113

IEEE TRANSACTIONS ON INSTRUMENTATION AND MEASUREMENT, VOL. IM-28, NO. 2, JUNE 1979

High-Accuracy

Analog Measurements
via Interpolated FFT

VIJAY K. JAIN, SENIOR MEMBER, IEEE, WILLIAM L. COLLINS, JR., MEMBER, IEEE,
AND

DAVID C. DAVIS,

MEMBER, IEEE

Abstract-By use of an interpolated fast-Fourier-transform

In this paper we propose certain FFT-based algorithms to
(FFT) algorithms are developed for multiparameter measurements achieve high accuracy measurement of the following
upon periodic signals. Eight pertinent measurements, such as fundamental frequency, phase, and amplitude, are made with enhanced parameters:
accuracy compared to existing algorithms, including taperedaverage voltage over the
a) Va,
window-FFT algorithms. For the more general case of nonharmonic
window
multitone signals also the method is shown to yield exact amplitudes
fundamental frequency
b) fi
and phases if the tone frequencies are known beforehand. These
measurements are useful in a variety of applications ranging from
amplitude of the fundamental
c) Al
analog testing of printed-circuit boards to measurement of Doppler
frequency
signals in radar detection.
phase of the fundamental
d) &l
I. INTRODUCTION

CONSIDERABLE interest is in evidence in the measurement of analog parameters from the sampled values of a
signal [1], [2]. It arises from a desire for greater accuracy,
repeatability, and simultaneous determination of several
parameters. Rapid advances in hardware technology, in
particular large-scale integration (LSI) implementation of
digital algorithms [3]-[5], have indeed spurred vigorous
activity and, as a result, we are perhaps on the verge of a new
era in instrumentation concepts, designs and capabilities. In
the past, the algorithms used for amplitude, phase and
frequency information have been primarily of the timedomain type. For instance, the amplitude was determined
via peak-to-peak measurement, the frequency via zero crossings, and the dc value through window averaging. Such
formulas possess the advantage of conceptual and practical
simplicity. And they are quite effective when moderate
accuracy say 1 percent or worse, can serve the purpose. Only
recently, the use of discrete-Fourier transform (DFT) or its
algorithmic version fast-Fourier transform (FFT),1 has been
proposed by some researchers with a view toward higher
resolution and accuracy. However, these methods also
produce somewhat erroneous results [1] due to the wellknown spill (or leakage) effect, despite the use of taperedwindow multipliers [6], [18]. Therefore, when a high degree
of accuracy is desired or if the signal quality is poor
(manifested by a low SNR or the presence of a nonperiodic
component), the usual time-domain methods as well as the
tapered window DFT methods prove to be unsatisfactory.
Manuscript received July 29, 1977; revised November 27, 1978 and
February 26, 1979.
V. K. Jain is with the Department of Electrical Engineering University
of South Florida, Tampa, FL 33620.

W. L. Collins, Jr., and D. C. Davis are with the Honeywell Information

Systems, Inc., Tampa, FL 33684.
1 The terms "discrete Fourier transform" and "fast Fourier transform"
will be used interchangeably in this paper.

e) AM,

OtM,

frequency

M = 2, ... amplitudes and phases of the

harmonics
true dc voltage level of the signal
root-mean-square value of the
signal
total harmonic distortion.

f) Vdc

g) Vrms

h) THD

The proposed algorithms require simple calculations

beyond the DFT of the signal. Software implementation is
therefore straightforward. However, recognizing the distinct
possibility of complete implementation in hardware, the
paper also considers the effect of arithmetic roundoff for
small word-lengths [7], [8]. For the sake of brevity, the effect
of roundoff is presented only for parameters a), b), c), and d).
The more general case of a multitone signal, with nonharmonically related tones, is also considered. It is shown by use
of interpolated FFT that the amplitudes and phases of the
constituent sinusoids can be determined with high accuracy.
Exactly, if the arithmetic roundoff errors can be ignored. We
shall begin with this general case in Section II. The remainder of the paper is devoted to periodic signals, where, as
stated above, all of the eight pertinent parameters are
determined, including the fundamental frequency. A brief
discussion of the DFT and the effect of finite word-length
upon it is given in Appendix A.
II. AMPLITUDES AND PHASES OF

MULTITONE SIGNALS

Consider the sampled multifrequency signal

M

x(kA)= E x.(kA)
m=1
M

Z Amsin (2ltfmkA+4Jm),

m =1

0018-9456/79/0600-0113$00.75 ( 1979 IEEE

k0O,

1,

, N-1

(1)

14

IEEE TRANSACTIONS ON INSTRUMENTATION AND MEASUREMENT, VOL.

IM-28, NO. 2, JUNE 1979

where it is assumed that the sampling frequency is greater so that the real and imaginary parts of the constituent
than two times the highest constituent frequency. Suppose sinusoids of x(t) become known. From these the amplitudes
that the frequency resolution (fo = 1/NA) is fine enough and phases can be determined in the usual way:
that the frequencies of the signal lie in separate bins. Thus
Am= X2 +Y2
(hla)
Lmfo <fm = (L + m)f0 Am fo < (Lm + )
(2)
(I lb)
Om= arctan (Ym/Xm).
where 4, are distinct integers and bm are suitable fractions
between zero and one, including zero. By use of the defining
Examples of application to multifrequency signals are
relation (equation (Al)Appendix) the DFT [9] of (1) can given in [10].
be shown to be
III. MEASUREMENT ALGORITHMS FOR
m
- i)
7
r(Am
PERIODIC SIGNALS
=
S(i) -0.5j E 1exp [j(a(m i) + 4m)] sin ir(A,m - i)/N
m=1
Let x(kA), k = 0, .., N - 1 be the samples of a periodic
signal
x(t). Again it is assumed that the sampling frequency
m
sin
+ i)/N]| Am
-exp [-j(a(Q,Z + i) + mr)]
exceeds the Nyquist rate so that aliasing of spectra does not
occur. Since the sampling interval A is considered fixed, we
(3) will simplify the notation and denote the sampled signal
where a = ir(N - 1)/N. To manipulate (3) into a more useful values as x(k) or Xk. Each of the eight parameters
enumerated in Section I is considered here. Derivations of
form let us define the interpolation functions
the algorithms are sketched only briefly for reasons of space;
Ssos (z)= sin (az) sin z/
(4) details may be found in [10].
sin 7rz/N
A. Average over the Observation Window (V,v)
Csos (z) cos (az) sin z/
(5) Letting i = 0 in (Al) one obtains the expected result

~~~~~~~sin

Also let

N-1

(12)
k=O
(6)
S(i) = u+ jvj
Am exp (fjtm) = Xm + jYm.
(7) Note that Va, = S(O)/N is the average over the observation
window; the dc level is determined later in (29).
Then, equating the real and imaginary parts of (4) one
obtains
B. Fundamental Frequency (f1)
M
Suppose the fundamental component is described by
E {[Csos (Am - i) - Csos (Am + i)]Xm
x1(t) = A1 sin (27rf1t + 01)
(13)
- [Ssos (Am - i) - Ssos (Am + i)]Ym} = -2Vj (8a)
wheref1 = 4fo; i.e., the fundamental frequency is A times the
M
frequency resolution achieved from the T = NA second
E {[Ssos (Am i) + Ssos (Am + i)]Xm
observation window. Note that the real number A also
the number of signal cycles contained in the obserdenotes
+ [Csos (Am - i) + Csos (Am + i)]Ym} = 2Ui. (8b)
vation window. For satisfactory accuracies to be achieved, it
Assuming that the frequencies of the constituen^t signals are is recommended that at least twenty cycles of the signal be
known, a set of distinct indices i is chosen. For the frequency obtained so that typically A > 20. Let A = 1 + ( where
fm choose i to be equal to Lm or Lm + 1; if distinctness is not 0 < ( < 1 and I is an integer.
violated choose i = Lm whenever bm < 1. Call this set of
By use of the defining relation (Al), or by setting M = 1 in
indices as I. Note that this set contains M indices.
(3), the DFT of (13) is found to be
Then (8), with i ranging over I, can be written in a matrix
form as

S(0) = Z x(k)-NVav.

Fz= w

(9) S(i) =

where

-0.5jA,1

exp [j(a(Q

- i)

0)] sin r( - i)/N

sin

z=[Xl Y, , XM,YM]
w = [-2V1, 2U1, , -2VM, 2UM]T
and the 2M x 2M matrix F is defined in the obvious way.
Assuming that the inverse of this matrix exists one obtains
the solution
z=

F-lw

irQA + i)

exp [-j(a( + i) + 0j]sn(+)

sin 7rQA +

i)/N]_14(14)

where a = r(N - 1)/N. The largest two spectrum lines are,

as one would expect, I S(l) and I S(l + 1)1 where the
interval [1, 1 + 1) encloses A. To proceed further, we will
make an approximation. We shall ignore the second term in
(14) when evaluating it at i = l and at i = I + 1. WhenXA > 20
and N = 1024 or higher, this approximation can be shown

115

JAIN et al.: HIGH-ACCURACY ANALOG MEASUREMENTS VIA INTERPOLATED FFT

to contribute no more than 0.04 percent error in frequency. D. Phase of Fundamental

Then the largest two spectral components are given by
From (15) we obtain

-jO.5A1 exp [j(ab + kA)]

S(l) =

sin

nr/N

S(l + 1)

(l -)
- -jO.5A1 exp [j(a(b - 1) + &)] s -in
(15b)
From (14) and (15) we observe that a single sinusoid results
in a single line (plus negative image) spectra only when i is an
integer, i.e., when the signalfrequency is an integer multiple of
1/NA.2 In general, however, the spectrum possesses a
sin v/sin (v/N) profile. This phenomenon is commonly referred
as spillover or leakage effect [6]. To counter leakage we will
use interpolation between the Ith and (I + 1)th DFT components. First a further approxinmation is made. For N equal
to 1024 or greater, the sine terms in the denominators of
(iSa) and (15b) may be replaced by their respective arguments incurring an error of no more than 0.015 percent.
Thus

I S(l)

05A I

sin n6i

(20a)
(20b)

01 = Phase {S(l)} - ab + rc/2

(15a)

01 = Phase {S(l + 1)}-a(b- 1) + r/2.

Recall that a = ir(N - 1)/N. Clearly, either of the above two
equations can be used to compute 1. We recommend using
the one corresponding to the largest spectral line.
E. Harmonic Amplitudes and Phases (AM, I'M)
Consider the Mth harmonic with frequency
fM - Mf1

(21)
where AM = MA and m . AM < m + 1. That is the frequency
fm is contained in the interval [mfo, (m + 1)fO) where
fo = 1/T was defined to be the frequency resolution of the
FFT. Then proceeding analogous to subsections C and D
=

AM fo

above we can show that

21bM
N

(16a)

I S(m)

(22a)

Isinl nMI

IS(m + 1)
(22b)
I sin r(1 - bM) I
IS(l 1) 0(1 - )/N 5A I(1 -3)/N
(23a)
O'M = Phase {S(m)} - abM + i/2
(16b)
OmM= Phase {S(m + 1)} - a(bM- 1) + /2. (23b)
Denoting the ratio of the two magnitudes as Lx, i.e.,
The amplitude AM and the phase O'M can be computed using
(22) and (23). We recommend using (22b) and (23b) only
_ IS(I+ 1)
when IS(m + 1)1 is greater than IS(m)I.
(17a)
i S(1) t
It should be noted that the highest harmonic is given by
it follows immediately from (16) that
(Note <y> = integer part of y) (24)
Mmax= (
27r(1 - M)

AM=-

(17b)

3=

Thus we obtain the fundamental frequency

as

=
f,=
t1)fo
fo ((11:I
1 + L)) NA'

(18)

C. Amplitude of Fundamental
From (16) we obtain3
Al

A=

2rbN IsinI S(l)nbI

27r(1N ) IS(l + 1)I
Isin r(1 - 3)1
-

whereft is the fundamental frequency.

F. True dc Level (Vdc)
In general, the true dc level of the analog signal x(t) will be
different from the average of x(k) over the N-point window
k =O, ", N -i1. To demonstrate this, consider for the
moment that the signal x(t) is a biased monotone, i.e.,

(25)
x(t) = Vdc+ A, sin (2nf1t + I').
(19a) Assume that
f1, A1, and 01 have been determined as
discussed in Sections III-B-III-D. Now
(19b)
(26)
x(k) = Vdc + A1 sin (2rf1kA + I1)

Either of the above equations can be used to compute A 1. so that

We recommend using (19b) only when S(l + 1) 1 is larger
A N-1
than IS(l)L.
Vav = Vdc + N E Imag. exp
k=O

Note that

I When 6

-O

as

0,

-*

0, the limit of sin

7r3/sin nrb approaches

7rb/sin (7rb/N)
1.

approaches

N.

A 1 sin 7rf1NA s

+
Vdc
sin
=VcN

7rf1 A

[j(2irf1kA + '1)]

(7rmf(N

1)A +

I1).

(27)

116

IEEE TRANSACTIONS ON INSTRUMENTATION AND MEASUREMENT, VOL. IM-28, NO. 2, JUNE 1979

....

S1 (l

TABLE I
FFT OF A SIGNAL WITH dc, FUNDAMENTAL AND
THIRD HARMONIC COMPONENTS (SECTION V)

)+S3 ( i)

s1 ( i )
--- -

|S(i)
iI

Frequency
Index

492.648
83.377

-0.0500

19
20
21

960.053
5749.332
1435.643

-0.8048
-0.8363
2.2747

59

197.037
518.243

0.4854
0.3775

775.002

-2.8689
-2.9688

s 3 (i )

60
61
62

Fig. 1. Effect of harmonic interference.

Phase S(i)

0.0

222.902

Thus we find that the true dc level is related to the window

average V5,, by
Vdc=VSVNisin
sin
Vdc-V"N sin
nION sin"

(aA~+ 0ti)

(28)

where, we recall, a = r(N - 1)/N, A = f1!]fo. More generally, when the signal x(t) contains harmonics in addition
to the fundamental,
1 Mmax

Vdc= Vav N

AM

sin 7rA

(a. M + QM)

sin , N sin

remedy is to modify S(m) and S(m + 1) (used in Section

III-E) as follows:
SM(m) = S(m) - S(m)
= S(m) + 0.5jA1
exp

(29)

Vrms

[Vd, + '(A' + ... + AMx)]112.

H. Total Harmonic Distortion (THD)

The THD is computed as follows:

sin

(32a)
SM(m+ 1)=S(m+

G. Root-Mean-Square Value (Vrms)

The rms value of the periodic signal is given by

exp

(30)

ir(1A -

m)
[j(a(. - m) + 01)] sin r(A - m)/N

1)+0.5jA,

U((A

m-

1)

exP[j(a(.. -m-1)

sin

+ 1)] sin

ir(A - m - 1

r(A

1)/N
(32b)

where S 1(m) denotes the spillover of the fundamental at

frequency mf0. Now the formulas (22) and (23) may be used
to compute Ak, Ok, k = 2, . . A more rigorous method is to
solve for A1, 1, A2, 42, etc., simultaneously as shown in
Mmax
1/2
//V
1Mmax
THD=
THD-~2- E2Am / (~ 2+iM- 4).
Section II. This, however, involves inversion of a 2Mmax
(
(31)
Am2
dimensional square matrix so that the gain in accuracy
could well be offset through computational roundoff. Unless
IV. HARMONIC INTERFERENCE
a large word-length, say 32 bits, were available, the simpler
In Sections III-B-III-E, it was tacitly assumed that the approach (corrections via (32) and use of (22) and (23)) is
effect of harmonic interference in the FFT components is recommended.
negligible. Specifically, it was assumed that the largest two
V. AN EXAMPLE
spectral lines S(l) and S(l + 1) were the contributions of only
the fundamental frequency. In truth, S(l) and S(l + 1) conConsider that
tain minor spillovers from the second, third, and possibly x(t) = 0.2 + 6.0 sin
[27r(20.2)fot + 0.1] + sin [27r(60.6)fot]
higher harmonics (see Fig. 1). It is shown in [10] that even
with a 33 percent third harmonic, the error in the fundamen- wherefo = 1/2.048 Hz; A = 0.001 s and N = 2048. The FFT
tal frequency caused by ignoring the spill at 1 and 1 + 1 is no of x(kA) is given in Table I for some of the frequencies where
greater than 0.025 percent. Similarly A 1 and 4 , are relatively the spectrum amplitude is significant.
insensitive to the harmonic spillover. However, the compua) Vav = S(O)/N = 0.2406.
tation of the harmonic amplitudes and phases is not robust
b) The largest spectrum lines are found to occur at 1 = 20
relative to the spillover from the fundamental. A simple and 1 + 1 = 21.

:1Ac

JAIN et al.: HIGH-ACCURACY ANALOG MEASUREMENTS VIA INTERPOLATED FFT

TABLE II
COMPARISON OF PARAMETERS OBTAINED FROM RAW FFT,
TAPERED-WINDOW FFT, AND INTERPOLATED-FFT

From tapered-window FFT

BlackmanBlackmanHarris 74dB
Harris 92db

From
raw FFT

Hamming

20.0

20.0

20.0

20.0

From
Interpolated
FFT

20.1998

A1

5.6140

5.8124

5.8857

5.9097

6.001

0.734

0.7292

0.7283

0.7283

0.1071

A3

0.7568

0.8795

0.9257

0.9411

0.9977

13

-1.2981

-1.2619

-1.2568

-1.2566

-0.0029

[18] for

See reference

window

weights

Formulas (17) and (18) give

a = 0.2497

A = 20.199811.

3 = 0.199811

c) From (19a),
A1 = 6.001.

d) From (20),
= A TAN2(-4310.5156, 3810.2954)

-_r (2047)
(0.19926) + 2
2048
- 0.1071 radian.

e) Since the spectrum shows significant values only

around the third harmonic, we compute only A3, 03. First
note that
A3 = 3(i) = 60.59943.
Since the spectral line S(61) is larger than S(60) 1, we
shall use it to compute A3 and 43. From (32b) we obtain

S3(61) = S(61) = S1(61)

= (-746.3538- j208.7312)

117

[17], and then utilize the sharpened spectral lines. A comparison of the frequency and amplitude estimates is given in
Table II. Although the use of tapered-windows (Hamming
and Blackman-Harris windows) yields an improvement
over the raw FFT results, the scalloping loss [18] is
significant enough to cause large errors. The present method
clearly yields the best results. The reader interested in
non-FFT based techniques, such as maximum entropy
method, demodulation algorithms, and the pencil-offunctions method is referred to references [2] and [16].
VI. ARITHMETIC ROUNDOFF ERRORS

As stated in Section II, the effect of arithmetic roundoff

errors becomes significant when the algorithms of Section
III, or any alternative ones, are implemented with a small
word length (with a mantissa of under 16 bits). Since high
accuracy in measurement is of concern, the effect of arithmetic roundoff errors on computed parameters must be
evaluated carefully. Indeed, if the accuracies desired are
prespecified, these considerations may well determine the
minimum word length and window size for the particular
application. It should be noted that the effect of arithmetic
roundoff depends, to some extent, upon the nature of the
signal under consideration. For the analysis presented here,
we will pick either the worst case signal (one that produces
largest arithmetic roundoff) or a signal that is deemed to be
of greater interest than others. Also, for reasons of space,
only the first four parameters will be considered. The
discussion of each is rather brief; however, the details can be
found in [10].
A. Average Over the Window (VYa)
If an exact representation of the number 1.0 is used in the
arithmetic [8] then, as stated in Appendix, the noise-tosignal power ratio (NSR) given by (A5) is invalid for the zero
frequency. For a worst case signal, consisting of a predominant dc component plus a small ripple, it is shown in [10] that
the NSR is given by

NSR VarS(0)
S(0) 12 - 22

- (- 12.0314 + j31.3065)
= 772.5590 /-2.8256 radian.

Then from (22b) and (23b),

A3 = 0.9977
43 = -0.00294 radian.
The remaining three parameters f), g), h) are calculated
routinely via (24), (29), (30), and (31).
Remarks: Let us compare this method with some alternative procedures. We can employ the spectral lines directly,
i.e., use the local maxima at i = 20 and 61. Or, we could
premultiply the data with a tapered window, such as the
Hamming window Xk = WkXk,
Wk

0.54 + 0.46

cos

ir(k

N/2)/(N/2)

(33)
(33)

For example, if b = 11 bits, the fractional error from (33)

turns out to be 0.04 percent. Let us compare this error with
that obtained for a conventional consecutive summation
and division by N. To examine arithmetic roundoff in
consecutive addition note that4

(xo + X1)
((Xo

Xl)

+ X2)

(Xo + xl)(1

(XO

X2)(1

SUM =

N-1

+ X1 +

E X=
k
k=0

N-1
k0=

1)
+

Xk +

2) + (Xo
N-1k

k=i1

i=O

X1)1

xi/k

4 Throughout, a prime on a variable or expression denotes its roundedoff value.

118

IEEE TRANSACTIONS ON INSTRUMENTATION AND MEASUREMENT, VOL.

IM-28, NO. 2, JUNE 1979

where gk are independent E-distributed random variables. C. Amplitude of Fundamental (A1)

Since division by N = 2n changes only the characteristic of
For this analysis also, we assume that the signal consists of
the number, there is no additional roundoff involved, so that a single sinusoid. Now, from (19a)

[14]

A'1 =

Var {Vd}

N2 Z

(kVdc)%SE

27r

IS'(l) I

A1(1 + '1 + 12)

where 4 has a variance of 2.5nU2 and 42 has a variance no

greater than nU2. We have made use of the fact that
(sin 7rb'/b') - (sin rb/b) <. 0.27r2(6'- 6) and that 6 does
not exceed 0.5 (for otherwise (19b) will be used). Then the
noise-to-signal ratio is given by [15].
(34)
Var A'2
3U.(4
= 22.9n 2.
(38)
NSR=
NSR- 2
2
Al
For b = 11 and N = 1024, the fractional error from (34)
turns out to be 0.521 percent-versus 0.04 percent computed For b = 11 and N = 1024, the fractional error in fundamenabove via the FFT method. Indeed, a comparison of (33) and tal amplitude, as given by (38), is 0.89 percent. Higher
(34) shows that the FFT method is superior to consecutive accuracy can, of course, be achieved with larger word
lengths. No attempt will be made to compare this accuracy
summation and division by N.
with that of simple methods such as the one consisting of use
of A1 = (Vmax - Vmin)/2. The latter method leads to erronB. Fundamental Frequency (f])
results when harmonics are present, and is also highly
eous
For this analysis we assume that the signal consists of a susceptible
to errors caused by additive noise. The FFTsingle sinusoid. Now,
based method, not surprisingly, yields A 1 that is quite robust
relative to the influence of harmonics as well as additive
S'(l) = (1 + CJ)S(l)
noise.
S'( + 1) = (I + C2)S(l + 1)
where C and 2 are uncorrelated, zero mean and have a D. Phase of Fundamental (&)
variance equal to 2.5nU2 [7] (n = Log2 N; see Appendix).
Again, we assume that the signal consists of a single
Then
sinusoid. Now from (20a)
+
(39)
01 = Phase {S'(l)}- ab' + 7r/2
'( +
X =
1(
())
= 1+ S3)
51
()|)
1+ W
where
IS(l)I
IS'(lH
1 N(N - 1)(2N - 1) V2 (2
6
N2
The noise-to-signal ratio is, therefore,
N
Var {Vd} _

(35)

S'(l) = U'(l) + jV'(1) = S'(l)

exp

[j0'(l)].

(40)

For convenience of discussion, and without loss of

where C3 has a variance equal to 5no2, the sum of the generality,5 we assume that 0 < 0'(l) < r/4 holds. Then
variances of 1 and C2. In turn [15]
tan 0'(l) = V'(l) - V(l) (1 + q1l + q12) = tan 0(l)(1 + 13)
U'(l) -U(l)
(1 1 Var (x')
Var 6' = Var(1 +

(41)

5na?.
(36)
Since the fundamental frequency is given byf1 = (1 + 6)fo
and since 1 . 20, we have
(37
Var {f'} 5na
<

f2N

400

where 1 and 12 are random variables accounting for

computational roundoff in V'(l) and U'(l), respectively. The
variance of the equivalent sum random variable n3 can be
shown to be 2.5na'. Now using the Taylor series
approximation
tan (0 + A0)

tan 0 +

O2 0

and N = 1024, the fractional error in

For example, if b
frequency turns out to be 0.01 percent. This error compares or
quite favorably with that produced by simpler methods. The
A0 < [tan (0 + AO) - tan 0]
latter, typically consist in locating the crossings of the signal
at the estimated "true dc level" and counting the number of we obtain from (41)
samples for an integer number of cycles. Errors for this
0'(l) - 0(l) < tan 0(1)13
simple method range from 0.1 to 1 percent and may increase
if the signal is masked by noise. It is shown in Section VIII
(42)
< .3.
the interpolated FFT method (of computing the fundamen5
tal frequency) is quite robust relative to additive white noise
If 0 < U'(l) < V'(l), then tan- ' 0'(1) = 7t/2 - tan- 1 U'(l)/V'(1). Other
= 11

[10].

quadrants can be taken care of in the usual way.

JAIN et

al.:

HIGH-ACCURACY ANALOG MEASUREMENTS VIA INTERPOLATED FFT

119

Reflecting upon (39), it is recognized that

Var 11 = Var 0'(l) + Var (ab'),
so that using (33) and (27) we have
Var 01 = 2.5na2 + 7r2(5na2)
_ 52n 2
For b = 11 and N = 1024, the rms value ofthe error in phase
computation is (less than) 0.0064 radian. This represents 0.2
percent of one cycle. Needless to say, the presence of
harmonics does not influence the computation of the fundamental phase (because the spillover is generally not
significant). However, in a conventional approach to phase
measurement consisting of locating the crossing of the true
dc level, the effect of harmonics can be disastrous; with a
10-percent third harmonic, a phase error of up to 0.1 radian
can accrue.

TABLE III
FFT OF A SIGNAL WITH A FUNDAMENTAL SINUSOID,
THIRD HARMONIC, dc, INITIAL TRANSIENT AND NOISE
(SECTION VII)
Frequency
Index

|S(i)|
I

Phase

0
1

590.117
189.587

-0.1025

19
20
21

978.818
5785.969
1412.702

-0.8239
-0.8383
2.2776

59

179.873

0.4912

60
61
62

520.271 (532.764)

SO)

0.0

0.3655 (0.3168)
768.884 (755.324) -2.8704(-2.8384)
229.564
-2.9030

VII. SIMULATION RESULTS

A simulation example will be presented in this section.

The purpose is to demonstrate the effect of the presence of a)
an initial transient and b) an additive white noise. That is the
signal x(t) is taken to be of the form
x(t) = y(t) + p(t) + w(t)
or, in the sampled form
x(k) = y(k) + p(k) + w(k).
Here, y is the periodic signal, p the transient component, and
w the additive white noise. Both the transient part and the
noise part are so scaled that their rms value is approximately
five percent of the rms value of the periodic signal.
Specifically, let N = 2048 and

y(k) = 0.2 + 6.0 sin [

(20.2)k + 0.1 + sin [N (60.6)kj

p(k) = 1.9293e- k50

w(k) = (0.05)(4.3058)u(k)
where u(k) is a zero mean, unit variance, uncorrelated
sequence. The FFT of the signal is shown in Fig. 2(a) and a
partial listing given in Table III. The results of computations, as described in Sections III and IV, are presented in

the second column of Table IV. The true values are indicated
parenthetically in the first column. Clearly, the parameter
values obtained are quite accurate despite the effect of noise
and the initial transient.
One more experiment was conducted in which we let
p(k) = 0, and w(k) = (1.0)(4.3058)u(k) while y(k) remained
the same as before. That is, now the corrupting influence is a
100-percent additive noise while the transient term is set to
zero. The FFT of the corresponding signal is shown in Fig.
2(b). The computed parameters, listed in column three of
Table IV, demonstrate the noiseworthiness of the algorithms developed in the paper.
The simulation was carried out on an Interdata 8/32
computer wherein the FFT of the signal was computed via a

Note:

The values within parentheses are obtained

after subtracting the contribution of
the fundamental component S1(i).

TABLE IV
CALCULATED PARAMETERS FOR A SIMULATED SIGNAL
(SECTION VII)
Calculated Parameter Values
Experiment 1
y(k) + 5% noise
+ 5% transient
V

av

f 1 (20. 2f0)

Experiment 2
y(k) + 100% noise

0.288

0.2405

20.196f

20.204f

A1

(6.0)

6.025

6.2016

(0.1 rad)

0.116 rad

0.0928 rad

A3 (1.0)

0.9913

0.8295

(0.0)

0.0237

-0.3979

0.2307

0.2091

4.3236

4.4290

0.17068

0.1324

03
VDC
v

(0.2)

(44.3062)

THD (0.1707)
* Because

of the presence of the initial transient, it

is not possible to compute the true D.C. level
accurately in Experiment 1.

Fortran routine. However, Honeywell, Inc., have implemented the FFT algorithm in hardware; the overall system,
designed for high accuracy measurement of analog parameters from dc to 5 MHz, uses the algorithms presented
in the paper.
VIII. EFFECT OF ADDITIVE NOISE
Although the simulation example has demonstrated the
robustness ofcalculated parameters, it is useful to have some

120

IEEE TRANSACTIONS ON INSTRUMENTATION AND MEASUREMENT, VOL.

IM-28, NO. 2, JUNE 1979

rlnr)(

5000 F-

4000 H
LL
LL

3000 F(0
4

2000

1000

0'

.
0

....eee000@S00**
to

*.0...
30

20

5000

INDEX

FREQUENCY

2 9+1
6000

50

40

70

60
m m+1

(a)

4000 _L.
L.

3000 _-

C,
4

2 000

000 _

0. *0

........ I.. 1
-

0
0

0~~~~~~~~

10

3'0

20

k+l

40

FREQUENCY

50

INDEX

(b)

60

70

m m+i

Fig. 2. Magnitude FFT of signals considered in a simulation study (Section VII).

theoretical estimates of the bias and standard deviation in

the presence of noise. Let us assume a signal of form
x(t) = A1 sin (2irnf t + &1) + w(t)
where w(kA) is zero-mean white noise with mean power U2.
Then, using afirst-order analysis [15], validfor rms noise-tosignal ratios of up to 0.1, thefollowing bounds can be obtained.

eters can be accomplished via interpolation of the FFT of

the digitized signal. For general multifrequency signals also
the interpolation approach leads to an explicit formula for
the true amplitudes and phases provided the tone frequencies are known beforehand. It was demonstrated further that
these algorithms are more robust to computational roundoff, to additive white noise in the signal as well as to an
initial transient, compared to some commonly used timeFractional bias
domain formulas. Potentially, another benefit ofthe discrete
Fractional S.D.
Fourier approach appears to be the following. Before actual
0
2 a/AIX>N:
Frequency
the signal must generally pass through stages
measurement,
2 ao/(2NA2)
Amplitude
4.203a/A l N
of preamplification and prefiltering, which introduce small
but unavoidable distortion. Assuming it is characterized by
IX. CONCLUSIONS
a known frequency function H(f ), nonzero over the range
Accurate measurement of the parameters of a periodic
1/2A < f < 1/2A, this distortion may be inverted by dividsignal is important in many practical applications. We have ing the signal FFT by this function.
shown that a high accuracy measurement of the fundamenTwo further improvements, however, remain to be
tal frequency, amplitude and phase, and five other param- explored. First, if the additive noise is white and its spectral
-

121

JAIN et al.: HIGH-ACCURACY ANALOG MEASUREMENTS VIA INTERPOLATED FFT

p (E)
density known, then certain minimum variance estimates
may be obtained for the magnitude ofthe true FFT components [11], [12]. Certain alternative methods, such as the
2( b- )
maximum entropy method [2] and Pisarenko's method [2],
do yield optimal estimates in the presence of noise. However,
their computations are involved, for example requiring
correlation functions, inversion of a matrix and finding roots
of a polynomial, and they do not yield phase values. A
second possible improvement in our method pertains to the
-b
-b
-2
0
2
transient may perhaps be assessed and its undesirable effect
subtracted from the signal. Work on these topics is Fig. 3. Probability density of the roundoff error fraction (E-distribution).
underway.
APPENDIX
This is shown in Fig. 3. For convenience we call this as
DFT, FFT, AND ARITHMETIC ROUNDOFF
s-distribution. Note that the noise-to-signal power ratio is
Let x(kA) = x(k), k = 0, , N- 1 be the samples of a
band-limited signal x(t). It is assumed that the sampling
a2= Variance (e) = 2 -2b/3.
(A4)
frequency exceeds the Nyquist rate so that aliasing of spectra
b) The computed spectra S'(i) consists of the sum of the
does not occur. The DFT of the N-point sequence is given as
true spectra S(i) plus computational noise S(i). The variance
[7], [9],
of the noise term S(i) is of course dependent upon the nature
=
- (Al) of the signal x(k). If the signal is assumed to have uniform
S(i) = I x(k) exp(-j j ik)I
spectra, then it is shown by Oppenheim et al. [7] that7
N
k 0 i
while the inverse relationship is
NSR = Var IS(i)
2 2
(A5)
N
IN -1 S e
I
S(i)1
-I
lE
i
(A2)
1) () e p
When the signal is sinusoidal, simulation experiments show
Note that the ith DFT coefficient refers to a frequency ifo a 15 percent higher NSR. We assume a pessimistic NSR of
where A = 1/T is the frequency resolution achieved; 2.5nU2 for all signals considered.
c) The addition or multiplication of two numbers results
T = NA is the total duration of the signal. We assume that
in
roundoff error:
N = 2", i.e., N is an integral power of two. As is well known,
this permits a fast implementation of either of the above
(A6)
(y + z) = (y + z)(1 + c1)
relationships through a radix-2 FFT. Often, this algorithm is
(A7)
(yz)' = (yz)(1 + 22)
carried out in hardware which results in very high-speed of
computation, typically 20 ms for a 1024-point DFT. Higher where e, and 22 each have the c-distribution.
speeds are currently possible and still higher ones will
undoubtedly be achieved in the future.
REFERENCES
In the paper the complex DFT coefficients S(i), i = 0, , [1] D. C. Rife and G. A. Vincent, "Use of the discrete Fourier transform
N/2 are used to measure the analog parameters of the signal.
in the measurement of frequencies and levels of tones," Bell Syst.
Tech. J., vol. 49, pp. 197-228, February 1970.
Since high accuracy in measurement is of concern, it is
[2]
Proc. RADC Workshop on Spectrum Estimation, Rome Air Developthe
to
arithmetic
roundoff
errors
accrunecessary recognize
ment Center, Rome, NY, May 1978.
ing in the FFT implementation of (1). To avoid discussing [3] J. R. Mick, "Fast computational devices for digital signal processing," in Proc. National Telecom. Conf., pp. 10.4-1 to 10.4-5, Dec.
various types of number representations and arithmetics
1976.
that can be employed, we restrict attention to only one [4] H.
L. Logan and D. G. Forney, "A MOS/LSI multiple configuration
important case. The case of interest involves i) floating-point
9600 bps data modem," in ICC Conf. Rec. (Philadelphia, PA), 1976.
number representation and ii) 2's complement arithmetic. [5] P. J. VanGerwen, A. M. Verhoeckx, H. A. Van Essen, and F. A. M.
Snijders, "Microprocessor implementation of high-speed data
With this assumption, the following observations can be
modems," IEEE Trans. Commun., vol. COM-25, pp. 238-250, Feb.
made
1977.
a) Let z be a number stored in floating-point form with a [6] I. Koval and G. Gara, "Digital MF receiver using discrete Fourier
transform," IEEE Trans. Commun., vol. COM-21, pp. 1331-1335,
b-bit mantissa. Then the stored number is given by [13]
Dec. 1973.
V. Oppenheim and C. J. Weinstein, "Effects of finite register length
[7]
A.
z = z(1 + c)
(A3)
in digital filtering and the fast Fourier transform," Proc. IEEE, vol.
60, pp. 957-976, Aug. 1972.
where E is a zero-mean random variable with a uniform [8] D.
V. James, "Quantization errors in the fast Fourier transform,"
probability density function6
IEEE Trans. Acoust., Speech Signal Processing, vol. ASSP-23, pp.
277-283, June 1975.
< 2 b and 0 otherwise.
p() = {2(b- 1) for
[9] B. Gold and C. M. Rader, Digital Signal Processing. New York:
'

McGraw-Hill, 1969.

6 Experimental evidence indicates some departure from this ideal [7];

however, for our purposes the assumption of uniform distribution is
adequate.

7 An exception to formula (A5) arises when i = 0 and exact representation of the number 1.0 is utilized in the arithmetic.

122

IEEE TRANSACTIONS ON INSTRUMENTATION AND MEASUREMENT, VOL.

[10] V. K. Jain, W. L. Collins, and D. C. Davis, "High accuracy measurement of analog signal parameters via IFFT," Honeywell Res. Rep.,
1979.
[11] B. P. Agrawal and V. K. Jain, "Bandwidth compression of noisy
images," J. Comput. Elec. Eng., vol. 2, pp. 275-284, Oct. 1975.
[12] B. P. Agrawal, "Digital processing of two-dimensional signals,"
Ph.D. dissertation, Univ. South Florida, 1974.
[13] W. R. Bennett, "Spectra of quantized signals," Bell Syst. Tech. J., vol.
27, pp. 446-472, 1948.
[14] H. G. Tucker, An Introduction to Probability and Mathematical Sta-

IM-28, NO. 2, JUNE 1979

tistics. New York: Academic Press, 1962.

[15] M. Eisen, Introduction to Mathematical Probability Theory. Englewood Cliffs, NJ: Prentice-Hall, 1969.
[16] V. K. Jain, "Filter analysis by Grammian method," IEEE Trans.
Audio Electroacoust., vol. AU-21, pp. 120-123, Apr. 1973.
[17] R. B. Blackman and J. W. Tukey, The Measurement of Power Spectra
from the Point of View of Communications Engineering. New York:
Dover, 1958.

[18] F. J. Harris, "On the use of windows for harmonic analysis with the
discrete Fourier transform," Proc. IEEE, vol. 66, pp. 51-83, Jan. 1978.

Special Purpose Ammonia Frequency

Standard A Feasibility Study
DAVID J. WINELAND, DAVID A. HOWE, MICHAEL B. MOHLER,
AND HELMUT W. HELLWIG, SENIOR MEMBER, IEEE

Abstract We have investigated the feasibility of a special purpose frequency standard based on microwave absorption in ammonia
gas (N"5H3). Such a device would potentially fill a need in certain
communications and navigation applications for an oscillator which
10- 9) than that provided
has medium stability, greater accuracy ( -O
by crystal oscillators, but a cost significantly smaller than that of
more sophisticated atomic frequency standards. A device was constructed using a stripline oscillator near 0.5 GHz whose multiplied
output was frequency-locked to the absorption of the 3-3 line in
N15H3 (' 22.8 GHz). Output between 5 and 10 MHz was provided
by direct division from the 0.5-GHz oscillator. Observed stability
was 2 x 10- " from 10 to 6000 s, and reproducibility (accuracy) is
estimated to be + 2 x 10- 9. The unique features of this device, which
include 1) high-performance stripline oscillator, 2) digital servo
techniques, 3) unique oscillator-cavity servo, 4) pressure shift com-

pensation scheme, and 5) acceleration insensitivity, are discussed.

Areas for further study are noted.
I. INTRODUCTION

T HE SPECIAL purpose ammonia frequency standard

has grown out of a need for a frequency standard
satisfying specific requirements not found in other precision
oscillators. Briefly, these currently available oscillators can
be divided into two classes: the quartz-crystal oscillators
and atomic "clock" oscillators. The quartz-crystal oscillaManuscript received February 13, 1978; revised November 27, 1978 and
February 26, 1979. This work was supported by the Advanced Projects
Agency of the Department of Defense and was monitored by ARPA under
Contract # 3140.
D. J. Wineland, D. A. Howe, and M. B. Mohler are with the Frequency
and Time Standards Section, National Bureau of Standards, Boulder, CO

80302.
H. W. Hellwig was with the Frequency and Time Standards Section,
National Bureau of Standards, Boulder, CO. He is now with Frequency
and Time Systems, Inc., Danvers, MA.

tors, while having good short-term stability and low cost

(- $500 to $2000) suffer three major drawbacks: 1) The
frequency is not invariant between units and is related to the
macroscopic parameters such as the dimensions of
the quartz crystal. Therefore, calibration is required initially
and subsequent recalibration is required due to "aging" of
the crystal. 2) The crystal oscillator is sensitive to vibration
and shock. These environmental factors affect the macroscopic dimensions of the crystal, and therefore can cause
step shifts in frequency. 3) The quartz-crystal oscillator is
temperature dependent and when oven-controlled requires
long warm-up time.
Atomic oscillators provide stabilities from one part in
1010 to one part in 1013 per year. Their cost ranges from
approximately $4000 to above $20000 depending upon
performance. Their excellent frequency stability and intrinsic accuracy make recalibration unnecessary for most applications, however this higher level of performance is not
needed in many applications. In addition, their warm-up
time is generally long and their performance under severe
environmental conditions (acceleration, vibration, temperature, barometric pressure, and magnetic fields) is inadequate
in some cases.
Presently there exists a need in many communication and
navigation systems applications for low-cost oscillators with
accuracy better than that provided by crystal oscillators, but
not as high as available atomic standards. With these needs
in mind, we investigated the possibility of constructing an
oscillator with 10-9 accuracy and 10-1o stability, which
could also be made environmentally insensitive, have quick
warmup and the potential for small size, weight, power
consumption and low cost.
-

U.S. Government work not protected by U.S. copyright.