Harmonicity-based blind dereverberation for single-channel speech signals

IEEE Transactions on Audio, Speech and Language Processing

Volumn 15, Issue 1, 2007, Pages 80-95

  Nakatani, Tomohiro ^a   Kinoshita, Keisuke ^a   Miyoshi, Masato ^a  

^a NTT Communication Science Laboratories   (Japan)

Author keywords

Adaptive harmonic filter; Blind signal processing; Dereverberation; Harmonicity; Inverse filter; Speech signal; Time warping

Indexed keywords

ADAPTIVE HARMONIC FILTER; BLIND SIGNAL PROCESSING; DEREVERBERATION; HARMONICITY; INVERSE FILTER; SPEECH SIGNAL; TIME WARPING;

ACOUSTIC WAVES; ARCHITECTURAL ACOUSTICS; IMPULSE RESPONSE; PROCESSING; REVERBERATION; SPEECH ANALYSIS; SPEECH RECOGNITION; WEAVING;

SIGNAL PROCESSING;

EID: 34548571735     PISSN: 15587916     EISSN: None     Source Type: Journal    
DOI: 10.1109/TASL.2006.872620     Document Type: Article

References (26)

1. T. Houtgast, H. J. M. Steeneken, and R. Plomp, "Predicting speech intelligibility in rooms from the modulation transfer function I. General room acoustics," Acustica, vol. 46, pp. 60-71, 1980.
2. K. L. Payton, R. M. Uchanski, and L. D. Braida, "Intelligibility of conversational and clear speech in noise and reverberation for listeners with normal and impaired hearing," J. Acoust. Soc. Amer., vol. 95, no. 3, pp. 1581-1592, Mar. 1994.
3. B. S. Atal, "Effectiveness of linear prediction characteristics of the speech wave for automatic speaker identification and verification," J. Acoust. Soc. Amer., vol. 55, no. 6, pp. 1304-1312, 1974.
4. C. J. Leggetter and P. C. Woodland, "Maximum-likelihood linear regression for speaker adaptation of continuous density hidden Markov models," Comput. Speech Lang., vol. 9, pp. 171-185, 1995.
5. B. Kingsbury and N. Morgan, "Recognizing reverberant speech with rasta-plp," in Proc. IEEE Int. Conf. Acoust., Speech, Signal Process. (ICASSP), 1997, vol. 2, pp. 1259-1262.
6. M. Miyoshi and Y. Kaneda, "Inverse filtering of room acoustics," IEEE Trans. Acoust., Speech, Signal Process., vol. 36, no. 2, pp. 145-152, Feb. 1988.
7. A. V. Oppenheim and R. W. Schafer, Discrete-Time Signal Processing, 2nd ed. Englewood Cliffs, NJ: Prentice-Hall, 1999, ch. 5.
8. S. C. Douglas and X. Sun, "Convolutive blind separation of speech mixtures using the natural gradient," Speech Commun., vol. 39, pp. 65-78, 2003.
9. B. W. Gillespie and L. E. Atlas, "Strategies for improving audible quality and speech recognition accuracy of reverberant speech," in Proc. IEEE Int. Conf. Acoust., Speech, Signal Process. (ICASSP), 2003, vol. 1, pp. 676-679.
10. K. Furuya, "Noise reduction and dereverberation using correlation matrix based on the multiple-input/output inverse-filtering theorem (mint)," in Proc. Int. Workshop Hands-Free Speech Commun., 2001, pp. 59-62.
11. H. Buchner, R. Aichner, and W. Kellermann, "TRINICON: A versatile framework for multichannel blind signal processing," in Proc. IEEE Int. Conf. Acoust., Speech, Signal Process. (ICASSP), May 2004, vol. 3, pp. 889-892.
12. B. Yegnanarayana and P. S. Murthy, "Enhancement of reverberant speech using LP residual signal," IEEE Trans. Speech Audio Process., vol. 8, no. 3, pp. 267-281, May 2000.
13. M. Unoki, M. Furukawa, K. Sakata, and M. Akagi, "A method based on the MTF concept for dereverberating the power envelope from the reverberant signal," in Proc. IEEE Int. Conf. Acoust., Speech, Signal Process. (ICASSP), 2003, vol. 1, pp. 840-843.
14. J. R. Hopgood and P. J. W. Rayner, "Blind single channel deconvolution using nonstationary signal processing," IEEE Trans. Speech Audio Process., vol. 11, no. 5, pp. 476-488, Sep. 2003.
15. T. Houtgast and H. J. M. Steenken, "A review of the MTF concept in room acoustics and its use for estimating speech intelligibility in audioria," J. Acoust. Soc. Amer., vol. 77, no. 3, pp. 1069-1077, Mar. 1985.
16. A. S. Bregman, Auditory Scene Analysis-The Perceptual Organization of Sound. Cambridge, MA: MIT Press, 1990.
17. M. Weintraub, "A computational model for separating two simultaneous talkers," in Proc. IEEE Int. Conf. Acoust., Speech, Signal Process. (ICASSP), 1986, vol. 11, pp. 81-84.
18. T. Nakatani, "Computational Auditory Scene Analysis Based on Residue-Driven Architecture and Its Application to Mixed Speech Recognition," Ph.D. dissertation, Dept. Appl. Anal. Complex Dynamical Syst., Kyoto Univ., Kyoto, Japan, Mar. 2002.
19. T. Abe and M. Honda, "Sinusoidal modeling based on instantaneous frequency attractors," in Proc. IEEE Int. Conf. Acoust., Speech, Signal Process. (ICASSP), 2003.
20. R. J. McAulay and T. F. Quatieri, "Speech analysis/synthesis based on a sinusoidal representation," IEEE Trans. Acoust., Speech, Signal Process., vol. ASSP-34, no. 4, pp. 744-754, Apr. 1986.
21. L. Cohen, Time-Frequency Analysis. Englewood Cliffs, NJ: Prentice- Hall, 1995.
22. T. Nakatani and M. Miyoshi, "Blind dereverberation of single channel speech signal based on harmonic structure," in Proc. IEEE Int. Conf. Acoust., Speech, Signal Process. (ICASSP), 2003, vol. 1, pp. 92-95.
23. T. Nakatani and T. Irino, "Robust and accurate fundamental frequency estimation based on dominant harmonic components," J. Acoust. Soc. Amer., vol. 116, no. 6, pp. 3690-3700, Dec. 2004.
24. M. R. Schroeder, "New method of measuring reverberation time," J. Acoust. Soc. Amer., vol. 37, pp. 409-412, 1965.
25. B. Yegnanarayana and B. S. Ramakrishna, "Intelligibility of speech under nonexponential decay conditions," J. Acoust. Soc. Amer., vol. 58, pp. 853-857, Oct. 1975.
26. K. Kinoshita, T. Nakatani, and M. Miyoshi, "Fast estimation of a precise dereverberation filter based on speech harmonicity," in Proc. IEEE Int. Conf. Acoust., Speech, Signal Process. (ICASSP), Mar. 2005, pp. 1073-1076.