Dereverberation by using time-variant nature of speech production system

Eurasip Journal on Advances in Signal Processing

Volumn 2007, Issue , 2007, Pages

  Yoshioka, Takuya ^a   Hikichi, Takafumi ^a   Miyoshi, Masato ^a  

^a NTT Communication Science Laboratories   (Japan)

Author keywords

[No Author keywords available]

Indexed keywords

ALGORITHMS; COMPUTER SIMULATION; PROBLEM SOLVING; REVERBERATION; SIGNAL FILTERING AND PREDICTION;

DEREVERBERATION; INVERSE FILTERING; ROOM ACOUSTIC SYSTEM; SPEECH SIGNAL; TIME-VARIANT NATURE;

SPEECH PROCESSING;

EID: 34548555989     PISSN: 16876172     EISSN: 16876180     Source Type: Journal    
DOI: 10.1155/2007/65698     Document Type: Article

References (28)

1. L. R. Rabiner R. W. Schafer Digital Processing of Speech Signals. Prentice-Hall Upper Saddle River, NJ, USA 1983
2. M. I. Gurelli C. L. Nikias EVAM: an eigenvector-based algorithm for multichannel blind deconvolution of input colored signals. IEEE Transactions on Signal Processing 43 1 1995 134 149
3. K. Furuya Y. Kaneda Two-channel blind deconvolution of nonminimum phase FIR systems. IEICE Transactions on Fundamentals of Electronics, Communications and Computer Sciences E80-A 5 1997 804 808
4. S. Gannot gannot@siglab.technion.ac.il M. Moonen Marc.Moonen@esat. kuleuven.ac.be Subspace methods for multimicrophone speech dereverberation. EURASIP Journal on Applied Signal Processing 2003 11 2003 1074 1090
5. T. Hikichi hikichi@cslab.kecl.ntt.co.jp M. Delcroix marc.delcroix@cslab. kecl.ntt.co.jp M. Miyoshi miyo@cslab.kecl.ntt.co.jp Blind dereverberation based on estimates of signal transmission channels without precise information on channel order. IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '05) Philadelphia, Pa, USA 1 2005 1069 1072
6. M. Delcroix T. Hikichi M. Miyoshi Precise dereverberation using multichannel linear prediction. IEEE Transactions Audio, Speech and Language Processing 15 2007 2 430 440
7. B. Yegnanarayana yegna@iitm.ernet.in P. S. Murthy Enhancement of reverberant speech using LP residual signal. IEEE Transactions on Speech and Audio Processing 8 3 2000 267 281
8. B. W. Gillespie H. S. Malvar D. A. F. Florêncio Speech dereverberation via maximum-kurtosis subband adaptive filtering. IEEE Interntional Conference on Acoustics, Speech, and Signal Processing (ICASSP '01) Salt Lake, Utah, USA 6 2001 3701 3704
9. B. W. Gillespie L. E. Atlas Strategies for improving audible quality and speech recognition accuracy of reverberant speech. IEEE International Conference on Accoustics, Speech, and Signal Processing (ICASSP '03) Hong Kong 1 2003 676 679
10. N. D. Gaubitch P. A. Naylor D. B. Ward On the use of linear prediction for dereverberation of speech. Proceedings of International Workshop on Acoustic Echo and Noise Control (IWAENC '03) Kyotp, Japan 2003 99 102
11. T. Nakatani K. Kinoshita M. Miyoshi Harmonicity-based blind dereverberation for single-channel speech signals. IEEE Transactions, Audio, Speech and Language Processing 15 2007 1 80 95
12. K. Kinoshita kinoshita@cslab.kecl.ntt.co.jp T. Nakatani nak@cslab.kecl.ntt.co.jp M. Miyoshi miyo@cslab.kecl.ntt.co.jp Efficient blind dereverberation framework for automatic speech recognition. Proceedings of the 9th European Conference on Speech Communication and Technology Lisbon, Portugal 2005 3145 3148
13. P. S. Spencer P. J. W. Rayner Separation of stationary and time-varying systems and its application to the restoration of gramophone recordings. IEEE International Symposium on Circuits and Systems (ISCAS '89) Portland, Ore, USA 1 1989 292 295
14. J. R. Hopgood jrh1008@eng.cam.ac.uk P. J. W. Rayner pjwr@eng.cam.ac.uk Blind single channel deconvolution using nonstationary signal processing. IEEE Transactions on Speech and Audio Processing 11 5 2003 476 488
15. O. Shalvi E. Weinstein New criteria for blind deconvolution of nonminimum phase systems(channels). IEEE Transactions on Information Theory 36 1990 2 312 321
16. K. Abed-Meraim E. Moulines P. Loubaton Prediction error method for second-order blind identification. IEEE Transactions on Signal Processing 45 3 1997 694 705
17. B. Theobald S. Cox G. Cawley B. Milner Fast method of channel equalisation for speech signals and its implementation on a DSP. Electronics Letters 35 16 1999 1309 1311
18. D.-T. Pham J.-F. Cardoso Blind separation of instantaneous mixtures of nonstationary sources. IEEE Transactions on Signal Processing 49 9 2001 1837 1848
19. K. Matsuoka M. Ohya M. Kawamoto A neural net for blind separation of nonstationary signals. Neural Networks 8 3 1995 411 419
20. Acoustical Society of Japan ASJ Continuous Speech Corpus. http://www.mibel.cs.tsukuba.ac.jp/jnas/instruct.html
21. H. Kuttruff Room Acoustics. Elsevier Applied Science London, UK 1991
22. W. B. Kleijn K. K. Paliwal Speech Coding and Synthesis. Elsevier Science Amsterdam, The Netherlands 1995
23. A. Gorokhov gorokhov@lss.supelec.fr P. Loubaton Blind identification of MIMO-FIR systems: a generalized linear prediction approach. Signal Processing 73 1-2 1999 105 124
24. J. Jacod A. N. Shiryaev Limit Theorems for Stochastic Processes. Springer New York, NY, USA 1987
25. P. Comon Independent component analysis, a new concept? Signal Processing 36 3 1994 287 314
26. A. Hyvärinen J. Karhumen E. Oja Independent Component Analysis. John Wiley & Sons New York, NY, USA 2001
27. T. Yoshioka T. Hikichi M. Miyoshi H. G. Okuno Robust decomposition of inverse filter of channel and prediction error filter of speech signal for dereverberation. Proceedings of the 14th European Signal Processing Conference (EUSIPCO '06) Florence, Italy 2006
28. S. F. Boll Suppression of acoustic noise in speech using spectral subtraction. IEEE Trans Acoust Speech Signal Process 27 2 1979 113 120