Professional Documents
Culture Documents
HSU Square Root Kalman
HSU Square Root Kalman
HSU Square Root Kalman
T HIS PAPER is concerned with square root Kalman reduced to ranges between lOeN/* and 10N/‘. Thus the
algorithms for adaptive equalization of fading disper- square root algorithm achieve accuracies that are compara-
sive high-frequency (HF) radio channels. High-speed adap- ble with a conventional Kalman algorithm that uses twice
tive communication systems are constrained by the non- the numerical precision [3].
ideal characteristics of HF channels such as bandwidth The application of conventional Kalman filtering theory
restrictions, signal fading, and multipath dispersion. These to the adaptive equalization techniques has been consid-
channel limitations present the major obstacles to achiev- ered by Godard [14]. In this paper, new algorithms based
ing increased digital transmission rates with the required on the Kalman/Godard algorithm and the Carlson/Bier-
accuracies. man [3], [6], [12] (or Gentleman [13]) U-D covariance fac-
High data-rate HF transmission (up to 9600 bits/s in 3 torization filter are proposed for updating the tap gains of
KHz bandwidths) can be achieved by use of a single carrier a decision-feedback equalizer for fading dispersive HF
bandwidth-efficient modulation that is demodulated with a channels. These new algorithms are applicable to complex
decision feedback equalizer (DFE) operated adaptively [I]. (i.e., angle-modulated) input signals and time-varying
The DFE coefficients can be updated adaptively by gradi- channels. Theoretical formulation and mechanization pro-
ent, Kalman algorithms, etc. It is found [2] that only the cedures for the square root Kalman algorithms are given
Kalman algorithm provides a tracking rate sufficient to here. Computer simulation indicates that good error-rate
follow the time-varying HF channel. Unfortunately, the performance can be achieved by these algorithms for rapidly
Kalman algorithm has been found to be sensitive to com- fading HF radio channels.
puter roundoff errors; numerical accuracy due to roundoff The computational requirements of the square root algo-
sometimes degrades performance to the point where the rithms for an equalizer with N taps are 1.5 N2 complex
results become meaningless [3]. Simulations that we have multiplications per cycle, which are roughly equivalent to
conducted in fixed-point arithmetic have shown the perfor- the requirements of the conventional Kalman algorithm.
mance of a DFE with conventional Kalman updating does Typically, the keying rate of an HF modem with 3 KHz
indeed degrade rapidly as the computer-word size is transmission bandwidth is set to 2400 bits/s. A decision
decreased. Recently, Falconer and Ljung [4] have shown a feedback equalizer with 15 feedforward taps and 14 feed-
“fast Kalman algorithm” which requires only n rather than back taps can handle multipath time spread up to 6.25 ms.
n2 operations per cycle. In fact, the fast Kalman algorithm The multipath spread is less than 5 ms in typical HF
was found, independently, by us and by Lim and Mueller channels [ 151. The least-squares lattice structure introduced
by Morf et al. and adapted by Satorius and Park [16] for
Manuscript received March 25, 1980; revised July 6, 198 1. equalizer adjustment algorithm is a linear (all zero) equali-
The author is with the Sylvania Systems Group, Communications
Systems Division, GTE Products Corporation, 77 A Street, Needham zation technique. It is well known [17] that linear filters are
Heights, M A 02 194. unable to cope with severe fading-dispersive channels. The
least squares lattice decision-feedback equalizer formulated input to the feedback section is the output symbol decision
by Shensa [ 181 requires about 97 N multiplications per sequence from the nonlinear symbol detector. The equalizer
cycle, where N is the number of stage (in a N-zero, N-pole output can be expressed as
channel). The least-squares lattice DFE requires more com- M N-M- I
putation than the square root algorithms shown in this ik = 2 ajYkij + 2 b/ik-j> (2.4
paper for multipath time spreads up to 6.25 ms. Conse- j=O j=l
quently, DFE’s with square root Kalman algorithms are where ik is an estimate of the k th information symbol and
more efficient than DFE’s with least-squares lattice algo- {fk-,; . .,‘k-N+M+I) are P reviously detected symbols. The
rithm for HF modem implementation. The square root decision Ik is formed by quantizing the estimate fk to the
Kalman algorithms described are shown to be remarkably nearest information symbol. The least mean-square error
stable with rapid convergence and tracking abilities for (LMSE) criterion provides a practical means for selecting
short computer-word sizes. The implementation of these the equalizer tap coefficients {a,} and {b,}. Based on the
algorithms for high-speed HF modems has been accom- assumption that previously detected symbols in the feed-
plished recently [ 191. back section are correct, the minimization of the mean
square error
II. COMMUNICATIONS
SYSTEMMODEL
G2 = EIIk - fk12 (2.5)
The HF channel can be modeled as a tapped delay-line leads to the following set of linear equations for the tap
with time-varying coefficients followed by addition of coefficients of the feedforward filter
zero-mean white-Gaussian noise, v(t) with variance, cr,”
[13]. The tapped delay-line has taps spaced at no more
than the symbol interval T and nonzero tap coefficients
corresponding to discrete multipath propagation. The dis-
m = O,l;..,M, (2.6)
crete tapped delay-line model can combine the transmis-
sion filter, the channel, the matched filter, the symbol rate f; = Ai- (2.7)
samples and a noise whitening filter [l]. The tap coeffi- :Hk)
cients are assumed to be independent complex-Gaussian
random variables { fi, i = 0,. . . , L} with zero means and nk=j$, (2.8)
fixed variances, E{ ] f; I’}. The tap coefficients are generated
by filtering white-Gaussian noise through a filter whose
bandwidth is on the order of the fade rate. where es2and ui’, are the variances of the signal Ik and noise
It can be shown that passage of the random signals nk, respectively. The tap coefficients of the feedback filter
through the discrete tapped delay-line results in a received are given in terms of coefficients for the feedforward filters
signal sequence {Rk} which can be expressed as by the following expression
tics of the channel. Generally, this is not known to the The state transition matrix @(k, k - 1) is defined in a
receiver so that iterative procedures must be used to de- system modeled as
termine the tap coefficients. The Kalman/Godard filter is C(k) = @(k, k - l)C(k - 1) + W(k), (3.8)
designed to update the equalizer tap coefficients, adap-
tively, from the received signal samples. During the train- where W(k) is N-dimensional vector of zero-mean white-
ing period, the equalizer continuously seeks to minimize noise process. It is assumed that the noise processes W and
the deviation of its sampled output signal from an ideal c are statistically independent. The parameter 5’ denotes
reference signal internally generated by the receiver. When the approximate final mean square error, E[] c 1’1. The
the residual distortion is small enough, the equalizer is matrix Q(k) represents the covariance matrix of W . Gener-
switched into the decision-directed mode, using as a refer- ally, the state transition matrix is not known to the receiver
ence a reconstructed signal obtained by quantizing the in the equalization process. Nevertheless, one can assume
output signal of the equalizer to the nearest symbol. Then that the optimum tap-gain values are randomly varying
the equalizer has the ability to self-adapt to changes in about a mean value [14], i.e.,
channel characteristics during the transmission process. = C( k - l)opt + AC(k),
c(%t (3.9)
The Kalman/Godard algorithm will be described here for
updating the DFE coefficients. This algorithm is by far the where AC(k) is considered to be a noise process. The
most powerful, i.e., it is fastest, known adaptive algorithm matrix Q(k) is then reduced to
for adjusting the equalizer coefficients.
Q(k) = +W))@C(k))*‘] (3.10)
-At time k, the @puts { y,, JJ~+,, . . . ,yK+ M} and decisions
(Ik--l> Ik--2,. . .Jk-,v+M+J are in the feedforward and and, at each step from (3.7), the predicted-error covariance
feedback sections of decision feedback equalizer, respec- matrix is given by
tively. The signals can be represented by a vector
P’(k) = P(k - 1) + Q(k). (3.11)
X(k)’ = (Yk, Y~+I,’ ’ ‘,Yk+M,
Exact calculation of the correlation matrix Q(k) is impos-
‘ik-,,~k-2,‘.‘,~k~N+M+,), (34 sible because the optimum tap coefficients are unknown to
where t denotes the transpose. the receiver. It has been shown [14] that C(k) converges to
The equalizer tap coefficients at time k are represented C(k)*,t within less than 2N steps. If the matrix Q(k) is
by the vector assumed to be negligible for a slowly time-varying channel,
then one has [ 141
C(k - I)‘= (ao, al,-“,aM, b,, bz,-,b,+,v-,I,
and
X*(m)X(m)’
1-’ (3.12)
then the U-D factors of D(k - 1) - a-‘V(k - l)V*‘( k - utilized for updating the tap gains. On the other hand, the
1) can be obtained by evaluating the following ordered convergence rate of the a1gorith.m is primarily controlled
equations recursively for j = 2,. . . , N: by the parameter 6. A large value of [ will produce a slow
rate of stable convergence, whereas a small value of 5 will
d, = d,(k - l)$ (4.12) result in a fast rate of convergence but at the sacrifice of
stability. From (4.9) to (4.22), it is found that d,(l) is
d, = dj(k - l)?, (4.13) significantly different from dj(l), j = 2,3, * * *, N for small
J values of [(e.g., E = 0.001; (Y, = 1; d,(l) = 0.001; dj(l) = 1,
and for i = : 1,2; f *,j - 1, j = 2,3,-s. , N; q e 1). The magnitude disparity of d, and
dj in the start-up period can significantly disturb the
vi*vj equalization process. The disparity can also be ,expected to
Fij = - - (4.14)
oljd, ’ degrade the performance of the algorithm if a periodic
resetting scheme as described in Section VII is adopted.
where
The algorithm can be modified in such a way that it is
fi = x1*, (4.15) stable for various values of 5 without sacrificing the rate of
1-l convergence. A revised square-root formulation described
4 = 2 pi,j(k - 1)x,* + xj*, j = 2,3;*.,N, below will serve this purpose. This algorithm is more suited
i=l for implementing the receiver with a processor of limited
(4.16) word size.
vj = d,(k - l)& j= 1,2;..,N, (4.17) Theorem 3: If (3.6) is modified according to the follow-
ing equation
a, = E + qfi*, (4.18)
aJ = aj-, + vj4.*, j= 2,3;**,N. (4.19) P(k) = P(k - 1) + Q(k) - G(k)X(k)‘P(k - l),
(54
Theorem 2: The intermediate Kalman gain G and the
updated error covariance factors U and D can be obtained then the intermediate Kalman gain G and the updated
from the following algorithm error covariance factors U and D can be obtained from the
following algorithm
gj = vj> j = 1,2;.e,N, (4.20)
gj = v~o/, j = 1,2;.+,N, (5 4
d,(k) = (1 + q)& (4.21)
d,(k) = (1 + q)d,, j= 2,3;..,N, (4.22) d,(k) = d,( k - I)h (E + hr)
(5.3)
’ ((~1 + hr) ’
‘J = -h/ffj- 12 j= 2,3;.+,N, (4.23)
(0(/-l + ht)
Pii = PiJ(k - 1) + gi*hj, i = 1,2;..,j - 1, d,(k) = dj(k - l)h, (aj+ h,) , J = 2,3,.-.,N,
(4.24)
(5.4)
gj = g, + vjpTj(k - l), i = 1,2;..,j - 1,
(4.25) j= 2,3;..,N, (5.5)
A, = - (aJ-,‘+ h,) ’
where g, on the right-hand side of (4.25) is gi obtained from
(j - 1)th iteration, and Pij(k) = PrJ(k- 1) ’ gi*‘j, i = 1,2;..,j - 1,
G’= tg,,g2>->g,v)> (4.26) (5.6)
and the final Kalman gain is given by g, = gi + vjpTj(k - l), i = 1,2,.-e ,j - 1, (5.7)
G(k) = G/a,. (4.27)
where gi on the right-hand side of (5.7) is gi obtained from
The square root Kalman algorithm is given by Theorems (j - 1)th iteration, and
1 and 2.
f, = x1*,
V. A REVISEDSQUAREROOTFORMULATION j-l
h, = aNq, (5.13) 0.001 gave identical results for the same sequence {Ik} and
the same sequence of noise samples nk after equalization.
h, = 1 + q, (5.14)
PI = Pij? (6.14)
j= 2,3;.*,N, (5.24)
Aj = - (ajel’+ h,) ’ Pij = P* + gi*xjY (6.15)
TABLE I
COMPUTATIONALREQUIRRMENTSFORSQUAREROOTKALMANALGORITHMS
Equivalent Equivalent
Real Real Real Computer
Algorithms Multiplications Additions Reciprocals Storage
Square
Root 6N2 f IlN 6N2 + 6N N N2+8N+ 14
Algorithm
Revised
Square
Root 6N2+ llN+ 1 6N2+7N+ 1 N+ 1 N2t8Nf 16
Algorithm
(6.9) (6.10), and (6.12) are modified as Figs. 1 and 2 depict the convergence properties of the
1 equalizer tap weight with a two-path fading channel model
Y=- a, + h,’ (6.19) in the first 40 iterations. The DFE with square root and
revised square root Kalman algorithms converges very
d,(k) = d,tk - l)h,(E + h,)y, (6.20) rapidly following the initial start-up. The speed of conver-
P = aj-, + h,, (6.21) gence and stability of the algorithms will depend on the
parameters h, 6, and q chosen for various design objectives.
1 Fig. 3 shows the convergence properties of the equalizer
y=-----aj + h, ’ (6.22)
tap-weight with a two-path fixed channel using the revised
where h, = aNq. square root Kalman algorithm. It is clear that the conver-
The computational requirements of the square root algo- gence of a DFE with the revised square root algorithm is
rithms for an equalizer with N taps are listed in Table I. extremely fast and stable on fixed channels. These results
The requirements may be slightly different for various indicate that the square root Kalman algorithms can also
signal processors. be used to obtain high stability and fast start-up on tele-
phone channels.
VII. SIMULATION RESULTS Figs. 4 and 5 depict the performance of the DFE for 8 +
In this section we present some results of computer PSK with a two-path fading channel model using the
simulation of the properties and performance of the algo- square root and revised square root Kalman algorithms.
rithms presented in the previous sections. W e have used the The quantized symbol ik is assumed to be equal to the
fading dispersive HF radio channel model described previ- actual transmitted symbol I,. For purposes of comparison,
ously. A complex Gaussian noise is also generated and the error rate based on exact knowledge of the equalizer
coefficients (LMS sense) is also illustrated. Both the perfor-
added to the received signal.
This section contains the properties of convergence, the- mance based on theoretical tap-weights and square root
Kalman filters are depicted. In Fig. 4, the performance of
oretical and simulated performance results for the DFE
the DFE with the square root and revised square root
with 8-ary (8 - $) phase-shift keying (PSK) modulation.
Extensive simulations were conducted in the investigation Kalman algorithms exhibit 1.25 and 1 dB degradation
of the performance of the receiver using the DFE with respectively related to the ideal performance at P, = 10e3.
square root Kalman algorithms for mitigating the inter- The performance of the DFE with the revised square root
symbol interference and signal fading. The channel fade algorithm is slightly better than that with the square root
rate is 1 Hz and the channel model consists of two fading algorithm. This result is due to the improvement of stabil-
paths (except in Fig. 3). The received signal is regulated by ity of the revised square root Kalman algorithm. In Fig. 5,
an AGC filter in the front end with an AGC constant, the revised square root Kalman algorithm has been mod-
X = 0.02. The DFE consists of two sections, a feedforward ified to incorporate the periodic resetting of the U-D
section and a feedback section. The numbers of feedfor- factors according to (7.1) and (7.2). The resetting of matrices
ward and feedback taps are set to be greater and equal to may slightly degrade the performance in floating-point
the number of interfering symbols given in the discrete arithmetic (as indicated in Fig. 5) but will considerably
channel model, respectively. The initial values of U and D improve the performance in fixed-point arithmetic. The
resetting scheme is used to prevent the propagation of
in the square root algorithms are
roundoff errors associated with the adaptation history.
d, = 1.0, j= 1,2;..,N, (74
pij = (0.0, o.o), i = 1,2;*.,N;
APPENDIX
-j = i + 1, i + 2;..,N, (7.2)
The U-D factors of (5.18)
where x, y, and (x, y) denote the real part, imaginary part
and complex quantities. P= D’(k - 1) - cC’V’(k - l)V*‘(k - 1) (A.11
760 IEEE TRANSACTIONS ON INFORMATION THEORY, VOL. IT-28, NO. 5, SEPTEMBER 1982
0.8 -
0.7 -
0.6 -
z
d
I I I
0 10 20 30 40
N U M B E R OF ITERATION
Fig. 1. Properties of convergence of the equalizer tap weight (real part of the main tap shown; DFE; S+J; two-path fading
channel; 1 hz fade rate; N = 5; SNR = 20 dB).
0.8 - I vv-
AA
THEORETICAL TAP GAIN
0.7 - I
zd /vi.,
,/'.i.,.,*,.,.~'~.-.-. /'
0.6
I I
0 10 20 30 40
N U M B E R OF ITERATION
Fig. 2. Properties of convergence of the equalizer tap weight (real part of the main tap shown; DFE; 8+; two-path fading
channel; 1 Hz fade rate; N = 5; SNR = 20 dB).
may be found by forming the quadratic form X’pX*, Let us suppose wN = --a-‘, then (A.2) becomes
- -- X’PX* = X’D’X* + w,X’YV*‘X*
X+X* = X’U*D U’X*
= F*~DF
N-l
where
F= u’x*, (A.31
f, = xl*, (A.41
i I
j-1 N-l
f, = x j&,x; + x*
J'
j = 2,3, . . . . N. (A4 +WNUN*XN* 2 vjx, .
i=l j=l
HSU: SQUARE ROOT KALMAN FILTERING
761
1.0
0.9
0.8
TAP GAIN OBTAINED BY REVISED SDUARE ROOT KALMAN ALGORlTHM
(t = 0.001: 9 = 0.081
0.7
0.6
z
3 0.5
3
0.4
0.3
0.2
0.1
0.0 I t I I
10 20 30 40
NUMBER OF ITERATION
Fig. 3. Properties of Convergence of the Equalizer tap weight (main tap shown; DFE; S+; two-path fixed channel; N = 5;
SNR = 30 dB).
THEORETICAL
Fig. 4. Error rate performance (DFE; two-path fading channel; 1 Hz fade rate; X = 0.02).
162 IEEE TRANSACTIONS ON INFORMATIONTHEORY,VOL.IT-28,~o. 5, SEPTEMBER 1982
10-l
i
E 10-2 TAP WEIGHT
d
ii
5
b
c FOR EVERY 100
3 ITERATIONS)
2
m 10.:
0
E
lo”
10-l I I I \ I 1
0 5 10 15 20 25 30
Fig. 5. Error rate performance (DFE; two-path fading channel; 1 Hz fade rate; X = 0.02).
WrJ:,
wp,=- (A.8)
dN ’
wNvN
ll,N= -u * j=1,2;..,N-1, (A.91
dN J’
N-l
= jz, d;lXj12fWN-I (A. 10)
Similarly, the inductive reduction follows recursively, and after N - 1 steps the quadratic form has been reduced to