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Least Squared Error FIR Filter Design with Transition Bands

C. Sidney Burms, Fellow, IEEE, Admadji W . Soewito, and Ramesh A. Gopinath, Student Member, IEEE

Abstract-This paper proposes the use of transition bands and transition functions in the ideal amplitude frequency response to allow the analytical design of optimal least squared error FIR digital filters with an explicit control of the transition band edges. The paper derives design formulas for approximations to ideal frequency responses which use pth order spline transition functions. It also derives a mixed analytical and numerical method for zero-error weight in the transition bands, and pass and stopband error weighting functions with an integral squared error approximation. A variable order spline transition function is developed and a method given for choosing the optimal order to minimize the integral squared approximation error.

I. INTRODUCTION HE two most popular linear phase FIR filter design techniques are the windowing method [ 1]-[5] and the minimum Chebyshev error method [6]-[8], [ l ] , [3]. In the case of windowing method, an optimal least squared error approximation to an ideal low-pass filter is truncated by multiplying the infinitely long ideal impulse response by a relatively simple time domain window. This method has the advantage of being easy to implement and of reducing the Gibbs phenomenon in the frequency response approximation. It has the disadvantage of destroying the minimum squared error optimality of the original approximation and of having implicit effects on the frequency response. Minimizing the Chebyshev error in FIR filter design can be done in a general way using linear programming techniques [8], [9] but is more efficiently carried out by using the Parks-McClellan algorithm [6], [l], [3] which applies the Remez exchange algorithm to give an equalripple frequency domain approximation and, therefore, a minimum Chebyshev error. Both the squared error and the Chebyshev error measures have physical and intuitive motivations. Energy and power are functions of the square of a signal and, therefore, are often good measures of a signal or its ability to do something. Because of Parsevals theorem [lo], a good squared error approximation in the frequency domain implies a good squared error approximation in the time do-

Manuscript received May 15, 1989; revised May 21, 1990. This work was supported by the National Science Foundation and Texas Instruments, Inc. The authors are with the Department of Electrical and Computer Engineering, Rice University, Houston, TX 77251-1892. IEEE Log Number 9107665.

main. The maximum value of a signal is also often used to determine if a signal exists or to trigger some effect. The Chebyshev error is particularly important if the signals are known to be sinusoidal or very narrow band. It must also be controlled in order to prevent overflow or some other problem. Interest, importance, or statistical considerations may dictate a weighting of either of these error measures. Window functions applied to signals are used to truncate or segment long signals in order to use DFTs or other block or finite length operations. They are important in short-time Fourier analysis. Their use in filter design seems to come from a need to have a simple method which will minimize or, at least, reduce the Chebyshev error that results from the Gibbs phenomenon in a least squared error approximation of an ideal frequency response that has a discontinuity. A variety of window functions are used with the choice being made as a tradeoff between the amount of reduction of the ripples and the amount of spreading of the transition region of the filter frequency response and on the simplicity of calculating the window. The purpose of this paper is to propose that ifa reduction of approximation ripple is desired, it can be obtained directly and optimally by using the Parks-McClellan algorithm or linear programming rather than using windows to modify a least squared error approximation. If a least squared error approximation is desired, along with a simultaneous reduction of the approximation ripple, we propose the explicit use of a transition band with a spline as a transition function. This allows an analytical solution to the approximation, eliminates the Gibbs phenomenon, allows explicit control of the transition bandwidth, and is as easy to calculate and implement as the window functions. A very interesting result of this paper shows the spline transition function may take on fractional orders and retain its desirable properties. A formula for optimally choosing the spline order which minimizes the integral squared approximation error over the total frequency band is analytically derived and then empirically modified. Analysis of this design method shows very small approximation error and good Chebyshev error behavior even though that measure was not the original criterion. If only the pass and stopband squared errors are of interest, we show how to design an FIR filter with a transition band having no contribution to the approximation
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error, therefore, requiring no transition function. It is shown that the results obtained using the variable order spline transition function with optimal choice of the order are close to the case with no contribution due to the transition band. The use of splines gives almost as good an approximation as more complicated numerical methods and is as simple to evaluate as the window method. Design formulas are derived for this case and for a weighted squared error approximation.

the amplitude response is given by

N- 1

A(w) =

n=O

c h ( n ) cos ( w ( M - n)).
(w(M - n))

(7)

Using the symmetry of h ( n ) in (6) allows a simplification for N odd to give

M- I

A(w)

n=O

2h(n)

COS

+ h(M)

(8)

and for N even 11. LINEAR PHASEFIR FILTERFREQUENCY RESPONSE N/2- 1 A l e n g t h 4 digital filter with an input sequence x ( n ) A ( w ) = n = O 2h(n) cos (w(M - n)). and an obtput sequence y ( n ) is defined and often implemented by In order to temporarily simplify the notation, for N odd, N-l h ( n ) is shifted to be symmetric about n = 0 which makes y(n) = h(m)x(n - m) (1) the filter noncausal, the phase shift zero, and H ( w ) = m=O A (a). shifted impulse response is denoted The and its frequency response can be expressed by the disA(n) = h ( M - n) (9) crete-time Fourier transform of the impulse response as which makes h even or symmetric about n = 0. Equation N- I (6) becomes ~ ( w = C /z(n)e-jwn ) (2)

n=O

h(n) = h(-n).
Equations (7) and (8) become A(w) = A(w)
=
n=

which is a function of the continuous frequency variable w and is periodic with period 2 a . This formulation assumes a normalized sampling frequency of one sample per second. The complex frequency response H ( w ) can be written in terms of a real-valued amplitude A ( w ) and a phase function [ 3 ] as
~ ( w = A(a)eJO"" )

-M

h(n) cos (an)

(10)

(3)

n= I

c 2h(n) cos (an) + h(0)

a

where A (U) may be positive or negative, and 8 ( U ) is continuous. For the phase to be a linear function of frequency with the smallest constant of linearity M consistent with causality and for A ( w ) to be real, with

and the inverse Fourier transform is

h(n) =

M
and

(N - 1)/2

(5)
(6)

When N is odd, in addition to shifting h ( n ) to give h (n), it further simplifies the notation of (10) and of matrix equations used later to define another sequence that starts at n = 0 rather than n = - w by

s*
0

A ( w ) cos (wn) d w .

(1 1)

h(n) = h(N - 1 - n).

In other words, the phase shift of an FIR filter is linear if and only if the impulse response is symmetric and the filter is causal if and only if the constant of phase linearity is at least (N - 1)/2. Note that M is a sort of delay and is an integer for an odd length filter and an odd multiple of a half-integer for an even length filter. Another special form of FIR filter exists which has a phase response that is linear plus a constant 90". This phase response is equivalent to the impulse response being antisymmetric or odd-symmetric [ 3 ] . These systems are used to implement differentiators and Hilbert transformers and the methods derived in this paper also can be applied to them. Since the frequency response H ( w ) is the Fourier transform of the impulse response h ( n ) , using (2), (3), and ( 5 ) ,

which allows a simple sum of in (10).

M

fi ( n )to replace the formula

(13)

A(o) =

n=O

c 2A(n) cos (wn)

This definition of f ( n ) is not necessary for even length or z antisymmetric filter design.

111. LEASTSQUARED ERRORFIR FILTERDESIGN The basic frequency domain FIR filter design problem is to choose the best N filter coefficients { h ( n ) } to approximate a desired frequency response. This requires first explicitly choosing a desired ideal response and an allowed class of filters. Next, a criterion of the "goodness"

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1.2

of the approximation must be set, and finally, a method must be derived to choose the best approximation from the allowed class as measured by the criterion. This process is often used iteratively. After the best filter is designed and evaluated, the desired response or the allowed class or the criterion of approximation might be changed and the filter then redesigned and reevaluated [3]. If the desired amplitude frequency response is &(U) and the allowed class of filters is the class of length-N linear phase FIR filters, the average integral squared approximation error can be defined as

-1

- 0 . 2 n h -

o:i

o.is

oi

o.is

0) :

0,;s

o i

0 . 4 5 4 5

Frequency in H2

Fig. 1 . Ideal frequency response with no transition band. f0

0.225 Hz.

where A ( @ )is the amplitude response of the actual filter calculated from (8) or (10). Using Parsevals theorem [lo], the error defined above can also be expressed in the time domain by
m

which gives

h,(n)

= .

sin (won) nn

6 =

n = -m

C ~h,(n) - h(n>I2

(15)

where the infinitely long sequence &(n) is symmetric about n = 0 and is the inverse discrete-time Fourier transform of &(a) given by

As shown in (17), the length-N filter h ( n ) which has a frequency response A ( U ) that is an optimal least squared error approximation toAAd(w) obtained by a simple symis metric truncation of hd(n) to give a l e n g t h 4 sequence h(n). The truncated impulse response must be shifted to make it causal, giving sin (wo(n - M)) for 0 5 n otherwise One can note that the filter coefficients are the Fourier series coefficients of an expansion of the periodic frequency response &(CO), but we want to emphasize that for the filter design problem, this is a result of the originally posed least squared error approximation rather than the other way around. The method should be thought of as a least squares approximation problem rather than a Fourier expansion method to keep in mind the criterion of optimality and to allow generalization. The minimization of the squared error causes ripples or oscillations in A (U) near the discontinuity of Ad (U). This is known as the Gibbs phenomenon [IO] and the size of the ripple nearest to the discontinuity is approximately 9 % of the discontinuity. This ear or overshoot is necessary to allow A ( o ) to rapidly change from one to zero. As the length N of the filter is increased, the width of these oscillations decreases, the area under them decreases, and, therefore, the integral of the squared error decreases, but the maximum height remains at approximately 9 % . Any change of the problem to reduce the overshoot that does not remove the discontinuity in & ( U ) will necessarily increase the total squared error and destroy the optimality of the original approximation. A high-pass filter can be obtained from the low-pass prototype given in (20)by either subtracting the ideal lowpass frequency response from one, or by moving the ideal frequency response to be centered around w = n rather

( N - 1)

and where h (n)isJhe shifted set of filter coefficients being sought. Because h ( n ) is finite in duration, for odd N the summation in (15) can be divided into two parts as
M
E

(20)

C n = -M

I$(n) - h(n)(*+

n=M+1

2hd(n). (17)

This exprysion clearly shows that to minimize E , the N values of h(n) sh?uld be chosen equal to the corresponding N values of hd(n). The residual approximation error for the optimal h ( n ) is then given by the second summation. Although shown here for N odd, symmetric truncation also yields an optimal approximation for N even [3]. IV. APPROXIMATION A N IDEALLOW-PASS OF RESPONSE The basic ideas of frequency domain approximation will be developed for an ideal low-pass filter. The amplitude frequency response of an ideal low-pass digital filter is shown in Fig. 1 . This ideal response is unity for frequencies between zero and wo and is zero for frequencies between wo and n (or, fo and 0.5 in hertz). The symmetric, infinitely long impulse response hd(n) that exactly realizes this ideal is given by the inverse discrete-time Fourier transform from (16) which for this & ( a ) becomes
(18)

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1.2

than w = 0 by multiplying h ( n ) by (- 1)". A bandpass or band reject filter can likewise be constructed from a lowpass prototype by subtraction and/or modulation. This process is possible for a straightforward integral squared error approximation because of the linearity and orthogonality of the Fourier transform. It is not possible if a Chebyshev error measure is used except for specialized cases. Unfortunately, this analytical approach to the least integral squared error approximation of an ideal frequency response does not allow independent control of the pass and stopbands. Independent control is possible by using a weighted squared error, but that requires numerical solution of simultaneous equations, which is covered in Section IX. V. THE SPLINEFUNCTION TRANSITION BAND In order to eliminate the Gibbs' phenomenon, add flexibility to the design process, and retain the optimal approximation, a transition band is introduced in the ideal frequency response Ad(w).A transition function is defined to continuously connect the passband value of one of the stopband value of zero which removes the discontinuity in Ad(w)that caused the overshoot in A ( w ) . This is shown in Fig. 2 . Choosing a transition function which makes Ad(w)continuous,&not only removes the Gibbs' phenomenon, but causes h,(n) to asymptotically decrease faster as n increases. This means the truncation of hd(n) may cause less approximation error but perhaps only for large N . If the transition function is chosen such that Qth gerivative of Ad(w) exists everywhere in 0 Iw I a, h,(n) will asymptotically decrease at least as fast as 1 / n Q +I [ 101. Because of this and mathematical tractability, we choose spline functions [ 113, [ 121 as attractive transition functions [3], [13]. If the transition band is defined from w I to w2 and a first-order spline [3]-[17], [18] is used for the transition function as shown in Fig. 2 , the ideal amplitude response is given by
0 5 W I W ~

a"

-200 .

r
0.2
OI

0.05

0.1

0.15

0.2

0.25

0.3

0.35

0.4

0.45

Frequency in Hz

Fig. 2 . Ideal frequency response with first-order spline transition band. f0 = 0.225. d = 0.05 Hz.

where
WO

w2
~

+
2

WI

is the average of the passband and stopband edges and

A =

~2

- WI

(23)

w2CEwIa

which is a continuous function but its derivative is not continuous. Using the inverse Fourier transform in (16) to calculate hd(n)gives
a

cos (wn) d o which after several operations becomes

hd(n) =

an

(22)

is the transition band width in radians. Where it is not clear from context, a subscript will indicate whether the units are radians per second or hertz. Notice that (22) has the form of the h d ( n )obtained from (19) without a transition band multiplied by a weighting function. Indeed this is the same form as a window method, but comes from a very different formulation of the problem. A similar weighting function was used by Ormsby in a filter design problem [ 141, by Hamming [ 171 who credits Lanczos [19], and by Gautschi [20] in improving the convergence of the Fourier series, and by Merserau et al. [18] in modifying the use of the Kaiser window. We also see that by making A,(w) continuous, hd(n)now asymptotically drops off as 1 /n' rather than as 1/n. Equation (22) can be derived by a simpler approach than direct integration. If a rectangle of width A and height l / A is defined in the frequency domain, the ideal frequency response with the transition function shown in Fig. 2 can be obtained by convolution with the simple ideal frequency response in Fig. 1 . Convolution in the frequency domain corresponds to multiplication in the time domain, therefore (22) can be obtained by multiplying the inverse transform of the rectangular pulse times the inverse transform of the ideal frequency response with no transition band which is given in (19) with a band edge wo which is the average of w I and w2. That is exactly the form given in (22). If a second order spline (two sections of parabolas) is used for the transition function, the ideal frequency response can be constructed by convolving an ideal, notransition-band frequency response with a triangle pulse of width A which in turn can be calculated by convolving two rectangular pulses of width A / 2 . In the time domain

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LSE FIR FILTER DESIGN

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this is done by multiplying (19) by the square of the inverse transform of the half width-pulses. This is easily extended to a pth order spline [21], [3], [ 131 with the impulse having the form
(24)

This impulse response asymptotically drops off as 1 /np I and is obtained by simply multiplying the original notransition-band impulse response by a sinc function raised to thep power. This more general form of weighting function was also used by Schussler and Steffen [21] in addressing an interpolation problem. The effects of p on the approximation ar? somewhat subtle. Although the asymptotic dropoff of hd(n) is greater for larger p , the length where the asymptotic behavior becomes dominant becomes very large. In all cases there is a particular value of p that results in the integral squared approximation error for a given N and A and that value should be used. There are other transition functions that could be used. The impulse response for a transition function that is a raised cosine is given by sin (won) cos (An/2) h,(n) = Rn 1- (A/P)~~~

an analytical method for design that is as simple and efficient as the window method yet gives explicit control over the transition band and separates it from the truncation process while keeping a well defined measure of optimality. The spline power p is a parameter that can be used to control the nature of the approximation, much as the parameter fi in the Kaiser window can be used. This method can also be used in conjunction with the methods described in Sections VI1 and VIII. VI. ANALYSIS SPLINEFUNCTION OF TRANSITION BANDS Designing FTR filters using the spline transition function method requires understanding the relationships among wo, A , N, and p . Normally, wo is given by the application and cannot be altered by the filter designer, A is somewhat fixed by specifications but can be changed by the designer in some cases, N is also loosely fixed by execution speed, memory considerations and group delay constraints, and the spline order p is completely free to be chosen by the designer to further reduce the approximation error or achieve other goals. In using spline transition functions, the modification of the ideal hd(n) to eliminate the Gibbs effect is a separate process from the truncation which gives the causal filter. Hence, an important question is what length is best for a given &(U) and p . The answer is seen by considering the integral squared approximation error as a function of the filter length N . The error defined in the frequency domain by (14) is more easily and accurately evaluated in the time domain from the second term in (17). For small error Values, it is difficult to accurately evaluate (14) numerically while it is easy to evaluate (17). This error is shown as a function of N for several values of p in Fig. 3. The transition width is A, = f2 - fi = 0.05 Hz and the nominal band edge isfo = 0.225 Hz assuming a normalized sampling rate of one sample per second. This display shows the interesting and important results that there are clearly preferred lengths and spline orders. For a particular p , the error monotonically decreases (as it must) but there are regions that are relatively flat, indicating that increasing the length in those regions does not decrease the error significantly. On the other hand, there are regions of rapid decrease between the flat regions. It is, clear that one should truncate the infinitely long ideal hd(n) just to the right of one of the regions of rapid decrease to get the maximum effect of each term of R (n). The reason for the shape of the error curve is seen from (22) and (24) where the ideal hd(n) is multiplied by the sinc weighting function. The flat regions of the error curve correspond to the regions near the zeros of the sinc weighting function. The zeros of the sinc function are functions of the ratio of A which is the transition band width to p which is the spline order. It is this combination of the spline order and the transition band width that primarily determines the best filter lengths. The width of the flat regions increases for increasing p .

(25)

and for a raised cosine plus first-order spline that will give an additional continuous derivative
h,(n)
=
~

sin (won) .sin (An/2) rn An/2

1[

1(26)

These results can be easily extended. The spline transition functions could be built up by convolution of different width narrow pulses. The transition function could have several sections of trigonometric functions or a mixture of splines and trigonometric functions of various orders. Initial investigations do not indicate how these would be chosen nor do they indicate that more complicated transitions functions would have any advantages. Although derived and illustrated for odd lengths, the formulas also hold for even length filters when shifted and truncated as was done for the simple filter in (20). For the case of an even N , M would be an odd multiple of a halfinteger. The derivation involves interleaving zeros between terms in the even length h(n) and then shifting to make it symmetric. This is explained in [3]. An important new extension of the design method presented in this section is allowing noninteger values of p. Although the ideas of convolving rectangles and of polynomial splines are no longer valid, the function in (24) is well defined and has properties that are obvious interpolations of those for integer values of p as long as the sinc function is positive. This property is developed and used later in this paper when choosing an optimal order p. The use of spline transition functions developed in this section gives a powerful and flexible design tool. It gives

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r I

1
08-

0.05

01

0.15

0.2

0.25

0.3

0.35

0.4

0.45

Filter Length

Frequency in Hz
=

Fig. 3 . Integral squared approximation error versus filter length. f 0 0.225, d = 0.05 Hz.

Fig. 5. Ideal frequency response with spline transition function. f0 0.025, d = 0 . 3 H z . f l = O . l , f 2 = 0.4 Hz. p = 1, 2, 8, 100.

1
5
w

2 7 _1 ____7

s
-25 0

10

20

30

40

50

60

70

:;;I,,
-160

_ 0.2 _
0.3

,~-_-~._-.
0.2 0.4 0.25 0.3 0.35 0.4 0.45
0.5

0.05

Ill

0.15

Order of Transition Spline Function

Average Filter Band Edge f0

Fig. 4 . Integral squared approximation error versus spline order. f 0 = 0.0225, d = 0.05 Hz.

Fig. 6. Integral squared error versus band edge. N

129, p

4.

The effect of the spline order are easily seen in Fig. 3 where it is clear that the best p depends on the chosen length and transition bandwidth, which in turn depends on the smallness of integral squared approximation error desired. Displaying the same data in a different form in Fig. 4 shows the importance of the spline order. The longer filters have a rather sharp minimum indicating that care should be taken in choosing p . Notice that the value of p giving the minimum error for a particular length N is approximately a linear function of the length. That property is exploited in Section XI where a method for choosing the optimal p is given. The relationship of spline order and transition band width is somewhat complicated because of the nature of splines. Because higher order splines match more derivatives of the spline to A d ( w ) at w l and w2, there is a narrowing of the effective transition bandwidth. A high order spline with a given A is similar to a first-order spline with a narrower A . This is illustrated in Fig. 5 where the ideal &(a) is shown f o r p = 1, 2, 8, and 100. Indeed, as p goes to infinity, A d ( w ) goes back to the ideal rectangle function with a band edge at wo illustrated in Fig. 1. Fig. 6 shows that the integral squared approximation error is essentially independent of w, if A and N are fixed. This characteristic is clear from (24) where wo shows up

only in the sine term which simply oscillates as a function of n.

VII. ZERO ERROR WEIGHTING IN BAND

THE

TRANSITION

For certain applications it is desirable to minimize the integral squared error over part of the total frequency band 0 Iw < n but omit any contribution from the remainder [22]-[24]. This is done for the case when the Chebyshev error is minimized using the Parks-McClellan algorithm. For example one may desire to minimize the error over the pass and stopband of a filter response, and have no measure of error over the transition band. This idea is now developed for the squared error criterion by a mixed analytical and numerical method. If the pass and stopbands are truly all that are of interest, then rather than choose a spline transition function for the ideal A d ( @ ) , the error measure is modified to include no contribution from the transition band. This modification of (14) is
E = T

[1
O

WI

(&(U)

- A ( @ ) ) dw

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or, using a more general notation,

E

1 jn [ A d @ )- A(w)I2 d w a

(27)

where the integral is over all frequencies Q included in the error which in this case are Q = (0 Iw Iwl)U (a2 Iw Ia). Unfortunately, using this error measure imposes a metric such that the Fourier transform basis functions are no longer orthogonal, Parseval's theorem does not hold, and, therefore, simple truncation does not result in optimal approxim!tions. Let h (n) be the odd length N FIR filter response shifted to be symmetric about n = 0 and with an amplitude A ( o ) given by
A(w)
=
n

a ( m - n)

-M

h ( n ) cos (wn)
cm%n=

w ) + sin ((m + n )a (lm-+sin ((m + n)w2) n)

or using (10)
A(w)
=

n= I

which from (13) is

A(w) =

n=O

c 2h(n) cos (wn)

(28)

that minimizes the error of (27). To find this minimum, substitute (28) into (27) and set the gradient of E equal to zero by

2= 2 a
ai(m)
a
*

j, [A&)

c 2h(n) cos (wn)

M
n=O

1
(29)

which are elements of a symmetric matrix in the form of the sum of a Toeplitz matrix and a Hankel matrix [25]. We now define a modified matrix as having its first column half of the first column of C and all other columns equal to the corresponding ones in C. This absorbs the effects of the center term in an odd length h (n)not coming in equal pairs. This modification is not necessary for even length filters. Using this modified matrix (31) becomes

cos (wm) d w

which gives M 1 equations for m = 0, 1 , . . . , M . After multiplying both sides by C / 2 n and reversing the l order of integration and summation, (29) requires for each m

hd =

The M 1 elements of h in (33) are the terms in h ( n ) , the optimal FIR filter impulse response which minimizes (27) and which are found by solving the M 1 equations using

h
=
n=O

= c-lhd.

c &(n)a jn cos (wn) cos (wm)dw

hd =

(30)

which in matrix form is

CA

(31)

where h d is the vector with elements h d ( n ) which are the inverse Fourier transform of &(U) with the response in the transition band set to zero. For the response shown in Fig. 2, this is cos (wn) do =
~

Notice that if there is no transition-band, w 1 = w2, = I , and there is no modification of h d . A similar formula was also derived by Tufts et al. [23], by Schiissler [26], and by Oetken et al. [27] in addressing similar problems. An alternate derivation of (34) gives additional insight into this important result and provides a structure for evaluating the integral squared approximation error in the time domain. Define a new ideal amplitude response over the complete frequency range of 0 Iw Ia :

sin ( w i n ) . an

A(w)
(32)
&(U)

c 2h(n) cos (wn) + h(0)

' a + wl -

for n f m
w2

sin (2w1n) - sin (2w2n) 2an forn

=

m # 0

ch.
+

(33)

(34)

I0 Iw1

<

<

wz

(35)

w2 Iw Ia

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D is modified in the same manner as C to correct for the

effects of an odd length filter by replacing the first column by one half the former values to give D . Using the decomposition in (36) allows (42) to be written as

ho(n) =
where

S'

&(U) COS (an) dw =

R,(n) -t h,(n)

(36)

hf = D[hd
Solving for h ( n ) as

ht and

substituting in (42) gives the optimal

f; = h d

+ hf = h d + (1
R
=

+n

j"

or
&(U) cos (wn) dw

(37)

W2

which reduces to

h
and
W2

h,(n)

A ( w ) cos (wn) dw.

WI

(39)

The sequence h,(n) cannot be found directly since A ( q ) is ngt yet known. Substituting h which is the truncated ho = (hd h,) into (28) and that into (39) gives

.
R

h,(n) =

1 Ic
rw2 r

24(m) cos (om) cos (wn) dw.

WI

m=O

Reversing the order of integration and summation gives

M

h,(n) =

2
m=O

h(m) -

i s:

cos (om) cos (wn) dw

(41)

which can be written in matrix form as

hf = Dh
where D is an M

(42)

+ 1 by M + 1 matrix with entries

U2

d,,,n =

cos (wn) cos (om) dw

WI

(43)

Equation (45) is seen to be equivalent to (34) by noting that + D = Z from (30) and (43). This second formulation of the error minimization with zero error weighting over the transition band retains the orthogonality of the Fourier transform basis functions by defining the error integral over all of the frequency band 0 i w Ia. Because of this, Parseval's theorem can be used to derive a time domain formula for the approximation error. This is done in Section VI11 by summing twice the squares of &n) 6, ( n ) from M + 1 to a large value as indicated in (17). The design of even length symmetric and even or odd length antisymmetric filters do not require the definition of h ( n ) or of or and, therefore, are somewhat more straightforward than for odd symmetric length filters. Since the squared error is calculated over Q , when comparing this error with values averaged over the complete zero to a range, it might make sense to multiply this error by n / Q for a fair comparison. The design method of this section is a remarkable result. It gives an exact solution to the least integral squared error approximation with a zero wejght transition band by using an analytical solution for h d ( n ) and numerically modifying it with a matrix C which can by analytically derived. It also allows an accurate calculation of the approximation error which is very important in the design process. Although illustrated here for a low-pass filter,

where A o ( o ) = A (U) over the transition band and, therefore, results obtained in a least squared error approximation of Ao(w) over 0 5 w 5 n using (14) is the same as a least squared error approximation to Ad (U) over the pass and stopbands using (27). A ( w ) is the actual frequency response of the specific l e n g t h 4 filter. Using this trick, we have restored the orthogonality of the cosine functions by using the full frequency band, yet have an error contribution only over the pass and stop bands when ho(n)is truncated to length N . The approach taken here is to calculate an infinitely long ho(n) which is the inverse Fourier transform of A o ( w ) , then truncate it to give a length-N h ( n ) which has an actual frequency response of A (U). Because of the linearity of the Fourier transform, the impulse response corresponding to this particular A o ( o ) can be written as a sum of two terms

which after integration gives sin ((m - n)w2) - sin ((m - n)wJ

dm,n

I +
w2

n(m
sin ( ( m

n)

+ n)w2) - sin ( ( m + n ) w l ) n ( m + n)
for n # m

- wl

sin (2w2n) - sin (2wln) 2 an

forn

m # 0

ht]

D[hd

+ ht].
D)-'Dhd

(.) 44

[I
(I

+ z - D]h,
-

D)-%,.

(45)

BURRUS et al.: LSE FIR FILTER DESIGN

1335
0,

the approach can be applied to any problem where A d ( w ) has an analytic inverse Fourier transform. There is no question of choosing a spline or other transition function to use with this method, but unfortunately, we now have no control over the transition band and must solve M 1 simultaneous equations which are ill conditioned for large N a n d A.

VIII. ANALYSIS ZERO ERROR OF TRANSITION BANDS Design using the zero error transition band from Section VI1 removes the spline order p from the design consideration. Here it is wo, A , and N that must be chosen with N and A being more flexible than wo in most cases. Recall that changing the clock rate or the sampling rate for a fixed wo has the same effect as changing wo. This option is sometimes overlooked. When no transition band error is used, as was the case in Section VII, the relationship of integral squared approximation error to filter length and to transition band width is shown in Fig. 7. Comparing this with the results in Fig. 3, we see that the error for zero error transition band is somewhat below that for the spline case, even if the optimal spline order is chosen for each length. While this method directly addresses the desired approximation regions, it has numerical problems for longer lengths and wide transition bands. Even the efficient algorithms used in Linpack [28] and Matlab [29] have trouble with filters where NA > 12 because (31) and (42) are near singular. The erratic behavior of the error curves in Fig. 7 for A = 0.2 and A = 0.3 is caused by numerical errors in solving (34) or (45). If the frequency response of filters designed with Nand A in these regions is calculated, the effects of the numerical errors show up only in the transition band. Alternative algorithms [25], the use of a small transition band weight as developed in Section IX, and the approach suggested by Vaidyanathan and Nguyen [30] of posing this approximation as an eigenvalue problem have been tried, but all have numerical problems for large A and N. Notice that the error curves are fairly straight lines on the semilog plot. This indicates an empirical formula could easily be derived to relate the squared error, length, and transition bandwidth. This would be useful in choosing a length from given specifications and to better see the tradeoffs involved. From the error analysis of the zero error transition band case, it can be seen that much of the error for the spline transition function approximation may come from the transition band rather than the pass and stopbands if the spline order is not carefully chosen. Fig. 8 shows that, similar to the case with the spline transition function, there is essentially no error dependency on the average band edge wo, therefore, only the effects of A and N need be analyzed. Although the design method of this section is developed to minimize the integral squared approximation error, it is interesting and useful to also analyze its Chebyshev error characteristics by examining the frequency

.2O0L

20

40

60

80

.-

100

120

Fig. 7 . Pass and stopband integral squared approximation error versus N . f0 = 0.225 Hz.

0-

! 3
c

.5L

B g

0.05
0.1

-10-

;
s
n
-15-20 -

s
0.1s

Fig. 8 . Pass and stopband integral squared approximation error versusf0. N = 33.

Filter Length N d=O

0.05
0.1

0.15

0.2

0.25

0.3

0.35

0.4

0.45

0.5

Average Passband Edge f0

Frequency in Hz

Fig. 9 . Frequency response of FIR filter with zero error transition band. N = 3 7 , f O = 0.25, d = 0.1 Hz.

response curve. Fig. 9 shows the magnitude frequency response of a length 37 linear phase FIR low-pass filter withfo = 0.25 Hz and the transition bandwidth A = 0.1 Hz. These are similar specifications to those for the example of Oppenheim and Schafer using the Kaiser window in [ 1, p. 4561. If there is no error measure at all in the transition band, the frequency response in this band could become unde-

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lEEE TRANSACTIONS ON SIGNAL PROCESSING, VOL. 40, NO. 6, JUNE 1992

sirable in order to minimize the pass and stopband errors. This happens under certain circumstances in the Chebyshev design method. If the frequency response of filters designed with N A > 12, where the simultaneous equations are ill conditioned, is examined, all of the numerical errors seems to show up as frequency response errors in the transition band. It is not clear, at present, why this happens. IX. A WEIGHTED INTEGRAL SQUARED ERROR CRITERION In some applications it is desired to design a linear phase FIR filter using a weighted squared error [22], 1311, [32] of the form

Using the same approach that we used in (29) gives

error is the second term in (17) and the hd(n)is calculated from (22) or (24), or perhaps (25) or (26). This is not easily done for the case of the zero error transition band described in Section VI1 or the weighted error method of Section IX. Because of the error definition in (27), the filter cannot be found directly by truncating a single ideal long filter and the error cannot be found from the truncated part. The problem is solved by using the powerful trick described in the second error formulation in Section VII. An ideal frequency response A o ( w ) is defined by (35) such that the actual frequency response of the finite filter A ( w ) and A o ( o ) are equal over the transition band. Because of this definition, truncation of the infinitely long inverse Fourier trapsform ho(;) is the optimal finite length impulse response h ( n ) . This ho(n) is defined as a sumpf ! O components as was done in (36). We partition the ho, W h d , and h, which are defined in (36)-(39) into finite length M 1 vectors and their infinite remainders by

j, W2(w)A,(w)cos
=
n=O

(wm) dw

h ( n ) = ho(n)

foro

In IM

c R(n) n

ho2(n)= ho(n + M
(47)

+ 1)
o

for 0

In

(52)

W2(w)cos (wn) cos (am) do

h,](n) = h,(n>
h,,(n> = h,(n
h,,(n>
=

for

I n 5 :

(53)
In

which in matrix form is

+ M + 1)

for

(54) (55)

h, =
where

cwh

(48)

h,(n)

for 0 5 n cc M

h,,(n) = h,(n

h , ( n )= -

7 F a

'S

W2(w)A,(w) COS (wn) dw

(49)

and the elements of the M

Cwm.n

+ 1 by M + 1 matrix C, are
(50)

A rectangular 00 uating (43) for 0 In I03 and halving the first column. The lower partition of this matrix is defined as 0 2 with elements

+ M + 1) for 0 s n. (56) by M + 1 matrix D is defined by eval+ M + 1, n). +D2h

I r a

W 2 ( w )cos (an) cos (am) do.

d,(m, n) = d ( m

(57)

Solying (48) for the optimal integral weighted squared error h (n) gives

Using the definitions in (36) and (42), we have the result we are seeking
hO2

hd2

(58)

c;Ihw.

If the integrals in (49) and (50) can be analytically evaluated, the solution of the weighted squared error approx1 equaimation is again easily obtained by solving M tions. Notice that if a low-pass filter is being designed with a different cpnstant wFighting function in the pass and stop band, h,(n) = h,(n) and the effects of the weighting show up only in C, which is easily calculated.

which allows the error to be easily calculated from the inner product
E =

2h,T,hO2

(59)

Yithout further solving any siI;nultaneous equations if h(n) has already been found. If h ( n ) has not been found, then (33) can be used and the squared error can be found from (59) as
hO2

= hd2

+ DC-'hd,

(60)

OF ERROR X. CALCULATION THE APPROXIMATION Efficient and accurate evaluation of the approximation error for a filter design is necessary for analysis and for choice of the filter parameters. For optimal approximation to the ideal frequency response with a spline transition function, the error calculation is simply the sum of the squares of the analytically determined ideal impulse response from beyond the filter length to a large value. This

and (60). This will obviously be much slower than calculating the error for the spline transition function because M + 1 simultaneous equations must be solved for each length N . It may be possible to speed these calculations by using the special structure of C [25]. The approach described in this section using the artificially defined A o ( w ) can be used to find the integral

BURRUS er a l . : LSE FIR FILTER DESIGN

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squared error of any length-N linear phase FIR filter. If the filter is not obtained by truncating ho(n),the first term in the error sum of (17) is not zero and must be included. For the general case, ho(n)is found from

Solving for p as a function of N gives

ho = h,

+ Dh

(61)

where D has more rows than columns in order that h, be long enough to give a good determination of the error. The error is then calculated from (15) by
E =

2(h, -

h)T(ho- h )

(ho(0) - / ~ ( 0 ) ) ~ .(62)

The last term in (63) corrects for the fact that there is only one h(0) term in (15) pr (17), but all the other terms occur in pairs. Recall that h ( n ) is finite in length as explicitly shown in (17). This method of calculating the integral squared error can be used to evaluate and compare filters designed by other methods such as with windows or the Parks-McClellan algorithm. It is used in the next section to calculate the integral squared error over the pass and stop bands of the spline transition function which are approximations over the total band. XI. DESIGN USINGOPTIMAL SPLINE TRANSITION FUNCTIONS This section contains a principal contribution of the paper which is an FIR filter design method using the spline transition function developed in Section V, but with an optimally chosen spline power p to give the least approximation error. Although the spline function used as a transition function was developed and defined in terms of sections of a pth order polynomial and was constructed by convolving rectangle functions together p times, the formula derived in (24) can be meaningfully evaluated for noninteger values of p . The function has properties that are the expected interpolations of those using integer values of p . The approximation error curves for noninteger p fall between those given in Fig. 3 and form a continuum of curves with an optimal p existing for each filter length N. A problem could exist for a noninteger p is the sinc function which is raised to the p power is negative. Fortunately, that does not occur in our method. An optimal spline transition function can be selected by finding the value of p that minimizes the integral squared approximation error as illustrated for integers in Fig. 3. This value gives an error curve with its knee at the desired length N. This will occur at a value of N equal to or slightly less than that given by the first zero of [sinc (An/2p)lP. This is when
-

Substituting this into (24) gives a formula for the optimal spline FIR filter design method. The accurate value of p that minimizes the error is not easy to find analytically because of the denominator of the sinc function and its being raised to the pth power. Numerical experimentation with error analysis of the fixedp spline design similar to that shown in Fig. 4 but with a wide variety of N, A, and wo indicates the value of p that minimizes the total integral error of (14) is fairly well approximated by p
=

0.62 AfN

(66)

with A given in hertz. If only the pass and stopband errors are of interest, a slightly different formula forp should be used to minimize error defined in (27). Initial experiments show a value of A slightly larger than ( f 2 - f , ) should be used and that the value of p , using A = 1.12 ( f 2 - f,), that minimizes the pass and stopband errors is approximated by p = KAN where 0.453 (67)

+ 0.386/NA

NA cr 1.25 1.25 5

K =

0.774 - 0.0251NA [0.648

< NA < 5 (68)

< NA.

An

2P

=7r

(63)

or when the total length of h ( n ) is

47rP N=---+l. A

The results on these formulas are preliminary and further work is being done to develop empirical formulas for p that will be more accurate. We will evaluate this new method by considering the integral squared approximation error which is the focus of this paper. Care must be taken in comparing errors defined with the two different criteria of (14) and (27). In order to allow meaningful evaluation and comparisons of an averaged error, the results calculated from (27) are multiplied by T / Q to give the average over pass and stopbands and not include the transition band. A fair comparison of a weighted error in (46) would be even more difficult. We will use as an example an ideal frequency response withf, = 0.225 and a transition band defined by f l = 0.2 andf2 = 0.25 Hz. Fig. 10 presents four approximation errors as a function of filter length. The upper curve is the total approximation error given by (14) for the ideal response having an optimal spline transition function with the order given by (66) which minimizes the total approximation error. The second curve is the approximation error of the same filter, but calculated only over the pass and stopband by (27). The third curve is the error given by (27) but with A = 0.056 HZ and p chosen by (67) to minimize the pass and stopband error. The fourth and lowest curve in the pass and stopband error of the optimal filter designed by the numerical

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IEEE TRANSACTIONS ON SIGNAL PROCESSING, VOL. 40, NO. 6. JUNE 1992

Total Error with Spline TF

PB+SB Error with Spline TF, d

1.12(f2-fl)

20

40

60
Filter Length

80

100

120

Frequency in Hz
=

Fig. 10. Integral squared approximation error versus filter length. f 0 0 . 2 2 5 , f 2 - f l = 0.05 H z .

Fig. 11. Frequency response of FIR filter with optimal spline transition function. N = 3 7 , f O = 0.25, d = 0.1 Hz.

method of (34) or (45). The errors of the lower three curves were all multiplied by a/Q to allow fair comparison with the upper curve. The maximum length was limited by the numerical problems when obtaining the fourth curve. For lengths less than N = 25, the total integral squared error over 0 Iw Ia for the optimal spline design is less than the error of the same filter averaged over Q = (0 I w Iwl) U (w2 Ia . This indicates that the shorter spline ) designs have less of their error in the transition band than the pass and stopbands and the longer ones have more. Examination of these four error curves shows several interesting features. For lengths less than N = 40, the optimal spline and the optimal zero-error transition band methods have essentially the same approximation error. Above that length, the errors diverge somewhat. Examination of the frequency response of these longer filters show that according to (66), the optimal spline transition function has a higher orderp, and, therefore, looks like a narrower transition band as was seen in Fig. 5 . In other words, lengthening the optimal spline filter both decreases the approximation error and decreases the effective transition bandwidth to less than A while the optimal zero-error transition band method only decreases the original pass and stopband error. Recall that in the limit, as N 03, the frequency response of an optimal spline filter will approach the ideal of Fig. 1, even for a nonzero A, while that is not the case for the optimal zero-error transition band design. This effective narrowing is taken into account by using a A larger than (f2 - fi) in curve three which gives results close to those of the numerical method used to obtain the filter for curve four. As was true for the fixed order spline method and the zero-error transition band of Section VII, the approximation error of the optimal spline method is relatively independent of the average passband edge wO. Fig. 11 presents the magnitude frequency response of a length 37 FIR filter with the same band edge and transition bandwidth specifications as used for the zero-error transition band example shown in Fig. 9, but in this case with a spline chosen to minimize the squared error using
+

(66) to give p = 2.25. This response has Chebyshev performance which compares favorably with that of the filter designed by numerical methods to give zero error weighting in the transition band shown in Fig. 8 and with the design using a Kaiser window shown in [ l , p. 4561, and is much better than the results of any other type of window even though reducing the Chebyshev error was not the goal. The FIR filter design method presented in this section uses empirically derived formulas (66) and (67) to choose an optimal order spline transition function which is much simpler and faster to implement in (24) than the numerical methods necessary for the zero-error transition band method, yet gives approximations that are almost as good. It keeps a control over the amplitude response in the transition band and gives an effective narrowing of the transition bandwidth with very little sacrifice of the pass and stopband approximations. It has some of the optimizing properties of a numerical method and the simplicity of a windowing method.

XTI. CONCLUSIONS The use of spline transition functions to design linear phase FIR digital filters gives formulas with a similar form to those designed with window functions. However, the method presented in this paper is based on a different formulation. The multiplying functions should be viewed as weighting functions rather than windows. They are not specified in the time domain to gently truncate the ideal impulse response in order to reduce the Gibbs phenomenon. They are the result of a frequency domain specification of a transition width and function. The use of spline transition functions is not a modification of a least squared error approximation trying to partially achieve a Chebyshev design, but the construction of a desired & ( U ) such that an analytical least squared error approximation is possible. The use of windows (except for the Kaiser window) does give a tradeoff of approximation ripple and transition width, but it is an implicit control. The use of spline and other transition functions allows an explicit control of transition and re-

BURRUS et a/ : LSE FIR FILTER DESIGN

1339

tains the least squared error optimality. The ad hoc aspect of choosing the spline order p was removed by showing how to choose it to minimize the integral squared approximation error for a desired error or filter length. This paper has also shown how to design optimal squared error FIR filters with the error having no contribution from the transition band and, therefore, not requiring any kind of transition function. A similar formulation of the problem allows a weighting of the squared error over the frequencies of interest. This allows an exact minimization over only the frequencies of interest with explicit control of the error weighting with no add hoc choices, however, in comparison to the use of spline functions, it has two disadvantages. It gives no control over transition band behavior and it requires the solution of M simultaneous equations which are ill conditioned for filter with A N > 12. Error analysis methods are derived and used to show the relationship of the various parameters. It was shown that optimal choice of spline transition functions can give pass and stopband approximations almost as good as the zero-error transition band without requiring the solution of simultaneous equations and while maintaining control in the transition band. This gives a very powerful new linear phase FIR filter design method that has the simplicity of the window method yet has the optimality and explicit band-edge control for least squared error design that the Parks-McClellan method does for least Chebyshev error design. These methods using transition bands can be applied to even and odd order high-pass, bandpass, and band reject filters as well as low-pass filters. Because of the nature of the least squared error criteria used here, a low-pass filter can be mFved to a bandpass location by modulation (multiplying h ( n ) by cos (U,)) or by adding two or more nonoverlapping frequency responses and have the results still be optimal. This is not possible in general with a Chebyshev approximation. The approach can also be applied to differentiators [3], equalizers, Hilbert transformers, and other signal processing applications. Further work is being done in comparing the squared error of filters designed using the methods described in this paper with those designed by a minimum Chebyshev error method, and comparing the Chebyshev error of filters designed by both methods. Although this paper has stressed the design of optimal least squared error filters, one of the reasons for introducing the transition band was to eliminate the Gibbs phenomena and how well that was done needs further evaluation.

REFERENCES
[ I ] A. V. Oppenheim and R. W. Schafer, Discrete-Time Signal Processing. Englewood Cliffs, NJ. Prentice-Hall, 1989. [2] L. B. Jackson, Digital Filters and Signal Processing, 2nd ed. Boston, MA: Kluwer, 1988. [3] T. W. Parks and C. S . Burrus, Digiral Filter Design. New York: Wiley, 1987. [4] J. F. Kaiser, Nonrecursive digital filter design using the Io-sinh window function, in Proc. IEEE ISCAS-74, 1974, pp. 123-126; also DSP Committee, Ed, Digital Signal Processing I I , Selected Reprints. New York: IEEE, 1979. [5] Y. H. Ha and J . A. Pearce, A new window and comparison to standard windows. IEEE Trans. Acoust., Sueech. Sianal Processina. \.,oI. 37, no. 2, pp. 298-301, Feb. 1989. 1 W. Parks and J . H. McClellan, Chebyshev approximation for r. rionrecursive digital filters with linear phase, IEEE Trans. Circuit rheory, vol. 19, pp. 189-194, Mar. 1972. r. W. Parks, L. R. Rabiner, and J. H. McClellan, On the transition width of finite impulse-response digital filters, IEEE Trans. Audio ?lectroacoust., vol. 21, no. 1, pp. 1-4, Feb. 1973. <. Steiglitz, T . W. Parks, and J . F. Kaiser, METEOR: A conitraint-based FIR filter design program, to be published. <. Steiglitz and T. W. Parks, What is the filter design problem?, n Proc. Princeton Con$ Inform. Sei. Syst. (Princeton, NJ), Mar. 1986, pp. 604-609. 3 . C. Champeney, A Handbook of Fourier Theorems. Cambridge: :ambridge University Press, 1987. 1.de Boor, A Practical Guide to Splines, Applied Mathematical Sci:nces, 27. New York: Springer, 1978. P. M. Prenter, Splines and Variational Methods. New York: Wiley, 1975 (Wiley Classics Edition), 1989. Z. S . Burrus, FIR filter design using spline function transition Jands, in Proc. Asilomar Con$ Circuirs, Syst., Comput. (Pacific Grove, CA), Nov. 1988. I . F. A. Ormsby, Design of numerical filters with applications to missile data processing, J . Ass. Comput. Mach., vol. 8, no. 3, pp. 440-466, July 1961. I C. T.-T. Kao, The spline approximation in system simulation by digital computer, Ph.D. dissertation, Rice Univ., Dep. Elec. Eng., Houston, TX, 1972. H. W. Schussler, Digifale Systeme zur Signalverarbeitung. Berlin: Springer, 1973 (in German). R. W. Hamming, Digital Filters, second ed. Englewood Cliffs, NJ: Prentice-Hall, 1983. R. M. Merserau, Z . S . Shen, and M. H. Hayes, Effect of ideal transition specification on window designs, in Proc. EUSIP CO-88 (Grenoble, France), Sept. 1988; a shorter version was presented at the ASSP DSP Workshop, Lake Tahoe. CA, 1988. C. Lanczos, Applied Analysis. Englewood Cliffs, NJ: Prentice-Hall, 1956. Walter Gautschi, Attenuation factors in practical Fourier analysis, Numer. Math., vol. 18, pp. 373-400, 1972. H. W . SchuBler and P. Steffen, A hybrid system for the reconstmction of a smooth function from its samples, Circuits, Syst., Signal Processing, vol. 3, no. 3, pp. 295-314, 1984. 1P. E. Fleischer, Digital realization of complex transfer functions, Simulafion, vol. 6, pp. 171-180, Mar. 1966. D. W. Tufts, D. W. Rorabacher, and W . E. Mosier, Designing simple, effective digital filters, IEEE Trans. Audio Electroacoust., vol. 18. no. 2, pp. 142-158, June 1970. D. W . Tufts and J. T. Francis, Designing digital low-pass filterscomparison of some methods and criteria, IEEE Trans. Audio Elecrroacoust., vol. 18. no. 4, pp. 487-494, Dec. 1970. G. A. Merchant and T. W. Parks, Efficient solution of a Toeplitzplus-Hankel coefficient matrix system of equations, IEEE Trans. Acoust., Speech, Signal Processing, vol. 30, no. 1, pp. 40-44, Feb. 1982. Lehrstuhl fur H. W. Schiissler, Digitale Signalverarbeitung I!. Nachrichtentechnik, Universitat Erlangen-Nurnberg, Erlangen, Germany, 1978, 1988 (in German). G. Oetken, T. W. Parks, and H. W. Schupler, New results in the design of digital interpolators, IEEE Trans. Acoust. , Speech, Signal Processing, vol. 23, pp. 301-309, June 1975; also in DSP Committee, Ed., Digital Signal Processing II, Selected Reprints. New York: IEEE, 1979. J. J. Dongarra, J. R. Bunch, C . B. Moler, and G . W. Stewart., LINPACK Users Guide. Philadelphia, PA: SIAM, 1979.

ACKNOWLEDGMENT The authors appreciate the suggestions and contributions from J. F. Kaiser of Bell Communications Research, H. W . Schiissler of the University of Erlangen, C. M. Loeffler of the Applied Research Labs, Austin, TX, and the DSP groups at Rice University and M.I.T. The analysis and experimental work was done using the Matlab computer software system [29].

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[29] C. Moler, J. Little, and S . Bangert, Matlnb Users Guide. South Natick, MA: The Math Works, 1989. 1301 P. P. Vaidyanathan and T . Q. Nguyen, Eigenfilters: A new approach to least squares FIR filter design and applications, including Nyquist filters, IEEE Trans. Circuits Syst., pp. 80-95, Jan. 1987. [31] D. C . Farden and L. L. Scharf, Statistical design of nonrecursive digital filters, IEEE Trans. Acoust., Speech, Signal Processing, vol. 22, no. 3, pp. 188-196, June 1974. [32] V. R. Algazi and M. Suk, On the frequency weighted least square design of finite duration filters, IEEE Trans. Circuits Sysr., vol. 22, no. 12, pp. 943-953, Dec. 1975. [33] DSP Committee, Ed., Digital Signal Processing II, Selected Reprints. New York: IEEE, 1979.

1989. He received an IEEE S-ASSP Senior Award in 1974, a Senior Alexander von Humboldt Award in 1975, a Senior Fulbright Fellowship in 1979, and the IEEE S-ASSP Technical Achievement Award in 1985. He was named an SP Society Distinguished Lecturer in 1990-1991, CAS Society Distinguished Lecturer in 1991-1992 and was elected an IEEE Fellow in 1981. He is coauthor (with T.W. Parks) of two books: DFTIFFTand Convolution Algorithms and Digital Filter Design.

C. Sidney Burrus (S55-M61-SM75-F81) was born in Abilene, TX, on October 9, 1934, He re. ceived the B,A,, B,s,E,E., and M,S, degrees from ~i~~ University, Houston, TX,in 1957, 1958, and 1960, respectively, and the Ph.D. degree from Stanford University, Stanford, CA, in 1965. From 1960 to 1062, he taught at the Nuclear Power School in New London, CT, and during the summers of 1964 and 1965, he was a Lecturer in Electrical Engineering at Stanford University. In 1965, he joined the faculty at Rice University where he is now Professor and Chairman of Electrical and Computer Engineering. From 1972 to 1978, he was Master of Lovett College at Rice University. In 1975 and again in 1979, he was a Visiting Professor at the Institut fur Nachrichtentechnik, Universitat Erlangen-Nurnberg, West Germany, and in 1989-1990 he was a Visiting Professor at M.I.T. Dr. Burrus is a member of Tau Beta Pi and Sigma Xi. He received teaching awards from Rice University in 1969, 1974, 1975, 1976, 1980, and

Admadji W. Soewito was born in Yogyakarta, Indonesia, on May 3, 1955. He received the B.S. degree in 1980 from Bandung Institute of Technology in Indonesia, the M.E.E. degree in 1985 from Philips International Institute in the Netherlands. and the Ph.D. degree in 1990 from Rice University in Houston. TX. His research interests are in digital signal processing and filter design and he currently works for the National Electronics and Electrical Research Institute (Len-Lipi) in Bandung, Indonesia.

Rarnesh A. Gopinath (S90) was born in Palghat,

India, on November I , 1965. He received the Bachelor of Technology degree from the Indian Institute of Technology, Madras, India, in 1987, and the masters degree from Rice University in 1990, and is currently working on the doctoral degree in electrical engineering at Rice University. His research interests are in the areas of digital filtering, time-frequency analysis, wavelets and filter banks, and computational complexity theory