PROCEEDINGS

IEEE, OF THE

904

VOL. 6 1 , NO. 6 , JUNE 1919

Digital Interpolation Beamforming for low-Pass

and Bandpass Signals
ROGER G. PRIDHAM

AND

RONALD A. MUCCI, MEMBER, IEEE

Abstrtrct-Digital timedomain beamforming requires that samples of

the sensor signals be available at a sufficient rate to realize accurate
time delays for beam steering. For many applications, this input rate,
which may be significantly higher than the Nyquist rate required for
waveform reconstruction, places stringent requirements
on A/D converter hardware and transmission b b l e bandwidth. Recently, a technique referred to as digital interpolation beamforming was introduced
which greatly relaxes the sampling requirement and provides substantial
hardware savings through more flexibledesign options. In this approach,
thesensorchannels
need only be sampled at arate whichsatisfies
aliasingrequirements.
Thevernierbeamdelayincrements
are then
synthesized using digital interpolation which can be implemented at the
beamformer input or output to minimize digital processing complexity.
Previously, thisconceptwaspresentedforthe
case of low-pass
signals. This paperextendsthisworkbyexaminingtherelationship
betweeninterpolationandbeamformingfortheimportant
class of
bandpass signals. Specifically,
sampling
methods
are
discussed
wherebythe
originalwaveform
can be reconstructedfromsamples
taken at a rate consistent with the bandwidth
of the bandpass signal.
Beamformer
implementations
are
presented
which
utilize
these
bandwidth-sampling techniques in conjunction with interpolation and
which compute beam output points at the generally low rate dictated
by the signal bandwidth. The interpolation beamformer achieves timedelay quantization (beam-steering accuracy) independent of both the
input and output sampling rates. This approach generally requires less
hardware than conventional procedures. Interpolation fdter characteristics dictated by the bandwidth-sampling procedure are described and
digital
efficientmethods of implementationemployingnonrecursive
bandpass and low-pass fiiters are presented.

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.

form reconstruction, i.e., the sampling rate 6- is significantly

larger than the Nyquist rate. This impacts directly on the A/D
converter requirements and possibly digital data transmission
and storage requirements.
An efficientbeamforming procedure which utlizes hydrophone data sampled at an interval consistent with the Nyquist
rate appeared recently in the literature [2]. The
beamsteering
delays necessary for time-domainbeamforming are realized
with digital interpolation
filters;
hence,
the
terminology
digital interpolation beamforming. Basically, the procedure
involves processing zero-padded sensor data with a linear interpolationfilter.Theadditional
digital processing requiredto
implement the interpolation
can be minimized by appropriately
placing theinterpolation
filter atthebeamformerinput
(Fig. 2(a)) or output (Fig. 2(b)). Frequently, the interpolation
filter characteristics can be realized with the digital filters required to shape thespectrum of the beam outputpriorto
post-beamformer signal processing. Thus ahardware savings
may be realized since there exists the flexibility to optimally
partitionA/Dconverterand
cable bandwidthrequirements
and, digital processing complexity.
Theconcept of digital interpolation beamforming can be
developed for both low-pass andbandpass signals. However,

0018-9219/79/0600-0904$00.75 0 1979 IEEE

PRIDHAM BEAMFORMING
AND
INTERPOLATION
MUCCI: DIGITAL

)1!

905

HILBERT
TRANSFORM

'1

'1

I
L

alnA1

__-_---__---___

i
I

(C)

Fig. 3. Sampling methods forbandpasssignals.

(a) Analytic signal
sampling. (b) Second-order sampling. (c) Quadrature sampling.

the original work [ 2 ] concentrated on the low-pass case both

because of its practical importance and because it requires
the most straightforward sampling and digital interpolation
procedures. The bandpass case is equally important but more
complex because of the bandwidth-sampling techniques that
must be employed. This work extendsthe earlier work to
formulate canonical-interpolation beamformerstructuresfor
bandpass signals. These structures are applicable to the broad
class of bandwidth-sampling techniques. However, the following discussion concentrates primarily onthree of the more
common bandwidth-samplingprocedures.Theseinclude
uniform sampling of the analytic signal (Fig. 3(a)), second-order
samplingprocedures(Fig.
3(b)), and uniform sampling of
quadrature components (Fig. 3(c)).
Utilizing these sampling procedures, the dataneed be sampled
only at a rate consistent with the signal bandwidth W, rather
than the highest frequency component fH,of the signal spectrum. Thus these sampling methods provide the most efficient
means (i.e., fewer data samples persecond are required) of
characterizing thedata, especially when theconditionthat
W <<fH is satisfied. The increased complexity of these
samplingprocedures
is generally offsetby
the efficiency
achieved for data representation. In the beamforming application, thistranslates into a potential hardware savings in the
areas of A/D conversion andinputdata
transmission and
storage. For someapplications, the signal conditioning may
actually be simplified as in the case where complex demodulation is used to implement the frontend bandpass filters with
low-pass filter sections.These
low-pass filtersare easier to
implement than corresponding bandpass sections. In addition,
the digital filters which are typically required to band-limit the
beam output, can perform some or all of the required interpolationoperations. In this case, interpolation filteringmay
only resultina
fractional increasein thetotalnumber
of

arithmetic operations performed by the existing band-limiting

filters.
In the following section,threebandwidth samplingprocedures commonly employed for bandpass signals are reviewed
briefly.This
sectionis also used to develop thenotation
utilized throughoutthe paper. In Section 111, theinterpolation procedure (resampling) for low-pass signals is briefly
described and extended to include bandwidth
resampling of
bandpass signals. New material is also given on the frequencydomain interpretationof "reduced" or subfilters.
. The development of interpolation beamforming is presented
in Section IV for both low-pass signals and bandpass signals.
First, the low-pass beamformer structure is reviewed. Second,
theextension tothe bandpassapplication is given. Various
methods of implementation are described for each ofthe
threebandwidth samplingprocedureswhich
have thefunctional structures shown inFig. 2.
A fundamentallydifferentbeamformerstructure
is also
introduced which operates directly on a frequency translated
replica of the bandpass-sensor sequence tocompute a frequency translated replica of the beam-output sequence. This
new technique is functionally a time-domain beamformer but
it possesses attributes of bothtimeandfrequencydomain
beamforming.It
provides potential hardware savings
even
when interpolation is not employed.
An example of beam-patternperformance is presented for
a uniformly spaced line array using quadrature samplingin
Section V. Finally,results and conclusionsarepresentedin
Section VI.

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

where X(O) denotestheFouriertransform

band-limited low-pass signal such that

X ( w )= 0,

for

101

> 27rW

of x ( t ) . For a

(2)

there is n o overlap of the rn = 0 and rn # 0 terms in (1) if

A-1 > 2 W . Thus in principle, thefunction x ( t ) can be reconstructedfrom samples takenatthe
interval ( 2 W ) - ' . If
A - 1 < 2 W , then overlap occurs and the m = 0 term cannot be
isolatedbyfilteringsince
it is contaminated by the m # 0
spectra. This overlap which is referred to as aliasing, is illustrated in Fig. 4.

906

PROCEEDINGS OF THE IEEE, VOL. 6 7 , NO. 6 , JUNE 1979

C. Uniform Sampling of an Analytic Signal

When uniform sampling of x ( ? ) at the rate 2W samples per

second is inadequate, a technique which can be used is to
remove the negative frequency components in the signal prior
to A/D conversion. This is equivalent to forming the analytic
signal [ 41 PA ( t ) which is defined as
P A ( t )= x ( t ) + i x g ( t )

-?

(7)

where the caret symbol (") is used to denote a complex signal

or sequence and x&) is the Hilbert transform of x ( t ) . x&)
is obtained by the filtering operation

x H ( t )= x ( t ) * [-(nt)-']

(8)

where the asterisk (*) denotes convolution. In the frequency

domain this operationhas the representation
X H ( W )= -i sign ( w )X ( w )

(9)

since -(nt)-' is thk inverse Fourier transform of -i sign (a).

The Fourier transform of the analytic signal X A (a),
is
zy(o),

XA(w)=
Fig. 4 . Illustration of aliasing effect for a low-pass signal.

X(w),

r..

o>o
0=

o<o.

In the case of the bandpass signal where X ( 0 ) = 0, one has

X A ( w )= 2X( a)u(0)

If A-' > 2W, thenreconstruction

operation

filtering involves the

00

h(t - n A x) ( n A )

x"(t)=
?I=--

(10)

(3)

where the symbol (") is used to denote a reconstructed signal

or sequence. Foranexact analog reconstruction, h ( t ) must
be the impulseresponse of an ideal low-pass filter of bandwidth W, i.e.,

1)
(1

where U ( o )is the unit step function.

In terms of the positive and negative frequencyspectra,
X+(w)and X - ( o ) , respectively, which are illustrated in Fig. 5,
one has that

X(0)= X+(w)+ XJO)

(12 )

where

X+(w)=

otherwise

and

This response has the Fourier transform

Thus it is evident that
X A (a)
=

is the rectangle function that is defined by

where rect (a)
rect ( w )=

1,

lot< 1/2

0, I w l > 1/2.

B. Bandpass: Sampling and Reconstruction

For a bandpass signal of bandwidth W confined to theregion
( f o - W / 2 ,f o + W / 2 )where f o > W ,simple uniform sampling at
2W is generally inadequate. However, procedures do exist for
sampling bandpass signals of bandwidth W (Hz)at the rate 2W
samples/s such that information is retained to reconstruct the
original waveform. Three such procedures, commonly referred
to as quadrature sampling, secondader sampling and analyticsignal sampling are briefly reviewed here. For a more detailed
description, see [ 31.

=+(a).

(14)

Fig. 3(a) illustrates the analog signal conditioning required to

obtain 5 ? (~t ) from x ( t ) .
It is clear fromthis discussion that the complex sequence
j ? ~ ( m Ahas
) the Fourier transform Z A [exp ( i w A ) ] shown in
Fig. 5 , i.e.,
'ZA [exp

( 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

It is readily shown that x2(mA)has the transform

x0

X2 [ exp (iwA)] = A-'

X(o + 2nmA-')

m=-m

. exp [ -i(w + 2nmA-')(u].

(20)

The use of xz(mA) introduces a new degree of freedom which

can be used to eliminate aliasing.
The original waveform is recovered by properly filtering and
combining the two sequences, i.e.,
m

x1 ( m a ) h ~ ( -t mA)

x(t) =
m=-m

xz(mA)h,(-t

+ mA - a).

(21)

m=-m

For exact reconstruction an idealbandpassfilter is required

with frequency transfer characteristics, H,(w) given by
(0,

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 )

where wo = 2nf0. This filter has the complex impulse response

h(t)=

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)

where CY is the temporal offset of the staggered sequence. The

sequence x1 (ma),has the Fourier transformgiven by

x, [exp(iwA)]

and k is the largest integer for which (k - 1) A-' <fo

= 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

An alternative procedure, which eliminates the need for im-

( 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-

PROCEEDINGS O F THE IEEE, VOL. 67,1979

NO. 6 , JUNE

908

strutting the in-phase and quadrature components with a low-

pass filter, i.e.,

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).

Thus direct uniform sampling of x ( t ) at the interval 1(2fo)-'

( 2 4 ~ yieldssamples
of the in-phase componentsmodulated
by
m
(- 1)".
A delayed version of the quadrature component is obtained
where h ( t ) denotes the filter impulseresponse. The original
waveform is obtained by weighting and combining the in-phase in a similar fashion. Incorporating ( 3 0 ) , (31), and (19) into
( 2 3 ) yields
and quadrature components as follows:
5 -00

?(t)= Zz(t) cos o0t- %e(?)

sin oot.

(25)

For exact reconstruction, h ( t ) must correspond to the ideal

low-pass filter of bandwidth W / 2 .
There area
number of important similarities that exist
amongthethreebandwidth
sampling procedures discussed
which should be noted. For example, the procedure for
generating the analytic signal involved a Hilbert transform. However, the analytic sequence$A ( m a ) can also be obtained using
quadraturedemodulation.
If the complexbaseband
signal
pB(r) is sampled at the rate A - ' , the resulting sequence has
the transform
XB[exp ( i o A ) ] = 2A-'

5 X + ( o+ oo+ 2nmA-'1.

(26)

m=-w

Clearly, from (15) ?A ( m a ) is obtained by multiplication of

x^,(mA) with exp ( i o o m A ) . This multiplication is eliminated
if A is chosen to be an integer multiple off;', i.e.,

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)

Hence, samples of the in-phase andquadraturecomponents

can be obtained using second-order sampling with the appropriate selection of the sampling interval A and intersequence
sample delay a.

111. INTERPOLATION (RESAMPLING)

Reconstruction of the original waveform from the sequence
x ( m A ) was discussed previously. However, in applications
such as digital beamforming a new sequence is desired which
corresponds to sampling the original waveform at a higher rate.
The procedure employed for both
increasing and decreasing
the original sampling rate for band-limited low-pass signals has
been treated in the literature [ 51-[ 71 and is briefly reviewed
herefor completeness. Because of their desirable properties
[51 - [ 7 ] , the use of finite impulse response (FIR) filters for
interpolation is emphasized. In addition,the basic concept
is extended to include bandpass signals as well.

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

PRIDHAM AND MUCCI: DIGITALINTERPOLATION BEAMFORMING

1
.iotQfl

ZERO
PAD

pw

IMTERPOLATION
FILTER

"I.

Fig. 6 . Digital interpolation process.

transform %'[ex, (iwAo)l given by [ 5 1 ,

L-1

%'[exp (ioAo)] = L - '

'Y{exp [i(w- 2nkA,')6]}.

(37)

k=O

A result which will be useful for digital beamformer analysis

isobtained
byapplyingthisexpression
to atime-delayed
sequence of the form X [ ( m - M ) 61. Using this identity, and
noting that x [(m- M ) 6 ] , where M is a fixed integer, has the
transformexp ( - i M d ) X[exp ( i d ) ] , then, from (37), the
sequence x [ ( m L - M )61 has the transform
L-1

Xh[exp (iwAo)] = L - '

exp [-i(w - 2nkAi1)M61

k=O

. X{exp

[i(w - 2nkA;') SI}.

(38)

If the sequence x(rn6) is approximately band-limited' so that

lX[exp ( i d ) ] l = O for I w l > n A i ' , then the k # O terms do
not overlap the k = 0 term and
Xh[exp (ioAo)] = L-' X[exp (i06)l exp ( - i o M G )
for IwI

< TAG'

(39)

Xh(.)

2-

2-

Fig. 7 . Illustration of thetransforms of thesequences.

(b) h ( k 6 ) . (c) z ( k 6 ) . (d) x ( k 6 ) .

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 - '

. JD {exp [i(w - 2nkA;') 6 I}

0 {exp [i(w- 2nkA;') SI}

used to band-limit a sequence prior to a sample rate reduction

[ 9 ] . Thefrequency-domaininterpretation of (34) is readily
developed. Specifically, if hD(k6) and u(k6) have the Fourier
(iw6)] and 0 [exp ( i d ) ] , respectively,
transforms
[exp
then x"(rn6) has the transform

L- 1
k=O
*

(36)

Fig. 7 shows the sense in which the transform %[exp ( i d ) ]

of theinterpolated
sequence approximatesthetransform
X[exp ( i d ) ] of the desired sequence for an example where
K = 2. i t is observed that the transforms differ due to the
finite passband andstopband
ripple of theinterpolation
filter.Thefilter
passbandripple
introducesanamplitude
distortion in the desired band while the stopband ripple gives
rise to out-of-band componentscenteredat k2nA-l. These
componentscontributetotheinterpolationerror,butin
principle, can be controlled by increasing the filter complexity
if necessary. However, if x"(k6) is downsampled to form the
sequence x"(kL6), then the out-of-band components may fold
into the signal band. When this happens, the resulting in-band
interpolation error is irreversible.
The effect of downsampling a sequence by a factor of L to
effect the output sampling interval L6 = A. can be determined
by noting that if a sequence y(rn6) has the Fourier transform
3 [exp ( i d ) ] , then the sequence y ( m L 6 ) =y(rnAo) has the

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.

1 exp [-i(w - 2nkA,')M6]

k=O

J
C
,{exp [i(w- 2nkA;')

SI}

X {exp [ i(w - 2nkA;') AI}

(40)

and $h[exp (ioAo)] is an approximation to L-' exp (-iwM6)

X [exp (iwAo)l.
As mentioned previously, the k = 0 term is an amplitude distorted version of the desired sequence in the band IwlG nA,'.
For this term there is no phase shift with respect to the desired
transform L-' exp (-iwM6) X [exp (iwAo)]. The k # 0 terms,
however, represent energy folded back into the band of interest. These terms can induce an apparent phase shift between
the interpolated and the desired sequences in the signal band.
Clearly, it is desirable to design the interpolation filter so that
the energy which folds in the signal band upon downsampling
is kept to acceptable levels.
When the upsampling ratio K equals the downsampling ratio
L , that is when A. = A, the folding of the out-of-band c o m p e
nents can be interpreted in terms of the amplitude and phase
9

'In the sense that aliasing requirements are satisfied.

PROCEEDINGS OF THE IEEE, VOL. 67, NO. 6 , JUNE 1979

910

where
L-1

XM[exp ( i w A ) ]= L-'

exp [ - i ( w - 2nkAP-')M61

k=O

*XD[exp [i(o- 2nkA-')61.

(43)
Equation ( 4 2 ) follows since for A . = A,
X{exp [i(o
- 2nkA;')AI} = X[exp ( i w A ) ] ( 4 4 )
which simply means that the transform of the data sequence
perfectly overlaps itself for frequency shiftsof A-'. If A . # A ,
equation ( 4 0 ) cannot be writtenintheform
of ( 4 2 ) which
means that a time invariant subfilter cannot be defined. This
is a consequence of the fact that when K f L , the composite
filter changes its relative position with respect to the paddedzero sequence as interpolated points are generated. Thus the
subfilter is in effect time varying.
B. Resampling of Bandpass Signals

The ideal filter characteristics required for reconstruction of

the analog waveform have beendescribed for the bandpasssampling techniques also. Not onlyare thesefilters unrealiFig. 8. Subfdter sequences for L = 3.
zable but many times it is desirable to represent the original
waveform as a sampled sequence. Hence, if the reconstruction
distortion of a "reduced" orsubfilter.
Specifically, for a filter characteristics can be realized approximatelywith a
sequence which is sampled at the interval A , resampling at the digital fiiter, then thedesired sequence can be obtained directly.
same interval A with a time shift of M6 can be achieved by Generally, the reconstructed waveform cannot be sampled at
the rate A consistent with the bandwidth W(Hz) but must be
filtering the given sequence with the subfiiterhM(mA) where
sampled at the rate 6 consistent with the highest frequency
hM(mA) = hD [(mL - M ) 61
component. If S is greater than A , then interpolation is neces= hD(mA - M 6 )
( 4 1 ) s a r y . The higher sampling rate can be realized by appropriately
zero padding thesampled sequence prior to fiitering.
instead of the full or composite filter hD(m6). This follows
The exact resampling procedure varies somewhat depending
since for a given value of M, the relative position of hD(m6) upon the bandwidth sampling procedure. A complex bandpass
andthepadded
zeros is invariant. The zeros inthedata
phase is required forthe
filter of constantamplitudeand
sequenceeffectively
eliminatethe
coefficients which they analytic signal. These fiitercharacteristics
can be realized
multiply.
approximatelywith a FIRfilter using thecomplex coeffiFig. 8 shows the subfilter coefficients for a 19-point sym- cients f ; (m6)
~
given by
metric filter with L = 3. Note that the M = 0 filter is symmet( m a ) = hp(m6) exp ( i w o m(64)5 )
ric while the M = 1 and M = 2 filters are not. As a result of
0

where L = K is important for interpolationbeamforming. This

is true, since a departure from linear phase results in frequency
dependent time delay errors that impact on spatial beam patterns and array directivity.
In thefrequencydomain,the
subfilter interpretation is
readily developed by substituting 4, = A into (40). This yields

where

otherwise

'Because of

the asymmetry of these reduced fdters, it has generally

for L = 2 , onecannottakeadvantage
of
beenbelievedthat,except
boththecompositefitersymmetry
and thepaddedzeros in an upsampling fdter. Recently, however, it
has been pointed out [IO] that
this is possible because the same data-coefficient product can
be used
for conjugate subfiter outputs.

and A = L 6 . The sequence corresponding to the reconstructed

waveform x"(m6)is simply the real part of ?A ( m 6 ) ,i.e.,

x"(m6)= Re

{'A

(m6)).
(48)

911

PRIDHAM AND MUCCI: DIGITALINTERPOLATIONBEAMFORMING

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

Fig. 9. Realization of passbandphasediscontinuityat

f 2 using bandpass fdters of constant amplitude and passband response;
f, and f,
denote lower and upper passband cutoffs, respectively.

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)

where N is a positive integer, then the phase discontinuity is

removed
and the desired filter characteristics can be achieved
+
u 2 [ ( m - k ) G I h s [ ( N C - 1 - k ) 6 1 (49)
k=o
is not necessarily a
approximately witha single filter.This
singificant restriction on the sampling frequency.
where the zero-padded sequences u1( m a )and vz ( m a ) are
Forquadrature sampling, thereconstruction is performed
with low-pass filters. However, since thereconstructed sexi(mL-'A),
m = 0, +L, f 2 L , * *
ui(m6) =
(50) quence is sampled at the rate 6-', which in general is higher
otherwise
than the original sampling rate A - ' , the in-phase and quadrato,
ture sequences are zero padded prior to filtering. If the lowf o r i = 1, 2.
pass filter is implemented with a FIR filter then the resampled
Thecompositefilterimplementation
can be realized by sequence is given by
properly modulating and phasing two low-pass FIR filters of
Ne- 1
appropriate bandwidths, i.e.,
x ( m 6 ) = cos (worn&)
h L ( k 6 ) q [ ( m- k) 61
h s ( m 6 ) = 2 h L l ( m 6 ) Re {Al exp (i\klm6)}
k=o
Nc- 1

+ 2 h ~ ~ ( mRe6 ){ A ,

exp (i\kzm6)}

where A l and A z are as previously defined by ( 2 2 ) ;

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

PROCEEDINGS OF THE IEEE, VOL. 6 7 , NO. 6 , JUNE 1979

------._

.
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

procedures are incorporated into the beamforming operation

and the interpolation beamforming concept extendedto bandpass signals as well.

anXn(t -

7),

(56)

k)61

k,,?*),

where v, -denotes the zero-padded sequence such that

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.

Substituting 2, for x, in (58) yields

n=l

where x n ( t ) denotes the output of the nth sensor of an array

of NE sensors, r, denotes the time delay required to compensate for the delay experienced by a coherent wavefront propagating froma specifieddirection t o thenthsensor,and
a, which describes digital interpolation beamforming where interdenotes the shading coefficient of the nth sensor. If the beam- polation is performed prior to beamforming. This implemenformer is implemented digitally as shown in Fig. 10, then the tation is shown in Fig. 2(a).
As analternative,considerinterchangingtheorderof
the
sensor outputs are sampled in time at the rate fs = 6- prior
to beamforming to provide the digital sequences x,(m6) where summations which yields
n = l , . . . , N E . Conventionally, the required time delays are
realized by shifting the sampled sequences an integer number
M , , of sampling intervals 6, i.e., 7, = M,6 and, hence, the
k =O
n-1
accuracy of the delays is controlled bythe sampling frequency.
processillustrated
The accuracy of the discrete time approximation to
bo(t) whichcan be interpretedasthetwo-step
in
Fig.
2(b).
First,
beamforming
is
performed
onthe
zerodepends on how accuratelyMn6 approximates 7., A beam for
padded sequences v,(m6) producing z(m6) where
which

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

Next, b(mAo) is obtained fromz ( m 6 ) via interpolation, i.e.,

Nc-1

G(m&) =

h&6)z[(mL

- k)61.

(64)

k =O

When interpolation filtering is done prior to beamforming,

the basic operation of thebeamformer is equivalent t o the
conventional
downsampling
approach
where
the
outputsample rate is lower than the input-sample rate.
If the interpolation filter is implemented at the beamformer output, the
beamformer
operates
in fundamentally
a
different
way;
namely, the beamformer processes the zero-padded sequences,
and the output rate is greater than the input rate. Because of
this property, this beamformer structure is properly referred
upsampling
beamformer
by
analogy
with
an
where A. = L6 is thebeamautput
sample rate which is to as an
bounded in the manner A 2 A. 2 6 and Mn6 approximates upsampling filter. Foran upsamplingbeamformer,downsampling to the desired rate occurs in the interpolation filter
the desired steering delay for the nth channel. Note that the
shading coefficients are suppressed to simplify notation with- used for waveform reconstruction.
As an illustration of the operation of the downsampling and
out loss of generality.
line arraywithseven
Since the beamformer output rate is less than the input rate, upsampling beamformer, considera
this conventional structure is properly referred to as a down- equally-spaced elements which are designated E l through E,.
sampling
beamformer
because
of the analogy
with
the TableI gives the elementdelaysrequired
to formthefirst
synchronousbeamoffbroadside
in termsoftherequired
downsampling filter given in Section 111.

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-')

Interpolatlon Filter Requirements

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

SAMPLES ALONG LINES COMBINED 7

El

E2

E4

use of interpolation beamforming reduces the A/D conversion

rate since, typically, fa is significantly less than fs . The price
paid for this reduction as indicated in Table I1 is an increase in
digital processing requirements. However, these computations
can be minimized by appropriately locating theinterpolation filter at the beamformer input or output. If NB < N E ,
implementing interpolation at the beamformer output is computationally more efficient than implementing it at the beamformerinput asindicated
in Table 11. However, in many
applications, much of theinterpolation can be incorporated
into the post-beamforming digital filters and hence, a reduction in theadditional processing requiredforinterpolation
can be realized by performing the interpolation at the beam
output.
It is instructive atthispointto
consider thefrequencydomain representation of the sampled beam output. By
noting that sequences of the form x,[(mL - M n ) 6 ] have the
Fourier transform given by (38), the transform of b(mA0) is
by inspection,
NE

E5

ELEMENTS

L-1

$[exp ( i o A o ) ] = L-'

exp [-i(o- 2nkAi')Mn6]

n = 1 k=O

Fig. 1 1 . (a) Example of downsamplingbeamformer ( 3 to 1 ) . (b)Example of upsamplingbeamformer (1 to 3 ) . (X -data Sample; 0padded zero.)

vernier delay quantization 6. Figs. 1 l(a) and (b)give snapshots

of corresponding beamformingperiods for adownsampling
and upsampling beamformer, respectively. In both cases, the
sampling rate ratio is 3 : 1. For the downsampling case, which
is illustrated in Fig. 1 l(a), the beamformation proceeds with
element samples being added in with the proper relative time
delay. Completed beams are available at the interval A .
Beamformation for the upsampling beamformer (Fig. 1 l(b))
is similar to that which occurs for the downsamplingbeamformerexceptforthe
effectivezeropadding
of theinput
sequences. Completed prebeam outputs, whichmustsubsequently be interpolated, are available at the interval 6. However, theaddedterms
consistprimarily of zeros. A microprogrammable beamformer [ 111 can be programmed to take
advantage of theseperiodicallyrecurringzeros.
Thusthe
higher beam outputratedoesnotimpactonbeamformer
computational throughput.
Table I1 summarizes the sampling and computationalrequirements of conventionalandinterpolation
beamforming. The

( " 6 ) has even symmetry.

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

K[exp (i(w- 2nkA--')6]

(67)

PROCEEDINGS O F THE IEEE, VOL. 6 7 , NO. 6 , JUNE 1979

914

Equations ( 6 5 x 6 7 ) can be used to computeboth single

frequency and broad-band beam patterns and hence, to compare the sidelobe structures of the conventional and interpolation beamforming procedures.

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

where 2An(t)denotes the analytic represent$ion of the nth

hydrophone output. The analyticsamples of bA ( m A are
)
NE

&(mA) =

2*,,(mA - r n )

n=l

where the minimum uniform sampling interval A necessary for

waveform reconstruction is equal to W-' for bandpass signals
of bandwidth W. Hence, theanalyticrepresentation of the
beam output can be computed by properly delaying and
summing theanalyticrepresentation
of the sensor outputs
sampled at the rate W-' consistent with the signal bandwidth.
Theexacttime
delays 7, are generally approximated by
M n 6 , where 6 is a fixed interval in time and M, is an integer
suchthat
M
.
6
approximates 7, withsufficientaccuracy.
Commonly 6 < A and interpolation is required to resample
the analytic signals at the approximate time instants Mn6 required for beam steering.
The interpolation can be accomplished conceptually by first
producing the sequence $ ~ ~ ( r n 6where
),

7KEoRYp+=

ZERO
PAD

'a

(b)
12. Beamforming foranalytic signals.(a) Rebeamforming interpolation. (b) Postbeamforming interpolation.

If A is constrained to be an integer multiple L of 6 then this

zero-padded sequence is next smoothed with a complex bandpass filter matched to the signal bandwidth to eliminate undesirable replicas of the signal spectra centeredat frequencies
Nfo for integer values of N not divisible by L. (For noninteger
values of L a similar procedure exists where the sample rate
is appropriately increased by an integer multiple greater than
. L and sequentially decreased to achieve the desired sampling
interval [61.) For a FIR implementation of the interpolation
filter, the resampled sequence ?A,(m6), is given by
NP- 1
k =O

where the i~p(k6)denote the NC complex filter coefficients.

Notethat complexfiltercoefficients
which produce a passband of width W centered at fo can be easily obtained from
the real coefficients h ~ p ( F of
) a low-pass FIR filter of band) h ~ p ( k 8 exp
)
(i2nfok6).
width W / 2 by equating h ~ p ( k 6 to
The approximate analytic beam output EA ( m A )which
,
results
is given by

(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

An appropriateprocedure is to choose a = K 6 , where K

is an integer, and to compute the interleaved sequence b(mA KS) in the following manner:
NE Nc-1

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)=

h ~ , ( k S ) ~ l ~ [-( Km -LM n - k)61

n=l

(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

Once again, interpolation is generally necessary to resample

the signals to obtain the time delay MnS which approximates
7, with sufficient accuracy. Data samples at the time instants
required for the vernier beamsteering delays can be computed
from the two sequences x l n ( m A )and xzn(mA)as discussed in
Section 11, where

The beamformer implementation for secondarder sampling

is depicted in Fig. 13 for both prebeamformer and postbeamformerinterpolations.The
mostefficient methodofimplementationdependsuponthenumber
of array elements,
number of beams formed simultaneously, and the post beamforming filter requirements.

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

where b z ( f ) and b Q ( f ) denotethe in-phase and quadrature

components respectively of b ( f ) . Expressing the beam output
in terms of the quadrature representation of the sensor output yields

x2,(mA) = x,(mA - a).

(75b)

b ( t )=

and

NE

xZ,(t - 7,) cos wo(t- 7 , ) - xQ,(f - 7,) sin wo(t- 7,)

n=l

Using a FIR filter, theresampling operation is

(81)
Nc- 1

where x l , ( f ) and X Q n ( f ) denote the in-phase and quadrature

components, respectively, of x n ( f ) . Equating(80)and (81)
yields

k =O

NE

bZ(t) =

where for i = 1 , 2

[Xl,(f

- 7,) cos wO7n + X Q n ( f - 7,) sin 007n)]

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 )=

[xzn(t - 7,) sin w07, - X Q n ( f - 7 , ) cos 007,1.

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.

Hence, the quadrature components of the beam output can

be obtainedfromthequadraturecomponents
of the sensor
outputs.
For a spectral bandwidth of W, the quadrature components
need only be specified at time instants spaced A apart where
A < W-',that is,
NE

bdmA) =

[xzn(mA- 7,) cos w07,

n=l

+x~,,(mA
- 7,) sin w o ~ n ]

(83a)

916

PROCEEDINGS OF THE IEEE, VOL. 6 7 , NO. 6 , JUNE 1979

DIGITAL
BEAMFORMER

Un~l

t-7-TBANOPASS

OlGlTAL
BEAMFORMER
N

Ill

I
I

Fig. 13. Beamforming for second-ordersampling.(a)

Rebeamforming
interpolation. (b) Postbeamforming interpolation.

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

h,p(k6)uQn(mA - Mn6 - k6) cos Wo7n

k=O
As before,the 7,, aregenerallyapproximated
by Mn6. The
resampledsequences of the in-phaseand quadrature components, xzn(m6) and xQn(m6),respectively, can be obtained by
smoothingthe zero-paddedsequences ur,(m6) and uon(mS) where h ~ p ( k 6 )denotesthekthcoefficient
of the low-pass
with a low-pass filter where
filter consisting of NC coefficients. The corresponding canonical implementation is shown in Fig. 14 for prebeamforming
interpolation.Thefiltering
can be performed atthebeamformer output aswell.
)
not have t o implement
The interpolation filter h ~ p ( k 6 does
(84) otherwise.
as high an upsampling ratio as do the bandpass interpolation
foreitheranalytic
signal or secondarder
Using a FIR filterforinterpolation,theapproximate
beam fiitersemployed
sampling. The reason for this is that the low-pass quadrature
outputs, Z d m A ) and ZQ(mA),are given by
signals can be aligned for beamsteering with delays which are
accurate t o a fraction of a wavelength of the highest frequency
component present in the quadrature signals. Thus 6 depends
n=l I k=o
on W rather than on fo + W / 2 . This is a consequence of the
fact that beamformation is being implemented on a frequency
Nr- 1
1
translated replica of the bandpass spectrumof the signal.
The information regarding the carrier frequency 00 = 2nfo
is contained in the sine/cosine shading coefficients that weight
the quadrature components prior to beamforming. Although

917

PRIDHAM AND MUCCI: DIGITAL INTERPOLATION BEAMFORMING

'l=="OTl

ZERO
PAD

- LOWPASS
FILTER

1
--

ZERO
PA0

INTERPOLATION

OUMRATURE
SAYLING

I
LOWASS
FILTER

..

0161TAL
BEAMFORMER

s
CEsn*0-Nt

LOWPASS
FILTER

=N~=-CI-N~

Fig. 14. Interpolation beamforming for quadratic sampling.

the 7, must be specified accurately for these coefficients, this

does not impact on eitherfA or fs .
Themultiplication by cos W O T , and sin W O T , whichare
beam dependent as well as element dependent can be eliminated by beamforming the complex analyticsequence

2~(rnA)= [xl(rnA)+ ix&nA)l

xl imi.Mltl
TIME DELAY

I
I

&>
I

exp [iuornA1.

(86)

By choosing A such that

TIME
DELAY

x~(m:-M~cl

I,=lWkb

A = Ilfo

(87)

where I is an integer, the multiplication by exp (iwornA) is

eliminated. Even if this condition is not satisfied, it is noted
that eachchannelneed
only be modulated once to form all
beams so thatthenumber
of multiplies is proportional to
NE ratherthan NENB. Of course, thecomplexbeamformer
now computes samples of the analytic signal representation
of the beam output rather than thebaseband representation of
the beam output.

Fig. 1 5 . Synchronous digital beamformer.

which is assumed to be 1500 m/s. For a sampling frequency

a beam pointingdirection
of
e, = 1 1.5O.
The beam patternforthesynchronousimplementation
depicted in Fig. 15. was computed using (65).The
result
is shown in Fig. 16.
Similarly, the beam pattern was computed using (67)for
theinterpolation beamformerdepicted in Fig. 14. An input
V. SPATIALBEAM PATTERNS
sampling rate of 10 kHz is assumed. Hence, an interpolation
The
performance
and
computational
requirements
for
ratio of 10 : 1 is required to achieve the effective sampling rate
synchronousandinterpolation
beamforming arecompared,
of100
kHz. Theinterpolation is performedwithalownextfor abandpass application.The
passband of interest pass FIR
filter.
The fiter coefficients were synthesized
is assumed tooccupy
aband
of W = 5kHz centeredat
using a computerized design alogrithm described in the literafo = 10 kHz.
ture [ 121 which minimizes the maximum absolute error
For comparison, the single-frequency beam pattern is com- between theactual
filterresponse
andthe
ideal low-pass
puted at the center frequency
fo for both the synchronous
characteristics (also see [ 131). The resulting filter frequency
andaninterpolationbeamformerincorporatingquadrature
characteristics and coefficientsareshown
in Figs. 17(a) and
sampling foran
unshaded array consisting of twentyane (b), respectively. Using this symmetric set of 31 filter coeffiomnidirectionalelements
of half-wavelength spacing. The cients, over 32 dB of out-of-bandrejection is obtained. The
first synchronous beam off-broadside satisfies
beam pattern for the interpolation
beamformerimplementation is shownin Fig. 16. Note thatthe sidelobesare only
( d / c ) sin e, = I/fs
(88)
slightly affected by the interpolation process.
The results indicate that the A/D conversion rate and beamwhere 6, denotes the beam pointing direction, d denotes the
beamforming can be
interelement spacing, and C denotesthepropagation
speed former throughput rate for synchonous

fs = 100kHz,oneobtains

918

PROCEEDINGS OF THE IEEE, VOL. 67, NO. 6, JUNE 1979

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

Fig. 16. Singlefrequencybeampatterncomparisonforsynchronous

beamformer versus digital interpolation beamformer with quadrature
sampling.

-Y.O

relaxed using interpolation. However, additional circuitry is

required not only for interpolation but for amplitude weighting of the quadrature sequences prior to beamforming for the
baseband quadrature beamformer.
This prebeamforming amplitude weighting may be incorporatedintothespatialshading
process in applicationsrequiring a shaded array. Also, if the sample rate is as prescribed
in (87) thentheadditionalcomputations
requiredprior to
beamformingare eliminated. This restriction of the sample
rate produces quadrature samples which represent the analytic
signal as well. Thepostbeamforminginterpolation is implemented now with a bandpass filter as shown in Fig. 18. This
bandpass filter can be designed directly as described in [ 141[ 151 or by proper design of a low-pass filter with passband
cutoff of W / 2 Hz. In either case, the complex beam output
needs to be computed only at the low rate W consistent with
the bandwidth of interest.
VI.

00.0

SUMMARY
AND CONCLUSIONS

In this paper, the novel concept of the digital interpolation

beamformer, which is formulated for low-pass signals in [ 2 ] ,
is generalized to treattheimportant
case where the sensor
outputs are bandpass signals. The new beamforming structures
given hereare compatiblewith abroad class of bandwidthsampling techniques that provide the most efficient representation of bandpass signals. Specific sampling techniques
considered
include
analytic-signal sampling, secondader
sampling, and quadraturecomponent sampling.
As in the low-pass case, interpolation beamforming provides
additional design flexibility by permitting hardwarecomplexityto be partitioned between A/D converter and cable
bandwidth, anddigital processing. For specific applications,
it offers the potential of fewer A/D converters, and reduced
datastorage
and transmission requirements.For
bandpass
signals, these hardware savings increase rapidly as the ratio of
bandwidth to center frequency decreases.
An importantfeature of theinterpolationbeamformer
is
that the unsampling ratio K can differ from the downsampling
ratio L . This permits oneto achieve timedelayquantizationindependent
of. boththeinputandoutput
sampling
rate.

0)
Fig. 17. (a) Filter frequency response for a symmetric set of 31 coefficients.(b)Filterunitsampleresponseforasymmetricset
of 31
coefficients.

Fig. 18. Digitalinterpolationbeamformerwithcenterfrequency

passband an integer multiple of sample rate.

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
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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
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434, Feb. 1978.
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throughput
for
an interpolation beamformer is actually
pp. 1219-1226, July 1959.
decreased over that required by a direct implementation.
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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
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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.
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R. E. Crochiereand L.R. Rabiner,Furtherconsiderationon
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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.
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nowdepend
onthe
highest frequencycomponent
of the
reduction, ZEEE Trans. Audio
Electroacoust.,
vol. AU-20,
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no. 4 , pp. 308-315, Oct. 1972.
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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
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Dec. 1973.
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H. S. Hersey, D. W. Tufts, and J. T. Lewis, Interactive minimax
Interpolation beamforming is also appropriatefor analog
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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.