EURASIP Journal on Audio, Speech, and Music Processing
Volume 2008 (2008), Article ID 156960, 9 pages
doi:10.1155/2008/156960
Research Article

Frequency-Domain Adaptive Algorithm for Network Echo Cancellation in VoIP

Xiang (Shawn) Lin,1 Andy W. H. Khong,1 Milŏs Doroslovăcki,2 and Patrick A. Naylor1

1Department of Electrical and Electronic Engineering, Imperial College London, London SW7 2AZ, UK
2Department of Electrical and Computer Engineering, The George Washington University, Washington, DC 20052, USA

Received 1 November 2007; Accepted 8 April 2008

Academic Editor: Sen Kuo

Copyright © 2008 Xiang (Shawn) Lin et al. This is an open access article distributed under the Creative Commons Attribution License, which permits unrestricted use, distribution, and reproduction in any medium, provided the original work is properly cited.

Linked References

  1. B. Goode, “Voice over internet protocol (VoIP),” Proceedings of the IEEE, vol. 90, no. 9, pp. 1495–1517, 2002.
  2. H. M. Chong and H. S. Matthews, “Comparative analysis of traditional telephone and voice-over-internet protocol (VoIP) systems,” in Proceedings of the IEEE International Symposium on Electronics and the Environment (ISEE '04), pp. 106–111, Phoenix, Ariz, USA, May 2004.
  3. H.-G. Kang, H. K. Kim, and R. V. Cox, “Improving the transcoding capability of speech coders,” IEEE Transactions on Multimedia, vol. 5, no. 1, pp. 24–33, 2003.
  4. G. L. Choudhury and R. G. Cole, “Design and analysis of optimal adaptive de-jitter buffers,” Computer Communications, vol. 27, no. 6, pp. 529–537, 2004.
  5. A. Raake, “Short- and long-term packet loss behavior: towards speech quality prediction for arbitrary loss distributions,” IEEE Transactions on Audio, Speech, and Language Processing, vol. 14, no. 6, pp. 1957–1968, 2006.
  6. M. M. Sondhi and D. A. Berkley, “Silencing echoes on the telephone network,” Proceedings of the IEEE, vol. 68, no. 8, pp. 948–963, 1980.
  7. J. Radecki, Z. Zilic, and K. Radecka, “Echo cancellation in IP networks,” in Proceedings of the 45th International Midwest Symposium on Circuits and Systems (MWSCAS '02), vol. 2, pp. 219–222, Tulsa, Okla, USA, August 2002.
  8. J. Benesty, T. Gänsler, D. R. Morgan, M. M. Sondhi, and S. L. Gay, Advances in Network and Acoustic Echo Cancellation, Springer, Berlin, Germany, 2001.
  9. D. L. Duttweiler, “Proportionate normalized least-mean-squares adaptation in echo cancelers,” IEEE Transactions on Speech and Audio Processing, vol. 8, no. 5, pp. 508–518, 2000.
  10. J. Benesty and S. L. Gay, “An improved PNLMS algorithm,” in Proceedings of the IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP '02), vol. 2, pp. 1881–1884, Orlando, Fla, USA, May 2002.
  11. J. Cui, P. A. Naylor, and D. T. Brown, “An improved IPNLMS algorithm for echo cancellation in packet-switched networks,” in Proceedings of the IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP '04), vol. 4, pp. 141–144, Montreal, Quebec, Canada, May 2004.
  12. H. Deng and M. Doroslovački, “New sparse adaptive algorithms using partial update,” in Proceedings of the IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP '04), vol. 2, pp. 845–848, Montreal, Quebec, Canada, May 2004.
  13. T. Aboulnasr and K. Mayyas, “Complexity reduction of the NLMS algorithm via selective coefficient update,” IEEE Transactions on Signal Processing, vol. 47, no. 5, pp. 1421–1424, 1999.
  14. S. M. Kuo and D. R. Morgan, “Active noise control: a tutorial review,” Proceedings of the IEEE, vol. 87, no. 6, pp. 943–973, 1999.
  15. A. Carini and G. L. Sicuranza, “Analysis of transient and steady-state behavior of a multichannel filtered-x partial-error affine projection algorithm,” EURASIP Journal on Audio, Speech, and Music Processing, vol. 2007, Article ID 31314, 15 pages, 2007.
  16. H. Deng and M. Doroslovački, “Proportionate adaptive algorithms for network echo cancellation,” IEEE Transactions on Signal Processing, vol. 54, no. 5, pp. 1794–1803, 2006.
  17. E. R. Ferrara, “Fast implementations of LMS adaptive filters,” IEEE Transactions on Acoustics, Speech, and Signal Processing, vol. 28, no. 4, pp. 474–475, 1980.
  18. J. W. Cooley and J. W. Tukey, “An algorithm for the machine calculation of complex Fourier series,” Mathematics of Computation, vol. 19, no. 90, pp. 297–301, 1965.
  19. S. Haykin, Adaptive Filter Theory, Information and System Science, Prentice-Hall, Upper Saddle River, NJ, USA, 4th edition, 2002.
  20. J. J. Shynk, “Frequency-domain and multirate adaptive filtering,” IEEE Signal Processing Magazine, vol. 9, no. 1, pp. 14–37, 1992.
  21. J.-S. Soo and K. K. Pang, “Multidelay block frequency domain adaptive filter,” IEEE Transactions on Acoustics, Speech, and Signal Processing, vol. 38, no. 2, pp. 373–376, 1990.
  22. A. W. H. Khong, P. A. Naylor, and J. Benesty, “A low delay and fast converging improved proportionate algorithm for sparse system identification,” EURASIP Journal on Audio, Speech, and Music Processing, vol. 2007, Article ID 84376, 8 pages, 2007.
  23. I. Pitas, “Fast algorithms for running ordering and max/min calculation,” IEEE Transactions on Circuits and Systems, vol. 36, no. 6, pp. 795–804, 1989.
  24. P. A. Naylor and W. Sherliker, “A short-sort M-Max NLMS partial-update adaptive filter with applications to echo cancellation,” in Proceedings of the IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP '03), vol. 5, pp. 373–376, Hong Kong, April 2003.
  25. P. O. Hoyer, “Non-negative matrix factorization with sparseness constraints,” Journal of Machine Learning Research, vol. 5, pp. 1457–1469, 2004.
  26. J. Benesty, Y. A. Huang, J. Chen, and P. A. Naylor, “Adaptive algorithms for the identification of sparse impulse responses,” in Selected Methods for Acoustic Echo and Noise Control, E. Hänsler and G. Schmidt, Eds., pp. 125–153, Springer, Berlin, Germany, 2006.
  27. A. W. H. Khong and P. A. Naylor, “Efficient use of sparse adaptive filters,” in Proceedings of the 40th Asilomar Conference on Signals, Systems and Computers (ACSSC '06), pp. 1375–1379, Pacific Grove, Calif, USA, October-November 2006.
  28. A. W. H. Khong and P. A. Naylor, “Selective-tap adaptive filtering with performance analysis for identification of time-varying systems,” IEEE Transactions on Audio, Speech, and Language Processing, vol. 15, no. 5, pp. 1681–1695, 2007.
  29. R. Ahmad, A. W. H. Khong, and P. A. Naylor, “Proportionate frequency domain adaptive algorithms for blind channel identification,” in Proceedings of the IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP '06), vol. 5, pp. V29–V32, Toulouse, France, May 2006.