A Decision-Directed Bayesian Equalizer
Michael E. Wilhoyte Tokunbo Ogunfunmi
Applied Signal Technology
400 N. California Ave
Sunnyvale, CA 94086
ABSTRACT
This paper investigates a new decision-directed
Bayesian equalizer based on the blind Bayesian
equalizer introduced by Iltis, Shynk, and Giridhar. The
blind equalizer of Iltis suflers @om high complexity
making it impractical for sofhvare implementation of
QAM signals beyond 4 states and with bit rates greater
than 16 kbps. Complexity of the decison-directed
Bayesian equalizer is compared to the blind version. It
is shown that even though the decision directed
implementation is less complex, implementation of it in
software still seems limited to BPSK and QPSK with bits
rates less than 30 kbps. It is also demonstrated through
simulation results that even for initially poor channel
estimates, the decision-directed equalizer still
converges to a close approximation to the actual
channel impulse response.
1. INTRODUCTION
A decision-directed Bayesian equalizer is introduced. In
the following, decision-directed means that the weights
of the adaptive filter are updated based on an internally
generated desired response or decision. The maximum
a-posteriori (MAP) algorithm was first introduced by
Abend and Fritchman 111. It was shown that the MAP
algorithm is the optimal symbol by symbol detection
scheme assuming that the channel is known apriori.
Howver, in most cases, the channel charmeristics are
not known very accurately and are generally non-
stationary. Recently, Iltis, Shynk, and Giridhar
[2,3,4,5,6] introduced a new blind approximation to the
optimal MAP algorithm based on Bayesian probabilities
of the channel coefficients. Up until the development
the blind Bayesian equalizer (BBE), there were
generally two classes of blind equalizers: Bussgang-type
equalizers which minimize a non-quadratic
performance surface, and blind equalizers based on
higher order statistics. The Bussgang-type equalizers,
such as CMA [7], are “property restoral” type
algorithms that use gradient descent to update the filter
058-6393/96 $5.000 1996 IEEE
Proceedings of ASILOMAR-29
Santa Clara University
Dept. of Electrical Engineering
Santa Clara., CA 9505 1
coefficients. Thus, these type of blind equalizers
implicitly invert the transfer function of the channel.
The problems with the Bussgang equalizers are their
slow convergence as well as their ill-convergence, that
is, since the cost function is non-quadratic, there is a
possibility that the algorithm will converge to an
undesired local minima. In addition, channel inverting
equalizers tend to strike a compromise between
achieving enough gain in the nulled regions and
enhancing the noise in those regions. This makes
equalization of channels with deep spectral nulls
dif€icult. In contrast, the BBE does not invert the
channel. Instead, a worst case channel impulse length
is assumed and all MNb+lpossible symbol sequences
are formed and fed through channel estimates, where M
is the length of the symbol alphabet and N b + l is the
worst case length of the channel. The output of the
channel estimates are then compared to the incoming
sample which is used to 1) compute likelihood
probabilities which are then used to select the most
probable sequence given particular channel estimates,
and 2) to update the channel coefficients for each
sequence. The problem with this technique is that for
every hypothesized sequence, there is a corresponding
channel estimate. This means that the update
computations for the channel estimates grow
exponentially with the channel length and the
modulation complexity. This paper investigates the
possibility of using one channel estimate for all
hypothesized sequences and then use a decision dlrected
approach to update the channel coefficients. Thus the
new algorithm is a decision-directed Bayesian equalizer
(DDBE). Two algorithms are investigated for coefficient
updating: the LMS algorithm, and the Kalman filter.
Section 2 gives a brief description of the BBE. Section 3
gives a description of the DDBE. Section 4 presents
simulation results, and conclusions are given in section
5.
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 2. BLIND BAYESIAN EQUALIZATION
This section will outline the derivation of the BBE
presented in [2]. A discrete time signal and channel
model is assumed
Nb -1
r(k)= Eb,(k)d(k-n)+n(k),
n=O
where d@) is the current transmitted symbol, r(k) is the
output of the matched filter, and {b,,(k)} are the time-
varying channel coefficients. The symbol r’ is used to
denote a collection of all the matched filter outputs up to
time k. Thus r = {r(k),r(k- l),...,r(O)). Likewise,
d: = { d i ( k ) , d i ( k - l ) , ..., d j ( 0 ) }which denotes the ith
of Adk+’ symbol sequence up to time k. The conditional
probabilities of the ith symbol sequence given a received
matched filter output sequence are given by
~~
Figure I : Block Diagram of LMS BBE
for i = 42,.. .,A4 ,and where p(d: Irk ) denotes the
probability of d: given r , and c is a normalization
326
constant so that the mm of the conditional probabilities
over all possible symbol sequences is exactly 1.
Equation 1 are the conditional probabilities that give
rise to the optimum MAP algorithm derived in [11. The
subopbmum Bayesian equalizer operates on a set of
truncated symbol sequences denoted by
which has the same number of terms as the impulse
response of the channel. In [2], two versions of the
algorithm are given; one version uses a Kalman filter to
update the channel caefficients, and the other version
uses the LMS algorithm which is much simpler to
implement. During this study, the LMS version of the
BBE was considered. A biock diagram of the LMSBBE
is given in Figure 1. Each element has an associated
channel estimate denoted as 6, ( k ) . Each of the N
possible subsequences are filtered by their
corresponding channel estimate and compared to the
input signal r@). The error signal at stagej is then used
to compute the conditional probabilities which gwe rise
to the current transmitted symbol and is used to update
the channel estimate
2.1 COMPUTATIONALCOMPLEXITY OF BBE
The complexity of the LMSBBE algorithm is given in
Figure 2. In the table below N = AdNb+’. Since a
complex multiply is the same as 4 real multiplies and 2
real adds, the computations are given in terms of total
real multiplies and real adds. Most DSP processors,
(TMS320C50 for example) can do a multiply and
accumulate with data move in one cycle. T h s allows
FIR filtering to be done in L clock cycles, where L =
filter length. In [2], a complexity analysis is performed
and it was determined that for N b = 2 , and employing
a QPsK modulation with a symbol rate of 8 kbaud
would take four 50 ns devices assuming little overhead.
Variable Real Adds RealMult Divides
Figure 2: Complexity Table For M S B B E
3. DECISION DIRECTED BAYESIAN
EQUALIZER
The motivation for tlus algorithm comes from the fact
that for many digital signals, a channel estimate may be
calculated using the framing structure common in many
digital systems. For example, the digital A M P S (IS-54)
and the GSM cellular standards both transmit training
sequences or sync bits which have very good
autocorrelation properties. That is, the correlation
function of these sequences is exactly a delta function
for lags up to 5. Thus, ifx(k) is modulated transmitted
discrete time signal which is corrupted by a channel
h(k), then the received signal is y(k) = x(k)*h(k). A
possible channel estimation algorithm is to employ a
matched filter to the incoming sequence ye), where the
matched filter coefficients are the conjugated time
reversed modulated training sequence s(k). Thus when
the incoming signal is the modulated training sequence,
we rset
s*(-k)*y(k) = s*(-k)*s(k)*h(k) = 6(k)*h(k)= h(k)
This technique shall be referred to at the cross
correlation technique for channel estimation. The
decision-directed approach would thus initialize the
channel estimate with this initial estimate given above
(or some other estimate), and then further improve this
estimate with an adaptive technique.
A block diagram of the decision-directed Bayesian
equalizer (DDBE)is given in Figure 3.
b
ddJf”b i r k )
e, (k) 3 i, (k) produces the M L
estimatefor r@)
I
K ( k ) = K ( k , k - 1 ) - G ( k ) . ( d j v N b) H .K(k,k-1)
Figure 3: Decision-Directed Bayesian Equalizer
Note that there is only one channel estimate 6 ( k )that is
used for all signal estimates. The error signal which
corresponds to the maximum likelihood estimate for
r(k)is fed back to update the channel estimate. The
output symbols at time k are denoted by 6 ( k ) . In order
for this algorithm to work, an initial channel estimate
must be supplied to bootstrap the system. If an initial
channel estimate is too poor, then the channel
coefficients may not converge. However, it was shown
through simulation studies that even if the initial
channel estimate is poor, the channel coefficients m a y
still converge. Thus, there is good indication that this
method is robust enough for real applications.
327
Two methods were used to update the channel
coefficients, namely the LMS algorithm, and the
Kalman filter. The update equations for the LMS
algorithm are
6(k+1)= 6(k)+pedj”b&*(k), where
~ ( k=)r(k)- Pj (k) ,
and where j is the index which results in the maximum
likelihood estimate of r(k). The Kalman filter
implementation is given by first defining the process
equation and the measurement equation. Assuming a
random walk for the channel coefficients, we get
b(k + 1) = b ( k ) + ~ l ( k ) ,
where {v (k)} is a stationary zero mean noise process
and whose correlation matrix is q I . Thus the above
equation may be viewed as the process equation of the
channel coefficients for the Kalman filter where the
state transition matrix @(k, k + 1) is equal the identity
matrix. The measurement equation may be written as
r,(k)=djPNb .b(k)+v2(k), where
the measurement matrix C(k) = dj”’””, and the state
vector is b(k) . The Kalman filter algorithm for channel
estimation is summarized below.
a(k)=r(k)-d;”bb(k-l)
6 ( k ) = 6(k - 1)+ G(k)a * (k)
K(k + 1, k) = K ( k )+ q I ,where
o2is the estimated noise variance, and q is the variance
of the weight noise process.
3.1 COMPUTATIONAL C O M P L E m Y OF DDBE
The computational complexity of the DDBE using the
LMS algorithm to update the channel coefficients are
given in Figure 4.
Variable Real Adds Real Mult Divides
4N(N& +1) 0
YY 0 0
 AdFNblrk) N(ll+M-1) 33N 0 Two cases were studied: case 1 initialized the algorithm
70 with a fairly good channel estimate obtained from the
2Nb sNb
Figure 4: Complexity Table For DDBE
To get a clear understanding for how these two
algorithms compare in terms of their computational
complexity, the tables in Figures 2 and 4 were used to
compute the number of 50 ns devices required assuming
varylng conditions, It was assumed that a multiply takes
one clock cycle as well as an addition. It was also
assumed that a divide takes 30 cycles. An overhead of
20% was added to account for lines of code that are not
directly related to the computations given in the tables
of Figures 2 and 4. Figure 5 gives a plot of the required
number of 50 ns devices versus the symbol rate
assuming a QPSK modulation with a channel length,
N + 1 ,equal to three.
~
Figure 5: Required Number of50 ns Devices Vs.
Symbol Rate For QPSK Modulation With Nb = 2
This graph pretty much corresponds to the calculation
made in [2] which reported 4 devices required for a
symbol rate of 8 kbaud. In the graph, this point is
actually closer to 5 devices which is probably due to the
20% overhead. This graph shows that approximately a
40% savings is acheved using the decision-directed
approach. Note that both graphs are linear as expected
since the symbol rate is varylng and not the parameters
of the tables
4. SIMULATION RESULTS
Simulations of the DDBE were performed assuming a
QPSK modulation and with N b = 2 . The channel
model assumed was the following:
H ( z ) = l + 0.99exp(jz).z-' +0.3exp(jz/4).z"
328
cross correlation channel estimation technique discussed
in section 3.0,and case 2 initialized the algorithm with
a poor channel estimate. Figures 6 and 7 show the
results for case 1, and figures 8 and 9 show the results
for case 2. In Figures 6 and 8, the solid line is the actual
channel magnitude response and the dashed line shows
the initial estimate. Figures 7 and 9 show the channel
coefficient learning curves using the LMS algorithm
and the Kalman filter. Note that in both cases, the
Kdman filter converges faster than the LMS as
expected. Also note that even when the channel estimate
is rather poor, both the LMS and the Kalman filter
converge to a close approximation to the actual channel
estimate. However in this case, it takes roughly twice
the number of iterations to achieve convergence. This is
due to the initially poor symbol decisions being made
because of the poor channel estimate. More work should
be done to investigate the convergence properties of the
decision-directed Bayesian equalizer, but a general rule-
of-thumb for convergence is a 10% symbol error rate.
5. CONCLUSIONS
A decision directed algorithm was investigated for the
blind Bayesian equalizer. The advantage of the decision
directed approach over the blind approach is reduced
computational complexity. However, the complexity of
the DDBE still grows exponentially with the channel
length, so its implementation in firmware seems
limited to QPSK and BPSK modulation with multipath
channels of about 2 symbols, and with bit rates of about
30 kbps or less. The disadvantage of this approach is
that an initial channel estimate must be provided to
'bootstrap" the algorithm. However, obtaining channel
estimates for certain signal standards is quite feasible,
and as was demonstrated through simulation results,
even for initially poor channel estimates, the DDBE
may still converge allowing demodulation of the data.
6. REFERENCES
[l] K. Abend, B.D. Fritchman, "Statistical detection for
communication channels with intersymbol
interference", Proceedings of the IEEE, Vol 58, pp.
779-785, May 1970.
[2] R. A. Iltis, J. J. Shynk, and K. Giridhar, "Bayesian
algorithms for blind equalization using parallel adaptive
filtering", Personal Copy.
[3] R. A. Iltis, "Bayesian maximum-likelihood sequence
estimation algorithm for a priori unknown channels and
symbol timing", IEEE Joumal on Selected Areas in
Communications, Vol 10, pp. 579-588, Apr. 1992.
[4] R. A. Iltis, J. J. Shynk, and K. Giridhar, “Recursive
Bayesian algorithms for blind equalization”, Proc.
Twenty-FiJh Asilomar Con$ on Signals, Systems, and
Computers, Pacific Grove, Calif.,pp. 710-715 (Nov.
1991).
i* -31
J -I
[5] K. Giridhar, J. J. Shynk, and R. A. Iltis,
‘Bayesiddecision-feedback algorithm for blind f *.
adaptive equalization”, Optical Engineering, pp. 1211-
1223, Vol. 31, June 1992.
8
[6] R. A. Iltis, J. J. Shynk, and K, Giridhar, “Bayesian i*.*.t1..

blind equalization for coded waveforms”, IEEE, pp

0211-0215,1992.
Figure 7: Channel Coeficient Leaming Curves For
[7] J. R. Treichler, and B. G. Agee, “A new approach to DDBE Initialized With Channel Estimate of Figure 6.
multipath correction of constant modulus signals,”
IEEE Transactions on Acoust. Speech and Sig. Proc.,
I. ,
Vol. ASSP-31, pp. 459-472, April 1983. 1
[8] S. Haylan, Adaptive Filter Eheory, 2nd ed.,
RenticeHall, Englewood Cliffs, NJ (1991).

-111.

-a- ’
-0.4 I
.- . 0 .Z e.4

-7:
-21

-a 4 . 4
;v:
-0.1 0 @.a 0.4
Figure 8: Magnitude Response of The Poor Channel
Estimate Compared to Actual

CL.....I CrrfI.l..* .”l....L f.l....

Figure 6:Magnitude Response ofActual Channel Filter

Compared to Estimate Obtained From Cross
Correlation Technique.

Figure 9: Channel Coe@cient Leaming Curves For

DDBE Initialized With The Channel Estimate of Figure
8.

329