Location via proxy:   [ UP ]  
[Report a bug]   [Manage cookies]                
Skip to main content

TRINICON-based Blind System Identification with Application to Multiple-Source Localization and Separation

  • Chapter
Blind Speech Separation

This contribution treats blind system identification approaches and how they can be used to localize multiple sources in environments where multipath propagation cannot be neglected, e.g., acoustic sources in reverberant environments. Based on TRINICON, a general framework for broadband adaptive MIMO signal processing, we first derive a versatile blind MIMO system identification method. For this purpose, the basics of TRINICON will be reviewed to the extent needed for this application, and some new algorithmic aspects will be emphasized. The generic approach then allows us to study various illustrative relations to other algorithms and applications. In particular, it is shown that the optimization criteria used for blind system identification allow a generalization of the well-known Adaptive Eigenvalue Decomposition (AED) algorithm for source localization: Instead of one source as with AED, several sources can be localized simultaneously. Performance evaluation in realistic scenarios will show that this method compares favourably with other state-of-the-art methods for source localization.

This is a preview of subscription content, log in via an institution to check access.

Access this chapter

Subscribe and save

Springer+ Basic
$34.99 /Month
  • Get 10 units per month
  • Download Article/Chapter or eBook
  • 1 Unit = 1 Article or 1 Chapter
  • Cancel anytime
Subscribe now

Buy Now

Chapter
USD 29.95
Price excludes VAT (USA)
  • Available as PDF
  • Read on any device
  • Instant download
  • Own it forever
eBook
USD 129.00
Price excludes VAT (USA)
  • Available as PDF
  • Read on any device
  • Instant download
  • Own it forever
Softcover Book
USD 169.99
Price excludes VAT (USA)
  • Compact, lightweight edition
  • Dispatched in 3 to 5 business days
  • Free shipping worldwide - see info
Hardcover Book
USD 169.99
Price excludes VAT (USA)
  • Durable hardcover edition
  • Dispatched in 3 to 5 business days
  • Free shipping worldwide - see info

Tax calculation will be finalised at checkout

Purchases are for personal use only

Institutional subscriptions

Preview

Unable to display preview. Download preview PDF.

Unable to display preview. Download preview PDF.

Similar content being viewed by others

References

  1. H. Buchner, R. Aichner, and W. Kellermann, “Blind source separation for convolutive mixtures exploiting nongaussianity, nonwhiteness, and nonstation-arity,” in Proc. Int. Workshop Acoustic Echo and Noise Control (IWAENC), Kyoto, Japan, pp. 223-226, Sept. 2003.

    Google Scholar 

  2. H. Buchner, R. Aichner, and W. Kellermann, “Blind source separation for convolutive mixtures: A unified treatment,” in Y. Huang and J. Benesty (eds.), Audio Signal Processing for Next-Generation Multimedia Communica-tion Systems, Kluwer Academic Publishers, Boston, pp. 255-293, Feb. 2004.

    Google Scholar 

  3. H. Buchner, R. Aichner, and W. Kellermann, “TRINICON: A versatile frame-work for multichannel blind signal processing,” in Proc. IEEE Int. Conf. Acoustics, Speech, Signal Processing (ICASSP), Montreal, Canada, vol. 3, pp. 889-892, May 2004.

    Google Scholar 

  4. H. Buchner, R. Aichner, and W. Kellermann, “A generalization of blind source separation algorithms for convolutive mixtures based on second-order statis-tics,” IEEE Trans. Speech Audio Process., vol. 13, no. 1, pp. 120-134, Jan. 2005.

    Article  Google Scholar 

  5. H. Buchner, R. Aichner, and W. Kellermann, “Relation between blind system identification and convolutive blind source separation,” in Proc. Joint Work-shop Hands-Free Speech Communication and Microphone Arrays (HSCMA), Piscataway, NJ, USA, Mar. 2005 (additional presentation slides with more results downloadable from the web site www.LNT.de/lms/).

  6. H. Buchner, R. Aichner, J. Stenglein, H. Teutsch, and W. Kellermann, “Simultaneous localization of multiple sound sources using blind adaptive MIMO filtering,” in Proc. IEEE Int. Conf. Acoustics, Speech, Signal Processing (ICASSP), Philadelphia, PA, USA, Mar. 2005.

    Google Scholar 

  7. M. Hofbauer, Optimal Linear Separation and Deconvolution of Acoustical Con-volutive Mixtures, Dissertation, Hartung-Gorre Verlag, Konstanz, May 2005.

    Google Scholar 

  8. C.H. Knapp and G.C. Carter, “The generalized correlation method for esti-mation of time delay,” IEEE Trans. Acoust., Speech, Signal Processing, vol. ASSP-24, pp. 320-327, Aug. 1976.

    Article  Google Scholar 

  9. M.S. Brandstein and D.B. Ward, Microphone Arrays: Signal Processing Tech-niques and Applications, Springer, Berlin, 2001.

    Google Scholar 

  10. R.O. Schmidt, “Multiple emitter location and signal parameter estimation,” IEEE Trans. Antennas Propagation, vol. AP-34, no. 3, pp. 276-280, Mar. 1986.

    Article  Google Scholar 

  11. R. Roy and T. Kailath, “ESPRIT - estimation of signal parameters via rota-tional invariance techniques,” IEEE Trans. Acoust., Speech, Signal Processing, vol. 37. no. 7, pp. 984-995, July 1989.

    Article  Google Scholar 

  12. B. Champagne, S. Bedard, and A. Stéphenne, “Performance of time-delay estimation in the presence of room reverberation,” IEEE Trans. Speech Audio Process., vol. 4, pp. 148-152, Mar. 1996.

    Article  Google Scholar 

  13. J.P. Ianniello, “Time delay estimation via cross-correlation in the presence of large estimation errors,” IEEE Trans. Acoust., Speech, Signal Processing, vol. ASSP-30, no. 6, pp. 998-1003, Dec. 1982.

    Google Scholar 

  14. J. Scheuing and B. Yang, “Disambiguation of TDOA estimates in multi-path multi-source environments (DATEMM),” in Proc. IEEE Int. Conf. Acoust., Speech, Signal Processing (ICASSP), Toulouse, France, 2006.

    Google Scholar 

  15. J. Benesty, “Adaptive eigenvalue decomposition algorithm for passive acoustic source localization,” J. Acoust. Soc. Am., vol. 107, pp. 384-391, Jan. 2000.

    Article  Google Scholar 

  16. J. Chen, Y. Huang, and J. Benesty, “Time delay estimation” in Y. Huang and J. Benesty (eds.), Audio Signal Processing for Next-Generation Multimedia Communication Systems, Kluwer Academic Publishers, Boston, pp. 197-227, Feb. 2004.

    Google Scholar 

  17. A. Lombard, H. Buchner, and W. Kellermann, “Multidimensional localization of multiple sound sources using blind adaptive MIMO system identification,” in Proc. IEEE Int. Conf. Multisensor Fusion and Integration for Intelligent Systems (MFI), Heidelberg, Germany, Sept. 2006.

    Google Scholar 

  18. S. Haykin, Adaptive Filter Theory, 4th ed., Prentice-Hall, Englewood Cliffs, NJ, 2002.

    Google Scholar 

  19. A. Hyvärinen, J. Karhunen, and E. Oja, Independent Component Analysis, Wiley & Sons, Inc., New York, 2001.

    Book  Google Scholar 

  20. S.C. Douglas, “Blind separation of acoustic signals” in M. Brandstein and D. Ward (eds.), Microphone Arrays: Signal Processing Techniques and Applications, pp. 355-380, Springer, Berlin, 2001.

    Google Scholar 

  21. J.-F. Cardoso and A. Souloumiac, “Blind beamforming for non gaussian sig-nals,” IEE Proceedings-F, vol. 140, no. 6, pp. 362-370, Dec. 1993.

    Google Scholar 

  22. S. Araki et al., “Equivalence between frequency-domain blind source separa-tion and frequency-domain adaptive beamforming,” in Proc. IEEE Int. Conf. Acoustics, Speech, Signal Processing (ICASSP), Orlando, FL, USA, pp. 1785-1788, May 2002.

    Google Scholar 

  23. M. Miyoshi and Y. Kaneda, “Inverse filtering of room acoustics,” IEEE Trans. Acoust., Speech, Signal Processing, vol. 36, no. 2, pp. 145-152, Feb. 1988.

    Article  Google Scholar 

  24. 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 Communication (HSC), Kyoto, Japan, pp. 59-62, Apr. 2001.

    Google Scholar 

  25. M.I. Gürelli and C.L. Nikias, “EVAM: An eigenvector-based algorithm for mul-tichannel blind deconvolution of input colored signals,” IEEE Trans. Signal Process., vol. 43, no. 1, pp. 134-149, Jan. 1995.

    Article  Google Scholar 

  26. K. Furuya and Y. Kaneda, “Two-channel blind deconvolution of nonmini-mum phase FIR systems,” IEICE Trans. Fundamentals, vol. E80-A, no. 5, pp. 804-808, May 1997.

    Google Scholar 

  27. S. Amari et al.,“Multichannel blind deconvolution and equalization using the natural gradient,” in Proc. IEEE Int. Workshop Signal Processing Advances in Wireless Communications, pp. 101-107, 1997.

    Google Scholar 

  28. S. Choi et al., “Natural gradient learning with a nonholonomic constraint for blind deconvolution of multiple channels,” in Proc. Int. Symp. Independent Component Analysis Blind Source Separation (ICA), pp. 371-376, 1999.

    Google Scholar 

  29. B.W. Gillespie and L. Atlas, “Strategies for improving audible quality and speech recognition accuracy of reverberant speech,” in Proc. IEEE Int. Conf. Acoustics, Speech, Signal Processing (ICASSP), Hongkong, China, Apr. 2003.

    Google Scholar 

  30. K. Matsuoka and S. Nakashima, “Minimal distortion principle for blind source separation,” in Proc. Int. Symp. Independent Component Analysis Blind Signal Separation (ICA), San Diego, CA, USA, Dec. 2001.

    Google Scholar 

  31. H. Sawada, R. Mukai, S. Araki, and S. Makino, “A robust and precise method for solving the permutation problem of frequency-domain blind source separa-tion,” IEEE Trans. Speech Audio Process., vol. 12, no. 8, Sept. 2004.

    Google Scholar 

  32. H. Liu, G. Xu, and L. Tong, “A deterministic approach to blind identification of multi-channel FIR systems,” in Proc. IEEE Int. Conf. Acoustics, Speech, Signal Processing (ICASSP), Adelaide, Australia, Apr. 1994.

    Google Scholar 

  33. H.-C. Wu and J.C. Principe,“Simultaneous diagonalization in the frequency domain (SDIF) for source separation,” in Proc. Int. Symp. Independent Com-ponent Analysis Blind Signal Separation (ICA), pp. 245-250, 1999.

    Google Scholar 

  34. C.L. Fancourt and L. Parra, “The coherence function in blind source separa-tion of convolutive mixtures of non-stationary signals,” in Proc. Int. Workshop Neural Networks Signal Processing (NNSP), 2001, pp. 303-312.

    Google Scholar 

  35. T.M. Cover and J.A. Thomas, Elements of Information Theory, Wiley & Sons, New York, 1991.

    Book  MATH  Google Scholar 

  36. R. Aichner, H. Buchner, F. Yan, and W. Kellermann, “A real-time blind source separation scheme and its application to reverberant and noisy acoustic envi-ronments,” Signal Processing, vol. 86, no. 6, pp.1260-1277, 2006.

    Article  MATH  Google Scholar 

  37. M. Kawamoto, K. Matsuoka, and N. Ohnishi, “A method of blind separation for convolved non-stationary signals,” Neurocomputing, vol. 22, pp. 157-171, 1998.

    Article  MATH  Google Scholar 

  38. R. Aichner, H. Buchner, and W. Kellermann, “Exploiting narrowband effi-ciency for broadband convolutive blind source separation,” EURASIP Journal on Applied Signal Processing, vol. 2007, pp. 1-9, Sept. 2006.

    Google Scholar 

  39. T. Nishikawa, H. Saruwatari, and K. Shikano, “Comparison of time-domain ICA, frequency-domain ICA and multistage ICA for blind source separation,” in Proc. European Signal Processing Conference (EUSIPCO), vol. 2, pp. 15-18, Sept. 2002.

    Google Scholar 

  40. K. Yao, “A representation theorem and its applications to spherically-invariant random processes,” IEEE Trans. Inform. Theor., vol. 19, no. 5, pp. 600-608, Sept. 1973.

    Article  MATH  Google Scholar 

  41. J. Goldman, “Detection in the presence of spherically symmetric random vec-tors,” IEEE Trans. Inform. Theor., vol. 22, no. 1, pp. 52-59, Jan. 1976.

    Article  MATH  Google Scholar 

  42. H. Brehm and W. Stammler, “Description and generation of spherically invari-ant speech-model signals,” Signal Processing, vol. 12, pp. 119-141, 1987.

    Article  Google Scholar 

  43. S. Araki et al., “The fundamental limitation of frequency-domain blind source separation for convolutive mixtures of speech,” IEEE Trans. Speech Audio Process., vol. 11, no. 2, pp. 109-116, Mar. 2003.

    Article  MathSciNet  Google Scholar 

  44. H. Sawada et al., “Spectral smoothing for frequency-domain blind source sepa-ration,” in Proc. Int. Workshop Acoustic Echo and Noise Control (IWAENC), Kyoto, Japan, Sept. 2003, pp. 311-314.

    Google Scholar 

  45. M.Z. Ikram and D.R. Morgan, “Exploring permutation inconsistency in blind separation of speech signals in a reverberant environment,” in Proc. IEEE Int. Conf. Acoustics, Speech, Signal Processing (ICASSP), Istanbul, Turkey, June 2000, vol. 2, pp. 1041-1044.

    Google Scholar 

  46. H. Buchner, R. Aichner, and W. Kellermann, “A generalization of a class of blind source separation algorithms for convolutive mixtures,” in Proc. Int. Symp. Independent Component Analysis Blind Signal Separation (ICA), Nara, Japan, Apr. 2003.

    Google Scholar 

  47. T. Kim, T. Eltoft, and T.-W. Lee, “Independent vector analysis: an extension of ICA to multivariate components,” in Proc. Int. Conf. Independent Component Analysis Blind Signal Separation (ICA), Mar. 2006.

    Google Scholar 

  48. A. Hiroe, “Solution of permutation problem in frequency domain ICA using multivariate probability density functions,” in Proc. Int. Conf. Independent Component Analysis Blind Signal Separation (ICA), pp. 601-608, Mar. 2006.

    Google Scholar 

  49. P. Smaragdis, “Blind separation of convolved mixtures in the frequency domain,” Neurocomputing, vol. 22, pp. 21-34, July 1998.

    Article  MATH  Google Scholar 

  50. D.H. Johnson and D.E. Dudgeon, Array Signal Processing, Prentice Hall, New Jersey, 1993.

    MATH  Google Scholar 

  51. H. Wang and M. Kaveh, “Coherent Signal-Subspace Processing for the De-tection and Estimation of Angles of Arrival of Multiple Wide-Band Sources,” IEEE Trans. Acoust., Speech, Signal Processing, vol. ASSP-33, no. 4, pp. 823-831, Aug. 1985.

    Google Scholar 

  52. H. Teutsch and W. Kellermann, “Acoustic source detection and localization based on wavefield decomposition using circular microphone arrays,” J. Acoust. Soc. Am., vol. 120, no. 5, Nov. 2006.

    Google Scholar 

  53. W.R. Hahn and S.A. Tretter, “Optimum processing for delay-vector estimation in passive signal arrays,” IEEE Trans. Inform. Theory, vol. IT-19, pp. 608-614, May 1973.

    Article  Google Scholar 

  54. M. Wax and T. Kailath, “Optimum localization of multiple sources by passive arrays,” IEEE Trans. Acoust., Speech, Signal Processing, vol. ASSP-31, no. 5, pp. 1210-1218, Oct. 1983.

    Article  MathSciNet  Google Scholar 

  55. P.E. Stoica and A. Nehorai, “MUSIC, maximum likelihood and Cramer-Rao bound,” IEEE Trans. Acoust., Speech, Signal Processing, vol. 37, pp. 720-740, May 1989.

    Article  MATH  MathSciNet  Google Scholar 

  56. J.C. Chen, R.E. Hudson, and K. Yao, “Maximum-likelihood source localization and unknown sensor location estimation for wideband signals in the near-field,” IEEE Trans. Signal Process., vol. 50, pp. 1843-1854, Aug. 2002.

    Article  Google Scholar 

  57. Y. Bard, Nonlinear Parameter Estimation, Academic Press, New York, 1974.

    MATH  Google Scholar 

  58. W.H. Foy, “Position-location solutions by Taylor-series estimation,” IEEE Trans. Aerosp. Electron. Syst., vol. AES-12, pp. 187-194, Mar. 1976.

    Article  Google Scholar 

  59. R.O. Schmidt, “A new approach to geometry of range difference location,” IEEE Trans. Aerosp. Electron., vol. AES-8, pp. 821-835, Nov. 1972.

    Article  Google Scholar 

  60. H.C. Schau and A.Z. Robinson, “Passive source localization employing intersecting spherical surfaces from time-of-arrival differences,” IEEE Trans. Acoust., Speech, Signal Processing, vol. ASSP-35, no. 8, pp. 1223-1225, Aug. 1987.

    Article  Google Scholar 

  61. J.O. Smith and J.S. Abel, “Closed-form least-squares source location estima-tion from range-difference measurements,” IEEE Trans. Acoust., Speech, Signal Processing, vol. ASSP-35, no. 12, pp. 1661-1669, Dec. 1987.

    Article  Google Scholar 

  62. Y.T. Chan and K.C. Ho, “A simple and efficient estimator for hyperbolic loca-tion,” IEEE Trans. Signal Process., vol. 42, no. 8, pp. 1905-1915, Aug. 1994.

    Article  MathSciNet  Google Scholar 

  63. Y.T. Chan and K.C. Ho, “An efficient closed-form localization solution from time difference of arrival measurements,” in Proc. IEEE Int. Conf. Acoustics, Speech, Signal Processing (ICASSP), 1994, vol. 2, pp. 393-396.

    Google Scholar 

  64. Y. Huang, J. Benesty, G.W. Elko, and R.M. Mersereau, “Real-time pas-sice source localization: an unbiased linear-correction least-squares approach,” IEEE Trans. Speech Audio Process., vol. 9, no. 8, pp. 943-956, Nov. 2001.

    Article  Google Scholar 

  65. J.S. Abel and J.O. Smith, “The spherical interpolation method for closed-form passive source localization using range difference measurements,” in Proc. IEEE Int. Conf. Acoustics, Speech, Signal Processing (ICASSP), vol. 1, pp. 471-474, 1987.

    Google Scholar 

  66. T.I. Laakso et al., “Splitting the unit delay,” IEEE Signal Processing Mag., vol. 13, pp. 30-60, 1996.

    Article  Google Scholar 

  67. M.S. Brandstein and H.F. Silverman, “A robust method for speech signal time-delay estimation in reverberant rooms,” in Proc. IEEE Int. Conf. Acoustics, Speech, Signal Processing (ICASSP), Munich, Apr. 1997.

    Google Scholar 

  68. A. Stéphenne and B. Champagne, “A new cepstral prefiltering technique for estimating time delay under reverberant conditions,” Signal Processing, vol. 59, pp. 253-266, 1997.

    Article  MATH  Google Scholar 

  69. R. Aichner, H. Buchner, S. Wehr, and W. Kellermann, “Robustness of acoustic multiple-source localization in adverse environments,” in Proc. ITG Fachtagung Sprachkommunication, Kiel, Germany, Apr. 2006.

    Google Scholar 

  70. M. Krinidis et al., “An audio-visual database for evaluating person tracking algorithms,” in Proc. IEEE Int. Conf. Acoustics, Speech, Signal Processing (ICASSP), Philadelphia, PA, USA, Mar. 2005.

    Google Scholar 

Download references

Author information

Authors and Affiliations

Authors

Editor information

Editors and Affiliations

Rights and permissions

Reprints and permissions

Copyright information

© 2007 Springer

About this chapter

Cite this chapter

Buchner, H., Aichner, R., Kellermann, W. (2007). TRINICON-based Blind System Identification with Application to Multiple-Source Localization and Separation. In: Makino, S., Sawada, H., Lee, TW. (eds) Blind Speech Separation. Signals and Communication Technology. Springer, Dordrecht. https://doi.org/10.1007/978-1-4020-6479-1_4

Download citation

  • DOI: https://doi.org/10.1007/978-1-4020-6479-1_4

  • Publisher Name: Springer, Dordrecht

  • Print ISBN: 978-1-4020-6478-4

  • Online ISBN: 978-1-4020-6479-1

  • eBook Packages: EngineeringEngineering (R0)

Publish with us

Policies and ethics