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Equalization of Loudspeaker and Room Responses Using Kautz Filters: Direct Least Squares Design

Abstract

DSP-based correction of loudspeaker and room responses is becoming an important part of improving sound reproduction. Such response equalization (EQ) is based on using a digital filter in cascade with the reproduction channel to counteract the response errors introduced by loudspeakers and room acoustics. Several FIR and IIR filter design techniques have been proposed for equalization purposes. In this paper we investigate Kautz filters, an interesting class of IIR filters, from the point of view of direct least squares EQ design. Kautz filters can be seen as generalizations of FIR filters and their frequency-warped counterparts. They provide a flexible means to obtain desired frequency resolution behavior, which allows low filter orders even for complex corrections. Kautz filters have also the desirable property to avoid inverting dips in transfer function to sharp and long-ringing resonances in the equalizer. Furthermore, the direct least squares design is applicable to nonminimum-phase EQ design and allows using a desired target response. The proposed method is demonstrated by case examples with measured and synthetic loudspeaker and room responses.

References

  1. Greenfield R, Hawksford MO: Efficient filter design for loudspeaker equalization. Journal of the Audio Engineering Society 1991,39(10):739–751.

    Google Scholar 

  2. Karjalainen M, Piirilä E, Järvinen A, Huopaniemi J: Comparison of loudspeaker equalization methods based on DSP techniques. Journal of the Audio Engineering Society 1999,47(1–2):15–31.

    Google Scholar 

  3. Karjalainen M, Paatero T, Mourjopoulos JN, Hatziantoniou PD: About room response equalization and dereverberation. Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA '05), October 2005, New Paltz, NY, USA 183–186.

    Google Scholar 

  4. Miyoshi M, Kaneda Y: Inverse filtering of room acoustics. IEEE Transactions on Acoustics, Speech, and Signal Processing 1988,36(2):145–152. 10.1109/29.1509

    Article  Google Scholar 

  5. Mourjopoulos JN: Digital equalization of room acoustics. Journal of the Audio Engineering Society 1994,42(11):884–900.

    Google Scholar 

  6. Mourjopoulos JN: Comments on 'analysis of traditional and reverberation-reducing methods of room equalization'. Journal of the Audio Engineering Society 2003,51(12):1186–1188.

    Google Scholar 

  7. Neely ST, Allen JB: Invertibility of a room impulse response. Journal of the Acoustical Society of America 1979,66(1):165–169. 10.1121/1.383069

    Article  Google Scholar 

  8. Radlovic BD, Kennedy RA: Nonminimum-phase equalization and its subjective importance in room acoustics. IEEE Transactions on Speech and Audio Processing 2000,8(6):728–737. 10.1109/89.876311

    Article  Google Scholar 

  9. Paatero T, Karjalainen M: Kautz filters and generalized frequency resolution: theory and audio applications. Journal of the Audio Engineering Society 2003,51(1–2):27–44.

    Google Scholar 

  10. Hatziantoniou PD, Mourjopoulos JN: Generalized fractional-octave smoothing of audio and acoustic responses. Journal of the Audio Engineering Society 2000,48(4):259–280.

    Google Scholar 

  11. Worley JW, Hatziantoniou PD, Mourjopoulos JN: Subjective assessments of real-time room dereverberation and loudspeaker equalization. Proceedings of 118th Audio Engineering Society Convention, May 2005, Barcelona, Spain Paper 6461

    Google Scholar 

  12. Mäkivirta A, Antsalo P, Karjalainen M, Välimäki V: Modal equalization of loudspeaker-room responses at low frequencies. Journal of the Audio Engineering Society 2003,51(5):324–343.

    Google Scholar 

  13. Fielder LD: Analysis of traditional and reverberation-reducing methods of room equalization. Journal of the Audio Engineering Society 2003,51(1–2):3–26.

    Google Scholar 

  14. Paatero T, Karjalainen M: Equalization of audio systems using Kautz filters with logarithmic allocation of frequency resolution. Proceedings of 120th Audio Engineering Society Convention, May 2006, Paris, France Paper 6767

    Google Scholar 

  15. Broome PW: Discrete orthonormal sequences. Journal of the Association for Computing Machinery 1965,12(2):151–168.

    Article  MathSciNet  Google Scholar 

  16. Kautz WH: Transient synthesis in the time domain. IRE Transactions on Circuit Theory 1954, 1: 29–39.

    Article  Google Scholar 

  17. Walsh JL: Interpolation and Approximation by Rational Functions in the Complex Domain. 2nd edition. American Mathematical Society, Providence, RI, USA; 1969.

    Google Scholar 

  18. Mourjopoulos J, Clarkson P, Hammond J: A comparative study of least-squares and homomorphic techniques for the inversion of mixed phase signals. Proceedings of IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '82), May 1982, Paris, France 7: 1858–1861.

    Article  Google Scholar 

  19. Härmä A, Karjalainen M, Savioja L, Välimäki V, Laine UK, Huopaniemi J: Frequency-warped signal processing for audio applications. Journal of the Audio Engineering Society 2000,48(11):1011–1031.

    Google Scholar 

  20. Moore BCJ, Peters RW, Glasberg BR: Auditory filter shapes at low center frequencies. Journal of the Acoustical Society of America 1990,88(1):132–140. 10.1121/1.399960

    Article  Google Scholar 

  21. Härmä A, Paatero T: Discrete representation of signals on a logarithmic frequency scale. Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA '01), October 2001, New Paltz, NY, USA 39–42.

    Google Scholar 

  22. Parks TW, Burrus CS: Digital Filter Design. John Wiley & Sons, New York, NY, USA; 1987.

    MATH  Google Scholar 

  23. Hayes MH: Statistical Digital Signal Processing and Modeling. John Wiley & Sons, New York, NY, USA; 1996.

    Google Scholar 

  24. Young TY, Huggins WH: 'Complementary' signals and orthogonalized exponentials. IRE Transactions on Circuit Theory 1962, 9: 362–370.

    Article  Google Scholar 

  25. Brandenstein H, Unbehauen R: Least-squares approximation of FIR by IIR digital filters. IEEE Transactions on Signal Processing 1998,46(1):21–30. 10.1109/78.651163

    Article  Google Scholar 

  26. Paatero T: Generalized linear-in-parameter models—theory and audio signal processing applications, Doctoral thesis. Helsinki University of Technology, Laboratory of Acoustics and Audio Signal Processing (Report no. 75), Espoo, Finland, November 2005.

    Google Scholar 

  27. Paatero T: An audio motivated hybrid of warping and Kautz filter techniques. Proceedings of European Signal Processing Conference (EUSIPCO '02), September 2002, Toulouse, France 627–630.

    Google Scholar 

  28. Linkwitz SH: Active crossover networks for noncoincident drivers. Journal of the Audio Engineering Society 1976,24(1):2–8.

    Google Scholar 

  29. Linkwitz SH: Passive crossover networks for noncoincident drivers. Journal of the Audio Engineering Society 1978,26(3):149–150.

    Google Scholar 

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Correspondence to Matti Karjalainen.

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Open Access This article is distributed under the terms of the Creative Commons Attribution 2.0 International License (https://doi.org/creativecommons.org/licenses/by/2.0), which permits unrestricted use, distribution, and reproduction in any medium, provided the original work is properly cited.

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Karjalainen, M., Paatero, T. Equalization of Loudspeaker and Room Responses Using Kautz Filters: Direct Least Squares Design. EURASIP J. Adv. Signal Process. 2007, 060949 (2006). https://doi.org/10.1155/2007/60949

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