1836 IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 19, NO. 6, AUGUST 2011
when the source is located near the microphone centroid, but
employs a variable center-clipping threshold whose value is de-
rived based on the absolute values of the received microphone
signals in order to work better, when the source is located closer
to one of the microphones. The proposed approach achieves
better convergence performance for different source positions
in comparison to both NL-NLMS and XMNL-NLMS.
 Audio Signal Processing for Next-Generation Multimedia Communica-
tion Systems, Y. Huang and J. Benesty, Eds.. Norwell, MA: Kluwer,
 J. Benesty, M. M. Sondhi, and Y. Huang, Handbook of Speech Pro-
cessing. Secaucus, NJ: Springer-Verlag, 2008.
 J. Benesty, T. Gänsler, D. R. Morgan, M. M. Sondhi, and S. L. Gay
Advances in Network and Acoustic Echo Cancellation. New York:
 J. Benesty, D. R. Morgan, and M. M. Sondhi, “A better understanding
and an improved solution to the speciﬁc problems of stereophonic
acoustic echo cancellation,” IEEE Trans. Speech Audio Process., vol.
6, no. 2, pp. 156–165, Mar. 1998.
 M. M. Sondhi and D. R. Morgan, “Acoustic echo cancellation for
stereophonic teleconferencing,” in Proc. IEEE Workshop Applicat.
Signal Process. Audio Acoust., 1991, pp. 141–142.
 S. Shimauchi and S. Makino, “Stereo projection echo canceller with
true echo path estimation,” in Proc. IEEE Int. Conf. Acoust., Speech,
Signal Process., 1995, pp. 3059–3062.
 M. M. Sondhi, D. R. Morgan, and J. L. Hall, “Stereophonic acoustic
echo cancellation-An overview of the fundamental problem,” IEEE
Signal Process. Lett., vol. 2, no. 8, pp. 148–151, Aug. 1995.
 J. Benesty, D. R. Morgan, J. L. Hall, and M. M. Sondhi, “Stereophonic
acoustic echo cancellation using nonlinear transformations and comb
ﬁltering,” in Proc. IEEE Int. Conf. Acoust., Speech, Signal Process.,
1998, pp. 3673–3676.
 J. Benesty, D. R. Morgan, J. L. Hall, and M. M. Sondhi, “Synthesized
stereo combined with acoustic echo cancellation for desktop confer-
encing,” in Proc. IEEE Int. Conf. Acoust., Speech, Signal Process.,
1999, pp. 148–158.
 K. Mayyas, “Stereophonic acoustic echo cancellation using lattice or-
thogonalization,” IEEE Trans. Speech Audio Process., vol. 10, no. 7,
pp. 517–525, Oct. 2002.
 T. Gänsler and J. Benesty, “New insights into the stereophonic acoustic
echo cancellation problem and an adaptive nonlinearity solution,” IEEE
Trans. Speech Audio Process., vol. 10, no. 5, pp. 257–267, Jul. 2002.
 M. Ali, “Stereophonic acoustic echo cancellation system using time-
varying all-pass ﬁltering for signal decorrelation,” in Proc. IEEE Int.
Conf. Acoust., Speech, Signal Process., 1998, pp. 3689–3692.
 T. Tangsangiumvisai, J. A. Chambers, and A. G. Constantinides, “Time
varying all-pass ﬁlters using spectral-shaped noise for signal decorre-
lation in stereophonic acoustic echo cancellation,” in Proc. Int. Conf.
Digital Signal Process., 2002, pp. 87–92.
 J. Herre, H. Buchner, and W. Kellermann, “Acoustic echo cancellation
for surround sound using perceptually motivated convergence enhance-
ment,” in Proc. IEEE Int. Conf. Acoust., Speech, Signal Process., 2007,
 J. M. Valin, “Perceptually-motivated nonlinear channel decorrelation
for stereo acoustic echo cancellation,” in Proc. Hands-Free Speech
Commun. Microphone Arrays (HSCMA), 2008, pp. 188–191.
 S. Emura, Y. Haneda, A. Kataoka, and S. Makino, “Stereo echo cancel-
lation algorithm using adaptive update on the basis of enhanced input-
signal vector,” Signal Process., vol. 86, pp. 1157–1167, Jun. 2006.
 A. W. H. Khong and P. A. Naylor, “Stereophonic acoustic echo can-
cellation employing selective-tap adaptive algorithms,” IEEE Trans.
Speech Audio Process, vol. 14, no. 3, pp. 785–796, May 2006.
 M. Bekrani, A. W. H. Khong, and M. Lotﬁzad, “Neural network based
adaptive echo cancellation for stereophonic teleconferencing applica-
tion,” in Proc. Int. Conf. Multimedia Expo, 2010, pp. 1172–1177.
 M. Bekrani, M. Lotﬁzad, and A. W. H. Khong, “An efﬁcient quasi
LMS/newton adaptive algorithm for stereophonic acoustic echo can-
cellation,” in Proc. IEEE Asia Paciﬁc Conf. Circuits Syst., 2010.
 S. Haykin, Adaptive Filter Theory
. Englewood Cliffs, NJ: Prentice-
 D. R. Morgan, J. L. Hall, and J. Benesty, “Investigation of several types
of nonlinearities for use in stereo acoustic echo cancellation,” IEEE
Trans. Speech Audio Process., vol. 9, no. 6, pp. 686–696, Sep. 2001.
 J. B. Allen and D. A. Berkley, “Image method for efﬁciently simu-
lating small-room acoustics,” J. Acoust. Soc. Amer., vol. 65, no. 4, pp.
943–950, Apr. 1979.
 S. Attallah, “The wavelet transform-domain LMS adaptive ﬁlter with
partial subband-coefﬁcient updating,” IEEE Trans. Circuits Syst. II: Ex-
press Briefs, vol. 53, no. 1, pp. 8–12, Jan. 2006.
 K. Mayyas, “New transform-domain adaptive algorithms for acoustic
echo cancellation,” Digital Signal Process., vol. 13, no. 3, pp. 415–432,
Mehdi Bekrani was born in Gorgan, Iran, in 1979.
He received the B.Sc. degree from Ferdowsi Uni-
versity of Mashhad, Mashad, Iran, in 2002, and the
M.Sc. and Ph.D. degrees from Tarbiat Modares Uni-
versity, Tehran, Iran, in 2004 and 2010, respectively,
all in electrical engineering.
He is currently a Research Fellow at Nanyang
Technological University, Singapore. His current
research interests include acoustic signal processing
and their applications.
Andy W. H. Khong (M’06) received the B.Eng. de-
gree from Nanyang Technological University, Singa-
pore, in 2002 and the Ph.D. degree from the Depart-
ment of Electrical and Electronic Engineering, Im-
perial College London, London, U.K., in 2005. His
Ph.D. research was mainly on partial-update and se-
lective-tap adaptive algorithms with applications to
mono- and multi-channel acoustic echo cancellation
for hands-free telephony.
He is currently an Assistant Professor in the
School of Electrical and Electronic Engineering,
Nanyang Technological University, Singapore. Prior to that, he served as
a Research Associate in the Department of Electrical and Electronic En-
gineering, Imperial College London, from 2005 to 2008. His postdoctoral
research involved the development of signal processing algorithms for vehicle
destination inference as well as the design and implementation of acoustic
array and seismic fusion algorithms for perimeter security systems. He has also
published works on acoustic blind channel identiﬁcation and equalization for
speech dereverberation. His other research interests include human-computer
interfaces, source localization, speech enhancement, and blind deconvolution.
Mojtaba Lotﬁzad was born in Tehran, Iran, in 1955.
He received the B.S. degree in electrical engineering
from AmirKabir University of Technology, Tehran,
in 1980, and the M.S. and Ph.D. degrees from the
University of Wales, Cardiff, U.K., in 1985 and 1988,
He joined the Department of Electrical and
Computer Engineering, Tarbiat Modares University,
Tehran, Iran. He has also been a Consultant to sev-
eral industrial and governmental organizations. His
current research interests are in signal processing,
adaptive ﬁltering, speech processing, and specialized processors.