2014 7th Advanced Satellite Multimedia Systems Conference and the 13th Signal Processing for Space Communications
Workshop (ASMS/SPSC)
A Novel Approach for Symbol Timing Estimation
Based on the Extended Zero-Crossing Property
Wilfried Gappmair1, Stefano Cioni2, Giovanni E. Corazza3, Otto Koudelka1
1
Institute of Communication Networks and Satellite Communications, Graz University of Technology, Austria
2
European Space Agency, ESA/ESTEC Noordwijk, The Netherlands
3
Department of Electronics, Computer Science and Systems, University of Bologna, Italy
AbstractCarrier- and data-blind recovery of the sym-
bol timing is of paramount importance in digital receivers,
i.e., detailed knowledge about carrier frequency and phase
or any pilot sequences is not necessary for proper operation.
In this context, feedforward algorithms are particularly
useful in packet-oriented systems, where rapid and stable
acquisition of the major transmission parameters is essen-
tial for subsequent processing stages. In the current paper,
we propose a novel approach for blind estimation of the
symbol timing, which needs just one or two samples per
symbol. The new method is characterized by introducing a
second filter in parallel to the receiver matched filter. Un-
der Nyquist conditions, this filter exhibits an impulse re-
sponse satisfying the extended zero-crossing property, i.e., it
vanishes for all integer multiples of the symbol period,
including the origin! Using this idea for a suitably designed
timing estimator, it is shown that the annoying jitter floor
typical for most timing estimators and caused by pattern
noise can be avoided.
KeywordsBlind feedforward recovery, symbol timing,
extended zero-crossing property, low-rate sampling.
I. INTRODUCTION
Symbol timing recovery (STR) is an indispensable
task in digital receivers [1], [2]. One of the major reasons
is the fact that many synchronization algorithms for carri-
er frequency and phase require the symbol timing to be
established in advance. This means that STR methods
should be carrier-blind, i.e., knowledge of carrier infor-
mation is not necessary to operate the related STR algo-
rithm properly. Furthermore, it is to be noticed that, when
carrier details are not known to the receiver, the underly-
ing data are usually not available or their detection is not
reliable enough. Therefore, it is most welcome, when the
envisaged STR technique follows a carrier-blind and non-
data-aided (NDA) approach.
Feedforward (FF) algorithms for STR are an attractive
option in packet-oriented systems, where rapid and stable
acquisition is of paramount importance [3]. The Oerder
and Meyr (O&M) estimator [4], normally implemented
with four samples per symbol, represents perhaps the most
prominent example in this respect. Mainly motivated by
the computational load of O&M schemes or modifications
thereof [5], an FF solution with only two samples per
symbol has been suggested by Lee in [6], whose bias can
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be circumvented via a simple compensation coefficient
derived in [7]. Guided by the idea of a prefilter to avoid
the annoying jitter floor, for the first time addressed by
Franks and Bubrouski [8] and in the sequel applied to
Gardner synchronizers [9], [10], it could be shown in [11]
that this approach works for FF concepts as well.
With respect to the receiver matched filter (MF), pre-
filters are arranged in series, which might cause problems
in several scenarios due to the additional delay. In the
current contribution, however, we follow a completely
different approach by introducing a second filter operated
in parallel to the MF so that the delay problem is avoided.
In case the noise power is known to the receiver, we end
up with an STR estimator, which needs just one sample
per symbol taken at the MF output and the output of the
parallel filter. On the other hand, if the noise power is not
known, which is perhaps more relevant from the practical
point of view, the algorithm requires twice the sampling
rate at the MF output.
Moreover, the parallel filter is to be designed such that
the jitter floor disappears. For real-valued Nyquist pulses
used for baseband shaping, we will show that this goal is
achieved, if the filter satisfies the extended zero-crossing
(XZC) property, i.e., the impulse response vanishes for all
integer multiples of the symbol period, including the
origin!
The remainder of the paper is organized as follows. In
Section II, we introduce the signal model used for analyti-
cal and simulation work, including the design of the XZC
filter. The novel STR approach is developed in Section
III, with emphasis on the computation of the mean estima-
tor value and the jitter variance. Numerical results are
given in Section IV and, finally, conclusions are drawn in
Section V.
II. SIGNAL MODEL
Let the independent and identically distributed (i.i.d.)
symbols ci = ai + jbi be zero-mean and normalized to unit
variance such that EC[|ci|2] = 1, where EC[] denotes expec-
tation with respect to the M-ary symbol alphabet C. The
unit-energy baseband pulse h(t) is assumed to satisfy the
Nyquist condition; guided by practical reasons, we will
59
 2014 7th Advanced Satellite Multimedia Systems Conference and the 13th Signal Processing for Space Communications
Workshop (ASMS/SPSC)

concentrate on a root-raised cosine shape with roll-off Because H1(f) is proportional to the first-order deriva-
factor , where 0 1. Furthermore, it is assumed that tive of H(f), the additional filter needed for the new STR
the receiver signal r(t) is rotated by the carrier phase approach will be denoted as derivative matched filter
[, ) and shifted in time by [T/2, T/2). Therefore, (DMF); for = 0.25, the spectral evolution of H(f) and
with T as the symbol period, we have H1(f) is illustrated in Fig. 2. Observe also that the DMF
output is furnished by
r (t ) = e j ci h(t iT ) + w(t ) . (1)
i
x 1 (t ) = r (t ) h1 (t ) = e j s1 (t ) + n1 (t ) , (6)
In this context, w(t) represents a zero-mean white
with n1(t) = w(t) h1(t) and
Gaussian noise process with independent real and imagi-
nary parts, each of variance w2 = 1/(2s), where s = s1 (t ) = ci g1 (t iT ) . (7)
Es/N0 is the mean signal-to-noise ratio (SNR) per symbol. i

In the sequel, r(t) passes the receiver matched filter h*(t) In this context, it is to admit that the DMF concept has
as shown in Fig. 1. For root-raised cosines, it is to be been suggested in the open literature for a completely
observed that h*(t) = h(t) so that the corresponding out- different scenario, namely for NDA recovery of the carrier
put signal can be written as frequency via a closed-loop design [1].
x 0 (t ) = r (t ) h(t ) = e j s0 (t ) + n0 (t ) , (2) Nevertheless, by detailed inspection of Fig. 2, it is
clear that the sharp edges of H1(f) at f = 12+T involve a
where denotes convolution and n0(t) = w(t) h(t) is a
fairly slow roll-off in the time domain so that the related
zero-mean non-white Gaussian process with variance
FIR filter must be implemented with a correspondingly
2 w2 . Finally, by defining the raised cosine (RC) function
large number of taps. In order to reduce the computational
as g(t) = h(t) h(t), the signal part is given by
complexity, it is suggested that H1(f) is shaped by an addi-
s0 (t ) = ci g (t iT ) . (3) tional filter function P(f), i.e., H 1 ( f ) = H1(f) P(f); the
i overbar indicates that signals at the DMF output are
Matched
x0(t) shaped by P(f). Of course, P(f) has the same spectral sup-
filter
Symbol timing
port as H1(f) and, perhaps more important, it must not
r(t) yk
Derivative
recovery violate the XZC property introduced previously. This is
MF
x1(t) given, when it satisfies

Fig. 1. Dual-filter framework for symbol timing recovery. 1 1

P f + = P f , | f | < , (8)
As already mentioned in the introductory section, we 2T 2T 2T
will introduce a second filter operated in parallel to the as could be shown for a similar problem investigated in
receiver MF. We propose a filter design so that the convo- [13]. Among the shapes fulfilling the condition in (8), we
lution of the impulse response of the filter, henceforth will focus for pragmatic reasons on an RC-based profile,
denoted by h1(t), and the signal part of (1) is zero at all i.e., P ( f 21T ) = cos 2 |fT | , where | f | < 2T and 0 < .
sampling instants t = kT, i.e., h(t) h1(t)|t = kT = 0, k Z. Fig. 2 displays the spectrum of H 1 ( f ) , when forming the
Excluding k = 0, this condition would be satisfied by g(t). DMF with = 0.25.
However, if we consider g1(t) = tg(t) instead of g(t), then
the XZC property is established. This means that h1(t) is
determined by the integral equation h(t) h1(t) = g1(t), 1.5

which might be solved most elegantly by shifting the 1

problem to the frequency domain. Doing this, we have 0.5

that
0

1 G ( f )
H ( f ) H1 ( f ) = , (4) -0.5 __
H (fT )
j 2 f -.- H1 (fT )
-1 H1 (fT )

where H(f), H1(f), and G(f) are the Fourier transforms of -0.75 -0.5 -0.25 0 0.25 0.5 0.75 1

h(t), h1(t), and g(t), respectively [12]. Then, with G(f) = fT

H2(f), we simply obtain
Fig. 2. Spectral evolution for MF, DMF, and RC-shaped DMF ( = =
1 H ( f ) 0.25).
H1 ( f ) = 2t h(t ) = h1 (t ) . (5)
j f

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Workshop (ASMS/SPSC)

III. DEVELOPMENT AND ANALYSIS OF THE Putting all pieces together, we finally arrive at
NOVEL ESTIMATOR SCHEME
U 0 2 w2 0
Considering MF as well as DMF signals described in U0 = = cos(2 ) , (15)
20
(2) and (6), the corresponding T-spaced samples are im-
mediately obtained as which establishes a relationship between observables x0,k,
x0, k = x0 ( kT ) = e s0, k + n0, k ,
j
(9) processed by (11), and the (normalized) timing offset .
Unfortunately, the noise power 2 w2 has to be known in
x1, k = x1 ( kT ) = e j s1, k + n1, k . (10) advance to obtain (15). If this is not realistic, we could
resort to E[|x0,k|2] E[|x0, k1/2|2], which gives after some
In this context, the noise parts n0,k and n1,k are zero- algebra
mean non-white Gaussian variates, whereas the signal
parts s0,k and s1,k are simply provided by s0(kT) and s1(kT). U 0 = E[| x0,k |2 ] E[| x0,k 1 2 |2 ]
Both sorts of samples, x0,k and x1,k, will be processed ap- = E[| s0,k |2 ] E[ | s0,k 1 2 |2 ] (16)
propriately in the FF algorithm providing a timing esti- = 40 cos(2 ) .
mate , the interpolator is fed with to generate the cor-
rected samples yk, as it is sketched in Fig. 3. Hence, by means of an additional normalization step,
we have that
x0(t)
MF Interpolator yk U 0
U0 = = cos(2 ) . (17)
40
FF estimator
DMF
x1(t) Note that for processing U 0 solely one sample per
symbol is needed, whereas for U 0 two of them are re-
Fig. 3. Feedforward scheme for symbol timing recovery. quired. Note also that (15) and (17) are functions of the
(normalized) timing error, which could be used for a re-
A. Analysis of MF and DMF Output covery method by inverting the relationship as follows:
In order to develop the new STR estimator, we first = 21 arccos(U0). However, since this approach does not
compute E[|x0,k|2], which denotes expectation of |x0,k|2 with provide unambiguous estimates over the full range of
respect to data and noise. By taking into account that errors [1/2, 1/2), it is not really helpful in practice.
E[|n0,k|2] = 2 w2 and that n0,k is statistically independent As a consequence, we will take the DMF output into ac-
from s0,k, we obtain count as well.
U 0 = E[| x0,k |2 ] = E[| s0,k |2 ] + 2 w2 . (11) To this end, we analyze the expected value of
Re[ x0*, k x1, k ] . Recalling that the signal and noise contribu-
Evaluating E[|s0,k|2], it is to be recalled that the unit- tions in x0,k and x1,k are statistically independent, this
variance symbols ci C are i.i.d. so that, by applying the yields
auxiliary result (33) achieved in the Appendix,
U 1 = E[Re{x0*,k x1,k }]
E[| s0,k |2 ] = g 2 [( k )T ] (18)
k = Re{E[ s0*,k s1,k ]} + Re{E[n0*,k n1,k ]} .
(12)
1 m


=
T G( f ) G T
m
f e j 2m df ,

Then, by introducing A() = Re{E[ s0*,k s1,k ]} , the first
term in (18) is furnished as
where we used for convenient reasons the normalized
timing offset = T . Since G(f) is bandlimited by 12+T , A( ) = Re g [( k )T ] g1[( k )T ]
k
the relationship boils down to (19)
1
m j 2 m
E[| s0,k | ] = 0 + 20 cos(2 ) ,
2
(13) = Re
T
G1 ( f ) m G T f e df ,
with by tacitly assuming that the data symbols ci in s0,k and s1,k
(1+ ) 2 T are i.i.d. and that the DMF output is shaped by P(f).
1
0 = G ( f ) df = 1 , Again, the second line in (19) is due to (33) derived in the
2

T (1+ ) 2 T 4 Appendix. Regarding that G(f) is bandlimited and that

(14)
1
(1+ ) 2 T
1 G1 ( f ) = G1 ( f ) P( f ) = j 21 G(f f ) P( f ) , the equation
0 =
T
(1 ) 2 T
G ( f ) G f df = .
T 8 reduces to

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sin(2 )
(1+ ) 2 T
G( f ) 1 For comparison purposes, we furnish also the jitter
T 2 (1) 2T f
A( ) = P( f ) G f df . (20) variance for the maximum-likelihood (ML) NDA algo-
T
rithm [16], which is bounded by
In order to complete the analysis of (18), we must also
1
have a look at the contribution introduced by the noise ML
2
= . (27)
components of x0,k and x1,k, i.e., we have to compute 2 L s
Bn = E[n0*,k n1,k ] . Shifting the problem to the frequency Because of the mathematical intricacies, we skip the
domain, we directly obtain detailed analysis of the jitter variance in general form.
Instead, the focus is on the special, but important case
Bn = 2 w2 H ( f ) H 1 ( f ) df . (21) with negligible timing errors.

By examination of (15) and (17), it is obvious that

Because H(f) and H 1 ( f ) are counter-symmetric, as
can be seen by inspection of Fig. 2, it is clear that the
integral in (21) is zero so that Bn does not contribute to
(18). Hence, the normalized value of (18) develops as
2 2
U 1 = U 1 = A( ) = sin(2 ) ,
1 1
where 1 = A( ) evaluated at = 0. With P ( f 21T ) =
U0 1, when 0, so that we may assume that U 0 is
approximated by this value as well. In the next step, by
finding out the contribution of U 1 , we have to analyze the
term Re[ x0*, k x1, k ] as can be seen from (25). With 0, it
is clear that x0, k e j ck + n0,k and x1,k n1,k , which
(22) gives vk = Re[ x0*,k x1,k ]|0 Re[ e j ck* n1,k ] in the low-
noise range. In consequence, if s >> 1, we have that
U 1 << U 0 such = 21 arg(U 0 + jU 1 ) 21 U 1 = 1 U 1 . 1

cos 2 |fT | , | f | < 2T and 0 < , the coefficient 1 is Putting all pieces together, this yields
given by
E[vk2 ] 12 E[| n1,k | ]
2

02 = 2 = = , (28)
A( ) 2 sin
2
2 0
12 L 12 L
1 = = 2
= . (23)
=0 4 2 2 3
=
where
Inspecting in detail the relationships (15), or (17), and (1+ ) 2 T
3
(22), we are now able to formulate the STR algorithm as E[| n1,k |2 ] = 4 w2 | H ( f ) P( f ) | df = . (29)
2

32 2 s
1
(1 ) 2 T
1
= arg(U 0 + jU 1 ) . (24) In order to obtain , which specifies the bandwidth of
2
P(f), we require that 02 is equal to ML
2
. To this end, the
Based on an observation length of L symbols at MF ratio (, ) = 0 ML is introduced and for {0.05,
2 2

and DMF output, U 0 and U 1 are obtained by approximat- 0.25} plotted in Fig. 4 as a function of , where 0 < .
ing U 0 or U 0 in (15) or (17), respectively, and U 1 in It is immediately verified that 02 approaches ML 2
, when
(22) with . This will be used in the next section, where nu-
1 L 1 merical results are presented in terms of the mean estima-
U 0 = | x0,k |2 , tor value and the jitter variance.
L k =0
1 L 1
U 0 = (| x0,k |2 | x0,k 1 2 |2 ) , (25)
5

L k =0 -.- = 0.25
4
1 L 1 - - = 0.05

U 1 = Re[ x0*,k x1,k ] .

L k =0 3
(, )

B. Jitter Variance 2

Powerful estimators exhibit no bias, i.e., = E[ ] , 1

and the jitter variance 2 = E[( ) 2 ] is close to the
Cramer-Rao lower bound (CRLB) as the theoretical limit
0.05 0.1 0.15 0.2 0.25
[14]. For computational simplicity, the latter is often em-
ployed in its modified form [15], which is for an observa-
tion length of L symbols and RC shapes with excess Fig. 4. Evolution of (, ) = 02 ML
2
.
bandwidth (roll-off factor) given by
IV. NUMERICAL RESULTS
1
MCRLB = . (26) Due to the related intricacy, we do not provide full
[ 3 (1 + 3 ) 8 2 ] 2 L s
1 2 2

analysis of the timing estimator in terms of bias and jitter

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Workshop (ASMS/SPSC)

variance. Instead, we will resort in the following to nu-

merical means verifying this way the major figures of XZC
O&M
merit.

0.01 mLee

Norm. jitter variance

We start with the mean error value (MEV), i.e., E[ ] , 0.001

which is approximated by averaging the estimates in (24)

over a number of simulation runs large enough to avoid
0.0001

any side effects. For 4-PSK, L = 100, and = = 0.25, 0.00001

Fig. 5 shows the results parameterized by different SNR

__
MCRLB
- - ML-NDA

values; the impulse response for MF as well as DMF 1. 106

5 10 15 20 25 30

spans a window of 21 symbol periods. For better visibil- Es/N0 [dB]

ity, simulation results are indicated by dots in different
Fig. 6. Evolution of the normalized jitter variance for 4-PSK (L = 100,
style, which have been interpolated by a cubic spline. As
= = 0.25).
can be seen, the estimates deviate from the ideal curve the
more the (normalized) timing offset 1/2; needless to
say that the phenomenon is more pronounced at lower XZC
O&M
SNRs and smaller values of L. The same behavior could

0.01 mLee

be observed with modulation schemes other than 4-PSK,

Norm. jitter variance

0.001
including multi-level signals like 16-QAM or 16-APSK,
and other values of and , although not shown because 0.0001
of limited space.
0.00001
With the same parameter setting used in Fig. 5, the jit-
__
MCRLB
- - ML-NDA
ter variance is illustrated in Fig. 6. As already mentioned 1. 106
5 10 15 20 25 30
previously, the jitter floor vanishes for the XZC estimator.
Es/N0 [dB]
Note that, for = , the XZC performance is close to the
ML-NDA bound, according to the result achieved previ- Fig. 7. Evolution of the normalized jitter variance for 16-QAM (L =
ously (see Fig. 4). For comparison purposes, the evolution 100, = = 0.25).
of the modified CRLB is plotted as well, indicating a gap Furthermore, the XZC estimator has been tested under
between ML-NDA bound and the theoretical limit, which very extreme SNR and roll-off conditions as they are
might be narrowed only by an increase of . Also for discussed by the channel model group for the new DVB-S
comparison purposes, the diagram includes the normal- standard [17], i.e., SNR values as low as -10 dB and =
ized jitter performance of O&M and modified Lee (mLee) 0.05. Due to the small roll-off factor, both MF and DMF
algorithms, both exhibiting a non-negligible jitter floor. are designed now for an impulse response spanning a
window of 81 symbol periods. Using 4-PSK signals, the
0.4
jitter performance is depicted in Fig. 8. An observation
window with L = 104 had to be chosen to achieve useful
Es /N0 = 3 dB
Es /N0 = 9 dB

0.2 results in the very low SNR range. Again, no jitter floor is
observed and the XZC performance is close to that of the
ML-NDA bound. Although not shown in the diagram to
MEV

avoid an overload of the plot, the same behavior has been

-0.2
verified for QAM and APSK signals as well.
-0.4

-0.4 -0.2 0 0.2 0.4 XZC

0.01 O&M
= /T mLee
Norm. jitter variance

0.001
Fig. 5. Evolution of the mean error value for 4-PSK (L = 100, = =
0.25). 0.0001

Using the same parameters as in Fig. 6, simulation re- 0.00001

sults for 16-QAM are visualized in Fig. 7. No significant

1. 106 __

discrepancy is detected with respect to the 4-PSK results;

MCRLB
- - ML-NDA

the only difference is that the jitter floor for O&M turns -5 0 5 10 15 20 25 30

out be somewhat increased. Es/N0 [dB]

Fig. 8. Evolution of the normalized jitter variance for 4-PSK (L = 104,

= = 0.05).

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Workshop (ASMS/SPSC)
V. CONCLUSIONS
A new approach for carrier-blind NDA recovery of the
symbol timing has been introduced in the current paper.
The developed feedforward estimator needs just one or
two samples per symbol. However, it requires the opera-
tion of a second filter in parallel to the receiver matched
filter (MF). Assuming real-valued Nyquist pulses for
baseband shaping, the filter design must satisfy the ex-
tended zero-crossing (XZC) property. It could be shown
that this corresponds to the first-order derivative of the
receiver MF in the frequency domain; sharp edges causing
a slow roll-off in the time domain might be smoothed by a
further shaping function not violating the XZC character-
istic.
Since the second filter is operated in parallel to the
MF, no additional delay needs to be taken into account.
This is in contrast to conventional prefilter structures,
arranged in series to the MF and frequently suggested in
the open literature to avoid the annoying jitter floor.
Furthermore, just to keep the analytical part tractable,
it has been decided to focus on a signal model, which does
not include phase noise effects and impairments caused by
residual frequency offsets. Preliminary simulation results
have shown that this would have an impact insofar as the
jitter floor does not disappear, depending on the standard
deviation of the phase noise and/or the percentage of the
residual frequency error with regard to the symbol rate.
However, detailed investigations are out of scope for this
paper and will be deferred to future activities. The non-
vanishing floor is easily explained by the violation of
Nyquist and XZC conditions; the same happens, if the
root-raised cosine shape is distorted by a nonlinear chan-
nel, e.g., a high-power amplifier operated close to the
saturation point.
Some words are also due in view of the computational
complexity. To this end, we assume that the timing esti-
mate in (24) is evaluated by replacing U 0 and U 1 with
U 0 and U 1 in (25). It is obvious that we need six addi-
tions and multiplications per symbol, all of them real-
valued. On the other hand, for O&M as well as mLee
algorithms, eight additions and multiplications per symbol
are necessary. For all algorithms, the argument of a com-
plex number might be extracted via a simple look-up
table. Finally, comparing the complexity of the involved
filter structures, no discrepancy is observed between XZC
and mLee algorithm, if we accept that parallel and prefil-
ter are both implemented with the same number of taps.
With O&M estimators using four samples per symbol,
however, we would need twice the number of taps for MF
and prefilter.
978-1-4799-5893-1/14/$31.00 2014 IEEE
APPENDIX
In the sequel, an alternative solution of the infinite
sum
q( ) = x ( kT ) y ( kT ) (30)
k
is derived, which turns out to be most useful in case that
the spectra of x(t) and y(t) are bandlimited.
In the first step, by introducing the definition zk() =
x(kT ) y(kT ), the corresponding Fourier transform is
obtained as [12]
Z k ( f ) = F [ z k ( )] = e j 2kfT X ( f v ) Y ( v ) dv , (31)
where X(f) and Y(f) are the Fourier transforms of x(t) and
y(t), respectively. Then, by applying the Poisson identity
1
e
k
j 2 kfT
( f m T )
T m
(32)
and after some straightforward algebraic manipulations,
the relationship in (30) can be re-written as
q( ) Z k ( f ) e j 2f df
k
(33)
1
m j 2m T
=
T Y (v ) m X T v e dv .
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