Abstract - In order to overcome the limitations of the fullband

and the previously proposed subband structures a new subband

structure is proposed. This newly proposed subband adaptive
filtering (SAF) structure will converge faster at a lower compu-
tational cost. The interband aliasing is introduced by the
downsampling process, degrades the performance. So the inter-
band aliasing is extracted and removed from the subband sig-
nal with the help of bandwidth increased analysis filter. Here,
an almost alias free SAF structure with critical sampling is pro-
posed to have faster convergence and lower computational com-
plexity. The software used in this paper is mat lab for the simu-
lation purpose with a signal processing tool box.

Keywords: Adaptive filtering, interband aliasing, LMS algorithm,
multirate signal processing, and subband adaptive filtering.

I. INTRODUCTION
One promising method that improves the performance
and reduces the computational cost is subband adaptive
filtering (SAF). It has the potential for a faster conver-
gence and a lower computational complexity than a
fullband structure. However, a subband structure suf-
fers from two deficiencies. First, the interband aliasing
that is introduced by the downsampling process re-
quired in reducing the data rate is unavoidable and de-
grades the performance. Second, the filter bank intro-
duces additional computation and system delay. For
these reasons various SAF structures were proposed. In
SAF, signals are decomposed into a number of subband
signals using an analysis filter bank, and the adaptive
filtering is performed on each subband. The result in
each subband is combined into an output using a syn-
thesis filter bank. For a critically sampled SAF with M
subbands, the kth subband error E
k
(z), shown in Fig.1,
is

Where X(z), H(z), H
k
(z) and G
k
(z) represent the z-
transforms of input signal, unknown system, the kth
analysis filter, and the kth adaptive filter to respective-
ly.

Now W
i
m
= e
-j2i/M
and the kth analysis filter H
k
(z) is a
bandpass filter .Equation (1) can be written as


Where

Fig.1. Block diagram of the adaptive filtering in the kth sub band.
When the filter bank is real-valued, the terms H
k
(z
1/M

W
k
M
) and H
k
(z
1/M
W
k+1
M
) are adjacent to H
k
(z
1/M
) and
when H
k
(z)
k
(z)is designed with high enough stopband
attenuation, is approximately zero. The second and
third terms on the right-hand side of (2) are the errors
introduced by interband aliasing due to the overlapping
between H
k
(z
1/M
W
k
M
) and H
k
(z
1/M
) and between H
k
(z
1/M

W
k+1
M
) and H
k
(z
1/M
) . In order to reduce |E
k
(z)| to a
value close to zero, the adaptive filter G
k
(z) has to
match the two different frequency responses, that is,
both H
k
(z
1/M
W
k
M
) and H
k
(z
1/M
) which have considerable
overlap and both H
k
(z
1/M
W
k+1
M
) and H
k
(z
1/M
) which
also have considerable overlap between them. Of
course, this is impossible. These mismatches result in
a large |E
k
(z)| around = k and (k+1) even after con-
vergence, therefore, the SAF with critical sampling has
a large MSE at the output.

II. ALIAS FREE SAF WITH CRITICAL
SAMPLING
The interband aliasing components are caused by
downsampling the signal, the downsampling process is
essential in almost all multirate signal processing for
making the overall data rate nearly equivalent to that of
the input. Fig.2 shows the magnitude responses
ALIAS FREE SUBBAND ADAPTIVE FILTERING A NEW APPROACH

1
CHAPPIDI ASHOK,
2
B.RAMESH BABU
1,2
Department of Electronics & Communication Engineering,, Dr. Samuel George Institute of Engineering & Technology,
Markapur, Prakasam (Dist.) A.P., INDIA
e-mail: chappidiashok@gmail.com
1
, rameshbabu.bonige@gmail.com
2

International Journal of
Systems , Algorithms &
Applications
I II I
J JJ J
S SS S
A AA A A AA A
Volume 2, Issue ICTM 2011, February 2012, ISSN Online: 2277-2677 39
ICTM 2011|June 8-9,2011|Hyderabad|India
( ) ( ) ( ) ( )
( ) ( ) ( ) 1
1 1
) (
/ 1
1
0
/ 1
/ 1 / 1
1
0
/ 1
i
m
M
M
i
i
m
M
k
k
i
m
M i
m
M
M
i
i
m
M
k k
W z W z H
z G
M
W z W z H W z H
M
z E

=

=
( ) ( ) ( ) [ ] ( ) ( ) ( ) ( ) [ ]
( ) ( ) ( ) ( ) [ ]
( ) ( ) ( ) ( ) 2
.
1 / 1 1 / 1
1 / 1 / 1 / 1
/ 1 / 1 / 1 / 1
z W z X W z H
z G W z H W z X W z H
z G W z H z X z H z G z H z E M
k
k
M
M k
M
M
k
k
k
M
M k
M
M k
M
M
k
k
k
M
M M M
k k
M
k
+
+
+ =
+ +
+
( ) ( ) ( ) ( ) ( )
( ) ( ) ( ) 3
/ 1
1
1 ,
1
/ 1
/ 1 / 1
1
1 ,
1
/ 1
i
M
M
M
k k i
i
i
M
M
k
k
i
M
M i
M
M
M
k k i
i
i
M
M
k k
W z X W z H
z G W z X W z H W z H z

+
=

+
=

=


X
k
e
j
of the signal that has passed through the kth analy-
sis filter, whose transition bandwidth is

/M and its
down sampled version X
d
k
e
j
for k = 0, 1,.., M-1. Hence-
forth, only the frequency interval of k (k+1) will
be considered, since the frequency responses in the other
intervals are just the frequency-shifted and the frequency
-flipped versions of the response in this interval. As
shown in Fig.2 (b), the frequency interval of k
(k+1) is divided into three subintervals as

1
= {/k k +

2
= {/k+

(k+1) -

3
= {/(k+1) -

(k+1) }

For
1
the error E
k
(e
j
) can be approximated as



For
2,


finally, For
3,
it can be approximated as


Fig.2. Magnitude responses of the signals in the kth subband. (a)
Output of the kth analysis filter. (b) Its downsampled version.

From (4) and (6), the kth adaptive filter G
k
(e
j
) has to
match the desired response H(e
j
) while simultaneously
matching the aliased response H(e
j(-2k)/M
) at
1
and
the aliased response H(e
j(-2(k+1))/M
) at
3
inorder to
ma ke | E
k
( e
j
) ze r o. Thi s i s i mpos s i b l e .

Fig.3 .Alias-free SAF structure with critical sampling in the kth sub-
band.

In Fig.3, the kth subband portion of the proposed SAF
structure is shown. H
k
(e
j
) is a linear-phase bandwidth-
increased analysis filter of the order 2MN
d
whereis an
integer and determines the order of H
k
(e
j
) the interband
aliasing component X
b
k
(e
j
) is extracted using the band-
width-increased analysis filter H
k
(e
j
) and then its deci-
mated version X
bd
k
(e
j
) is subtracted from the subband
signal X
d
k
(e
j
) to obtain the almost alias-free X
n
k
(e
j
).
The spectral dips of X
n
k
(e
j
) are reduced using a mini-
mum-phase filter W(e
j
), and the result X
w
k
(e
j
) is used
as an input of the adaptive filter G
k
(e
j
) . The same anal-
ysis is also applied to D
k
(e
j
), which is the kth subband
signal of the desired signal D(e
j
) and thus the almost
alias-free subband desired signal D
w
k
(e
j
) with flat spec-
trum is obtained and used as a desired signal of the
adaptive filter G
k
(e
j
). The output ofG
k
(e
j
), E
w
k
(e
j
),
passes through 1/W(e
j
) to negate the effect of W(e
j
),
and then E
k
(e
j
) is obtained. In the kth subband, X
d
k
(e
j
)
is given as
Where
The analysis filters H
k
(e
j
) can be designed such that the
overlapping
k
(e
j
) between nonadjacent terms can be
much smaller than the overlapping between adjacent
ones and thus negligible for k = 0, 1 M-1. In order to
eliminate the aliasing term X
d
k
(e
j
) from the aliasing
components should be obtained first. To do so, the out-
put signal of the analysis filter in X
k
(e
j
) fig.4(a) is sub-
tracted from the interpolated signal X
i
k
(e
j
) in Fig. 4(b),
and the result is X
i
k
(e
j
) as

Where M is multiplied to X
i
k
(e
j
) since the amplitude of
X
k
(e
j
) is reduced by M during the downsampling proce-
dure. As shown in Fig.4(c), X
s
k
(e
j
) includes only the
interband aliasing components in k/M (k+1)/M
The interband aliasing components are extracted by
passing X
s
k
(e
j
) through kth bandwidth-increased analy-
sis filter H

k
(e
j
) , whose magnitude response is almost-
flat from k to (k+1)/M as shown in fig.4(d); this is the
reason why the term bandwidth-increased is used for
H

k
(e
j
) . In general, the frequency interval where the
magnitude response of the passband of the filter bank is
almost-flat is narrower than that of filter H

k
(e
j
) for a
perfect reconstruction (PR) of the filter bank.
ALIASFREESUBBANDADAPTIVEFILTERINGANEWAPPROACH
International Journal of
Systems , Algorithms &
Applications
I II I
J JJ J
S SS S
A AA A A AA A
Volume 2, Issue ICTM 2011, February 2012, ISSN Online: 2277-2677 40
ICTM 2011|June 8-9,2011|Hyderabad|India
( ) ( ) ( ) [ ] ( ) ( )
( )
( ) ( ) [ ]
( )
( )
( )
( ) ( ) 4
/ 2 / 2 / 2
/ / /
M k j M k j
k
j
k
M k j
M j M j
k
j
k
M j j
k
e X e H e G e H
e X e H e G e H e E M



+

( ) ( ) ( ) [ ] ( ) ( ) ( ) 5
/ / / M j M j
k
j
k
M j j
k
e X e H e G e H e E M


( ) ( ) [ ] ( ) ( )
( )
( ) ( ) [ ]
( )
( )
( )
( ) ) 6 .(
) ( .
1 2 ( ) 1 2 (
1 2
M k w j M k w j
k
j
k
M k jw
M jw M jw
k
j
k
M jw
k
e X e H
e G e H
e e H e G e H ej E M
+ +
+

+

( )
( ) ( ) ( ) ( )
( )
( ) ( )
( ) ( ) ( ) ( )
( ) [
( ) ( ) ( ) ( ) ( )
( )] ( ) ) 7 (
1
1
1 2 ) 1 ( 2
2 2 / /
2 2



j
k
M k j M k j
k
M k j M k j
k
M j M j
k
M i j M i j
k
j d
k
e e X e H
e X e H e X e H
M
e X e H
M
e X
+ +
+
=
+ +

( )
( )
( )
( )
( )

+
=

=
1
1 ,
1
/ 2 / 2
) 8 (
1
M
k k i
i
M i j M i j
k
j
k
e X e H
M
e

( ) ( ) ( )
( )
( ) ( )
( )
( ) ) 9 (
1
1
/ 2
1
0
/ 2

=
=
=
M
i
M i j
k
j
k
M
i
M i j
k
j
k
j u
k
j s
k
e X
e X e X
e X e X M e X






Fig. 4. Magnitude responses of the signals of the proposed SAF
structure. (a) Output signal of the kth analysis filter. (b) Interpolated
version of X
d
k
(e
j
). (c) Resulting signal after subtracting X
k
(e
j
)
from M.X
i
k
(e
j
). (d) Kth bandwidth-increased analysis filter. (e)
Extracted interband aliasing component.

In 0, H

k
(e
j
) overlaps predominantly with X(e
j(-2/
M)
) when i= k and I = k+1, thus X
b
k
(e
j
) is given as




which is shown in Fig.4(e), thus X
b
k
(e
j
). is given as

Where


After down sampling X
b
k
(e
j
), an unwanted interband
aliasing occurs due to the overlapping between X
b
k
(e
j
).
in - 0 represented in (11) and X
b
k
(e
j
) in 0
represented in (10).The downsampled interband aliasing
component is




As shown in Fig.4, the kth almost alias-free subband
input signal is given as


by inserting (7) and (13) into (14) and then representing
H

k
(e
j
) as e
-jNd
k (k+1), X
n
k
(e
j
). is given by







With X
k
(e
j
)= H
k
(e
j
)X(e
j
), X
n
k
(e
j
) at the three subinter-
vals i.e., is given as (16). This analysis also applied to
D
n
k
(e
j
) as shown in Fig.4, and thus D
n
k
(e
j
) is given as
(17). If X
n
k
(e
j
) and D
n
k
(e
j
) are used for adaptive filter-
ing in subbands instead of X
d
k
(e
j
) and D
d
k
(e
j
) in the kth
subband, the output of the adaptive filter E
k
(e
j
) is given
as (18). This is not the case as in (4) and (6) where G
k
(e
j
) has to match the desired H(e
j/M
) while simultane-
ously matching the aliased responses H(e
j(-2k)/M
) at

1
and H(e
j(-2k)/M
) at
3
in order to make |E
k
(e
j
)|
zero, From (18), G
k
(e
j
) needs to match just the desired
response H(e
j/M
) to make |E
k
(e
j
)|zero. Therefore, the
interband aliasing error can be zero.

Fig.5 (a) Subband input with interband aliasing. (b) Downsampled
version of the extracted interband aliasing component. (c) Almost
alias-free subband input. (d) Filter for the spectral flatness of the
almost alias-free subband signal. (e) Almost alias-free subband input
with the flat spectrum.















As shown in Fig.5c, X
n
k
(e
j
) has spectral dips at the sub-
intervals
1
and
3
and so does D
n
k
(e
j
) . The spectral
dips at
1
and
3
can be reduced by using a minimum-
phase filter W(e
j
) that has the following ideal magni-
tude response





The design procedure of is as follows: first, we design a
minimum-phase bandpass filter with passband
2
and
transition bands
1
and
3
.The designed minimum-
phase bandpass filter is inverted to give W(e
j
), and thus
W(e
j
) is designed to be a minimum-phase filter. The
output of W(e
j
) in the kth subband X
w
k
(e
j
) is given as
X
w
k
(e
j
) = W(e
j
) X
n
k
(e
j
) -(20)
The kth desired signal D
w
k
(e
j
) is obtained using W(e
j
)
in the same way as X
w
k
(e
j
). Therefore, when X
w
k
(e
j
)
ALIASFREESUBBANDADAPTIVEFILTERINGANEWAPPROACH
International Journal of
Systems , Algorithms &
Applications
I II I
J JJ J
S SS S
A AA A A AA A
Volume 2, Issue ICTM 2011, February 2012, ISSN Online: 2277-2677 41
ICTM 2011|June 8-9,2011|Hyderabad|India
( ) ( ) ( )
( )
( )
( )
( ) ( )
( ) ( ) [ ] ) 10 (
/ 1 2 / 2

jM
k
M k j
k
M k j
k
j
k
j s
k
j
k
j b
k
e M e X e X
e H
e X e H e X
+ +
=
=
+
( ) ( )
( )
( )
( ) ( )
( ) ( ) [ ] ) 11 (
/ 1 2 / 2 jM
k
M k j
k
M k j
k
j
k
j b
k
e M e X e X e H e X + + =
+ + +
( )
( )
( )
( )
( )

=

=
1
1 ,
1
/ 2 / 2
) 12 (
M
k M k M i
i
M i j M i j
k
jM
k
e X e H e M


( )
( )
( )
( )
( )
( )
( )
( )
( ) ( )
( ) [ ]
( )
( ) ( )
( ) ( )
( ) ( ) ) 13 (
/ / 1 2 / / 2
/ 1 2 / 2 /
1
0
/ 2 / 2
M j
k
M k j
k
M j
k
M k j
k
M k j
k
M k j
k
M j
k
M
t
M l j s
k
M l j
k
j bd
k
e X e H e X e H
e X e X e H
e X e H e X M



+
+

=

+
+
=

( ) ( ) ( ) ) 14 (
j bd
k
j d
k
jN j n
k
e X e X e e X
d
=

( )
( )
( )
( )
( ) ( )
( ) [ ]
( )
( )
( )
( ) ( )
( ) [ ]
( )
( )
( ) ( )
( ) ( )
( )
( )
( ) ( )
( ) [ ] ( ) ) 15 (
/ / 1 2 / 2
/ / 1 2 / 2
/ 1 2 / 2 /
/ 1 2 / 2 /
M j
k
M k j
k
M k j
k
jN
M j
k
M k j
k
M k j
k
M k j
k
M k j
k
M j
k
M k j
k
M k j M j
k
jN
j n
k
e X e H e H e
e X e H e H
e X e X e H
e X e e X e
e X M
d
d

+
+
+
+

+
+
+ +

( )
( )
( ) [ ] ( ) ( )
( ) ( ) ( )
( ) ( )
( ) [ ] ( ) ( )

=
+

, ,
) 16 ( , ,
, ,
.
3
/ / / 1 2
2
/ / /
1
/ / / 2

M j M j
k
M k w j
k
jN
M j M j M j
k
jN
M j M j
k
M k w j
k
jN
j n
k
e X e H e H e
e X e H e H e
e X e H e H e
e X M
d
d
d


and D
w
k
(e
j
) are used for adaptive filtering in subbands
instead of X
d
k
(e
j
) and D
d
k
(e
j
) in the kth subband, is giv-
en as E
w
k
(e
j
) (21). As represented in (21), the kth adap-
tive filter G
k
(e
j
) needs to match only H(e
j/M
) as in (18)
achieving the spectral flatness. The output of E
k
(e
j
) the
Adaptive filter is obtained by passing E
w
k
(e
j
) through 1/
W(e
j
) to negate the effect of W(e
j
). Since W(z)is de-
signed to be a minimum-phase filter, 1/W(z) is imple-
mented by inverting W(z). The bandwidth-increased
analysis filters can be designed by cosine modulation of
the prototype filter. The analysis and synthesis filters of
the CMFB are obtained by cosine modulation of the pro-
totype linear-phase lowpass filter. If p[n]is the impulse
response of the FIR prototype filter, the impulse re-
sponses of kth analysis and synthesis filters, h
k
[n] and, f
k
[n]are given as





Where L is the order of the FIR prototypefilter, and
k
=
(-1)
k
(/4) for k=0,1,...M-1. The term
k
on the right-hand
side of (22) and (23) is used to cancel the interband ali-
asing component at the output of the filter band and sig-
nificant distortion around frequencies =0 and = re-
spectively to obtain PR. However, the bandwidth-
increased analysis filter band does not need the PR con-
dition, thus
k
is discarded here. Therefore, if p[n] is
the impulse response of the liner-phase FIR bandwidth-
increased prototype filter, the impulse response of the
kth bandwidth-increased analysis filter h
k
[n] is given as
follows
Where L is the order of the FIR bandwidth-increased
prototype filter for k=0,1,...M-1. In order to design the
magnitude response of the kth bandwidth-increased anal-
ysis filter to be unit from k/M(k+1)/M the magni-
tude response of the prototype lowpass filter should be
unit from 0 to /2M Such condition on the magnitude
response of the prototype filter causes amplitude distor-
ton to H
0
(e
j
) around =0 and to H
M-1
(e
j
)around =
but no amplitude distortion for k=1,2...M-2

II. SIMULATIONS
Four simulation experiments were conducted.

A .White Noise Input - The simulations using the pro-
posed SAF were conducted by varying the number of
subbands, that is, for M = 2, 4, 8, and 16.The MSE evo-
lutions of the proposed SAF and the fullband NLMS
algorithm are shown in Fig.6. The figure shown that the
convergence rates of both the proposed SAF and the
fullband NLMS algorithms were similar for a white
noise input, however, the fullband reached lower steady
state MSE than the proposed after convergence.

Fig.6. MSE performance for the proposed SAF algorithm for differ-
ent values of M and the fullband NLMS algorithm for a white noise
input.

B. Colored Input- The simulation using a colored input
was conducted. The MSE evolution is shown in Fig.7.
The proposed SAF algorithm performed better than the
fullband NLMS algorithm. Increasing the number of
subbands led to better convergence rate, but larger MSE
after convergence

Fig.7.MSE performance for the proposed SAF algorithm for different
values of M and the fullband NLMS algorithm for a colored input.
C. Speech - An acoustic echo cancellation (AEC) exper-
iment was conducted with real speech sampled at 8
kHz. The residual echoes using the proposed eight-
channel SAF algorithm and the fullband NLMS algo-
rithm are shown in Fig.8. It shows the proposed SAF
algorithm cancelled the echo better than the fullband
NLMS algorithm.

Fig.8. Residual echo signals using the proposed eight-channel alias-
free SAF algorithm and the fullband NLMS algorithm for speech
input.

D. Comparison of Performances - The performance of
the proposed SAF algorithm was compared to those of
the SAF algorithms when M = 4. For a colored noise of
length 200000, simulation result shown in Fig.9 was
obtained. The proposed SAF achieved faster conver-
gence rate and lower MSE than the other algorithms.
ALIASFREESUBBANDADAPTIVEFILTERINGANEWAPPROACH
International Journal of
Systems , Algorithms &
Applications
I II I
J JJ J
S SS S
A AA A A AA A
Volume 2, Issue ICTM 2011, February 2012, ISSN Online: 2277-2677 42
ICTM 2011|June 8-9,2011|Hyderabad|India
[ ] [ ] ( ) ) 22 (
2
5 . 0 cos 2
|
|

\
|
+ |

\
|
+ =
k k
L
n k
M
n p n h

[ ] [ ] ( ) ) 23 (
2
5 . 0 cos 2
|
|

\
|
|

\
|
+ =
k k
L
n k
M
n p n f

[ ] [ ] ( ) ) 24 (
2
5 . 0 cos 2
|
|

\
|
|

\
|
+ =
L
n k
M
n p n h
k

Fig.9. Comparison of the performances of the proposed alias-free
SAF algorithm and the conventional SAF algorithms for a colored
input when M = 4.

III. CONCLUSION
In this paper, a structure with critical sampling that is
virtually alias-free is proposed. The interband aliasing is
extracted in each subband using the bandwidth-increased
FIR linear-phase analysis filters and then subtracted
from each subband signal. The almost alias-free sub-
band signals have spectral dips, so the spectral dips are
reduced using a filter for the spectral flatness and then
the outputs are used for adaptive filtering in each sub-
band. Simulations results show that the proposed sub-
band structure achieves similar convergence rate to the
fullband structure for white noise input and better con-
vergence rate than both the equivalent fullband at lower
computational complexity and the conventional SAF
structures for colored input.

IV. REFERENCES
[1] Gilloire and M. Vetterli, Adaptive filtering in subbands, in
Proc.IEEE Int. Conf. Acoust., Speech, Signal Process., Apr. 1988,
pp. 15721575.
[2] S. S. Pradhan and V. U. Reddy, A new approach to subband
adaptive filtering, IEEE Trans. Signal Process., vol. 47, no. 8, pp.
655664, Aug. 1999.
[3] P. P. Vaidyanathan, Multirate Systems and Filter banks. Eng-
lewood Cliffs, NJ: Prentice-Hall, 1993.
[4] R. D. Koilpillai and P. P. Vaidyanathan, Cosine-modulated FIR
filter banks satisfying perfect reconstruction, IEEE Trans. Signal
Process, vol. 40, no. 4, pp. 770783, Apr. 1992.
[5] S. Haykin, Adaptive Filter Theory. Englewood Cliffs, NJ: Pren-
tice- Hall, 1996.
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Volume 2, Issue ICTM 2011, February 2012, ISSN Online: 2277-2677 43
ICTM 2011|June 8-9,2011|Hyderabad|India