298

COMMUNICATIONS,
I E E E TRANSACTIONS
ON

design, as a member of the Basic Systems Study Group of the Radar

Engineering Department. In 1952 he joined the teaching staff of the
Polytechnic Institute of Brooklyn, where he is currently Professor
of Electrical Engineering. He served as Head of the Electrical Engineering Department from 1961 to 1965. During the year 1965 to
1966 he was an NSF Science Faculty Fellow at the Laboratoire de

VOL.
NO. COM-22,

3,

MARCH

1974

Physique, Ecole Normale Superieure, Paris, France. For the 19731974 academic year heis on leave as a Visiting Professor at Columbia
University, New York, N. Y. He is the author of various technical
papers and several books in the field of communications, including
the second edition of InformationTransmission,Modulation,
and
Noise, and a forthcoming book on signal processing.

Use of Limit Cycle Oscillations to Obtain Robust

Analog-to-Digital Converters

Abstract-High quality analog-to-digital conversions are obtained

using simple and inexpensive circuits that require no high-precision
components. Samples of the analog signal are cycled rapidly through
a coarse quantizer while the roundoff error is fed back and subtracted from theinput. By means of this feedback, the coarsequantizations are caused to
oscillate between levels, keepingtheir running
average representativeof the input. A binary coding of the quantized
values, summed over Nyquist intervals, provides a high resolution
PCM output. The precision is determined by a product of the cycle
rate and the spacing of the coarse quantization levels. The system
is surprisingly tolerant of inaccuracies in gains and threshold settings; indeed, it has many of the desirable properties of classical
feedback servomechanisms. An 8-bit limit cycling converter intended
for 1-MHz signal bandwidths has been fabricatedof standard components that, in total, cost less than $150.

linearityanddc level settingsare usually adequate fol

video and speech transmission. On the other hand, then
canbe noappreciablediscontinuitiesin
the amplitudc
transmission characteristic; this fault is common of somt
well known A/D converter techniques.
Thecircuit describedhere is basedonatechniquc
proposed byCutler
[la] andelaboratedbyBrahn
[13] and by Riliura et al. [14]. We will show that it if
verytolerant of circuitimperfection. I n essence, eacl
input sample is coarsely quantized and digitally coded ir
a circuit that is not necessarily precise. An estimate o
the quantization error, together with any spurious errors
is subtracted from theinputvalue,andthe
differencc
becomes the input for the next cycle. After the proces;
I. INTRODUCTION
has repeated many times a t high speed, an average of thl
ECENT' technicalliterature [I]-[ll] is burdened digital outputs occurring in each sample time is a usefu
withtechniques for convertinganalog signals into digital representation of the input signal. By this process
digital form, yet there still is need
to improve the methods speed of operation replaces precision of circuit elements
The tradeoff is
invaluable
for designing integrate(
of connectinganalogsources
tothe
powerfuldigital
circuits.
Simple
inexpensive
implementations
of the con
systems thatare
becoming
available
for processing,
verter
have
provided
high
resolution.
For
example
storing,andtransmittinginformation.The
difficulty of
Picturephone@
signals
have
been
coded
into
8-bit
word
A/D conversions stems from the need for high precision in
a
t
2
megawords/s
using
standard
circuit
component
nonlinearcircuits thatmustoperateatfastrates.
For
cost less than $150. Alogarithmicall
example, resolutions of about 1 in 200 a t 2 megasamples/s that,intotal,
companded
coder
suitable for speech is significantly les
are desirable for videotelephone signals, and resolutions of
expensive.
D/A
converters
basedonrelatedtechnique
about 1 in 10 000 a t 8 kilosamples/s are needed for speech
have
been
implemented
both
for uniform and compandel
signals [SI. Providing reliable circuits
having
these
signals.
These
converters
will
be
described in a laterpapel
orders of absolute precision is difficult and expensive.
The technique tobe described here eliminates muchof the 11. A CIRCUIT FOR PULSEYCODE MODULATIOI
need for precision circuit components in A/D converters
(PCM)
that are intended for use in communication systems.
It
Thecircuit, shown in Fig. 1, uses a negative feed
takesadvantage of tradeoffs that are peculiar to comback
loop containing an integrator and a coarse quantize
munications, in contrast to the requirements needed for
that
spans
thesignal range. A digital representationof th
instrumentation. Two percent tolerances in gain stability,
quantized signal increments the contents of an accumu
lator. After signals have cycled around the feedback loo
Paper approved by the Associate Editor for Data Communica-

tions of the IEEE Communications Society for publication without

oral presentation. Manuscript received September 24, 1973.
The author is with the Bell Laboratories, Holmdel, N. J. 07733.

@ Registered
service mark of the American Telephone an
Telegraph Company.

299

CANDY:
E

SAMP

CYCLE
CLOCK

rn

Cl&

TABLE I
THEQUANTIZATION
q IS A REPRESENTATION
O FT H ES U MO FT H E
DIFFERENCES
BETWEENTHE INPUTAND THE PREVIOUS
QUANTIZATION

La

INTEGRATOR
DIGITALDIGITAL

ANALOG
INPUT

Input

Integrand

Error
en = (X,

Xn

QUANTIZER

- qn-1)

I n = In-1

Quantization
en

qn

= [In1

DELAY
I

WANTIZED
ANALOG SIGNAL

CY L E
CLOCK

8
8.2
8.2
8.2
8.2
8.2

-0.8
0.2

Fig. 1. Outline of the A/D converter in which digital0.2

connections
are drawnwiththicker
lines thanthe analog connections.

a set number of times, an output is taken from the accumulator and its content set to
zero before cycling for
the next sample.
Table I illustrates the action of an algorithm using a
quantized signal that assumes only integer values and an
input amplitude equal to
8.2. It generates quantized values
that repeat themselves with the pattern 88 988. In fact,
the quantizedsignal oscillates between the two levels
that bracket the input amplitude with duty ratio that
attempts to keep the average quantized value equalto the
input.
In order to describe the action of the circuit with linear
equations, we will employ the technique described in
[lo]. The quantization will be approximated by addition
of an error, E , that is assumed to be uncorrelated with the
input, x. We then have the equation
I

.-Nr

where T is the cycle period, and q(nr) the nth quantized

signal. The integrator has unit output
for a unit input
persisting for r seconds. Since q ( t ) is a sampled function,
its integralmay be writtenas a summation;then reorganizing ( 1 ) and dividing by N , the number of cycles
in a sample time, we get the result
I N

q(nr) = -!-.

x ( t ) dt

N.r)
+ 4N

(2)

N o
Nr
which shows that, for appropriatestarting conditions,
the average of the quantized signals represents the average
input with a quantization error that is N times less than
the errorassociatedwith
the coarse quantizer.High
resolution is obtained by cycling the signals repeatedly
around the feedback loop until N is sufficiently large. For
the application to Picturephone signals, the quantizer has
sixteen levels andthe cycle frequencyis 32 MHz;an
average of sixteen quantizations provides 8-bit resolution
at 2-MHz sample rate.
The Appendix gives a more thorough analysis
of the
system. The following sections illustrate the more important properties with simple examples.

8.4
8.6
7.8
8.0
8.2
8.4

0.2
0.2

and sets both the integrator and the digital accumulator

t o zero content before processing each sample. Analysis
given in the Appendix indicates that inaccuracy of loop
gain or zero level can have significant influence on the
digital output unless N is especially large.Aprocedure
that avoids the need for high precision connects the analog
1, and
signal directly to the subtractingcircuitinFig.
cycles data continuously around the feedback loop without everresetting the integrator.The digital ouput is
periodically read from the digital accumulator, which is
then reset to zero content. Most of the following discussion
is concerned withthis continuousmode; itsaction resembles that of aservosystem
because, for baseband
frequencies, the effective gain of the integrator is large.
Thus distortion and inaccuracies in the forward path of
the feedback loop have only a second-order effect on the
overall properties of the converter.
The continuousmode is suitablenot only for single
input signals, but also for time multiplexed inputs. Section
VI1 will discuss this application.
IV.RESOLUTION
This section is concerned with the noise introduced into
the signal by the converter when its parameters are ideal.
Effects of circuitinaccuracies will be discussed in the
next section. As before, let r be the cycle interval, and let
T = N r be the sampleinterval. Processing of the Kth
[ ( K - 1)Nlth
output sample commences afterthe
cycle and endswith the (KN)th cycle. Applying (1) a t
the start and finish of this sample, we have
(K-1) N
1 (K-UT
C q(n.1 = ;
~ ( t dt)
( K T - T ) (3)

and
KN

p(nr)
0

:r

+ (KT).

x ( t ) dt

(4)

Taking the difference of these equations and dividing by

N gives the important general result
1

KN
(K-1)N

111. OPERATINGMODES
The initial condition for the integral in (2) is dependent
on the application, and severaldifferentprocedures are
feasible. Anobviousone uses asampled and held inputEquation

q(n7) =

KT

x ( t ) dt

(K-l)T

+ (KT)

( K T- T )
. (5)
N

(5) shows that the average of the quantized

300

IEEE TRANSACTIONS ON COMMUNICATIONS, MARCH

TIME-AMPLITUDE

BR IGHTNESS

1974

TIME-AMPLITUDE

BRIGHTNESS
(b)

(4

Fig. 2.

(a) Response of PCM to a ramp input. (b) Response of

the limit-cycle converter t o a ramp input.

values over the sample period representsthe average value

of the input signal with error given by 1/N of the difference between the quantization error in the last
cycle
and the error in the lastcycle of the previous sample.
It is well known [7] that quantizationnoise e has
uniformprobability.densityspanningmagnitudes
up to
half the quantization stepsize u and a flat spectral density,

The noise introduced by the converter is the firstdifference of suchquantization noise; ithasatriangular
probability densit.y function spanning magnitudes up to a
step size and baseband spectral density given by the first
quarter cycle of a sinewave, Le.,

Thus, using a coarse quantizer with

L levels provides
resolutions that are comparable to what is obtained from
ordinary PCM converters having N L levels. Although the
rms noise from this converter is 3 dB greater than with

ordinary PCM, it usually is lessannoyingbecause it i;

composed of higher frequency components. The picture;
in Fig. 2 compare the visibility of the noise introduced b;
ordinary 5-bit PCM with the noise introduced by a 5-bi
limit cycling converter. The input was a sawtooth voltagl
and in each case, the decoded output is displayed both a
a trace on an oscilloscope and as a brightness modulatiol
of atelevisiondisplay.
The response of thisconverte
resembles that of PCM with patterned dither [SI.
Equation(5) shows thattheoutput signal is repre
sentative of the averageinput signalover
theentir,
sample interval,notjust
overasmall
aperture withil
the sampleinterval,as
is the case withmostordinar:
PCM coders. Such avcraging is equivalent to a low-pas
filter with spectral response [sin ( w T / 2 ) / ( T w / 2 ) ] ,whicl
is plotted in Fig. 3. I n many applications, this filterin!
action is useful, especially when signals have
significan
out-of-band energy. The effective gain of the filter fall,
4 dB in the
signal band, but this drop
is easily compensate(
by preemphasis.Alternatively, it may becorrected b:
using a sampled and held input plus the technique fo
avoiding errors caused by transients that is described ir
Section VII.

301

CANDY:LIMITCYCLEOSCILLATIONS

TABLE I1
THE INTEGRATOR
IS 0.2

THEGAININ
Integrand Error

Input
en = (2,

8.48
8.52
8.36

0.2
0.2
-0.8
0.2

0.2

- qn-1)

8.2
8.2
8.2
8.2
8.2

In

In-1

+ 0.2~n

Quantization
qn

= [In1

8.40
8.44

8
8
8

8.40

8
8

TABLE I11
THEGAININ THE INTEGRATOR
IS 1.3
Integrand Error

Input
rn

= (2, -

E,

qn-1)

In

In-1

+ 1.3~n

Quantization
Qn

8.40
8.66
7.62

0.2
-0.8
7.88

0.2

8.14

1 1
00.4

Fig. 3.

0.2
0.2

8.2
8.2
8.2
8.2
8.2

= [In1

8
9
8
8
8
8

8.40

\
1

0.1

0.4
R E L A T I V E B A S E BAND FREOUENCY

Components of the transmission gain. (a)

01 = 1. (c) 01 = 0.75. (d) 01 = 0.5.

I
a =

.o
1.9. (b)

V. CIRCUIT PARAMETERS AND THEIR EFFECT

ON THE SIGNAL
A . Choice of Loop Gain and Circuit Stability
Stability is a'primary consideration in the design of the
circuit in Fig. 1. The example given in Table I shows that
the algorithm used in this converter is stable, in thc sense
that oscillations of the quantized value are bounded for a
steadyinput, i.e., ithasalimit
cycle. Thisexample
assumes unity gain in all elements of the feedback loop.
Tables I1 and I11 give the correspondingresponses for
gains 0.2 and 1.3 in the integrator. In all three examples,
theinitial state of the integrator is such that the first
amplitudepresented
tothe
quantizeris
8.4. The algorithmsillustratedinTables
I, 11, and I11 are stable,
and the average quantized signal equals the input amplitude. Aninvestigation of asimilarfeedbackloop,describedin [lo], demonstrates that the circuit is stable
for values of gains less than 2. However, for gainsin
excess of 4/3, there is a tendency for the signal to oscillate
between levels that are remote from the input amplitude.
This may cause the quantizer to overloadand thus increase the error E . It is recommended t.hat loop gains be
kept less than 1.5.
Using gains that differ from unity distorts the transient
response of the circuit, as Fig. 4 demonstrates.These
graphs show thequantized signal, q, responding to a
step change of input forvariousgains, a. The effective
transmission through the system is given by (19) in the

TIME 3

q10 Y

I-

!I

TIME-

0
(C)

Fig. 4. Responses of the feedback loop to step changes' of input,

0.33 t o 8.20, withloop gains. (a) 01 = 1. (b) a = 0.5. (c) 01 = 1.5.

Appendix; it can be expressed as

I H(w) I

[l

a[sin ( w T / 2 ) / ( w T / 2 ) ]
. (8)
(a - 1)2
2 ( a - 1) cos ( w r ) ] 1 ' 2

Fig. 3 shows the two major components of this function

plottedagainstbasebandfrequency
for N = T / r = 16,
and gains a = 1.9, 1, 0.75, and 0.5. Evidently,with

302

COMMUNICATIONS,

ON

IEEE TRANSACTIONS

MARCH

1974

N 2 16, gain tolerances of f 5 0 percentshouldbe

acceptable for most

communication
applications.
Thus
designing an amplifier to perform the integrating and subtracting opefations is simple, especially because its noise
and dc offset can be related to equivalent signals added
directly to the input. The
amplifier specifications need
be no more stringent than that of ordinary relay amplifiers.

B. Threshold Settings

Providing accurately positioned voltage thresholds can

be a very difficult task, especially when fast decisions are
required.Ideally, the threshold levels should be placed
centrallywithrespect
tothequantization
levels [7],
but fortunately, use of the continuous mode of operation
relaxes the requirement on threshold positioning.
Slow wholesale drift of theset of thresholdshas insignificant effect pn the signal, because the
integrator
can provide dc coGrection a t negligible input cost.
Displacement of individual thresholds with respect to
other ones can be studied by simulations similar to those
describedinTables
I, 11, and 111. A useful empirical
method for predicting the consequences of displaced
thresholds is to define a local gain of the quantizer as the
ratio of theoutput level spacing t o the corresponding
thresholdspacing.Thereusuallyislittlepractical
difficulty in providing circuits for which hhe effective loop
gains have atolerancerange
of f 2 5 percent.
for" all
p
INPUT AMPLITUDE
signal amplitudes.
(C)
I n practice,circuitperformanceshavebeenfound
to be very tolerant of threshold spacing. This is possibly Fig.'5. (a) Graph of q for a ramp input. (b) Effect of displacin
level L3 by 20 percent of a step size. (c) Effective dc transmissio
becauseadecrease in the spacing of some thresholds is
characteristics.
usually accompanied by an increase in the spacing of
adjacent ones..Thus any tendency tooscillate a t increased log, ( L ) geometrically weighted sources activated by t h
amplitude is localized.
g. binary code. Use of weightedsourcesrequires.mucl
greater precision inorderto avoidlargeerrors
inste]
C. Alignment of the Quantization Levels
size a t transitions of major bits of the code.
t
The quantized signal appears in Fig.1 in two forms; the
OF TIMIN(
analog signal q and the digital signal q'. We have seen, VI.PRACTICALCONSIDERATIONS
CIRCUITOPERATIONS
in (5), that the average of the analogvaluescanbea
good representation of the input, but isit an average of the
Thecircuit uses two clocks: afast clock that time
digitalvalues that produces theoutput. Discrepancies decisions of the coarse quantizer and the accumulation c
between the analog and the digital
values affect theol!tput digital numbers, and a slower clock that times the outpu
directly, since mismatch is notprotectedby
feedback. gate. In order that these clocks be correctly phased, it ha
However,one of theattractivefeatures
of thisA/D
been advisable to generate them as part of the convert6
converter is that discrepancies in the coarse quantization circuitry.Fig.6illustratesthemethod
used. The fat
levels resultinasmoothnonlinearity
of transmission, clock CF is divided in frequency by the factor N to t h
rather than introduction of noise or sharp discontinuity.
desired samplingfrequency C S . Thephase of this slol
Fig. 5 illustratesthe effect diagramaticallybytracing
clock is set by means of a .timing input pulse C H , whic
the quantized signal for ramp inputs. The course quanti- resets the divider if there is significant phase discrepanc
zation levels are uniformly spaced in Fig. 5 (a), and one between C S and CH. For television application, CF hs
step size isincreased by 20 percent in Fig. 5 (b) . Fig. beena32-MHz
signal, C S a2-MHz signal, and CH
5 (c) shows thk effective dc transmission characteristic for horizontal drive signal.
the two cases. This graphwas obtained by assuming linear
VII. TIME-SHARED OPERATION
interpolation of valuesbetween the qdantization levels.
It is advisable that the analog signal q be constructed
The continuous mode of operation does not reset
th
by adding together L equal binary sources, one activated integrator betweensamples.Therefore,
data are earrie
by each of thethresholds, rabbr than ,by permuting overfromsample
to sample.Thiswouldbevery
ur

303

CANDY:LIMITCYCLEOSCILLATIONS

SAMPLE CLOCK

Fig.6.Generating
suitable clocks. The sample clock is automatically rephased if it deviates by more t h a n f 1 / 1 6 of a sample
interval from clock C H .

desirable for time-sharedapplicationwhereconsecutive

samples come from unrelated sources [SI. The circuit can
be used forsuchapplications,
however, withouthaving
to actually reset the integrator. Thecycle frequency would
be increased in order to provide more cycles per sample.
Then, the first few cycles are used to clear residual transients from the system. The quantized outputs
for these
cycles are ignored by holding the accumulator at zero
content.When the accumulator is released, the action
of the circuit is substantially the same as previously
described. The noise, however, now has a flat frequency
spectrum but still has triangular probability density and
is 3 dB greater thanthe
corresponding PCM noise.
Ordinary PCM quality could be obtained by increasing
N by afactor of two or moreand rounding off the
digital output words. Indeed, this is the method that was
used to obtain the pictures shown in Fig. 2(a).

CYCLE
CLOCK ,
I

_______ _______ ____-________-_

I

SIXTEEN WIRECONNECTION

I
J

CYCLE

qc.
ET

CONCLUSION
Themethod of analog-to-digital conversion described
hasmany
of the desirableproperties
of the classical
feedbackmethod used in servomechanisms. Instead of
making an accurate evaluation of each signal sample, or a
succession of approximations
requiring
precise components.andexactingadjustments,
it uses the limit
cycles in a coarsely quantizing feedback coder to give a
precise determination of the averageinput value. The
main limitation of the method described is the need for
fast cycle times. There is littlepractical difficulty in
using it forvideotelephone,
but broadcast television
signals would challenge present circuit art. The technique
is particularly attractive for speech signals, because the
levels of the coarse quantizer can be companded. Then
the interpolativeaction of the feedback loop provides
uniform resolutionbetween
each pair of levels. Such
quasi-linear companding is used for transmitting telephone signals [SI.
The circuit in Fig. 7 is a practical design for an A/D
converter; it is capable of 40-MHz cycle rates, providing
8-bit outputs at2.5 megawords/s. Several of these circuits
have been put t o use in interframe coding experiments.

APPENDIX
A. Analysis of the System with Arbitrary Loop Gain
The diagram in Fig. 8 shows a linear representation of
the basic systemandnotation
for the signals. Small

R: REGISTERS

CLOCK

Fig. 7. Schematic circuit of an 8-bit A/D converter.

Fig. 8. Representation used for analysis of the A/D converter.

letters, z ( t ) , represent
time
functions
and capitals,
X ( , ) , represent the corresponding spectral function. will
be the cycle time and T = N T , the sample time. We can
describe the feedback by the following equation:

where A accounts for the zero offset of the integrator with

respect to the quantizer, and (Y is a gain constant. Now let

304

IEEE TRANSACTIONSONCOMMUNICATIONS,MARCH

then (9) can be written, in terms of sample functions, as

M

q(M7)

{x1(n7)- q(n7 -

= a

7))

+ A + (MTj.

(11)

1974

Let us first consider the effects of resamplingon the

signalitself, considering the noise later. If the original
input z ( t ) is restricted to baseband,thenthe
sampled
functions X ' ( M 7 ) and 3 (MT)
will contain energy only
in the frequency bands

The equivalent spectral relationship [lo], [14] is

and
where 2-' = exp ( - j w r ) , a cycle delay. I n order to complete the description of the entire system, we must introduce the action of the digitalaccumulator. It is convenient to regard its actionas filtering followed by resampling. The filter is a simple running summation over
N previous cycles generating the function

i(M7)

c
MT

= -

(13)

q(n.1

X(w)

X(w

+ $) ,

for integer i.

Resampling [14] these functions at intervals Nr will n o

affecttheshape
of theirbasebandspectraldensities
Therefore, the baseband spectrum of the output signal i;

Y(w) =

(M-NIr

a[sin (wT/2)/(wT/2)] X(,)

cos w ~ ( a 1) - j ( a - 1) sin WT

with spectral density

where
Using (12), we have
-Q(w) =

a[(I - Z-N)/(l- Z-l)][X'(~)/N]

1
Z-'(a - 1)

(1 - 2-9 ( E ( w ) A )
[jwN(l
Z-'(a - l ) ]

(15)

This expression confirms that the system is stable [15] for

< 2.
The initial transient described by the term

+ Z-'(a

(1 - Z-N)A/jw(l

1))

bill decay rapidly to zero if a is near unity. For realistic

values, it will influence the output only during the first
few cycles of the first sample. It could be significant if the
integrator were periodically reset, but here we are interested in the continuous mode of operation; A will be
henceforth ignored.
Now define
X(w) =
X(w)

X'

(w)

N T

Mr

Z(t)

dt,

E(w) (1 - 2 - N ) ( 1 - Z-'(a - 1)
N

+ Z-Z(a - 1)2 - *..)

( 20

Therefore, the output noise samples are given by

{ ( K T ) - ( K T - T ) ) - (a - . l ) { e ( K T -

(1 - 2--N)
(1 - 2-')

- ( K T- T

is the spectral density of the time function

?(MT)=

The magnitude of this function is illustratedinFig. 3

It should be emphasized that when the input is not banc
limited, the result (19) appliesonly for special case8
For example, using arguments similar to those given il
the following, while discussing noise, it can be shown tha
using a sampled and held input makes the output mor
sensitive to gain a.
Finally, we consider the effect of gain on the outpu
noise. The noise ~ ( t is) not limited to baseband, therefor(
the resampling operation is moreeasily studied in the tim
domain. From (18), the output noise can be written as

(17)

(M-Nh

which represents a running average

of the input over a
Nyquist interval. Equation (15) becomes
1

and it remains to sample this function at times K N T

KT.

- T)}

T)

+ (a - 1)2{c(KT- 27)

(KT- T - 27)) -

....

(21

The effect of using gainsslightly different from unit?

(a - 1) << 1, .is to increase the noise approximately b
the factor
1
[l - (a - 1)2]"2
[a(2 -

")]"2'

(22

which, for practicalvalues of a, is less than 5 percer

negligibl
increase in the rms noise, and therefore,a
amount.

IEEE TRANSACTIONS ON COMMUNICATIONS, VOL.

COM-22,

NO.

3,

MARCH

ACKNOWLEDGMENT
The author would like to thank R. H.Bosworth and C.
Vetrano for constructing circuits, and his associates in the
systemsResearchLaboratory
for helpfuladvice
and
encouragement.
REFERENCES
[I] H. Schmid, ElectronicAnalog/DigitalConversions.
New York:
Van Nostrand-Reinhold, 1970.
[a] D. F. Hoeschele, Jr., Analog-to-DigitallDigital-to-AnalogConversionTechniques.
New York: Wiley, 1968.
[3] D. H. Sheingold and R. A. Ferrero, Understanding A/D and
D/A converters, I E E E Spectrum, vol. 9, pp. 47-56, F p t . 1972.
[4] D. N. Kaye, FOCUS
on A/D and D/A converters, Electron.
Des., vol. 21, pp. 56-65, Jan. 4, 1973.
1.51 D. Kesner, J. Barnes,and T. Henry, Analog-to-digital converters, Motorola Monitor, vol. 11, pp. 23-27, Apr. 1973.
[6] R. C. Kime, The charge-balancing A-D converter: An alternative to dual-slope integration, Electronics, vol. 46, pp. 97-100,
MR.V
24
.__ _1973.
, ~
[7] K. W. Cattermole, Principles of Pulse Code Modulation. New
York: American Elsevier, 1969.
[SI H. H. Henning and J . W. Pan, I32 channel bank: System
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*
James C. Candy (61) was born in Crickhowell, Wales, on September 27, 1929. He
received the B.Sc. and Ph.D. degrees in engineering from the University of Wales, Bangor,
in 1951 and 1954, respectively.
From 1954 to 1956 he worked for S. Smith
and Sons, Guided Weapons Department,
Cheltenham, England. He studied nuclear
instrumentation at the Atomic Energy Research
Establishment,
Harwell, England,
from 1956 to 1959 and was a Research Associate at theUniversity of Minnesota, Minneapolis, from 1959 to 1960.
In 1960, he joined Bell Laboratories, Holmdel, N. J., where he has
investigated digital circuits and pulse transmission methods, and is
now concerned with methods for processing video signals.

Ambiguity Resolutionin Systems Using Omega

for PositionLocation
GABRIEL FRENKEL,

MEMBER, IEEE, AND

Abstract-The lane ambiguity problem prevents the utilization

of the Omega system for many applications such as locating buoys
and balloons. The method of multiple lines of position introduced
herein uses signals from four or more Omega stations for ambiguity
resolution. The coordinates of the candidate points are determined
firstthrough the use of the Newton iterative procedure. Subsequently, a likelihood function is generated for each point, and the
ambiguity is resolved by selecting the most likely point. The method
was tested through simulation.

I.INTRODUCTION

MEGA is a global radio navigation system consisting

of anetwork of eight transmittingstations.Each
Omega station transmits threebasic frequencies which are
Paper approved by the Associate Editor for Space Communication of the IEEE Communications Society for publication without
oralpresentation.Manuscript
received January 5, 1973; revised
June 27, 1973. This work was supported in part by NASA>Goddard
Space Flight Center, under Contract NAS5-21717.
The authors are with the Computer Sciences Corporation, Falls
Church, Va. 22046.

DIWAICAR G. GAN,

MEMBER, IEEE

integral multiples of a fundamental frequency, f

kHz. These are given by

1.133

Sf

fi

10.2 kHz

fi

11.33 kHz

lOf

f3

13.6 kHz

12f.

Anextremely importantpotential application of the

Omega system is as a means of locating platforms such as
buoys, balioons, andother low cost packages. Sucha
system was developed by NASA under the name Omega
PositionLocationExperiment
(OPLE).The three frequencies from a number of Omega stations, such as the
three shown in Fig. 1, are relayed by the platform whose
position is t o be determinedthrough the satellite to a
groundstation.Herethethree
signals originatingfrom
any particular Omega station are in effect combined to
yield a waveform at the fundamental frequency of 1.133
kHz.The difference intimes of arrival of two signals
is measured and hence, the difference in path lengths from