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IOSR Journal of Electronics and Communication Engineering (IOSR-JECE)
e-ISSN: 2278-2834,p- ISSN: 2278-8735. Volume 7, Issue 4 (Sep. - Oct. 2013), PP 83-87
www.iosrjournals.org
www.iosrjournals.org 83 | Page
A Decisive Filtering Selection Approach For Improved
Performance Active Noise Cancellation System
Abha Nyati1
, S K Dargar2
, **Deepak Vyas
1
(Department of E & C Engineering, Pacific University, (PAHER) Udaipur, Rajasthan, India)
2
(Faculty, Department of E & C Engineering, SPSU University, Udaipur, Rajasthan, India)
**
(Faculty, Department of E & C Engineering, Pacific Institute of Technology, Udaipur, Rajasthan, India)
Abstract : In this work we present a filtering selection approach for efficient ANC system. Active noise
cancellation (ANC) has wide application in next generation human machine interaction to automobile Heating
Ventilating and Air Conditioning (HVAC) devices. We compare conventional adaptive filters algorithms LMS,
NLMS, VSLMS, VSNLMS, VSLSMS for a predefined input sound file, where various algorithms run and result
in standard output and better performance. The wiener filter based on least means squared (LMS) algorithm
family is most sought after solution of ANC. This family includes LMS, NLMS, VSLMS, VSNLMS, VFXLMS, FX-
sLMS and many more. Some of these are nonlinear algorithm, which provides better solution for nonlinear
noisy environment. The components of the ANC systems like microphones and loudspeaker exhibit
nonlinearities themselves. The nonlinear transfer function create worse situation. This is a task which is some
sort of a prediction of suitable solution to the problems. The Radial Basis Function of Neural Networks (RBF
NN) has been known to be suitable for nonlinear function approximation [1]. The classical approach to RBF
implementation is to fix the number of hidden neurons based on some property of the input data, and estimate
the weights connecting the hidden and output neurons using linear least square method. So an efficient novel
decisive approach for better performing ANC algorithms has been proposed.
Keywords - Adaptive filters, Winner filter ANC, Least mean square, N LMS, VSNLMS, RBF.
I. OVERVIEW
Acoustic Noise Cancellation is a method for reducing undesired noise. It is achieved by introducing a
canceling “anti-noise” wave through secondary sources. These secondary sources are interconnected through an
electronic system using a specific signal processing algorithm for the particular cancellation scheme. Noise
cancellation makes use of the notion of destructive interference. When two sinusoidal waves are superimposed,
the resultant waveform depends on the frequency amplitude and relative phase of the two waves. If the original
wave and the inverse of the original wave encounter at a junction at the same time, total cancellation occurs. The
challenges are to identify the original signal and generate the inverse noise cancellation on inverse wave without
delay in all directions where noises interact and superimpose. We meet in our everyday life. Echo phenomena
are interesting and entertaining, but their presences in communication networks are undesirable and represent a
serious problem. Echo is delayed and degraded version of original signal which travels back to its source after
several reflections or because of some other reason. Nature of echo signal can be either acoustic or electrical,
and in order to reduce its undesired effect we employ echo cancellers. Design of echo cancellers requires an
application of adaptive filter theory. Echo cancellers must work within specific time limits so adaptive
algorithms must provide fast convergence of filter parameters. We‟ve been applying echo cancellers
successfully for many years, but we always tend to improve them and increase their efficiency.
The wiener filter based least means squared (LMS) algorithm family is most sought after solution of
ANC. This family includes LMS , Fx-LMS, VFx-LMS, FsLMS and many more . Then there are Kalman filter
algorithms which are basically based on recursive least square algorithm. Some of these are nonlinear algorithm,
which provides better solution for non linear noisy environment. The components of the ANC systems like
microphones and loudspeaker exhibit nonlinearities themselves. The non linear transfer function of primary and
secondary path may itself create worse situation.
II. BACKGROUND LITERATURE REVIEW
Active Noise Control (ANC) includes an electromechanical system or electro acoustic system that
cancels the unwanted noise based on the principle of superposition of an anti-noise wave of equal amplitude.
The customary ways for active noise cancellers uses passive techniques such as enclosures, barriers, and
silencers to attenuate the undesired noise. These passive silencers are valued for their high attenuation over a
broad frequency range. However, they are relatively large, costly, and ineffective at low frequencies.
Mechanical vibration is another related type of noise that commonly creates problems in all areas of
transportation and manufacturing, as well as with many household appliances. S.M. Kuo and D. R. Morgan
A Decisive Filtering Selection Approach for Improved Performance Active Noise Cancellation System
www.iosrjournals.org 84 | Page
(1996) in their paper have emphasized the practical aspects of ANC systems in terms of adaptive algorithms and
DSP implementations for real-world applications [4]. Widrow and Hoff (1960) developed the least mean square
algorithm (LMS). This algorithms is a member of stochastic gradient algorithm algorithms, and because of
robustness and low computational complexity, it has been used in wide spectrum of applications [6]. Least
square solution is not very practical in the actual implementation of adaptive filters, this is because we know all
the past samples of the input signal, and as well the desired output signal must be available at every iteration.
Performance with less computational complexity compared to the Second-order VFXLMS algorithm [9]. Debi
Prasad das and G. panda in 2004 has shown a new approach for nonlinear processes using Filtered-s LMS
(FsLMS). They proved that using a nonlinear controller for a linear device can achieve superior performance
when the secondary path is linear non minimum phase and the reference noise is non-gaussian and predictable.
III. PROCEDURE AND APPROACH
The experimental work has been done on MATLAB, audio processor, capable to perform real time
processing on signals. Various program based on individual algorithm were written first e.g. LMS, NLMS,
FSLMS, VSLMS. Each of them is considered as a separate hidden neuron of network. In case two primary noise
signal is chosen to be a logistic chaotic type x(n+1)=λx(n)[1-x(n)] λ=4, x(0)=0,.9[2]
All the neurons were allowed to run for 600, 1200 and 1800 iterations, and respective obtained results were
compared with standard threshold value. LMS algorithm weight update equation is as
w(n+1)=w(n)-µe(n)v(n) (1)
v(n)=x(n)*a(n) where a(n) is approximation of secondary path transfer function. w(n) if filter weight , µ is
step size factor[2]. The calculation of threshold depends upon number of parameter and it may vary according to
surrounding situation and instantaneous noise characteristics. We had adopted threshold parameter δk Which
was based on following equations. For each input Xn
δk(Xn) =exp( -1/σ2( ||XN- µK|||2
)) (2)
Where σ is variance, µ is is weight factor of particular neuron [5]. This approach is based on technique of RBF
neural network. We are using this technique to prune less effective neurons from network and finally running
systems with successful neuron only. In different conditions different neuron may give better results. The
survival chances of neurons depend upon its algorithm efficiency in given circumstances. The step µwas size
varied factor between .05, 0.125 and 0.5 depending upon the optimization required between complexity and
amount of noise cancellation. We have done experimental studies on various inputs for different algorithms e.g.
LMS, NLMS, VSLMS, VSNLMS etc. Finally for simulations trial we opted one particular acoustic signal of
linear nature, which was producing a sound in the frequency range of low decibels.
IV. RESULT AND ANALYSIS
0 100 200 300 400 500 600
0.05
0.055
0.06
0.065
0.07
0.075
0.08
Mean square error convergence for LMS algorithm for step size =0.125
Number of Iteration
MeanSquareError
0 100 200 300 400 500 600
-1
-0.8
-0.6
-0.4
-0.2
0
0.2
0.4
0.6
0.8
1
MSE for NLMS algorithm for step size 0.125
No. of iterations
MeanSquareError
0 200 400 600 800 1000 1200
-1.5
-1
-0.5
0
0.5
1
1.5
MSE for NLMS algorithm for step size 0.125
No. of iterations
MeanSquareError
0 100 200 300 400 500 600
0
0.002
0.004
0.006
0.008
0.01
0.012
0.014
0.016
0.018
0.02
MSE for VSLMS algorithm for the step size 0.125
No. of iterations
MeanSquareError
A Decisive Filtering Selection Approach for Improved Performance Active Noise Cancellation System
www.iosrjournals.org 85 | Page
0 100 200 300 400 500 600
0
0.002
0.004
0.006
0.008
0.01
0.012
0.014
0.016
0.018
0.02
MSE for VSNLMS algortihm with step size range 0.125
rm=iterations
MeanSquareError
0 200 400 600 800 1000 1200 1400 1600 1800
0
0.005
0.01
0.015
0.02
0.025
MSE for VSLMS algorithm for the step size 0.125
No. of iterations
MeanSquareError
Table1. MSE of Various algorithms for number of iterations=600, 1200 and 1800.
Iteration Applied Algorithm Step Size Mean error
Avg ERLE
(dB)
600 LMS 0.050 0.05163 -12.8710
600 LMS 0.125 0.03505 -14.5531
600 LMS 0.500 0.09041 -10.4378
1200 LMS 0.050 0.04132 -13.8384
1200 LMS 0.125 0.05299 -12.7581
1200 LMS 0.500 0.08390 -10.7624
1800 LMS 0.050 0.03000 -15.2288
1800 LMS 0.125 0.04297 -13.6683
1800 LMS 0.500 0.07039 -11.5249
600 Normalized-LMS 0.050 0.01177 -19.2922
600 Normalized-LMS 0.125 0.04031 -13.9459
600 Normalized-LMS 0.500 0.05047 -12.9697
1200 Normalized-LMS 0.050 0.01780 -17.4958
1200 Normalized-LMS 0.125 0.02354 -16.2819
1200 Normalized-LMS 0.500 0.02983 -15.2535
1800 Normalized-LMS 0.050 0.06708 -11.7341
1800 Normalized-LMS 0.125 0.02847 -15.4561
1800 Normalized-LMS 0.500 0.02574 -15.8939
600 Variable step size-LMS 0.050 0.02059 -16.8634
600 Variable step size-LMS 0.125 0.01673 -17.7650
600 Variable step size-LMS 0.500 0.02059 -16.8634
1200 Variable step size-LMS 0.050 0.01630 -17.8781
1200 Variable step size-LMS 0.125 0.02016 -16.9551
1200 Variable step size-LMS 0.500 0.02039 -16.9058
1800 Variable step size-LMS 0.050 0.03980 -14.0012
1800 Variable step size-LMS 0.125 0.03672 -14.3510
1800 Variable step size-LMS 0.500 0.02120 -16.7366
600 VS Normalized-LMS 0.050 0.00168 -27.7547
600 VS Normalized-LMS 0.125 0.00212 -26.7469
600 VS Normalized-LMS 0.500 0.02246 -16.4859
1200 VS Normalized-LMS 0.050 0.01277 -18.9381
1200 VS Normalized-LMS 0.125 0.01537 -18.1333
1200 VS Normalized-LMS 0.500 0.01893 -17.2285
1800 VS Normalized-LMS 0.050 0.00281 -25.5098
1800 VS Normalized-LMS 0.125 0.02989 -15.2447
1800 VS Normalized-LMS 0.500 0.03098 -15.0892
A Decisive Filtering Selection Approach for Improved Performance Active Noise Cancellation System
www.iosrjournals.org 86 | Page
Figure 3. Graphical analysis of variations in mean error for algorithms for number of iterations=600, 1200 and 1800.
V. COMPUTATION AND COMPLEXITY
Application of RBF neural network criteria to eliminate non efficient neuron where neuron is the
different algorithm, for the selection of proficient algorithm can be selected for ANC. The performance of the
algorithm depends upon the type of input. We found VSNLMS is doing better corresponding to the input
„sound‟. Here we used and apply the criteria for deciding threshold value [1]. And the simplified formula is
given as
( 3)
where µk(x) is response of kth
hidden unit. In our case, implementation of the concept of RBF neural network
approach can be used after simplification σ = 1 , and mean square error µ variance is normalized by factor 1/20
Threshold criteria results in selection of the efficiently working neuron as efficient algorithm can be given in
simplified manner, where Th is the threshold value and MSE is Mean square error 1/20 is the normalizing factor
as :
(4)
VI. CONCLUSION AND FUTURE WORK
It was found out those in particular noisy conditions most suitable algorithm can be sort out using
decisive method using Radial basis function in ANC method. This method can also be applied to noise
cancellation, vibration cancellation and signal estimation problems. Average of the MSE obtained 0.3175
deviations from mean when calculated; it is observed that Variable step size Normalized LMS is performing
better than the other utilized application for ANC of “Sound” input for no. of iterations 1800 with step size 0.5
as the threshold criteria results =Mean error-(1/20)Average. 0.0068 is the least value amongst the all applied
algorithm, hence pruning of all other algorithm inculcate the selection of that algorithm as survival neuron.
Next good performing algorithm is NLMS itself for 600 iterations of step size 0.05. The surprisingly good
performance of the LMS is due to linear nature of the „sound‟ input noise. Here in this project have tried to
reduce such weed signals by using basic adaptive approach. But in real time applications which involve
thousands of such weed signals and knowing the fact that noise, at any cost, cannot be cancelled to cent percent,
this basic approach cannot do us a good favour and keeps this research topic active in upcoming years. The
future task is to apply this novel concept for Kalman and Wiener filter‟s family algorithm simultaneously. It is
also being tried to implement it for outer noise removal in automobiles and noise cancellation in head phones.
Noise performance can be upgraded by finding proper step factor and threshold value. The algorithm level up
gradation of threshold and decisive criteria may improve performance
ACKNOWLEDGEMENTS
We pay our special thanks to Mr. Himanshu Purohit, Prof. D. Das, for sharing good research ideas and
suggestions. We wish to put on record the appreciative original work of all the authors of various technical
papers which we have referred in initiation of work without whom it was very difficult to achieve successful
completion.
A Decisive Filtering Selection Approach for Improved Performance Active Noise Cancellation System
www.iosrjournals.org 87 | Page
REFERENCES
[1] S.M. Kuo and D.R.Morgan, Active Noise Control –Algorithms and DSP Implementations. New York: Wiley, 1996.
[2] Lu Yingwei, N.Sundararajan, P.Saratchandran “Adaptive Nonlinear Systems Identification Using Minimal Radial Basis Function
Neural Networks “in IEEE 1996.
[3] Shoji Makino, Member, IEEE, Yutaka Kaneda, Member, IEEE and Nobuo Koizumi, “Exponentially weighted step size NLMS
adaptive filter based on the statistics of a room impulse response”, IEEE Trans. on speech and audio Processing, vol. 1, No.1,
pp.101-108, Jan 1993.
[4] S. Qureshi, “Adaptive Equalization,” Proceedings of the IEEE, vol.73, No.9, pp.1349-1387, Sept. 1985.
[5] S. Haykins, “Adaptive Filter Theory”, Prentice Hall, New Jersey, 1996.
[6] Raymond H. Kwong and Edward W. Johnston, “A Variable Step Size LMS Algorithm”, IEEE Transactions on Signal Processing vol.
40, No.7, pp. 1633-1642, July, 1992.
[7] Tyseer Aboulnasr, Member, IEEE, and K.Mayyas, “A robust variable step size LMS type algorithm: Analysis and Simulations”,
IEEE Trans. On Signal Processing vol. 45, No.3, pp. 631-639, March 1997.
[8] Dargar S K, Himanshu Purohit, Mahajan “An Iterative Pruning Approach of Neural Network for Proficient Noise Cancellation” in
International Journal Engg Advanced Technology, ISSN: 2249 – 8958, Volume-2, Issue-4, April 2013.
[9] Debi Prasad Das ,”Filtered –s LMS Algorithm for Multichannel Active control Of Nonlinear Noise Processes” in IEEE Transactions
on Speech And Audio Processing , vol 14 ,no.5 September 2006.
[10] Debi Das and Ganpati Panda, “Active Mitigation of Nonlinear Noise Processes Using A Novel Filtered-s LMS Algorithm” in IEEE
Transactions of Speech and Audio Processing, vol. 12, No.3, May 2004.
Abha Nyati1
, is M. Tech scholar from Pacific University PAHER, Udaipur pursuing specialization in Digital
Communication. She has completed her degree of Bachelor of Engineering in electronics and communication with
honors from Mohanlal Sukhadia University, Rajasthan. She has good academic records and has academic
experience in teaching. Her area of research interest is digital communication, signal processing and wireless
communication.
Shashi Kant Dargar2
, is working as assistant professor in the department of Electronics and Communication
Engineering at Sir Padampat Singhania University, Udaipur. Presently, and pursuing Ph.D. He has completed his
degree of Bachelor of Engineering from University of Rajasthan, Jaipur in specialization with Electronics and
Communication Engineering. He has completed his degree of Masters of Technology in Digital Communication
from Rajasthan Technical University, Kota. He has been contributing in academics from seven years. His Area of
research interest is signal processing, wireless communication, Microwave & Communication Engineering.
**Deepak Vyas, is working as assistant professor in the department of Electronics and Communication Engineering
at Pacific Institute of Technology, Udaipur. Presently, he is pursuing Ph.D. He has completed his degree of Bachelor
of Engineering in specialization with Electronics and Communication Engineering. He has completed his degree of
Masters of Technology in Digital Communication from Rajasthan Technical University, Kota. He has been
contributing in academics form over seven years. His Area of research interest is Communication Engineering. He
has guided many projects for undergraduate and post graduate students in his academic career.

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A Decisive Filtering Selection Approach For Improved Performance Active Noise Cancellation System

  • 1. IOSR Journal of Electronics and Communication Engineering (IOSR-JECE) e-ISSN: 2278-2834,p- ISSN: 2278-8735. Volume 7, Issue 4 (Sep. - Oct. 2013), PP 83-87 www.iosrjournals.org www.iosrjournals.org 83 | Page A Decisive Filtering Selection Approach For Improved Performance Active Noise Cancellation System Abha Nyati1 , S K Dargar2 , **Deepak Vyas 1 (Department of E & C Engineering, Pacific University, (PAHER) Udaipur, Rajasthan, India) 2 (Faculty, Department of E & C Engineering, SPSU University, Udaipur, Rajasthan, India) ** (Faculty, Department of E & C Engineering, Pacific Institute of Technology, Udaipur, Rajasthan, India) Abstract : In this work we present a filtering selection approach for efficient ANC system. Active noise cancellation (ANC) has wide application in next generation human machine interaction to automobile Heating Ventilating and Air Conditioning (HVAC) devices. We compare conventional adaptive filters algorithms LMS, NLMS, VSLMS, VSNLMS, VSLSMS for a predefined input sound file, where various algorithms run and result in standard output and better performance. The wiener filter based on least means squared (LMS) algorithm family is most sought after solution of ANC. This family includes LMS, NLMS, VSLMS, VSNLMS, VFXLMS, FX- sLMS and many more. Some of these are nonlinear algorithm, which provides better solution for nonlinear noisy environment. The components of the ANC systems like microphones and loudspeaker exhibit nonlinearities themselves. The nonlinear transfer function create worse situation. This is a task which is some sort of a prediction of suitable solution to the problems. The Radial Basis Function of Neural Networks (RBF NN) has been known to be suitable for nonlinear function approximation [1]. The classical approach to RBF implementation is to fix the number of hidden neurons based on some property of the input data, and estimate the weights connecting the hidden and output neurons using linear least square method. So an efficient novel decisive approach for better performing ANC algorithms has been proposed. Keywords - Adaptive filters, Winner filter ANC, Least mean square, N LMS, VSNLMS, RBF. I. OVERVIEW Acoustic Noise Cancellation is a method for reducing undesired noise. It is achieved by introducing a canceling “anti-noise” wave through secondary sources. These secondary sources are interconnected through an electronic system using a specific signal processing algorithm for the particular cancellation scheme. Noise cancellation makes use of the notion of destructive interference. When two sinusoidal waves are superimposed, the resultant waveform depends on the frequency amplitude and relative phase of the two waves. If the original wave and the inverse of the original wave encounter at a junction at the same time, total cancellation occurs. The challenges are to identify the original signal and generate the inverse noise cancellation on inverse wave without delay in all directions where noises interact and superimpose. We meet in our everyday life. Echo phenomena are interesting and entertaining, but their presences in communication networks are undesirable and represent a serious problem. Echo is delayed and degraded version of original signal which travels back to its source after several reflections or because of some other reason. Nature of echo signal can be either acoustic or electrical, and in order to reduce its undesired effect we employ echo cancellers. Design of echo cancellers requires an application of adaptive filter theory. Echo cancellers must work within specific time limits so adaptive algorithms must provide fast convergence of filter parameters. We‟ve been applying echo cancellers successfully for many years, but we always tend to improve them and increase their efficiency. The wiener filter based least means squared (LMS) algorithm family is most sought after solution of ANC. This family includes LMS , Fx-LMS, VFx-LMS, FsLMS and many more . Then there are Kalman filter algorithms which are basically based on recursive least square algorithm. Some of these are nonlinear algorithm, which provides better solution for non linear noisy environment. The components of the ANC systems like microphones and loudspeaker exhibit nonlinearities themselves. The non linear transfer function of primary and secondary path may itself create worse situation. II. BACKGROUND LITERATURE REVIEW Active Noise Control (ANC) includes an electromechanical system or electro acoustic system that cancels the unwanted noise based on the principle of superposition of an anti-noise wave of equal amplitude. The customary ways for active noise cancellers uses passive techniques such as enclosures, barriers, and silencers to attenuate the undesired noise. These passive silencers are valued for their high attenuation over a broad frequency range. However, they are relatively large, costly, and ineffective at low frequencies. Mechanical vibration is another related type of noise that commonly creates problems in all areas of transportation and manufacturing, as well as with many household appliances. S.M. Kuo and D. R. Morgan
  • 2. A Decisive Filtering Selection Approach for Improved Performance Active Noise Cancellation System www.iosrjournals.org 84 | Page (1996) in their paper have emphasized the practical aspects of ANC systems in terms of adaptive algorithms and DSP implementations for real-world applications [4]. Widrow and Hoff (1960) developed the least mean square algorithm (LMS). This algorithms is a member of stochastic gradient algorithm algorithms, and because of robustness and low computational complexity, it has been used in wide spectrum of applications [6]. Least square solution is not very practical in the actual implementation of adaptive filters, this is because we know all the past samples of the input signal, and as well the desired output signal must be available at every iteration. Performance with less computational complexity compared to the Second-order VFXLMS algorithm [9]. Debi Prasad das and G. panda in 2004 has shown a new approach for nonlinear processes using Filtered-s LMS (FsLMS). They proved that using a nonlinear controller for a linear device can achieve superior performance when the secondary path is linear non minimum phase and the reference noise is non-gaussian and predictable. III. PROCEDURE AND APPROACH The experimental work has been done on MATLAB, audio processor, capable to perform real time processing on signals. Various program based on individual algorithm were written first e.g. LMS, NLMS, FSLMS, VSLMS. Each of them is considered as a separate hidden neuron of network. In case two primary noise signal is chosen to be a logistic chaotic type x(n+1)=λx(n)[1-x(n)] λ=4, x(0)=0,.9[2] All the neurons were allowed to run for 600, 1200 and 1800 iterations, and respective obtained results were compared with standard threshold value. LMS algorithm weight update equation is as w(n+1)=w(n)-µe(n)v(n) (1) v(n)=x(n)*a(n) where a(n) is approximation of secondary path transfer function. w(n) if filter weight , µ is step size factor[2]. The calculation of threshold depends upon number of parameter and it may vary according to surrounding situation and instantaneous noise characteristics. We had adopted threshold parameter δk Which was based on following equations. For each input Xn δk(Xn) =exp( -1/σ2( ||XN- µK|||2 )) (2) Where σ is variance, µ is is weight factor of particular neuron [5]. This approach is based on technique of RBF neural network. We are using this technique to prune less effective neurons from network and finally running systems with successful neuron only. In different conditions different neuron may give better results. The survival chances of neurons depend upon its algorithm efficiency in given circumstances. The step µwas size varied factor between .05, 0.125 and 0.5 depending upon the optimization required between complexity and amount of noise cancellation. We have done experimental studies on various inputs for different algorithms e.g. LMS, NLMS, VSLMS, VSNLMS etc. Finally for simulations trial we opted one particular acoustic signal of linear nature, which was producing a sound in the frequency range of low decibels. IV. RESULT AND ANALYSIS 0 100 200 300 400 500 600 0.05 0.055 0.06 0.065 0.07 0.075 0.08 Mean square error convergence for LMS algorithm for step size =0.125 Number of Iteration MeanSquareError 0 100 200 300 400 500 600 -1 -0.8 -0.6 -0.4 -0.2 0 0.2 0.4 0.6 0.8 1 MSE for NLMS algorithm for step size 0.125 No. of iterations MeanSquareError 0 200 400 600 800 1000 1200 -1.5 -1 -0.5 0 0.5 1 1.5 MSE for NLMS algorithm for step size 0.125 No. of iterations MeanSquareError 0 100 200 300 400 500 600 0 0.002 0.004 0.006 0.008 0.01 0.012 0.014 0.016 0.018 0.02 MSE for VSLMS algorithm for the step size 0.125 No. of iterations MeanSquareError
  • 3. A Decisive Filtering Selection Approach for Improved Performance Active Noise Cancellation System www.iosrjournals.org 85 | Page 0 100 200 300 400 500 600 0 0.002 0.004 0.006 0.008 0.01 0.012 0.014 0.016 0.018 0.02 MSE for VSNLMS algortihm with step size range 0.125 rm=iterations MeanSquareError 0 200 400 600 800 1000 1200 1400 1600 1800 0 0.005 0.01 0.015 0.02 0.025 MSE for VSLMS algorithm for the step size 0.125 No. of iterations MeanSquareError Table1. MSE of Various algorithms for number of iterations=600, 1200 and 1800. Iteration Applied Algorithm Step Size Mean error Avg ERLE (dB) 600 LMS 0.050 0.05163 -12.8710 600 LMS 0.125 0.03505 -14.5531 600 LMS 0.500 0.09041 -10.4378 1200 LMS 0.050 0.04132 -13.8384 1200 LMS 0.125 0.05299 -12.7581 1200 LMS 0.500 0.08390 -10.7624 1800 LMS 0.050 0.03000 -15.2288 1800 LMS 0.125 0.04297 -13.6683 1800 LMS 0.500 0.07039 -11.5249 600 Normalized-LMS 0.050 0.01177 -19.2922 600 Normalized-LMS 0.125 0.04031 -13.9459 600 Normalized-LMS 0.500 0.05047 -12.9697 1200 Normalized-LMS 0.050 0.01780 -17.4958 1200 Normalized-LMS 0.125 0.02354 -16.2819 1200 Normalized-LMS 0.500 0.02983 -15.2535 1800 Normalized-LMS 0.050 0.06708 -11.7341 1800 Normalized-LMS 0.125 0.02847 -15.4561 1800 Normalized-LMS 0.500 0.02574 -15.8939 600 Variable step size-LMS 0.050 0.02059 -16.8634 600 Variable step size-LMS 0.125 0.01673 -17.7650 600 Variable step size-LMS 0.500 0.02059 -16.8634 1200 Variable step size-LMS 0.050 0.01630 -17.8781 1200 Variable step size-LMS 0.125 0.02016 -16.9551 1200 Variable step size-LMS 0.500 0.02039 -16.9058 1800 Variable step size-LMS 0.050 0.03980 -14.0012 1800 Variable step size-LMS 0.125 0.03672 -14.3510 1800 Variable step size-LMS 0.500 0.02120 -16.7366 600 VS Normalized-LMS 0.050 0.00168 -27.7547 600 VS Normalized-LMS 0.125 0.00212 -26.7469 600 VS Normalized-LMS 0.500 0.02246 -16.4859 1200 VS Normalized-LMS 0.050 0.01277 -18.9381 1200 VS Normalized-LMS 0.125 0.01537 -18.1333 1200 VS Normalized-LMS 0.500 0.01893 -17.2285 1800 VS Normalized-LMS 0.050 0.00281 -25.5098 1800 VS Normalized-LMS 0.125 0.02989 -15.2447 1800 VS Normalized-LMS 0.500 0.03098 -15.0892
  • 4. A Decisive Filtering Selection Approach for Improved Performance Active Noise Cancellation System www.iosrjournals.org 86 | Page Figure 3. Graphical analysis of variations in mean error for algorithms for number of iterations=600, 1200 and 1800. V. COMPUTATION AND COMPLEXITY Application of RBF neural network criteria to eliminate non efficient neuron where neuron is the different algorithm, for the selection of proficient algorithm can be selected for ANC. The performance of the algorithm depends upon the type of input. We found VSNLMS is doing better corresponding to the input „sound‟. Here we used and apply the criteria for deciding threshold value [1]. And the simplified formula is given as ( 3) where µk(x) is response of kth hidden unit. In our case, implementation of the concept of RBF neural network approach can be used after simplification σ = 1 , and mean square error µ variance is normalized by factor 1/20 Threshold criteria results in selection of the efficiently working neuron as efficient algorithm can be given in simplified manner, where Th is the threshold value and MSE is Mean square error 1/20 is the normalizing factor as : (4) VI. CONCLUSION AND FUTURE WORK It was found out those in particular noisy conditions most suitable algorithm can be sort out using decisive method using Radial basis function in ANC method. This method can also be applied to noise cancellation, vibration cancellation and signal estimation problems. Average of the MSE obtained 0.3175 deviations from mean when calculated; it is observed that Variable step size Normalized LMS is performing better than the other utilized application for ANC of “Sound” input for no. of iterations 1800 with step size 0.5 as the threshold criteria results =Mean error-(1/20)Average. 0.0068 is the least value amongst the all applied algorithm, hence pruning of all other algorithm inculcate the selection of that algorithm as survival neuron. Next good performing algorithm is NLMS itself for 600 iterations of step size 0.05. The surprisingly good performance of the LMS is due to linear nature of the „sound‟ input noise. Here in this project have tried to reduce such weed signals by using basic adaptive approach. But in real time applications which involve thousands of such weed signals and knowing the fact that noise, at any cost, cannot be cancelled to cent percent, this basic approach cannot do us a good favour and keeps this research topic active in upcoming years. The future task is to apply this novel concept for Kalman and Wiener filter‟s family algorithm simultaneously. It is also being tried to implement it for outer noise removal in automobiles and noise cancellation in head phones. Noise performance can be upgraded by finding proper step factor and threshold value. The algorithm level up gradation of threshold and decisive criteria may improve performance ACKNOWLEDGEMENTS We pay our special thanks to Mr. Himanshu Purohit, Prof. D. Das, for sharing good research ideas and suggestions. We wish to put on record the appreciative original work of all the authors of various technical papers which we have referred in initiation of work without whom it was very difficult to achieve successful completion.
  • 5. A Decisive Filtering Selection Approach for Improved Performance Active Noise Cancellation System www.iosrjournals.org 87 | Page REFERENCES [1] S.M. Kuo and D.R.Morgan, Active Noise Control –Algorithms and DSP Implementations. New York: Wiley, 1996. [2] Lu Yingwei, N.Sundararajan, P.Saratchandran “Adaptive Nonlinear Systems Identification Using Minimal Radial Basis Function Neural Networks “in IEEE 1996. [3] Shoji Makino, Member, IEEE, Yutaka Kaneda, Member, IEEE and Nobuo Koizumi, “Exponentially weighted step size NLMS adaptive filter based on the statistics of a room impulse response”, IEEE Trans. on speech and audio Processing, vol. 1, No.1, pp.101-108, Jan 1993. [4] S. Qureshi, “Adaptive Equalization,” Proceedings of the IEEE, vol.73, No.9, pp.1349-1387, Sept. 1985. [5] S. Haykins, “Adaptive Filter Theory”, Prentice Hall, New Jersey, 1996. [6] Raymond H. Kwong and Edward W. Johnston, “A Variable Step Size LMS Algorithm”, IEEE Transactions on Signal Processing vol. 40, No.7, pp. 1633-1642, July, 1992. [7] Tyseer Aboulnasr, Member, IEEE, and K.Mayyas, “A robust variable step size LMS type algorithm: Analysis and Simulations”, IEEE Trans. On Signal Processing vol. 45, No.3, pp. 631-639, March 1997. [8] Dargar S K, Himanshu Purohit, Mahajan “An Iterative Pruning Approach of Neural Network for Proficient Noise Cancellation” in International Journal Engg Advanced Technology, ISSN: 2249 – 8958, Volume-2, Issue-4, April 2013. [9] Debi Prasad Das ,”Filtered –s LMS Algorithm for Multichannel Active control Of Nonlinear Noise Processes” in IEEE Transactions on Speech And Audio Processing , vol 14 ,no.5 September 2006. [10] Debi Das and Ganpati Panda, “Active Mitigation of Nonlinear Noise Processes Using A Novel Filtered-s LMS Algorithm” in IEEE Transactions of Speech and Audio Processing, vol. 12, No.3, May 2004. Abha Nyati1 , is M. Tech scholar from Pacific University PAHER, Udaipur pursuing specialization in Digital Communication. She has completed her degree of Bachelor of Engineering in electronics and communication with honors from Mohanlal Sukhadia University, Rajasthan. She has good academic records and has academic experience in teaching. Her area of research interest is digital communication, signal processing and wireless communication. Shashi Kant Dargar2 , is working as assistant professor in the department of Electronics and Communication Engineering at Sir Padampat Singhania University, Udaipur. Presently, and pursuing Ph.D. He has completed his degree of Bachelor of Engineering from University of Rajasthan, Jaipur in specialization with Electronics and Communication Engineering. He has completed his degree of Masters of Technology in Digital Communication from Rajasthan Technical University, Kota. He has been contributing in academics from seven years. His Area of research interest is signal processing, wireless communication, Microwave & Communication Engineering. **Deepak Vyas, is working as assistant professor in the department of Electronics and Communication Engineering at Pacific Institute of Technology, Udaipur. Presently, he is pursuing Ph.D. He has completed his degree of Bachelor of Engineering in specialization with Electronics and Communication Engineering. He has completed his degree of Masters of Technology in Digital Communication from Rajasthan Technical University, Kota. He has been contributing in academics form over seven years. His Area of research interest is Communication Engineering. He has guided many projects for undergraduate and post graduate students in his academic career.