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IEEE, OF THE
904
AND
I. INTRODUCTION
ONVENTIONAL time-domainbeamforming
is accomplished by combining the signals from an array of hydrophone sensors as shownin
Fig. 1. By matching the
beamformer time delays to the signal propagation delays of a
pressure field which is incident from a specific direction, the
amplitude of a coherent wavefront can be enhanced relative to
background noise and directional interference. Generally, the
sensor outputs are shaded or amplitude weighted prior to summation to reduce the beams sidelobe structure.
The beamformer function can be implemented in a variety
of ways. True analog continuous-time beamformersutilizing
analog delay and sum networksrepresent a straightforward
approach. Typically however, the beamformer is implemented
as either a discrete-time or a sampled data system and the required operations (multiplication, addition and time delay)are
performed digitally [ 11.
Formany digital beamformingapplications, the sampling
interval 6 required to properly match the propagation delays
for beamsteering is much smaller than that required for waveManuscript received September 5 , 1978; revised December 21, 1978.
R. G. Fridham is with the Submarine Division of Raytheon Company,
Portsmouth, RI 02871.
R. A. Mucci was with the Raytheon Company, Portsmouth, RI. He is
now with Bolt Beranek and Newman, Inc., Cambridge, MA 02138.
SULER
ZERO
OIGITAL
IHTERWLATION
PAD
DIGITAL
BEAMFORMER
ZERO
PA0
INTERWLATION
(b)
Fig. 2. Interpolationbeamformingstructures.
(a) Prebeamforming
interpolation. (b) Postbeamforming interpolation.
PRIDHAM BEAMFORMING
AND
INTERPOLATION
MUCCI: DIGITAL
)1!
905
HILBERT
TRANSFORM
'1
'1
I
L
alnA1
__-_---__---___
i
I
(C)
11. SAMPLING
PROCEDURESAND WAVEFORM
RECONSTRUCTION-A
REVIEW
Sampling andreconstruction procedures for low-pass and
bandpass signals are reviewed now to facilitate the development of digital interpolation and interpolation beamforming
concepts.
A . Low-Pass: Sampling and Reconstruction
For the low-pass case, a signal of bandwidth W(Hz) is specifiedbysamples
taken at theuniform interval A = (2W)-',
where thebandwidth
W includes the significant frequency
componentsofthe
signal andincludesaguard-band
which
ensures against unacceptable levels of aliasing. If x ( t ) is
A-', thentheFourier transsampled uniformlyattherate
form of the sampled sequence, X[exp ( i w A ) ] ,is
OD
X [exp ( i w A ) ]=
x ( n A ) exp ( - i n w A )
n=-oD
= A-1
(1)
X ( w + 2nmA-')
m=-m
X ( w )= 0,
for
101
> 27rW
of x ( t ) . For a
(2)
906
-?
(7)
x H ( t )= x ( t ) * [-(nt)-']
(8)
(9)
XA(w)=
Fig. 4 . Illustration of aliasing effect for a low-pass signal.
X(w),
r..
o>o
0=
o<o.
00
h(t - n A x) ( n A )
x"(t)=
?I=--
(10)
(3)
1)
(1
(12 )
where
X+(w)=
otherwise
and
1,
lot< 1/2
0, I w l > 1/2.
=+(a).
(14)
( i o A ) ] = 2A-'
X+(O + 2nmA-').
(1 5)
mr-00
Since theFouriertransform
of the sampled analytic signal
consists of replicas of the positive frequency spectra of X(o),
the waveform is specified by W complex samples per second or
2W real samples per second because the positive spectra do not
overlap for A-' > W.
PRIDHAM AND
INTERPOLATION
MUCCI: DIGITAL
BEAMFORMING
907
x0
X(o + 2nmA-')
m=-m
(20)
x1 ( m a ) h ~ ( -t mA)
x(t) =
m=-m
xz(mA)h,(-t
+ mA - a).
(21)
m=-m
0 < w <wO
- nA-'
where
Fig. 5. Fourier
transform
of sampled
analytic
sequence
;A(mA),
where sampling rate A-' is greater than signal bandwidth W.
Analog reconstruction .^A ( t ) from .^A (mA) requires a complex bandpass filter with the transfer function
H(o)=
1,
wo-nW<o<oo+nW
0,
otherwise
(1 6 )
sin (nWt)
(iwo
exp
nt
t).
(1 7)
D. Second-Order Sampling
plementing the Hilbert transform,is
t o use second-order
sampling. Second-order sampling of x(t), which is illustrated
in Fig. 3(b) yields two sets of uniformly spaced samples which
are interleaved, Le.,
= x(mA)
(184
and
XZ
( m a ) = x(mA
- CY)
(18b)
x, [exp(iwA)]
= X[exp
(iwA)].
(1
9)
(2.24
- (2A)-'.
E. Quadrature Sampling
The final procedure considered involves characterizinga
waveform using uniformly spaced samples of itsquadrature
components.It
is well known [3],[4]thatany
bandpass
waveform can be represented as
x(t) = xZ(t) COS w 0 t - xQ(t) sin mot
( m a )x 1
p = 2nd-1
(23)
where x&) and XQ(f) are the in-phase and quadrature components of x(t), respectively. If x(?) is constrained to a frequency band of width W then xl(t) and XQ(f)are low-pass
signals of bandwidth W/2. Thus it is evident that x(t) is specified by the sequences xz(mA) and xQ(mA) where A-' = W.
This results in the total sampling rate of 2W real samples per
second.
A commonapproach to obtainingthequadrature components is to employquadraturedemodulation
as shownin
Fig. 3(c). In one channel x(r) is modulated by 2 cos w o t and
low-pass filtered to remove the sum frequency term to obtain
x ~ ( t ) . Similarly, X Q ( ~is) obtained from the sine channel. The
original waveform x(t), can be reconstructed byfirstrecon-
908
follows
x1 ( m a ) = x ( m A )
Z z ( t )=
xz(mA)h(t - mA)
(244
m=-w
= x z ( m A ) COS ( o o m A ) - X Q ( ~ Asin) ( ~ 0 m A ) ( 3 2 )
= (- l)"'xZ(mA).
(25)
5 X + ( o+ oo+ 2nmA-'1.
(26)
m=-w
A = Af= Kf;'.
When this is satisfied, one has
(27)
x2 ( m a ) = x z ( m A
- a)COS [ o o ( m A - a ) ]
- x p ( m A - a) sin [ o 0 ( m A- a ) ]
= (- a).
(33)
A. Resampling of LowPassSignals
If the original waveform x ( t ) has been sampled at a rate A ,
XB[exp ( i o A : ) ]= 2(Af)-'
X + [ o+ 2n(m + K ) ( A ' ) - ' I .
mr-w
which is adequateforreconstruction,interpolation
canbe
used to effecta higher samplingrate KA-' where K is assumed
( 2 8 ) to be an integer. The resampling of the sequence at the higher
rate KA-' can be viewed as the two-step process shown in
Making the substitutionk = m + K yields
Fig. 6 . First the sequence is upsampledbyeffectively interleaving K - 1 zeros between each data point. Then, the zeroXB[exp ( i o A f ) ]= 2(A')-'
X + ( o + 2n(Af)-'kl
padded sequence is smoothed using a digital filter to obtain
k=-m
estimates of the signal attherate KA-'. if theFIR fiiter
= X A [exp ( i o( A
2 9f ) l .
for interpolation or resampling is characterizedby the unit,
6 = A / k , then
sample response h ~ ( k 6 )where
Thus waveform reconstruction to obtain x~ ( t ) can be achieved
using the quadrature-sampled sequence 9B(mA'). It should be
noted that while ?A (mA') = 2B(mAf),in general ?A ( t )# ?,&).
k=O
The complex demodulationrequiredprior
to sampling to
obtain the quadrature components can also be eliminated by
where u(m6) denotes the zero-padded sequence defined as
properly applying the second-order sampling techniques. This
x(mM-'),
m = 0,fK,*2K, *
procedure requires that [ 41
u(m6) =
(35)
otherwise
Io,
A=Z(2fo)-' < W-', I = 1,2, * * *
(30)
and Nc denotes the number of filter coefficients. The symbol
and
(") is used to indicate that %(ma) is an approximation to the
a = (4f0)-' + K A
(3 1) sequence x(m6). Since the filter output rate 6-' exceeds the
input rate A - ' , this fiiter can be referred to an "upsampling
where K is an integer. The resulting sequence, x1 ( m a ) , is as filter." This is in contrast to a "downsampling filter" which is
909
1
.iotQfl
ZERO
PAD
pw
IMTERPOLATION
FILTER
"I.
(37)
k=O
k=O
. X{exp
(38)
< TAG'
(39)
Xh(.)
2-
2-
and
denotesthe Fourier transform of the sequence
delayed M6 in time and sampled at a rate of A,' samples per
second.
It is also useful forsubsequent discussion toapply this
result to thetime-delayed interpolated sequence x"[(rn - M ) 6 1 .
When theinterpolated sequence ?[(m- M ) 6 ] is downsampled by L , its transform is similarly given by
L- 1
(a) u n ( k 6 ) .
f h [ e x p (iwAo)] = L - '
L- 1
k=O
*
(36)
exp [ - i ( w - 2nkA;')MSI
=L-'
!T[exp ( i d ) ] = x D [ e x p ( i u s ) ] O[exp ( i d ) ]
= X D [exp ( i d ) ] X[exp (iwA)l.
k=O
J
C
,{exp [i(w- 2nkA;')
SI}
(40)
910
where
L-1
XM[exp ( i w A ) ]= L-'
exp [ - i ( w - 2nkAP-')M61
k=O
where
otherwise
'Because of
x"(m6)= Re
{'A
(m6)).
(48)
911
and
BAWDPAS FILTER
2k' - 1 = k
I+
J
In,
f2 f3
(5 1d)
where k is the largest integer for which (k - 1) A-' <fo (2A)-', and hLl(m6) and h ~ z ( m 6denote
)
the mthcoefficients
of two low-pass FIR fdters of bandwidths 2nk'A-' - 2w0 and
00 - 2nA-l (k' - 1/2), respectively, each
of which have N ,
coefficients. Hence, the two sets of filter coefficients required
for interpolation, h (m6)and h2 ( m a ) ,are given by
A bandpass filter is also requited for waveform reconstruction for second-order sampling. The passband is characterized
in frequency by a constant amplitude. However, in general, a
discontinuity in the phase responseoccurs inthe passband
(see equation 22). This can be realized by utilizing two passband filters of constant amplitude and phase as shown in Fig.
Thetwo
filter implementation generally produces unac9. If hs(m6), m = 0, 1, . . ,Nc - 1 denotes the unit-sample ceptable passband ripple in the vicinity of the discontinuity.
response for the composite FIR filter implementation then the As an alternative approach, the discontinuity in phase can be
resampled sequence is given by
eliminated by proper selection of the sampling frequency fs.
If f, is chosen such that the center frequency of the passband
Nc- 1
fo is an integer multiple off,, i.e.,
x"(m6)=
u1 [ ( m- k) 61 h s ( k 6 )
k=O
fs
=folN
(53)
+ 2 h ~ ~ ( mRe6 ){ A ,
exp (i\kzm6)}
IV. INTERPOLATIONBEAMFORMING
The application of the sampling procedures and resampling
techniques discussed above to beamforming is described in this
section. First, the basic concept of interpolation beamforming
for low-pass signals, which has been presented in the literature
[ 2 ], is reviewed briefly with the addition of some new material. Second,boththebandwidthamplingand
resampling
912
------._
.
I
------_----------
The
concept
of digital interpolation
beamforming
can be
demonstrated by a simplemodification t o thisconventional
digital beamforming process. The sensor outputs are sampled
at the rate fa,,zero padded and smoothed via a digital interpolation filter as discussed in Section I11 to obtain estimates
of. the sampled data at the time instants required for the beamsteeringdelays. If theinterpolationor
resamplingisimplemented with a FIR filter
of unit-sample response h ~ ( k 6for
)
k = O ; * - , N c - 1,then
I
I
Nc-1
anXn(t -
7),
(56)
k)61
k,,?*),
un(m6) =
m = 0,fK, f2K,
10,
NE
b ( t )=
k =O
A . Low-Pass Application
Theanalogbeam
output b ( t ) , computedbysummingthe
delayed sensor outputs, is given by
h&6)vn[(m
?,(mS) =
* *
otherwise.
n=l
z(m6) =
is referred to as a synchronous or exact beam. When (57) is
not satisfied exactly,the beam is calleda nonsynchronous
is evident thattheaccuracy of a
orapproximatebeam.It
nonsynchronous beamcanonly
be improved by decreasing
the vernier timedelay interval 6. Generally, fs = 6 - is significantly higher than the minimum sampling rate fa = A-1. requiredforwaveformreconstruction
to meetrequirements
on beam patterns. The beam output b(mAo) is given by
NE
u,[(m - Mn)61.
(63)
n=1
G(m&) =
h&6)z[(mL
- k)61.
(64)
k =O
PRIDHAM AND
INTERPOLATION
MUCCI: DIGITAL
BEAMFORMING
TABLE I
ELEMENT
DELAYS
FOR FIRST
BEAMOFFBROADSIDE
(I Is THE REFERENCE
ELEMENT)
Element
913
TABLE I1
GENERAL
SAMPLING
AND BEAMFORMING
REQUIREMENTS* FOR CONVENTIONAL
AND INTERPOLATION DIGITAL BEAMFORMERS
(NE = NUMBER
OF ELEMENTS,
NC = NUMBER
OF FILTER
COEFFICIENTS, NE = NUMBER
OF BEAMS,
fs = A-',fs = 6-')
A,D Rale
Type
Adds/s
Samples/s
E2
26
E3
Interpolation
Beam-
KE
fa
36
E4
46
E5
56
E6
Multiplids
I
I E CN
NE fA
Interplation Beamformer
Post-beamformer
Interpolation
(N
1) fa
I L
- 1) fa
C
N (N
B
HBNCfA
2
66
*This table assumes that h
El
E2
E4
E5
ELEMENTS
L-1
$[exp ( i o A o ) ] = L-'
Fig. 1 1 . (a) Example of downsamplingbeamformer ( 3 to 1 ) . (b)Example of upsamplingbeamformer (1 to 3 ) . (X -data Sample; 0padded zero.)
Xn[exp i ( o - 2 n k A i 1 ) 6 ] .
(65)
Corresponding to (65), the Fourier transform of the approximate beam output &mAo) obtain with interpolation is from
(40)
L-1
%[exp ( i o A o ) ] = L-'
NE
1 exp [ - i ( w - 2nkAi')Mn6]
k=O n = 1
JC(exp [ i ( w - 2nkAi1)6]
X,[exp ( i ( w - 2nkAi')AI
(66)
x(-)
where
denotes the frequency characteristic of the interpolation filter. Forthe
special case where A . = A , this
becomes
L-1
NE
exp [ - i ( w - 2nkA--')Mn6]
L-'
!&exp ( i o A ) ] =
n=l
k=O
(67)
914
B. Bandpass Application
It was demonstrated that the beam output can be computed
using hydrophone data sampled at a rate Ai1 consistent with
the highest frequency component in the signal spectrum of
interest rather than at the generally higher rate 6-', required
for beam steering. Interpolation is usually required to realize
the steering delays forthe lower sampling frequency Ai'.
Here, the interpolationbeamforming concept, used in conjunction withbandwidth-sampling
techniques, is extended to
include the general class of signals possessing bandpass characteristics. Themore
efficient datarepresentation
provided
by the bandwidth sampling for bandpass signals limited to a
band of W(Hz) centered at fo such that fo > W/2 results in
additional hardware savings in areas such as A/D conversion,
data transmission and storage and beamformer throughput
requirements.
The following is adescription of various beamformerconfigurations intended primarily for bandpass applications which
incorporatebothinterpolationandthe
previously described
bandwidthsampling techniques. Only the case where A0 = A
is given to simplify the development. Theextension to the
more general case is as given for the low-pass case.
Fig.
C. Analytic Signal Sampling
Theanalyticrepresentationforthe
beam output z A ( t ) is
n=l
&(mA) =
2*,,(mA - r n )
n=l
7KEoRYp+=
ZERO
PAD
'a
(b)
12. Beamforming foranalytic signals.(a) Rebeamforming interpolation. (b) Postbeamforming interpolation.
(72)
Clearly, theinterpolation can be performedeither prior to
or following the delay and summing operations as shown in
Fig. 12. The most computationally efficient location of the
interpolation filter depends upon the number of sensors and
number of beams formed simultaneously. If there are more
beams(sensors) than sensors(beams) thentheinterpolation
is performedmore efficiently attheinput(output)
of the
PRIDHAM AND
INTERPOLATION
MUCCI: DIGITAL
BEAMFORMING
beamformer. An exception to this occurs when the interpolation filtercharacteristics can be incorporated into the postbeamformer filter characteristics which are usually needed to
shape the spectrum of the beam output.
915
D. Second-Order Sampling
&A
In the case of secondarder sampling, the beam output b ( f ) ,
is sampled to form twosequences
b l ( m A ) = b(mA)
- KS)=
(734
k=O
N E Nc-1
h ~ , ( k Suzn[(mL
)
- K - M n - k)61.
n=l
k=O
and
b z ( m A )= b ( m A - a)
where A is the minimum sampling interval necessary for waveform reconstruction. In terms of the sensor data, b l ( m A ) and
bz(mA) can be expressed as
NE
xbnl ( m A )- =
7,)
(744
n=l
and
NE
b z ( m A )=
x,(mA - a - T ~ ) .
(79)
(73b)
(74b)
n=l
E. QuadrafureSampling
Commonly, for bandpass signals, the postbeamformer signal
processing is performed on the quadrature components of the
beam output. If the signal spectrum of interest occupiesa
frequency band centered at fo ,then the quadrature representation of the beam output b ( f ) ,is given by
b ( f )= bZ(f)cos &)of - b Q ( f )sin &)Of
(80)
x l n ( m 4=xn(mA)
(754
(75b)
b ( t )=
and
NE
n=l
k =O
NE
bZ(t) =
where for i = 1 , 2
[Xl,(f
fl=l
(824
otherwise,
and h ~ ~ ( k and
6 ) h ~ , ( k 6 )denote the kth coefficients of the
interpolation filter requiredforsecondarder
sampling as
discussed in Section 111. Substituting 2n for x, in (74) yields
NE
&A)
NE
b Q ( f )=
n=i
(82b)
Nc-1
hDl(kS)uln[(ml- M,,- SI
=
n=l
and
k=O
Second-order sampling of the beam output requires the computation of the additional sequence b"(mA- a) in order to
reconstruct the beam output, i.e., the two sequences g l ( m A )
and g2(mA)completely specify the beam output.
bdmA) =
+x~,,(mA
- 7,) sin w o ~ n ]
(83a)
916
DIGITAL
BEAMFORMER
Un~l
t-7-TBANOPASS
OlGlTAL
BEAMFORMER
N
Ill
I
I
and
and
NE
NE
bQ(mA)=
[xzn(mA-
7,)
ZQ(mA)=
sin W 0 7 n
n=l
- xQn(mA - 7,)
n=l
COS WO?,]
(83b)
Ne-1
Ne- 1
h ~ p ( k S ) ~ z ~ -( m
A - k6) sin 0 0 7 ,
Mn6
~o
917
ZERO
PAD
- LOWPASS
FILTER
1
--
ZERO
PA0
INTERPOLATION
OUMRATURE
SAYLING
I
LOWASS
FILTER
..
0161TAL
BEAMFORMER
s
CEsn*0-Nt
LOWPASS
FILTER
=N~=-CI-N~
xl imi.Mltl
TIME DELAY
I
I
&>
I
exp [iuornA1.
(86)
TIME
DELAY
x~(m:-M~cl
I,=lWkb
A = Ilfo
(87)
fs = 100kHz,oneobtains
918
OD
0.0
-10.0
-20.0
0.1
02
03
04
05
:1.0.D
UI
UI
.no
I ,
.YO
I I
nI
,x
,
110
ILI
no
nm
o
i
71111A
do
4.0
-Y.O
00.0
SUMMARY
AND CONCLUSIONS
0)
Fig. 17. (a) Filter frequency response for a symmetric set of 31 coefficients.(b)Filterunitsampleresponseforasymmetricset
of 31
coefficients.
of
PRIDHAM AND
INTERPOLATION
MUCCI: DIGITAL
BEAMFORMING
919
Interpolation beamforming has the disadvantage of requiring Dimarco, A. C. Callahan, and W. C. Knight for helpful discussions on the subject of beamforming, and S. M. Kay for
extra digital processing to implement the interpolation filtering. However, forboth low-pass and bandpass signals, the providing a careful review of the final manuscript.
interpolation can be performed at the beamformer input or
REFERENCES
output,dependihgonthe
relative number of beams and
elements, to minimize the extra processing. In addition, most
D. E. Dudgeon,Fundamentals
of digital array processing,
Proc. ZEEE, vol. 65, pp. 898-904, June 1977.
systemsrequire digital filters for bandlimiting.Thesefilters
R. G. PridhamandR.
A. Mucci, A novel approach t o digital
may provide a large percentage of the required interpolation
beamforming, J. Acoust. SOC. Amer., vol. 63(3), pp. 425processing. It is possible that in these cases, the computational
434, Feb. 1978.
D. A. Linden, A discussion of sampling theorems, Proc. IRE,
throughput
for
an interpolation beamformer is actually
pp. 1219-1226, July 1959.
decreased over that required by a direct implementation.
0. D. Graceand
S. P. Pitt,Samplingandinterpolationof
bandlimited signals byquadraturemethods,
J. Acoust. SOC.
In addition to the generalization of the interpolation beamA m e r . , v o l . 4 8 , no. 6 , pp. 1311-1318, 1970.
former, thispaper
introducesanotherimportantconcept;
R. W. Schaferand L. R. Rabiner, A digital signal processing
namely, beamforming on frequency translated versions of the
approach t o interpolation, h o c . ZEEE, vol. 61, pp. 692720, June 1973.
sensor signals. This structure is especially compatible with
R. E. Crochiere and L. R. Rabiner, Optimum FIR digital filter
systems where the sensor data is initiallybasebanded
to
implementationfordecimation,interpolationandnarrowband
filtering, ZEEE Trans. Acoust.Speech, Signal Processing, vol.
simplify implementation of thefrontend
bandpassfilters.
ASSP-23, pp. 444-456, Oct. 1975.
Ratherthan modulating the filtered signals back up to the
R. E. Crochiere, L. R. Rabiner,andR.
R. Shively, A novel
passband,this new technique can be used to beamform the
implementation of digital phase shifters, Bell Syst. Tech. J . ,
Vol. 54, Pp. 1497-1502, Oct. 1975.
basebanded signal directly to generate the basebanded beam
R. E. Crochiereand L.R. Rabiner,Furtherconsiderationon
output. Even if interpolation is notused,thebeamformer
the design of decimators and interpolators,ZEEE Trans. Acous?.
throughput requirements are reduced since the vernier delays
Speech, Signal Processing, vol. ASSP-24, pp. 296-31 1 , 1976.
A. W. Crookeand J. W. Craig, Digital filtersforsample-rate
nowdepend
onthe
highest frequencycomponent
of the
reduction, ZEEE Trans. Audio
Electroacoust.,
vol. AU-20,
basebanded signal rather than that of the bandpass signal.
no. 4 , pp. 308-315, Oct. 1972.
N. Narasimka and A. Peterson, On using the symmetry of FIR
The beamforming techniques given hereshould
be confiltersfor digital interpolation, ZEEE Trans. Acoust.Speech,
sidered in the design of future sonarsystems.
In order to
Signal Procesnhg, vol. ASSP-26, p. 267, June 1978.
assess potential hardware savings, a range of system parameters
Raytheon
Company,
The
Microprogrammable
Beamformer.
Portsmouth, RI: Raytheon Company Publication, June 1974.
must be considered. These include thenumber
of array
J. H. McClellan, T. W. Parks,and L.R. Rabiner, A computer
elements, system bandwidth, number of beams formed simulprogram for designing optimum FIR linear phase digital filters,
taneously, cable bandwidthrequirementsand
signal condiZEEE Trans. AudioElectroacoust., vol. AU-21, pp. 506-526,
Dec. 1973.
tioner requirements.
H. S. Hersey, D. W. Tufts, and J. T. Lewis, Interactive minimax
Interpolation beamforming is also appropriatefor analog
design of linear-phase nonrecursive digital filters subject to upper
discrete-timesystems such as those configured from chargeand lower functionconstraints, ZEEE Trans. AudioElechoacous?., vol. AU-20, pp. 171-173, June 1972.
coupled device (CCD) technology.
ACKNOWLEDGMENT
The
authors
thank
the
support
and
encouragement
of
their colleagues atRaytheonCompany;
especially, W. J .
L. R. Rabiner, J. F. Kaiser, and R. W. Schafer, Some considerations in the design of multiband finiteimpulse response digital
filters, ZEEE Trans. Acoust.,Speech, Signal Processing, vol.
ASSP-22, no. 6 , pp. 462-472, Dec. 1974.
M. McCallig,
Design
ofnonrecursive
digital filters to meet
maximum and minimum frequency response, Ph.D. dissertation,
Purdue University, Lafayette, IN, 1975.