DocumentCode
3607394
Title
Convolutive Blind Source Separation Using an Iterative Least-Squares Algorithm for Non-Orthogonal Approximate Joint Diagonalization
Author
Saito, Shinya ; Oishi, Kunio ; Furukawa, Toshihiro
Author_Institution
Dept. of Manage. Sci., Tokyo Univ. of Sci., Tokyo, Japan
Volume
23
Issue
12
fYear
2015
Firstpage
2434
Lastpage
2448
Abstract
In this paper, we present an approach of recovering signal waveforms of speech sources from observed signals in noisy and reverberant environments. The approach is based on approximate joint diagonalization estimate to provide interference suppression of source signals and reduce echoes and distortions of separated signals. In the proposed approach, the mixing matrix is estimated by minimizing the constrained direct least-squares (LS) criterion in direct model. Exclusively under the condition where the estimated mixing matrix is not of full rank, it is replaced by a full-rank matrix. The unmixing matrix from the estimated mixing matrix is obtained by setting the frequency response of the composite mixing-unmixing filter to identity matrix. The cross-spectral density diagonal matrices of the source signals are precisely estimated by minimizing the indirect LS criterion in indirect model. These operations are fulfilled by using alternating least-squares algorithm. The correlation between the interfrequency power ratios is used to prevent a misalignment permutation of the unmixing matrix. Finally, we compare the proposed BSS with a number of conventional BSS methods in noisy and reverberant environments under both artificial and actual conditions.
Keywords
blind source separation; distortion; echo suppression; filtering theory; frequency response; iterative methods; least mean squares methods; matrix algebra; minimisation; reverberation; speech processing; BSS method; alternating least squares algorithm; composite mixing-unmixing filter; constrained direct least square criterion minimisation; convolutive blind source separation; cross-spectral density diagonal matrix; direct model; echo reduction; frequency response; full-rank matrix; identity matrix; interference suppression; iterative least squares algorithm; mixing matrix estimation; noisy environment; nonorthogonal approximate joint diagonalization; reverberant environment; separated signal distortion; signal waveform recovery; source signal; speech source; unmixing matrix; Blind source separation; Lagrange multipliers; Least square methods; Microphones; Noise measurement; Speech processing; Alternating least-squares (ALS) algorithm; approximate joint diagonalization (AJD); blind identification; blind source separation (BSS); convolutive linear mixture; method of Lagrange multipliers;
fLanguage
English
Journal_Title
Audio, Speech, and Language Processing, IEEE/ACM Transactions on
Publisher
ieee
ISSN
2329-9290
Type
jour
DOI
10.1109/TASLP.2015.2485663
Filename
7286760
Link To Document