Multi-microphone speech dereverberation and noise reduction using relative early transfer functions

IEEE/ACM Transactions on Audio Speech and Language Processing

Volumn 23, Issue 2, 2015, Pages 240-251

  Schwartz, Ofer ^a   Gannot, Sharon ^a   Habets, Emanuel A P ^b  

^a BAR ILAN UNIVERSITY   (Israel)

^b UNIVERSITY OF ERLANGEN NUREMBERG   (Germany)

Author keywords

Dereverberation; generalized sidelobe canceller; minimum variance distortionless response (MVDR) beamforming; multichannel Wiener filter; relative transfer function

Indexed keywords

ACOUSTIC NOISE; ADAPTIVE SYSTEMS; ALGORITHMS; ANTENNA LOBES; ARCHITECTURAL ACOUSTICS; BEAMFORMING; ESTIMATION; MEAN SQUARE ERROR; MICROPHONES; QUALITY CONTROL; REVERBERATION; SIGNAL TO NOISE RATIO; SPEECH; SPEECH COMMUNICATION; TRANSFER FUNCTIONS;

DEREVERBERATION; GENERALIZED SIDELOBE CANCELLERS; MINIMUM MEAN-SQUARE ERROR ESTIMATORS; MINIMUM VARIANCE DISTORTIONLESS RESPONSE; MINIMUM VARIANCE DISTORTIONLESS RESPONSE BEAMFORMER; MULTICHANNEL WIENER FILTER; PERCEPTUAL EVALUATION OF SPEECH QUALITIES; SINGLE-CHANNEL ALGORITHMS;

SPEECH RECOGNITION;

EID: 84921681684     PISSN: 23299290     EISSN: None     Source Type: Journal    
DOI: 10.1109/TASLP.2014.2372335     Document Type: Article

References (56)

1. E. A. P. Habets, "Single-and multi-microphone speech dereverberation using spectral enhancement," Ph.D. dissertation, Technische Univ. Eindhoven, Eindhoven, The Netherlands, Jun. 2007.
2. M. Triki and D. T. Slock, "Iterated delay and predict equalization for blind speech dereverberation," in Proc. Int. Workshop Acoust. Echo Noise Control (IWAENC), 2006.
3. 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.
4. S. Subramaniam, A. P. Petropulu, and C. Wendt, "Cepstrum-based deconvolution for speech dereverberation," IEEE Trans. Speech Audio Process., vol. 4, no. 5, pp. 392-396, Sep. 1996.
5. 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.
6. S. Gannot and M. Moonen, "Subspace methods for multimicrophone speech dereverberation," EURASIP J. Adv. Signal Process., pp. 1074-1090, 2003.
7. M. Miyoshi and Y. Kenda, "Inverse filtering of room acoustics," IEEE Trans. Audio, Speech, Lang. Process., vol. 36, no. 2, pp. 145-152, Feb. 1988.
8. M. Kallinger and A. Mertins, "Multi-channel room impulse response shaping-a study," in Proc. IEEE Int. Conf. Acoust., Speech, Signal Process. (ICASSP), 2006, pp. 101-104.
9. A. Mertins, T. Mei, and M. Kallinger, "Room impulse response shortening/reshaping with infinity-and p-norm optimization," IEEE Trans. Audio, Speech, Lang. Process., vol. 18, no. 2, pp. 249-259, Feb. 2010.
10. W. Zhang, E. A. Habets, and P. A. Naylor, "On the use of channel shortening in multichannel acoustic system equalization," in Proc. Int. Workshop Acoust. Echo Noise Control (IWAENC), 2010.
11. I. Kodrasi and S. Doclo, "Robust partial multichannel equalization techniques for speech dereverberation," in Proc. IEEE Int. Conf. Acoust., Speech, Signal Process. (ICASSP), 2012, pp. 537-540.
12. J. D. Polack, "La transmission de l'énergie sonore dans les salles," Ph.D. dissertation, Univ. du Maine, Le Mans, France, 1988.
13. K. Lebart, J. Boucher, and P. Denbigh, "A new method based on spectral subtraction for speech dereverberation," Acta Acust. united with Acust., vol. 87, pp. 359-366, 2001.
14. E. A. P. Habets, "Single-channel speech dereverberation based on spectral subtraction," in Proc. Asilomar Conf. Signals, Syst., Comput., 2004, pp. 250-254.
15. E. A. P. Habets, "Multi-channel speech dereverberation based on a statistical model of late reverberation," in Proc. IEEE Int. Conf. Acoust., Speech, Signal Process. (ICASSP), Mar. 2005, vol. 4, pp. 173-176.
16. E. A. P. Habets, S. Gannot, and I. Cohen, "Dual-microphone speech dereverberation in a noisy environment," in Proc. IEEE Int. Symp. Signal Process. Inf. Technol., 2006, pp. 651-655.
17. H. W. Löllmann and P. Vary, "Low delay noise reduction and dereverberation for hearing aids," EURASIP J. Adv. Signal Process., vol. 2009, p. 1, 2009.
18. E. A. P. Habets, S. Gannot, and I. Cohen, "Late reverberant spectral variance estimation based on a statistical model," IEEE Signal Process. Lett., vol. 16, no. 9, pp. 770-773, Sep. 2009.
19. T. Yoshioka, T. Nakatani, and M. Miyoshi, "Integrated speech enhancement method using noise suppression and dereverberation," IEEE Trans. Audio, Speech, Lang. Process., vol. 17, no. 2, pp. 231-246, Feb. 2009.
20. M. Togami and Y. Kawaguchi, "Noise robust speech dereverberation with kalman smoother," in Proc. IEEE Int. Conf. Acoust., Speech, Signal Process. (ICASSP), 2013, pp. 7447-7451.
21. L. Griffiths and C. W. Jim, "An alternative approach to linearly constrained adaptive beamforming," IEEE Trans. Antennas Propag., vol. 30, no. 1, pp. 27-34, Jan. 1982.
22. S. Affes and Y. Grenier, "A signal subspace tracking algorithm for microphone array processing of speech," IEEE Trans. Speech Audio Process., vol. 5, no. 5, pp. 425-437, Sep. 1997.
23. S. Gannot, D. Burshtein, and E. Weinstein, "Signal enhancement using beamforming and nonstationarity with applications to speech," IEEE Trans. Signal Process., vol. 49, no. 8, pp. 1614-1626, Nov. 2001.
24. R. Talmon, I. Cohen, and S. Gannot, "Convolutive transfer function generalized sidelobe canceler," IEEE Trans. Audio, Speech, Lang. Process., vol. 17, no. 7, pp. 1420-1434, Sep. 2009.
25. R. Talmon, I. Cohen, and S. Gannot, "Multichannel speech enhancement using convolutive transfer function approximation in reverberant environments," in Proc. IEEE Int. Conf. Acoust., Speech, Signal Process. (ICASSP), Taipei, Taiwan, Apr. 2009, pp. 3885-3888.
26. E. A. P. Habets, "Single-and multi-microphone speech dereverberation using spectral enhancement," Ph.D. dissertation, Technische Univ. Eindhoven, The Netherlands, Jun. 2007.
27. E. A. P. Habets and S. Gannot, "Dual-microphone speech dereverberation using a reference signal," in Proc. IEEE Int. Conf. Acoust., Speech, Signal Process. (ICASSP), 2007, vol. 4, pp. 901-904.
28. E. A. P. Habets, "Towards multi-microphone speech dereverberation using spectral enhancement and statistical reverberation models," in Proc. Asilomar Conf. Signals, Syst., Comput., 2008, pp. 806-810.
29. B. Cauchi, I. Kodrasi, R. Rehr, S. Gerlach, A. Juki, T. Gerkmann, S. Doclo, and S. Goetze, "Joint dereverberation and noise reduction using beamforming and a single-channel speech enhancement scheme," in Reverb Challenge. IEEE Audio, Acoust., Signal Process. TC, Florence, Italy, 2014.
30. E. A. P. Habets and J. Benesty, "A two-stage beamforming approach for noise reduction and dereverberation," IEEE Trans. Audio, Speech, Lang. Process., vol. 21, no. 5, pp. 945-958, May 2013.
31. K. U. Simmer, J. Bitzer, and C. Marro, "Post-filtering techniques," in Microphone Arrays: Signal Processing Techniques and Applications. New York, NY, USA: Springer, 2001, vol. 3, pp. 39-60.
32. R. Balan and J. Rosca, "Microphone array speech enhancement by bayesian estimation of spectral amplitude and phase," IEEE Sens. Array Multichannel Signal Process. Workshop, pp. 209-213, 2002.
33. B. D. Van Veen and K. M. Buckley, "Beamforming: A versatile approach to spatial filtering," IEEE Acoust., Speech, Signal Process. Mag., vol. 5, no. 2, pp. 4-24, Apr. 1988.
34. H. Cox, R. Zeskind, and M. Owen, "Robust adaptive beamforming," IEEE Trans. Audio, Speech, Lang. Process., vol. ASSP-35, no. 10, pp. 1365-1376, Oct. 1987.
35. A. T. Parsons, "Maximum directivity proof for three-dimensional arrays," J. Acoust. Soc. Amer., vol. 82, p. 179, 1987.
36. J. Allen, D. A. Berkley, and J. Blauert, "Multimicrophone signal-processing technique to remove room reverberation from speech signals," J. Acoust. Soc. Amer., vol. 62, no. 4, pp. 912-915, 1977.
37. H. Kuttruff, Room acoustics. New York, NY, USA: Taylor and Francis, 2000.
38. K. Lebart, J.-M. Boucher, and P. Denbigh, "A new method based on spectral subtraction for speech dereverberation," Acta Acust. united with Acust., vol. 87, no. 3, pp. 359-366, 2001.
39. K. B. Petersen and M. S. Pedersen, "The matrix cookbook," Technical Univ. of Denmark, 2008, pp. 7-15.
40. E. A. P. Habets, J. Benesty, I. Cohen, S. Gannot, and J. Dmochowski, "New insights into the MVDR beamformer in room acoustics," IEEE Trans. Audio, Speech, Lang. Process., vol. 18, no. 1, pp. 158-170, Jan. 2010.
41. A. Spriet, M. Moonen, and J. Wouters, "Spatially pre-processed speech distortion weighted multi-channel Wiener filtering for noise reduction," Signal Process., vol. 84, no. 12, pp. 2367-2387, Dec. 2004.
42. L. J. Griffiths and C. W. Jim, "An alternative approach to linearly constrained adaptive beamforming," IEEE Trans. Antennas Propag., vol. AP-30, no. 1, pp. 27-34, Jan. 1982.
43. K. U. Simmer, J. Bitzer, and C. Marro, "Post-filtering techniques," in Microphone Arrays: Signal Processing Techniques and Applications, M. S. Brandstein and D. B. Ward, Eds. Berlin, Germany: Springer-Verlag, 2001, ch. 3, pp. 39-60.
44. E. A. P. Habets, S. Gannot, and I. Cohen, "Late reverberant spectral variance estimation based on a statistical model," IEEE Signal Process. Lett., vol. 16, no. 9, pp. 770-773, Sep. 2009.
45. E. A. P. Habets, S. Gannot, and I. Cohen, "Speech dereverberation using backward estimation of the late reverberant spectral variance," in Proc. IEEE Conv. Elect. Electron. Eng. Israel (IEEEI), 2008, pp. 384-388.
46. N. Dal Degan and C. Prati, "Acoustic noise analysis and speech enhancement techniques for mobile radio applications," Signal Process., vol. 15, no. 1, pp. 43-56, 1988.
47. E. A. P. Habets and S. Gannot, "Generating sensor signals in isotropic noise fields," J. Acoust. Soc. Amer., vol. 122, pp. 3464-3470, Dec. 2007.
48. E. A. P. Habets, "A distortionless subband beamformer for noise reduction in reverberant environments," in Proc. Int. Workshop Acoust. Echo Noise Control (IWAENC), Tel-Aviv, Israel, Aug. 2010.
49. M. Souden, J. Chen, J. Benesty, and S. Affes, "An integrated solution for online multichannel noise tracking and reduction," IEEE Trans. Audio, Speech, Lang. Process., vol. 19, no. 7, pp. 2159-2169, Sep. 2011.
50. M. Taseska and E. A. P. Habets, "MMSE-based blind source extraction in diffuse noise fields using a complex coherence-based a priori SAP estimator," in Proc. Int. Workshop Acoust. Signal Enhancement (IWAENC), Aachen, Germany, Sep. 2012.
51. R. Hendriks and T. Gerkmann, "Noise correlation matrix estimation for multi-microphone speech enhancement," IEEE Trans. Audio, Speech, Lang. Process., vol. 20, no. 1, pp. 223-233, Jan. 2012.
52. "ITU-T, Perceptual evaluation of speech quality (PESQ), an objective method for end-to-end speech quality assessment of narrowband telephone networks and speech codecs," Int. Telecomm. Union (ITU-T) Rec., Feb. 2001, p. 862.
53. D. Huggins-Daines, M. Kumar, A. Chan, A. W. Black, M. Ravishankar, and A. I. Rudnicky, "Pocketsphinx: A free, real-time continuous speech recognition system for hand-held devices," in Proc. IEEE Int. Conf. Acoust., Speech, Signal Process. (ICASSP), 2006, pp. 185-188.
54. D. B. Paul and J. M. Baker, "The design for the wall street journalbased CSR corpus," in Proc. Workshop Speech Natural Lang. Assoc. for Comput. Linguist., 1992, pp. 357-362.
55. J. B. Allen and D. A. Berkley, "Image method for efficiently simulating small-room acoustics," J. Acoust. Soc. Amer., vol. 65, no. 4, pp. 943-950, Apr. 1979.
56. P. M. Peterson, "Simulating the response of multiple microphones to a single acoustic source in a reverberant room," J. Acoust. Soc. Amer., vol. 76, no. 5, pp. 1527-1529, Nov. 1986.