Optimized Multirate Filter Banks for Radar Pulse Compression

Andreas Steffen George S. Moschytz

Siemens-Albis AG Institute for Signal and Information Processing

Albisriederstr. 245 Swiss Federal Institute of Technology
CH-8047 Zurich, Switzerland CH-8092 Zurich, Switzerland

Abstract a length L x T . Usually the matched filter impulse response

Many modern radar systems transmit broadband chirp wave-
forms of long duration, thereby realizing time-bandwidth prod-
ucts in the order of 10’ t o lo4. Thus matched filtering in a
digital radar receiver requires FIR filters possessing up to sev-
eral thousand taps. Fast convolution based on an FFT with
blocksize B of at least twice the matched filter length L , has
been known for quite some time. With increasing blocksize,
the limited numerical accuracy of fixed point hardware and the
growing latency in heavily pipelined F F T processors become
serious problems. In this paper we propose a novel structure,
consisting of a multirate filter bank with analysis and synthe-
sis filters based on an F F T of a much smaller size B x
and B channel filters with very sparse coefficients, so that for
high time-bandwidth products the computational complexity
becomes even smaller than for the standard fast convolution
method. Applying a least-squares optimization algorithm on
the sparse set of channel filter coefficients minimizes the side-
lobes of the matched filter output signal.
is additionally weighted by a window sequence w [ n ] ,in order
t o keep down the time or range sidelobes occurring in addition
to the main correlation peak at the filter output, so we get the
response ~ [ n=]w[n]s*(-n).
All examples presented in this paper will be based on a linear
chirp s ( t ) with bandwidth P = 0.7 and duration r = 1440.
The primary design goal is t o achieve a minimum sidelobe
suppression of at least 50 dB.
“Perfect Convolution” Filter Banks
a Recently it has been shown [2],how a multirate filter bank
with an arbitrary subsampling factor N , can be used to imple-
ment a running convolution of an infinitely long input sequence
z [ n ]with a finite filter impulse response c[n] of length L . The

Chirp Waveforms
Let s ( t ) = a(t)ejw(t)be the complex-valued baseband repre-
sentation of a continuous-time FM waveform s ( t ) of duration
r . In radar applications the amplitude a ( t ) of the transmitted
waveform is usually held constant. The simplest FM wave-
form is a linear chirp [I], the name coming from the fact that
its instantaneous frequency
Figure 1: Multirate filter bank with B channels subsampled
f ( t ) = -- P
1 dP = -t,
by N , consisting of signal analysis filters Gk(z),channelfilters
27r d t T
(ti 5 r / 2
C&) and output synthesis filters Fk(z).
is linearly swept over a total bandwidth /3 during the pulse
duration r . block diagram of a general multirate filter bank is shown in
In a digital radar receiver the complex baseband signal s ( t ) is Fig. 1. An analysis filter bank with individual filters G k ( z ) ,
sampled at a constant rate fc, which we set to unity, without k = 0,1,. . . , B - 1, splits the input sequence z [ n ]into B sub-
loss of generality, and obtain the discrete-time sequence bands, which are subsequently decimated by a factor N. Each
subband signal zk[m]is then filtered at a reduced rate by a
z [ n ]= s ( n -I-A), -0.5 5 A < 0.5 (’4 complex-valued FIR channel filter
with the integer index n marking the sampling instant. We
M-1
have also introduced a real-valued random variable A, because
the exact time of arrival of the radar return is not known. In C,(z) = Ck[m]Z-m (3)
m=O
order t o avoid aliasing, the sampling rate fc must he sufficiently
higher than the approximate bandwidth P of the chirp signal, of length A4 =: L / N . The channel filter coefficients c k [ m ] are
which leads t o the condition p < 1. obtained from the matched filter response c[n]of length L , by
The finite impulse response c[n]of the digital matched filter applying it to a second analysis filter bank, which is not shown
or pulse compression filter [ l ]is
, defined as the complex con- in Fig. 1, and which has filters N k ( z ) , IC = 0,1, ...,B - 1,
jugate of the time-reversed waveform s ( n ) , and therefore has and subsamples the outputs by the same factor N . Finally

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 a synthesis filter bank with filters F k ( z ) interpolates the B The strong diagonal structure apparent in the plot of Fig. 2
channel signals y k [ m ] to form the output signal @[,I. immediately inspires the idea of zeroing all sufficiently small
Among the various “perfect convolution” methods presented coefficients c k [ m ] off the diagonal, thereby substantially re-
i n [a],all of which were derived from fast algorithms for com- ducing the computational effort required in implementing the
puting polynomial products, the approach based on DFT- channel filters Ck(z). Unfortunately the massive spectral leak-
modulated analysis and synthesis filter banks is most suited for age due to the sin(z)/z shape of the individual DFT bandpass
processing chirp waveforms. In this case the individual anal- filter responses Hk(ejw),keeps the number of negligible coef-
) Hk.2) actually compute the unweighted
ysis filters G ~ ( zand ficients rather low, even for large time-bandwidth products.
short time Fourier transform or spectrogram of the sequences Therefore no significant savings can be achieved with “perfect
z [ n ]and c[n],respectively. convolution” filter banks.

.............
........ ........
....
B x M Frequency Selective Filter Banks
......... 0.0
..0.0.0
0.0.D.0.0.0
0
..... 64 x 47
A further idea [3], which was also explored in [4],is to use
...
. O O ~ . D . O . D . O . ~ . O . D . O . ~ . ~

....
..0.0.*

.........
D...O.*.O.D 0.0.

longer DFT-modulated analysis and synthesis filters G ~ ( z ) ,

............ ...
0...0.0.0.0.0.0.0.~.0
0...0.0...0...00.0.
O.O.0.D.O.O.rn.O
0..0.0.0.0.0.0
a...
e.o.a..o.o..o
0.0.0.0 H k ( z ) and Fk(z), having a more selective frequency response.
o.o.a......
oa.OO~......o.o.o.o.o.D..... Scale
.....
.00.000......0.0.0.0.0.*....
o..ooooo........o.o.o.o.o....
.D..oO.Oo........o.o.o.o.....
0..00.00........D.0.0.0-....
0
.
.0
0.
0
0.
0.
.
..
..
0
.e
.a
.
~.
..
OdB
The signal energy is then restricted to a few neighbouring chan-
nels centered about the instantaneous frequency of the chirp.
This tremendous improvement is evident from Fig. 4. With
-6 dB
0
3
.....
......
...
o.o.o....o.......o....o...... -12 dB ................................................
B x M
...... ..o.o..o.......o....o.o.o
o...o........o...~o.o.o.O.o.
o.o.o......lr.o..Oo.oo.o..
64 X 51
-24 dB

-36 dB

0.0.0.0.e.0.D.0.000.
-48 dB
-
Scale
.......
.......
........ ..........
.O.O.O.O.O.D.D.e..q.

.......
.........
0.0.0.0.0.0.D...000.
0.-

............ OdB
...... ..........
O.o.O.a.O.0

....................
..................
s.0.0.e
0.0
-60 dB
0

-6dB
0
0 m --i 46
-12 dB

-24 dB

-36 dB

-48 dB

-60 dB
-3: . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
0 1 50
m -+
[dB1
-10 Figure 4: Magnitudes of the coefficients q [ m ]of a multirate
filter bank with frequency selective analysis/synthesis filters.
-20
All coefficients with magnitudes below a level of -38 dB were
-30 zeroed.

-40 the largest coefficient normalized to have unit magnitude, all

channel filter coefficients c k [ m ] with magnitudes below a cer-
-50 tain level were zeroed. To achieve a minimum sidelobe sup-
pression of at least -50 dB, a level of -38 dB was chosen for
-60
this particular example. The minimum number of coefficients
-70 I I I
is obtained, if the instantaneous frequency f ( m N ) of the chirp,
defined by (l), changes between adjacent blocks of N samples
by approximately one DFT bin of width 1/B, i.e.
Figure 3: Magnitude of the compressed pulse y[n]for a “per-
fect convolution” multirate filter bank. m
- = 0.5 - f ( m N ) , m = 0,1, ....M - 1
B
Because the system response of a “perfect convolution’’ mul- for a down chirp. Using (1) and assuming B = 2 N , we can
tirate filter bank is time-invariant, the output sequence $[n] solve for the optimum DFT size
shown in Fig. 3 is equal to the linear convolution of the FIR
] the input sequence z [ n ] ,save for an
filter sequence ~ [ nwith
additional system delay of N - 1 samples [a].
BOpt= :/ M fi (4)

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 Usually the nearest power of two or four is chosen, which allows
the use of an efficient radix-2 or radix-4 F F T algorithm.
Since this modified multirate filter bank does not possess the
"perfect convolution property", it becomes a periodically time-
varying system with period N . This causes additional side-
lobes in the output signal jj[n], which vary with the time of
arrival of the transmitted signal s(t), relative t o the timing
of the subsampling circuitry. By designing lowpass filter pro-
totypes G ( z ) , H ( z ) and F ( z ) with a high enough stop band
attenuation, the aliasing terms can be kept as small as desired.
of the multirate filter bank depicted in Fig. 1, where WN =
.
N 1s the N t h root of unity and the terms X ( K $ z ) , z =
1,.. . ,N - 1, are aliased versions of the input signal X ( z ) . If
we define by
1
U t ) ( = )= - F k ( z ) G k ( W t ) (6)
N
the combined filter response of the analysis bandpass filter
Gk(W$z), rotated by 2 ~ i / Nin the complex z-plane and its
corresponding synthesis filter Fk(z), then using ( 3 ) , we can
rewrite ( 5 ) as

Y(2)= c
N-1 E-1 M-1
U~'(Z)X(~*)Z-mNck[m]

-20 Note the linear dependence of the output signal Y ( t ) on the

channel filter coefficients ck[m]. We now select the positions,
-30
denoted by the tuple (ICl, m T ) ,of those coefficients Ck,[mr], r =
-40
1,.. . ,R, which we will free for optimization. The remaining
coefficients are set t o zero. With the definition of the set
-50
C = { ( h , m 1 ) , . . ., ( I C , , ~ , ) , . . . , ( ~ R , ~ R ) }
-60
we can write the transfer function Y ' ( z ) of the multirate filter
-70 I I I bank with sparse channel filters as
-r -.5r 0 .5r T
N-1

Figure 5 : Worst-case envelope of the compressed pulse ?(z) = ck[m] uy(z)x(w$z)z-mN (7)
(k,m)€C t=O
g[n] for a multirate filter bank with frequency selective a n a l
ysis/synthesis filters of length 3B. Since the filters F k ( z ) and G ~ ( zhave
) a high stopband at-
tenuation, the combined analysis/synthesis frequency response
If the subband signals are oversampled by a factor of two, i.e. lUf)(eJW)Iusually becomes very small for i = 1,...,N - 1.
by the convenient choice B = 2N, we can considerably relax Thus in a first step we will neglect all aliasing terms in (7) and
the transition widths of the bandpass filters. In that case anal- proceed with the time-invariant response
ysis and synthesis filter lengths of about three times the DFT
blocksize B are sufficient t o guarantee a stopband attenuation Y ( z )= Ck[rn] U k ( z ) X ( z ) P N (8)
greater than 60 dB and range sidelobe levels below -50 dB can (k,m) E C
be achieved. Due to the periodically time-varying nature of where, for ease of notation, we have introduced the equality
the multirate filter bank, N different output sequences jj[n] Uk(2) = U ( O ) ( z ) .
must be computed. If we always keep the highest value at Due to the random variable A introduced in (2) the exact
every time instant n, we get the worst-case envelope of the position t o of the output correlation peak can fall anywhere
compressed pulse, plotted in Fig. 5 . The aliasing errors mani- between two sampling instants. Therefore, depending on the
fest themselves mainly as attenuated and shifted copies of the actual value of the parameter A, the discrete-time output se-
main correlation peak, spaced B samples apart. This is a phe- quence $[.] can take on any value of the continuous-time out-
nomenon typical of DFT-modulated filter banks. DUt SiWd
(9)
Optimized Channel Filters
obtained by interpolating the output sequence obtained by set-
The major advantage of the method described in the preced- ting A = 0. Inserting (8) into (9) we get the relationship
ing section is its ease of use. Once the analysis and synthesis
filter banks are designed, any FIR filter response c[.] designed Y(t) = Pk,m(t) C k b I ( 10)
( k m )E C
t o match the input sequence ~ [ n ]can , be fed into the analysis
filters H k ( z ) and the channel filter coefficients ck[m] just ap- with the functions pk,m(t) defined as
pear at the other end. But as is well known from FIR filter
design, windowing methods with subsequent truncation of the
filter impulse response are far from optimal.
Hence we propose in this paper a direct optimization of the
nonzero channel filter coefficients ck[m]. We set out from the
periodically time-varying transfer function [2]

N-1B-1
(5)

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 so that the output signal (10) becomes the dot product

Y ( t )= P Y t ) c
As stated earlier, our primary design goal is the minimization
of the maximum sidelobe level. In our work we have chosen a 32 64 - 12 26
least-squares approach by minimizing the total sidelobe energy
E,, obtained by integrating the squared magnitude Table 1: Number of complex multiplication per output sam-
ple. a) direct FIR implementation, b) fast convolution with
Iv(t)12= ?/"(t)Y(t) = C"P*(t)P" c
B M 2L, c) fcst convolution with B M f i ,d) with selective
starting a distance r, from the left and from the right side of filter banks, e) with optimized channel filters.
the main correlation peak situated at t = to. By defining the
the optimized channel filters require about 30% less nonzero
.................................................. coefficients than the truncated channel filters. Table 1 gives
BxM
64 x 51
an overview of the number ptot of complex multiplications per
output sample, needed for the various implementations of a
matched filter with length L = 1471. ptot is the sum of p ~ m
... ..............
.. . .. (channel filters), ~ F F T(radix-2 FFT/IFFT) and p ~ (selec-
p
. . &&
...
tive analysis/synthesis polyphase filters).
OdB
... ... ... e
...
.. .. ..
-6 dB Conclusion
Tk ... ... ...
e

... ... ... -12 dB We have shown how multirate filter banks based on a DFT of
.. .. .. small size can be used for matched filtering of chirp waveforms.
.. .. .. -24 dB
... ... ... These novel structures have several advantages:
... ... ... -36 dB
First, for large time-bandwidth products, the number of mul-
..- ... ,. ... tiplications becomes smaller than for the standard method.
,... ... ... -48 dB Second, with an optimal blocksize B M 4,the processing gain
... ... ... is evenly distributed between the analysis and the synthesis
.. .. .. -60 dB filter banks, so that a chirp signal experiences the maximum
...
.. .. ..
. . . . . . . . .. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . - gain in every stage of the forward, as well as the inverse FFT.
. .
I 51:I
m- Simple fixed point arithmetic with static scaling is therefore
sufficient. With a large transform, the whole processing gain
Figure 6: Optimized channel filter coefficients ck[m]. occurs in the IFFT, so that the forward F F T becomes very
region S = {t : t < to - r, U t > t o 4-r,} we can express the sensitive to numerical overflows caused by sinusoidal jammers.
sidelobe energy as Third, as radar systems become increasingly more agile in re-
sponding to changing environments, the huge delays incurred
E, = Iy(t)12dt= c H Qc by pipelining a large transform can pose serious problems.
With a matched filter based on an FFT of small size, sys-
with Q being a R x R positive definite and Hermitian square tem latency is reduced considerably, making highly adaptive
matrix defined by systems possible.

References
Setting up the matrix Q is actually the computationally most [l] C. E. Cook and M. Bernfeld. Radar Signals. Academic
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can be replaced by summations, if the vector function p ( t ) is
[2] M. Vetterli. Running FIR and IIR filtering using multirate
evaluated on a fine enough grid along the time axis. A sampling
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frequency of 4fc will give satisfactory results.
ing, ASSP-36(5):730-738, May 1988.
The minimization of the sidelobe energy can now be formulated
as the quadratic programming problem [3] A. Steffen. Digital Pulse Compression Using Multirate Fil-
ter Banks. PhD thesis, ETH Zurich, 1991.
minimize E, = c H Qc (12)
[4]C. Cafforio, C. Prati, and F. Rocca. Synthetic aperture
subject to the linear constraint y(t0) = p'(t0) c = 1, which radar focusing with polyphase filters. Signal Processing,
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from [5]and modified appropriately to handle complex param- [5] G . W. Medlin, J. W. Adams, and C. T. Leondes. La-
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The resulting optimized channel filter coefficients are plotted multirate applications. IEEE Trans. Circuits Syst., CAS-
in Fig. 6. To achieve the same 50 dB sidelobe suppression, 35(10):1210-1219, October 1988.

661

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