Optimized speech dereverberation from probabilistic perspective for time varying acoustic transfer function

IEEE Transactions on Audio, Speech and Language Processing

Volumn 21, Issue 7, 2013, Pages 1369-1380

  Togami, Masahito ^a   Kawaguchi, Yohei ^a   Takeda, Ryu ^a   Obuchi, Yasunari ^a   Nukaga, Nobuo ^a  

^a HITACHI LTD   (Japan)

Author keywords

Dereverberation; expectation maximization algorithm; local Gaussian modeling; multichannel filtering; time varying acoustic transfer function

Indexed keywords

ACOUSTIC TRANSFER FUNCTIONS; DEREVERBERATION; EXPECTATION-MAXIMIZATION ALGORITHMS; LOCAL GAUSSIAN MODELING; MULTI-CHANNEL FILTERING;

ALGORITHMS; ITERATIVE METHODS; REVERBERATION;

OPTIMIZATION;

EID: 84875663456     PISSN: 15587916     EISSN: None     Source Type: Journal    
DOI: 10.1109/TASL.2013.2250960     Document Type: Article

References (34)

1. M. I. Gurelli and C. L. Nikias, "EVAM: An eigenvector-based algo rithm for multichannel blind deconvolution of input colored signals," IEEE Trans. Signal Process., vol. 43, no. 1, pp. 134-149, Jan. 1995.
2. S. Gannot and M. Moonen, "Subspace methods for multi microphone speech dereverberation," EURASIP J. Appl. Signal Process., vol. 2003, no. 11, pp. 1074-1090, Oct. 2003.
3. M. Miyoshi and Y. Kaneda, "Inverse filtering of room acoustics," IEEE Trans. Acoustic., Speech, Signal Process., vol. 36, no. 2, pp. 145-152, Feb. 1988.
4. Y. Huang, J. Benesty, and J. Chen, "A blind channel identification-based two-stage approach to separation and dereverberation of speech signals in a reverberant environment," IEEE Trans. Speech Audio Process., vol. 13, no. 5, pp. 882-895, Sep. 2005.
5. M. Delcroix, T. Hikichi, and M. Miyoshi, "Precise dereverberation using multi-channel linear prediction," IEEE Trans. Audio, Speech, Lang. Process., vol. 15, no. 2, pp. 430-440, Feb. 2007.
6. E. Moulines, P. Duhamel, J. F. Cardoso, and S. Mayrargue, "Subspace methods for the blind identification of multichannel FIR systems," IEEE Trans. Signal Process., vol. 43, no. 2, pp. 516-526, Feb. 1995.
7. E. A. P. Habets, "Speech Dereverberation using statistical reverberation models," in Speech Dereverberation, P. A. Naylor and N. D. Gaubitch, Eds. New York, NY, USA: Springer, 2010.
8. T. Hikichi, M. Delcroix, and M. Miyoshi, "Inverse filtering for speech dereverberation less sensitive to noise and room transfer function fluctuations," EURASIP J. Adv. Signal Process., vol. 2007, no. 1, p. 62, 2007.
9. S. Bharitkar, P. Hilmes, and C. Kyriakakis, "Robustness of spatial average equalization: A statistical reverberation model approach," J. Acoust. Soc. Amer., vol. 116, no. 6, pp. 3491-3497, Dec. 2004. (Pubitemid 40029930)
10. H. Hacihabiboglu and Z. Cvetkovic, "Multichannel dereverberation theorems and robustness issues," IEEE Trans. Speech Audio Process., vol. 20, no. 2, pp. 676-689, Feb. 2012.
11. K. Lebart and J. M. Boucher, "A new method based on spectral subtrac tion for speech dereverberation," Acta Acoustica, vol. 87, pp. 359-366, 2001. (Pubitemid 32699291)
12. J. Even, H. Saruwatari, K. Shikano, and T. Takatani, "Blind signal extraction based joint suppression of diffuse background noise and late reverberation," in Proc. Eur. Signal Process. Conf., 2009, pp. 1534-1538.
13. N. D. Gaubitch, M. R. P. Thomas, and P. A. Naylor, "Dereverbera-tion using LPC-based approaches," in Speech Dereverberation, P. A. Naylor and N. D. Gaubitch, Eds. New York, NY, USA: Springer, 2010.
14. D. H. Johnson and D. E. Dudgeon, Array Signal Processing-Concepts and Techniques. Englewood Cliffs, NJ, USA: Prentice-Hall, 1993.
15. J. Benesty, J. Chen, Y. Huang, and J. Dmochowski, "On microphone-array beamforming from a mimo acoustic signal processing perspective," IEEE Trans. Speech Audio Process., vol. 15, no. 3, pp. 1053-1065, Mar. 2007.
16. J. Capon, "High resolution frequency-wavenumber spectrum analysis," Proc. IEEE, vol. 57, no. 8, pp. 1408-1418, Aug. 1969.
17. K. Furuya and A. Kataoka, "Robust speech dereverberation using multichannel blind deconvolution with spectral subtraction," IEEE Trans. Speech Audio Process., vol. 15, no. 5, pp. 1579-1591, Jul. 2007.
18. K. Kinoshita, M. Delcroix, T. Nakatani, and M. Miyoshi, "Suppression of late reverberation effect on speech signal using long-term multiple-step linear prediction," IEEE Trans. Speech Audio Process., vol. 17, no. 4, pp. 534-545, May 2009.
19. N. Q. K. Duong, E. Vincent, and R. Gribonval, "Under-determined reverberant audio source separation using a full-rank spatial covari-ance model," IEEE Trans. Speech Audio Process., vol. 18, no. 7, pp. 1830-1840, Sep. 2010.
20. M. Togami, Y. Kawaguchi, R. Takeda, Y. Obuchi, and N. Nukaga, "Multichannel speech dereverberation and separation with optimized combination of linear and non-linear filtering," in Proc. Int. Conf. Acoust. Speech, Signal Process., 2012, pp. 4057-4060.
21. M. Togami and K. Hori, "Multichannel semi-blind source separation via local Gaussian modeling for acoustic echo reduction," in Proc. Eur. Signal Process. Conf., 2011, pp. 496-500.
22. T. Nakatani, T. Yoshioka, K. Kinoshita, M. Miyoshi, and B.-H. Juang, "Speech dereverberation based on variance-normalized delayed linear prediction," IEEE Trans. Speech Audio Process., vol. 17, no. 7, pp. 1717-1731, Sep. 2010.
23. T. Yoshioka, T. Nakatani, M. Miyoshi, and H. G. Okuno, "Blind separation and dereverberation of speech mixtures by joint optimization," IEEE Trans. Speech Audio Process., vol. 19, no. 1, pp. 69-84, Jan. 2011.
24. T. Nakatani, T. Yoshioka, K. Kinoshita, M. Miyoshi, and B.-H. Juang, "Blind speech dereverberation with multi-channel linear prediction based on short time Fourier transform representation," in Proc. Int. Conf. Acoust. Speech, Signal Process., 2008, pp. 85-88.
25. M. Miyoshi, M. Delcroix, K. Kinoshita, T. Yoshioka, T. Nakatani, and T. Hikichi, "Inverse filtering for speech dereverberation without the use of room acoustics information," in Speech Dereverberation. New York, NY, USA: Springer, 2010, pp. 271-310.
26. A. P. Dempster, N. M. Laird, and B. D. Rubin, "Maximum likelihood from incomplete data via the EM algorithm," J. R. Statist. Soc., ser. B, vol. 39, pp. 1-38, 1977.
27. D. A. Harville, Matrix Algebra From a Statistician's Perspective. New York, NY, USA: Springer-Verlag, 1997.
28. N. Murata, S. Ikeda, and A. Ziehe, "An approach to blind source separation based on temporal structure of speech signals," Neurocomput., vol. 41, no. 1-4, pp. 1-24, Oct. 2001.
29. Y. Suzuki, F. Asano, H. Y. Kim, and T. Sone, "An optimum computergenerated pulse signal suitable for the measurement of very long impulse responses," J. Acoust. Soc. Amer., vol. 97, no. 2, pp. 1119-1123, Feb. 1995.
30. TIMIT Corpus, [Online]. Available: http://www.ldc.upenn.edu/Cat-alog/ CatalogEntry.jsp?catalogId=LDC93S1
31. "Perceptual evaluation of speech quality: An objective method for end-to-end speech quality assessment of narrow-band telephone networks and speech coders," ITU-T P.862, 2001.
32. K. Kokkinakis and P. C. Loizou, "Evaluation of objective measures for quality assessment of reverberant speech," in Proc. Int. Conf. Acoust. Speech, Signal Process., 2011, pp. 2420-2423.
33. T. Yoshioka, H. Tachibana, T. Nakatani, and M. Miyoshi, "Adaptive dereverberation of speech signals with speaker-position change detection," in Proc. Int. Conf. Acoust. Speech, Signal Process., 2009, pp. 3733-3736.
34. K. Kumatani, J. McDough, and B. Raj, "Block-wise incremental adaptation algorithm for maximum kurtosis beamforming," in Proc. Workshop Applicat. Signal Process. Audio Acoust., 2011, pp. 229-232.