Selective-tap blind signal processing for speech separation.
In this paper, we propose a new blind multichannel adaptive filtering scheme, which incorporates a partial-updating mechanism in the error gradient of the update equation. The proposed blind processing algorithm operates in the time-domain by updating only a selected portion of the adaptive filters. The algorithm steers all computational resources to filter taps having the largest magnitude gradient components on the error surface. Therefore, it requires only a small number of updates at each iteration and can substantially minimize overall computational complexity. Numerical experiments carried out in realistic blind identification scenarios indicate that the performance of the proposed algorithm is comparable to the performance of its full-update counterpart, but with the added benefit of a highly reduced computational complexity.
Kokkinakis
Kostas
K
Center for Robust Speech Systems, Department of Electrical Engineering, University of Texas at Dallas, Richardson, TX 75080, USA. kokkinak@utdallas.edu
Loizou
Philipos C
PC
D000465:Algorithms; D003198:Computer Simulation; D006801:Humans; D008297:Male; D008962:Models, Theoretical; D016571:Neural Networks, Computer; D009716:Numerical Analysis, Computer-Assisted; D010364:Pattern Recognition, Visual; D015203:Reproducibility of Results; D012815:Signal Processing, Computer-Assisted; D012984:Software; D013060:Speech; D013061:Speech Acoustics; D049250:Speech Recognition Software; D013997:Time Factors