1122 J EEE TRANSACTJ ONS ON SI GNAL PROCESSING. VOL. 40. NO.

5, MAY 1992
A VLSI Architecture for Simplified Arithmetic
Fourier Transform Algorithm
Irving S. Reed, Fellow, IEEE, Ming-Tang Shih, Member, IEEE, T. K . Truong, Senior Member, IEEE,
E , Hendon, and D. W. Tufts, Fellow, IEEE
Abstract-The arithmetic Fourier transform (AFT) is a num-
ber-theoretic approach to Fourier analysis which has been
shown to perform competitively with the classical FFT in terms
of accuracy, complexity, and speed. Theorems developed in a
previous paper for the AFT algorithm are used here to derive
the original AFT algorithm which Bruns found in 1903. This is
shown then to yield an algorithm of less complexity and of im-
proved performance over certain recent AFT algorithms. A
VLSI architecture is suggested for this simplified AFT algo-
rithm. This architecture uses a butterfly structure which re-
duces the number of additions by 25% of that used in the direct
method.
I . INTRODUCTION
HE Fourier representation of a periodic signal is im-
T portant to Fourier analysis and the discrete Fourier
transform. The Fourier series of a periodic sequence cor-
responds to the discrete Fourier transform (DFT) of a fi-
nite-length numerical sequence. The most efficient method
for computing the DFT is the fast Fourier transform (FFT)
[I ]. However, computation of the FFT is still complicated
and time consuming, especially in terms of the number of
needed complex multiplications.
At the beginning of this century, in 1903, a mathema-
tician named H. Bruns [2] developed a method for com-
puting the coefficients of a Fourier series of a periodic
function using the Mobius inversion formula for series.
This technique of Fourier analysis was considered again
by Wintner [3] in a 1945 monograph.
A similar algorithm was rediscovered recently [4] by
Tufts and Sadasiv and called the arithmetic Fourier trans-
form (AFT). This AFT is based on the Mobius inversion
of series and can be used to compute finite Fourier coef-
ficients of even periodic function. The advantages of the
AFT over the FFT is that this method of Fourier analysis
needs only addition operations with one exception,
namely, multiplications by an inverse-integer scale factor
at one stage of the computation.
The AFT developed in [4] was extended very recently
in [5] to compute the Fourier coefficients of both the even
and odd components of a periodic function. This latter
algorithm can be used to find the Fourier coefficients of
any complexed-valued periodic function. The main draw-
back of this AFT algorithm is the oversampling problem,
i.e., the higher the required accuracy is, the greater the
number of samples needs to be. However, the AFT al-
gorithm can be used with sampling rates close to the
Nyquist rate (see [ 5] ) . Another disadvantage in this latter
algorithm is that the amount of computation needed for
the odd components of a periodic function is apparently
greater than for the even components. As a consequence,
the number of real additions needed to realize the N-point
Fourier coefficients of real-valued waveform in this latter
algorithm is N2/2, see [9], [lo].
This paper consists of three parts. First, the AFT al-
gorithm in the form developed in [5] is outlined. This al-
gorithm is used then to derive Bruns original method in
[2] for finding Fourier coefficients. It is shown here that
Bruns AFT algorithm is more balanced than the algo-
rithm in [5] in the amount of computation needed for com-
puting the even and odd coefficients of a Fourier series.
In fact, the matrix needed to compute the odd coefficients
is identical in Complexity to the matrix needed for the even
coefficients. Also, this new version of Bruns AFT tech-
nique is compared with the previous AFT algorithms in
[5] on the basis of accuracy, complexity, and speed. Fi-
nally, an architecture of this simplified AFT is suggested
in order to reduce the number of additions. This reduction
in the addition operations is accomplished primarily by
decomposing the Bruns alternating average into two or-
dinary averages. Then a systolic array structure is devel-
oped to propagate the required k-point averages into 2k-
point averages, etc. As a result, a parallel Fourier trans-
form is developed which is suitable for VLSI implemen-
tation.
Manuscript received J uly 31, 1989; revised February 12, 1991. This work
was supported in part by SRC 88-DP-075 and i n part by NASA 7-100.
I . S. Reed, M.-T. Shih, and E. Hendon are with the Department of Elec-
trical Engineering, University of Southern California. LOS Aneeles. CA
11. THE FOURI ER ANALYSIS BY USING THE MOBIUS
INVERSION FORMULA
Reed et al. [51 introduced the arithmetic Fourier trans-
-
90089-0272.
pulsion Laboratory, Pasadena, CA 91 109.
sity of Rhode Island, Kingston, RI 02881.
form as fol1ows:Let a real function A( t ) be a finite Fourier
series of period T. The Fourier series of A( t ) has the form
bn sin 2nnfot (1)
T. K. Truong is with Communication SystemResearch Section, J et Pro-
D. W. Tufts is with the Department of Electrical Engineering, Univer-
IEEE Log Number 9106572. n = I n = 1
N N
A( t ) =a. + c an cos 2nnfot +
1053-587X/92$03.00 0 1992 IEEE
REED er al . : VLSI ARCHITECTURE
A( 0. 2)
A (0.3)
A(0.4)
A(0. 5)
A(0. 6)
A(0. 7)
1123
'
where f, =1 / T . ao. the zeroth harmonic. is the mean of
A( t ) , and a, and b, are nonvanishing coefficients in the
interval 1 I n I Nand vanishing elsewhere. The zeroth
harmonic a,, in (1) is given by
a0 =T o j T A( t ) dt.
( 2)
If the mean in (1) is removed, one obtains
N
A(t ) =A( t ) - a. = C a, cos 2anht
n =l
N
+ b, sin 2nnfot. (3)
n =l
Now shift the periodic function A( t ) in (3) in time by an
amount aT, where - 1 <a <1, as follows:
A(t +a ~ ) =C a, cos 2 a n ( h t +a)
N
n = I
N
+ c b, sin 27rn(fot +a)
n = I
N
=C c,(a) cos 2anfot
, = I
N
+ d,(a) sin 2anfot (4a)
n =l
wherefor -1 <a <1
c,(a) =a, cos 2ana +6, sin 2ana
(4b)
and
&( a) =-a, sin 2ana +b, cos 2ana.
(4c)
Dejinition 1: Let S(n, a) be the nth average
1 ,-'
S(n, a) =- C A( mT/ n +a T )
( 5)
n m = O
of the n values A( mT/ n +a T ) for ( m =0, 1, 2, ,
n - 1) of A( t ) shifted by the amount aT where - 1 <a
<1.
The following two theorems proved in [ 5] contain the
mathematics needed to describe the recent versions of the
AFT algorithm as exposed in [3]-[5].
*
meorem 1 (5, theorem 41: The coefficients c,,(a) in
(4b) are given by the Mobius inversion formula for a finite
series as follows:
Wl n l
/ = 1
c,(a) = c p ( OS( l n , a) (6)
where S(n, a) is the nth average defined in ( 5) of Defini-
tion 1 and ~ ( ( 1 ) is the Mobius function defined by
p(Z) =1
if 1 =I
=(- 1)' if 1 =pIp2 . . pr ,
where the p , are distinct primes
=o i f p2 1 1 for any prime p .
meorem 2 (5, theorem 51: The Fourier coefficients a,
and b, of the Fourier series in (3) for n =2'! (2m +I ) are
computed by
a,, =C,(O) (74
b, =(- 1)'"~,( 1 / 2h+2) (7b)
and
where mand k are determined by the factorization, n =
2k(2m +1).
If nearest neighbor or zero-order interpolation is used,
it is shown in [5] that a simple matrix can be used to com-
pute the a, and that rlog2(N)1 matrices are needed to
compute the b, where rx1 denotes the least integer
greater than or equal to x. To illustrate this, let A( f ) be a
periodic function with period T =1. Also assume that 2N
=10 and At =T/ 2N =0.1. The mean value, i.e., the
zeroth harmonic a. of A( t ) , is
9
a0 = kzo 4 k A t )
and
A( kAt ) =A( kAf ) - a. fork =0, 1, 2, * * . , 9.
To compute a, for 1 I n I 5 by Theorem 2, let a =0.
If zero-order interpolation is used to obtain S(n, 0) for the
case 1 I n 5 5 , then S(n, 0) appears in matrix notation
as follows:
0 0
0 0
0 1
0 1
1 0
0 0
0 1
0 0
0 1
1 0
0 0
0 0
1 0
0 0
1 0
I] I O
1 0
IEEE TRANSACTIONS ON SIGNAL PROCESSING, VOL. 40, NO. 5, MAY 1992 1124
By (7a) in Theorem 2, the even Fourier coefficients are
[ 5 /,I
Next to compute b2 one needs k =1 so that CY =1 / 2
obtained as
=1/8. Then, from (7b), one obtains
a, =C,(O) = c p( l ) S( l n, 0).
I = I
Thus the even coefficients a, for n =1, 2, . .
obtained approximately in matrix notation as follows:
, 5 are
where S(2, 1/8) =[A( Od) +A(0.5)]/2 and S(4, 1/8)
=[A(0.1) +A(0.4) +A(0.6) +A(0.9)]/4. To com-
pute b4, again one must have k =2 so that CY =1/2
=1 / 16. Thus from (7b) one has
(8)
To compute bl , b3, and b5, note by (7b) in Theorem 2
for n =1, 3 , 5 , that k must be zero, i.e., k =0. Hence
(y =1/2k+2 =1/4 and
(1 1) b4 =(0 0 0 1 0)
fern=1,2;** , 5
or in matrix form,
0 0
0 0
0 0
0 0
1 0
0 0
0 0
0 1
1 0
1 0
0 0
0 1
0 0
0 1
1 0
1 0
1
A(0.2)
A(0.3)
A(0.4)
A(0. 5)
A(0.6)
A(0.7)
Also by (7b) one has
where S(4, 1/16) =[A(0.1) +A(0.3) +A(0.6) +
A (0. S)] /4.
In this example one observes that if nearest neighbor
interpolation of the sampling points is used, then the three
different matrices, given in (9)-(ll), are needed to com-
pute the odd Fourier coefficients b, for n =1, 2, 3 , 4, 5 ,
whereas by (8) only one matrix is needed to compute the
a,. In the next section Bruns original AFT is developed
from the above Theorems 1 and 2. A use of Bruns alter-
nating averages makes it possible to compute the coeffi-
i) . (9) cients a, and b,, each with a single matrix of similar com-
plexity. As a consequence, for Bruns version of the AFT
algorithm the computational complexity needed for com-
puting the even and odd coefficients is identical.
forn =1, 3, 5
so that in matrix form
S(1, $)
f:) =(% -% -:) ( f:
S( 5,
REED er al. : VLSl ARCHITECTURE
I I25
111. BRUNS ARITHMETIC FOURIER TRANSFORM AND ITS
GENERALIZATION
Bruns' original AFT in [2] used weighted, signed,
averages of discrete values of A(t) where the weights were
alternatively +1. For the original AFT, Bruns in [2] de-
veloped a special Mobius-type inversion formula for these
alternating averages. In this section Bruns' original AFT
is derived more directly from the formula for c, (a) in (6)
of the fundamental Theorem 1 of Section I1 [ 5, theorem
41. Then the Fourier coefficients a, and b, are shown to
have expressions which are equivalent to Bruns' original
formula.
Dejinition 2: Define B(2n, a) to be the 2nth Bruns al-
ternating average,
of the 2n values A(k/2nT +CYT) for (m =0, 1, 2,
. . . , 2n - 1) of A( t ) in (1) shifted by the amount aT
where -1 <CY <1.
Theorem 3: The coefficients c, (a) in (4b) are given by
the Mobius inversion formula for finite series as follows:
W/nI
c,(a) = C p((1)~(2n1, CY) (13)
1 = 1 , 3 , 5 ; . .
where B(2n, a) is the 2nth Bruns alternating average de-
fined in Definition 2.
Proof? It is readily shown that the coefficients c,(a)
in (4b) are periodic functions of a with period l /n, i.e.,
c,(a +l /n) =c, (a). Now shift the C, (CY) by a half-
period as follows:
c, a +- =a, cos (2ana +n) +b, sin (2nna +a)
( 2 3
=-a, cos 27rna - b, sin 27rna =-c,(a).
Hence by (14) c,(a) can be reexpressed as
The substitution of (5) and (6) into (15a) yields
m= O (nl 2n l )
In - I
- C A ~ T + - T + ~ T
In - 1
- C A (7 2m +1 T +a T ) ] .
m=O
In the latter summation l etj =2m +I, thenj and 1 are
both even or both odd.
In this last change of variables the last term in (15b)
becomes
nl - 1 - 2m+l
m=O
a ( j , 1 ) = C A (7 T +a T )
2(nl - I ) + I
; = 1
2nl - 2
= 2 / T +
j = l (2n1
2nl + / - 2
+ ; =C 2nl A( - &T +c Y T )
if j and 1 are both even;
2nl - I
;=/
2nl f 1 - 2
+j =2 n / +I c A ( - & T + ~ T )
i f j and 1 are both odd.
Hence, setting k =j +2nE in the second term of o ( j , l ) ,
given above, one has
2nl - 2
if k and 1 are both even;
2nl - 1 1- 2
= c A L T + ~ T + C A - T + ~ T
k = l (2d ) k = l ( l 1 )
if k and 1 are both odd.
Thus finally,
2111- 2
a ( j , 1 ) = c A (& +a T ) i f 1 is even.
k = 0 , 2 . 4 ; ' .
Hence, using the last result for a ( j , I ) in the last term
of (15b), the expression within the brackets of (15b) be-
comes
2m +1
In - 1 In- 1
m=O c A & T + ~ T ) - m=O c A ( ~ T + ~ T )
21n - 2
k = 0 , 2 , 4 ; ' .
In - I
2m+1
- m= O C A ( T T + C Y T )
=0 if 1 is even;
2n/ - I
- A (& T +CYT) if l i s odd.
k = 1. 3. 5 ' . '
I I26 IEEE TRANSACTIONS ON SIGNAL PROCESSING, VOL. 40. NO. 5, MAY 1992
Finally, a substitution of this result into (15b) yields
I
IN/,]
I = 1,3,5; ' .
2 h - I
* [ k = O , 2 3 , . . .
P"n1
= C p(1)~(2n1, a>.
I = 1. 3. 5, ' ' '
Hence the theorem is proved.
From (13), the c,(a) are calculated by the alternating
averages B(2n1, a), which are simple weighted sums of
sampled values of A( t ) . Thus the previous requirement to
have a zero-mean function A( t ) is eliminated in this cal-
culation. The Fourier coefficients a, and b, are computed
now by use of (13) in Theorem 3. These formulas corre-
spond to (7a) and (7b) in Theorem 2.
The above method for calculating the Fourier coeffi-
cients by (13), (7a), and (7b) is a generalized form of
Bruns' method [2] of Fourier analysis. The following
shows that this generalized form in (13) can be evaluated
to yield Bruns' exact formulas in [2].
Theorem 4: The Fourier coefficients a, and b, of the
Fourier series in (1) are computed by
a0 =- A( t ) dt ( 1 64
T o s'
IN/,]
a, = c p(l)B(2nf, 0) ( 16b)
/=1,3,5;..
and
I Nl nl
b, = p(1)(-1)('-1'/2B(2nl, 1/4nl) (16c)
I = 1.3.5: ' .
for (n =1, 2,
Proof: The zeroth order harmonic a. is obtained
from (2). Also, the even coefficients a, in (1) are easily
obtained from (7a) and Theorem 3 as follows:
W/nl
a, =c,(o). = C p((1)~(2nl , 01.
* 9 , N ) .
I = 1,3,5, ' . .
can see first from (4b) that b, =c,(l/4n). Thus
I = 1.3.5. ' ' '
. A - T + - T
( ki 4n
1 2"1 -
P"nl
= C p(1) - c (-1y
1=1,3,5:.. 2nl 1;=0
- A (in1 - T + I T ) . 4nl (17a)
But
2nI - 1 + ( I - I )/2
2nl - I
4nl
1
2nI - I + ( I - 1)/2
+ J =2n/ (-1)IA (". . -T) 2nl 4nl
(kl 4nl
2nI - 1
k =( I - 1)/2
= C (-l)kA - T + - T
Since A( t +T ) =A( t ) , then (17b) becomes
j = ( / - 1)/2
2,- I
The substitution of (17c) into (17a) yields
1 2 n / - 1
I N/nl
I = 1.3.5.. ' '
b, = C p( f ) ( - l ) o- 1) ' 2- 2nl k = o C (-l )k
* A - T + - T
( l l 4nl
To calculate the odd Fourier coefficients b, in (I ), one Hence the theorem is proved.
I I27
REED et al . : VLSI ARCHITECTURE
From (16b) and (16c) it is evident for zero-order interpolation that both the a, and b, can be obtained from single
To compute a, set (Y =0. By (12) B(2nl, 0) for n =1, 2, - , 5 are obtained using nearest neighbor interpolation
matrices. To illustrate this in detail, consider again the same example given in Section 11.
in matrix form as follows:
1 0 0 0 0 - 1
0 0 - 1 0 1
1 0 -1
1 -: -: 1 - 1 1 -
-1 1 -1 1 -1
0
0
0
0
-1
0
-1
-1
1
1
A(0.2)
A(0.3)
A(0.4)
A(0.5)
A(0.6)
A(0.7)
- 1
- 1
From the above and (16b) in Theorem 4, coefficients a,
for n =1, 2, * , 5 are obtained finally in matrix form
as From the above and (16c) the coefficients b, for n =1,
2, * , 5 are obtained finally in matrix form as
Note that the matrices in (1 8) and (19) have precisely the
Same computational complexity.
To compute the b, by the Bruns method, the Bruns al-
, 5 are ternating average B(2n, 1/ 4n) for n =1, 2,
computed from (15) in matrix form as follows:
*
A( 0. 2)
A( 0. 3)
A(0.4)
A(0.5)
A(0.6)
A(0.7)
0 0 0 1 0 0 0
0 1 0 0 - 1 0 1
0 1 0 - 1 1 0 - 1
(194
1 - 1 0 1 - 1 1 -1 0
1 - 1 1 -1 1 - 1 1 -1
I128 IEEE TRANSACTIONS ON SIGNAL PROCESSING, VOL. 40, NO. 5, MAY 1992
TABLE I
COMPARISON OF THE DIFFERENT METHODS FOR CALCULATI NG A N M-POINT
IV. COMPARISON OF DIFFERENT METHODS FOR
COMPUTING THE AFT AFT
Bruns Original Method AFT Method in 151
In order to calculate 6, the shift a needed in the alter-
nating average B(2n, a) of Bruns' method is equal to
1/4n. Note also that the row of the matrix used to Cal- Additions ( 1/ 2) M2 (3/ 8)M2
M (3/ 2)M
Larger
culate B(2n, 1 /4n) in (19a) is a cyclic shift of the corre-
sponding row for calculating B(2n, 0) in (18a). The shifted Er&lexity Low Higher
structure of these two matrices may be useful when the
entries of the matrix are shifted serially into a signal pro-
cessor.
Multiplications
Small
The Bruns alternating average B(2nZ, a) in (12) differs
from S(n, a) in (5) in that it uses only an even number of
samples. This fact overcomes the requirement in (5) for
using a zero-mean function.
From (5) and (12) the number of operations needed to
calculate B(2nl, a) is twice that needed for S(nZ, a). How-
ever, the calculation of a, and b, in (16b) and (16c) needs
only the terms B(2n1, a) for odd 1, only half of the terms
of type S(n1, a) used for the AFT algorithm in [5].
Hence the total number of additions needed for a direct
computation of the Bruns method is approximately the
same as for the previous method in [5]. This fact is sup-
ported by the results shown in Table I : In the Bruns
method the number of additions is greater than the method
in [5]. However, the number of multiplications by scale
factors are less in the Bruns technique. As a result, the
total number of operations required in Bruns method is
approximately the same as the method in [5]. Finally the
calculation of the zero-mean function A(t) in [5] is a time-
consuming process not needed in the Bruns AFT. As a
consequence Bruns method should have a better through-
put than the method in [5]. This is an important factor for
designing a real-time parallel signal processor.
An approximate error analysis of Bruns' method is
given in Appendix A. It appears to show a 1.25-dB re-
duction in noise level compared with the method given in
[5]. However, a computer simulation for calculating the
coefficients of a Gaussian periodic function with zero-or-
der interpolation shows that in actuality there is only about
a 0.92-dB gain in signal-to-noise ratio by Bruns' method
compared with the method in [5]. This is illustrated in
TABLE I1
COMPARISON OF THE RELATI VE ROOT-MEAN-SQUARE ERROR WITH
DIFFERENT AFT'S I N THE COMPUTATION OF A GAUSSIAN
WAVEFORM WI TH 1024 SAMPLES
AFT Method in [ 5] Bruns Method
1.41 X 1.27 x 10- ~
Definition 3: Let W(n, a) be the n-point average
(20)
1 n - l
W(n, a) =- c A( mT/ n +a T )
n m = O
of the n values A( mT/ n +aT) for (m =0, 1, 2, . * ,
n - 1) of A( t ) shifted by the amount aT where - 1 <a
<1.
Note by (3) that average W(n, a) in (20) is related to
average S(n, a) in (5) by the relation, W(n, a) =S( n, a)
+a. where a. is the zeroth harmonic of the periodic func-
tion A( t ) .
Theorem 4: The Bruns 2n-point alternating average
B(2n, a) decomposes into
B(2n, a) =- W(n, a) - w n, a +-
2 l L ( 231 (21)
and the 2n-point average W(2n, a) decomposes into
W(2n, a) =- W(n, a) +w n a +-
2 l [ ( ' 231 (22)
Table I1 where the error is the root-mean-square error be-
form using an inverse FFT of the coefficients.
tween the original waveform and the reconstructed wave-
where w(n, a) is the n-point average Of A( t ) given in Def-
inition 3.
Proof: First, by the definition of B(2n, a) in (12), it
V. A VLSI ARCHITECTURE FOR ARITHMETIC FOURIER
is not difficult to show that
TRANSFORM 1 2 n - l
B(2n, a) =- C (-l)*A
In this section one shows that Bruns alternating average
2n m = O -
B(2n, a) can be decomposed into two n-point averages.
Then a systolic array structure is developed which uses
n-point averages to obtain 2n-point averages and so forth.
Both of the Bruns alternating averages B(2n, 0) and B(2n,
1/4n) for calculating the even and odd components of a
signal can be generated by the same such architecture. As
a result, a VLSI architecture is suggested here to reduce
the number of computations of the arithmetic Fourier
transform.
n - I
=I [ ' - 2 nm=o c (-1)2mA
+- 1 n - l C (-1)2m+lA f* T +a T ) ]
n m=O
REED er a/ . : VLSI ARCHITECTURE
2n
1 n - l
- - c A - T + a T + - T
nm=o
Also, by the definition in (20),
+- CA - T + a T + - T
n m=O 2n
and the theorem is proved.
From (21) and (22) it is evident that both the B(2k, a)
and W(2n, a) can be obtained from the n-point averages
W(n, a) and W(n, a +1 /2n). The basic operations given
in (21) and (22) constitute what might be called the AFT
butterfly as shown in Fig. 1. A systolic array based on the
AFT butterfly is constructed such that the averages W(2n,
a) are propagated by the corresponding previous averages
W( n, a). The Bruns alternating averages B(2n, a) needed
for calculating the Fourier coefficients are obtained at the
same butterfly level but with possibly different signs. The
structure of this method is shown in Fig. 2. In Fig. 2,
both B(2n, 0) and B(2n, 1/4n) are obtained in the same
structure. Thus a substantial reduction of the number of
additions is accomplished by this method.
To illustrate this in detail, again consider the example
developed in Section 111. To compute B(2, 0), B(4, 0),
andB(8,O)in(17a)andB(2, 1/4)andB(4, 1/8)in(18a),
one needs to calculate W(2, 0), W(2, 1/4), W(2, l /8),
and W(2, 3/8) first. As it is shown in Fig. 3,
W(2, 0) =k[W(l, 0) +W(1, 1/2)]
=i[A(O) +A(0.5)]
/4) =i W(1, 1/41 +W(1, 3/4)1
=i[A(0.3) +A(0.8)]
/8) =i[W(1, 1/8) +W(1, 5/81]
=;[A(O.l) +A(0.7)]
W(2, 3/ 8) =;[W(l, 3/8) +W(1, 7/8)]
=;[A(0.4) +A(0.9)].
As a consequence one has
W(4, 0) =i [ W( 2, 0) +W(2, 1/4)]
1 I29
B(2j ,a ) =1/2 [ W(j , a ) - W(j ,a +1/2j ) 1
W(2j , a ) =112 [ W(j . a ) +WO, a +1/2j) I
Fig. 1. Butterfly structure of the AFT.
W( k . l l 4 k ) B ( 2k , 1/ 4k ) '>(
Fig. 2. General tree of butterfly structure for calculating Bruns' altei
averages.
mating
and
W(4, 1/8) =i [ W( 2, l /8) +W(2, 3/8)].
Finally the required alternating averages are obtained
as outputs from the butterfly tree as follows:
B(2, 0) =k[W(l, 0) - W(1, 1/2)]
=i[A(O) +A(0.5)]
B(4, 0) =i[W(2, 0) - W(2, 1/4)]
B( 8, 0) =5[W(4, 0) - W(4, l/8)]
B(2, 1/4) =i[W(l, 1/4) - W(1, 3/4)]
=;[A(0.3) - A(0.8)]
and
B(4, 1/8) =i[W(2, 1/8) - W(2, 3/8)].
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IEEE TRANSACTIONS ON SIGNAL PROCESSING, VOL. 40, NO. 5, MAY 1992
W (2, 3/8) W( 4 , 1 / 8 )
Fig. 3. Example of butterfly tree for calculating Bruns' alternating averages B(2, 0), B(4, 0) , B(8, 0) , B(2, 1/4), and B(4,
1/8).
4(2 A
4(3 A
I
I
I
I
I
I
I I
I I
I I
I
I I
I
I
B(6.0), B(6.1112)
8( 12, 0) , 8( 12. 1/ 24)
I
1I
I
~ ( K A t)
Fig. 4. VLSI architecture for the arithmetic Fourier transform.
Note that the averages B( 2, a), B(4, a) and B( 8, a) are
calculated in the same group. Also, the computations of
W( 2, a) and "(4, a) are inherent in those computations
needed for B(8, a). The division-by-two operations in (21)
or (22) are achieved by one bit shifts when the data is
represented in binary form. The number of additions
needed to calculate the alternating averages is O( N2) (see
Table I and [ 5] ) . As a result, an approximate 25% reduc-
tion in the number of additions is accomplished by this
new method when the number of coefficients is large.
A VLSI architecture for the arithmetic Fourier trans-
form, using zero-order interpolation, is shown in Fig. 4.
1131
REED er a/. : VLSI ARCHITECTURE
If the real function A( t ) is sampled at points t =mT/ k
where 0 I m <k and T is the period, then the zeroth
order harmonic is the sample average. That is, aO =(1 / k )
=OA( mT/ k) . Also, the different required terms of form
B( 2n, a) can be computed by a use of the method shown
in Fig. 2 and zero-order interpolation. Finally, the even
and odd harmonics, i.e., a, and b,,, are computed by
means of ( 16b) and ( 16c).
z k - I
VI. CONCLUSION
A computationally balanced AFT algorithm for Fourier
analysis and signal processing is developed in this paper.
This algorithm does not require complex multiplications,
which is one of the obstacles needed for developing faster
Fourier transform algorithms. Finally, this new, efficient
AFT algorithm is shown to be identical to Bruns' original
AFT algorithm.
The simple weight values {- 1 , 0 , l } required by the
Mobius function reduce substantially the number of mul-
tiplications required by a conventional forward FFT. The
error analysis in Appendix A shows a better upper bound
for this algorithm than the result previously obtained in
[ 5 ] . Also, this generalized algorithm is suited ideally to
parallel processing and data-flow methods. Finally, a
newly found VLSI architecture is shown to reduce the
number of additions. As a consequence, this simplified
AFT algorithm is well suited to a VLSI implementation.
APPENDIX A
Fourier analysis of a random signal is used to develop
on error analysis of Bruns' original method. This error
analysis is derived by the method used in [ 5 ] . First con-
sider the error in computing the Fourier series of A( t ) due
to zero-order interpolation. The mean-square error over a
period is given by
E[ e21 =E /f 1' [ A@) - AO( t ) l 2 dr]
N
1
=- E
C (U, -
+ C (b, - b!o')2
2 n = ~ n = 1
('41)
where U(:' and b!" are obtained by using zero-order inter-
polation.
/ N
One has
F,(CX) =b, - bLo'
[Nl nl 21n- I
21n [ A (2 +a) - A0 (2 +a)]
where AO ( ( m/ 21n) +a) is the nearest neighbor sample to
A( ( m/ 2l n) +a).
Assume the sampling errors c, ( m, 1, a) act like inde-
pendent, identically distributed, zero-mean, random vari-
ables for different values of m, I, n, and a. That is, the
sampling errors behave like
E[ cn( m, 1, a) en, ( m I , 1 ', a11
0 i f m #m' or
1 #I ' orn #n' or a #a'
a: if m =m' and
I =1' and n =n' and a =a'.
(A31
=r
Then one obtains the following bound on the error dis-
persion of the Fourier coefficients as follows:
[N/nl 2In- 1
a:" = c C - ' p ( 1 ) ' 2 E[ c, ( m, I , a)I2
I = 1.3.5:. . m=O (21n)'
a 2 W/nI 1
<' C -
a2
2n I = 1. 3. 5; ' 1
(A4) I L [l nN - I nn +I n2 +y ]
2n
where y =0.534 is Euler's constant.
The variance a:,, of error F,(a) =c,(a) - ~ ~ ~ ' ( a ) is
functionally independent of the value a, hence one finds
the E[ e 2 ] is given by
N
~ [ e ~ ] = C a:,,. ('45)
n = I
By this result, one gets the error bound
(In N +y)(ln N +y +In 2 - 1) +
(A61
In order to find a: one defines cJ =A( t ) - A( t J) , where
tJ - ( A / 2 ) <t <tJ + +(A/ 2) and tJ is the sampling
time and A =1 / MO is the sampling interval. Then by the
first mean value theorem of calculus, one has
a: =E[A(t ) - A(t,)I2 =E[A(t, ) +&t j ) ( t - t J)
where t J E ( t ] , t ) f ort >tJ or 4; E ( r , t J ) f ort <tJ.
Let A =1 / MO. Then if A( t ) is a section of a Gaussian
process, the random variable &E,), 4, E ( t ] , t ) , is indepen-
dent of the random variable, r =t - tJ where - A / 2 <
r I A / 2 . Next if r is subjected to the uniform distribu-
tion, then
- A(5)12 =E [ A 2 ( t j > ( f - t J ) 2 1 (A71
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IEEE TRANSACTIONS ON SIGNAL PROCESSING, VOL. 40, NO. 5, MAY 1992
is the dispersion 7 . Also it is proved in [8] that
E[;1(4j)]2 =-&4(0) (A91
where RAA( 7) is the autocorrelation function of A( t ) . Thus
a substitution of (A8) and (A9) into (A7) yields
A2 ..
. f =--
12 RAA(0).
Now consider a random signal with a Gaussian power
spectrum density,
where W is the m.s. bandwidth. Then, one obtains
RM( 0 ) =-PA(2nW)2. (A 12)
Thus for zero-order interpolation this yields finally the
relative error bound
+In 2 - 1) +- ( ~ T W ) ~ (A13)
for Fourier analysis. For a Gaussian waveform with N =
512, A =1/1024, T =1 and interval, the bandwidth is
approximately W 2: 1.0. Hence
1 6
By an argument similar to that used above for zero-
order interpolation, one can obtain an error bound for lin-
ear interpolation which is given by
+In 2 - 1) +- R$(O)
(A15)
1 6
where R$ is the fourth-order derivative of RAA.
From (A14), the upper error bound appears to be better
than that found in [ 5 ] . This result shows the signal-to-
noise ratio in Bruns method is better by a factor of 4 over
the previous method in [ 5 ] .
REFERENCES
J . W. Cooky and J . W. Tukey, An algorithmfor the machine cal-
culation of complex Fourier series, Math. Computation, vol. 19,
H. Bmns, Grundlinien des Wissenschajilichnen Rechnens. Leipzig,
1903.
A. Wintner, An Arithmerical Approach to Ordinary Fourier Series,
Baltimore, MD, privately published monograph, 1947.
D. W. Tufts and G. Sadasiv, The arithmetic Fourier transform,
IEEEASSP Mag. , pp. 13-17, J an. 1988.
I. S . Reed, D. W. Tufts, Xiaoli Yu, T. K. Truong, M. T. Shih, and
X. Yin, Fourier analysis and signal processing by use of the Mobius
inversion formula, IEEE Trans. Acoust. , Speech, Signal Process-
ing, vol. 38, no. 3, Mar. 1990.
pp. 297-301, 1965.
[6] E. 0. Brigram, The Fast Fourier Transform. Englewood Cliffs, NJ :
Prentice-Hall, 1974.
[7] L. K. Hua, Introduction to Number Theory. Berlin, Heidelberg, New
York: Springer, 1982.
[8] A. Papoulis, Probability, Random Variable, and Stochastic Pro-
cesses. New York: McGraw-Hill, 1984.
[9] N. Tepedelenliglu, A note on the computational complexity of the
arithmetic Fourier transform, IEEE Trans. Acoust. , Speech, Signal
Processing, vol. 37, no. 7, pp. 1146-1 147, J uly 1989.
[l o] D. W. Tufts, Comments on A note on the computational complex-
ity of the arithmetic Fourier transform, IEEE Trans. Acoust.,
Speech, Signal Processing, vol. 37, no. 7, pp. 1147-1 148, J uly 1989.
Irving S. Reed (SM69-F73) was born in Seat-
tle, WA, on November 12, 1923. He received the
B.S. and Ph.D. degrees in mathematics fromthe
California Institute of Technology, Pasadena, in
1944 and 1949, respectively.
From1951 to 1960 he was associated with Lin-
coln Laboratory, Massachusetts Institute of Tech-
nology, Lexington. From 1960 to 1968 he was a
Senior Staff Member with the Rand Corporation,
Santa Monica, CA. Since 1963 he has been a Pro-
fessor of Electrical Engineering and Computer
Science at the University of Southern California, Los Angeles. He holds
the Charles Lee Power Professorship in Computer Engineering at USC. He
is also a Consultant to the Rand Corporation, the MITRE Corporation, and
a Director of Adaptive Sensors, Inc. His interests include mathematics,
VLSI computer design, coding theory, stochastic process, and information
theory.
Dr. Reed is a member of the National Academy of Engineering and re-
ceived the 1989 IEEE Richard W. Hamming Medal.
Ming-Tang Shih (M90) was born in Kaohsiung,
Taiwan, on December 19, 1953. He received the
B.S. and M.S. degrees in electrical engineering
from the National Cheng-Kung University,
Tainan, Taiwan, in 1976 and 1978, respectively,
and the Ph.D. degree from the University of
Southern California, Los Angeles, in 1990.
From1980 to 1990 he was with the Electronic
Research and Service Organization, Hsin-Chu,
Taiwan. He served as the Design Engineer and the
Section Manager involved in data conversion and
telecommunication integrated circuit design. Since 1990 he has been with
the Computer and Communication Research Laboratories, Hsin-Chu, Tai-
wan. He is currently a project leader of ISDN design. His research interests
include communication system, signal processing, coding theory, and VLSI
design.
T. K. Truong (M82-SM83) was born in Cho-
lon, Vietnam, on December 4, 1944 He received
the B S degree in electrical engineering fromthe
National Cheng Kung University, Taiwan, China,
in 1966, the M.S. degree in electrical engineering
fromWashington University, St. Louis, MO, in
1971, and the Ph.D. degree fromthe University
of Southern California, Los Angeles, in 1976
Since 1976 he has been with the Communica-
tion SystemResearch Section, SystemEngineer-
ing Technical Staff of the J et Propulsion Labora-
tory, California Institute of Technology, Pasadena Also he is currently an
Adjunct Associate Professor with the Department of Electrical Engineer-
ing, University of Southern California, Los Angeles, and a Consultant to
the Department of Radiology, Memorial Hospital, Long Beach, CA His
research interests are in the areas of mathematics, VLSI architecture, cod-
ing theory, X-ray reconstruction, and digital signal processing
REED er al . : VLSI ARCHI TECTURE
I I33
E. Hendon was born in Birmingham, AL, on
March 1, 1958. She received the B.A. degree in
physics and the M.S. degree in electrical engi-
neering from Vanderbilt University, Nashville,
TN, in 1980 and 1982, respectively, and the Ph.D.
degree fromthe University of Southern Califor-
nia, Los Angeles, in 1990.
She has been working with TRW, Inc., in Re-
dondo Beach, CA, since 1982. Presently, she is
with Southern Research Technology, I nc., Bir-
mingham, AL. Her areas of interest are radar, de-
tection theory, and signal processing.
D. W. Tufts (SSS-M6l-SM78-F82) received
the B.A. degree in mathematics from Williams
College, Williamstown, MA, in 1955, and the
S.B., S.M., and Sc.D. degrees in electrical en-
gineering from the Massachusetts Institute of
Technology, Cambridge, in 1957, 1958, and
1960, respectively.
From 1960 to 1967 he was at Harvard Univer-
sity, Cambridge, MA, first as a Research Fellow
and Lecturer and later as an Assistant Professor of
ADDlied Mathematics. Since 1967 he has been a
..
Professor of Electrical Engineering at the University of Rhode Island,
Kingston, RI. He has been a consultant to Bell Telephone Laboratories,
Sanders Associates, Inc., and other companies.