DocumentCode :
2199839
Title :
Time domain blind source separation of non-stationary convolved signals by utilizing geometric beamforming
Author :
Aichner, Robert ; Araki, Shoko ; Makino, Shoji ; Nishikawa, Tsuyoki ; Saruwatari, Hiroshi
Author_Institution :
NTT Commun. Sci. Labs., Kyoto, Japan
fYear :
2002
fDate :
2002
Firstpage :
445
Lastpage :
454
Abstract :
We propose a time-domain blind source separation (BSS) algorithm that utilizes geometric information such as sensor positions and assumed locations of sources. The algorithm tackles the problem of convolved mixtures by explicitly exploiting the non-stationarity of the acoustic sources. The learning rule is based on second-order statistics and is derived by natural gradient minimization. The proposed initialization of the algorithm is based on the beamforming principle. This method leads to improved separation performance, and the algorithm is able to estimate long unmixing FIR filters in the time domain due to the geometric initialization. We also propose a post-filtering method for dewhitening which is based on the scaling technique in frequency-domain BSS. The validity of the proposed method is shown by computer simulations. Our experimental results confirm that the algorithm is capable of separating real-world speech mixtures and can be applied to short learning data sets down to a few seconds. Our results also confirm that the proposed dewhitening post-filtering method maintains the spectral content of the original speech in the separated output.
Keywords :
FIR filters; acoustic signal processing; adaptive signal processing; blind source separation; convolution; filtering theory; learning (artificial intelligence); minimisation; speech processing; statistical analysis; time-domain analysis; algorithm initialization; beamforming; computer simulations; dewhitening post-filtering method; frequency-domain BSS; geometric beamforming; geometric initialization; learning rule; long unmixing FIR filters; natural gradient minimization; nonstationary acoustic sources; nonstationary convolved signals; post-filtering method; real-world speech mixtures; scaling technique; second-order statistics; sensor positions; separation performance; short learning data sets; source locations; speech spectral content; time domain blind source separation; time-domain BSS algorithm; Acoustic sensors; Array signal processing; Blind source separation; Finite impulse response filter; Frequency domain analysis; Laboratories; Source separation; Speech; Statistics; Time domain analysis;
fLanguage :
English
Publisher :
ieee
Conference_Titel :
Neural Networks for Signal Processing, 2002. Proceedings of the 2002 12th IEEE Workshop on
Print_ISBN :
0-7803-7616-1
Type :
conf
DOI :
10.1109/NNSP.2002.1030056
Filename :
1030056
Link To Document :
بازگشت