Promoting convergence in multi-channel blind signal separation using PNLMS

The proportionate normalized least-mean squares (PNLMS) adaptation algorithm exploits the sparse nature of acoustic impulse responses and assigns adaptation gain proportional to the absolute value of filter coefficients, thereby resulting in faster convergence. In the past it has shown to improve convergence of acoustic paths in echo-cancellation applications. In this paper, we investigate the use of PNLMS algorithm for blind speech separation and show that with a careful selection of operating parameters the PNLMS algorithm greatly helps promote convergence of the un-mixing filters when compared to the conventional normalized least-mean-squares (NLMS) adaptation. The PNLMS based blind speech separation is suitable for real-time implementation as it promises faster convergence and requires only a modest increase in complexity as compared to the NLMS algorithm.

[1]  Donald L. Duttweiler,et al.  Proportionate normalized least-mean-squares adaptation in echo cancelers , 2000, IEEE Trans. Speech Audio Process..

[2]  Dirk Van Compernolle,et al.  Signal separation by symmetric adaptive decorrelation: stability, convergence, and uniqueness , 1995, IEEE Trans. Signal Process..

[3]  Hiroshi Sawada,et al.  Natural gradient multichannel blind deconvolution and speech separation using causal FIR filters , 2004, IEEE Transactions on Speech and Audio Processing.

[4]  Jont B. Allen,et al.  Image method for efficiently simulating small‐room acoustics , 1976 .

[5]  Dennis R. Morgan,et al.  Exploring permutation inconsistency in blind separation of speech signals in a reverberant environment , 2000, 2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100).

[6]  Kiyohiro Shikano,et al.  Blind source separation based on a fast-convergence algorithm combining ICA and beamforming , 2006, IEEE Transactions on Audio, Speech, and Language Processing.

[7]  Changshui Zhang,et al.  Fast nonlinear autocorrelation algorithm for source separation , 2009, Pattern Recognit..

[8]  S. Haykin Unsupervised adaptive filtering, vol. 1: Blind source separation , 2000 .

[9]  Walter Kellermann,et al.  A generalization of blind source separation algorithms for convolutive mixtures based on second-order statistics , 2005, IEEE Transactions on Speech and Audio Processing.

[10]  Meir Feder,et al.  Multi-channel signal separation by decorrelation , 1993, IEEE Trans. Speech Audio Process..

[11]  Scott C. Douglas,et al.  Natural gradient multichannel blind deconvolution and speech separation using causal FIR filters , 2005, IEEE Trans. Speech Audio Process..