Adaptive wave eld synthesis for broadband active sound eld

reproduction: Signal processing

a)
Philippe-Aubert Gauthier
b
and Alain Berry
Groupe dAcoustique de lUniversit de Sherbrooke, Universit de Sherbrooke, 2500 boul. de lUniversit,
Sherbrooke, Qubec, Canada, J1K 2R1
Received 26 April 2007; revised 23 January 2008; accepted 24 January 2008
Sound eld reproduction is a physical approach to the reproduction of the natural spatial character
of hearing. It is also useful in experimental acoustics and psychoacoustics. Wave eld synthesis
WFS is a known open-loop technology which assumes that the reproduction environment is
anechoic. Areal reective reproduction space thus reduces the objective accuracy of WFS. Recently,
adaptive wave eld synthesis AWFS was dened as a combination of WFS and active
compensation. AWFS is based on the minimization of reproduction errors and on the penalization
of departure from the WFS solution. This paper focuses on signal processing for AWFS. A classical
adaptive algorithm is modied for AWFS: ltered-reference least-mean-square. This modied
algorithm and the classical equivalent leaky algorithm have similar convergence properties except
that the WFS solution inuences the adaptation rule of the modied algorithm. The paper also
introduces signal processing for independent radiation mode control of AWFS on the basis of plant
decoupling. Simulation results for AWFS are introduced for free-eld and reective spaces. The two
algorithms effectively reproduce the sound eld and compensate for the reproduction errors at the
error sensors. The independent radiation mode control allows a more exible tuning of the
algorithm. 2008 Acoustical Society of America. DOI: 10.1121/1.2875269
PACS numbers: 43.38.Md, 43.60.Tj, 43.50.Ki NX Pages: 20032016
I. INTRODUCTION
Following in the footsteps of the constantly evolving
digital signal processing and the relatively recent advent of
multichannel audio, spatial audio has gained more attention
in the past few decades from researchers and practitioners for
applications such as high-delity sound reproduction, music
reproduction, virtual reality display, interactive multisensory
environments, and spatial sound Camurri and Ferrentino,
1999; Wilson, 2001; AES Staff Writer, 2005; Woszczyk et
al., 2005. The interest for immersion and convincing multi-
sensory environments is not new and various techniques for
spatial audio have been introduced in the past Kleiner et al.,
1993; Kendall, 1995; Verheijen, 1997; Poletti, 2000; Rum-
sey, 2001; Davis, 2003.
Sound eld simulation as a spatial sound technique
attempts to reproduce the physical stimulus wave eld of
spatial hearing. Sound eld reproduction was investigated by
researchers in the past few decades Berkhout et al., 1993;
Nelson et al., 1997; Verheijen, 1997; Poletti, 2000; Choi and
Kim, 2004; Takane and Sone, 2004. Recently one of the
most active and related matters is room compensation Spors
et al., 2003; Gauthier et al., 2005a; Betlehem and Abhaya-
pala, 2005; Fuster et al., 2005; Gauthier and Berry, 2006,
which is essential for sound eld reproduction in a real re-
production space.
This paper deals with the problem of sound pressure
eld reproduction using adaptive digital signal processing
applied to adaptive wave eld synthesis AWFS initially
introduced by Gauthier et al. 2005b see also Gauthier and
Berry, 2006. The paper focuses on digital signal processing
realization for AWFS.
Within this introductory section, the problem of sound
pressure eld reproduction and a review of corresponding
technologies are presented. Section II A introduces WFS.
Section II B presents the ltered-reference least-mean-square
FXLMS realization of AWFS. Section II C presents AWFS
on the basis of independent radiation mode control including
the plant singular value decomposition SVD Gauthier et
al., 2005b; Gauthier and Berry, 2006. Appendices AC de-
tail some aspects of AWFS based on independent radiation
mode control. Digital signal processing simulation results are
reported in Sec. III for AWFS using the proposed algorithms.
Section IV discusses the results.
A. Sound eld reproduction
In sound eld simulation, the main objective can be gen-
erally stated as the aim to recreate a given physical quantity,
such as sound pressure, sound intensity Choi and Kim,
2004, spatial diffuseness Merimaa and Pulkki, 2005, etc.,
over a certain space. This can be achieved using a reproduc-
tion system including electroacoustical sources and receiv-
ers, signal processing, and the desired physical output de-
scription. In several cases, use of adaptive ltering implies
the minimization of a cost function which is representative of
the reproduction objective Gauthier and Berry, 2006. Adap-
tive signal processing for spatial sound reproduction has
a
Portions of this work were presented in Wave eld synthesis, adaptive
wave eld synthesis and ambisonics using decentralized transformed con-
trol: Potential applications to sound eld reproduction and active noise
control, 150th ASA meeting, Minneapolis, MN, October 2005.
b
Author to whom correspondence should be addressed. Electronic mail:
philippe_aubert_gauthier@hotmail.com.
J. Acoust. Soc. Am. 123 4, April 2008 2008 Acoustical Society of America 2003 0001-4966/2008/1234/2003/14/$23.00
been studied in various forms by researchers examples are:
Nelson, 1994; Asano and Swanson, 1995; Ise, 1999; Takane
et al., 1999; Radlovi et al., 2000; Santilln, 2001; Choi and
Kim, 2004; Gauthier et al., 2005a: Gauthier and Berry,
2006.
B. Wave eld synthesis WFS
Work on WFS originates from the theoretical investiga-
tions by Berkhout Berkhout et al., 1993; Verheijen, 1997. It
is assumed that the reproduction space is a free eld. From
the simple source formulation of the KirchhoffHelmholtz
integral theorem Williams, 1999, WFS operators are de-
ned to link a given simple virtual source typically creat-
ing spherical or plane waves in a planar listening area, fed
by a monophonic signal, to a loudspeaker array which repro-
duces the virtual sources acoustic eld also called the target
sound eld. The problem usually concerns reproduction in
the horizontal plane with a nite number of discrete sources
using appropriate simplications of the integral formulation
Verheijen, 1997. Some WFS studies have investigated spa-
tial aliasing Spors and Rabenstein, 2006; Corteel, 2006,
objective performance in-room situation Klehs and Sporer,
2003; Sporer and Klehs, 2004; Gauthier and Berry, 2007
and compensation of listening room acoustics Spors et al.,
2003; Fuster et al., 2005.
The benet of current WFS prototypes is their effective-
ness in creating a rich spatial impression over a broad area
surrounded by loudspeakers. One of the WFS drawbacks is
related to synthesis operators which rely on an important
assumption: the reproduction rooms response is not consid-
ered in the process except in specic WFS research Spors
et al., 2003. This typically introduces reproduction errors
that AWFS can possibly reduce Gauthier and Berry, 2007.
Active room compensation for WFS is an active research
topic for objective sound eld reproduction Elliott and Nel-
son, 1989; Asano and Swanson, 1995; Bouchard and Qued-
nau, 2000; Santilln, 2001; Spors et al., 2003; Gauthier et
al., 2005b; Fuster et al., 2005.
C. Adaptive wave eld synthesis AWFS
In a recent paper Gauthier and Berry, 2006, AWFS was
dened as a possible practical compromise between WFS
and active room compensation. AWFS is based on a cost
function to be minimized by the reproduction sources. The
cost function is dened from measurements of the repro-
duced sound eld using error sensors at a number of discrete
locations, and it is a quadratic function of: 1 the reproduc-
tion errors and 2 the adaptive lters departure from the
WFS lters. Using a singular value decomposition SVD of
the plant matrix frequency response functions between re-
production sources and error sensors Gauthier and Berry,
2006, it was shown that the underlying AWFS mechanism is
the independent control of radiation modes. The radiation
modes correspond to the left and right singular vectors Lan-
caster and Tismenetsky, 1985 from the plant SVD. This sug-
gested a decoupled implementation of AWFS which is prac-
tically developed in this paper from a signal processing
perspective.
II. SIGNAL PROCESSING FOR ADAPTIVE WAVE
FIELD SYNTHESIS
For a target sound eld produced by a virtual monopole
with amplitude A Pa m, az is dened as the z transform
of discrete-time sequences having the frequency response
functions
ae
j

= Ae
j
1
/r
1
e
j
2
/r
2
. . . e
j
m
/r
m
. . . e
j
M
/r
M

T
,
1
where
m
is the time delay which corresponds to r
m
/ c. The
time convention is e
jt
, c m/s is the sound phase velocity
and r
m
=x
m
m
x
o
represents the distance between the virtual
source in x
o
and the mth error sensor in the reproduction
space Fig. 1, x
l
l
is the reproduction source l position in
three-dimensional space x
1
l
, x
2
l
, . . . , x
L
l
, x is the position
of an arbitrary point in the reproduced wave eld, and x
m
m
is
the position of the mth error sensor x
1
m
, x
2
m
, . . . , x
M
m
.
A. Review of WFS operators for AWFS
According to Verheijen 1997, the following general
WFS operators are obtained in the frequency domain:
Qx
l
l
, e
j

=
jk
2
A cos
e
jkr
o

e
jT
s

r
o

r
ref
r
ref
+ r
o

2
and the reproduced sound eld in free space is
px, e
j
=

Qx
l
l
, e
j

e
jkr
r
dL, 3
where A is the target pressure eld amplitude virtual source
monopole amplitude, Qx
l
l
, e
j
is the driving function ex-
pressed as a monopole amplitude density for the reproduc-
tion sources along the line L, k=/ c is the wavenumber and
e
jkr
/ r with r=x
l
l
x is the complex sound pressure pro-
FIG. 1. Symbol convention for the WFS operators and lters denition. The
virtual source is located in x
o
. The reproduction source l is located in x
l
l
;
x
ref
describes points which belong to the reference line; x describes any
eld point and x
m
m
is the mth sensor position. L is the reproduction source
line, the virtual source is on the left of the source line and the reproduction
space is on the right of the source line. All sources and sensors are located
in the x
1
x
2
plane.
2004 J. Acoust. Soc. Am., Vol. 123, No. 4, April 2008 P. Gauthier and A. Berry: Adaptive wave eld synthesis
duced at x by a unitary amplitude monopole source in x
l
l
.
The exponent jT
s
has been articially added in Eq. 2
to add a modeling delay of samples. The sampling period
is T
s
s. It is assumed that the virtual source and the repro-
duction sources are both in the horizontal plane x
1
, x
2
. All
other variables are shown in Fig. 1.
The reference line is chosen as a circle centered at the
virtual source and passing through the center of the repro-
duction source array Verheijen, 1997; Gauthier and Berry,
2006.
Practical WFS implementation using discrete reproduc-
tion sources on a nite line replaces Eq. 3 by
px, e
j
=

l=1
L

s
Qx
l
l
, e
j

e
jkr
r
, 4
where
s
is the reproduction source separation distance m.
For WFS, a reproduction source is active i.e.
Qx
l
l
, e
j
0 if 90 where is shown in Fig. 1.
Truncation effects are reduced using a spatial weighting win-
dow which reduces the amplitude of the reproduction
sources at the end of the activated set. In this paper, half-
Hanning window with a half length of 1 m is applied to the
driving function Eq. 2 at the ends of the active set of
reproduction sources Gauthier and Berry, 2006.
B. FXLMS for AWFS
The classical FXLMS algorithm Elliott, 2001 is
adapted for the implementation of AWFS in a discrete-time
domain.
Figure 2 presents the generalized sound eld reproduc-
tion system for M reproduction points Nelson, 1994; Nelson
et al., 1997. In this case, the problem is to reproduce a set of
signals the desired signals d
M
n, n is a sample index at a
set of M positions in the reproduction space using L repro-
duction sources. The term wz is the adaptive control lter
vector and Gz the acoustical plant responses z transform
between reproduction sources and error sensors. Here, xn
is the reference signal that is used to dene a given simple
source eld prescribed in space and time domains through
the target vector az and d
M
n d
M
z =azz

xz is the
target wave eld at the M error sensors, and is the mod-
eling delay.
The system is characterized by one reference signal and
L FIR nite impulse response control lters each including
I coefcients w
li
. The multichannel reproduction system
can thus be described by the reproduction error vector capi-
talized subscript pairs denote matrix dimensions
en
M1
= dn
M1
Rn
MLI
wn
LI1
, 5
with the error vector structured as
en = e
1
n . . . e
M
n
T
. 6
The desired signals are
dn = d
1
n . . . d
M
n
T
. 7
The control lter vector is dened by
wn = w
0
T
n . . . w
I1
T
n
T
, 8
w
i
n = w
1i
n . . . w
Li
n
T
9
and the matrix of ltered reference is
Rn =

r
1
T
n r
1
T
n 1 . . . r
1
T
n I + 1
r
2
T
n r
2
T
n 1 . . . r
2
T
n I + 1
] ] ]
r
M
T
n r
M
T
n 1 . . . r
M
T
n I + 1

, 10
where a ltered reference vector was introduced as
r
m
n = r
1m
n . . . r
Lm
n
T
. 11
The ltered reference r
lm
n corresponds to the reference sig-
nal convolved with the plant impulse response from repro-
duction source l to error sensor m
r
lm
n =

j=0
J1
G
lmj
xn j, 12
where the J-coefcient plant impulse responses are stocked
G
LMJ
.
The general cost function for multichannel AWFS is
Gauthier and Berry, 2006
J
AWFS
= Ee
T
nen + w
T
nwn
+ wn w
WFS

T
wn w
WFS
, 13
where the rst term represents the summation of the qua-
dratic reproduction errors, the second term represents a con-
trol effort penalty Elliott, 2001, the third term corresponds
to a quadratic penalty for the coefcients wn deviation
from w
WFS
the time-domain WFS operators. The math-
ematical expectation is denoted E. The time-domain WFS
operators are obtained from inverse discrete-time Fourier
transform of the WFS operator Qx
1
l
, e
j
dened in Eq. 2.
In Eq. 13 the WFS operators w
WFS
have been arranged
similarly to the control lter coefcient vector in Eq. 8.
The stochastic gradient, using instantaneous values of
the gradient of Eq. 13 is
J
AWFS
w
= 2R
T
nRn + + Iwn 2R
T
ndn
2w
WFS
. 14
Using a classical gradient algorithm Elliott, 2001 wn
+1 =wn J
AWFS
/ w, a multichannel recursive algo-
rithm is obtained
FIG. 2. Generalized sound eld reproduction using adaptive ltering. Capi-
talized subscripts denote signal dimensions.
J. Acoust. Soc. Am., Vol. 123, No. 4, April 2008 P. Gauthier and A. Berry: Adaptive wave eld synthesis 2005
wn + 1 = 1 + wn + w
WFS
+ R
T
nen,
15
which corresponds to a modied version of the multichannel
leaky LMS algorithm Elliott, 2001 where , 1+,
w
WFS
, and R
T
nen are, respectively, the convergence
coefcient =2, the leakage term, the WFS operators co-
efcients w
WFS
re-injection in the solution and the adaptive
lter update according to the instantaneous errors. The WFS
departure term in Eq. 13 introduces a leakage term in the
lter update Eq. 15 plus a bias which is proportional to
w
WFS
w
WFS
. In Eq. 15, the matrix R
T
n has been re-
placed by R
T
n which uses a plant model to compute the
estimated ltered reference.
Analysis of convergence of FXLMS applied to AWFS is
detailed in the remainder of this section. Using these results,
one may evaluate the convergence behavior of the multi-
channel AWFS in comparison with standard FXLMS or
leaky FXLMS.
Equation 15 is rewritten using the error vector deni-
tion of Eq. 5
wn + 1 = wn + w
WFS
+ R

ndn
R
T
nRn + + Iwn, 16
which allows the derivation of the asymptotic lter coef-
cients w

=lim
n
Ewn
w

=
ER
T
ndn + w
WFS
ER
T
nRn + + I
, 17
where it is assumed that Ew

=w

and Ew
WFS
=w
WFS
since w
WFS
is constant for a xed virtual source position.
Equation 17 shows that the reproduction lters after con-
vergence of the adaptation algorithm are partly dened by
the WFS operators to an extent determined by and partly
by the leaky LMS error attenuation.
Using Eq. 17, and Eq. 15,
Ewn + 1 w

= I ER
T
nRn
+ + IEwn w

. 18
The eigenvalue decomposition of the matrix
ER
T
nRn ++I is introduced
ER
T
nRn + + I = QTQ
1
, 19
where T is a diagonal matrix of eigenvalues and Q is the
matrix of the eigenvectors. The transformed adaptive lter is
tn = Q
1
Ewn w

. 20
One can rewrite Eq. 18 as a set of uncoupled update equa-
tions for the various convergence modes since T is diagonal
tn + 1 = I Ttn. 21
Each convergence mode will converge provided that 01
T
i
1 with T
i
being the ith eigenvalue of the decompo-
sition introduced in Eq. 19. This allows the following cri-
terion for the convergence factor which would provide stable
convergence Morgan, 1980; Elliott, 2001
0 2
RT
i

T
i

2
, 22
where R indicates the real part of the argument. This is a
typical FXLMS result for which convergence properties in
relation with plant responses are known Elliott, 2001, thus
supporting that modied FXLMS for AWFS has the same
convergence properties as classical FXLMS. Since the penal-
ization terms +I in Eq. 19 form a diagonal matrix,
they have the effect of stabilizing standard FXLMS algo-
rithm by compensating for potentially small or negative ei-
genvalues T
i
Elliott, 2001, where T
i
is the ith eigenvalue
of ER
T
nRn. This effect can be shown by rewriting Eq.
22 using the eigenvalue decomposition of ER
T
nRn as
QTQ
1
0 2
RT
i
+ +
T
i
+ +
2
. 23
Both the control effort penalization and the WFS penal-
ization have the same effect in terms of convergence con-
dition and may both equally stabilize a system by compen-
sating for small negative eigenvalues T
i
.
Although sound eld reproduction using AWFS as de-
ned by the adaptation algorithm of Eq. 15 is readily ap-
plicable, some well-known disadvantages of multichannel
FXLMS also apply to AWFS. These are related to the slow
FXLMS convergence properties and computational load as-
sociated with an increasing number of lter coefcients I and
reproduction sources L. Most of the slow convergence prop-
erties of the multichannel FXLMS are related to the cross-
coupling paths and the non-at frequency responses of the
electroacoustical multichannel plant between the sources and
the sensors Elliott, 2001; Bai and Elliott, 2004. Since a
sound eld reproduction system like a AWFS system implies
many reproduction sources, this might limit the application
of FXLMS for AWFS. With this in mind, it would be inter-
esting to transform the multichannel problem in a set of in-
dependent single-channel systems. This is possible on the
basis of independent radiation mode control Gauthier and
Berry, 2006 and is related to the principal-component LMS
PC-LMS algorithm Cabell and Fuller, 1999; Elliott, 2000;
Bai and Elliott, 2004.
C. AWFS with independent radiation mode control
This section presents the system preconditioning for in-
dependent radiation mode control using parallel single-
channel adaptive systems as already suggested in a previous
paper Gauthier and Berry, 2006. Independent radiation
mode control is based on singular value decomposition
SVD of the plant matrix Ge
j
. Further details are pro-
vided in Appendix A.
The control architecture is shown in Fig. 3. The upper
branch is the target wave eld denition. The second branch
represents the xed WFS contribution, obtained by the WFS
operators projection on the null space of Ge
j
. This part is
2006 J. Acoust. Soc. Am., Vol. 123, No. 4, April 2008 P. Gauthier and A. Berry: Adaptive wave eld synthesis
non-adaptive. Note that the corresponding two dashed blocks
in this gure are combined in a single operation with an
additional delay of
SVD
samples see Appendix B. The
third branch includes the independent adaptive controllers
w
m
, the rst synthesis lter matrix
s
Gz, the reproduction
source signals y
L
n, and the plant response Gz to produce
the reproduced sound pressures p
M
n. The bottom branch
includes the uncoupled plant model z
2
SVD
, the independent
single-channel LMS gradient algorithms and the analysis l-
ter matrix
a
Gz to transform the reproduction errors into
the pressure mode basis. Details about the synthesis and
analysis lters are given in Appendices AC.
The independent LMS algorithms operate on the decou-
pled plant. This means that each single-channel independent
LMS algorithm controls a single radiation mode obtained
from the SVD of Ge
j
.
Several preconditioning methods have been published
for feedforward control Cabell and Fuller, 1999; Cabell et
al., 2001; Elliott, 2001, 2000; Bai and Elliott, 2004. Precon-
ditioning using plant decoupling has been treated using prin-
cipal component LMS PC-LMS algorithm, based on SVD,
for tonal disturbances Cabell and Fuller, 1999 or, equiva-
lently, SVD controller for tonal disturbances Elliott, 2000.
SVD decoupling has been recently extended for broadband
signals using a decoupling lter formulation in the frequency
domain and inverse transformation towards the time domain
for two-channel cross-talk cancelation Bai and Elliott,
2004. This paper includes considerations for the null space
of the plant response which has proven to be of importance
for AWFS Gauthier and Berry, 2006 and for the realization
of singular vectors reordering and modication to reduce
the time-domain leakage in the decoupling lters synthesis
and analysis lters. Moreover, the AWFS approach implies
the projection of the WFS operators on the source mode
basis. This differs from Bais work which was also devoted
to a different application. All these special considerations are
described in the Appendices to lighten the main content of
the paper.
For the sake of brevity, the following developments are
presented for system uncoupling and whitening. Minor
modications are needed when the system is only decoupled.
Since the control system shown in Fig. 3 can be considered
as a set of independent single-channel radiation-mode con-
trollers, for the mth independent radiation mode controller
the instantaneous reproduction error is
e
m
n = d

m
n p
m
n, 24
where d

m
n is the mth element of dn passed through the
analysis lter matrix
a
Gz and p
m
n is the mth element of
pn passed through the analysis lter matrix. This equation
can be rewritten as follows:
e
m
n = d

m
n w
m
T
nxn 2
SVD
, 25
where w
m
n is the I-coefcient vector associated with the
control lter for the mth radiation mode and the reference
signal vector is xn =xnxn1 . . . xnI 1
T
. In this
case the cost function associated with the mth radiation mode
is expressed as
J

m
= Ee
m
2
n +
m
w
m
n w
WFS
m

T
w
m
n w
WFS
m
,
26
where w
WFS
m
is the I-coefcient vector of the WFS operators
projected on the mth rst synthesis lter see Appendix C.
The gradient of this cost function is
J

m
w
m
= 2r
xd

m
2
SVD
+ 2R
xx
0w
m
n + 2
m
w
m
n
2
m
w
WFS
m
27
with r
xd

m
t =Ed

m
n+txn and R
xx
t =Exn+tx
T
n.
Using Eq. 25, the instantaneous value of the gradient be-
comes
J

m
w
m
= 2e
m
n + 2
SVD
xn + 2
m
w
m
n 2
m
w
WFS
m
.
28
A single-channel LMS algorithm w
m
n+1 =w
m
n

m
/ 2 J

m
/ w
m
with a convergence coefcient
m
for the
mth controller gives
w
m
n + 1 = 1
m

m
w
m
n +
m
e
m
n + 2
SVD
xn
+
m

m
w
WFS
m
. 29
The adaptation rule is, however, based on future values of
the reproduction error e
m
n+2
SVD
so that a modication of
this algorithm is introduced
w
m
n + 1 = 1
m

m
w
m
n +
m
e
m
nxn 2
SVD

+
m

m
w
WFS
m
, 30
where xn2
SVD
represents the ltered reference in the
SVD-transformed domain which includes decoupling and
whitening. This last equation can be implemented for each
radiation mode controller for practical applications of AWFS
based on plant decoupling. If plant whitening is excluded, it
is still possible to use the last equation as an adaptation al-
gorithm since xn2
SVD
would then correspond to a de-
FIG. 3. Block diagram of the AWFS least-mean-square adaptive digital
processing implementation based on independent radiation mode control.
Subscripts denotes signal dimensions.
J. Acoust. Soc. Am., Vol. 123, No. 4, April 2008 P. Gauthier and A. Berry: Adaptive wave eld synthesis 2007
layed version of the reference signals. This implementation
is related to the DXLMS delayed-reference LMS algorithm
Ahn and Voltz, 1996. More than simplifying the computa-
tion by replacing a multichannel implementation by a set of
independent controllers, the amount of operation is also re-
duced since the ltered reference is now replaced by a simple
delay operation xn2
SVD
in Eq. 30. In the FXLMS
algorithm Eq. 15, the computation of the ltered refer-
ence R
T
n involved a multichannel convolution. This is an
advantage of AWFS based on independent radiation mode
control.
Denoting w
m

=lim
n
Ew
m
n and nding the math-
ematical expectation of Eq. 30, one nds
w
m

=
r
xd

m
2
SVD
+
m
w
WFS
m
R
xx
0 +
m
I
=
Ed

m
nxn 2
SVD
+
m
w
WFS
m
Exn 2
SVD
x
T
n 2
SVD
+
m
I
. 31
The structure of Eq. 31 is very similar to the one obtained
for the multichannel FXLMS case Eq. 17, except that it
applies independently to each radiation mode. Such imple-
mentation therefore allows for the independent tuning of the
convergence properties of each radiation mode. This is the
second advantage of AWFS based on independent radiation
mode control.
III. SIMULATIONS RESULTS
In order to demonstrate the feasibility of an AWFS sys-
tem based on the adaptive signal processing described in this
paper, numerical simulations were performed.
A. Free eld
The system shown in Fig. 4 was simulated using a
simple free-eld model: The reproduction sources were rep-
resented by monopoles with a radiated sound pressure eld
e
jkr
/ r where rm is the distance from the reproduction
source and k the wavenumber. There are four error sensors
and 24 reproduction sources. The four error sensors are sepa-
rated by 0.175 m along x
1
and x
2
. The source array has a
diameter of 2 m and the source separation is 0.2611 m. The
impulse responses from the reproduction sources to the error
sensors were then computed using an inverse fast Fourier
transform IFFT of the frequency-domain response func-
tions. Since the function e
jkr
/ r generally involves fractional
time delays Laakso et al., 1996, Hanning windows were
applied in the time domain centered on the main impulsion
of the response to reduce time aliasing. For the free-eld
simulations, the circular group of monitor sensors shown in
Fig. 4 is used. The sampling frequency is 1200 Hz.
Results of time-domain AWFS simulations for this case
are shown in Figs. 57. Appendix B presents the synthesis
and analysis lters. The results, shown in Fig. 5, are de-
scribed in terms of the E
LS
performance criterion dened by
E
LS
n = en
T
en 32
and corresponds to the sum of the squared reproduction er-
rors either at the error microphones top of Fig. 5 or at the
monitoring microphones bottom of Fig. 5. In Fig. 5, E
LS
is
also smoothed with a moving-average lter to enhance the
gure readability.
12
11
13
10
14
9
15
8
16
7
17
x
2
6
18
5
19
4
20
3
21
2
22
1
23
24
x
1
Sources
Error sensors
Monitor sensors
Monitor sensors
Virtual source
1

2
3

4
FIG. 4. Conguration of the reproduction source, error sensor and monitor
sensor arrays. The error sensors numbers are identied with arrows. Two
different groups of monitor sensors are used: a linear array of eight monitors
or a circular array of 24 monitors. : virtual source in x
o
=0, 4, 0 m.
0 2 4 6 8 10
10
-4
10
-3
10
-2
10
-1
10
0
E
L
S
(
n
)
0 2 4 6 8 10
10
-2
10
-1
Time [s]
E
L
S
(
n
)
(
m
o
n
i
t
o
r
s
)
WFS
AWFS, FXLMS, =0
AWFS, FXLMS, =10
AWFS, SVD
WFS
AWFS, FXLMS, =0
AWFS, SVD
AWFS, FXLMS, =10
FIG. 5. Learning curves for reproduction errors at the error sensors and at
the monitor sensors using FXLMS and independent radiation mode control
algorithms in free eld.
2008 J. Acoust. Soc. Am., Vol. 123, No. 4, April 2008 P. Gauthier and A. Berry: Adaptive wave eld synthesis
The xed WFS solution is obtained by setting the con-
vergence coefcient =0 and the penalization parameter
=0 in Eq. 15. Since =0, the control lters then coincide
with the WFS solution which is used as the initial value for
the control lters. We used =0.00004 and =0 for the rst
FXLMS case while =0.00004 and =10 were used for the
second FXLMS case. As for the AWFS algorithm based on
independent radiation mode control, each radiation mode is
associated with its own values of the convergence coef-
cients
1
=0.0002,
2
=
3
=0.001,
4
=0.004 and penaliza-
tion parameters
1
=
2
=
3
=
4
=1. In all cases, the size of
the control lters was set to 256 and was set to 0. The
convergence coefcients were chosen to approach the maxi-
mum value for stable convergence. A modeling delay of 16
samples, =16, was used in these simulations. The SVD
delay was set to 128,
SVD
=128. The rst and second syn-
thesis lters and the analysis lters all include 256 coef-
cients.
Figure 5 shows that the different algorithms effectively
reduce the reproduction error in comparison with WFS. The
independent radiation mode control implementation implies
a delayed adaptation caused by the supplementary delay in
the synthesis and analysis lters a total of 2
SVD
samples
delay. Although the FXLMS algorithm can reach a similar
residual criterion as the AWFS algorithm based on indepen-
dent radiation mode control, the independent implementation
allows for ne tuning of the convergence properties of each
mode, which is not possible with the FXLMS algorithm.
This is highlighted in Fig. 6, which shows the 24 control
lters after convergence for the FXLMS and SVD-based al-
gorithms. Note that for the SVD-based algorithm, these con-
trol lters correspond to the multiplication of the adaptive
lter matrix M256, with M=4 with the rst synthesis
lter matrix
s
GzLM256 along with the addition of
the null space synthesis lters matrix
s
GzL256.
Clearly, both algorithms result in a similar set of control
lters. However, for the SVD-based algorithm, the presence
of the higher-order SVD radiation modes is visible from
sources 112 where the spatial variation of the control lters
with reproduction source number is more pronounced
Gauthier and Berry, 2006. This is also shown in Fig. 7
where the total energies quadratic sums of all the coef-
cients of the control lters are shown for each reproduction
source. This more pronounced variation for the reproduction
sources 112 is due to the faster convergence of higher-order
radiation modes in independent radiation mode control, be-
cause it is based on modal convergence coefcients
m
and
penalization
m
. With the FXLMS algorithm, these radiation
modes are not allowed to converge in the given adaptation
time. Indeed, see Appendix B, the higher-order radiation
modes correspond to more pronounced spatial variations of
the corresponding rst synthesis lters.
B. Semi-innite space delimited by a rigid oor
and two hard walls
To investigate the effectiveness of the AWFS algorithms
for active sound eld reproduction in non-anechoic environ-
ments, other simulations were performed where the acousti-
cal responses included three rigid surfaces. The sources and
the sensors are 1.22 m above a rigid oor. Two perpendicular
walls are also included as planes located at x
2
=2 m and
x
1
=6 m. The conguration is shown in Fig. 8. For the
simulation results presented here, the linear array of moni-
toring sensors shown in Fig. 4 is used.
Although this reproduction space is simpler than a real
listening room, it provides enough departures with respect to
the free eld situation to investigate AWFS in a reective
reproduction space.
1
6
12
18
24 1
64
128
192
256
-0.02
0
0.02
Samples
Source no
1
6
12
18
24 256
320
384
448
512
-0.02
0
0.02
Samples
Source no
A
m
p
.
A
m
p
.
FIG. 6. Examples of the L resulting control lters for AWFS by FXLMS
top and by independent radiation mode control bottom in free eld.
1 4 8 12 16 20 24
10
-6
10
-5
10
-4
10
-3
10
-2
Reproduction source no
E
n
e
r
g
i
e
s
o
f
t
h
e
c
o
n
t
r
o
l
f
i
l
t
e
r
s
AWFS by FXLMS
AWFS with SVD
FIG. 7. Total energies of the L control lters shown in Fig. 6.
x
3
x
1
x
2
Infinite rigid wall (x
1
=6)
Infinite rigid wall (x
2
=-2)
Infinite rigid floor (x
3
=0)
Reproduction
source array
FIG. 8. Circular reproduction source array in three-dimensional semi-
innite space.
J. Acoust. Soc. Am., Vol. 123, No. 4, April 2008 P. Gauthier and A. Berry: Adaptive wave eld synthesis 2009
Appendix B presents the rst synthesis lters
s
Gz
and analysis lters
a
Gz each of which includes 256 coef-
cients. A modeling delay of 16 samples, =16, was used in
these simulations.
A comparison of WFS and AWFS is reported in Fig. 9 in
terms of the learning curves for the various algorithms. The
following values were used: =0 and =0 for WFS. We
used =0.00004 and =0 for the rst FXLMS case while
=0.00004 and =10 were used for the second FXLMS
case. As for the AWFS algorithm based on independent ra-
diation mode control, the convergence and penalization pa-
rameters were adjusted for each individual radiation mode:

1
=0.00015,
2
=
3
=0.0004,
4
=0.003 and
1
=1,
2
=
3
=0.1,
4
=0.01. The adaptation coefcients were chosen to
approach the maximum value for stable convergence. It is
possible to reduce the corresponding penalization parameter
so that the convergence of the higher-order modes is not
blocked by excessively large penalization parameters. In all
cases, the size of the control lters was 256 and was set to
0. According to the learning curves of Fig. 9, it is clear that
AWFS provides a reduction of the reproduction error both at
the error and monitor sensors when compared to WFS.
The effects of the reective surfaces on sound eld re-
production in space are shown in Figs. 1012, which present
the reproduced impulse responses instantaneous sound pres-
sure at the monitoring sensors for the linear monitor array
shown in Fig. 4. For the WFS solution, one can clearly dis-
tinguish the undesirable reections reaching the monitor ar-
ray in Fig. 10. The FXLMS case, Fig. 11, is obtained for a
penalization parameter =10. In this gure, one can note the
reduction of the reections. This reduction of room effects is
most notable for the oor and late reections. The attenua-
tion of the room effects is noticeable at the central monitor-
ing sensors this region corresponds to the control region
error sensors position, that is monitor sensors Nos. 4 and
5. The comparison between Figs. 11 and 12 shows how the
AWFS algorithm based on independent control effectively
reduces the undesirable reections at the monitoring sensors.
1
2
3
4
5
6
7
8
1
16
32
48
64
80
-0.1
0
0.1
0.2
Monitor no Samples
I
n
s
t
a
n
t
a
n
e
o
u
s
s
o
u
n
d
p
r
e
s
s
u
r
e
[
P
a
]
Direct wave fronts Walls and floor
Floor
First wall
and floor
FIG. 10. Reproduced thin lines and virtual thick dashed lines impulse
responses at the monitor sensor array linear array shown in Fig. 4 for WFS
using the system with a rigid oor and two hard walls.
1
2
3
4
5
6
7
8
1
16
32
48
64
80
-0.1
0
0.1
0.2
Monitor no Samples
I
n
s
t
a
n
t
a
n
e
o
u
s
s
o
u
n
d
p
r
e
s
s
u
r
e
[
P
a
]
FIG. 11. Reproduced thin lines and virtual thick dashed lines impulse
responses at the monitor sensor array linear array shown in Fig. 4 for
AWFS by FXLMS using the system with a rigid oor and two hard walls.
1
2
3
4
5
6
7
8
257
272
288
304
320
336
-0.1
0
0.1
0.2
Monitor no Samples
I
n
s
t
a
n
t
a
n
e
o
u
s
s
o
u
n
d
p
r
e
s
s
u
r
e
[
P
a
]
FIG. 12. Reproduced thin lines and virtual thick dashed lines impulse
responses at the monitor sensor array linear array shown in Fig. 4 for
AWFS based on independent radiation mode control using the system with a
rigid oor and two hard walls.
0 2 4 6 8 10
10
-4
10
-3
10
-2
10
-1
10
0
E
L
S
(
n
)
0 2 4 6 8 10
10
-2
10
-1
Time [s]
E
L
S
(
n
)
(
m
o
n
i
t
o
r
s
)
WFS
AWFS, FXLMS, =0
AWFS, SVD
AWFS, FXLMS, =10
WFS
AWFS, FXLMS, =0
AWFS, SVD
AWFS, FXLMS, =10
FIG. 9. Learning curves for reproduction errors at the error sensors and at
the monitor sensors using FXLMS and independent radiation mode control
algorithms using the system with a rigid oor and two hard walls.
2010 J. Acoust. Soc. Am., Vol. 123, No. 4, April 2008 P. Gauthier and A. Berry: Adaptive wave eld synthesis
IV. CONCLUSION AND PERSPECTIVES
Adaptive wave eld synthesis, as introduced by the au-
thors Gauthier and Berry, 2006, was originally proposed as
a solution for active sound eld reproduction and active
room compensation for WFS. AWFS is based on minimiza-
tion of the reproduction errors with a departure penalty from
the WFS solution. Such penalization is what makes the
AWFS original. This paper furthered the understanding and
denition of AWFS by analyzing broadband signal process-
ing and implementation on the basis of two algorithms:
modied FXLMS and independent radiation mode control
based on plant decoupling by singular value decomposition.
It was shown theoretically that the application of classi-
cal adaptive algorithm for active noise control such as
FXLMS provides a practical implementation frame for
AWFS. The convergence properties of the modied FXLMS
algorithm are very similar to the properties of the classical
leaky FXLMS algorithm. The main difference between this
classical algorithm and its modied counterpart for AWFS is
the constant reintroduction of a scaled part of the WFS solu-
tion at each sample in the adaptation rule. Plant SVD and
AWFS based on independent radiation mode control were
then introduced. The corresponding adaptation equation
showed that it is possible to control independently the con-
vergence properties of each of the radiation modes. This is a
major advantage of plant decoupling for AWFS along with
reduction of the computational burden partly caused by the
simplication of the adaptation rules and of the computation
of the ltered reference signals. In comparison with other
published results on plant preconditioning by SVD Bai and
Elliott, 2004, further practical considerations for rst syn-
thesis and analysis lters were presented in the Appendices.
It was shown that either a reordering algorithm or a sign
change tracking algorithm both working in the frequency
domain are needed before any tractable time-domain imple-
mentation of AWFS can be achieved. Moreover, special at-
tention was given to the practical null space denition which
is of great importance for AWFS.
Simulations results for free eld and in a semi-innite
space bounded by three rigid surfaces were presented. The
rst case was introduced to validate the AWFS algorithms in
a simple situation. It proved the workability of the algo-
rithms. The second case was introduced as a demonstration
of the interest of AWFS: active sound eld reproduction and
active room compensation. A simple semi-innite space was
chosen as an example where identiable discrete reected
wave fronts impinge on the sound eld reproduction system.
In this example, it was demonstrated that in-plane AWFS can
effectively be used to compensate for undesirable room ef-
fects such as discrete reections in the vicinity of the error
sensors. These included reections from vertical and hori-
zontal surfaces. It is therefore possible to compensate for
reections coming from horizontal surfaces with a planar
horizontal system using AWFS and the presented congura-
tion. Recent experiments with AWFS support these observa-
tions.
From all these remarks and from the detailed results
presented in Sec. III, we conclude that AWFS can be used for
active sound eld reproduction and active room compensa-
tion according to the original AWFS denition Gauthier and
Berry, 2006. An experimental AWFS system using the con-
guration depicted in Fig. 4 was recently evaluated in sev-
eral environments: hemi-anechoic chamber, laboratory space,
and reverberation chamber. Experimental evaluations of
physical AWFS performance for active sound eld reproduc-
tion is thus a topic of current research.
As for most recent WFS systems, the conguration re-
tained here was a two-dimensional distribution of reproduc-
tion sources. Simplicity and lower cost are the usual reasons
behind this choice. Such a conguration was selected for
illustration purposes and because it corresponds to our ex-
perimental system not presented in this paper. In all cases,
the presented algorithms and AWFS concepts are not limited
to a given specic conguration. As an example, use of com-
pact sensor arrays can easily be extended to three dimen-
sions. Another example is a three-dimensional reproduction
source array.
Recent experiments with AWFS in real reproduction
spaces also conrmed the importance of the higher-order ra-
diation modes for the enlargement of the effective control
region. This is only possible with the independent radiation
mode control algorithm. Experiments also showed an even
better performance of AWFS in several reproduction envi-
ronments.
Other issues related to AWFS are open questions:evalu-
ation of system size on performance, possible use of several
compact error sensor arrays in a larger distributed reproduc-
tion region, effects of modeling delay, supplementary consid-
erations for causality, modication of other classical adaptive
algorithms frequency domain implementation, sparse adap-
tation and others Garas, 1999; Elliott, 2000, or various
types of penalization for AWFS and both subjective and
objective evaluations of AWFS performance in several repro-
duction rooms.
ACKNOWLEDGMENTS
This work has been supported by NSERC Natural Sci-
ences and Engineering Research Council of Canada,
NATEQ Fond Qubecois de la Recherche sur la Nature et
les Technologies, VRQ Valorisation Recherche Qubec
and Universit de Sherbrooke. This research has been con-
ducted in collaboration with CIRMMT Centre for Interdis-
ciplinary Research in Music Media and Technology, McGill
University.
APPENDIX A: SYSTEM DECOUPLING AND
WHITENING USING SINGULAR VALUE
DECOMPOSITION SVD IN THE FREQUENCY
DOMAIN
The electroacoustical plant Gz is represented in the
frequency domain by Ge
j
=Gz
z=e
j where j =

1. Us-
ing the SVD, the plant matrix can be expressed as
Ge
j

ML
= Ue
j

MM
e
j

ML
V
H
e
j

LL
A1
with
J. Acoust. Soc. Am., Vol. 123, No. 4, April 2008 P. Gauthier and A. Berry: Adaptive wave eld synthesis 2011
e
j
=

1
e
j
0
] ] 0
MLM
0
M
e
j

ML
,
A2
where U is the column matrix of the left singular vectors, V
is the column matrix of the right singular vectors, e
j
is a
pseudo-diagonal matrix of the real singular values
i
e
j

ordered on the main diagonal by decreasing amplitude. The

left singular vectors are dened as pressure modes i.e., ra-
diation modes evaluated at the error sensor array and the
right singular vectors are dened as source modes i.e., ra-
diation modes evaluated at the reproduction source array.
Superscript
H
denotes Hermitian transposition. In Eq. A2,
the plant matrix rank r was assumed to be M for notation
convenience and we assume LM.
The rst radiation mode synthesis lter LM matrix
s
Ge
j
is dened by
s
Ge
j

LM
= Ve
j

LL
e
j

LM
+
, A3
where the pseudo inversion
+
of the pseudo-diagonal ma-
trix is dened as follows:

+
e
j
=

1/
1
e
j
0
] ] 0
MML
0 1/
M
e
j

T
.
A4
The rst synthesis lters
s
Ge
j
are used to dene repro-
duction source inputs in the frequency domain that corre-
sponds to the source modes not belonging to the null space.
Each column of
s
Ge
j
is a set of L synthesis lters that
creates one of the source modes. According to this denition,
one notes that: 1 the
s
Ge
j
columns cover the range of
the matrix Ge
j
by its multiplication with the plant FRF
matrix Ge
j
and 2 the rst radiation mode synthesis lter
matrix
s
Ge
j
removes from Ve
j
the null space of
Ge
j
by post-multiplication by
+
e
j
in Eq. A3. Alter-
natively, the 1/
i
can be replaced by 1 on the main diagonal
of Eq. A4 so that the plant is decoupled but not whitened.
In this specic case, the same terminology and notation, for
the synthesis lters, are used. For the simulations reported in
this paper, the plant is only decoupled, but not whitened.
An analysis lter matrix
a
Ge
j

MM
is dened from
the SVD
a
Ge
j

MM
= U
H
e
j

MM
, A5
The analysis lter matrix transforms the physical sound pres-
sures p
M
n or e
M
n in pressure mode basis giving p
M
n
or e
M
n. Each column of
a
Ge
j
is a set of M analysis
lters that create a pressure mode. The rst step of plant
decoupling is the multiplication of the plant matrix by the
rst synthesis lter matrix
Ge
j

s
Ge
j
= Ue
j
e
j
V
H
e
j
Ve
j

+
e
j

= Ue
j
. A6
The second step is the pre-multiplication of Eq. A6 by the
analysis lter matrix
a
Ge
j
Ge
j

s
Ge
j
= Ie
j
, A7
since Ve
j
and Ue
j
are unitary matrices. I is the identity
matrix. Equation A7 allows plant decoupling and pre-
whitening by pre- and post-multiplication with synthesis and
analysis lters. This property is of fundamental importance
for practical implementation of AWFS based on independent
radiation mode control Gauthier and Berry, 2006. If, as
noted earlier, the plant is only decoupled, one has:
a
Ge
j
Ge
j

s
Ge
j
=e
j
, where e
j
is the rst
MM submatrix of e
j
.
The second radiation mode synthesis lter matrix
s
Ge
j
producing the null space of Ge
j
is
s
Ge
j

L1
= Ve
j

LL
i
L1
, A8
where i is a block vector dened by i
L1
=0
1M
0
1LM

T
. This lter matrix produces all the ra-
diation modes which belong to the null space of the plant
matrix Ge
j
so that, according to the null space denition
Lancaster and Tismenetsky, 1985, Ge
j

s
Ge
j
=0
M1
.
APPENDIX B: TIME-DOMAIN FILTERS FROM SVD
IN THE FREQUENCY DOMAIN
The procedure used to generate time-domain decoupling
lters from the frequency response matrices
s
Ge
j
and
a
Ge
j
has been described by Bai and Elliott 2004. For
AWFS, a similar procedure must also be applied to
s
Ge
j
. Other procedures such as WFS projection on the
SVD basis and radiation mode reordering are also original
aspects.
The frequency response matrices, temporarily denoted
Fe
j
, are calculated for simulations or measured for
practical implementations. Using inverse discrete-time Fou-
rier transform DTFT, a discrete-time sequence matrix Fn
is obtained: Fe
j

=2/Nk
Fn, where k represents a fre-
quency index and N the number of samples in the frequency
domain. A time delay of
SVD
samples is then added for
causality purposes. Fn is then transformed to produce the
lter coefcient matrix Fz.
Although this procedure is readily applicable to derive
the synthesis and analysis lters, the SVD decoupling being
independently computed at each frequency, the complex fre-
quency response functions may not be directly passed into
inverse DTFT to create time-domain lters without risk. The
limitation comes from the SVD denition that involves a
singular value matrix e
j
in which the singular values are
ranked in decreasing order of magnitude. Left and right sin-
gular vectors are accordingly ranked in the matrices Ue
j

and Ve
j
. Some singular values may thus coincide or cross
over a frequency range so that singular vector ordering arbi-
trarily switches between the two singular values. At this
point, the SVD calculation with classical numerical algo-
rithms will introduce a switch in the radiation mode ordering
2012 J. Acoust. Soc. Am., Vol. 123, No. 4, April 2008 P. Gauthier and A. Berry: Adaptive wave eld synthesis
so that the frequency responses
s
Ge
j
,
s
Ge
j
,
a
Ge
j
and their components go through abrupt transitions
over the frequency range of interest. Such discontinuities
amplitude and phase as function of frequency produce
longer impulse responses when inverse DTFT is applied. Ap-
plying a simple smoothing algorithm to the SVD matrices in
the frequency domain that is radiation mode reordering so
that they smoothly change shape over the reproduction
source array as function frequency before taking the inverse
DTFT proved to greatly decrease the length of the impulse
responses of the lters. This constitutes an originality of the
present work in comparison with Bai and Elliott 2004.
Note that to increase the robustness of the reordering
algorithm for the simulations reported in this paper, an arti-
fact has been introduced for a practical SVD decoupling
demonstration. A random variation normal distribution with
zero mean and 1 mm variance of the source and sensor po-
sitions in the horizontal plane proved, along with standard
SVD algorithm and smoothing process, to enhance the de-
coupling results. Placing of such random artifact in the the-
oretical conguration is somewhat more representative of a
real situation than the otherwise geometrically perfect con-
guration.
1. Examples
To illustrate these ideas, an example of plant decoupling
based on SVD and smoothing process is introduced for the
free-eld condition in Figs. 1316 for the conguration
shown in Fig. 4. Note that this example does not include
plant whitening. In Fig. 13, the 24 synthesis lter impulse
responses for each source mode corresponding to the rst
synthesis lters matrix are shown with 256 points in the
time domain with a sampling frequency of 1200 Hz a con-
servative approximation of the minimal spatial aliasing fre-
quency for the given reproduction source separation distance
is 634 Hz, at least two reproduction sources per wavelength,
so that a sampling frequency of 1200 Hz covers the fre-
quency range of interest. These impulse responses have
been delayed by 128 samples
SVD
=128. Each of the sub-
FIG. 13. Free-eld theoretical synthesis lters in the time domain for the
source modes 14. Each plot includes 24 synthesis lters to create the given
source mode with 24 reproduction sources. Each lter includes 256 coef-
cients. The small plots on the right display the algebraic sum of the lters
response along the 256 coefcients.
FIG. 14. Free-eld theoretical analysis lters in the time domain for the
pressure modes 14. Each plot includes four analysis lters to construct the
pressure mode from four pressure sensors. Each lter includes 256 coef-
cients.
Source mode input
P
r
e
s
s
u
r
e
m
o
d
e
o
u
t
p
u
t
1 2 3 4
1
2
3
4
FIG. 15. Plant decoupling: 10 log
10
of the total energies of the impulse
responses of individual components of the 44 matrix
a
Ge
j
Ge
j

s
Ge
j
using the synthesis and analysis lters shown in
Figs. 13 and 14.
J. Acoust. Soc. Am., Vol. 123, No. 4, April 2008 P. Gauthier and A. Berry: Adaptive wave eld synthesis 2013
plots represents the 24 impulse responses of a column of the
rst synthesis lter matrix
s
Gz
LM
to produce y
L
n from
y
M
n see Fig. 3. Without the smoothing algorithm before
inverse DTFT, the time leakage of these lters had proven to
be much more important. Figure 14 shows the four analysis
lters for each pressure mode in the time domain using

SVD
=128 samples delay. Each of the subplots represents the
four impulse responses of a column of the analysis lter
matrix
a
Gz
MM
to transform the reproduction errors e
M
n
into e
M
n. To illustrate the fact the higher-order source
modes imply more spatial variations than lower-order radia-
tion modes, a set of small subplots are included in Fig. 13.
They show the algebraic sums of the lter coefcients over
the 256 coefcients. Clearly, the rst source mode involves
an in-phase radiation from all the reproduction sources on
the average. The other modes involve more source-to-source
variations, but they show simple spatial distributions. As an
example, it is clear that the fourth source mode appears in
AWFS based on independent radiation mode control shown
in Fig. 7. In contrast, for AWFS based on FXLMS algorithm,
the rst radiation dominates more in Fig. 7. Another point
should be noted: both the synthesis and analysis lters ap-
proach simple multipolar patterns. The rst radiation mode
implies an in-phase radiation from the reproduction sources
and in-phase capture from the error sensors. The higher-order
radiation modes imply an alternation of in-phase and out-of-
phase average over the reproduction source and error sensors
arrays; these are then approaching simple multipolar patterns
Gauthier and Berry, 2006.
To test the validity of the SVD decoupling using these
lters, the plant decoupling has been simulated using the
synthesis and analysis lters described above. The total en-
ergies of the impulse responses in individual components of
the 44 matrix
a
Ge
j
Ge
j

s
Ge
j
are shown in Fig.
15. Clearly, this matrix is diagonal. This proves that broad-
band plant decoupling using SVD as described herein is ef-
fective.
For the simulated case with three rigid surfaces, the syn-
thesis and analysis lters impulse responses are shown in
Figs. 16 and 17. The same small subplots on the right of Fig.
16 show the algebraic sums of the lter coefcients along the
256 coefcients. These lters show a more complex response
than those of the free-eld case see Figs. 13 and 14, but
they have a mean spatial distribution approaching the lters
for the free-eld case. The synthesis lters of the rst source
mode clearly show an in-phase response from all the repro-
duction sources this; is in agreement with our previous the-
oretical results Gauthier and Berry, 2006. Again both the
synthesis and analysis lters approach simple multipolar pat-
terns. The rst radiation mode implies an in-phase radiation
from the reproduction sources and in-phase sensing at the
error sensors. The higher-order radiation modes imply an al-
ternance of in phase and out of phase over the reproduction
source and error sensors arrays, they then approach simple
multipolar patterns. Note that, the synthesis and analysis l-
ters only include plant decoupling. The SVD delay was set to
128,
SVD
=128.
FIG. 16. Synthesis lters in the time domain for the source modes 14 for
the system with a rigid oor and two hard walls. Each plot includes 24
synthesis lters to create the given source mode with 24 reproduction
sources. Each lter includes 256 coefcients. The small plots on the right
display the algebraic sum of the lters response along the 256 coefcients.
FIG. 17. Analysis lters in the time domain for the pressure modes 14 for
the system with a rigid oor and two hard walls. Each plot includes four
analysis lters to construct the pressure mode with four pressure sensors.
Each lter includes 256 coefcients.
2014 J. Acoust. Soc. Am., Vol. 123, No. 4, April 2008 P. Gauthier and A. Berry: Adaptive wave eld synthesis
APPENDIX C: PROJECTION OF THE WFS
OPERATORS ON THE SOURCE MODES
AND PRACTICAL SYNTHESIS OF THE
NULL SPACE OF Ge
j

The AWFS algorithm based on SVD decoupling requires

the projection of the WFS operators on the synthesis lters
Gauthier and Berry, 2006.
Once the rst radiation modes are reordered, it is pos-
sible to write the column-reordered matrix Ve
j
as a block
matrix: Ve
j

LL
=Ve
j

LM
Ve
j

LLM
where V
was used to create the rst synthesis lter according
to the method described in Appendix B while V is used
for the second synthesis lters. In a similar manner, e
j

is made of two blocks:

ML
=
MM
0
MLM
. The pro-
jection of the WFS operators on the rst radiation modes
is a vector of dimension M dened by q
WFS
e
j

=e
j
V
H
e
j
q
WFS
e
j
. Each component of q
WFS
repre-
sents a projection of the WFS operators on the given source
mode multiplied by the corresponding singular value. The
multiplication by the singular value is needed since the rst
synthesis lters are made from Ve
j

1
e
j
according to
the method described in Appendix B. If the system is only
decoupled, the projection of the WFS operators becomes
q
WFS
e
j
=V
H
e
jm
q
WFS
e
j
. Taking the inverse discrete-
time Fourier transform DTFT of a component of q
WFS
gives the impulse response w
WFS
m
for the mth source mode
see Eq. 29, that is the projection of the WFS lters on the
source mode m. This impulse response is non-causal and an
additional delay of
SVD
is added to make it causal. The
importance of the resulting lter projection of the WFS op-
erators on the rst synthesis lters is clear in light of the
algorithm described in Sec. II C.
For the null space basis, any smoothing or reordering
operation on the corresponding source modes is by far more
complicated and less effective than for the rst synthesis
lters. For the projection of the WFS operators on the null
space, a different approach is used. Rather than directly pro-
jecting the WFS operators on the null space with
V
H
e
j
q
WFS
e
j
, a simple approach is to substract the rst
radiation mode contributions from the WFS operators so
that q
WFS
null
e
j
=I
LL
Ve
j

LM
V
H
e
j

ML
q
WFS
e
j

where both q
WFS
null
e
j
and q
WFS
e
j
are L1 vectors. The
vector q
WFS
null
e
j
is the portion of q
WFS
e
j
which belongs
to the null space. This was dened without using the projec-
tion of the WFS operators on the null space basis. Since the
impulse responses corresponding to q
WFS
null
e
j
are non-
causal, an additional delay of
SVD
is included in the impulse
responses for the synthesis of the WFS operators projected
on the null space basis.
AES Staff Writer 2005. Multichannel audio systems and techniques, J.
Audio Eng. Soc. 53, 329335.
Ahn, S.-S., and Voltz, P. J. 1996. Convergence of the delayed normalized
LMS algorithm with decreasing step size, IEEE Trans. Signal Process.
44, 30083016.
Asano, F., and Swanson, D. C. 1995. Sound equalization in enclosures
using modal reconstruction, J. Acoust. Soc. Am. 98, 20622069.
Bai, M. R., and Elliott, S. J. 2004. Preconditioning multichannel adaptive
ltering algorithms using EVD- and SVD-based signal prewhitening and
system decoupling, J. Sound Vib. 270, 639655.
Berkhout, A. J., de Vries, D., and Vogel, P. 1993. Acoustic control by
wave eld synthesis, J. Acoust. Soc. Am. 93, 27642778.
Betlehem, T., and Abhayapala, T. D. 2005. Theory and design of sound
eld reproduction in reverberant rooms, J. Acoust. Soc. Am. 117, 2100
2111.
Bouchard, M., and Quednau, S. 2000. Multichannel recursive-least-
squares algorithms, and fast-transversal-lter algorithms for active noise
control and sound reproduction systems, IEEE Trans. Speech Audio Pro-
cess. 8, 606618.
Cabell, R. H., and Fuller, C. R. 1999. A principal component algorithm
for feedforward active noise and vibration control, J. Sound Vib. 227,
159181.
Cabell, R. H., Palumbo, D., and Vipperman, J. 2001. A principal compo-
nent feedforward algorithm for active noise control: Flight test results,
IEEE Trans. Control Syst. Technol. 9, 7683.
Camurri, A., and Ferrentino, P. 1999. Interactive environments for music
and multimedia, Multimedia Syst. 7, 3247.
Choi, J.-W., and Kim, Y.-H. 2004. Manipulation of sound intensity within
a selected region using multiple sources, J. Acoust. Soc. Am. 116, 843
852.
Corteel 2006. On the use of irregurlarly spaced loudspeaker arrays for
wave eld synthesis, potential impact on spatial aliasing frequency, Pro-
ceedings of the 9th International Conference on Digital Audio Effects
(DAFX), Montreal, Canada, September 1820, 2006.
Davis, M. F. 2003. History of spatial coding, J. Audio Eng. Soc. 51,
554569.
Elliott, S. J., and Nelson, P. A. 1989. Multiple-point equalization in room
using adaptive digital lters, J. Audio Eng. Soc. 37, 899907.
Elliott, S. J. 2000. Optimal controllers and adaptive controllers for mul-
tichannel feed-forward control of stochastic disturbances, IEEE Trans.
Signal Process. 48, 10531060.
Elliott, S. 2001. Signal Processing for Active Control Academic, London.
Fuster, L., Lpez, J. J., Gonzlez, A., and Zuccarello, P. D. 2005. Room
compensation using multichannel inverse lters for waveeld synthesis
systems, convention paper No. 6401, AES 118th Convention, Barcelona,
Spain, May 2831.
Gauthier, P.-A., Berry, A., and Woszczyk, W. 2005a. Sound-eld repro-
duction in-room using optimal control techniques: Simulations in the fre-
quency domain, J. Acoust. Soc. Am. 117, 662678.
Gauthier, P.-A., Berry, A., and Woszczyk, W. 2005b. Wave eld synthe-
sis, adaptive wave eld synthesis and ambisonics using decentralized
transformed control: Potential applications to sound eld reproduction and
active noise control, J. Acoust. Soc. Am. 118, 1967.
Gauthier, P.-A., and Berry, A. 2006. Adaptive wave eld synthesis with
independent radiation mode control for active sound eld reproduction:
Theory, J. Acoust. Soc. Am. 119, 27212737.
Gauthier, P.-A., and Berry, A. 2007. Objective evaluation of room effects
on wave eld synthesis, Acta. Acust. Acust. 93, 824836.
Garas, J. 1999. Adaptive 3D Sound Systems Technische Universiteit Eind-
hoven, Eindhoven.
Ise, S. 1999. A principle of sound eld control based on the Kirchhoff-
Helmholtz integral equation and the theory of inverse system, Acust.
Acta Acust. 85, 7887.
Kendall, G. S. 1995. A 3-D sound prime: Directional hearing and stereo
reproduction, Comput. Music J. 19, 2346.
Klehs, B., and Sporer, T. 2003. Wave eld synthesis in the real world:
Part 1 - In the living room, convention paper No. 5727, AES 114th con-
vention, Amsterdam, The Netherlands, March 2225, 2003.
Kleiner, M., Dalenbck, B.-I., and Svensson, P. 1993. Auralizarion-An
overview, J. Audio Eng. Soc. 41, 861875.
Laakso, T. I., Vlimki, V., Karjalainen, M., and Laine, U. K. 1996. Split-
ting the unit delay, IEEE Signal Process. Mag. 13, 3060.
Lancaster, P., and Tismenetsky, M. 1985. The Theory of Matrices Aca-
demic, Orlando, FL.
Merimaa, J., and Pulkki, V. 2005. Spatial impulse response rendering I:
Analysis and synthesis, J. Audio Eng. Soc. 12, 11151127.
Morgan, D. R. 1980. An analysis of multiple correlation cancellation
loops with a lter in the auxiliary path, IEEE Trans. Acoust., Speech,
Signal Process. 28, 454467.
Nelson, P. A. 1994. Active control of acoustic elds and the reproduction
of sound, J. Sound Vib. 177, 447477.
Nelson, P. S., Kirkeby, O., and Takeuchi, T. 1997. Sound elds for the
production of virtual acoustic images, J. Sound Vib. 204, 386396.
Poletti, M. A. 2000. A unied theory of horizontal holographic sound
J. Acoust. Soc. Am., Vol. 123, No. 4, April 2008 P. Gauthier and A. Berry: Adaptive wave eld synthesis 2015
systems, J. Audio Eng. Soc. 48, 11551182.
Radlovi, B. D., Williamson, R. C., and Kennedy, R. A. 2000. Equaliza-
tion in an acoustic reveberant environment: Robustness results, IEEE
Trans. Speech Audio Process. 8, 311319.
Rumsey, F. 2001. Spatial Audio Focal, Oxford.
Santilln, A. O. 2001. Spatially extended sound equalization in rectangu-
lar rooms, J. Acoust. Soc. Am. 110, 19891997.
Sporer, T., and Klehs, B. 2004. Wave eld synthesis in the real world:
Part 2 - In the movie theatre, convention paper No. 6055, AES 116th
Convention, Berlin, Germany, May 811, 2004.
Spors, S., Kuntz, A., and Rabenstein, R. 2003. An approach to listening
room compensation with wave eld synthesis, Proceedings of the AES
24th International Conference, Banff, Canada, June 2628, 7082.
Spors, S., and Rabenstein, R. 2006. Spatial aliasing artifacts produced by
linear and circular loudspeaker arrays used for wave eld synthesis, con-
vention paper, AES 120th Convention, Paris, France, May 2023, 2006.
Takane, S., Suzuki, Y., and Sone, T. 1999. A new method for global
sound eld reproduction based on Kirchhoffs integral equation, Acust.
Acta Acust. 85, 250257.
Takane, S., and Sone, T. 2004. A new theory for active suppression of
reected sound waves from the walls based on KirchhoffHelmholtz
boundary integral equation, Acoust. Sci. & Tech. 25, 3744.
Verheijen, E. N. G. 1997. Sound Reproduction by Wave Field Synthesis,
Ph.D. thesis Delft University of Technology, Delft.
Williams, E. G. 1999. Fourier Acoustics - Sound Radiation and Neareld
Acoustical Holography Academic, London.
Wilson, S. 2001. Information Arts: Intersection of Art, Science, and Tech-
nology MIT Press, Cambridge.
Woszczyk, W., Cooperstock, J., Roston, J., and Martens, W. 2005. Shake,
rattle, and roll: Getting immersed in multisensory, interactive music via
broadband networks, J. Audio Eng. Soc. 53, 336344.
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