Digital Signal Processing 20 (2010) 736742

Contents lists available at ScienceDirect

Digital Signal Processing
www.elsevier.com/locate/dsp
Blind identication of multichannel systems driven by impulsive signals
Wanzhi Qiu

, Syed Khusro Saleem, Minh Pham

National ICT Australia, Department of Electrical and Electronic Engineering, The University of Melbourne, Parkville, Victoria 3010, Australia
a r t i c l e i n f o a b s t r a c t
Article history:
Available online 6 August 2009
Keywords:
Blind system identication
Impulsive signal
Multichannel system
Sparse representation
L
1
-norm optimization
We study the problem of estimating the impulse responses of multiple FIR channels driven
by an unknown impulsive signal. A deterministic method which deals with arbitrarily
short excitation signals is proposed by utilizing subspace properties of the data covariance
matrix. A variant of the proposed method is also presented which exploits the L
1
-norm
convex optimization to improve the estimation accuracy when the impulse responses of the
system to be identied are sparse. The proposed methods have the potential to improve the
performances of communication, exploration seismology and mechanical signature analysis
systems.
2009 Elsevier Inc. All rights reserved.
1. Introduction
Blind system identication allows recovery of systems impulse response without knowing its input, and has applications
in communications, seismic analysis and image restoration [1].
Methods based on second order statistics (SOS) have been proposed (see [24], for example) to blindly identify a mul-
tichannel system (up to a constant scalar) under some mild conditions. One major advantage of SOS-based methods, as
compared with methods based on higher-order statistics, is the ability to identify the system using much shorter input
sequences. Meanwhile, methods using deterministic models have also been proposed [5,6]. While making use of the SOS
implicitly, the deterministic approach does not impose any statistic restrictions on input signals (e.g. whiteness or station-
ariness) and has been shown to be more ecient in dealing with short input signals. However, the methods in [5,6] are
developed primarily for communication systems where long sequences of input symbols are assumed and are not applicable
when the input signal is shorter than the impulse response of the system to be identied.
The ability to accomplish blind system identication with extremely short excitation signals are essential in certain
practical applications since these input signals are impulsive in nature wherein. For example, in mechanical signature anal-
ysis [7], an impact signal is created by colliding the material under test with a known object. The output signal ltered
by the material is analyzed for diagnostic and monitoring purposes. In exploration seismology, a charge of dynamite is
exploded in the earth and a geophone is used to receive the reected and/or diffracted signal [8]. This signal is used to
estimate the reection coecients, which are associated with the impulse responses of the various layers in the earth and
unravel its physical characteristics. In both applications, the input signals are not exactly known; nor can they be treated
as impulses of unit duration. Even in communications where continuous input sequences are normally assumed, there are
cases when short, burst-like packets (e.g. acknowledgments and certain request/control commands) are used [9].
In all the above-mentioned applications the excitation signals injected to the unknown systems are impulsive in nature
and do not exhibit any periodicity or statistics. As a result, no statistical properties on the input signals can be assumed. On
the other hand, the entire system outputs associated with full duration of the input signal should be utilized in identifying
*
Corresponding author. Fax: +61 3 93481682.
E-mail addresses: wanzhi.qiu@nicta.com.au (W. Qiu), khusro.saleem@nicta.com.au (S.K. Saleem), minh.pham@nicta.com.au (M. Pham).
1051-2004/$ see front matter 2009 Elsevier Inc. All rights reserved.
doi:10.1016/j.dsp.2009.08.004
W. Qiu et al. / Digital Signal Processing 20 (2010) 736742 737
the system since they become available due to the impulsive nature of the excitation signal. This paper complements the
work in [26] by proposing a deterministic method, in the sense that the input is treated as a deterministic signal, for blind
identication of multichannel systems driven by an impulsive signal of arbitrarily short length. Successful application of the
proposed method in these systems could lead to improved performances.
It has become well known that most of systems in real applications actually have sparse representations, i.e., with a
large portion of their coecients equal to zero [10].
For example, it has been shown that communication channels exhibit great sparseness. In particular, high-denition
television (HDTV) channels [11] and the hilly terrain (HT) delay prole in broad-band wireless communications are very
sparse [12]. Underwater acoustic channels also exhibit sparseness [13]. This has led to the recent advance in compressed
sensing (CS) techniques. One of the key ndings of CS is that L
1
-norm optimization promotes sparsity. That is, as compared
with its popular L
2
-norm counterpart, L
1
-norm optimization produces better results for sparse representations.
A variant of the proposed method will also be presented which exploits the L
1
-norm convex optimization to improve
the estimation accuracy when the system to be identied has sparse impulse responses.
This paper is organized as follows. Section 2 presents the data model. Section 3 develops the proposed method. Section 4
proposes the variant for systems with sparse coecients. Section 5 presents simulation results and Section 6 concludes the
paper.
2. The data model
The following notational conventions will be used. A bold lower case (a) and capital ( A) represent a column vector and
matrix, respectively. A
T
and A
H
denote transpose and complex-conjugate transpose of A, respectively. a denotes estimate
of a.
Consider the following multichannel FIR system (for clarity purposes, noise is omitted in the equations and will be dealt
with in the simulation):
y
i
(k) =
L
i

j=0
h
i
( j)s(k j), i = 1, . . . , M, (1)
where {h
i
(k), k = 0, 1, . . . , L
i
1} and {y
i
(k), k = 0, 1, . . . , L
i
+ N} are the impulse response and output of the i-th channel,
respectively, and {s(k), k = 0, 1, . . . , N} is the common input to the M channels.
Let d = N + L, where L = max{L
1
, L
2
, . . . , L
M
}, the output can be uniquely represented by the (d +1) 1 vector:
y
i
=
_
y
i
(0) y
i
(1) . . . y
i
(L
i
+ N) 0 . . . 0
_
T
, i = 1, . . . , M. (2)
The channel coecients can be similarly represented by (L +1) 1 vectors and the input signal by a (N +1) 1 vector:
h
i
=
_
h
i
(0) h
i
(1) . . . h
i
(L
i
) 0 . . . 0
_
T
, i = 1, . . . , M, (3)
s =
_
s(0) s(1) . . . s(N)
_
T
. (4)
By dening an operator Toepliz() which forms a (W + 1) (D + W + 1) Toeplitz matrix from the (D + 1) 1 vector
a = [a(0) a(1) . . . a(D)]
T
, i.e.,
Toepliz(a, D, W) =

a(0) a(1) . . . a(D)

a(0) a(1) . . . a(D)
.
.
.
.
.
.
. . .
.
.
.
a(0) a(1) . . . a(D)

(W +1) rows
we can rewrite (1) into a matrix form:
Y
i
= H
i
S, i = 1, . . . , M, (5)
where
Y
i
= Toepliz( y
i
, d, d),
H
i
= Toepliz(h
i
, L, d),
S = Toepliz(s, N, d + L). (6)
The purpose of blind system identication is to nd {h
i
} from {Y
i
}.
As in [26], it is assumed that the maximal channel order L is known and the channel polynomials {H
i
(z)} do not share
any common roots, where
H
i
(z) =h
i
(0) +h
i
(1)z + +h
i
(L
i
)z
L
i
, i = 1, . . . , M.
Note that, unlike in [24], no assumption about the statistics of the input signal is imposed. The input signal is virtually
arbitrary as long as it is not an all-zero sequence. In addition, the restrictions on the length of the input as in [5,6] are not
required since the data model (5) caters for input signals of arbitrary lengths.
738 W. Qiu et al. / Digital Signal Processing 20 (2010) 736742
3. The proposed method (L
2
method)
Dene the operator Stack() which stacks up M matrices of the same size (A
1
, A
2
, . . . , A
M
) into a new matrix, i.e.,
Stack(A
1
, . . . , A
M
) =

A
1
.
.
.
A
M

.
We now apply subspace principles to nd the following vector of the system impulse responses:
h = Stack(h
1
, h
2
, . . . , h
M
).
From (5), we have
Y = HS,
where Y and H are two Sylvester matrices given by
H = Stack(H
1
, H
2
, . . . , H
M
),
Y = Stack(Y
1
, Y
2
, . . . , Y
M
). (7)
We then form a covariance matrix,
R = Y Y
H
= HS S
H
H
H
(8)
and compute its eigendecomposition:
R = (U
r
U
o
)
_

o
_
(U
r
U
o
)
H
(9)
where
r
contains r non-zero eigenvalues, U
r
and U
o
contain the principal and null eigenvectors, respectively. Note that
when noise is absent, the diagonal matrix
o
= (
r+1
, . . . ,
M(d+1)
) in (9) is an all-zero matrix. Otherwise, its diagonal
entries will be nonzero positive numbers. Although it is impossible here to accurately estimate the noise variance, these
diagonal entries do indicate the noise level and can be represented by
=
1
M(d +1) r

_
M(d+1)

i=r+1

i
. (10)
It has been shown in [2] that H has full column rank since the polynomials {H
i
(z), i = 1, . . . , M} are co-prime and it has
more rows (M(d +1)) than columns (d + L +1). In addition, one can easily verify that, due to the Sylvester structure of S,
the matrix S S
H
is of full rank as long as s is not an all-zero vector. Therefore, r = rank(H) = d + L + 1, and (8)(9) imply
that the columns of H and the columns of U
r
= [u
r+1
u
r+2
. . . u
M(d+1)
] span the same subspace. Due to the orthogonality
between U
r
and U
o
, we have the following relationship:
U
H
o
H =0. (11)
It is known [3] that such an equation involving the full rank generalized Sylvester matrix H and null-space U
o
has a unique
(up to a constant scalar) solution for H (or equivalently h). We now solve it by transforming it to
M(d+1)

i=r+1
_
_
u
H
i
H
_
_
2
2
=0,
which is equivalent to
M(d+1)

i=r+1
_
_
G
H
i
h
_
_
2
2
=0,
where
G
i
= Stack(G
i1
, G
i2
, . . . , G
i M
), i = r +1, . . . , M(d +1), (12)
G
ik
= Toepliz(g
ik
, d, L), k = 1, 2, . . . , M, (13)
g
ik
=
_
u
i
_
(k 1)(d +1)
_
u
i
_
(k 1)(d +1) +1
_
. . . u
i
_
k(d +1) 1
__
T
. (14)
This leads to
h
H
Q h =0,
W. Qiu et al. / Digital Signal Processing 20 (2010) 736742 739
where
Q =
M(d+1)

i=r+1
G
i
G
H
i
. (15)
This means that, under the constraint h
2
= 1, h is the eigenvector of Q corresponding to the zero eigenvalue. When noise
exists, h can be identied (up to a constant scalar) by the least eigenvector of Q . Our proposed method (Proposed_L
2
) for
general (i.e., not necessarily sparse) systems is now summarized as follows:
Step 1. Form the covariance matrix R = Y Y
H
using (6)(7).
Step 2. Compute U
o
the null space of R, form matrix Q using (12)(15), and estimate h (up to a constant scalar) by the
eigenvector of Q corresponding to the smallest eigenvalue.
4. The variant for sparse systems (L
1
method)
In the previous section, the approach to solve (11) is the L
2
-norm optimization which minimizes the quadratic function
x
H
Q x and, therefore, has an ecient closed-form solution. When h is sparse (i.e., most of its coecients are zeros), the
L
2
-norm optimization will almost never nd a sparse solution; producing instead a non-sparse solution with many nonzero
elements [14]. In this situation, the L
1
-norm optimization which has been proven [10,14] to be a powerful tool in producing
sparse solutions can be applied.
Unfortunately, (11) cannot be directly used in L
1
-norm optimization since this will lead to trivial (i.e., all-zero) solutions.
Constraining the solution to h
2
= 1 does not help since this will create a non-convex L
1
-norm optimization problem
and, therefore, convergence is not guaranteed. The challenge here is to build a linear relationship which avoids trivial L
1
solutions between the unknown (h) and the available data { y
i
}. In fact, it can be shown that h and the data vector
y = Stack( y
1
, y
2
, . . . , y
M
)
are related by a matrix dened by the input s dened in (4). This is because that (5) can be re-written as
y
i
= Sh
i
, i = 1, 2, . . . , M,
where
S =
_
Toepliz(s, N, L)
_
T
. (16)
Therefore, we have
y = Sh (17)
where S is a block diagonal matrix dened by S as
S =

S 0 0 . . . 0
0 S 0 . . . 0
.
.
.
.
.
.
0 0 0 S

. (18)
Now we need to have an estimate of s so that S can be constructed. Going back to the original data model (1), it is obvious
that the polynomial dened by the input signal s:
S(z) = s(0) + s(1)z + + s(N)z
N
is actually the greatest common division (GCD) of the polynomials dened by the system outputs { y
i
}:
Y
i
(z) = y
i
(0) + y
i
(1)z + + y
i
(L
i
+ N)z
L
i
+N
, i = 1, . . . , M.
Therefore, S(z) (or equivalently s) can be found from {Y
i
(z)} using a GCD estimation algorithm. We will use the algorithm
in [15] for this purpose.
Since a GCD scaled by any nonzero constant is also a GCD, s can only be estimated in this way up to a constant scalar.
With the estimated s, the reformulated data model (17) allows h to be obtained (also up to a constant scalar) by L
1
-
minimization. Note that this is strict convex optimization and convergence is guaranteed. The minimization generally takes
the following form [16]:

h = argmin
x
_
x
1
+ y Sx
2
2
_
(19)
where balances the sparsity of the solution with the delity to the data, and should be inversely proportional to the noise
level. That is, the more accurate the data is, the bigger weight it should have in the optimization objective function. The
value of needs to be tuned for each particular application and some research towards determining analytically [17] is
currently going on. In the proposed method we set = / where is a positive constant and is given by (10).
We are now ready to summarize the Proposed_L
1
method for identifying systems with sparse impulse responses:
740 W. Qiu et al. / Digital Signal Processing 20 (2010) 736742
Fig. 1. NRMSE vs. SNR, non-sparse systems.
Step 1. Use the method in [15] to estimate s.
Step 2. Form matrix S using estimated s according to (16) and (18), and obtain the estimate of h (up to a constant scalar)
through L
1
-norm minimization (19).
5. Simulation
We evaluate the performances of the proposed methods against that of the LS method in [5]. Although the LS method
was primarily developed for continuous input sequences, it can also be applied for impulsive inputs certifying certain length
constraints. Note that here we are not intending to have a comprehensive performance comparison between the proposed
methods with the LS method since they are targeting different applications. What we would like to conrm is that our
proposed methods provide meaningful results and they are capable of dealing with arbitrarily short input signals.
In the simulation, the input consists of QPSK symbols and the three-ray multi-path channel system with M = 4 and
L = 5 in [5, Table III], is used and modied as follows to generate channels with desired sparsity. The sparsity parameter K
is introduced to specify the number of nonzero elements in each channel vector h
i
. In order to keep the order of channels
to be L, h
i
(0) and h
i
(L) are kept unchanged. The other K 2 nonzero indexes are uniformly chosen from {1, . . . , L 1} and
the remaining channel coecients are set to zero. This process is done independently for each channel.
For each test, one realization of the input s and channels with desired sparsity is generated and 100 runs are then
conducted where, in each run, an independent noise vector w
i
is added to the output y
i
(i = 1, . . . , M), with w
i
containing
i.i.d. Gaussian variables with zero mean and variance
2
i
.
The SNR for each channel is dened as
SNR
i
= 10log
10
y
i

2
2
(d +1)
2
i
, i = 1, . . . , M,
and chosen to be the same for all the channels. The normalized root-mean-square-error for the j-th test is measured by
NRMSE
j
=
1
h
2

_
1
100
100

i=1
_
_
h
o
i
h
_
_
2
2
where h
o
i
denotes the normalized estimate in the i-th run. The normalization is to remove the effect of the arbitrary
constant scalar on the error evaluation and is done by xing one element of the estimate

h
i
to the true value. That is,
h
o
i
=

h
i
(h(l)/

h
i
(l)), where h(l) = 0, l {0, 1, . . . , M(L +1)}. The nal NRMSE is the averaged NRMSE
j
over 50 independent
tests. In the simulation, = 0.1 is xed for all the scenarios and the convex programming package from [18] is used in
solving (19).
Figs. 1 and 2 show the cases when the system is not sparse (K = 6). In Fig. 1 the length of input signal is xed to N = 15
while SNR varies from 25 dB to 45 dB. The results show how the three methods improve while SNR increases. Also shown
W. Qiu et al. / Digital Signal Processing 20 (2010) 736742 741
Fig. 2. NRMSE vs. N, non-sparse systems.
Fig. 3. NRMSE vs. sparsity.
here is that, in this non-sparse situation, the performances of the proposed methods (Proposed_L
2
and Proposed_L
1
) are
barely distinguishable.
Fig. 2 shows the case when SNR = 30 dB while N varies from 1 to 35. The LS method is not applicable when N < L +1. It
can be seen that the proposed methods deal with short inputs well and their performances remain relatively constant when
N increases. The LS method is shown to improve signicantly while N increases and outperform the proposed methods
when N 20 (i.e., N 4L). Once again, Proposed_L
2
and Proposed_L
1
are shown to perform very closely for non-sparse
systems.
Fig. 3 shows the case when SNR = 30 dB and N = 15 while K varies from 2 to 6. It can be seen that when h is
relatively sparse (i.e., smaller K), Proposed_L
1
(proposed specically for sparse systems) performs better than Proposed_L
2
.
When the system impulse responses become dense (i.e., K is close to L + 1), the two methods give virtually identical
performances.
742 W. Qiu et al. / Digital Signal Processing 20 (2010) 736742
6. Conclusion
We have proposed a deterministic method for blind identication of multichannel systems. A variant of the proposed
method is also presented which utilizes the L
1
-norm convex optimization to improve the estimation accuracy when the
system to be identied has sparse impulse responses. The capability of the proposed methods of dealing with impulsive
input signals of arbitrarily short lengths has been demonstrated. The proposed methods, therefore, have the potential to
improve the performances of communication, exploration seismology and mechanical signature analysis systems.
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Wanzhi Qiu received his B.E. and M.E. from University of Electronic Science and Technology of China and Ph.D. from the University
of Melbourne, all in Electrical and Electronic Engineering. He then held senior research and engineering positions with Motorola, NEC
and Bandspeed, working on GSM, WCDMA, Bluetooth, ADSL and 3G mobile projects. He is currently a senior researcher with National
ICT Australia (NICTA). His research has primarily been in the areas of blind system identication, wireless sensor networks and signal
processing.
Syed Khusro Saleem received his B.S.E.E. from Northwestern University in 1992. He received his M.Eng.Sc. and Ph.D. in 1994 and
1997, respectively, both in Electrical and Electronic Engineering from the University of Melbourne, Australia. Dr. Saleem has had various
roles ranging from Research Scientist to Project Leader at the Australian Defence Science & Technology Organisation, NEC Australia and
Bandspeed. He is currently employed by National ICT Australia Ltd as a Project Leader where he oversees the Water Information Net-
works research program. Dr. Saleems interests range from software engineering and digital design to large scale system engineering and
management.
Minh Pham received the Bachelor and the Ph.D. degrees in Electrical Engineering from the University of Melbourne in 1994 and 2000,
respectively. He had worked in industry and academia organizations including the Australias Commonwealth Scientic and Industrial
Research Organisation (CSIRO). He was a research fellow at the University of Lancaster in the United Kingdom and the University of
Melbourne. He is presently a senior researcher at the National ICT Australia. His current research interests are in the following areas:
sensor network technology, electromagnetic eld theory, and digital signal processing.