Integrated speech enhancement method using noise suppression and dereverberation

IEEE Transactions on Audio, Speech and Language Processing

Volumn 17, Issue 2, 2009, Pages 231-246

  Yoshioka, Takuya ^a   Nakatani, Tomohiro ^a   Miyoshi, Masato ^a  

^a NTT Communication Science Laboratories   (Japan)

Author keywords

Dereverberation; Maximum likelihood (ML) estimation; Minimum mean square error (MMSE) estimation; Noise suppression; Speech enhancement

Indexed keywords

AUTOREGRESSIVE SYSTEMS; DEREVERBERATION; ESTIMATED PARAMETER; GAUSSIAN RANDOM VARIABLE; MAXIMUM LIKELIHOOD ESTIMATION METHOD; MAXIMUM-LIKELIHOOD (ML) ESTIMATION; MINIMUM MEAN SQUARE ERROR (MMSE) ESTIMATION; MINIMUM MEAN SQUARE ERROR ESTIMATE; NOISE SUPPRESSION; REVERBERATION TIME; SPECTRAL COMPONENTS; SPEECH ENHANCEMENT METHODS; SPEECH MODELS; SPEECH SIGNALS; SPEECH SPECTRA; STATIONARY NOISE; TIME INVARIANTS;

ARCHITECTURAL ACOUSTICS; BLOCK CODES; FREQUENCY BANDS; MEAN SQUARE ERROR; PARAMETER ESTIMATION; POLES; POWER SPECTRAL DENSITY; POWER SPECTRUM; PROBABILITY DISTRIBUTIONS; RANDOM PROCESSES; RANDOM VARIABLES; REVERBERATION; SIGNAL TO NOISE RATIO; SPEECH ENHANCEMENT;

MAXIMUM LIKELIHOOD ESTIMATION;

EID: 70350435249     PISSN: 15587916     EISSN: None     Source Type: Journal    
DOI: 10.1109/TASL.2008.2008042     Document Type: Article

References (37)

1. T. Quatieri, Discrete-Time Speech Signal Processing. Upper Saddle River, NJ: Prentice-Hall, 2002.
2. J. S. Lim and A. V. Oppenheim, "All-pole modeling of degraded speech, " IEEE Trans. Acoust. Speech, Signal Process., vol. ASSP-26, no. 3, pp. 197-210, Jun. 1978.
3. S. Gannot, D. Burshtein, and E. Weinstein, "Iterative and sequential Kalman filter-based speech enhancement algorithms, " IEEE Trans. Speech Audio Process., vol. 6, no. 4, pp. 373-385, Jul. 1998.
4. S. Srinivasan, J. Samuelsson, and W. B. Kleijn, "Codebook-based Bayesian speech enhancement for nonstationary environments, " IEEE Trans. Audio, Speech, Lang. Process., vol. 15, no. 2, pp. 441-452, Feb. 2007.
5. M. I. Gurelli and C. L. Nikias, "EVAM: An eigenvector-based algorithm for multichannel blind deconvolution of input colored signals, " IEEE Trans. Signal Process., vol. 43, no. 1, pp. 134-149, Jan. 1995.
6. S. Gannot and M. Moonen, "Subspace methods for multimicrophone speech dereverberation, " EURASIP J. Appl. Signal Process., vol. 2003, no. 11, pp. 1074-1090, 2003.
7. 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, 2007, 10.1155/2007/34013, article ID 34013..
8. S. Javidi, N. D. Gaubitch, and P. A. Naylor, "An experimental study of the eigendecomposition methods for blind SIMO system identification in the presence of noise, " in Proc. Int. Worksh. Acoust. Echo, Noise Contr., 2006, CD-ROM Proc..
9. 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.
10. B. W. Gillespie, H. S. Malvar, and D. A. F. Florêncio, "Speech dereverberation via maximum-kurtosis subband adaptive filtering, " in Proc. Int. Conf. Acoust., Speech, Signal Process., 2001, vol. VI, pp. 3701-3704.
11. N. D. Gaubitch, P. A. Naylor, and D. B. Ward, "On the use of linear prediction for dereverberation of speech, " in Proc. Int. Worksh. Acoust. Echo, Noise Contr., 2003, pp. 99-102.
12. T. Nakatani, K. Kinoshita, and M. Miyoshi, "Harmonicity-based dereverberation for single-channel speech signals, " IEEE Trans. Audio, Speech, Lang. Process., vol. 15, no. 1, pp. 80-95, Jan. 2007.
13. K. Kinoshita, T. Nakatani, and M. Miyoshi, "Spectral subtraction steered by multi-step forward linear prediction for single channel speech dereverberation, " in Proc. Int. Conf. Acoust., Speech, Signal Process., 2006, vol. I, pp. 817-820.
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, 2003.
15. M. Delcroix, T. Hikichi, and M. Miyoshi, "Precise dereverberation using multichannel linear prediction, " IEEE Trans. Audio, Speech, Lang. Process., vol. 15, no. 2, pp. 430-440, Feb. 2007.
16. T. Yoshioka, T. Hikichi, and M. Miyoshi, "Dereverberation by using time-variant nature of speech production system, " EURASIP J. Adv. Signal Process., vol. 2007, 2007, 10.1155/2007/65698, article ID 65698.
17. T. Yoshioka, T. Nakatani, T. Hikichi, and M. Miyoshi, "Overfittingresistant speech dereverberation, " in Proc. IEEE Worksh. Appl. Signal Process. Audio, Acoust., 2007, pp. 163-166.
18. T. Nakatani, B.-H. Juang, T. Hikichi, T. Yoshioka, K. Kinoshita, M. Delcroix, and M. Miyoshi, "Study on speech dereverberation with autocorrelation codebook, " in Proc. Int. Conf. Acoust. Speech, Signal Process., 2007, vol. I, pp. 193-196.
19. T. Nakatani, B. H. Juang, T. Yoshioka, K. Kinoshita, and M. Miyoshi, "Importance of energy and spectral features in Gaussian source model for speech dereverberation, " in Proc. IEEE Workshop Appl. Signal Process. Audio, Acoust., 2007, pp. 299-302.
20. 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.
21. H. Attias, J. C. Platt, A. Acero, and L. Deng, "Speech denoising and dereverberation using probabilistic models, " Adv. Neural Inf. Process. Syst., vol. 13, pp. 758-764, 2000.
22. K. Kinoshita, T. Nakatani, M. Delcroix, and M. Miyoshi, "Multi-step linear prediction based speech dereverberation in noisy reverberant environment, " in Proc. Interspeech, 2007, pp. 854-857.
23. M. Delcroix, T. Hikichi, and M. Miyoshi, "Dereverberation and denoising using multichannel linear prediction, " IEEE Trans. Audio, Speech, Lang. Process., vol. 15, no. 6, pp. 1791-1801, Aug. 2007.
24. S. F. Boll, "Suppression of acoustic noise in speech using spectral subtraction, " IEEE Trans. Acoust., Speech, Signal Process., vol. ASSP-27, no. 2, pp. 113-120, Apr. 1979.
25. R. Martin, "Noise power spectral density estimation based on optimal smoothing and minimum statistics, " IEEE Trans. Speech Audio Process., vol. 9, no. 5, pp. 504-512, Jul. 2001.
26. 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.
27. H. Wang and F. Itakura, "Dereverberation of speech signals based on sub-band envelope estimation, " IEICE Trans. Fund., vol. E74-A, no. 11, pp. 3576-3583, 1991.
28. Y. Avargel and I. Cohen, "System identification in the short-time Fourier transform domain with crossband filtering, " IEEE Trans. Audio, Speech, Lang. Process., vol. 15, no. 4, pp. 1305-1319, May 2007.
29. B. W. Gillespie and L. E. Atlas, "Acoustic diversity for improved speech recognition in reverberant environments, " in Proc. Int. Conf. Acoust., Speech, Signal Process., 2002, vol. I, pp. 557-560.
30. T. Nakatani, T. Yoshioka, K. Kinoshita, M. Miyoshi, and B.-H. Juang, "Speech dereverberation in short time Fourier transform domain with crossband effect compensation, " in Hands-Free Speech Commun, Mic. Arrays, 2008, pp. 220-223.
31. L. R. Rabiner and R. W. Schafer, Digital Processing of Speech Signals. Englewood Cliffs, NJ: Prentice-Hall, 1983.
32. A. van den Bos, "The multivariate complex normal distribution-A generalization, " IEEE Trans. Inf. Theory, vol. 41, no. 2, pp. 537-539, Mar. 1995.
33. Y. Ephraim, "Speech enhancement using a minimum mean square error short-time spectral amplitude estimator, " IEEE Trans. Acoust. Speech, Signal Process., vol. ASSP-32, no. 6, pp. 1109-1121, Dec. 1984.
34. S. Haykin, Adaptive Filter Theory, 3rd ed. Englewood Cliffs, NJ: Prentice-Hall, 1995.
35. X. L. Meng and D. B. Rubin, "Maximum likelihood estimation via the ECM algorithm: A general framework, " Biometrika, vol. 80, no. 2, pp. 267-278, 1993.
36. M. R. Portnoff, "Implementation of the digital phase vocoder using the fast Fourier transform, " IEEE Trans. Acoust. Speech, Signal Process., vol. ASSP-24, no. 3, pp. 243-248, Jun. 1976.
37. ASJ Continuous Speech Corpus, Acoustical Society of Japan [Online]. Available: http://www.milab.is.tsukuba.ac.jp/jnas/instruct.html.