International Journal on Recent and Innovation Trends in Computing and Communication

Volume: 4 Issue: 2

ISSN: 2321-8169
115 - 119

______________________________________________________________________________________

A Novel Method for the Application of Adaptive filters

for Active Noise Control System
Pradyumna Ku.Mohapatra

Jibanananda Mishra

Asistant Professor
Department of Electronics &Telecommunication
Orissa Engg. College,BBSR,Odisha
[email protected]

Associate Professor
Department of Electronics & Telecommunication
Orissa Engg. College, BBSR, Odisha

Abstract This paper introduces one novel method for active noise control .Though use of filtered-X LMS FIR Adaptive Filter mature in the
literature ,this expression illustrates the application of adaptive filters to the attenuation of acoustic noise via active noise control. The reference
signal is a noisy version of the undesired sound measured near its source. We shall use a controller filter length of about 44 msec and a step
size of 0.0001 for these signal statistics. The resulting algorithm converges after about 5 seconds of adaptation. We also realize adaptive
algorithm using IIR filter with active noise to overcome the ability of acoustic feedback . The direct form IIR filter structure, which faces the
difficulties of checking stability and of relatively slow convergence speed for noise composed of narrow band components with large power
inequality. To overcome these difficulties along with using the direct form IIR filters filtered-u LMS algorithm is used.
Keywords- Active Noise, Acoustic noise cancellation, Filtered-XLMS algorithm.

__________________________________________________*****_________________________________________________
I.

INTRODUCTION (HEADING 1)

Using an electro-acoustic system dimension sensors such as

microphones and output actuators such as speakers one can
reduce the intensity of an unwanted dB noise propagating
through the air. Usually the noise signal comes from any piece
of equipment, such as a revolving machine, so that it is
possible to measure the noise near its source. The end of the
active noise control scheme is to create an"anti-noise" that
attenuates the unwanted noise in a desired settle down region
using an adaptive filter. In view of the fact that of that
inadequacy of the physical barriers, active means to cut low
frequency noise (less than 500-1000 Hertz) have been look
into by researchers in the area of adaptive acoustic control.
Active noise control (ANC) plays an important role in which
noise can be reduced in the form of a small package of a DSP
controller, microphone(s), and loudspeaker(s). The ANC
systems are effective,it may be better or worse only when the
intended noise is periodic and so random noises like the white
noise will not be reduced. This problem differs from
traditional adaptive noise cancellation in that:

Only the attenuated signal is available but the

most wanted response signal cannot be directly
measured.
The secondary loudspeaker-to microphone error
path in its adaptation the ANC system must take
into account.
The application of adaptive filters to the attenuation of
acoustic noise via active noise control FXLMS FIR Adaptive
Filter is used. The path in which the anti-noise takes from the
output loudspeaker to the error microphone within the quiet

zone is called as secondary propagation path. Here we

considered as the impulse response which is band limited to
the range 160 - 2000 Hz with a filter length
of 0.1 seconds. For this, ANC We shall use a sampling
frequency of 8000 Hz. The structure of the ANC is normally
classified into two classes: feed forward control and feedback
control. In the feed forward control, a reference noise is
presumed to be usable to the adaptive filter. Depending on the
type of primary noise that can be shortened feed forward ANC
systems can be categorized as either a broadband or a
narrowband. In the broadband feed forward control case, a
reference noise is observed by a reference sensor
(e.g.,microphone), and thus noise correlating with the
reference noise can be scaled down. For ANC, IIR adaptation
has been used as filteredU algorithm but this combines the
main path adaptation with the cancellation of the reverse
feedback from secondary source to reference input [1].
II.

ADAPTIVE ANC SYSTEM

The ANC system using filter x-LMS algorithm are shown in

figure-1. The system is excited with i/p signal x (n) and output
of adaptive filter y (n) to cancel the primary noise at the error
microphone location. Though the use of Filtered-X LMS FIR
Adaptive Filter is mature in literature, adaptive filters are
defined for problems such as electrical noise cancelling where
the filter output is an estimate of a preferred signal. In many
applications, the adaptive filter works as a dynamic system
Controller which containing actuators and amplifiers etc. The
estimate (anti-vibrations or anti-sound) in this case can thus be
seen as the output signal from a dynamic system, i.e. a forward
path. A conventional adaptive algorithm such as the LMS
115

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International Journal on Recent and Innovation Trends in Computing and Communication

Volume: 4 Issue: 2

ISSN: 2321-8169
115 - 119

______________________________________________________________________________________
algorithm is likely to be unstable in this application due to the
phase shift (the delay) introduced by the forward path [1, 2].
The well-known filtered-x LMS-algorithm is, however, an
adaptive filter algorithm which is suitable for active control
applications [3].

estimation error of up to 0 90 , within the limit of slow

adaptation [2]. The maximum allowable step size for FXLM
algorithm
is
approximately[5]

Where

is the mean square value or power

of the filtered reference signal

N is the number of adaptive filter coefficients and

Equivalent sampled-time block diagram of the broadband

feedforwardANC systemshown in Fig. 1. In this figure, P (z) is the primary
path, S(z) is thesecondary path,W(z) is the control filter, and S(z) is the
secondary-pathmodel..

A.The FXLMS algorithm: Figure-1 shows the block

diagram of an ANC system in which the FXLMS algorithm is
applied.The output of the adaptive filter i.e y(n) can be obtain
as
Where w(n )=[
coefficient of signal vector

III.
the

and L is the filter length.

The adaptive filter updates coefficient of signal vector

using FXLMS algorithm which can be expressed as
Where is the
convergence

number of samples corresponding to the overall delay in the

secondary path. The the most significant factor which
influencing the convergence behavior of the ANC system is
delay in the secondary path, thus reducing the maximum step
size in the FXLMS algorithm.
B. Feedback path ANC :. The mere approach to solving the
feedback problem is to use a feedback cancellation filter that
models the feedback path from the secondary loud speaker to
the reference sensor, which is precisely the same technique
used in acoustic echo cancellation [2]. In the broadband feed
forward ANC, when a feedback path is present, the optimal
solution is generally an infinite impulse response (IIR)
function with poles and zeros. The purpose of the IIR filter can
be considerably more efficient for the realization than FIR
filters, because an IIR filter may require much fewer
coefficients than an FIR filter to model any resonance systems.

and
are

convergence factor that determines the

speed.

is the filtered reference signal vector and

is the impulse

response of the secondary-path estimation filter. The accurate

estimation of the secondary-path model the ANC system
requires FXLMS algorithm. This algorithm shows that when
secondary path, S(z) , follows the adaptive filter, this transfer
function must also be placed in the reference signal path. The
basic ANC system described above perform quite well in
reducing broadband as well as narrowband noise in ducts
under plane wave conditions. In this arrangement primary path
and secondary path are assumed to be linear in nature and so
represented by linear transfer functions. To implement
FXLMS algorithm the secondary path filter S(z) is to be
estimated first. Many offline and online methods are available
for identifying the secondary path filter. FXLMS algorithm
found to be tolerant to errors made in the estimation of
.

is the

ANC USES IIR FILTER

IIR adaptive filters have the ability to give up matching

Characteristics with lesser filter coefficients compared to FIR
adaptive filters. Fall in number of filter coefficients leads to
decrease
in
computational
complexity
for
ANC
implementation. ANC introduces poles in the system in the
case of acoustic feedback. So IIR adaptive filters can better
match the physical system as they have zeros as well as poles
whereas FIR filters have only zeros. The ANC using an IIR
filter is mature in the literature.The algorithm used in this
paper for IIR adaptive filter is called to filter-u recursive LMS
algorithm. The only drawback of this algorithm is that even
though experimentally it works well, the stability and global
convergence is not guaranteed [4].
IV.

SIMULATION OF ANC USING FXLMS:

To give emphasis to the difference we run the system with no

active noise control for the first 200 iterations. Listening to its
sound at the error microphone before cancellation, it has the
quality industrial "whine" of such motors. Once the adaptive
filter is enabled, the resulting algorithm converges after about
5 (simulated) seconds of adaptation. Comparing the spectrum
of the residual error signal with that of the original noise
signal, we see that most of the periodic components have been
attenuated considerably. The steady-state cancellation
performance may not be uniform across all frequencies,

FXLMS algorithm generally converge even with a phase

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_______________________________________________________________________________________

International Journal on Recent and Innovation Trends in Computing and Communication

Volume: 4 Issue: 2

ISSN: 2321-8169
115 - 119

______________________________________________________________________________________
however. Such is often the case for real-world systems applied
to active noise control tasks. Listening to the error signal, the
annoying "whine" is reduced considerably.

Primary Path Impulse Response

0.25
0.2
0.15

Coefficient value

0.1

Step-1:Generate sine wave with random phase.

Step-2: Generate synthetic noise by adding all sine waves.
Step-3:Propagate noise through primary path.
Step-4:Add measurement noise
Step-5:No noise control for first 200 iterations.
Step-6: Play noise signal.
Step-7: Show spectrum of original and attenuated noise.

0.05
0
-0.05
-0.1
-0.15
-0.2
-0.25

0.01

0.02

0.03

0.04
0.05
0.06
Time [sec]

0.07

0.08

0.09

0.1

Fig 5:Primary path impulse response

True Secondary Path Impulse Response

0.3

0.1

-10

-20

Power/frequency (dB/Hz)

Coefficient value

Power Spectral Density of the Noise to be Cancelled

0.2

-0.1

-0.2

-0.3

-0.4

0.01

0.02

0.03

0.04
0.05
0.06
Time [sec]

0.07

0.08

0.09

-30

-40

-50

-60

0.1

-70

0.2

0.4

0.6

Fig 2:The secondary path impulse response

0.8
1
1.2
Frequency (kHz)

1.4

1.6

1.8

Fig 6: PSD of the Noise to be cancelled

Secondary Identification Using the NLMS Adaptive Filter

5
Desired Signal
Output Signal
Error Signal

4
3

Active Noise Control Using the Filtered-X LMS Adaptive Controller

3
Original Noise
Anti-Noise
Residual Noise

Signal value

Signal value

0
-1

-1

-2
-2

-3
-4
-5

-3

0.5

1.5
Number of iterations

2.5

-4
0

x 10

Fig 3:The secondary identification using NLMS

3
Number of iterations

6
4

x 10

Fig 7:ANC Using the FXLM adaptive Controller

Secondary Path Impulse Response Estimation

0.25

Power Spectral Density of the Original and Attenuated Noise

True
Estimated
Error

0.2
0.15

0
Original
Attenuated

-10

Power/frequency (dB/Hz)

Coefficient value

0.1
0.05
0
-0.05
-0.1

-20

-30

-40

-50

-0.15
-60

-0.2
-0.25

0.01

0.02

0.03

0.04
0.05
0.06
Time [sec]

0.07

0.08

0.09

0.1

Fig 4:Secondary path impulse response estimation

-70

200

400

600

800
1000 1200
Frequency (Hz)

1400

1600

1800

2000

Fig 8:PSD of the Original and Attenuated Noise

117
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_______________________________________________________________________________________

International Journal on Recent and Innovation Trends in Computing and Communication

Volume: 4 Issue: 2

ISSN: 2321-8169
115 - 119

______________________________________________________________________________________
V.

SIMULATION OF ANC ANC USING

FIR FILTER

VI.

The limitations of FXLMS algorithm is the magnitude

response and the convergence rate of the adaptation since it
depends on loop gain of adaptation path. So the proposed
New filtered MFXLMS AlGORITHM is used to overcome
not only the said problem but to achieve more predictive
power. In this case We choose number of taps of filter is 20.
step-size parameter mu=0.001. We have to compute reference
singals d1= x*Hp1(Z) and d2=x*Hp2(Z). Aso compute
filteroutput s1=r1*w and s2=r2*w where r1=x*Hs1(z)and
r2=x*Hs2(z) update weights and plot noise and filtered signal

SIMULATION OF ANC USING

IIR FILTER
Generate signal

1
0.5
0
-0.5
-1

0.1

0.2

0.3

0.4

0.5

0.6

0.7

0.8

0.9

0.1

0.2

0.3

0.4

0.5

0.6

0.7

0.8

0.9

2
1
0
-1
-2

0.9
0.8

0.2

0.7

0.1

0.6

0.5

-0.1

0.4

-0.2

0.1

0.2

0.3

0.4

0.5

0.6

0.7

0.8

0.9

0.1

0.2

0.3

0.4

0.5

0.6

0.7

0.8

0.9

0.3

1.5

0.2

1
0.1

0.5
0

0.5

1.5

2.5

-0.5

Fig9 : Generate signal

reference singal d1
0.6

0.6

0.4

0.4

0.2

0.2

-0.2

-0.2

0.5

1.5

Fig12: Noise and filtered signals using IIR Filter

VII.
0

0.5

filteroutput s1
0

-0.1

-0.1

-0.2

-0.2

-0.3

-0.3

0.5

1.5

filteroutput s2

-0.4

-1

reference singal d2

1.5

-0.4

0.5

1.5

Fig10: Noise and filtered signals

error signal e1
0.5
0
-0.5
0.5
0
-0.5
0.5
0
-0.5

0.2

0.4

0.6

0.8
1
1.2
error signal e2

1.4

1.6

1.8

0.2

0.4

0.6

0.8

1.4

1.6

1.8

0.2

0.4

0.6

0.8

1.4

1.6

1.8

1
1.2
error signal

1.2

This paper deals with active noise control using a filtered-X

LMS FIR Adaptive Filter. We also realize adaptive algorithm
using IIR filter with active noise to overcome the ability of
acoustic feedback .We have to concentrate more to design
efficient filter structures based on new topologies to deal with
the ANC problems. Different novel methods based on
evolutionary and bio-inspired techniques be developed with
the basic purpose to optimally train the weights of the adaptive
filter structures.
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0.5
0
-0.5

[3]
0

0.2

0.4

0.6

0.8

1.2

1.4

1.6

1.8

Fig 11: Error signals

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IJRITCC | February 2016, Available @ http://www.ijritcc.org

_______________________________________________________________________________________

International Journal on Recent and Innovation Trends in Computing and Communication

Volume: 4 Issue: 2

ISSN: 2321-8169
115 - 119

______________________________________________________________________________________
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