve the convergence rate and the computational load. Computer simulations show the benefits obtained with these proposed algorithms.

1:40, SPTM-L1.3
SUBBAND MMSE BOUNDS FOR FILTERBANK ADAPTATION IN SYSTEM MODELING NON-UNIFORM SUBBAND ADAPTIVE FILTERS
J. GRIESBACH, M. LIGHTNER, D. ETTER
Minimum mean square error estimators have been developed for non-uniform subband adaptive filters (SAFs) in system modeling configurations. The next step towards practical implementation of non-uniform SAFs involves developing an adaptive algorithm to control the non-uniform filterbank's bandwidths and decimation factors. This paper constructs such an algorithm using subband minimum mean square error (MSE) bounds to suggest decimation factors. A numerical simulation shows that a non-uniform SAF can achieve lower MSE with lower complexity than a equivalent uniform SAF.

2:00, SPTM-L1.4
A NEW CRITICALLY SAMPLED NON-UNIFORM SUBBAND ADAPTIVE STRUCTURE
R. VASCONCELLOS, M. PETRAGLIA, R. ALVES
In this paper, a non-uniform subband adaptive structure with critical sampling of the subband signals is derived. With the assumption of non-overlapping between non-adjacent analysis filters, the resulting structure yields exact modeling of an arbitrary FIR system. An LMS-type adaptation algorithm, which minimizes the sum of the subband squared-errors, is obtained for updating the coefficients of the proposed structure, resulting in significant convergence rate improvement for colored input signals when compared to the fullband LMS algorithm. Computer simulations illustrate the convergence behavior of the proposed non-uniform subband adaptive filter in the applications of system identification and acoustic echo cancellation.

2:20, SPTM-L1.5
DYNAMIC STRUCTURES FOR NON-UNIFORM SUBBAND ADAPTIVE FILTERING
A. OAKMAN, P. NAYLOR
Subband adaptive filters suffer degraded performance when high input energy occurs at frequencies coincident with subband boundaries. This is seen as increased error in critically sampled systems and as reduced asymptotic convergence speed in oversampled systems. To address this problem a dynamic frequency decomposition scheme is presented which aims to control the frequency of subband boundaries such that they avoid spectral regions of high input energy. An efficient structure for this is described, which maintains the low complexity advantage of subband systems. Simulation results show reductions in MSE of around 5-10dBs in the critical case and convergence improvement in the oversampled case, in addition to increased robustness to coloured inputs in both cases.

2:40, SPTM-L1.6
ROBUST MICROPHONE ARRAYS USING SUBBAND ADAPTIVE FILTERS
B. FARHANG-BOROUJENY, W. NEO
A new adaptive beamformer which combines the idea of subband processing and the generalized sidelobe canceller structure is presented. The proposed subband beamformer has a blocking matrix that uses coefficient-constrained subband adaptive filters to limit target cancellation within an allowable range of direction of arrival. Simulations compare the fullband adaptive beamformer and the subband adaptive beamformer show that the subband beamformer has better performance than the fullband beamformer when the input signals to the microphone array are coloured. In reverberant environments, also, the proposed subband beamformer performs better than its fullband counterpart.