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BLIND ADAPTIVE CHANNEL ESTIMATION FOR OFDM SYSTEMS

Rafael Boloix-Tortosa, Franciso J. Simois-Tirado and Juan Jos e Murillo-Fuentes


DTSC en Universidad de Sevilla.
Paseo de los Descubrimientos s/n,
41092 Sevilla, Spain.
rboloix@us.es
ABSTRACT
In this paper, we face the problem of blind channel estimation
in zero padding OFDM systems. Based on the assumption
that the transmitted symbols are independent and identically
distributed, we propose a blind adaptive estimation algorithm
for minimum phase channels, based on Independent Compo-
nent Analysis, that exploits the particular structure of the ZP-
OFDM system model. Some included simulations show that
this novel algorithm can yield comparable performance as the
SVD methods with a reduced computational complexity.
Index Terms Adaptive Blind channel estimation, OFDM,
zero-padding (ZP).
1. INTRODUCTION
Orthogonal frequency division multiplexing is a multicarrier
transmission technique that enjoys high spectral efciency. It
has been proposed as transmission system for many of the
last generation systems: digital audio and video broadcast-
ing (DAB, DVB), digital subscriber lines (xDSL) and wire-
less local area networks (IEEE 802.11a, HIPERLAN/2). In
OFDM, blocks of symbols are transmitted in parallel over
several narrowband sub-channels or orthogonal sub-carriers.
In frequency selective fading environments the transmitted
symbols would suffer both inter-block (IBI) and inter-channel
interferences. In order to avoid IBI, a guard interval, no shorter
than the channel spread, is inserted in the transmitted sequence.
Among the alternatives, we focus in this work on zero padding
(ZP) OFDM systems which transmits no information during
the guard interval. This assures a complete elimination of
IBI at the receiver side and may achieve better performance
than the cyclic prex alternative (see [1],[2] and references
therein). Knowledge of the channel impulse response is im-
perative whenever coherent detection or bit loading is intended.
Blind channel estimation approaches can be taken into con-
sideration in order to avoid the payload of using training se-
quences or pilot tones. The majority of articles dealing with
This work was partially funded by Spanish government (Ministerio de
Educaci on y Ciencia TEC2006-13514-C02-02/TCM and Consolider-Ingenio
2010 CSD2008-00010) and the European Union (FEDER)
blind channel estimation in OFDM are based on subspace
techniques [3]. These are usually block methods that are
characterized by high computational cost. The algorithm pro-
posed in [4] in one of the few adaptive approaches to the sub-
space technique. The rest of adaptive schemes proposed so
far follow mainly a decision-directed structure.
In this work, we propose a blind adaptive channel esti-
mation algorithm for ZP-OFDM systems based on indepen-
dent component analysis (ICA) [5]. ICA applied to the blind
source separation problem (BSS) [6] aims to estimate a set
of unobservable sources from another set of observations. In
this paper we face the case of overdetermined BSS, i.e., the
number of observations available is greater than the number
of unknown sources. Overdetermined BSS is usually solved
applying a pre-whitening stage to reduce the number of obser-
vations. However, in ZP-OFDM systems the mixing matrix
has a particular and desirable known structure. That struc-
ture is lost if the pre-whitening step is applied. Therefore,
we seek a new adaptive BSS algorithm that ts that partic-
ular matrix structure. This was rst proposed in [7], where
a blind natural gradient based BSS algorithm was proposed
for equalization in ZP-OFDM. However, that algorithm was
based on a BSS algorithm for square matrices, not intended
for the overdetermined case. It was proved by simulations
to work ne, but its derivation lacked of mathematical rigor
including several rough approximations. In this paper we in-
clude the rigorously derivation of a suitable natural gradient
based ICA algorithm for efcient adaptive estimation of min-
imum phase channels in ZP-OFDM systems, exploiting the
appointed particular matrix structure.
2. SYSTEM MODEL
ZP-OFDM can be described by the baseband discrete-time
block equivalent model shown in Fig. 1. At the transmit-
ter, the information symbol sequence is divided into blocks of
N symbols. The i-th block passes through a serial to par-
allel converter to be arranged into vector s(i). Then, this
vector is modulated by the IFFT matrix F
H
N
, where F
N
is
the FFT matrix of dimensions N N with (k, r)-th entry
N
s
k
H(z) +
n(i)
S/P
s(i)
S/P P/S

1:N
M:1 1:M
m(i)

r(i)
F
H

Fig. 1. Discrete-time block equivalent model of ZP-OFDM.


(N)
1/2
exp(j2kr/N), for 0 k, r N 1. Next,
a sequence of L zero symbols is padded at the end of the
modulated block to avoid the IBI. This yields the block of
M = N + L transmitted symbols
m(i) = [I
N
0
NL
]
T
F
H
N
s(i). (1)
The symbols in m(i) are serialized and transmitted sequen-
tially through the multipath additive white noise channel with
impulse response h = [h
0
h
M
]
T
. It is assumed here that
the channel order is at most L (i.e., h
k
= 0, k > L).
At the receiver side, a serial to parallel conversion yields
the received block of symbols at instant i,
r(i) = Hm(i) +n(i) = H[I
N
0
NL
]
T
F
H
N
s(i) +n(i), (2)
where H is a lower triangular Toeplitz channel matrix of di-
mensions M M with rst column [h
0
h
L
0 0]
T
and
vector n(i) denotes the AWGN. Here, we assume perfect syn-
chronization with the transmitted sequence and carrier recov-
ery.
The received block of symbols r(i) in (2) is a linear in-
stantaneous mixture of the information symbols s(i). Con-
volutive mixtures (i.e., IBI between consecutive blocks) are
avoided thanks to the inserted zero symbols guard space. The
M N mixing matrix A = H[I
N
0
NL
] is Toeplitz and
is always guaranteed to be invertible, which assures perfect
symbols recovery in absence of noise [2]. Hence, the appli-
cation of overdetermined blind source separation techniques,
to obtain an estimation of the original information symbols
and matrix A, is a straightforward idea [8]. However, since
ZP-OFDM systems usually employ a large number of sub-
carriers the separation process is slow. Furthermore, blind
separation algorithms do not provide a known arrangement or
scaling of the recovered symbols.
In order to solve these problems, we propose to use a
modied ZP-OFDM receiver structure that will simplify the
mixing model. ZP-OFDM systems may be seen as a pre-
coded CP-OFDMsystems [9],[2]. This fact relies on the prop-
erty that every circulant matrix can be diagonalized by pre-
multiplication by IFFT matrix and post-multiplication by FFT
matrix. Thanks to the zero padding, the last L columns of
matrix H do not affect the received block in (2). Therefore,
H can be replaced by a circulant matrix C
M
of dimensions
M M with rst row [h
0
0 0 h
L
h
1
] and can be diag-
onalized as follows [2],
x(i) = F
M
r(i)
= F
M
C
M
[I
N
0
NL
]
T
F
H
N
s(i) +F
M
n(i)
= [F
M
C
M
F
H
M
]F
M
[I
N
0
NL
]
T
F
H
N
s(i) +F
M
n(i)
= Vs(i) +F
M
n(i), (3)
where, = F
M
C
M
F
H
M
= diag(

MF
M
h) is a diagonal
matrix with

MF
M
h = [H(0) H(2/M) H(2(M
1)/M)]
T
on its diagonal, i.e., the channel M-length FFT. The
pre-coding M N matrix is V = F
M
[I
NN
0]
T
F
H
N
. No-
tice that (3) is similar to a CP-OFDM system where M-length
pre-coded blocks of data symbols v(i) = Vs(i) are trans-
mitted over M narrowband sub-channels. Since the channel
is of order L, matrix may have, at most, L zero-diagonal
entries. However, the remaining (at least M-L=N) nonzero
entries guarantee the recovery of the symbols in s(i).
The mixing problem in (3) is much simpler than that in
(2). The number of unknowns is reduced to the M elements
on the diagonal of , while there were M N unknown el-
ements on A. This is the distinct advantage of the model in
(3). On the other hand, the search for a diagonal demixing
matrix will provide the original arrangement of the sources.
Also, the recovered symbols will differ from the true ones by
a complex scalar only (the same for all the sources).
3. BLIND ADAPTIVE CHANNEL ESTIMATION
In this section we develop an efcient blind adaptive algo-
rithm based on Blind Source Separation (BBS) to solve the
channel identication problem. The objective of BSS is to
obtain an output vector
y(i) = [y
0
y
N
]
T
= Bx(i) = BVs(i) +BF
M
n(i),
(4)
that yield the sources s(i). Let us dene B = V

D = V
H
D,
where D is M M diagonal matrix and ()

denotes matrix
pseudoinverse. Notice that V

= V
H
, therefore y(i) yields
the sources if D =
1
. Hence, by solving the BSS problem,
we will nd the inverse of the the channel M-length FFT, i.e.,
D = diag(

h
M
), where

h
M
= [1/H(0) 1/H(2/M)
1/H(2(M 1)/M)]
T
.
Many different methods have been proposed in the liter-
ature to solve the BBS problem [6]. We consider here ap-
proaches based on Independent Component Analysis (ICA)
and gradient descent for minimizing a suitable cost function.
The natural gradient (NG) [10] represents the steepest descent
direction of the cost function dened on a Riemannian space:

L(B) = G
1
L(B), (5)
where L(B) is the conventional Euclidean gradient,

L(B)
is the natural gradient, and G
1
is the inverse of the metric
tensor. We are interested in extending this approach to the
overdetermined BSS problem in (4). First, we introduce a Lie
group structure in our parameter space and the Riemannian
metric. Then, we derive the ICA based channel estimation
algorithm.
3.1. Lie group
We construct a Lie group structure on an subset Gl

(N, M)
B Mat(N, M)[B = V

D of Gl(N, M) B
Mat(N, M)[rank(B) = min(N, M), where matrix V is a
xed known matrix, and D is a diagonal matrix. The group
multiplication and inverse operations are given by
B
1
B
2
= V

(D
1
D
2
),
B

= V

D
1
, (6)
where B
i
= V

D
i
, is the multiplication operator, and ()

is
the inverse operator. The neutral element of the group is E =
V

I
M
. The inverse operation satises B

B = BB

= E.
Now, we have to dene a Riemannian or natural metric
in our space of interest, i.e., the inner product on the tangent
space T
B
of Gl

(N, M). Let us rst dene the inner prod-


uct on T
B
at the neutral element E of Gl

(N, M), i.e., the


Euclidean metric, as
X, Y)
euc
E
= tr(XY
H
), (7)
where X, Y T
B
. Such a metric is independent on the basis
point: X, Y)
euc
E
= X, Y)
euc
B
.
Since Gl

(N, M) is a Lie Group, any W Gl

(N, M)
denes an onto-mapping: B B W. The multiplica-
tion transformation maps a tangent vector Xat Bto a tangent
vector X Wat B W. Therefore, we can dene a natural
metric on Gl

(N, M), such that the right multiplication trans-


formation is isometric, i.e., it preserves the natural metric on
Gl

(N, M). We write it as follows


X, Y)
nat
B
= X B

, Y B

)
euc
BB

= X B

, Y B

)
euc
E
= X B

, Y B

)
euc
B
(8)
3.2. Natural gradient
For a risk function l(B) dened on Gl

(N, M), the natural


gradient

l(B) is dened by [11]
X,

l(B))
nat
B
= X, l(B))
euc
B
. (9)
Using the following notation N =

l(B) = V
H
D
N
, S =
l(B) = V
H
D
S
and N = SG
H
, by applying (8) we may
derive the diagonal matrix G
H
:
X,

l(B))
nat
B
= X, N)
nat
B
= X, l(B)G
H
)
nat
B
= X, SG
H
)
nat
B
= X B

, SG
H
B

)
euc
B
, (10)
where
X B

, SG
H
B

)
euc
B
= tr((X B

)(SG
H
B

)
H
)
= tr(V
H
(D
X
D
1
)(V
H
(D
S
G
H
D
1
))
H
)
= tr(V
H
D
X
D
1
D
H
G
1
D
H
S
(V
H
)
H
), (11)
and taking denition (9) into account it follows
X, l(B))
euc
B
= X, S)
euc
B
= tr(XS
H
)
= tr(V
H
D
X
(V
H
D
S
)
H
)
= tr(V
H
D
X
D
H
S
(V
H
)
H
). (12)
Hence, we have D
1
D
H
G
1
= I
M
, and

l(B) = l(B)G
H
= l(B)D
H
D. (13)
3.3. Adaptive solution
Our target is to make the components of y(i) in (4) as mutu-
ally independent as possible. To this end, we employ a cost
function L(B) = El(B) based on the Kullback-Liebler di-
vergence [10]:
l(B) = H(y, B) +
N

k=1
H(y
k
, B), (14)
where H(y, B) =

p(y, B) log p(y, B)dy, H(y


k
, B) =

p
k
(y
k
, B) log p
k
(y
k
, B)dy
k
, and p(y, B), p
k
(y
k
, B) are
the joint probability density function (pdf) of y and marginal
pdf of y
k
(k = 1, , M), respectively. The output signals y
are mutually independent iff L(B) = 0.
The risk function l(B) can be simplied for the overde-
termined BSS problem to [12]
l(B) = log [ det (BE
H
)[
N

k=1
log p
k
(y
k
, B), (15)
where det (BE
H
) is the determinant of matrix BE
H
, and E
is the neutral element of the Lie group Gl

(N, M). Since


B = V
H
Dis also in Gl

(N, M), then (15) yields


l(B) = log [ det (V
H
D(V
H
)
H
)[
N

k=1
log p
k
(y
k
, V
H
D).
(16)
We want to obtain the conventional gradient for this function
with respect to D. For the rst term, we have
d
dD

log [ det (V
H
D(V
H
)
H
)[

= D
H
. (17)
For the second term, let f =

N
k=1
log p
k
(y
k
, V
H
D), then
df
dD
l,l
=
N

k=1

k
(y
k
)(V
H
k,l
)

l
, (18)
where D
l,l
is the (l, l)-th entry of the diagonal matrix D,
(V
H
k,l
)

is the complex conjugate of the (k, l)-th entry of


matrix V
H
, and
k
(y
k
) =
p

k
(y
k
,V
H
D)
p
k
(y
k
,V
H
D)
y
k
|y
k
|
is a function of
the sources estimates. p

k
(y
k
, V
H
D) denotes differentiation
of p
k
(y
k
, V
H
D) with respect to y
k
. We can now write the
complex gradient of (15) with respect to Das
l(B) = D
H
+, (19)
where is the M M diagonal matrix that results from
taking the main diagonal of matrix V(y)x
H
, and (y) =
[
1
(y
1
)
N
(y
N
)]
T
. Application of (13) and (19) leads to
the natural gradient in [7]

l(B) = I +D
H
D. (20)
This natural gradient can be further simplied as follows. First,
x = Vs, neglecting the effect of the noise in (3). Also, the
outputs y are intended to be an estimation of the sources s.
And third, D
1
is an estimation of . Therefore, we can nd
taking the main diagonal of matrix V(y)y
H
V
H
D
H
,
and the term D(i)
H
in (20) equals the diagonal matrix that
results fromtaking the main diagonal of matrix V(y)y
H
V
H
.
We propose to use this natural gradient for the adaptive es-
timation of the inverse of the channels transfer function

h
M
,
i.e., the diagonal of D. The algorithm takes the following
form

h
M
(i + 1) =

h
M
(i) E1 p(i)

h
M
(i). (21)
where p(i) is a vector equal to the main diagonal of matrix
V(y(i))y
H
(i)V
H
, stands for the Hadarmard (point-wise)
product, and controls the learning rate.
In the BSS problem in (4) it is easy to show that simple
second order statistics are useful to ensure separation, there-
fore we propose to use (y(i)) = y(i). Then, each entry
of p(i) is the squared absolute value of each entry of vector
Vy(i). Also, between two consecutive steps of the algorithm,
we propose to perform a denoising process [13] by taking ad-
vantage of the fact that the channel is FIR of order at most
L. This process involves the application of the IFFT to the
estimated channels transfer function to remove the spurious
taps located beyond tap number L + 1 before switching back
to the frequency domain to continue with next step of the al-
gorithm. Finally, the proposed selection (y(i)) = y(i) in-
volves that the algorithm will only accurately estimate min-
imum phase FIR channels. If the channel is non-minimum
phase, only the absolute value of the channel will be accu-
rately blindly estimated. Pilot symbols may be used to (at
least partially) provide the algorithm with information about
the channel phase. The algorithm is able to estimate accu-
rately non-minimum phase channel if it has some initial infor-
mation about the phase. Exploring our method more deeply
and improving the performance for non-minimumphase chan-
nels are major directions for future work.
0 500 1000 1500 2000
25
20
15
10
5
0
Number of symbols
R
e
l
a
t
i
v
e

c
h
a
n
n
e
l

e
s
t
i
m
a
t
i
o
n

e
r
r
o
r

p
o
w
e
r

(
d
B
)
NG
Pilots
SVD
RLS
Fig. 2. Performance of the proposed NG based algorithm,
RLS [4], direct SVD and the estimated channel using the rst
2 blocks as pilots. SNR=15 dB. No fading.
4. EXPERIMENTAL RESULTS
We present now several simulation examples. We consider a
4-QAM transmitter in the practical context of HIPERLAN/2
(HL2), a multicarrier wireless system operating in the 5-GHz
band with N = 64 carriers and a zero padding of length
L = 16. In HL2, each frame contains 500 blocks, and among
them the rst two are pilots. The channel selected complies
with channel model A in HL2 [14] with high frequency se-
lectivity. In addition to the proposed natural gradient (NG)
based adaptive channel estimation, we also simulate the RLS
and direct SVD approaches, as in [4]. Once the estimated
channel impulse response

h is available, the complex scalar
ambiguity is solved by = min

|h

h|
2
. For the pro-
posed algorithm we select a normalized value = 1/(p
H
p).
The following results are an average of 100 simulation runs.
In every run the channel, noise and signals are generated ran-
domly, and every entry of

h
M
(0) in (21) is set to one.
Fig. 2 presents the performance of the algorithms under a
non-fading channel for SNR=15 dB. The relative channel esti-
mation error power E|h

h(i)|
2
/|h|
2
in dB is plotted
versus the number of OFDM symbols received. The parame-
ters for the RLS and direct SVD methods in [4] are selected to
assure the same steady state error for all schemes. We observe
that the proposed algorithm performs the same or slightly bet-
ter than the direct SVD method but at a much lower computa-
tional complexity. The convergence speed of the RLS method
is remarkably slower.
Depicted in Fig. 3 we present the bit error rate (BER)
of the proposed NG based algorithm, the RLS and the di-
rect SVD technique. We also include a scheme with perfect
channel knowledge and the estimated channel using the rst
5 10 15 20 25 30
10
4
10
3
10
2
10
1
10
0
SNR (dB)
B
E
R
NG
Exact channel
SVD
Pilots
RLS
Fig. 3. Averaged BERperformance of the proposed NGbased
algorithm, RLS [4], direct SVD, the estimated channel using
the rst 2 blocks as pilots and exact channel.
2 blocks as pilots. A zero forcing detector was implemented
after 1000 blocks of symbols. We observe an indistinguish-
able performance of the adaptive schemes as compared to the
one with perfect channel knowledge.
We examine now a more realistic scenario involving a
fading channel (model A, receiver speed 3 m/sec). Fig. 4
presents the performance of the algorithms for SNR=15 dB
along the HL2 frame. As we can see, the proposed NG based
algorithmis able to reduce the estimation error along the frame
with a performance comparable as the direct SVD method,
while the convergence speed of the RLS method is lower.
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25
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R
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