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Abstract
Although equalizers promise to improve the signal to noise energy ratio, zero forcing equalizers are derived
classically in a deterministic setting minimizing intersymbol interference, while minimum mean square error equalizer
solutions are derived in a stochastic context based on quadratic Wiener cost functions. In this contribution we show
that it is possible and in our opinion even simpler to derive the classical results in a purely deterministic setup,
interpreting both equalizer types as least squares solutions. This in turn allows the introduction of a simple linear
reference model for equalizers, which supports the exact derivation of a family of iterative and recursive algorithms
10
with robust behavior. The framework applies equally to multi-user transmissions and multiple-input multiple-output
11
channels. A major contribution is that due to the reference approach the adaptive equalizer problem can equivalently
12
be treated as an adaptive system identification problem for which very precise statements are possible with respect
13
to convergence, robustness and l2 stability. Robust adaptive equalizers are much more desirable as they guarantee a
14
much stronger form of stability than conventional in the mean square sense convergence. Even some blind channel
15
estimation schemes can now be included in the form of recursive algorithms and treated under this general framework.
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17
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19
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Index Terms
linear equalizers, least mean squares, minimum mean square error, zero forcing, reference modeling, blind
channel estimation, recursive adaptive filters, iterative adaptive filters, convergence, robustness, l2 stability
I. I NTRODUCTION
21
Modern digital receivers in wireless and cable-based systems are unimaginable without equalizers in some
22
form. The earliest mentions of digital equalizers were probably made in 1961 by E.Gibson [1, 2] and in 1964
23
by M.Rappeport [3]; they became popular due to the milestone work by R.W.Lucky [4, 5] in 1965 and in
24
1966, published in the Bell System Technical Journal, who also coined the expression Zero Forcing (ZF). In
25
1967 M.E.Austin introduced the Decision Feedback Equalizer (DFE) concept [6] that was soon to be thoroughly
This work has been funded by the NFN SISE project S10611-N13 (National Research Network Signal and Information Processing in
Science and Engineering).
Copyright (c) 2011 IEEE. Personal use of this material is permitted. However, permission to use this material for any other purposes must
be obtained from the IEEE by sending a request to pubs-permissions@ieee.org.
The author is with the Institute of Telecommunications at the Vienna University of Technology.
26
analyzed [7, 8]. A.Gersho proposed the Minimum Mean Square Error (MMSE) equalizer in 1969 [9] and D.M.Brady
27
introduced Fractionally Spaced Equalizers (FSE) in 1970 [10], which inspired a large amount of significant
28
work [1113]. In 1972 G.Forney included the Maximum Likelihood principle [14]. In the 80s overview papers
29
by S.U.H.Qureshi were popular [15, 16] and the major driving forces in research became blind equalization
30
methods, introduced by Y.Sato in 1975 [17] and D.N.Godard [18]. A first overview paper on blind methods was by
31
A.Benveniste and M.Goursat [19], although the breakthrough in this field came later in 1991 by L.Tang, G.Xu, and
32
T.Kailath [20, 21]. Further papers of significance are by J.M.Cioffi et al. [22, 23] as well as J.Treichler [24].
33
Extensions for Multiple-Input Multiple-Output (MIMO) transmissions start with A.Duel-Hallen [25] in 1992,
34
followed by many others, of which we cite just a few selected papers [2630]. More recent overview papers
35
36
37
In all the above literature, linear equalizers were derived either on the deterministic ZF approach for equalizers
38
of ZF type or on the stochastic MMSE approach for equalizers of the MMSE type. We present a new formulation
39
of the equalizer problem based on a weighted Least Squares (LS) approach. This deterministic view is very much
40
in line with Luckys original formulation, leaving out all signal and noise properties (up to the noise variance) but
41
at the same time offers new insight in the equalizer solutions, as they share common LS orthogonality properties.
42
This novel LS interpretation allows a very general formulation to cover a multitude of equalizer problems, including
43
44
45
In practice the equalizer problem is not yet solved once the solution is known, as it typically involves a matrix
46
inversion, a mathematical operation that is highly complex and challenging in low-cost fixed-point devices. Adaptive
47
algorithms are thus commonly employed to approximate the results. Such adaptive algorithms for equalization
48
purposes come in two flavors, an iterative (also off-line or batch process) approach as well as a recursive approach
49
(also on-line or data-driven process) that readjusts its estimates on each new data element that is being observed.
50
Both approaches have their benefits and drawbacks. If channel estimation has been performed in a previous step
51
(for various reasons), then the iterative algorithm based on the channels impulse response may be most effective.
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On the other hand, it is not required to compute first the channels impulse response if only the equalizer solution
53
is of interest. In particular in time-variant scenarios one may not have the chance to continuously estimate the
54
channel and compute equalizer solutions iteratively and therefore, a recursive solution that is able to track changes,
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56
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However, such adaptive algorithms require a deep understanding of their properties as selecting their only free
58
parameter, the step-size, turns out to be crucial. While for decades adaptive filter designers were highly satisfied
59
when they could prove convergence in the mean-square sense, more and more situations now become known, in
60
which this approach has proved to be insufficient, since despite the convergence in the mean square sense, worst
61
case sequences exist that cause the algorithm to diverge. This observation started with Feintuchs adaptive IIR
62
algorithm [36, 37] and the class of adaptive filters with a linear filter in the error path [38, 39] but has recently
63
arisen in other adaptive filters [40, 41] as well as in adaptive equalizers [42]. A robust algorithm design [43, 44], on
64
the other hand, is much more suited to solving the equalization problem as it can guarantee the adaptive algorithm
65
will not diverge in every case. In this contribution we therefore show how to design robust, adaptive filters for
66
linear equalizers.
67
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Table I provides an overview of the most common variables and their dimensions. The paper is organized as
69
follows. In Section II common transmission models in wireless systems are introduced along with their optimal
70
linear equalizer solutions. In Section III we propose a least squares linear reference model for ZF and a weighted
71
LS model for MMSE that allows the interpretation of equalizers as a standard system identification problem. In
72
Section IV we discuss iterative forms (i.e., algorithms that contain all required data and iterate to find an optimal
73
solution) and in Section V recursive forms (i.e., algorithms that take on new data for each adaptation step) of
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adaptive equalizers, as well as blind channel estimators for equalization, some of them well known in the literature,
75
others entirely new. Due to the novel reference approach, we are able to design them as robust algorithms and
76
provide convergence conditions. In Section VI we present selected Matlab experiments to validate our findings.
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78
79
Throughout this paper we assume that the individual transmit signal elements sk have unit energy, E[|sk |2 ] = 1,
80
and the noise variance at the receiver is given by E[|vk |2 ] = No . We are considering several similar but distinct
81
transmission schemes:
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83
Variable
Dimension
Comment
SNT M 1
SNT 1
S1
RNR 1
RNR SNT M
RNR SNT
RNR S
IRS1
et
NT 1
et,m
IRSNT 1
e,t,m
S1
C
l
RNR 1
ZF LS solution
MMSE
f,t,m
C
l
RNR 1
MMSE LS solution
fk
C
l
RNR 1
equalizer estimate
C
l
RNR 1
sk
C
l
sk,m
C
l
sk,t,m
C
l
rk
H
Hm
C
l
C
l
C
l
Ht,m
C
l
IR
IR
ZF
f,t,m
fk
ZF
v,t,m
C
l
SNT M 1
ZF modelling error
MMSE
v,t,m
C
l
SNT M 1
ZF
vk,t,m
C
l
ZF compound noise
MMSE
vk,t,m
C
l
84
(1)
85
Here, the vector sk = [sk , sk1 , ..., skS+1 ]T consists of the current and S 1 past symbols according to the
86
span L < S of the channel H, which is typically of Toeplitz form as shown in (2). The received vector is
87
defined as rk = [rk , rk1 , ..., rkR+1 ]T . Let the transmission be disturbed by additive noise vk , being of the
88
89
same dimension as rk .
rk
rk1
..
rkR+1
h0 h1 . . . hL1
.. ..
..
=
.
.
.
h0
h1
. . . hL1
RS
sk
sk1
..
.
skS+1
vk
vk1
..
.
vkR+1
(2)
90
on the received signal rk so that f H rk is an estimate of sk = eT sk for some delayed version of sk . A unit
91
vector e = [0, ..., 0, 1, 0, ..0]T with a single one at position {0, 1, ..., S 1} facilitates the description.
92
93
The transmissions follow the same form as described in (1), although with a different meaning as we transmit
94
over NT antennas and receive by NR . Such Multiple-Input Multiple-Output systems are generally referred
95
to as MIMO systems. The transmit vector sk = [s1,k , s2,k , ..., sNT ,k ]T is of dimension 1 NT , the channel
96
matrix H C
l
97
Here, NT is the number of transmit antennas, whilst NR denotes the number of receive antennas. As in the
98
previous case, we assume each entry of the transmit vector sk to have unit power. Unlike the earlier situation,
99
100
solution). For NR = NT both solutions coincide. In order to detect the various entries of the transmit vector
101
sk , we again employ a unit vector et : eTt sk = st,k . Note however that in contrast to the previous channel
102
model, a set of NT different vectors et ; t = 1, 2, .., NT will be employed in order to select all NT transmitted
103
symbols while in the frequency selective SISO case a single vector e is sufficient. Early work on linear
104
MIMO equalization can be found in [29, 35]. Note that precoding matrices are often applied in particular
105
in modern cellular systems such as HSDPA and LTE. In this case the concatenation of the precoding matrix
106
and the wireless channel has to be considered as a new compound channel. Such precoding matrices can also
107
have an impact on the dimension of the transmit vector sk as in many cases fewer symbols than rank(H) are
108
transmitted at each time instant k . A particular form of this is given when the precoding matrix shrinks to a
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vector, in which case we talk about beamforming where only one symbol stream is transmitted.
110
111
NR NT
, and thus the receive vector and the additive noise vector are of dimension 1 NR .
M
X
Hm sk,m + vk = Hsk + vk .
(3)
m=1
112
The channel matrices Hm ; m = 1, 2, .., M hereby denote matrices of dimension NR NT , describing flat
113
frequency MIMO transmissions. As before a unit vector is required to select the desired transmit symbol but
114
now we also have to select the user whose symbols we want to detect, thus a set of vectors et,m ; t = 1, 2, .., NT
115
is required for each user m = 1, 2, .., M : sk,t,m = eTt,m sk , where sk denotes a concatenated vector over all
116
M transmit vectors sk,m and the compound channel matrix H = [H1 , H2 , ..., HM ].
117
118
The previous concepts can be combined to model more complicated scenarios. Consider for example MU
119
transmissions over frequency selective MIMO channels as the most elaborate case. To simplify matters we
120
assume that all corresponding SISO channels require a memory of L taps and that all transmitters are
121
H2,1,1
H1,1,1
H2,2,1
H1,2,1
H=
.
..
H1,NR ,1 H2,NR ,1
...
HNT ,1,1
H1,1,2 ...
HNT ,1,M
...
HNT ,2,1
H1,2,2 ...
HNT ,2,M
,
(4)
122
where the indexes in Ht,r,m indicate the frequency selective SISO channel (of Toeplitz structure) of user m
123
from transmit antenna t to receive antenna r. Consequently, the compound channel matrix H C
l
124
NT X
M
X
Ht,r,m sk,t,m + vk ,
RNR SNT M
(5)
t=1 m=1
125
sk being a concatenated transmit vector, comprised of all (SISO) transmit vectors sk,t,m for each transmit
126
antenna t and user m at time instant k . Concatenating all NR receive vectors in a single vector rk , we obtain
127
the general transmission model rk = Hsk + vk , the channel being the compound matrix as defined above.
128
As before we can envision a unit vector e,t,m ; {0, 1, ..., S 1} that selects the transmit symbol at delay
129
lag , originating from transmit antenna t; t = 1, 2, .., NT of user m; m = 1, 2, .., M : sk,t,m = eT,t,m sk .
130
131
Although an SIR or SINR formulation is very intuitive for designing transmission systems, equalizers are usually
132
not derived in the context of maximizing SINR but in minimizing a mean square error (referred to as Minimum
133
MSE=MMSE) instead. The relation between MMSE equalizers and maximizing SINR is well known, see for
134
example [45, 46] and is being used successfully in recent research [47, 48]. We briefly re-derive it here in a
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136
137
To understand the vast amount of research and information available on this subject, one has to ask the question
138
What is the purpose of an equalizer?. While Luckys original work focused on the SU scenario, attempting
139
a minimax approach to combat Inter-Symbol-Interference (ISI), today we typically view the equalizer in terms
140
of Signal-to-Interference Ratio (SIR) or Signal-to-Interference and Noise Ratio (SINR). If a signal, say sk , is
141
transmitted through a frequency selective channel, a mixture of ISI, additive noise and signals from other users,
142
Multi-User Interference (MUI), is received. If signals are transmitted by multiple antennas, then additional so-called
143
spatial ISI (SP-ISI) is introduced. The ratio of the received signal power Ps and all disturbance terms before an
144
Ps
.
PISI + PSPISI + PMUI + No
(6)
145
The task of the equalizer is to improve the situation, i.e., to increase this ratio. A linear filter applied to the
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observed signal can for example result in an increased Ps0 > Ps , utilizing useful parts of PISI and PSPISI , while
147
0 < P
0
0
the remaining PISI
ISI and/or PSPISI < PSPISI and/or PMUI < PMUI is decreased. Unfortunately, the noise
148
power N0 as well as its power spectral density is in general also changed when an equalizer filter is applied. At
149
best it can be considered possible to achieve the post-equalization SINR (the index ae denotes after equalization):
SINRae =
Ps + PISI + PSPISI
Ps0
,
max
SINRbe = 0
0
0
0
PISI + PSPISI + PMUI + No
No
Equalizer strategy
(7)
150
where the equalizer manages to treat the ISI and SP-ISI as useful signal whilst at the same time eliminating the
151
MUI (for example by successive interference cancellation). The ratio of SINRbe to the eventually achieved SINRae
152
is considered as the equalizer gain. The purpose of this paper is to develop a unified view on the SINR and SNR
153
relation to the MMSE and ZF equalizer, which permits the introduction of a simple linear reference model as well
154
as its application in an adaptive system identification framework. In correspondence with the four transmission
155
scenarios described above, there also exist four related linear equalizers that will be discussed below.
156
157
We start with the simplest case: a single user scenario. Now consider the original equalizer problem of
158
maximizing the signal to interference ratio in the absence of noise in (6). By applying a linear filter f we
159
160
obtain one part that is proportional to the signal of interest, eT sk = sk and a remaining ISI part:
T H
2
e H f
PS
f H He eT HH f
max SIR = max
= max
=
max
f
f
f
f
PI
f H H (I e eT ) HH f
f H HHH f keT HH f k22
(8)
or, alternatively if the noise is also included to maximize the signal to interference-plus-noise ratio
max SINR = max
f
PS
f H He eT HH f
= max H
.
f
PI + N0
f [H (I e eT ) HH + N0 I] f
(9)
161
Here, PS and PI denote the signal and interference power (ISI only in the case of for SISO), respectively,
162
while I denotes the identity matrix. The introduction of our unit vector e now takes on a new interpretation.
163
Not only does it select the symbol we wish to detect from the (concatenated) transmit vector eT sk = sk ,
164
but it also selects the sub-channels He in H that carry relevant information for the equalizer. Note that both
165
terms, SIR as well as SINR, can uniquely be mapped into the following problem
SIR
PS
= max
= SIR 1
f
PS + PI
SIR + 1
(10)
SINR
PS
= max
= SINR 1.
f
PS + PI + No
SINR + 1
(11)
max
f
166
and correspondingly
max
f
167
As both provide a monotone mapping from SIR to SIR or SINR to SINR , we can equivalently maximize
168
these new terms and obtain (here shown on the example of SINR):
max SINR = max
f
169
f H He eT HH f
f H [HHH + No I] f
(12)
the solution of which is given by the eigenvector to the largest eigenvalue of the numerator term He eT HH :
fmax SINR = (HHH + No I)1 He = fMMSE ,
(13)
170
which is obviously proportional to the MMSE solution. More precisely: any vector f max SINR from (13)
171
maximizes the SINR but only one solution of this is also the MMSE solution, thus a better way to put it is
172
that the MMSE solution also maximizes SINR (but not the opposite).
173
174
175
f H He eT HH f
f H HHH f
(14)
(15)
176
Analog to the MMSE case, we can now state that any vector f max SIR maximizes the SIR but only one
177
178
179
180
181
(16)
182
183
The ZF and MMSE equalizer for this transmission scenario can be derived straightforwardly following the
184
previous concept. We thus briefly present the results. For the overdetermined case (NT > NR ) the ZF solution
185
186
(17)
(18)
187
188
solutions by o. We omit the index if the context makes it clear what is meant or if they coincide, as
189
190
191
(19)
192
Formally the solutions appear in the same form as before but note that in the SISO channel, could be
193
selected to maximize SINR, while now we have no such liberty and have to accept the SINR that comes out
194
195
196
We now show that this method can easily be extended to an MU scenario, where we have M users transmitting
197
simultaneously, as explained in (3). The linear equalizer that maximizes the SINR of the signal from transmit
198
199
f H Hm et,m eTt,m HH
mf
hP
i
M
H +N I f
fH
H
H
m
o
m
m=1
(20)
M
X
!1
Hm HH
m + No I
MUMMSE
Hm et,m = ft,m
,
(21)
m=1
200
which is proportional to the solution of the MMSE MU equalizer. Knowing the individual channels Hm ; m =
201
1, ..., M involved, it is thus straightforward to find the linear solution to optimize the SINR in an MU
202
scenario. The solution is typically overdetermined as long as NR < M NT . The corresponding ZF solution,
203
maximizing SIR, is obtained by omitting No . Correspondingly, an MU setup that maximizes SIR can be
10
204
computed simply by setting No to zero in the above solution. Note further that similar concepts occur for
205
equalizers in CDMA systems, where due to the signatures of the various users a rich interference is introduced
206
207
208
The linear equalizer that maximizes the SINR of the signal at time lag from transmit antenna t of user m,
209
210
211
(22)
(23)
NT X
M
X
!1
Ht0 ,m HH
t0 ,m
+ No I
MUMMSE
.
Ht,m e,t,m = f,t,m
(24)
t0 =1 m=1
212
In the case of the ZF solution we have to be careful. Depending on the actual numbers of antennas and users,
213
both situations can appear: if SNT M > RNR we have the overdetermined case, i.e., we simply omit No I in
214
(24), while for SNT M < RNR the underdetermined solution has to be taken:
!1
NT X
M
X
max SIR,u
MUZF,u
H
= Ht,m
f,t,m
Ht0 ,m Ht0 ,m
.
e,t,m = f,t,m
(25)
t =1 m=1
215
For the MMSE solution both are again identical. Such equalizers have also been referred to as interference
216
aware equalizers in the context of UMTS-HSDPA transmissions [5052], as they also take other users into
217
account.
218
For all such equalizer solutions (MMSE and ZF) corresponding views exist, interpreting the equalizer as the
219
linear solution that maximizes SINR or SIR. As long as we have a general transmission model rk = Hsk + vk ,
220
we always find that the MMSE solution also maximizes SINR, or alternatively, the ZF solution maximizes SIR.
221
III. A
222
While classical literature views the equalizer problem as minimizing a mean square error, we show in the following
223
section that this is in fact not required and a purely deterministic approach based on a least squares modeling is
224
possible. This approach in turn leads to the novel interpretation of the adaptive equalizer problem in terms of a
225
classic system identification problem. For such problems, however, a much stronger l2 stability and robustness has
226
been derived in the past to ensure convergence of the adaptive algorithms under worst case conditions. In order to
11
227
apply such robust techniques, we first have to show the equivalent system identification approach for equalizers.
228
We start with the ZF equalizer and then continue with its MMSE counterpart.
229
A. ZF Equalizer
230
(26)
231
with e,t,m indicating a unit vector with a single one entry at position , for transmit antenna t of user m, thus
232
eT,t,m sk = sk,t,m , the transmit signal at antenna t of user m that will be decoded at delay lag . Note that this form
233
of derivation requires no signal or noise information, focusing instead only on properties of linear time-invariant
234
systems of finite length (FIR); it thus entirely ignores the presence of noise. This is identical to Luckys original
235
formulations [4], where system properties were the focus and the particular case of NT = 1, M = 1 was considered.
236
237
238
If RNR < SNT M (for example in Luckys SISO frequency selective scenario, we have R < S ) the solution to
this problem is obviously given by
ZF,o
= (HHH )1 He,t,m ,
f,t,m
(27)
239
commonly known as the ZF solution. Note that this is a so-called overdetermined LS solution as we have more
240
241
242
(28)
243
244
245
246
Let us first consider the overdetermined case of Eqn. (27). As ISI does not vanish for finite length vectors, we
propose the following reference model for ZF equalizers
ZF,o
ZF,o
e,t,m = HH f,t,m
+ v,t,m
(29)
ZF,o
v,t,m
= I HH (HHH )1 H e,t,m .
(30)
247
ZF,o
The term v,t,m
models ISI, SP-ISI and MUI. The larger the equalizer length RNR , the smaller the ISI, e.g.,
248
ZF,o 2
measured in kv,t,m
k2 . The cursor position also influences the result.
249
12
Fig. 1.
The overdetermined ZF equalizer can be interpreted as a linear combination from two complementary subspaces, the equalizer in
the range of HH and the modeling error from its orthogonal complement.
250
251
Due to its projection properties we find that the outcome of the reference model lies in the range of HH with
ZF,o
an additive term v,t,m
from its orthogonal complement:
ZF,o
= 0,
Hv,t,m
(31)
ZF,o
ZF,o H
= 0,
Hv,t,m
f,t,m
(32)
ZF,o H
ZF,o H ZF,o
e,t,m = eT,t,m I HH (HHH )1 H e,t,m .
v,t,m = v,t,m
v,t,m
(33)
252
This last term can be interpreted as the energy of the modeling error but equally describes the remaining ISI power
253
0 + P0
0
PISI
SPISI + PMUI . Figure 1 illustrates the ZF equalizer reference model.
254
255
What changes for the underdetermined solution, i.e., if RNR > SNT M ? Such situations may occur if we simply
256
have two (NR = 2) sensors (antennas) for observations on an SU SIMO channel, resulting in a stacked matrix
257
H
HH = [HH
1 , H2 ]. In this case we find that
ZF,u
e,t,m = HH f,t,m
,
(34)
258
ZF,u
i.e., the equalizer removes all ISI, without introducing any additional modeling noise, thus v,t,m
= 0. Such
259
situations typically occur in SU situations (M = 1), either in frequency selective SIMO transmissions when the
260
number of observations R > S or in MIMO transmissions if NR > NT . In such cases the remaining ISI becomes
13
261
zero.
262
263
ZF,o
How does a received signal look after such ZF-equalization? We apply f,t,m
on the observation vector and obtain
ZF,o H
ZF,o H
ZF,o
f,t,m
rk = sk,t,m v,t,m
sk + f ZF,o H vk = sk,t,m + vk,t,m
.
264
265
(35)
Let us now consider the underdetermined case, i.e., RNR > SNT M . We then find that the ISI components
ZF,u
become zero, i.e., v,t,m
= 0. The outcome of the ZF equalizer now reads
ZF,u H
f,t,m
rk = sk,t,m + f ZF,u H vk .
(36)
266
While in the overdetermined case some ISI remains, the underdetermined case only displays the additive noise
267
term. Both relations serve as ZF reference model, as we apply them to adaptive algorithms further ahead. Note
268
ZF
that both ISI and additive noise are often treated as a compound noise vk,t,m
as indicated in (35). Summarizing the
269
270
271
ZF
Theorem 3.1: The LS solution f,t,m
from (27) for the overdetermined and (28) for the underdetermined case,
272
respectively, applied to the observation vector rk , defines a linear reference model in which the desired output signal
273
is sk,t,m , originating from a transmitted signal over antenna t of user m, and corrupted by additive compound
274
ZF . The compound noise is defined by f ZF H v and has an additional component in the overdetermined
noise vk,t,m
,t,m k
275
ZF,o H
ZF,o
case, i.e., the modeling noise v,t,m
sk , defined by the modeling error vector v,t,m
in (30).
276
B. MMSE Equalizer
277
MMSE solutions are typically derived [31] on the basis of signal and noise statistics, e.g., by
MMSE
f,t,m
= arg min E |f H rk sk,t,m |2 .
(37)
278
(38)
279
with an additional term according to the additive noise variance No . We consider here white noise; alternative forms
280
with colored noise, as originating for example from fractionally spaced equalizers, are straightforward; one only
281
14
282
283
This formulation (38) now reveals that the MMSE problem equivalently can be written as a weighted LS problem
1
with the weighting matrix HHH + No I 2 . Note that the last term of (38)
MMSE = eT,t,m I HH (HHH + No I)1 H e,t,m
(39)
284
defines the minimum mean square error. As the term is independent of f , it can thus be dropped when minimizing
285
286
(40)
Similarly to the ZF equalizer, an overdetermined solution for RNR < SNT M also exists here:
MMSE,o
f,t,m
= H(HH H + No I)1 e,t,m .
(41)
287
MMSE,o
MMSE , which is very different to the ZF
= f,t,m
Under white noise both solutions are in fact identical f,t,m
288
equalizer (for the proof see Appendix A). The same procedure can be applied to DFE or FSE structures, the noise
289
terms, however, are then partially zero or correlated and care needs to be taken when treating such cases. The
290
equivalence (13) to the maximum SINR solution, however, is preserved. Correspondingly to the reference model
291
for ZF equalizers in (29), we can now also define a reference model for MMSE equalizers
292
MMSE
MMSE
e,t,m = HH f,t,m
+ v,t,m
(42)
MMSE
v,t,m
= I HH (HHH + No I)1 H e,t,m .
(43)
293
Note however, that unlike in the case of the ZF solution the modeling error is not orthogonal to the MMSE
294
MMSE H HH f MMSE 6= 0. MMSE equalizers are typically designed on the basis of observations rather
solution, i.e., v,t,m
,t,m
295
than system parameters. Multiplying the signal vector with e,t,m we obtain
MMSE H
MMSE H
eT,t,m sk = sk,t,m = f,t,m
Hsk + v,t,m
sk .
296
297
and obtain
MMSE H
MMSE H
MMSE H
MMSE
f,t,m
rk = sk,t,m v,t,m
sk + f,t,m
vk = sk,t,m + vk,t,m
.
298
299
(44)
(45)
From classic equalizer theory it is well known that the remaining ISI energy of the ZF equalizer is smaller than
ZF
MMSE (see Appendix B)
that of the MMSE part. This can be shown by comparing the energies of v,t,m
and v,t,m
ZF
MMSE 2
kv,t,m
k22 kv,t,m
k2
(46)
15
300
while the remaining ISI and noise energy is smaller for MMSE:
ZF
ZF
MMSE 2
MMSE 2
kv,t,m
k22 + No kf,t,m
k22 kv,t,m
k2 + No kf,t,m
k2
301
(47)
Let us summarize the most important findings of this section in the following statement:
302
303
304
reference model in which the desired output signal is sk,t,m , originating from a transmitted signal over antenna t
305
306
307
308
In conclusion, the adaptive equalizer problem has thus taken on the form of an identification problem as depicted
309
ref
in Figure 2. The linear system with impulse response f,t,m
is estimated as f,t,m by an adaptive equalizer algorithm.
310
Here, ref stands for either MMSE or ZF. The outcome of the reference system is disturbed by the compound noise
311
ref
ref
vk,t,m
(see (45)) and constructs a noisy reference symbol sk,t,m . The adaptive filter with its output sk,t,m +
vk,t,m
312
313
ref
sk,t,m + vk,t,m
rk
- h
6
- f ref
,t,m
- ek,t,m
-
f,t,m
Fig. 2.
314
ref
sk,t,m + vk,t,m
315
Equalizer solutions requiring matrix inverses are highly complex and numerically challenging, in particular when
316
the matrix size is 50 or over [53]. In this section we apply the reference model description obtained in the previous
317
section to iterative algorithms, as they offer the possibility of implementing the equalizer in a robust manner, being
16
318
particularly suited to fixed-point implementations and furthermore avoid the need for an explicit matrix inversion.
319
An iterative algorithm, as referred to here, is one that possesses all data and attempts to achieve an optimal solution.
320
In the literature such algorithms are also referred to at times as off-line or batch algorithms since they require no
321
new data during their operation. In this contribution we show convergence conditions for numerous known and
322
novel algorithms but do not deal with the question of when to stop the iterations.
323
324
The design processes for iterative algorithms follow a common path. A channel H is applied in order to obtain
325
the desired equalizer estimate. Due to our ZF and MMSE reference models in (29) and (42), respectively, we can
326
327
matrix X (which in the simplest case of ZF is the channel matrix H) and our equalizer estimate. In the following,
328
we omit the indexes for the various solutions in order to simplify the description. An iterative gradient algorithm
329
; l = 1, 2, ...,
(48)
330
where the matrix B is considered to be positive definite and Hermitian. Given the set {X, B, e}, the algorithm
331
332
333
334
335
Theorem 4.1: The iterative-algorithm updating Equation (48) converges in the parameter error vector to zero,
fk = f ref if both the following conditions hold:
i.e., limk
1) The absolute value of all eigenvalues of I XBXH is smaller than one, or correspondingly
0<<
336
337
2
.
max {XBXH }
(49)
; l = 1, 2, ...,
(50)
I XBXH fl1 XB
v.
338
Thus, as long as the step-size guarantees that all eigenvalues of I XBXH are smaller than one in magnitude,
339
340
17
341
342
We show in the following examples that both conditions are satisfied for all ZF and MMSE type iterative equalizer
algorithms.
343
344
345
fl =
fl1 + H(e,t,m HH
fl1 )
346
; l = 1, 2, ....
(51)
ZF
With the reference model in (29) we can introduce the parameter error vector
fl = f,t,m
fl and obtain
ZF
fl = (I HHH )
fl1 Hv,t,m
.
(52)
347
ZF
ZF
We recognize that the noise condition is satisfied, as property Hv,t,m
= 0 for v
= v,t,m
from (31) holds
348
independent if we have an over- or underdetermined equalizer. Convergence conditions for the step-size are now
349
also readily derived, being dependent on the largest eigenvalue of HHH , i.e.,
0<<
2
.
max {HHH }
(53)
350
As computing the largest eigenvalue may be a computationally expensive task, simpler bounds are of interest, even
351
352
353
2
trace(HHH )
(54)
and can be computed with low complexity once the matrix H is known.
354
2) For an SU in a frequency selective SISO channel, the channel H is defined by a single Toeplitz matrix,
355
the largest eigenvalue of which can also be bounded by max |H(ej )|, with H(ej ) denoting the Fourier
356
transform of the channel impulse response. The corresponding condition for the step-size reads now
0<<
2
.
max |H(ej )|2
(55)
357
Such a step-size may be more conservative than the condition (53) but is also more practical to find. In the
358
simulation examples presented in Section VI the bound so obtained is very close to the theoretical value (53).
359
Note that for a SIMO case, i.e., NT = 1 but NR receive antennas, a similar bound can be found:
0<<
360
361
max
2
PNR
r=1 |Hr (e
j )|2
(56)
18
362
363
365
exhibits much slower convergence for some channels than for others, even for optimal step-sizes. The analysis of
1
the algorithm shows that the optimal matrix B that ensures fastest convergence is given by B = HHH
, which
366
is exactly the inverse whose computation we are attempting to avoid with the iterative approach. If, however, some
364
367
368
369
370
a-priori knowledge is present on the channel class (e.g., Pedestrian B or Vehicular A), then we can precompute the
mean value over an ensemble of channels from a specific class, for example E [HHH ]1 = R1
HH . In this case
the algorithm updates read:
H
fl =
fl1 + R1
HH H(e,t,m H fl1 )
; l = 1, 2, ...
(57)
fl =
fl1 + HR1
HH (e,t,m H fl1 )
; l = 1, 2, ...
(58)
371
when RNR > SNT M with the corresponding step-size bound in (49). An example of how this can be applied in
372
373
374
fk =
fk1 + X(He,t,m (HHH + No I)
fk1 )
(59)
375
MMSE
H MMSE
MMSE is the desired solution. Recalling (42), i.e., e
for which f ref = f,t,m
,t,m = H f,t,m + v,t,m we identify the
376
fk1 ] No f,t,m
+ HvMMSE
,t,m .
(60)
377
MMSE
MMSE = I HH (HHH + N I)1 H e
MMSE
Recalling with (43) that v,t,m
o
,t,m we find that No f,t,m + Hv,t,m = 0.
378
fk =
fk1 X(HHH + No I)
fk1 .
379
(61)
If we select X = I, we can identify B = HHH + No I and we find as a condition for convergence that
0<<
2
.
max {HHH + No I}
(62)
380
Alternatively, a speed-up algorithm is possible with X = (RHH + No I)1 as an approximation or even better with
381
19
382
383
In contrast to iterative algorithms, recursive algorithms try to improve their estimation in the presence of new
384
data that has not been previously applied. Such algorithms are sometimes referred to as on-line algorithms as they
385
require new data for each adaptation step. Due to this property they typically behave adaptively as well. In addition
386
to their learning behavior, recursive algorithms also have a tracking behavior that allows them to follow channel
387
changes, a feature that makes them very interesting for many applications. Furthermore, there is typically no need
388
for the channel matrix H to be known for such algorithms, offering a solution of very low complexity.
389
390
As previously in the case of iterative algorithms we start with the construction of an error signal. By applying
391
the observations Xk (which can be observation vectors rk or composite matrices of several of them) on the linear
392
393
394
A recursive algorithm thus continuously takes on new data values. Its update equations take on the form
fk = fk1 + Yk sk XH fk1
k
h
i
ref
fk1 + v
= fk1 + Yk XH
f
k
k
395
; k = 1, 2, ...,
(63)
396
The analysis strategy is thus to reformulate the updates (63) in terms of the parameter error vector
fk1 =
397
k to guarantee l2 stability
f ref fk1 and find conditions on the step-size and the modeling noise sequence v
398
of the recursive form. We refer to results from the standard literature [43, 44], repeated here for convenience of
399
the reader, only bringing the algorithms into the required formulation. Note that l2 stability is a much stronger
400
form than the mean square sense convergence that is mostly associated to adaptive filters. This form of stability
401
includes the desired property that there should be no data or noise sequences (or initial values) to cause divergence
402
of the algorithm. Algorithms that behave in this way are thus called robust. In practice the differences may not be
403
so obvious, as sequences that cause divergence may not often occur. On the other hand, a systematic search for
404
them can reveal that a trusted algorithm is in fact by no means not robust [40].
405
406
407
Theorem 5.1: Consider Update (63) in which both Xk = rk and Yk = Brk are vectors with a constant positive
k = vk rk .
definite Hermitian matrix B and noise v
1) The algorithm is called robust and l2 stable for step-sizes 0 < < maxk
1
rH
k Brk
20
408
409
410
411
412
413
2
).
rH
k Brk
P
2) Furthermore, if the noise vk is l2 bounded, i.e.,
k |
vk |2 < Sv < , the undistorted a priori estimation
error k ea,k = k rH
k fk1 becomes zero [43, 44].
3) If additionally the regression vector rk is of persistent excitation, the parameter error vector also becomes
zero limk fk1 = 0 [43, 44].
414
4) If the compound noise vk is not l2 bounded but has bounded variance, for a normalized step-size =
415
k r H
k Brk , we obtain convergence in the mean square sense for 0 < < 2 and the steady-state misadjustment
416
v2
2
[42].
417
There are even extensions towards non symmetric-matrices B [42, 54], which are not considered here. At first sight
418
these conditions may appear very similar to those mentioned previously for iterative algorithms; note, however, that
419
they are distinctively different as they involve observed data rather than estimated channels.
420
421
422
The first truly recursive algorithm we present here is the well-known LMS algorithm for equalization (see for
423
example Rappaport [32] for the SU SISO case). We identify here B = I. Starting with an initial value f0
H
fk = fk1 + k rk [s
k,t,m rk fk1 ]
; k = 1, 2, ...
(64)
MMSE
MMSE
= fk1 + k rk [rH
k,t,m
].
k (f,t,m fk1 ) + v
424
The desired l2 stability is achieved as long as the step-size condition is satisfied. According to [43, 44] for a
425
426
2
.
krk k22
(65)
In this case it makes sense to apply a time-variant step-size k with step-size condition
0 < k <
2
,
krk k22
(66)
427
which can easily be satisfied for example by a normalized step-size. Due to the linear MMSE reference model
428
we can now relate the adaptive equalizer algorithm to the well-known l2 stability of adaptive filters [43, 44]; for
429
430
431
432
vk , the compound noise sequence will not be bounded. We can thus not expect the recursive, adaptive MMSE
21
433
equalizer (without sophisticated step-size control) to converge to zero. If additive noise vk is absent, only the
434
underdetermined ZF solution can converge to its corresponding reference. In this case MMSE and ZF solutions are
435
identical.
436
Otherwise if additive noise remains unbounded but is of finite variance, a steady-state mismatch larger than
437
zero remains. Following Property 4 of Theorem 5.1, with a normalized step-size k = /krk k22 , we obtain a
438
misadjustment of
MMMSE =
MMSE 2
MMSE 2
kv,t,m
k2 + No kf,t,m
k2 .
2
(67)
439
In [42] simulation results are presented that validate such relations. Note that we can follow the same arguments as
440
before, by simply replacing the reference model with the ZF model. We then obtain the same stability conditions
441
ZF
ZF
kv,t,m
k22 + No kf,t,m
k22 .
2
(68)
442
We can thus not truly argue that the recursive algorithm leads to the MMSE or the ZF solution as there is a
443
considerable misadjustment for both algorithms. However, the distance to the two solutions in the mean square
444
sense is smaller for MMSE than for ZF (see Appendix B). As in the case of previous algorithms, a speed-up variant
445
446
447
448
Consider the following potential update for a recursive ZF algorithm (starting with an initial estimate f0 ) in which
449
we replace the regression vector rk of the previous recursive MMSE algorithm with Hsk :
H
fk = fk1 + k Hsk [s
k,t,m rk fk1 ]
; k = 1, 2, ...
(69)
ZF
ZF
450
The last line reveals that additional noise occurs, proportional to the parameter error vector. Algorithms of such
451
structure have been analyzed in [40, 42] and it has been shown that they exhibit convergence in the mean-square
452
sense with sufficiently small step-size, but also that they are not robust since unbounded noise sequences (unbounded
453
energy but with bounded variance) can be found that cause the algorithm to diverge even for very small step-sizes.
454
We thus conclude that in this set-up of recursive algorithms, there is no known robust form that truly achieves a
455
22
456
457
The situation is more difficult if Xk and Yk are truly matrices. In this case the theoretical results from the
458
literature are not directly applicable and thus need to be extended first. Such scenarios occur if the update is not
459
performed at every time instant but blocked, say after every T instances. Applied to the equalizer problem we can
460
461
462
463
464
465
466
467
Theorem 5.2: Consider the recursive block algorithm with update (63) for which Yk = Xk Bk with a positive
definite Hermitian matrix Bk .
1
1) The algorithm is called robust and l2 stable if the step-size can be selected so that 0
H/2 H
1/2 .
max (Bk X
k Xk Bk )
P
H/2
H/2
k k2 < Sv < , then
k k is bounded, i.e., K
2) Furthermore, if the noise sequence k kBk v
k=1 k kBk v
3) If additionally a persistent excitation condition is satisfied for k Xk , we also can conclude that the parameter
468
469
Iterative/recursive block MMSE algorithm: Consider a set of receiver vectors rk obtained over a time period
470
T . Stacking the corresponding receiver vectors rk in a matrix Rk = [rk , rk1 , ..., rkT +1 ] and correspondingly the
471
transmitted symbols Sk = [sk , sk1 , ..., skT +1 ], we can reformulate the previous transmission model in block form
472
to Rk = HSk + Vk in which we also stacked the noise vectors accordingly into Vk . We only have to assume that
473
the channel H remains constant during the period T . Applying (45) in matrix form, we obtain the reference model
MMSE H
MMSE H
MMSE H
MMSE
k,t,m
f,t,m
Rk = sk,t,m v,t,m
Sk + f,t,m
Vk = sk,t,m + v
.
T 1
MMSE C
k,t,m
is now in vector form, as is the compound noise v
l
474
475
each time instant k it is possible to first derive an iterative algorithm, starting with the initial value f0 :
fl =
fl1 + k Rk sk,t,m RH
f
k l1
476
; l = 1, 2....
(70)
T 1
. For
(71)
k,t,m
fl = I k Rk RH
k fl1 + k Rk v
; l = 1, 2, ....
(72)
477
Convergence conditions can now be derived on the basis of the largest eigenvalue of matrix Rk RH
k :
478
0 < k <
479
MMSE 6= 0. If we envision an ensemble of random data and noise sets S and V , convergence
f = [Rk RH ]1 Rk v
k,t,m
k
k
k
480
2
.
max {Rk RH
k }
Note, however, that due to the noise components in Rk a final error remains:
23
481
The entire algorithm remains an iterative algorithm and not a recursive method as long as k remains fixed. As
482
soon as k changes, the recursive aspect enters the algorithm. Note, however, that the convergence condition then
483
484
Based on the previous analysis, convergence in the mean sense remains. Also note that such algorithms are quite
485
popular, often appearing under the name of data-reuse adaptive filters [5560].
486
487
Recursive single-user block MMSE algorithm: Rather than applying an iterative method (or mixed itera-
488
tive/recursive method), we can reformulate the algorithm in terms of a purely recursive procedure in which new
489
data are being processed at every time instant k at a block basis of period T :
fk =
fk1 + k Rk sk,t,m RH
f
k k1
; k = 1, 2....
(73)
490
Note that the algorithm now tries to improve the solution at every time instant k under varying conditions
491
{Rk , sk,t,m }. To guarantee l2 stability, the step-size k needs to be changed over time:
0 < k <
1
.
max {Rk RH
k }
(74)
492
493
Theorem 5.2 we identify Yk = Xk = Rk and we can conclude robustness and l2 stabilty. Similar to the fast
494
convergent ZF algorithm, a fast convergent MMSE algorithm is also possible. However, it requires pre-computation
495
496
of the inverse of Rk RH
k at every time instant k , which is prohibitive from a complexity point of view. Alternatively,
1 can be employed, i.e., Y = R B
if the channel class is a priori known, a fixed mean value B = E [Rk RH
k
k
k ]
497
498
499
All iterative equalizer methods discussed so far require the knowledge of channel matrix H. The reference model
500
also facilitates deriving robust algorithms for channel estimation. While these are well known algorithms for system
501
identification once a training sequence is employed, it is of greater interest to derive algorithms when such training
502
sequences are entirely missing. In such situations we call this a blind algorithm. Let us first consider an iterative
24
503
algorithm that employs two SISO channels, say H1 and H2 , the cost function for which is given by
hB = arg
min
h=
h2
H
2
H1 h2 HH
2 h1
2
h1
2
H
= arg min
HH
1 , H2 h 2
h
2
= arg min
HH h
2 .
(75)
504
If h1 is proportional to the impulse response in channel Toeplitz matrix H1 and h2 proportional to H2 , then we
505
H
find HH
1 h2 = H2 h1 and so the cost function becomes zero and thus minimal. If we can guarantee R > S we can
506
0 as:
derive an iterative algorithm, starting at initial value h
l1 )
l = h
l1 H(HH h
h
; l = 1, 2, ...,
(76)
507
508
509
algorithm solving an LS problem for which the reference is zero, thus explaining the minus sign in (76). Utilizing
510
the compound channel H, step-size bounds for convergence of the algorithm can be derived as before. As it stands,
511
the algorithm does not yet make any sense, as the channel would need to be known in order to blindly estimate it.
512
A true blind algorithm does not know the channel matrices and has to operate on the received vectors, i.e.,
(1)T
(2)T
2
max {HHH } .
One can also view the blind algorithm above as a standard iterative
(1)H
513
rTk = [rk
514
0:
can thus formulate a recursive algorithm for blind channel estimation, starting with initial value h
, rk
k = h
k1 k Rk RH h
h
k k1
; k = 1, 2....
(2)H
, Rk
]. We
(77)
k = h
k1 + k Rk (RH hB RH h
h
k
k k1 )
H B
k1 k Rk (RH h
= h
k k1 Vk h ),
(1)H
(2)H
(78)
515
516
convergence some other conditions need to be satisfied: persistent excitation of sk , channels that are co-prime and
517
bounded noise [61]. The classic blind channel estimation can thus be straightforwardly formulated as a robust,
518
recursive algorithm as an alternative to the highly-complex SVD methods originally proposed in [20, 21], adaptive
519
, Vk
25
520
521
In order to corroborate our theoretical findings, we present selected Matlab examples, in which we elaborate
522
previous experimental settings, provided in [42], based on a set of seven channel impulse responses of finite length:
1
[k ] = 0.9k ; k = 0, 1, 2..M 1
1 0.9q 1
1
[k ] = 0.8k ; k = 0, 1, 2..M 1
1 0.8q 1
1
[k ]; k = 0, 1, .., M 1
1
1q +0.5q 2
1
[k ]; k = 0, 1, .., M 1
1
11.4q +0.6q 2
(1)
(2)
(3)
(4)
(5)
= k + 0.6k2
(83)
(6)
= k + k2
(84)
(7)
= k + k1 ,
(85)
hk
hk
hk
hk
hk
hk
hk
(79)
(80)
(81)
(82)
523
where we used the Kronecker delta k to describe a unit impulse and the delay operator q 1 [xk ] = xk1 . In [42]
524
the MMSE equalizer algorithm (64) was demonstrated to run on channel h(4) . An excellent agreement with the
525
526
527
Iterative ZF equalizer: In our first example we select the length of the channel to be M = 50 for which even the
528
first four impulse responses have decayed considerably. If we run an iterative receiver (also of 50 taps) according
529
to (51), the result for h(3) is depicted on the left-hand side of Figure 3, with f0 denoting the ZF solution and fl
530
denoting its estimate. Based on the convergence condition (53), it is possible to compute the exact step-size bound
531
(0.255), given the channel matrix H. Also shown in the figure are the conservative bound (54), which is the smallest
532
step-size (0.017) in the figure, resulting in the slowest convergence speed and (55), which is just a fraction smaller
533
(0.25 vs. 0.255) than the step-size bound. The average inverse autocorrelation R1
HH is computed over all seven
534
channels (79)-(85), and applied in the algorithms updates. This results in a considerable speed-up in the iterations
535
536
537
Non-robust alternative recursive ZF algorithm: In the second example we consider the recursive algorithm in
538
(69). We again apply channel h(3) for M = 50 taps and compare the outcome with the ZF solution (also of 50
539
taps). Additive noise was N0 = 0.01 and QPSK for data transmission over the channel. Figure 4 displays the
540
obtained relative system mismatch for various normalized step-sizes , i.e., = kHs1k k2 . On the left we display
2
26
iterative ZF
with speed up
.2
60
=
0
=0.017
=0.0
51
10
0.1
4
-20
=
0.
=0.255 128
-30
5
0.3
=
7
=0.
-10
fig1.m
=
0.9
iterative ZF
50
iteration l
50
Fig. 3.
0.2
-40
100
50
iteration l
100
Left: Convergence behavior of the iterative ZF algorithm (51), Right: Iterative ZF algorithm with speed-up variant (57).
541
the behavior for random data sequences. As expected for normalized step-sizes 0 < < 2 we observe convergence
542
in the mean square sense. On the right of the figure we show the behavior when searching for the worst case data
543
sequence. We observe that even for very small step-sizes, the algorithm diverges and thus behaves non-robustly as
544
predicted.
545
546
Blind channel estimation and equalizer: In the final example we simulated a blind channel estimation according
547
to (77). We selected two channels, channel hk and hk from the list above, but shortened them to five elements
548
(M = 5). We ran the experiment with additive noise No = 0.01 and averaged over 50 runs. The left-hand side of
549
Figure 5 depicts the results when using only the instanteneous regression vector rk while the right-hand side of
550
the figure applies a block mode with T = 5. As a figure of merit we compute the closeness of the two estimated
551
(3)
2|r12 |2 =
(4)
2
2
|hH
|hH
1 h2 |
2 h1 |
+
.
2 k2 kh2 k2 kh
1 k2
kh1 k22 kh
2
2
2
(86)
552
In Figure 5 the distance of the correlation coefficient to one, i.e., 1 |r1,2 |2 is depicted. In both cases the step-sizes
553
554
krk k22
27
2
=1.9
-10
=1.5
=1
-20
2000
unstable
for all
=0.5 1000
-30
=0.1
=0.1
-40
=0.5
25000
step k
50000
25000
step k
50000
Fig. 4. Left: Convergence behavior of the recursive ZF algorithm (69) under random data; Right: Convergence behavior under worst case
data sequences.
VII. C ONCLUSION
555
556
We have started with well-known SIR and SINR formulations for linear equalizers and shown their connection
557
to equivalent ZF and MMSE formulations. Due to an LS approach it is now possible to derive the classical
558
equalizer types with an alternative formulation, an LS formulation for ZF and a weighted LS formulation for MMSE
559
equalizers. This in turn resulted in a linear reference model for both. Based on such a linear reference model, it is
560
possible to derive iterative as well as recursive forms of equalizers that are robust. Conditions for their robustness
561
were presented, and in particular ranges for their only free parameter, the step-size, were presented to guarantee
562
robust learning. The reader may be disappointed not to encounter the classical algorithm from Lucky [4, 5]. As this
563
recursive algorithm is not l2 stable, it does not fit into our robustness framework and has been treated elsewhere
564
565
566
567
ACKNOWLEDGMENTS
The author wishes to thank the anonymous reviewers for their helpful comments, Sebastian Caban for preparing
high quality figures and Nick Shaw for his language corrections.
28
1-|r12|2
100
=0.1
=1.9
10-1
=1.5
10-2
=1.9
=1.5
=1
=0.5
=1
=0.5
10-3
Fig. 5.
2000
step k
4000
2000
step k
4000
Left: Convergence behavior of recursive channel estimator (77) with instantaneous observations (T = 1); Right: Convergence
A PPENDIX
568
569
570
A. Appendix A
We want to show that both MMSE solutions
MMSE
f,t,m
= (HHH + No I)1 He,t,m = H(HH H + No I)1 e,t,m
(87)
571
are equivalent. Although the reader can find the result also in the literature [64], we repeat it here for convenience.
572
573
(88)
(89)
= A1 C I [I + CH A1 C]1 CH A1 C
= A1 C[I + CH A1 C]1 .
574
(90)
29
575
576
B. Appendix B
577
We find
ZF,u 2
kv,t,m
k2 = 0,
(91)
ZF,o 2
kv,t,m
k2 = eT,t,m [I HH [HHH ]1 H]e,t,m ,
(92)
ZF,u 2
kf,t,m
k2 = eT,t,m [HH H]1 e,t,m ,
(93)
ZF,o 2
kf,t,m
k2 = eT,t,m HH [HHH ]2 He,t,m ,
(94)
MMSE 2
kv,t,m
k2 = eT,t,m [I HH [HHH + No I]1 H]2 e,t,m
{z
}
|
(95)
MMSE 2
kf,t,m
k2
2
eT,t,m HH Q He,t,m
(96)
578
resulting in
ZF,u 2
ZF,u 2
k2 = No eT,t,m [HH H]1 e,t,m ,
kv,t,m
k2 + No kf,t,m
(97)
1
ZF,o 2
ZF,o 2
k2 = eT,t,m [I HH [HHH ]1 Q
k2 + No kf,t,m
kv,t,m
[HHH ]1 H]e,t,m ,
(98)
MMSE 2
MMSE 2
kv,t,m
k2 + No kf,t,m
k2 = eT,t,m [I HH QH]e,t,m = eT,t,m [I HH HQ]e,t,m ,
579
580
(99)
(100)
581
582
ZF,u 2
obviously true. As for the underdetermined case kv,t,m
k2 = 0, the difference is also negative.
583
The notation > 0 for a matrix means that the matrix is positive definite.
(101)
30
584
As [(HH H)1 Q] > 0, we can conclude that the compound noise from the underdetermined ZF solution is larger
585
586
(102)
587
588
overdetermined ZF solution also results in an MSE larger than that of the MMSE solution.
589
C. Appendix C
590
591
(HH H)1 (HH H)1 > (HH H + No I)1 = Q we can conclude that the
1/2
H/2
substitute Xk Bk
fk = fk1 k Xk B XH fk1 + v
; k = 1, 2, ...,
k
k
k X
H fk1 + BH/2 v
= fk1 k X
.
k
k
k
592
593
as long as 0
594
from 0 to K:
H/2
max (Bk
1
HX
k B1/2 ) .
X
k
k
||fK+1 ||2 +
597
598
K
X
601
602
603
604
605
K
X
H/2
k ||Bk
k ||2 ,
v
(105)
k=1
P
H/2
k ||2 <
which already guarantees robustness and thus l2 stability. If additionally the noise is bounded, i.e., K
k=1 k ||Bk v
H fk1 ,
Sv < , we can conclude convergence of k ||ea,k || 0 as it must be a Cauchy sequence. Since ea,k = X
k
we require a persistent excitation condition for k Xk in order to conclude that the parameter error vector fk1 0
for k ||ea,k || 0.
R EFERENCES
599
600
(104)
Summing up the terms, we find the global relation over a time horizon
k=1
596
(103)
595
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34
2001 he is a full professor for Digital Signal Processing in Mobile Communications at the Vienna University of Technology where he founded
the Christian-Doppler Laboratory for Design Methodology of Signal Processing Algorithms in 2002 at the Institute for Communications and
RF Engineering (now institute of telecommunications). He served as Dean from 2005-2007. He was associate editor of IEEE Transactions
on Signal Processing from 2002-2005, is currently associate editor of JASP EURASIP Journal of Advances in Signal Processing, and JES
EURASIP Journal on Embedded Systems. He is elected AdCom member of EURASIP since 2004 and serving as president of EURASIP
from 2009-2010. He authored and co-authored more than 400 papers and patents on adaptive filtering, wireless communications, and rapid
prototyping, as well as automatic design methods.