Block Robust Algorithm for Network Echo Cancellation
the author of the article can submit here a request for assignment of a DOI number to this resource!
Cost of the service: euros 10,00 (for a DOI)
This paper is about an efficient implementation of adaptive filtering for echo cancelers. Recently a fast converging algorithm called Robust Proportionate Normalized Least Mean Squares (RPNLMS++) against double-talk has been proposed. This paper presents a realization of an improved version of the previous RPNLMS++ adaptive filter using block structure in which the filter coefficients are adjusted one per each output block. Then, an efficient implementation of the block filtering process is proposed using Number Theoretic Transforms (NTT) which can significantly reduce the computation complexity of filter implantation on Digital Signal Processor (DSP). Analyses of convergence properties, during single and double-talk, and complexity show that the new block adaptive filter permits fast implementations while maintaining performance equivalent to that of the widely used RPNLMS++ adaptive filter.
Copyright © 2013 Praise Worthy Prize - All rights reserved.
J. Benesty, T. Gansler, D.R. Morgan, M.M. Sondhi, S.L. Gay (2001). Advances in Network and Acoustic Echo Cancellation. Springer-Verlag.
D.L. Duttweiler (2000). Proportionate normalized least means square adaptation in echo cancelers. IEEE Trans. Speech Audio Processing, vol. 8, pp. 508-518.
D. L. Duttweiler (1978). A twelve-channel digital echo canceler. IEEE Trans.Commun., vol. COM-26, pp. 647–653.
T. Gänsler, S.L. Gay, M.M. Sondhi, and J. Benesty (2000). Double-talk robust fast converging algorithms for network echo cancellation. IEEE Transactions on Speech and Audio Processing. vol. 8, pp. 656-663.
R. C. Agarwal, C.S. Burrus (1975). Number theoretic transform to implement fast digital convolution. Proc. IEEE, vol. 63, pp. 550-560.
R. C. Agarwal, C. S. Burrus (1974). Fast convolution using Fermat number transform with application to digital filtering. IEEE trans. Acoust., Speech and Signal proces, ASSP-22, N°2.
G. A. Jullien (1981). Number Theoretic Techniques in Digital Signal Processing. Academic Press, vol. 80, Chap. 2, pp. 69-163.
S. Gudvangen and A. Patel (1995). Rapid synthesis of a macro-pipelined CMOS ASIC for the Fermat number transform. in Proc. NORSIG, 1-2 Sept. 1995, Stavanger, Norway.
M. M. Sondhi (1967). An adaptive echo canceler. Bell Syst. Tech. J, vol. XLVI-3, pp. 497–510.
P. J. Huber, Robust Statistics (1981). New York:Wiley, pp. 68–71, 109, 135–138.
S. L. Gay and S. Tavathia (1995). The fast affine projection algorithm. in Proc. IEEE Int. Conf. Acoustics, Speech, Signal Processing, pp. 3023–3026.
G. A. Clark, S.K. Mitra, and S.R. Parker (1981). Block implementation of adaptive digital filters. IEEE Trans. On Acoust., Speech and Signal Processing, vol. ASSP-29, pp. 744-752.
E. H. Baghious, G. Madre, H. Alaeddine, G. Burel (2004). Realization of adaptive blocks digital filters using Fermat Number Transform. International Symposium on Image/Video Communications over fixed and mobile networks (ISIVC’04), Brest, France, July 2004.
Gilloire A., Vetterli M (1994). Performance evaluation of acoustic echo controls : required values and measurement procedures. Annales des Télécommunications, vol 49, N◦7 − 8, pp. 368-372, 1994.
Recommandation UIT-T G.131 (08/96), Réduction de l’écho pour le locuteur.
Omid S. T., Mohsen A., Payman M., An FPGA-Based Implementation of Fixed-Point Standard-LMS Algorithm with Low Resource Utilization and Fast Convergence, (2010) International Journal on Computers and Software (IRECOS), 5 (4), pp.436-444.
- There are currently no refbacks.
Please send any question about this web site to firstname.lastname@example.org
Copyright © 2005-2021 Praise Worthy Prize