AE

U Int. J. Electron. Commun.

(2006) No. 7
c Gustav Fischer Verlag
Jena 1
Computationally-Efcient Methods for
Blind Decision Feedback Equalization of QAM Signals
Kevin Banovi c, Esam Abdel-Raheem, and Mohammed A.S. Khalid
Abstract This paper investigates computationally-efcient
methods for blind decision feedback equalization (DFE) that
reduce the complexity and power requirements of blind equal-
ization algorithms while maintaining their steady-state charac-
teristics for quadrature amplitude modulation (QAM) signals.
These include the power-of-two error (POT), selective coefcient
update (SCU), and frequency-domain block (FDB) methods. A
novel radius-directed stop-and-go (RSG) method is introduced,
which selectively adjusts the equalizer tap coefcients based on
the equalizer output radius. In addition, a new activation/de-
activation method based on the equalizer output radius is uti-
lized to control the feedback equalizer (FBE) of the DFEs. Sim-
ulation studies and analysis are providedfor empiricallyderived
cable and microwave channels and Ricean fading channels.
Keywords Adaptive ltering, decisionfeedbackequalizers, blind
equalization algorithms, fading channels
1. Introduction
Adaptive equalizers compensate for signal distortion
caused by intersymbol interference (ISI), whereby sym-
bols transmitted before and after a given symbol cor-
rupt the detection of that symbol. All physical channels
tend to exhibit ISI at high enough symbol rates [1], [2].
Blind equalization schemes improve the bandwidth ef-
ciency of a communication system by achieving equal-
izer tap adaptation without the transmission of a training
sequence [2][4]. Instead, blind equalization algorithms
utilize known symbols statistics for equalizer tap adapta-
tion until switching to the decision-directed mode after the
symbol error rate (SER) has been sufciently reduced.
Recently, quadrature amplitude modulation (QAM)
based communication standards were adopted for satellite,
cable, and very high speed digital subscriber line (VDSL)
applications. Blind equalization is recommended for both
the Pan-European satellite-based Digital Video Broad-
cast (DVB-S) [5] and cable-based (DVB-C) [6] standards.
Broadband standards for VDSL include provisions for
both single- and multiple-carrier modulation [7]. The lat-
ter uses carrierless amplitude-phase (CAP) or QAM and
requires the receiver to startup blindly. Although the Ad-
vance Television Systems Committee (ATSC) adopted 8-
vestigal side-band modulation (VSB) over 32-QAM for
terrestrial high denition television (HDTV) broadcast
Received July 2006.
K. Banovi c is with the Department of Electrical and Com-
puter Engineering, University of Toronto, 10 Kings College
Road, Toronto, ON M5S 3G4, Canada, E. Abdel-Raheem and
M.A.S. Khalid are with the Department of Electrical and Com-
puter Engineering, 401 Sunset Ave., Windsor, Ontario, N9B 3P4,
Canada. Email: banovic@eecg.toronto.edu, eraheem@uwindsor.ca,
mkhalid@uwindsor.ca
Fig. 1. Multirate system model for a decision feedback equalizer.
[8], blind decision feedback equalization (DFE) was cho-
sen over trained equalization. In eld tests conducted by
HDTV manufacturers, the blind DFE achieved a lower er-
ror rate and faster data acquisition than its trained counter-
part in time-varying terrestrial channels [9].
In mobile communication channels, such as those for
microwave radio, high order lters are needed to achieve
channel equalization. Equalizer tap adaptation is costly in
terms of power, memory, and computations and can be
impractical for mobile units. In this paper, we investigate
computationally-efcient methods for blind DFEs, which
reduce the complexity and power requirements of blind
equalization algorithms while maintaining their steady-
state characteristics for QAM signals. These include the
power-of-two error [10], [11], selective coefcient update
(also partial update) [12][14], and frequency-domain
block methods [15][18]. A novel radius-directed stop-
and-go method is introduced, which selectively updates
the equalizer tap coefcients based on the equalizer output
radius. This concept was conceived by Banovi c et. al. in
[19] for linear equalizers, where it was termed the se-
lective update method. In this reformulation for DFEs,
criteria is given for selection of the static bound pa-
rameter, analysis is provided for adjustment probability
and transient/steady-state performance, and a modied
method is proposed to reduce hardware complexity. In ad-
dition, a new activation/de-activation method based on the
equalizer output radius is utilized to control the feedback
equalizer (FBE) of the DFEs. Simulation studies for blind
DFEs employing the discussed methods are performed
over empirically derived cable and microwave channels
and for Ricean fading channels.
2. Fractionally-Spaced System Model
In this section, a signal model is constructed for the T/2-
spaced single-input single-output (SISO) baseband com-
munication system for a DFE, where T is the symbol pe-
riod and 1/T is the baud rate. A multirate model of the
system is illustrated in Fig. 1, where the index n de-
notes T-spaced quantities while k denotes T/2-spaced
quantities. A T-spaced source symbol s(n) is transmitted
through a pulse-shaping lter and modulated onto a T/2-
spaced propagation channel, whose impulse response is
2 Preprint submitted to Elsevier Science
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U Int. J. Electron. Commun.

(2006) No. 7
given by the nite series c
k

Nc1
k=0
, where N
c
is the chan-
nel length. This corresponds to the N
c
1 channel im-
pulse response vector of c = [c
0
, c
1
, . . . , c
Nc1
]
T
where
()
T
is the transpose operator and the channel is station-
ary (a time-varying channel can be used as long as it
does not vary faster than can be tracked by the equaliza-
tion algorithm). The source symbol is a random variable
that is independent and identically distributed (i.i.d.) with
zero mean and variance
2
s
= E[s(n)[
2
and is drawn
from a nite alphabet, which is given by the nite set
s
m
= s
m,R
+ s
m,I

M
m=1
for an M-QAM constellation,
where E is the expectation operator and the subscripts
R and I denote the magnitude of the real and imaginary
quantities, respectively.
The received T/2-spaced input signal u(k) is corrupted
by ISI and the additive white Gaussian noise (AWGN)
signal v(k). The baseband receiver consists of a N
f
-
tap T/2-spaced feedforward equalizer (FFE) and a N
b
-
tap T-spaced FBE to form a nonlinear DFE, where the
FFE removes the precursor ISI and the FBE removes
the postcursor ISI. The FFE and FBE tap-coefcients
are characterized by the nite series f
k

N
f
1
k=0
and
b
k

N
b
1
k=0
, respectively, which correspond to the N
f
1
vector f (n) =
_
f
0
(n), f
1
(n), . . . , f
N
f
1
(n)

T
and the
N
b
1 vector b(n) = [b
0
(n), b
1
(n), . . . , b
N
b
1
(n)]
T
, re-
spectively. The output of the FFE is baud spaced and
is formed by convolving the received T/2-spaced input
signal sequence with the FFE tap coefcients. The T/2-
spaced convolution matrix is constructed from the channel
impulse response vector and is dened as [2]
C
FS
=
_

_
c
0
c
1
c
0
.
.
. c
1
c
0
c
Nc1
.
.
. c
1
.
.
.
c
Nc1
.
.
.
.
.
. c
0
c
Nc1
c
1
.
.
.
.
.
.
c
Nc1
_

_
(1)
where C
FS
is an (N
c
+ N
f
1) N
f
matrix. The T-
spaced convolution matrix is formed by the odd rows of
(1) and is dened as
C =
_

_
c
1
c
0
c
3
c
2
c
1
c
0
.
.
.
.
.
. c
3
c
2
.
.
.
c
Nc1
c
Nc2
.
.
.
.
.
.
.
.
. c
1
c
0
c
Nc1
c
Nc2
c
3
c
2
.
.
.
.
.
.
.
.
.
c
Nc1
c
Nc2
_

_
(2)
where C is a P N
f
matrix and
P = (N
c
+ N
f
1)/2|. The regressor vector of
FFE input samples is comprised of the previous N
f
received T/2-spaced samples and is dened as
u(n) = C
T
s(n) +v(n) (3)
where s(n) = [s(n), s(n 1), . . . , s(n P + 1)]
T
is the P 1 transmitted source symbol vector and
v(n) = [v
0
(n), v
1
(n), . . . , v
N
f
1
(n)]
T
is the N
f
1
vector of AWGN samples. The FFE output is decimated
by a factor of two and is dened as
y(n) = u
T
(n)f (n) = s
T
(n)Cf (n) +v
T
(n)f (n). (4)
The DFE output signal is dened as
z(n) = x
T
(n)w(n) = u
T
(n)f (n) s
T
(n)b(n) (5)
where the N
w
1 vectors x(n) =
_
u
T
(n), s
T
(n)

T
and
w(n) =
_
f
T
(n), b
T
(n)

T
are the combined DFE in-
put regressor and tap coefcient vectors, respectively,
while s(n) = [ s(n), s(n 1), . . . , s(n N
b
+ 1)]
T
is
the N
b
1 regressor vector of past estimated symbol
points and N
w
= N
f
+ N
b
.
The FBE does not exhibit noise enhancement since it
utilizes past symbol estimates, which are assumed to be
correct [1]. When an incorrect symbol estimate is fed back
to the feedback tapped delay line, there is a greater like-
lihood of error propagation. Therefore, the SER must be
sufciently low before the FBE is activated. Initially, the
FFE is utilized to reduce the SER, while the FBE tap coef-
cients are xed at zero. After the SER is sufciently low,
the FBE is activated to reduce the postcursor ISI.
3. Blind Equalization Algorithms
3.1 Constant Modulus Algorithm
The constant modulus algorithm (CMA) [20], [21]
achieves channel equalization by penalizing the dispersion
of the squared output modulus, [z(n)[
2
, from the constant

2
c
. The cost function minimized by CMA is dened as
J
cma
=
1
4
E
_
_
[z(n)[
2

2
c
_
2
_
(6)
where
2
c
= E[s
m
[
4
/E[s
m
[
2
is the dispersion con-
stant. A gradient-descent equalizer tap adjustment algo-
rithm that minimizes J
cma
is dened as
w(n + 1) = w(n) + (
w
J
cma
) (7)
= w(n) + z(n)
_

2
c
[z(n)[
2
_
. .
e
cma
(n)
x

(n)
where is a positive stepsize,
w
is the gradient operator
with respect to the elements of vector w, e
cma
(n) is the
CMA error signal, and ()

denotes complex conjugation.

3.2 Multimodulus Algorithm
The multimodulus algorithm (MMA) [22], [23] achieves
channel equalization by penalizing the dispersion of
AE

U Int. J. Electron. Commun.
(2006) No. 7 Preprint submitted to Elsevier Science 3
z
R
(n) and z
I
(n) components squared from the constant

2
m
, where z(n) = z
R
(n) + z
I
(n). The cost function
minimized by MMA is dened as
J
mma
=
1
4
E
_
_
z
2
R
(n)
2
m
_
2
+
_
z
2
I
(n)
2
m
_
2
_
(8)
where
2
m
= Es
4
m,R
/Es
2
m,R
is the dispersion con-
stant. A gradient-descent equalizer tap adjustment algo-
rithm that minimizes J
mma
is dened as
w(n + 1) =w(n) + (
w
J
mma
) (9)
=w(n) +
_
e
mma
R
(n)
..
z
R
(n)
_

2
m
z
2
R
(n)
_
+ z
I
(n)
_

2
m
z
2
I
(n)
_
. .
e
mma
I
(n)
_
x

(n)
where e
mma
R
(n) and e
mma
I
(n) are the real and imaginary
components of the MMA error signal, respectively.
3.3 Decision-Directed Algorithm
The cost function minimized by the decision-directed
(DD) algorithm[24] utilizes the instantaneous error across
the slicer and is dened as
J
dd
=
1
2
E([z(n) s(n)[)
2
(10)
where s(n) = s
R
(n)+ s
I
(n) is the estimated QAM sym-
bol. A gradient-descent equalizer update algorithm that
minimizes J
dd
is dened as
w(n + 1) = w(n) +
_

w
J
dd
_
(11)
= w(n) + ( s(n) z(n))
. .
e
dd
(n)
x

(n)
where e
dd
(n) is the DD error signal. The DD algorithm
requires the mean-squared error (MSE) to be lower than a
specied threshold [25] and cannot be applied at the onset
of equalization.
4. Computationally-Efcient Methods
This section discusses computationally-efcient methods
that can be applied to both trained and blind adaptive
equalizers. As illustrated in Fig. 2, adaptive equalization
can be generalized into two operations: convolving the re-
ceived symbol sequence with the equalizer tap coefcients
and updating the equalizer tap coefcients. One method to
improve computational efciency is to simplify or reduce
the number of multiplications needed to realize the equal-
izer. Signed-error [26][28] and power-of-two error [10],
[11] are methods which simplify the multiplications in the
equalizer tap adjustment to shift operations when a power-
of-two step size is applied. The selective coefcient update
method [12], [13] reduces the number of multiplications
by updating only a subset of the total taps during an itera-
tion, while frequency-domain block algorithms [15][18]
perform time-domain convolution for a block of samples
in the frequency-domain.
4.1 Power-of-Two Error Method
The most common method of reducing the complexity
of an adaptive algorithm is to retain only the sign of the
error signal [26][28]. Signed-error algorithms simplify
the multiplications in the equalizer tap adjustment por-
tion to shift operations when a power-of-two step size is
applied. However, signed-error algorithms are character-
ized by rough convergence and high steady-state MSE.
An alternative method that avoids these characteristics is
the power-of-two (POT) error method [10], [11], which
quantizes the error signal of the respective algorithm to a
power-of-two. The general equalizer tap adjustment algo-
rithm for POT algorithms is dened as
w(n + 1) = w(n) + Qe(n)x

(n) (12)
where e(n) is the error signal of the respective algorithm
and Q is a nonlinear power-of-two quantizer, which
can be dened as [11]:
Qx =
_
_
_
csgn(x), [x[ 1
2
log2|x|
sgn(x), 2
L+2
[x[ < 1
csgn(x), [x[ < 2
L+2
(13)
where csgn() is the complex sign operator, L is the data
word length including the sign bit and is typically set to
either 0 or 2
L+1
.
The computational requirements for the POT method
equalizer tap update are specied in Table 1. When cou-
pled with a power-of-two step size, the multiplications are
reduced to shift operations while the equalizer tap adjust-
ment becomes shift and add operations.
4.2 Selective Coefcient Update Method
The complexity of an adaptive lter is proportional to
the number of its tap coefcients. By partially updating
the tap coefcients, the processor capacity can be uti-
lized more efciently while reducing the power consump-
tion [12][14]. The general equalizer tap adjustment al-
gorithm for selective coefcient update (SCU) algorithms
is dened as
w(n + 1) = w(n) + e(n)A
IP(n)
x

(n) (14)
where e(n) is the respective error signal and A
IP(n)
is
a diagonal matrix having P elements equal to one in the
positions indicated by I
P
(n) and zeros elsewhere, where
I
P
(n) is the N
w
1 update constraint vector. The up-
date constraint vector is determined through information
evaluation, which can be accomplished using a number of
methods. We will consider the xed and time-varying set-
membership criteria discussed in [14]. The rst method
is when P tap coefcients are updated during each itera-
tion, where P is a xed value between 0 < P < N
w
. The
equalizer input vector x(n) is sorted and the index posi-
tions that correspond to the largest P input samples are set
to one in I
P
(n), while all other positions are set to zero.
4 Preprint submitted to Elsevier Science
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U Int. J. Electron. Commun.

(2006) No. 7
Fig. 2. Direct form complex DFE with tap update portions indicated by dashed-boxes.
Table 1. Number of real arithmetic operations for the equalizer tap adjustment of a direct form complex DFE when = 2
n
.
Method Multiplications Additions Barrel Shifts FFTs/IFFTs Adjustment Percentage
None 4N
w
4N
w
2 0 1
FDB (FFE, FBE) 8N
f
, 4N
b
8N
f
, 4N
b
4N
f
, 2 3, 0 1/N
f
, 1
SCU 4P(n) 4P(n) 2 0 1
POT 0 4N
w
4N
w
+ 2 0 1
RSG 4N
w
4N
w
2 0 0 < F < 1
RSG-DPOT 0 6N
w
8N
w
+ 2 0 0 < F < 1
There is no restriction on the selection of P as long as the
stability or convergence is not compromised.
An alternate method is to let P vary with time, such that
P
min
P(n) P
max
. Initially, P(n) = P
min
and is in-
cremented by 1 until P(n) = P
max
or the regressor power
meets the following condition:
|A
IP(n)
x(n)|
2

p
|x(n)|
2
(15)
where |x| =
_
i
[x
i
[
2
is the two normand
p
is a xed
constant that ranges from 0 <
p
< 1.
The computational requirements for the xed and vari-
able SCU methods are specied in Table 1. These gures
do not include the overhead processing for the xed and
time-varying cases. One possible implementation of the
equalizer input power calculation is to immediately square
the input sample and apply the result to a separate tapped
delay line (TDL). The square of the latest input sample
would be added to an accumulator while the square of the
rst sample to leave the TDL would be subtracted to ob-
tain the current regressor power. This costs two real multi-
plications and four real additions per iteration while dou-
bling the storage elements for the TDLs.
4.3 Frequency-Domain Block Method
Frequency-domainblock (FDB) algorithms [15][18] use
a block of input samples and instantaneous error sam-
ples to update the equalizer taps once every B input sam-
ples, where B is the block length. Signicant reductions in
complexity are obtained by performing time-domain con-
volution in the frequency-domain, which is due in part
to the efciency of the fast Fourier transform (FFT) and
the inverse fast Fourier transform (IFFT) algorithms. For
DFE architectures, an exact frequency-domain implemen-
tation does not exist since the FBE would require future
symbol estimates. Therefore, in typical frequency-domain
DFE implementations, the FFE is implemented in the fre-
quency domain while the FBE is implemented in the time-
domain [29][31]. FDB algorithms can be realized using
the overlap-save or overlap-add sectioning methods. The
FDB implementation considered here is the overlap-save
method with a block size of B = N (where in this sec-
tion N
f
= N) since this is the most efcient value for the
FFT algorithms [15]. This corresponds to 2N frequency-
domain equalizer taps. The general equalizer tap adjust-
ment algorithm for FDB algorithms is dened as
F(nN+N) = F(nN)+T
_
gT
1
_
U
H
(nN)E(nN)
__
(16)
where here uppercase letters denote frequency-domain
quantities, g is the 2N 2N gradient constraint matrix,
and T and T
1
are the FFT and the IFFT, respectively,
while ()
H
is the Hermation transform (complex conju-
gation and transpose). The input signal matrix U(nN) is
comprised of two blocks of N input samples and is dened
as
U
T
(nN) = TT [u(nN N), . . . , u(nN + N 1)]
. .
u(nT)

(17)
where the T operator transforms the 2N 1 vector
into a 2N 2N diagonal matrix. The frequency-domain
equalizer output is Y(nN) = U(nN)F(nN), while the
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U Int. J. Electron. Commun.
(2006) No. 7 Preprint submitted to Elsevier Science 5
Fig. 3. Realization of the FDB method using the overlap-save sec-
tioning procedure.
N 1 time-domain equalizer output vector is dened as
y(nN) = kT
1
U(nN)F(nN) (18)
= [y(nN N + 1), . . . , y(nN)]
T
where k is the N 2N constraint matrix that ensures the
output result is a linear convolution [17]. The frequency-
domain error signal vector is dened as
E(nN) = T[0
T
N,1
, e
T
(nN)]
T
(19)
where e(nN) = [e(nN N + 1), . . . , e(nN)]
T
is the
time-domain error signal vector and 0
N,1
is the N 1
zero vector. The gradient and convolution constraint ma-
trices, g and k, respectively, are dened as [17]:
g =
_
I
N,N
0
N,N
0
N,N
0
N,N
_
k = [ 0
N,N
I
N,N
] (20)
where 0
N,N
is an N N zero matrix and I
N,N
is an
N N identity matrix.
The computational requirements for the FDB method
equalizer tap update are specied in Table 1. The FDB
method is applied to the FFE while the FBE is imple-
mented in the time-domain. As illustrated in Fig. 3, a to-
tal of three 2N-point FFTs and two 2N-point IFFTs are
needed to implement the FFE utilizing the FDB method,
where two FFTs and one IFFT are utilized specically for
the frequency-domain equalizer tap adjustment.
5. Radius-Directed Stop-and-Go Method
The radius-directed stop-and-go (RSG) method for QAM
signals selectively updates the equalizer tap coefcients
based on the equalizer output radius, r(n) = [z(n) s(n)[,
which is the Euclidean distance between the equalizer out-
put and estimated symbol. The equalizer tap coefcients
are only adjusted during iterations when r(n) > r
s
, where
r
s
is a constant bound. The general equalizer tap adjust-
ment algorithm for RSG algorithms is controlled by the
ag (n) and is dened as
w(n + 1) = w(n) + (n)e(n)x

(n) (21)
where e(n) is the error signal of the respective algorithm
and (n) is dened as
(n) =
_
1, if r(n) > r
s
0, otherwise
(22)
where the constant r
s
=
s
d/2,
s
is a user dened pa-
rameter, and d is the distance between symbol points. This
parameter is to be chosen between the following limits
2N
w

max
<
s

2
3
(23)
where
max
is the maximum adjustment error that can oc-
cur when the equalizer output is a constellation point (i.e.
when z(n) s
m

M
m=1
for a square M-QAM constella-
tion) and the upper limit of
s
corresponds to the mini-
mum level of MSE required for transfer to the DD algo-
rithm, which is denoted
dd
[32]. While there is no ad-
justment error for the DD algorithm in steady-state op-
eration, this is not the case for statistical mean algo-
rithms such as CMA and MMA, which have non-zero
updates for z(n) s
m

M
m=1
. These algorithms accentu-
ate the bottom of the bowl scenario of classical gradi-
ent search methods, where the equalizer tap coefcients
bounce around the optimal solution. As a result, these uc-
tuations cause the steady-state MSE to increase.
At the uppermost limit of (23), the number of equalizer
tap updates will be minimized at the expense of a high
steady-state MSE. As
s
approaches the lowermost limit,
the steady-state MSE will be equivalent to the original al-
gorithmwith slightly fewer updates. When selecting
s
, it
is important to note that when the MSE level is below
dd
,
the steady-state MSE for the original algorithm,
ss
, can
be approximated as the error across the slicer as follows

ss

= E
_
[ s(n) z(n)[
2
_
= E
_
r
2
(n)
_
. (24)
The relationship between the steady-state MSE for the se-
lective update method,
rsg
ss
, and
ss
for equalizers in the
decision-directed mode of operation, can be expressed as

rsg
ss
_

=
ss
, if
_

ss
r
2
s
and r
2
s
,
ss
_
or
ss
r
2
s
>
ss
, if
ss
r
2
s
and r
2
s

ss
.
(25)
This can be explained as follows: if r
2
s

ss
, then
r
2
s
E
_
r(n)
2
_
, which will cause a signicant reduction
in equalizer tap updates. This can effect
rsg
ss
construc-
tively or destructively, depending on the selection of r
s
. If
r
2
s

ss
, the steady-state MSE will approach r
2
s
since the
equalizer tap coefcients will only be updated once
rsg
ss
is
degraded. However, as r
2
s

ss
, the
rsg
ss
will decrease to
the point where
rsg
ss

=
ss
.
At the onset of equalization the equalizer output will be
a random i.i.d. value, which will result in the following
6 Preprint submitted to Elsevier Science
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(2006) No. 7
equalizer update probability:
Pr[update] =
d
2
(
s
d/2)
2

d
2
= 1

2
s

4
. (26)
This probability decreases as the equalizer adapts and
reaches a minimum when the equalizer is in steady-
state operation. During the initial stages of adapta-
tion, E r(n) r
s
causing the equalizer taps to be
updated frequently. This allows the respective algo-
rithm to maintain its transient characteristics. As the
E
_
r(n)
2
_

ss
E r(n) r
s
, the equalizer is in
steady-state operation and the number of equalizer tap up-
dates will be at a minimum. If the channel should experi-
ence sudden changes, the MSE will increase and the pro-
cess will repeat.
While the RSG method reduces the number of equalizer
tap adjustments, there are no reductions in the hardware
resources needed for implementation. The RSG method
can be modied to reduce the hardware complexity by
combining it with methods that simplify the multipli-
cations in the tap coefcient update, such as the POT
method. Here we propose retaining the rst two leading
ones of the error signal, which we term the double power-
of-two (DPOT) method. This is proposed to improve the
accuracy of the error signal estimate over the POT method,
while minimizing the added hardware complexity over
that method. The general equalizer tap adjustment algo-
rithm for RSG-DPOT algorithms is dened as
w(n + 1) = w(n) + (n)e
dpot
(n)x

(n) (27)
where e
dpot
(n) = Qe(n) + Qe(n) Qe(n),
e(n) is the error signal of the respective algorithm and
Q is the nonlinear power-of-two quantizer that was
dened in (13) for the POT method.
The computational requirements for the RSG and RSG-
DPOT methods are specied in Table 1. The RSG method
maintains the same hardware complexity but reduces the
number of equalizer tap adjustments and hence, the num-
ber of computations. However, in addition to reducing
the number of equalizer tap adjustments, the RSG-DPOT
method reduces the equalizer tap adjustment to shift and
add operations when a power-of-two step size is applied.
The calculation of r(n) requires two real multiplications,
three real additions and one real square root. However, if
the operand precision is sufcient, r
2
(n) can be utilized
to determine whether to adjust the equalizer taps, which
eliminates the square root function. Alternatively, a look-
up-table (LUT) could be utilized to implement the square
root function.
6. Simulation Study
This section presents simulation results for
computationally-efcient methods applied to CMA-
and MMA-based DFEs. The algorithms are simulated
over empirically derived cable and microwave channels
from the Signal Processing Information Base (SPIB,
located: http://spib.rice.edu/) and multi-tap
Ricean fading channels. The simulation environment
Calculate r(n)
r(n) < d/3
count < r
th
count > 0
count=count-1
Yes
No
Yes
Turn off DFE
Yes
No
No
count=count+1 Turn on DFE
Fig. 4. DFE activation and de-activation ow chart.
consists of a T/2-spaced channel in cascade with a 54-tap
DFE consisting of an 18-tap T/2-spaced FFE and a
36-tap T-spaced FBE, where the channel, FFE, and FBE
are modeled as complex FIR lters. The source symbol
sequence is randomly generated using an i.i.d. process and
is drawn from a normalized square QAM constellation.
The received equalizer input samples are generated by
convolving the source sequence with the channel impulse
response and adding AWGN.
The DFE is controlled by the activation/de-activation
method illustrated in Fig. 4, which utilizes r(n) to de-
termine whether to activate or de-activate the FBE of the
DFE. At the onset of equalization, the FFE is initialized
with a dual center spike of 1/

2 and the FBE is xed at

zero, while the variable count is set to zero. The FFE is
adapted blindly using CMA or MMA. During an iteration,
if r(n) < d/3, count will increment by one while less
than the user dened threshold value r
th
, where d/3 corre-
sponds to
dd
[32]. Once count reaches r
th
, the FBE will
be activated and count will saturate at r
th
. The adapta-
tion of the DFE is switched to the DD algorithm. How-
ever, if r(n) > d/3, count will decrement by one while
greater than zero. If the DFE is in the active state, once
count reaches zero, the DFE will deactivate. The FFE
will be adapted blindly using CMA or MMA and the FBE
will be xed at zero, while count will saturate at zero.
The rst set of simulation studies compare the stan-
dard computationally-efcient methods of FDB, SCU and
POT with RSG for CMA- and MMA-based DFEs. In these
simulations, the algorithms are compared with the FDB
algorithm since its performance is equivalent to that of
the original DFE. The second set of simulations com-
pare RSG-based methods that reduce the number of com-
putations and hardware complexity of the original RSG
method. The RSG method is combined with the SCU,
POT, and DPOT methods for CMA- and MMA-based
DFEs, where each method is compared to the original
RSG method. The simulation parameters for each method
are as follows:
p
= 0.875 for SCU, L = 16 and = 0
for POT, and
s
= 2 10
3/2
d
1
for RSG, while r
th
was
set to 63 for all DFEs. Quantitative simulation results are
presented for the steady-state MSE, the average time-to-
AE

U Int. J. Electron. Commun.
(2006) No. 7 Preprint submitted to Elsevier Science 7
convergence (TTC), the equalizer adjustment percentage
for RSG-based DFEs in transient and steady-state opera-
tion (TR/SS), and the tap coefcient update (TCU) per-
centage for SCU-based DFEs. The MSE curves are ob-
tained by averaging the instantaneous squared-error across
the slicer over 300 realizations. The TTC is calculated as
the number of symbols needed to reach 90% of the steady-
state MSE while
ss
is the average MSE over the the nal
10% of the estimated symbol sequence.
Microwave channel simulations were conducted for
CMA- and MMA-based DFEs over SPIB microwave
channels #1, 2, 4, 5, 8 10 and #1, 2, 4 6, 8 10, re-
spectively, for 64- and 16-QAM signals with a signal-to-
noise ratio (SNR) of 35dB, where each channel consists
of 208-300 complex T/2-spaced taps. These modulation
schemes were chosen since they are the largest schemes
applied to the data carriers in an orthogonal frequency
division multiplex (OFDM) frame for terrestrial DVB.
Step sizes of = 2
11
and = 2
10
were applied to the
equalizer tap adjustment for 64- and 16-QAM signals, re-
spectively. Simulation results are illustrated for channel #8
in Fig. 5 for CMA- and MMA-based DFEs with 64-QAM,
which are representative of the typical results obtained.
Quantitative results averaged over all channels are given in
Table 2. On average, in comparison with the FDB method,
the RSG method achieves a slightly longer TTC, while the
SCU and POT methods require signicantly more sym-
bols to converge for CMA- and MMA-based DFEs. All
methods are able to achieve similar steady-state MSE val-
ues. As expected, in transient operation, the equalizer ad-
justment percentage of RSG-based DFEs is high at above
91% for both 64- and 16-QAM, respectively, while in
steady-state operation, the percentage is below 67% and
56%. Less than 73% of the DFE tap coefcients are up-
dated for SCU-based DFEs. Of the RSG-based DFEs, the
RSG-DPOT method obtains a slightly longer TTC while
maintaining the properties of the RSG-based DFEs. The
results for CMA- and MMA-based DFEs for microwave
channels indicate similar transient and steady-state char-
acteristics for the efcient methods under consideration.
Therefore, without loss of generality, the remaining simu-
lations will be for MMA-based DFEs.
Cable channel simulations were conducted for MMA-
based DFEs over SPIB cable channels #1, 2 for 256-
and 64-QAM signals with an SNR of 40dB, where each
channel consists of 128 complex T/2-spaced taps. These
modulation schemes were selected since they are the
largest square schemes utilized for DVB-C. Step sizes of
= 2
12
and = 2
11
were applied to the equalizer tap
adjustment for 256- and 64-QAM signals, respectively.
Simulation results are illustrated for channel #1 in Fig. 6
for 256-QAM, which are representative of the typical re-
sults obtained. Quantitative results averaged over all ca-
ble channels are given in Table 3. On average, the RSG
method achieves a slightly longer TTC, while the SCU
and POT methods require signicantly more symbols to
converge. All methods are able to achieve similar steady-
state MSE values. The equalizer adjustment percentage for
RSG-based DFEs in transient operation is above 81% and
90% for 256- and 64-QAM, respectively, while in steady-
state operation, the percentage is below 38% and 30%.
Fig. 7. Ricean fading channel model used for simulations.
For SCU-based DFEs, 64% of the DFE tap coefcients
were updated. Of the RSG-based DFEs, the RSG-DPOT
method obtains a slightly longer TTC while once again
maintaining the properties of the RSG-based DFEs.
Fading channel simulations were conducted for MMA-
based DFEs over a 5-tap Ricean fading channel as illus-
trated in Fig. 7 for 16-QAM and quadrature phase shift
keying (QPSK) signals with a SNR of 40dB, where g(k)
is the Rayleigh fading process and K is the rice factor
which was set to K = 1dB, K = 2.5dB, and K = 5dB.
The Rayleigh fading processes applied to the channel taps
were independent, randomly initialized, and simulated us-
ing the normalized low-pass fading process of Jakes sim-
ulator [33]. The fading parameters were selected using the
IMT-2000 evaluation methodology for 3G wireless com-
munications [34], where the carrier and symbol frequen-
cies were set to 2GHz and 4.096MBaud, respectively.
The mobile transceiver was moving at a velocity of 12
km/h to represent a person jogging. Step sizes of = 2
9
and = 2
6
were applied to the equalizer tap adjust-
ment for 16-QAM and QPSK signals, respectively. Sim-
ulation results are illustrated in Fig. 8 for 16-QAM with
K = 1dB. Quantitative results are given in Table 4. On
average, for 16-QAM signals, the RSG method achieves
nearly identical TTC and steady-state MSE values, while
the SCU and POT methods obtain a signicantly larger
TTC and a higher steady-state MSE, where the perfor-
mance of the POT method is the worst for both. For QPSK,
the steady-state MSE of the SCU method is nearly iden-
tical to that of the original while the TTC properties re-
main the same for all methods. The equalizer adjustment
percentage for RSG-based DFEs in transient operation is
above 93%and 83%for 16-QAMand QPSK, respectively,
while in steady-state operation, the percentage is below
71% and 44%. For SCU-based DFEs, 42% of the DFE
tap coefcients were updated for 16-QAM and 48% for
QPSK. Of the RSG-based DFEs, the RSG-DPOT method
obtains a slightly longer TTC and slightly higher steady-
state MSE, while the RSG-SCU and RSG-POT methods
obtain a signicantly higher steady-state MSE and longer
TTC for 16-QAM signals..
7. Conclusion
This paper investigated computationally-efcient methods
for blind DFEs that reduced the complexity and power re-
quirements of blind equalization algorithms while main-
taining their steady-state characteristics for complex sig-
nals. New computationally-efcient methods methods
were proposed that reduced the number of computations
8 Preprint submitted to Elsevier Science
AE

U Int. J. Electron. Commun.

(2006) No. 7
and hardware requirements for blind DFEs, mainly the
RSG and RSG-DPOT methods. Simulation results for
static cable and microwave channels and for Ricean fad-
ing channels indicate that the new methods maintain the
transient and steady-state performance of the original al-
gorithm, while reducing their complexity.
Acknowledgement
The authors wish to thank Harb Abdulhamid and Raymond Lee for
their comments and suggestions during discussions on fading chan-
nels and the proposed methods.
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[17] Shynk, J. J.: Frequency-domain and multirate adaptive lter-
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[18] Shynk, J. J.; Gooch, R. P.; Witmer, D. P.; Chjan, C. K.; Ready,
M. J.: Adaptive equalization using multirate ltering tech-
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[19] Banovic, K.; Lee, R.; Abdel-Raheem, E.; Khalid, M. A. S.:
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current blind deconvolution for channel equalization. Proc.
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[26] Im, G.; Park, C.; Won, H.: A blind equalization with the sign
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bust, computationally efcient blind adaptive equalization.
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[29] Pancaldi, F.; Vitetta, G. M.: Frequency-domain equalization
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Comm. 53 (July 2005), 11271131.
[31] Kim, D. K.; Park, P.: Adaptive self-orthogonalizing per-tone
decision feedback equalizer for single carrier modulations.
IEEE Signal Processing Letters 13 (January 2006), 2124.
[32] Banovic, K.; Abdel-Raheem, E.; Khalid, M. A. S.: A novel
radius-adjusted approach for blind adaptive equalization.
IEEE Signal Processing Letters 13 (January 2006), 3740.
[33] Jakes, W. C.: Microwave mobile communications. Piscataway,
New Jersey: IEEE Press, 1994.
[34] Evaluation Group ARIB IMT 2000 Study Committee: Evalu-
ation methodology for IMT-2000 ratio transmission technolo-
gies. ARIB (June 1998).
Kevin Banovi c received his B.A.Sc. and
M.A.Sc. degrees in electrical engineer-
ing from the University of Windsor, On-
tario, Canada, in 2003 and 2006, respec-
tively. He is a candidate in the electri-
cal and computer engineering Ph.D. pro-
gram at the University of Toronto, Ontario,
Canada. His research interests include the
design and applications of signal process-
ing microsystems, adaptive signal process-
ing, high performance VLSI design and
eld-programmable logic.
AE

U Int. J. Electron. Commun.
(2006) No. 7 Preprint submitted to Elsevier Science 9
Esam Abdel-Raheem received his
B.Sc. and M.Sc. degrees from Ain Shams
University, Cairo, Egypt, in 1984 and
1989, respectively, and Ph.D. degree
from the University of Victoria, Canada
in 1995, all in Electrical Engineering.
Currently, he is an Associate Professor at
University of Windsor, Ontario, Canada.
Dr. Abdel-Raheems research elds of
interests are in digital signal processing,
signal processing for communications,
and VLSI signal processing. He is a senior member of the IEEE and
a member of the IEEE SPS tech. committee on Signal Processing
Education and IEEE CAS tech. committee on VLSI systems &
applications. He has served as the technical program co-chair for
IEEE ISSPIT 2004 & 2005.
Mohammed A.S. Khalid received the
Ph.D. degree in Computer Engineering
from the University of Toronto in 1999.
He is an Assistant Professor in Electrical
and Computer Engineering Department at
the University of Windsor. From 1999 to
2003, he was a Senior Member of Techni-
cal Staff in the Verication Acceleration R
& D Group (formerly Quickturn), of Ca-
dence Design Systems, based in San Jose,
California. His research and development
interests are in architecture and CAD for eld programmable chips
and systems, recongurable computing, digital system design and
hardware description languages.
10 Preprint submitted to Elsevier Science
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(2006) No. 7
0 0.25 0.5 0.75 1 1.25 1.5 1.75 2
x 10
4
30
27.5
25
22.5
20
17.5
15
12.5
10
Symbols
M
S
E

(
d
B
)


POTCMA
SCUCMA
RSGCMA
FDBCMA
(a) Efcient methods for CMA-based DFEs.
0 0.25 0.5 0.75 1 1.25 1.5 1.75 2
x 10
4
30
27.5
25
22.5
20
17.5
15
12.5
10
Symbols
M
S
E

(
d
B
)
POTMMA
SCUMMA
RSGMMA
FDBMMA
(b) Efcient methods for MMA-based DFEs.
0 0.25 0.5 0.75 1 1.25 1.5 1.75 2
x 10
4
30
27.5
25
22.5
20
17.5
15
12.5
10
Symbols
M
S
E

(
d
B
)


RSGPOTCMA
RSGSCUCMA
RSGDPOTCMA
RSGCMA
(c) Efcient methods for RSG-CMA-based DFEs.
0 0.25 0.5 0.75 1 1.25 1.5 1.75 2
x 10
4
30
27.5
25
22.5
20
17.5
15
12.5
10
Symbols
M
S
E

(
d
B
)
RSGPOTMMA
RSGSCUMMA
RSGSCUMMA
RSGMMA
(d) Efcient methods for RSG-MMA-based methods for DFEs.
Fig. 5. Comparison of efcient methods applied to CMA/MMA-based DFEs for SPIB microwave channel #8 with 64-QAM signals.
Table 2. Quantitative results averaged over SPIB static microwave channels.
64-QAM 16-QAM
DFE Method
ss
(dB) TTC (T) TR/SS TCU
ss
(dB) TTC (T) TR/SS TCU
FDB-CMA -26.68 11,246 -28.24 3,769
SCU-CMA -26.24 12,627 0.72 -28.24 5,123 0.67
POT-CMA -26.55 15,827 -28.24 5,236
RSG-CMA -26.67 11,284 0.92/0.63 -28.22 3,813 0.92/0.53
RSG-SCU-CMA -26.33 13,619 0.93/0.63 0.72 -28.25 5172 0.93/0.54 0.67
RSG-POT-CMA -26.67 16,324 0.94/0.66 -28.19 5291 0.94/0.55
RSG-DPOT-CMA -26.59 12,633 0.93/0.64 -28.20 4194 0.93/0.54
FDB-MMA -26.96 9,764 -28.36 3,664
SCU-MMA -26.91 15,449 0.65 -28.40 4,865 0.65
POT-MMA -26.93 13,739 -28.36 5,046
RSG-MMA -26.95 9,898 0.93/0.62 -28.34 3,686 0.93/0.52
RSG-SCU-MMA -26.77 15,322 0.93/0.64 0.65 -28.39 4,916 0.93/0.52 0.65
RSG-POT-MMA -26.80 13,880 0.94/0.65 -28.35 5,063 0.93/0.52
RSG-DPOT-MMA -26.91 10,836 0.93/0.63 -28.35 4,016 0.93/0.52
AE

U Int. J. Electron. Commun.
(2006) No. 7 Preprint submitted to Elsevier Science 11
0 0.25 0.5 0.75 1 1.25 1.5 1.75 2 2.25 2.5
x 10
4
35
32.5
30
27.5
25
22.5
20
17.5
15
12.5
10
Symbols
M
S
E

(
d
B
)
POTMMA
SCUMMA
FDBMMA
RSGMMA
(a) Efcient methods for MMA-based DFEs.
0 0.25 0.5 0.75 1 1.25 1.5 1.75 2 2.25 2.5
x 10
4
35
32.5
30
27.5
25
22.5
20
17.5
15
12.5
10
Symbols
M
S
E

(
d
B
)
RSGPOTMMA
RSGSCUMMA
RSGDPOTMMA
RSGMMA
(b) Efcient methods for RSG-MMA-based DFEs.
Fig. 6. Comparison of efcient methods applied to MMA-based DFEs for SPIB cable channel #1 with 256-QAM signals.
Table 3. Quantitative results averaged over SPIB static cable channels.
256-QAM 64-QAM
DFE Method
ss
(dB) TTC (T) TR/SS TCU
ss
(dB) TTC (T) TR/SS TCU
FDB-MMA -30.98 11,298 -31.79 4,805
SCU-MMA -31.11 13,298 0.64 -31.94 5,429 0.64
POT-MMA -30.87 16,539 -31.71 6,959
RSG-MMA -30.92 12,099 0.83/0.32 -31.71 4,910 0.94/0.28
RSG-SCU-MMA -31.02 13,980 0.82/0.32 0.64 -31.83 5,558 0.91/0.26 0.64
RSG-POT-MMA -30.72 17,629 0.84/0.37 -31.60 7,043 0.91/0.29
RSG-DPOT-MMA -30.90 13,616 0.83/0.32 -31.68 5,493 0.91/0.27
0 0.25 0.5 0.75 1 1.25 1.5 1.75 2 2.25 2.5
x 10
4
25
22.5
20
17.5
15
12.5
10
Symbols
M
S
E

(
d
B
)
POTMMA
SCUMMA
RSGMMA
FDBMMA
(a) Efcient methods for MMA-based DFEs.
0 0.25 0.5 0.75 1 1.25 1.5 1.75 2 2.25 2.5
x 10
4
25
22.5
20
17.5
15
12.5
10
Symbols
M
S
E

(
d
B
)
RSGPOTMMA
RSGSCUMMA
RSGDPOTMMA
RSGMMA
(b) Efcient methods for RSG-MMA-based DFEs.
Fig. 8. Comparison of efcient methods applied to MMA-based DFEs for Ricean fading channels with K=1dB for 16-QAM signals.
Table 4. Quantitative results for Ricean fading channels.
16-QAM QPSK
DFE Method
ss
(dB) TTC (T) TR/SS TCU
ss
(dB) TTC (T) TR/SS TCU
FDB-MMA -25.23 5,116 -28.86 888
SCU-MMA -24.62 5,972 0.42 -27.19 1,000 0.48
POT-MMA -24.16 6,386 -28.92 1,126
RSG-MMA -25.03 5,148 0.94/0.65 -28.65 876 0.84/0.38
RSG-SCU-MMA -24.44 5,926 0.94/0.67 0.42 -26.99 968 0.88/0.43 0.31
RSG-POT-MMA -24.01 6,472 0.95/0.70 -28.75 1,150 0.86/0.38
RSG-DPOT-MMA -24.77 5,430 0.95/0.66 -28.71 908 0.89/0.39