Title :
Audio source separation based on independent component analysis
Author :
Makino, Shoji ; Araki, Shoko ; Mukai, Ryo ; Sawada, Hiroshi
Author_Institution :
NTT Commun. Sci. Labs., NTT Corp., Kyoto, Japan
Abstract :
This paper introduces the blind source separation (BSS) of convolutive mixtures of acoustic signals, especially speech. A statistical and computational technique, called independent component analysis (ICA), is examined. By achieving nonlinear decorrelation, nonstationary decorrelation, or time-delayed decorrelation, we can find source signals only from observed mixed signals. Particular attention is paid to the physical interpretation of BSS from the acoustical signal processing point of view. Frequency-domain BSS is shown to be equivalent to two sets of frequency domain adaptive microphone arrays, i.e., adaptive beamformers (ABFs). Although BSS can reduce reverberant sounds to some extent in the same way as ABF, it mainly removes the sound from the jammer direction. This is why BSS has difficulties with long reverberation in the real world. If sources are not "independent," the dependence results in bias noise when obtaining the correct unmixing filter coefficients. Therefore, the performance of BSS is limited by that of ABF. Although BSS is upper bounded by ABF, BSS has a strong advantage over ABF. BSS can be regarded as an intelligent version of ABF in the sense that it can adapt without any information on the array manifold or the target direction, and sources can be simultaneously active in BSS.
Keywords :
acoustic signal processing; audio signal processing; blind source separation; convolution; decorrelation; independent component analysis; acoustical signal processing; adaptive beamformers; adaptive microphone arrays; array manifold; audio source separation; blind source separation; convolutive acoustic signal mixture; frequency-domain BSS; independent component analysis; jammer direction; long reverberation; mixed signals; nonlinear decorrelation; nonstationary decorrelation; reverberant sound reduction; sound removal; source signals; speech signal; target direction; time-delayed decorrelation; unmixing filter coefficient; Acoustic signal processing; Adaptive arrays; Adaptive signal processing; Blind source separation; Decorrelation; Frequency domain analysis; Independent component analysis; Microphone arrays; Source separation; Speech;
Conference_Titel :
Circuits and Systems, 2004. ISCAS '04. Proceedings of the 2004 International Symposium on
Print_ISBN :
0-7803-8251-X
DOI :
10.1109/ISCAS.2004.1329896