An Adaptive Transmultiplexer by Fast RLS Algorithms

and its Application to Multicarrier Communications

Dah-Chung Chang and Da-Long Lee
Department of Communication Engineering, National Central University
Jhongli City, Taoyuan 320, Taiwan
E-mail: dcchang@ce.ncu.edu.tw
Abstract We explore an adaptive transmultiplexer
(TMUX) technique used for multicarrier communication
systems and find the Wiener solution and adaptive
reconstruction algorithm. In order to obtain good
performance and low computational complexity, a fast
recursive least squares (RLS) algorithm is developed. By
using the polyphase decomposition method, the adaptive
receiver in a filter bank system can be formulated as a
multichannel filtering problem. Simulations show that the
proposed algorithm, called the block step-up step-down (BSUSD) algorithm, has the convergence rate close to the
standard RLS algorithm. Finally, the new adaptive TMUX
receiver is applied to the WLAN system compared to the
conventional IEEE 802.11a OFDM system to show the
superior performance for a filterbank-based muticarrier
communication system.

I. INTRODUCTION
The transmultiplexer (TMUX) is a bandwidth-efficient
communication technique that can simultaneously transmit
narrowband signals through a single wideband channel. The
conventional implementation of TMUXs used the discrete
Fourier transform (DFT) [1-2]. Since the filter bank theory
has been well developed in signal processing, the TMUX
performed by modulated filter banks has received great
attention [3-7]. The fundamental operation of TMUXs is
similar to that of frequency-division multiplexing. The
difference is that the TMUX system uses filter banks to
modulate/demodulate transmitted signals, all digitally. The
filter bank of a TMUX can be obtained from that of a
subband system. Koilpillai, Nguyen, and Vaidyanathan [6]
have shown elaborate theories to obtain a crosstalk-free
TMUX from an aliasing-free QMF bank. However, the
noise and channel distortion, which always exist in a
communication system, are not concerned.
Chen and Lin [7] applied Kalman filters to reconstruct the
transmitted signals. The drawback of this approach is that
parameters required in Kalman filters have to be known in
advance, which may be difficult in practical applications.
Ramachandran and Kabal [3] added a compensation filter,
which acts like an equalizer, in front of each separation
filter. Although their method can exactly cancel crosstalks
between different channels, residual intersymbol
interference remains. In this paper, we explore the adaptive
algorithm for the separation filter bank and find the Wiener
solution. The recursive least squares (RLS) is a well-known
adaptive algorithm with fast converges rate, but it requires
extensive computations. Although fast RLS algorithms,

x0 ( n )

x1 ( n )

f0

f1

v( m)

y( m)
+

x M1(n)

h1

x^ 0 ( n )

x^ 1 ( n )

z( m)

h0

:
:

f M 1

h M 1

^x

M1(n)

Transmission Channel

Combining Filters

Separation Filters

Figure 1. An M-band TMUX system

which can substantially reduce computations, have been

developed, they cannot be applied directly for this system.
This is due to the sampling rate conversion involved in the
TMUX system. By exploring polyphase structures, we
found that signal reconstruction in a TMUX system can be
viewed as a multichannel filtering problem. Using this
formulation, we not only simplify the derivation of the
Wiener solution, but also can apply multichannel fast RLS
algorithms. We develop our algorithm in light of the
stabilized block step-up step-down fast RLS (B-SUSD
FRLS) algorithm [8]. It is shown that the convergence rate
of the stabilized B-SUSD FRLS algorithm is close to that of
the standard RLS algorithm, and its performance
approaches to the Wiener filter.
The Orthogonal Frequency Division Multiplexing
(OFDM) technology has been widely adopted in modern
wireless communication systems, like WLAN, DVB-T, and
ultra-wideband (UWB) transmission. Although OFDM can
implement multicarreir communication systems by fast
IDFT/DFT algorithms, the spectrum sidelobe is very high so
that the subcarrier orthogonality can be seriously destroyed
due to noise, interference, imperfect synchronization and
channel estimation. The TMUX can use filter coefficients
with low sidelobe such that diminishes the influence
existing in the OFDM system. The adaptive TMUX receiver
is applied to the IEEE 802.11a system in the simulation. In
comparison with the conventional OFDM system, the
adaptive receiver effectively reduces the effect of losing
orthogonality and better performance is achieved.
II. THE MMSE RECEIVER FOR THE TRANSMULTIPLEXER
A. Polyphase Formulation

This work was supported in part by the National Science Council

of Taiwan under the contract 94-2220-E-008-003.

3247
1-4244-0355-3/06/$20.00 (c) 2006 IEEE
This full text paper was peer reviewed at the direction of IEEE Communications Society subject matter experts for publication in the IEEE ICC 2006 proceedings.

In Fig. 1, we depict an M-band TMUX system with

noise and channel distortion. The input signals are first
upsampled by M-fold expanders and interpolated by the
combining filter bank. The resultant signals are then
combined to a single channel signal, which is transmitted
over a communication channel. In the receiver, the
separation filter bank and decimators are used to reconstruct
original signals.
Let the length of the separation and combing filters be
L , the time index before upsampling be n , and that after
upsampling be m . The relation of m and n can be written
as m=Mn+l or m=Mn-l, where l=0,1,, M-1. The
reconstructed signal x i ( n ) in Fig. 1 can be expressed as

z0 ( n )

h0,i

z1 (n)

z ( m)

h1,i

xi (n)

S/P

zM 1 (n)

hM 1,i

xi ( n)

MMSE
Figure 2. The adaptive TMUX receiver structure

L 1

xi ( n ) = hi ( ) z ( Mn )

(1)

Let the channel impulse response be {c0 c1 L c N 1 }, the

where hi (n ) denotes the nth coefficient of the ith

separation filter. As we can see from (1), the input vector
for h () does not have the time-shift property. In other
words, the present input vector and the previous input
vector do not overlap L-1 elements. As a consequence, fast
RLS algorithms cannot be applied. Although we can still
apply fast RLS filtering before downsampling, it will
require M times computations. Here, we propose a structure
that can overcome this problem. Using this structure,
derivation of the Wiener solution also becomes simpler.
Let = Mk + l and K = L / M be an integer. Then,
we have

polyphase components of y (m ) be y l (n ) = y ( Mn l ) ,

=0

xi ( n ) =

where

and v (m ) is channel noise. For simplicity of representation,

we assume that P = ( N + L 1) / M
Define

v (n ) = [ v TS (n ) v TS (n 1) L v TS ( n K + 1)]TL1

where

y S (n ) = [ y 0 ( n ) y1 (n ) L y M 1 (n )]TM 1
v S (n ) = [v0 (n ) v1 (n ) L v M 1 (n )]TM 1

l =0 k =0
M 1 K 1

Then, we have the following equation

(2)
polyphase

polyphase components of z (m ) . Equation (2) can be

interpreted as a result of multichannel filtering. There are
M-channel input signals z l (n ) and M filters hli (n ),

i = 0,1,..., M 1. This leads to the reconstruction

structure depicted in Fig. 2. The device marked by MMSE
is some adaptive algorithm used for adjusting filter
coefficients as that we will develop in the next section. To
facilitate the derivation of the Wiener solution, we rearrange
hli (n ) and zl (m ), l = 0,1,..., M 1 , and express (2) as
a product of two vectors. Define
h i = [h Ti, 0 hTi,1 L h Ti ,K 1 ]TL1

(3)

z(n ) = Cy (n ) + v( n )

c0 c1 L c N 1
0 c c
L
0
1
C=
M

c0
0 K 0

0
c N 1 L 0
(7)
O
M

c1 L c N 1
We next relate y l (n ) to the input signal x (n ) . The
signal y (m ) can be expressed as follows:

y ( m) =

M 1

f (m Mk ) x (k )
i

(8)

where f i (m) denotes the mth coefficient of the ith

combining filter. Using the polyphase representation, we
can rewrite (8) as

yl (n) =
=

z S (n ) = [ z 0 (n ) z1 ( n ) L z M 1 (n )]TM 1

Thus, (2) can be written as

B. The Wiener Solution

i = 0 k =

h i ,k = [h0i (k ) h1i ( k ) L hM 1,i ( k )]TM 1

xi ( n ) = h z(n )

(6)

where

z(n ) = [ z TS ( n ) z TS ( n 1) L z TS (n K + 1)]TL1 (4)

T
i

is an integer.

y ( n ) = [ y TS ( n ) y TS (n 1) L y TS (n P + 1)]T( N + L1)1

h ( Mk + l ) z ( Mn Mk l )

components of hi (m) , and z l ( n ) = z ( Mn l ) are the

where

As we can see from Fig. 1, y (m ) is the transmitted signal

M 1 K 1

hli (k ) z l (n k )

l = 0 k =0
hli (n ) = hi ( Mn + l ) are the
=

and the polyphase components of v (m ) be v l ( Mn l ) .

M 1

f ( Mn Mk l ) x (k )
i

i =0 k =
M 1

( n k ) xi ( k )

i = 0 k =
M 1 K 1

f
i=0

(5)

li

k =0

li

( k ) xi ( n k )

3248

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(9)

f li (n ) = f i ( Mn l ) are the polyphase

components of f i (m) . To have a vector representation of

where

We can find the Wiener solution of h i by (16)-(18).

III. ADAPTIVE RECONSTRUCTION BY FRLS ALGORITHMS

(9), we first define

f l = [f

T
l ,0

T
0

T
l ,1

T
T
l , M 1 L1

T
1

T
M 1

f Lf

x S (n ) = [x (n ) x (n ) L z

(10)
T
L1

(n )]

(11)

where

fl ,i = [ f li (0) f li (1) L f li ( K 1)]TK 1

x i (n ) = [ xi (n ) xi (n 1) L xi (n K + 1)]TK 1

Then, (9) can be written as

y l (n ) = f lT x S (n )

(12)

Define an extended input vector as

x (n ) = [x TS (n ) x TS ( n 1) L x TS ( n P + 1)]TPL1(13)
Then, y (n ) can be obtained as:
(14)
y ( n ) = Fx (n )

To find the Wiener solution in (16) may require

extensive computations when the input or the environment
is time-varying. An adaptive filtering scheme will be more
useful in this case. For fast convergence, we consider the
RLS algorithm here. Using our formulation, we can apply
the multichannel fast RLS algorithm called the block stepup step-down (B-SUSD) algorithm [8]. This algorithm is a
multichannel extension of the FAEST algorithm [9]. Since
the stability problem may arise when computations are
performed in finite precision, a stabilized procedure is then
adopted. We summarize the stabilized TMUX FRLS
algorithm in Table I. Some notations are explained as
follows. The vector z L ( n ) is the input vector in (2) and
defined as

where

F=

z L (n ) = [ z T0 ( n ) z1T ( n ) L z TM 1 ( n )]TL1

F ( N + L 1)PL

where z i ( n ) = [ z i ( n ) z i ( n 1) L zi ( n K + 1)]K 1 .
By this representation, the corresponding separation filters
are

h i = [h 0T,i h1T,i L hTM 1,i ]T

T

where h l ,i = [ hli ( 0) hli (1) L hli ( K 1)]

and F = [f 0 f1 L f M 1 ]M L .
Finally, using (5), (6), and (14), we obtain
T

T
i

T
i

xi (n ) = h CFx (n ) + h v (n )

(15)

as shown in

Fig. 2. The two permutation matrices T and S are

defined such that

z L (n )
~z M (n )
T
z L+M ( n) = T
S
=

~z (n L) (19)

M
z L ( n 1)
T

The Wiener solution for the separation filters is to find the

optimal h i that minimizes the mean square error between

where

the desired signal d i (n ) and the reconstruction signal

xi ( n ) . The Wiener solution can be obtained in the

following matrix equation:

h i = Rzz1 Rzdi

(16)

where R zz is the correlation matrix of z(n ) and R zdi is the

cross-correlation matrix of z(n ) and d i (n ) . From (14) and
(6), we have that z(n ) = CFx(n ) + v( n ) . Then R zz can
be calculated by

Rzz = E{ [CFx(n ) + v (n )][CFx(n ) + v (n )]T }

(17)
= CFRxx F T C T + Rv
where E{} denotes the expectation operation, Rv is the
covariance matrix of the channel noise, and Rv = v I
2

where

is the standard deviation of the noise and I is an

L L identity matrix. The desired signal d i (n ) is the

delayed version of the corresponding input signal, i.e.,
x ( n D ) , where D accounts for the total effect of filter
banks delay and channel delay. Hence,
Rzdi = E{[Cy (n ) + v(n )] x (n D )}
= CF E{x(n ) x ( n D )}

~
z M ( n ) = [ z 0 (n ) z1 ( n ) L z M 1 (n )]T
T

Note that T = T and S = S . The main idea of the

stabilized B-SUSD FRLS algorithm is to update the Kalman
gain w L ( n ) by a blockwise step-up step-down procedure,
i.e.,

w L (n ) w L + M ( n + 1) w L ( n + 1)

(20)
The computational complexity of the stabilized TMUX
FRLS algorithm is on the order of O( 6 ML ) [8] while that
2

of the standard RLS algorithm is O( 2 L ) . Thus, when the

separation filter length is long, the stabilized TMUX FRLS
algorithm can save significant computations.
We perform some simulations to demonstrate
effectiveness of the proposed TMUX reconstruction. The
performances of the standard RLS algorithm and the
algorithm in [6] are also compared. A five band system is
used and each analysis/synthesis filter has 55 taps. The
prototype filters presented in the Table IV of [10] are
adopted. The reconstruction performance is measured by
the reconstruction signal to noise ratio (SNR). This is
defined as

(18)
3249

E{x 2 (n )}

SNR(n ) = 10 log10
2
E{[ x ( n ) x (n D )] }

This full text paper was peer reviewed at the direction of IEEE Communications Society subject matter experts for publication in the IEEE ICC 2006 proceedings.

TABLE I. THE STABILIZED TMUX FRLS ALGORITHM

35

Initials:

w L (0) = h l (0) := 0 L1 , A(0) = B (0 ) := 0 L M , M ( 0) := 1 ,

Mf (0) = Mb (0) := I M M , K i := 1, i = 1,2,3

25

reconstruction SNR, dB

Recursions:
The time updating of the Kalman gain:

e Mf ( n + 1) = ~z M ( n + 1) A T ( n) z L ( n)

Mf ( n + 1) = 1 Mf , 1 ( n )e Mf ( n + 1)
0 I M M f
Tw L + M ( n + 1) =
+
M ( n + 1)
w L ( n ) A(n )
Partition : c L ( n + 1) := Sw
L + M ( n + 1)
(n + 1)
M

(2)

(3)

20

15

10
(4)
5

Mf (n + 1) = e Mf ( n + 1) / M (n )
A(n + 1) = A(n ) + w L ( n ) MfT ( n + 1)
w L (n + 1) = c L (n + 1) + B( n ) M (n + 1)

100

200

300

400

500
iterations

600

700

800

900

1000

Figure 3. Comparison of learning curves for reconstructing AR(1) signal

with the subband input SNR=30 dB: (1) the Wiener filter, (2) the standard
RLS algorithm, (3) the stabilized B-SUSD FRLS algorithm, (4) the
Koilpillai et. al's algorithm.

e bM (n + 1) = ~z M (n + 1 L) B T ( n )z L ( n + 1)
~e b (n + 1) = b (n ) (n + 1)
M
M
M
e bM,i (n + 1) = K i e bM (n + 1) + (1 K i ) ~
e Mb ( n + 1), i = 1,2,3

30

M + L (n + 1) = M ( n ) + e MfT (n + 1) Mf ( n + 1)
M ( n + 1) = M + L ( n + 1) e bM,1,T (n + 1) M (n + 1)
Mb ,i (n + 1) = e bM,i ( n + 1) / M ( n + 1), i = 2,3
B( n + 1) = B( n ) + w L ( n + 1) Mb, 2 ,T (n + 1)
Mf ( n + 1) = Mf ( n ) + e Mf ( n + 1) MfT ( n + 1)
Mb ( n + 1) = Mb ( n ) + e bM, 3 (n + 1) Mb ,3,T ( n + 1)

(1) The Wiener Filter

25

(2) BSUSD FRLS Algorithm

(3) Koilpillai et al. Algorithm

reconstruction SNRc, dB

20

The time updating of the separation filter bank:

For l = 0 to M 1 Do

15

10

(1)

el (n + 1) = d l (n + 1) z TL (n + 1)h i ( n)

(2)

l ( n + 1) = el (n + 1) / M ( n + 1)
h i ( n + 1) = h i (n) + w L (n + 1) l (n + 1)

End For

The noise is assumed to be white Gaussian and the channel's

response to be {-0.077, -0.355, 0.059, 1, 0.059, -0.273}. The
input is a first-order AR signal with correlation coefficient
0.8.
In the first experiment, we show the convergence
characteristic of the proposed adaptive algorithm. Let the
input SNR be 30 dB and the forgetting factor for RLS
algorithms be 0.995. Fig. 3 shows the learning curves for all
algorithms. From the figure, we see that the convergence
behavior of the standard RLS and the stabilized TMUX
FRLS is very close, so is the steady-state reconstruction
SNR. The algorithm in [6] does not consider the channel
effect resulting in poor results. In the second experiment, we
compare the reconstruction SNRs for different input SNRs.
The result is shown in Fig. 4. As we find that the Wiener
filter outperforms the stabilized TMUX FRLS by 0.7 dB.
This is due to the fact that the forgetting factor used in RLS
is 0.995 instead of one. For all input SNRs, the Wiener and
the adaptive filtering perform a lot better than the
conventional separation filtering. At low input SNR, noise
dominates the distortion. Thus, the performance difference

(1)

30

(3)

15
20
10
channel signal to noise ratio, dB

25

30

Figure 4. Comparison of reconstruction performance for AR(1) signal

with noise and channel distortions.

between conventional separation filters and the proposed

filters is not as significant as that at high input SNR. At high
input SNR, the channel effect dominates. The proposed
separation filters act like equalizers and can achieve better
performance.
IV. APPLICATION TO THE MULTICARRIER
COMMUNICATION SYSTEMS
The OFDM system uses IDFT/DFT to implement a
multicarrier system. Data is modulated on the subcarriers
and the subcarriers are mutually orthogonal so that the
system can distinguish individual data on subcarriers after
removing the channel effect in the receiver. However, the
orthogonality is not satisfied with channel noise and
interference. Consider the following equation describing the
transmitted OFDM signal:

xOFDM (t ) =

t nTs N 1
j 2 f k ( t nTs )

T X [ k ]e
n =
s
k =0

(22)

3250
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where Ts is the symbol period, f = 1 / Ts is the subcarrier

frequency spacing, and [] denotes the rectangular
function. The OFDM signal is generated by IDFT for every
N data samples. Then the continuous-time OFDM signal can
be thought as multiplying the discrete-time samples by a
rectangular pulse with period of N data samples. It can be
shown that the spectrum of the OFDM signal is composed
of a shifting summation of N sinc functions with the
frequency spacing of f . The amplitude attenuation of the
first sidelobe for the sinc function is about 13.1dB, which is
very high compared to that for a typical filterbank system.
The high sidelobe value results in a significant loss of
orthogonality when noise or interference is introduced.
We use the proposed adaptive TMUX proposed to
compare the conventional OFDM system for the IEEE
802.11a application. The sample rate is 20 MHz,
modulation format is 64-QAM, active subcarrier number is
52, and the CP length is 16 samples. The OFDM system
uses two long training symbols for channel estimation by
least squares (LS) and linear minimum mean squared error
(LMMSE) algorithms. The 64-band filterbank system uses
128-tap coefficients for the combining filters and is
designed by the method in [11] with stopband attenuation of
about -25 dB. The amplitude response of the indoor channel
model used in the simulation is plotted in Fig. 5.
Fig. 6 shows the bit error rate (BER) curve comparison
of LS channel estimation, LMMSE channel estimation,
adaptively reconstructed separation filters with 128 taps.
For the adaptive TMUX system, the guard band can be
released since the sidelode of filterbank is lower than that of
the sinc function. From the results, we can see that the
adaptive TMUX systems either with guard bands or without
guard bands outperform the OFDM systems with LS or
LMMSE channel estimation. The adaptive receiver
minimizes the error between the received signal and the
transmitted signal, which includes minimizing the loss of
orthogonality due to noise as well. However, when SNR is
high, the loss of orthogonality is not significant and the
performance of the adaptive TMUX system approaches to
that of the OFDM system.
V. CONCLUSIONS
We have obtained the adaptive TMUX algorithm and the
Wiener solution by the property of polyphase
decomposition in a multirate system. By the multichannel
fast RLS algorithm, the B-SUSD algorithm can be used to
reduce computational complexity of the adaptive TMUX
system used for a multicarrier communication system. The
adaptive TMUX receiver can release the influence of loss of
orthogonality due to noise and imperfect channel estimation
compared to an OFDM system. The new algorithm is
applied to the IEEE 802.11a and the simulation shows that
the performance is better than the conventional OFDM
technique.
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