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3, MARCH 1993 501

Interpolation in Digital
Modems-Part I: Fundamentals
Floyd M. Gardner, Fellow, IEEE

Abstrucf- Timing adjustment in a digital modem must be SAMPLER

performed by interpolation if sampling is not synchronized to SIGNAL IN DATA OUT

the data symbols. This paper describes the fundamental equation PROCESSOR

for interpolation, proposes a method for control, and outlines the

signal-processing characteristics appropriate to an interpolator.
The material combines a review of previously known topics, 0. ANALOG RECOYCRY

presentation of new results, and a tutorial exposition of the

A companion paper will treat performance and implementa-


T IMING in a data receiver must be synchronized to
the symbols of the incoming data signal. In analog-

implemented modems, synchronization typically is performed
by a feedback loop that adjusts the phase of a local clock, or
by a feedforward arrangement that regenerates a timing wave
from the incoming signal. The local clock or the timing wave PROCESSOR
is used to sample (or strobe) the filtered output of the modem,
once per symbol interval. Message data are recovered from
the strobes. Timing of the strobes is adjusted for optimum C. DIGITAL RECOVERY
Fig. 1. Timing
CLOCK recovery methods.
detection of the symbols.
Implementation of the modem by digital techniques (a topic
of intense present activity) introduces sampling of the signal.
In some circumstances, the sampling can be synchronized
to the symbol rate of the incoming signal; see Fig. l(a) output-the same strobe values that would occur if the original
and (b). Timing in a synchronously sampled modem can be sampling had been synchronized to the symbols.
recovered in much the same ways that are familiar from analog Interpolation is a timing-adjustment operation on the signal,
practice. not on a local clock or timing wave. In this respect, it
In other circumstances, the sampling cannot be synchronized is radically different from timing adjustment in the better-
to the incoming signal. Examples include 1) digital processing known analog modems. Of all the operations in a digitally
of unsynchronized frequency-multiplexed signals, or 2) non- implemented modem, interpolation is perhaps the one with
synchronized digital capture and subsequent postprocessing of the least resemblance to established analog methods.
a signal. For one reason or another, the sampling clock must Several issues arise as follows.
remain independent of the symbol timing. See Fig. l(c) for a -What mathematical model of interpolation can be de-
nonsynchronized-sampling configuration. vised?
How is receiver timing to be adjusted, by digital methods, -How is interpolation to be controlled?
when it is not possible to alter the sampling clock? One answer -What characteristics are desirable in an interpolator for
is to interpolate among the nonsynchronized samples in such modems?
manner as to produce the correct strobe values at the modem -How is the interpolator to be implemented?
-What performance can be obtained? How large is the
Paper approved by the Editor for Synchronization and Optical Detection computing burden?
of the IEEE Communication Society. Manuscript received December 6, -What conceptual model is appropriate for interpolation?
1990; revised May 23, 1991. This work was supported under Contract
8022/88/NL/DG by the European Space Agency, Noordwijk, The Netherlands.
These are the matters treated in this paper and its ‘Ompanion
This paper was presented at the Second International Workshop on Digital [l].The first three issues are addressed here in Part I, and the
Signal Processing Techniques Applied to Space Communications (DSP’90), last three in Part 11 [I]. Attention is concentrated On high-
Politecnico di Torino, Turin, Italy, September 24-25, 1990.
m e author is with Gardner Research Company, Palo Alto, CA 94301,
‘peed methods, defined by a hardware-imposed constraint that
IEEE Log Number 9208042. no clock frequency can greatly exceed the signal sample rate.
027&0062/93$03.00 0 1993 IEEE


Interpolation as a Digital Signal Processing (DSP) opera-

tion has been covered extensively in the literature; excellent
examples and further references may be found in [2] and [3]. I I
By contrast, the role of interpolation in timing adjustment has
had comparatively meager attention [2, ch. 61, [4], [5]. In fact,
these latter references do not speak of “interpolation”, but of
“digital phase shifting” [2, ch. 61 and [4], or of “sampling-rate

conversion” [2, ch. 21 and [5].

It will be seen presently that the process of timing adjust-
Fig. 2. Elements of digital timing recovery.
ment includes substantially more than interpolation alone and
that “rate conversion” is a more accurate label. Nonetheless,
we will apply the term “interpolation” to denote all of the Assume ~ ( tto) be bandlimited so that it can be sampled
processes that are involved in adjustment of timing. at a rate l / T s without aliasing. If z ( t ) is not adequately
The term “interpolation” to describe the entire timing- bandlimited, aliasing will introduce distortion that causes
adjustment process appears to have been published first by a performance penalty. Interpolation is not an appropriate
a group at the Technical University of Aachen [6], [7]. The technique to be applied to wide-band signals.
term is also used by Bingham [8, p. 1671. Samples z(mT,) = z ( m ) are taken at uniform intervals
In light of the extensive DSP literature on interpolation, T,. The ratio T/Ts is assumed to be irrational, as indeed will
and of the large number of digitally implemented modems be true in all practical situations where the symbol timing
that have been built for voice-frequency telephone-line service, is derived from a source that is independent of the sampling
how is it that the literature on digital timing adjustment is so clock. These signal samples are applied to the interpolator,
sparse? which computes interpolants, designated y(lcTi) = y(k) at
Authors in the established DSP literature almost invariably intervals Ti. We assume that Ti = T / K where K is a small
restrict themselves to sampling-rate conversion by a rational integer.
factor, which can be modeled as a cascade of interpolation The data filter employs the interpolants to compute the
and decimation, each by integer ratios. Thus, the output is strobes that are used for data and timing recovery.
synchronized to the input. In the sequel, the interval Ti between interpolants is treated
But the inherent problem of fully digital timing adjustment as a constant, for simplicity of explanation. A practical modem
is that the signal sampling is not synchronized to the symbol must be able to adjust the interval so that the strobes can be
timing; the two rates are incommensurate and the sample times brought into synchronism with the data symbols of the signal;
never coincide exactly with desired strobe times. Recognition thus, the interpolation interval cannot be constant.
of incommensurability is vital to understanding the timing- All elements within the feedback loop contribute to the
adjustment problem. synchronization process. Timing error is measured by the tim-
Limitations of the DSP literature aside, why didn’t the ing error detector and filtered in the loop filter, whose output
timing adjustment problem arise more clearly in the design drives the controller. The interpolator obtains instructions for
of digitally implemented telephone-line modems? The answer its computations from the controller.
is that it indeed did arise, and was solved by the adaptive This paper concentrates on the interpolator and controller
equalizers that play so large a role in those modems. Besides alone, with little or no consideration of the data filter, the
correcting for transmission dispersion, an equalizer almost timing error detector, or the loop filter. One example of digital
incidentally also corrects the timing. For that reason, timing timing-error detectors may be found in [9], which also has
adjustment itself does not appear as a widely recognized, references to other examples. A n illustrative loop design and
distinct problem in the context of telephone-line modems. simulation may be found in Part I1 [l].
Digital implementation is now coming to higher speed com- The data filter is shown within the feedback loop, after the
munications links which do not require adaptive equalization. interpolator. That placement is not essential; the data filter
The need for digital timing adjustment must be faced by itself, could be outside of the loop, prior to the interpolator. A data
without embedding it inside an equalizer. filter inside the feedback loop introduces delay, with adverse
influence on loop stability.
Post placement may be advantageous when the data filter
111. MODEL
is more complicated than the interpolator-a likely situa-
tion-and when a relatively high sampling rate is employed for
A. Timing Loop interpolation. With postplacement, the data filter can decimate
Consider the feedback timing recovery of Fig. 2. (Feedfor- its output to the required strobe rate (just one or two samples
ward interpolation is also feasible, but not considered here.) A per symbol) and thereby save on computing burden. If the data
time-continuous, PAM signal z ( t ) is received. Symbol pulses filter is placed before the interpolator, then the sample rate
in z ( t )are uniformly spaced at intervals T . For simplicity, z ( t ) out of the data filter must be maintained high enough to avoid
is assumed to be a real, baseband signal, but those restrictions aliasing. On the other hand, simulation results [ l ] indicate that
can be removed without difficulty. quite modest sampling rates provide excellent results, even


Signal Analog Analog Froctionol

Samples Interpolated
Impulses Interpolants INPUT SAMPLE TIMES

c DAC c
xhTs) Y(kTi )

/ '"

at t = kTi
Fig. 4. Sample time relations.

Fig. 3. Rate conversion with time-continuous filter.

where 0 5 ,c& < 1. Timing relations are illustrated in Fig. 4.
Function arguments in (2) become m = m k - i and
with very simple interpolators. Thus, post placement may not
often be necessary.
(kT; - mT,) = (z pk)T,, and the interpolant is computed
at time kT; = (mk pk)Ts. Equation (2) can be rewritten as
B. Interpolator Equations
To derive a model for the interpolator, we recapitulate the
fundamental development of Crochiere and Rabiner [2, ch. 21.
The same basis underlies the adaptive rate convertor in [5].
Refer to Fig. 3, which shows a fictitious, hybrid ana- Equation (6) is the foundation of digital interpolation in
loddigital method of rate conversion. Convert the samples modems.
to a sequence of weighted analog impulses, which are applied If the interpolating filter has finite impulse response (FIR),
to a time-continuous, analog, interpolating filter with impulse then I1 and 12 are fixed, finite numbers and the digital filter
response hI(t). The time-continuous output of the filter is actually used for computing the interpolants has I = 12 -11 1 +
At this point, most DSP accounts of interpolation assume
that the ratio Ti/T, is rational. No such assumption will
be made here; real-world symbol rates are almost never
Observe that y(t)#z(t). There is no attempt, and no need to
synchronous with independent, fixed-rate sampling clocks.
recover the original waveform, contrary to most conventional
Assuming a commensurate ratio tends to obscure broader
interpolation. Since a modem is required to perform filtering
issues of control and implementation.
of signals there is no reason why some of the filtering cannot
When Ti is incommensurate with T,, the fractional interval
be included in the interpolator.
p k will be irrational and will change for each interpolant. If
Now resample y(t) at time instants t = kTi where Ti is
determined to infinite precision, PI, takes on an infinite number
synchronized with the signal symbols. In general, T;/T, is
of values, which never repeat exactly. This behavior is contrary
irrational; the sampling and symbol rates are incommensurate.
to that observed if Ti is assumed very nearly equal to T,-if
The new samples-the interpolants-are represented by
sampling is nearly synchronized. Then fik changes only very
y(kT;) = E z ( m T , ) h I ( k T i - mT,). (2) slowly; if p k is quantized, it might remain constant over many
m interpolations. If T, were commensurate with Ti, but not equal,
then jLk would cyclicly repeat a finite set of values, when the
Although the model includes a fictitious DAC and a fictitious
timing loop is in equilibrium.
analog filter, the interpolants in (2) can be computed entirely
digitally from knowledge of 1) the input sequence {z(m)},
2) the impulse response h l ( t )of the interpolating filter, and 3 )
the time instants mT, and kTi of the input and output samples. Fig. 5 presents the timing loop of Fig. 2 with expanded detail
These digitally computed interpolants have identically the for the controller. The interpolator performs the computations
same values as if the analog operations had been performed. of (6). The controller provides the interpolator with infor-
A more useful format is obtained by rearranging the index- mation needed to perform the computations. Other essential
ing in (2). Recognizing that m is a signal index, define a filter elements in the loop will not be treated here.
index An interpolant is computed from (6) using I adjacent
samples z ( m ) of the signal and I samples of the impulse
z = int[kTi/T,] - m ( 3 ) response h I ( t ) of the interpolating filter. The correct set of
where int[z] means largest integer not exceeding z. Also, signal samples is identified by the basepoint index mk and
define a basepoint index the correct set of filter samples is identified by the fractional
interval p k . Thus, the controller of Fig. 5 is responsible for
determining mk and pk, and making that information available
to the interpolator.
and a fractional interval
Once mk and p k have been identified by the controller, then
other elements load the selected signal and impulse-response

, .................



Fig. 5. Timing processor.

samples into the interpolation filter structure for computations.

-1 0

.................... rl(mkil)-r

These loading operations are regarded as part of the filter

implementation; some options are examined in Part I1 [ 13. (m kt I

The necessary control can be provided by a number- Fig. 6. NCO relations.

controlled oscillator (NCO). Assume that the signal samples
are uniformly clocked through a shift register at rate l / T s-and
That is to say, W ( m )is the synchronizer’s estimate of the
that the NCO is clocked at a rate synchronized to l / T s .
average frequency of interpolation l / T , , expressed relative
Provided that the interpolator is never called upon to per-
to the sampling frequency l / T s . The control word is an
form upsampling’ then the NCo ‘lock period can be Ts‘ If estimate because it is produced from filtering of multiple, noisy
upsampling is ever required, then a higher NCO clock rate is
measurements of timing error.
needed. Further discussion will concentrate on NCO clocking
To see how / L k can be extracted from the NCO, refer to Fig.
at rate l/Ts (downsampling only); modifications needed to
6, which is a plot of (fictitious) time-continuous q ( t ) versus
accommodate upsampling are readily devised once the basic
continuous time. In the figure, mkTs is the time of the sample-
principles are established.
clock pulse immediately preceding the kth interpolation time
The NCO is operated so that its average period is T,.
Recycling of the NCO register indicates that a new interpolant
ICT, = ( m k pk)TS.NCO register contents decrease through
zero at t = ICT,, and the zero crossing (underflow) becomes
is to be computed, using the signal samples currently residing
in the interpolator’s shift register. Thus, basepoint index is
known at the next clock tick at time ( m k +
l)Ts. Register
identified by flagging the correct set of signal samples, rather
contents q ( m k ) and q ( m k 1) are available at the clock ticks.
From similar triangles in Fig. 6, it can be seen that
than explicitly computing mk.
- (1 -Pk)Ts
A. Extraction of /lk ‘V(mk) 1 - V(m,k f 1 )
Fractional interval lLk can be calculated from the contents of which can be solved for pk as
the NCO’s register upon recycling, as will now be explained.
V(mk) -
Designate the NCO register contents computed at the mth
clock tick as q(m),and the NCO control word as W ( m ) Then .
/Lk =
1 - q(mk + 1 ) + q(mk) - w(mk)
the NCO difference equation is An estimate for P k can be obtained by performing the indi-
cated division of the two numbers q ( m k ) and W ( m k )that are
~ ( m=) [q(m- 1) - W ( m- l)]mod-1. (7) both available from the NCO. [Equation (9) is an estimate of
(A decrementing NCO is employed because it affords a the exact /Lk because its constituents W(mk) and q ( m k ) are
minor simplification in computation of / L k as compared to an ’0th estimates of the true frequency and phase.]
incrementing NCO.) To avoid division, recognize that l / W ( m )2~ T,/Ts;nom-
control word ~ ( [a positive
~ 1 is adjusted by the inal value of this ratio is designated to. Although the exact
timing-recovery loop so that output of the data filter is strobed / T ~is unknown and the (09 ex-

at near-optimal timing. Under loop equilibrium conditions, pressed to finite precision, can often be an excellent approxi-
~ ( will ~be nearly 1 constant. Contents of the NCO register mation to the true value. Therefore, the fractional interval can
be approximated by
(also a positive fraction) will be decremented by an amount
W ( m )each Ts seconds and the register will underflow each Pk E EOQ(mk). (10)
l / W ( m ) clock ticks, on average. Thus, the NCO period is
T, = T s / W ( m )and so
Represent the deviation in Eo from the true ratio of periods
TS as A[. This deviation causes a uniformly distributed error with
W(m,)E -. (8)
T, standard deviation A [ / ( [ o O ) in the calculated value of p k .

If the deviation of [ O is too large, then a first order correction If the filter has finite impulse response (FIR), then I filter
coefficients and I signal samples must be delivered to the filter
structure for each interpolation.
If an infinite impulse response (IIR) filter were employed,
reduces the standard deviation in pk to at2/([;fi), again a recursive structure would be required so that the computing
without requiring a division. effort could be finite. Let the filter have p poles and z zeros.
Timing errors arising from multiplying by the nominal <O Then for each interpolation it would be necessary to load the
using (10) instead of dividing by the exact W ( m )using (9) following information to be able to compute the interpolant:
cannot accumulate; the feedback loop removes any constant -p + +z 1 filter coefficients, as specified by p k .
error or trend. -z + 1 signal samples.
-p past outputs of the filter, calculated with the present
value of p k .
B. Interpolation Jitter
But those past outputs cannot be known for the present
Although the kth interpolation is computed for a time kTi = ,uk unless they were computed for all possible values of p k
(mk + pk)Ts, the interpolant is actually delivered coincident at every interpolation instant. That would ordinarily be an
with a clock tick no earlier than (mk l)Ts.Therefore, the unacceptable computing burden and so FIR filters are usually
output exhibits a timing jitter with peak-to-peak fluctuations preferred.
of T,,even if the sampling clock and received symbol rate Other reasons for selecting FIR filters have been given in
are entirely jitter free.
Timing jitter may be inconsequential if the received data
are consumed at the receiver location. A timing clock is B. Ideal Interpolation
provided by underflows of the NCO. Underflow marks give an
indication of correct data clocking to any downstream devices It is well known [ 3 ] that the bandlimited input signal x ( t )
because they have the same jitter as the data. (or its samples {x(kTi)}at times t = kTi) could be recovered
But often the data must be retransmitted over a synchronous from the samples {z(mT,)}by using the ideal filter with
link to a remote consumer. The underflow marks usually impulse response
cannot be retransmitted along with the data. Unless jitter is sinrt/T8
removed before retransmission, the jitter will be transmitted h i ( t )= ~

on the data stream.
Auer [ l o ] has pointed out that a near jitter-free clock can and transfer function
be retrieved from the NCO and used to reclock the data before
retransmission. His scheme employs the contents ~ ( mof)
the NCO register in a direct digital frequency synthesizer.
At each NCO clock tick, the register contents are used to The ideal filter is IIR and noncausal; it cannot be realized
address a table of sines to produce a sample sin 2 7 4 m ) . and so perfect recovery of x ( t ) is not possible with any
These sine samples are applied to a D/A convertor and then practical filter. Failure of a realizable filter to reconstruct x ( t )
filtered to yield an analog, low-jitter sinewave with frequency would be charged as distortion in conventional applications of
1/Ti = K / T , from which a symbol-rate clock can be derived. interpolation.
But perfect recovery is not required from an interpolator
C. Alternative Control Methods in a modem. It is only necessary that the filtered strobe
outputs of the modem have the correct values-a much less
An NCO is not the only possible control structure. An stringent requirement than perfect reconstruction of x ( t ) . An
alternative, suggested by M. Moeneclaey, is described in interpolating filter in a modem need not be nearly so precise
Appendix A. as some of the optimized interpolators found in the DSP
literature, such as in [ 3 ] .
Practical demands on the interpolation filter can be explored
by considering its frequency response Hl(f).
What properties are desirable in the interpolating filter’s
impulse response h i ( t ) or equivalently, via Fourier transfor- C. Stopband Response
mation, in its transfer function H i ( f ) ? Take heed that the
properties sought are those of the fictitious analog filter, despite The spectrum of the signal samples has periodic images,
the fact that all physical operations are performed digitally. spaced at a frequency interval l / T , . See Fig. 7. An inter-
polation filter is required to suppress those images prior to
resampling. Any image energy that is not suppressed will be
A. Duration of Impulse Response aliased by resampling and, if the sampling and symbol rates
In general, new filter coefficients [samples of h i ( t ) ]must are incommensurate, will constitute random interference to the
be reloaded or recomputed for each interpolation. The frac- output sequence { y( I C ) } .
tional interval pk-which specifies the filter-coefficient sample An ideal interpolation filter completely suppresses all input
values-never repeats if Ti and T, are incommensurate. frequency components above 1/2T, and the same stopband

without penalty by other linear filters in the system.

This relaxation in the passband means that interpolating
filters for use in modems can have much less stringent re-
quirements than would be imposed upon interpolation filters
that attempted to recover the orginal time function ~ ( t )The
0) Spectrum of x(t)
passband filtering allowable in a modem interpolator is not
counted as distortion.

If sampling in a digital modem is not synchronized with
b) Spectrum of x(mT,)
the data symbols, timing must be adjusted by interpolating
new samples among the original ones. “Interpolation” is
really a more-involved process that combines interpolation and
subsequent decimation by resampling.
A useful conceptual model includes a digital-to-analog
convertor, an analog, time-continuous interpolating filter, and
c) Spectrum of y(t)
a resampler, all fictitious, to produce the desired interpolants.
Exactly the same interpolants can be computed entirely dig-
itally from the input samples and knowledge of the sampled
impulse response of the fictitious analog filter. Equation (6)
underlies interpolation operations in digital modems.
An individual interpolant is specified by the signal samples
d) Folded Spectrum of y(kTi) (the basepoint set) that contribute to its value, and the filter
Fig. 7. Signal spectra. samples used for the computation. The basepoint set is identi-
fied by a basepoint index, and the filter samples are identified
behavior is desirable in a practical interpolation filter. Of by the fractional interval. These two pieces of information
course, no realizable filter can provide infinite attenuation over must be delivered to the digital interpolating structure by a
an entire stopband. Therefore, any practical filter will introduce controller. A number-controlled oscillator (NCO) can provide
some penalty because of incomplete suppression of images. these parameters via control algorithms presented in the text.
Fig. 7 illustrates spectra of various signals in the modem. Because the NCO is clocked synchronously with the signal
The top line of the figure shows the bandlimited spectrum samples, the modem output will exhibit timing jitter. This
of the input signal ~ ( t )Sampling
. generates periodic spectral jitter is inconsequential if the data are consumed locally to
images, as in the second line. Absence of aliasing is indicated the modem, because the NCO can provide a symbol clock
by the non-overlap of the images. with the same jitter as the data.
The time-continuous interpolating filter attenuates the im- If the data must be retransmitted synchronously, the jitter
ages in varying degree, so that the spectrum of y(t)-the may be intolerable. A jitter-free analog clock can be recovered
third line-consists of a main lobe around zero frequency, plus from the NCO and used to reclock the jittered data prior to
partially suppressed images at all integer multiples of l / T s . retransmission.
Upon resampling at rate l/Ti, all residual images fold in The fictitious analog interpolating filter should be FIR and
onto the desired signal. Fig. 7(d) sketches that part of the should provide good stopband suppression of the periodic
spectrum (not to scale) lying in the vicinity of zero frequency. images of the sampled input signal. Passband response of
The actual spectrum repeats with a period of l/Ti. If T,/T, is this filter is part of the overall filtering of the modem. In
irrational, the folded images are uncorrelated with the desired consequence, non-flat response in the passband is not charged
signal and will impair recovery of the data. Relative power as distortion, as it would be in a classical interpolator. A
in the folded images, or equivalently, image attenuation by designer has wide latitude in distributing overall filter response
H r ( f ) ,is a measure of the adequacy of the stopband response between the interpolating filter and other filters in the modem.
of the filter.
D. Passband Resvonse M. Moeneclaey has pointed out an alternative control
scheme that does not use an NCO. Two successive
A n ideal interpolator would pass all frequencies from 0
interpolations are performed for time instants
to 1/2T, with flat attenuation and with linear phase. In a
modem where signal filtering is to be performed anihow, there
is no need for flat transmission in the filter’s passband. The
kT, = VI,^: + pk)T,
interpolator merely contributes a portion of the filtering that
is required for the receiver. Any reasonable passband char-
acteristic is permissible, provided that it can be compensated (k + l ) T z= +
( ~ + + ipk+i)Ts.

Subtracting these two expressions and rearranging slightly REFERENCES

gives the recursion
L. Erup, F. M. Gardner, and R. A. Harris, “Interpolation in digital
modems-Part 11: implementation and performance,” to be published.
mk+l = mk Ti/Ts pk - /Lk+l. (-4.2) R. E. Crochiere and L. R. Rabiner, Multirate Digital Signal Processing.
Englewood Cliffs, NJ: Prentice-Hall, 1983.
By definition, mk+l is an integer. Then, since 0 5 p k + l < 1, R. W. Schafer and L. R. Rabiner, “A digital signal processing approach
to interpolation,” Proc. IEEE, vol. 61, pp. 692-702, June 1973.
mk+l + pk+l = mk 4-Ti/Ts + pk < mk+2 (-4.3) R. E. Crochiere, L. R. Rabiner, and R. R. Shively, “A novel imple-
mentation of digital phase shifters,” Bell Syst. Tech. J., vol. 54, pp.
whence the increment in sample count from one interpolation 1497-1502, Oct. 1975.
F. Takahata et al., “A PSK group modem for satellite communication,”
to the next is IEEE J. Select. Areas Commun., vol. SAC-5, pp. 648-661, May 1987.
M. Oerder, G. Ascheid, R. Haeb, and H. Meyr, “An all digital implemen-
tation of a receiver for bandwidth efficient communication,” in Signal
Processing III (Eusipco 1986), I. T. Young et al. Ed., pp. 1091-1094,
Elsevier, 1986.
Notice that a practical scheme must work with the increment G. Ascheid, M.Oerder, J. Stahl, and H. Meyr, “An all digital receiver
architecture for bandwidth efficient transmission at high data rates,”
rather than the sample count mk. Any finite-length counter of IEEE Trans. Commun., vol. 37, pp. 804-813, Aug. 1989.
mk would overflow eventually. J. A. C. Bingham, The Theory and Practice of Modem Design. New
To compute the fractional interval PIE, recognize that the York: Wiley, 1988.
F. M. Gardner, “A BPSWQPSK timing-error detector for sampled
fractional part fp[ ] of the increment is zero receivers,” IEEE Trans. Commun., vol. COM-34, pp. 423-429, May
E. Auer, “An advanced, variable data rate modem for Intelsat IDR/IBS
services,” Paper 1-3, Proc. 2nd In?. Workshop Digital Signal Processing
from which one may conclude TechniquesAppl. Space Commun., Turin, Italy, 24-25 Sept. 1990.

The true Ti/T, is not available. Instead, the synchronizer

produces a control word V ( m k ) N TiIT, to be used in
the recursions (A.4) and (AS). This control word is the
synchronizer’s estimate of the true interpolation period Ti
relative to the sampling period T,.
The alternative control method may be most useful in
systems where the data are consumed at the same location Floyd M.Gardner (S’49-A’54-SM’58-F’80) re-
ceived the B.S.E.E. degree from the Illinois Institute
as the data receiver, without reclocking. It is not immediately of Technology, Chicago, IL, in 1950, the M.S.E.E.
apparent how a jitter-free, time-continuous clock for retrans- from Stanford University, Stanford, CA, in 1951,
mission could be synthesized easily without the phase v(m) and the Ph.D. degree from the University of Illinois,
Urbana, IL, 1953.
that accumulates in an NCO. He has been a independent consulting engineer
since 1960, active in the fields of communications
and electronics. He is a specialist in synchronization
ACKNOWLEDGMENT and in phase-lock loops, and is the author of the
I wish to thank Dr. R. Harris and L. Erup of the European book Phaselock Techniques (New York: Wiley, 2nd
edition, 1979). In recent years he has been investigating algorithms for
Space Agency for their helpful critiques of the work as it digitally implemented modems.
progressed. Dr. Gardner is a Registered Professional Engineer in the State of California.