Skip to main content

Interface for Barge-in Free Spoken Dialogue System Based on Sound Field Reproduction and Microphone Array

Abstract

A barge-in free spoken dialogue interface using sound field control and microphone array is proposed. In the conventional spoken dialogue system using an acoustic echo canceller, it is indispensable to estimate a room transfer function, especially when the transfer function is changed by various interferences. However, the estimation is difficult when the user and the system speak simultaneously. To resolve the problem, we propose a sound field control technique to prevent the response sound from being observed. Combined with a microphone array, the proposed method can achieve high elimination performance with no adaptive process. The efficacy of the proposed interface is ascertained in the experiments on the basis of sound elimination and speech recognition.

References

  1. 1.

    Juang BH, Soong FK: Hands-free telecommunications. Proceedings of International Workshop on Hands-Free Speech Communication, April 2001, Kyoto, Japan 5–8.

    Google Scholar 

  2. 2.

    Hänsler E: Acoustic echo and noise control: where do we come from—where do we go? Proceedings of International Workshop on Acoustic Echo and Noise Control (IWAENC '01), September 2001, Darmstadt, Germany 1–4.

    Google Scholar 

  3. 3.

    Makino S, Shimauchi S: Stereophonic acoustic echo cancellation—an overview and recent solutions. Proceedings of 6th IEEE International Workshop on Acoustic Echo and Noise Control (IWAENC '99), September 1999, Pocono Manor, Pa, USA 12–19.

    Google Scholar 

  4. 4.

    Jung Y-W, Lee J-H, Park Y-C, Youn D-H: A new adaptive algorithm for stereophonic acoustic echo canceller. Proceedings of IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP '00), June 2000, Istanbul, Turkey 2: 801–804.

    Google Scholar 

  5. 5.

    Herbordt W, Kellermann W: Acoustic echo cancellation embedded into the generalized sidelobe canceller. Proceedings of European Signal Processing Conference (EUPSICO '00), September 2000, Tampere, Finlande 3: 1843–1846.

    Article  Google Scholar 

  6. 6.

    Buchner H, Spors S, Kellermann W: Wave-domain adaptive filtering: acoustic echo cancellation for full-duplex systems based on wave-field synthesis. Proceedings of IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP '04), May 2004, Montreal, Que, Canada 4: 117–120.

    Google Scholar 

  7. 7.

    Tatekura Y, Saruwatari H, Shikano K: Sound reproduction system including adaptive compensation of temperature fluctuation effect for broad-band sound control. IEICE Transactions on Fundamentals of Electronics, Communications and Computer Sciences 2002,E85-A(8):1851-1860.

    Google Scholar 

  8. 8.

    Benesty J, Morgan DR, Cho JH: A family of doubletalk detectors based on cross-correlation. Proceedings of 6th IEEE International Workshop on Acoustic Echo and Noise Control (IWAENC '99), September 1999, Pocono Manor, Pa, USA 108–111.

    Google Scholar 

  9. 9.

    Ochiai K, Araseki T, Ogihara T: Echo canceler with two echo path models. IEEE Transactions on Communications 1977,25(6):589-595. 10.1109/TCOM.1977.1093869

    Article  Google Scholar 

  10. 10.

    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 

  11. 11.

    Bauck J, Cooper DH: Generalized transaural stereo and applications. Journal of the Audio Engineering Society 1996,44(9):683-705.

    Google Scholar 

  12. 12.

    Tatekura Y, Saruwatari H, Shikano K: An iterative inverse filter design method for the multichannel sound field reproduction system. IEICE Transactions on Fundamentals of Electronics, Communications and Computer Sciences 2001,E84-A(4):991-998.

    Google Scholar 

  13. 13.

    Haykin S: Adaptive Filter Theory. 4th edition. Prentice-Hall, Englewood Cliffs, NJ, USA; 1991.

    Google Scholar 

  14. 14.

    Suzuki Y, Asano F, Kim H-Y, Sone T: An optimum computer-generated pulse signal suitable for the measurement of very long impulse responses. Journal of the Acoustical Society of America 1995,97(2):1119-1123. 10.1121/1.412224

    Article  Google Scholar 

  15. 15.

    Blauert J: Spatial Hearing. Revised edition. MIT Press, Cambridge, Mass, USA; 1997.

    Google Scholar 

  16. 16.

    Flanagan JL, Johnston JD, Zahn R, Elko GW: Computer-steered microphone arrays for sound transduction in large rooms. Journal of the Acoustical Society of America 1985,78(5):1508-1518. 10.1121/1.392786

    Article  Google Scholar 

  17. 17.

    Hayamizu S, Itahashi S, Kobayashi T, Takezawa T: Design and creation of speech and text corpora of dialogue. IEICE Transactions on Information and Systems 1993,E76-D(1):17-22.

    Google Scholar 

  18. 18.

    Lee A, Kawahara T, Shikano K: Julius—an open source real-time large vocabulary recognition engine. Proceedings of 7th European Conference on Speech Communication and Technology (EUROSPEECH '01), September 2001, Aalborg, Denmark 1691–1694.

    Google Scholar 

  19. 19.

    Lee A, Kawahara T, Takeda K, Shikano K: A new phonetic tied-mixture model for efficient decoding. Proceedings of IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '00), June 2000, Istanbul, Turkey 3: 1269–1272.

    Google Scholar 

  20. 20.

    Yamade S, Lee A, Saruwatari H, Shikano K: Unsupervised speaker adaptation based on HMM sufficient statistics in various noisy environments. Proceedings of 8th European Conference on Speech Communication and Technology (EUROSPEECH '03), September 2003, Geneva, Switzerland 2: 1493–1496.

    Google Scholar 

  21. 21.

    Itou K, Yamamoto M, Takeda K, et al.: The design of the newspaper-based Japanese large vocabulary continuous speech recognition corpus. Proceedings of 5th International Conference on Spoken Language Processing (ICSLP '98), November-December 1998, Sydney, Australia 7: 3261–3264.

    Google Scholar 

  22. 22.

    Itou K, Yamamoto M, Takeda K, et al.: JNAS: Japanese speech corpus for large vocabulary continuous speech recognition research. Journal of the Acoustical Society of Japan (E) 1999,20(3):199-206.

    Article  Google Scholar 

  23. 23.

    Rabiner L, Juang BH: Fundamentals of Speech Recognition. Prentice-Hall, Englewood Cliffs, NJ, USA; 1993.

    Google Scholar 

  24. 24.

    Deller JR Jr., Hansen JHL, Proakis JG: Discrete-Time Processing of Speech Signals. Macmillan, New York, NY, USA; 1993.

    Google Scholar 

  25. 25.

    Miyabe S, Takatani T, Mori Y, Saruwatari H, Shikano K, Tatekura Y: Double-talk free spoken dialogue interface combining sound field control with semi-blind source separation. Proceedings of IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP '06), May 2006, Toulouse, France 1: 809–812.

    Google Scholar 

Download references

Author information

Affiliations

Authors

Corresponding author

Correspondence to Shigeki Miyabe.

Rights and permissions

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.

Reprints and Permissions

About this article

Cite this article

Miyabe, S., Hinamoto, Y., Saruwatari, H. et al. Interface for Barge-in Free Spoken Dialogue System Based on Sound Field Reproduction and Microphone Array. EURASIP J. Adv. Signal Process. 2007, 057470 (2007). https://doi.org/10.1155/2007/57470

Download citation

Keywords

  • Information Technology
  • Transfer Function
  • Quantum Information
  • Speech Recognition
  • Control Technique