A Variable Tap-length Feedback Minimum Mean
M-estimate Adaptive Sub-band Filtering
Algorithm for Improved ANC
Ambika Prasad Chanda Abhishek Deb
Infosys Technologies Ltd Tata Consultancy Services Ltd
Bhubaneswar, Odisha, India-751024 Bangalore, Karnataka,India-560066
chand.iiit@gmail.com abhishekdeb1014@gmail.com
Abstract— Active Noise Canceller has become very common in
modern day to day electronic equipment. The adaptive filters
installed inside this canceller play a crucial role in noise
cancellation. The computational complexity as well as the
structural complexity of the filter is an important factor to be
considered for the overall performance. This depends on the
structure of the filter. The structure depends on the number of
taps. Generally filters used for Active Noise Cancellation have
fixed tap length and are long which result is slow convergence
and delay. Therefore it is necessary to have a trade-off between
the length of the filter and its convergence. In this paper a Sub-
band adaptive filtering technique is employed to reduce the
length of the filter and a new Minimum Mean M-estimate
algorithm having variable tap length is proposed. To be precise,
the advantages of two concepts are combined in this paper so that
the efficiency of the filter is maximum. They are the Sub-band
adaptive filtering and the variable tap length. The simulation and
results justify the proposal.
Keywords— Adaptive filters; Active Noise Cancellation (ANC);
Filtered-X Least Mean Square algorithm;Fractional Tap-
length;Sub-band Adaptive Filtering.
I. INTRODUCTION
Adaptive filters are largely used in various applications
such as Acoustic Echo Cancellation (AEC), ANC, channel
modelling, channel equalization etc. The Finite Impulse
Response (FIR) adaptive filter is usually prevalent because of
its intrinsic stability and Tapped Delay Line (TDL) feed
forward structure [1]. Practically, ANC method in a real-time
operation should have less computational complexity,
maximum noise removal and stability and robustness to input
signal. Hence, for noise cancellation algorithms, long adaptive
filters are a must, which in turn result in prodigious
computational load.
Linear feed forward FIR adaptive filters are primarily
fabricated of two components that is the delay unit and the
weight [2]. The tap-length is the overall quantity of weights in
an adaptive filter. It adversely affects the potential of the TDL
structure since it is here where the weights are continuously
upgraded by various adaptive algorithms. If the tap-length is
very less, then the system gets under modelled and its MSE
rises. If the tap-length is very high then it introduces extra
978-1-4799-7961-5/15/$31.00 ©2015 IEEE
AsutoshKar
EEE Department, BITS Pilani
(Hyd.),Telangana, India-500078
asu131@gmail.com
noise and complexity. Hence, an optimal value must be found
so as to maintain a trade-off between the adaptive filter
performance and complexity [3].
There are many existing algorithms that are employed to
ANC environment. The FX-LMS algorithm used for
cancelling noise was not that efficient in regards to its
convergence rate [4]. It is seen in the following sections that it
doesn’t cancel all of the noise. The Modified FX-LMS(M-FX-
LMS) algorithm was quite efficient but increased the
computational complexity from 2M to 3M [5].The Normalized
FX-LMS(N-FX-LMS) algorithm was designed, in which the
step size was normalized by help of the power of the signal
because it affected important properties such as performance,
stability and error after convergence [6]. But the method
which was used in estimating the power of the signal
employed the exponential window which failed when there
were sudden power changes of the signal which resulted in
poor convergence and more delay. The FX-Least Mean M-
estimate (LMM) algorithm was introduced which also
successfully cancelled noise by converging faster than the FX-
LMS. But it was not that fruitful in decreasing the impulsive
noise present in the input signal. For this a Modified FX-LMM
(M-FX-LMM) algorithm was proposed in which the step size
decreased faster than that of the FX-LMM algorithm. It was
successful in decreasing the impulsive noise present in the
input signal to less than 1.883.The problem with this algorithm
was that for the impulsive noise part this algorithm converged
slowly, on the other hand it converged faster for the
background noise. All the above algorithms used a fixed tap-
length for their functioning which led to either more structural
and computational complexity or to poor convergence rate.
So till now it is clear that there are mainly two problems
which are needed to be sorted out if an effective cancellation
of noise is required. Firstly, a solution must be found for the
long adaptive filters that are required for ANC. Secondly, an
optimum order or a range for the suitable tap-length must be
found for the adaptive filter. To solve the former, a concept
known as Sub-band adaptive filtering is employed which
successfully reduces the load on the filter. The above two
problems can be resolved by the Feedback Minimum Mean
M-estimate (FMMM) algorithm [7]. But the algorithm has
fixed tap length. Fixing the tap-length at a certain value
sometimes results in inescapable issues with the adaptive
design like deficient modelling and adaptation noise. The
fractional tap-length algorithm which combined the
advantages of segmented filter as well as the gradient descent
algorithm was proposed [8-10]. This algorithm had less
complexity than the earlier methods. But due to random use of
leaky factor it suffered from noise level variations. So in this
paper the VTL version of the FMMM algorithm is presented.
Here the order of the filter can be altered to any value so that
optimal performance is achieved and the trade-off between
tap-length keeping in view that the rate of convergence is
preserved. The results of the proposed VTL algorithm justify
the effectiveness of the algorithm in ANC environment.
II. PROBLEM FORMULATION
Acoustic noise results due to the coupling of the
microphone and loudspeaker as shown in Figure 1.The
situation gets worse when it is a digital communication. In the
Figure for simplicity it is shown that there are two speakers
involved in communication .One is the near end speaker and
the other is known as the far end speaker. The speech signal
from the near end speaker goes through the channel. On its
way, it gets added up by the noise present in the surrounding.
This corrupted signal goes through the channel and reaches the
far end speaker. In the far end speaker’s headphone or
receiver, a card is installed to cancel the noise present in the
corrupted signal [11]. This card contains a framework known
as ANC Framework. It is displayed as a shaded box in the
Figure 1.This framework works with the help of an adaptive
filter which is governed by an adaptive algorithm. There are
mainly three problems that are being encountered in the ANC
process.
Fig. 1. ANC Framework
The problems like filter length and convergence can be
resolved by the FMMM algorithm [11].This also helps in low
computational complexity. The users demand not only for a
low complexity and faster convergence but also better
performance at different situations. The FMMM algorithm has
fixed tap length which creates problem at certain situations. So
our main aim is to design an algorithm which performs
according to the situation having a variable tap length. So this
problem is addressed in this paper and corresponding solutions
are proposed.
III. PROPOSED ALGORITHM
Sub-band adaptive filtering technique is a part of the
concept of. Filter banks are used for spectral disintegration
filter bank theory and composition of signals. It is possible to
do this because of the arrangement of the low-pass and high
pass filters. They allow easy extraction of spectral
components, and so find effective use in many signal
processing fields such as ECG signal compression, Frequency
Division Multiplexing or Time Division Multiplexing
conversion, antenna systems and also in ANC. The main
reason for which this concept finds its maximum use in ANC
is that it reduces the computational complexity especially
when the noise is in the broadband and there is a long impulse
in the system.
d(i) + 
err(i)
ĉ(i)
∑
R( z )
−
+ Fb ( z ) L( z )
∑
+
o(i)
W ( z)
L̂( z ) MM
c′(i )
Fig. 2. Proposed Algorithm
In the FMMM algorithm instead of the mean square
objective function as in the FX-LMS algorithm and its family,
the mean M-estimate error objective function is utilized on the
basis of robust statistical estimation [12].
Fig. 2 shows the structure of the FMMM algorithm
with feedback. This structure is physically implemented to
control the vibrations in a mechanical system. R ( z ) is the
reference sensor, W ( z ) is the control unit, Fb ( z ) is the
feedback unit that generates secondary vibrations and there is
an error sensor in the end. Now vibration is also a type of
mechanical noise.
To cancel this noise FMMM algorithm is already derived
which is:
W (i + 1) = W (i ) - μ (i ) m(err  (i )C ′(i )
 (i ))err (1)
The problem with the above algorithm is its fixed tap
length. Selection of tap length in a particular environment is
not a petty task. The selection depends on behavior of the
system to be recognized, memory constraint, anticipated
performance, computational and structural complexity, noise
level, convergence rate etc. For example when the user is in
motion or the environment is constantly changing then in that
case the noise level is also constantly changing as a result the
optimum tap length has to vary from time to time. In most of
the filter designs, regrettably the tap-length is fixed at some
fixed value creating the problem of too short and too long log10 θ vt (i )
filters. So in the weight update equation we need to add an fo = (8)
extra term which will imply that the tap-length is changing (θ max − θ min )
after every iteration.
where θ vt (i) is the adjustable error positioning factor,
For convenience of derivation we change the notations of
(1), and the new equation is given by: (θ max , θ min ) is fixed bestowing to the system necessities.
Wvt (i ) (i + 1) = Wvt (i ) (i ) - μ (i )m(err
 (i )) 
err (i )C ′ (i ) (2) The MSE is the sum of Excess MSE (EMSE) and the system
vt (i ) vt (i ) vt (i )
noise as,
where vt (i ) is the instantaneous variable adaptive tap-length
and, [4] 2 2 2
E[(errvt (i ) (i )) ] = E[ M ex (i )] + E[t (i )] (9)
μ
μ (i ) = T , (3)
C ( i )C (i )[2 + vt (i )]
vt (i ) vt (i ) where, Mex (i) is the EMSE . It so happens that at the initial
stage the iteration factor increases to a big value and then in
μ is a constant and lies between 0 and 2. the advanced stage to a trivial value when the tap-length is
Now suppose the tap length is made to vary, for varied continuously .So the EMSE helps to update this factor.
example let it be increased, then if the new MSE found is very At steady state, the performance of the squared estimated error
close to the old MSE then it will imply that there is negligible is given by,
effect on the MSE when additional taps are added to the 2 (1 − fo ) 2
vt (i ) (i ) ] ≅ (1 + f ) σ t
current order. Let θP = M P-1 (∞) − M P (∞) be difference 
E[err (10)
o
between the converged MSE when the filter order is increased
The assumption which is taken here is that at
from VT − 1 to VT . Now the optimum order can be defined as
balanced state the error signal approaches the noise of the
VT′ that satisfies, present structure. Next, in time changing environment the tap
θP < η for all VT > VT ′ (4) length adaptation expression is given by, [3]
vtnf (i +1) = ⎡⎣vtnf (i) - Lf ⎤⎦ + [(errvt (i) (i)) - (errvt (i)-θ
2 2
Where η is a small positive number fixed as required by the (i)) ]Lf (11)
vt (i )
system.
Hence we get the equation of the tap-length adaptation for the
The selection of tap length can be done if the following cost next iteration as: [4-6]
function inequality is valid min{VT | M P-1 - M P ≤ η}
⎧ Lf ⎫
.During the process of selection it so happens that a deceitful ⎪ vtnf (i ) if vt (i )-vtnf (i ) > ⎪
optimum tap-length is found which builds misperception. So vt (i + 1) = ⎨ Lf ⎬ (12)
these quasi tap-lengths can be defined as given below: ⎪ ⎪
⎩ vt (i ) otherwise ⎭
Z < VT ′ and θ z < η (5)
vtnf (i ) , is the tap length and it can take decimal values.
where Z is named as the quasi-optimal filter order .The
width of the quasi-optimal filter order is found as S + 1 if the Logically the order of a filter cannot be fractional so this tap-
length is approximated to the closest integer value to obtain
inequality given above is fulfilled by a group of linked
integers Z , Z + 1,............., Z − S + 1 . Next, we cannot find the the optimal tap length. In (12) L f is the Leakage factor. The
steady state MSE directly, so exponential averaging is done, major role of this factor is to avoid the order to upsurge to an
unanticipated high value. L f is the step size for the adaptation
2
M vt ( i ) (i + 1) = (1 - ω )( errvt ( i ) ) (i + 1) + ω M vt ( i ) (i ) (6) of the filter. [3-6]
L f = min( L f ,max , L f (i + 1)) (13)
Where ω is the smoothing parameter which govern the
operative retention of the iterative process.
where ,
i-1 n-i n

errvt (i ) ( i ) = (1 - f o ) ∑ f errvt ( n ) + t ( n ) fo (7)  2 (i + 1)
err
n =0 opt L f (i + 1) = (14)
 2 (i + 1) + θ vt (i )
err
ss

Where n is the time interval, vt opt is the optimal choice of tap-

vt (i ) (i + 1) = f errvt (i ) (i ) + (1 - f )errvtopt (i ) (i + 1)

err  (15)
length. fo is a forgetting smoothing parameter which can be
estimated as,
θvtss is the adjustable error positioning constraint at steady A comparison of MSE is made for all the existing algorithms
in ANC environment on the basis of mean square error. The
state tap-length vtss and f is a partial load factor. At the noise added here is the additive white Gaussian noise with
stable state, zero mean and a finite variance. The input signal is taken as a
sinusoidal signal of frequency 6 mHz. The room impulse
2
(1 - f )σ t response as shown in Figure 3 is measured for an arena of
Lf → (16)
(1 + f )θ vtss (∞) dimension 12 ×14 ×10 ft3. The mode of simulation used here
is MATLAB.
Now θ vt is the difference between the MSE values as already
discussed. If this difference is large then the adaptation is slow
and also the reverse is true.

L f = min( L f ,max , λ L f ,max ) (17)

where,
2 2
[errvt ( i ) (i )) − ( errvt ( i )-θ (i )) ]
vt ( i )
λ= 2 2 2 (18)
( errvt ( i ) (i )) + f [ errvt ( i ) (i )) − ( errvt ( i )-θ (i )) ]
opt vt ( i ) Fig.5. MSE Vs. No. of Iterations where SNR=40dB

The above algorithm is the proposed variable tap-length

FMMM algorithm.
IV. RESULT
In this section the output of the proposed algorithm is shown
with FTL implementation.

Fig.6. MSE Vs. No. of Iterations where SNR=40dB for FTL and VTL
Fig. 4 shows the recorded output signal of the noise canceller
when the proposed algorithm is implemented. When the
proposed FMMM algorithm with FTL is applied the noise is
present only till around 150 iterations which is very less in
Fig. 3. Room impulse response plot comparison to other existing algorithms such as FXLMS or
NFXLMS etc. So this shows that this new algorithm cancels
noise effectively in less number of iterations.
From Fig.5 it is observed that when the simulation is done
at Signal-to-Noise-Ratio (SNR)=40 dB, the proposed
algorithm converges in about 3000 iterations. Again, the
simulation is done at SNR=0 dB, the proposed algorithm
converges in about 2800 iterations. This clearly proves that the
MSE performance and the convergence rate of the proposed
FMMM algorithm is far better and effective than the existing
ones.
From Fig. 6 it can be clearly observed that the proposed
VTL algorithm performs better than the FTL algorithm in
extreme conditions.
V. CONCLUSION
The Sub-band adaptive filtering technique is implemented to
Fig. 4. Proposed algorithm find the range of the length of the filter required for perfect
cancellation of noise. The proposed FMMM algorithm is
mathematically derived and simulated also. On the basis of
simulation, a comparison is presented which shows that the
proposed algorithm has better performance in terms of noise
cancellation as well as the convergence rate. So to conclude,
this paper is successful in combining the merits of the Sub-
band framework to apply in an ANC environment.
VI. REFERENCES
[1] S. J. Elliot, Signal Processing for Active Control, London, U.K.:
Academic Press, 2001.
[2] Asutosh Kar, Ambika Prasad Chanda, Mahesh Chandra, “An Improved
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Suppression”,Volume-1, pp.25-32,2014,Springer-Verlag,Germany.
[3] Asutosh Kar, Mahesh Chandra “A Minimized Complexity Dynamic
Structure Adaptive Filter Design for Improved Steady State Performance
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[4] Asutosh Kar and Mahesh Chandra “A Novel Variable Tap-length
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