International Journal of Engineering and Advanced Technology (IJEAT)
ISSN: 2249 – 8958, Volume-1, Issue-3, February 2012
“Performance Evaluation of Different Adaptive
Filtering Algorithms for Reduction of Heart
Sound from Lung Sound”
R. M. Potdar, Anup Mishra, Vinni Sharma, Tripti Roy
Abstract—Auscultation is the most important and effective
clinical technique for evaluating a patient’s respiratory
function. Auscultation of the chest is a diagnostic method
used by physicians, owing to its simplicity and
noninvasiveness. Hence, there is interest in lung sound
analysis using time and frequency domain techniques to
increase its usefulness in diagnosis. This proposed work is
focused on the application of adaptive filtering technique to
separate heart sound signal from lung sound signal. Lung
sound signal measurements are taken to aid in the diagnosis
of various diseases. The aim of this proposed work is to
filtering heart sounds from lung sounds. In medicine this
separation is made by doctors individually. This may lead to
some errors in listening the lung sounds. The method we will
use during this separation process is adaptive filtering. We will
use Matlab basically while doing mathematical calculations
and filtering methods.
Keywords : Auscultation, lung sound, heart sound, adaptive
filtering, different adaptive algorithms.
I. INTRODUCTION
Since the 1950s, techniques and equipment have
been developed that improve the detection and analysis of
respiratory sounds. Monitoring of the respiratory activity
is clinically useful in many patients affected by cardiac,
pulmonary and neurological diseases [6]. Direct
percussion had only been introduced shortly before,
palpation serving to evaluate transmission of vocalized
vibrations from the trachea through to the chest wall [1].
Generally, all LSSs originate from airways during
inspiration-expiration cycles. The LSS propagates
through lung tissues in the parenchyma and can be heard
over the chest wall using a sound transducer [2].
Manuscript Details February 2012
Mr. R. M. Potdar , Sr. Associate Professor ,Department of Electronics
& Telecommunication Engineering, Bhilai Institute of Technology
(BIT), Durg , Chhattisgarh, India, (E-mail ID -
ravi_potdar@rediffmail.com ).
Dr. Anup Mishra, Professor & Head, Department of Electrical &
Electronics Engineering, Bhilai Institute of Technology (BIT), Durg ,
Chhattisgarh, India.
Mrs. Vinni Sharma, . Associate Professor ,Department of Electronics
& Telecommunication Engineering, Bhilai Institute of Technology
(BIT), Durg , Chhattisgarh, India, (E-mail ID –
sharma.vinni@gmail.com ).
Ms. Tripti Roy , Assistant Professor ,Department of Electrical &
Electronics Engineering, Maha Maya College of Technology (MMCT),
Raipur, Chhattisgarh, India, Mobile – 09977717051, (E-mail ID –
triptiroy8401@gmail.com).
61
It has been shown that the intensity of breath
sounds increases with increasing airflow, [3] however,
the time and frequency domain combination of heart
sounds are an inherent interference in lung sound
analysis, for the simple fact that the heart cannot be made
to cease beating [4]. The main frequency components of
HSS are in the range 20-100 Hz. This is the range in
which LSS has major components [5]. In our proposed
work, we use different adaptive filtering algorithms for
separating heart sound from lung sounds with a reference
signal highly correlated to the noise component (i.e. heart
sound here in our work) of the input signal (i.e. lung
sound corrupted by heart sound here in our work) and not
correlated at all with the desired information component
of the primary input (i.e. lung sound in our work).
II. METHODOLOGY
A Lung sounds : Generally, lung sounds are produced
during inspiration and expiration cycles, and are found in
the frequency range 20-1200Hz. There are two types of
lung sound, namely - normal and abnormal lung sounds.
Normal breath sounds can be categorized into three
classes: bronchial, bronchovesicular, and vesicular
sounds. Each class of sounds is detected during
auscultation according to the characteristics described
below:
• Bronchial: Normally heard over the manubrium or
upper part of the sternum. Consist of high pitch and high
amplitude sounds similar to the sound of air blowing
through a tube. There is a brief pause between expiration
and inspiration, where expiration is three times as long as
inspiration. Abnormal if heard over the entire lung (where
vesicular sounds should be heard), indicating that the
lung is consolidated.
• Bronchovesicular: Heard over the anterior chest
between the scapulae and sternum, between the first and
second intercostal spaces. They possess pitch and
intensity characteristics midway between vesicular and
bronchial sounds. They can be heard during the
inspiratory and expiratory phases, which each last for
approximately the same duration and do not have a pause
in between them.
• Vesicular: Soft and low-pitch breezy sound heard over
the majority (and particularly over the base) of the lung.
Occurs as a result of changing airflow patterns in the
lungs. Inspiration is more audible than expiration. There
Performance Evaluation of Different Adaptive Filtering Algorithms for Reduction of Heart Sound from Lung Sound
is no pause between the two phases, but inspiration is
three times as long as expiration. Similar to the sound of
wind blowing through a tunnel.
B Heart sounds : Auscultation of the heart can reveal up
to four heart sounds :
S1: (First heart sound) “Lub” sound of tricuspid and
mitral valves closing and is heard the loudest at the apex
of the heart.
S2: (Second heart sound) “Dub” sound of pulmonic and
aortic valve closing and is lou flowing to the ventricles.
Sometimes there can be a “split S2” if there is a delay
between the two valves closing. The length of the delay
(or split) varies according to whether it is inspiration or
expiration phase.
S3: Follows the S2 and occurs from the vibrations of the
ventricle walls when the ventricles fill quickly with
blood.
S4: Second stage of ventricle filling and immediately
precedes S1. Occurs from the vibrations of the valves and
ventricle walls. A fourth heart sound may occur during
atrial systole where blood is forced into the ventricles. If
the ventricle is stiff, the force of blood entering the
ventricle is more vigorous, and the result is an impact
sound in late diastole, S4.
C Adaptive Filter : The so-called adaptive filter, is the use
of the result of the filter parameters a moment ago,
automatically adjust the filter parameters of the present
moment, to adapt to the unknown signal and noise[8].
Fig.1 Adaptive Filter
d[n] = Primary input (Lung sound contaminated with
heart sound)
d[n] = s[n] + x1[n]
s[n] = Lung sound
x1[n] = Heart sound
x[n] = Reference input (Heart sound)
y[n] = Adaptive filter output
e[n] = Output of the system or error signal
e[n] = d[n] – y[n]
The basic concept of adaptive noise canceling is shown
in figure. The main signal d[n] picked up by the main
sensors contains useful signal s[n] and interference signal
x1[n]. In order to remove noise from the main signal,
make use of the reference sensors to pick up related noise
and use for the input x[n] of adaptive filter. The
interference signal x1[n] is uncorrelated with the signal.
The combined signal and noise (d[n] = s[n] +x1[n]) form
the primary input to the canceller. A second sensor
receives a noise x[n] uncorrelated with the signal s[n] but
correlated in some unknown way with the noise x1[n].
This sensor provides the reference input to the canceller.
62
The noise x[n] is filtered to produce an output y[n] that is
as close a replica as possible of x1[n]. This output is
subtracted from the primary input d[n] to produce the
output of the system (e(n) = d(n) – y(n)). In the system
shown in figure, the reference input is processed by an
adaptive filter, which differs from the regular or fixed
filter. The adaptive filter automatically adjusts its own
impulse response through an algorithm that responds to
an error signal, which depends on the filter’s output. In
the noise canceling application the objective is to produce
an error signal that is a best fit in the least squares sense
to the signal s[n]. This is accomplished by feeding back
the system output to the adaptive filter and adjusting the
filter through an adaptive algorithm (LMS algorithm,
NLMS algorithm and RLS algorithm) to minimize the
total system output power. In an adaptive noise-canceling
system, in other words, the system output serves as the
error signal for the adaptive process.
Assume that s[n], x1[n], x[n], and y[n] are
statistically stationary and have zero means. Assume that
s[n] is uncorrelated with x1[n] and x[n], and suppose that
x[n] is correlated with x1[n].
The output e[n] is
e[n]=s[n]+x1[n]–y[n] (1)
Squaring both sides
e2[n]=s2[n]+[x1[n]–y(n)]2+2s(n)[x1[n]–y(n)] (2)
Taking the expected value for both sides, and realizing
that s[n] is uncorrelated with x1[n] and x[n] and with y[n]
results in
E[e2[n]] = E[s2[n]] + E[[(x1[n] – y[n])2] +
2E[s[n][x1[n] – y[n]]]
=E[s2[n]]+E[(x1[n]–y[n])2] (3)
The signal power E[s2[n]] will be unaffected as the
filter is adjusted to minimize E[e²[n]]. The minimum
output power will be given by
min E[e2[n]]=E[s2[n]]+min
E[s[n][x1[n]–y[n]]2] (4)
When the filter is adjusted so that E[e2] is minimized,
E[(x1[n] – y[n])2] is also minimized. The filter output
y[n] is then a best least squares estimate of the primary
noise x1[n]. Also when E[(No(n) – y(n))2] is minimized,
E[(e(n) – s(n))2] is also minimized, and that leads to
[e[n]–s[n]] ≈ [x1[n]–y[n]] (5)
The output e[n] will contain the signal s[n] plus noise.
From (1), the output noise is given by [x1[n] – y[n]].
Since minimizing E[e2] minimizes E[(x1 – y)2], and
since the signal in the output remains constant,
minimizing the total output power maximizes the output
signal-to-noise ratio. From (3), it seems that the smallest
possible output power is E[e2] = E[s2], and that will lead
to E[(x1 - y)2] ≈ 0. Therefore, y[n] = x1[n], and e[n] =
s[n]. In this case, minimizing output power causes the
output signal to be noise free[7].
III. RESULTS
However the diagnosing the lung sound through
auscultation is subjective and is largely depends upon the
hearing ability and skill of physician. The proposed work
is an attempt to make a tool for adaptively filtering the
 International Journal of Engineering and Advanced Technology (IJEAT)
ISSN: 2249 – 8958, Volume-1, Issue-3, February 2012

heart sound signal from lung sound signal to recognize 1 1 RLS 30.4937 28.1673 30.2877 27.6102 25.3076 2 7 .4 1 35
which adaptive filtering algorithm is found to be the most 1 2 QRDRLS 12.7940 10.7851 13.7428 12.7784 10.7468 1 3 .7 3 18
HSWR
useful alternative. The heart sound and lung sound were 13 43.6695 43.1486 46.0696 40.6704 40.1512 43.0668
LS
obtained from the web. SWR 32.92 32.40 35.31 29.95 29.44 32.34
14
The 4 types of sound signals are used for this LS 68 70 96 85 15 06
30.49 28.16 30.28 27.61 25.30 27.41
work. 3 Lung sound signals and one normal heart sound 15 HRLS
37 73 77 02 76 35
signal. The 3 Lung sound signals are 19.85 17.71 19.76 17.64 15.64 17.67
16 FTF
1. Bronchial lung sound. 24 00 28 53 18 20
2. Bronchovesicular lung sound. SWFT 32.24 31.66 34.66 29.19 28.68 31.57
17
F 55 99 62 71 06 62
3. Vesicular lung sound. 18 19.32 16.80 19.01 17.64 14.96 17.25
These signals are taken for the free data base AP
77 80 15 57 14 96
available in internet. The three lung sound signals are 19
APRU
19.32 16.80 19.01 17.64 14.96 17.25
77 80 15 57 14 96
sampled at sampling frequency 11025 Hz and the
20 Does Does Does
standard heart sound signal is sampled at sampling 20.01 17.20 19.55
BAP not not not
16 84 00
frequency 8012 Hz. For evaluations, first we chose first work work work
8000 samples of the lung sound signals and the standard 21 30.44 28.12 30.23 27.52 25.22 27.31
FDAF
92 33 63 42 29 42
heart sound signal. The lung sound signals are degraded 22 Does Does Does
by heart sound signal. The adaptive filter structure is used PBFD 27.52 25.22 27.31
not not not
AF 42 29 42
to evaluate the performance of different adaptive filtering work work work
algorithm. The input signal to the filter is lung sound 23 Does Does Does
PBUF 24.67 22.41 24.47
not not not
signal corrupted by standard heart sound signal. The DAF 08 05 17
work work work
reference input is a signal correlated with standard heart 24
TDAF
Does Does Does
10.25 10.71 13.52
sound signal. The output of the filter is the heart sound not not not
DCT 36 11 58
work work work
recovered lung sound signal. Table 1.1, 1.2 and 1.3 shows 25 TDAF 10.51 10.80 13.58 10.23 10.66 13.51
the signal to noise ratio, peak signal to noise ratio and DFT 46 92 80 87 26 84
normalized root mean square error for the entire data set 26 UFD 27.52 25.22 27.31 25.88 23.60 25.67
AF 42 29 42 23 32 39
considered. We compare the performances of different
27 Does Does Does
adaptive filtering algorithms on the basis of SNR, PSNR 27.66 25.36 27.47
GAL not not not
78 04 41
and NRMSE by varying the length of filter from 1 to 6. work work work
For evaluating performance of the method both objective 28 Does Does Does
13.91 11.97 14.47
LSL not not not
and subjective tests are conducted. In objective test, SNR work work work
85 18 73
of signal after adaptive filtering is computed. Other three 29 Does Does Does
QRD 13.91 11.97 14.47
parameters used for comparing results are time required not not not
LSL 85 18 73
work work work
for reconstruction of signal, peak signal to noise ratio
normalized root mean square error between lung sound
signal corrupted by heart sound signal and adaptively Table 1.2: Peak signal to noise ratio for different lung
filtered signal. Different adaptive filtering algorithms are sounds
applied on the three lung sound signals and effort is made S.NO FILT
to remove additive heart sound signal from the lung . ER
L=1 L=2
sound signal corrupted by heart sound signal. LEN
GTH
Table 1.1: Signal to noise ratio for different lung sounds ADA Bron Bron Vesic Bron Bron Vesic
PTIV chial chov ular chial chov ular
FILT E Lung esicu Lung Lung esicu Lung
ER ALG Soun lar Soun Soun lar Soun
L=1 L=2
LEN ORI d Lung d d Lung d
GTH THM Soun Soun
ADA Bron Bron S d d
S.NO PTIV Bron chov Vesic Bron chov Vesic NLM 42.29 36.98 40.39 38.07 32.36 35.96
1
E chial esicu ular chial esicu ular S 68 07 33 07 60 24
ALG Lung lar Lung Lung lar Lung BLM 60.19 55.39 58.57 54.36 49.54 52.73
2
S 69 03 61 20 21 70
ORI Soun Lung Soun Soun Lung Soun
DLM 60.43 55.58 58.77 54.59 49.73 52.93
THM d Soun d d Soun d 3
S 59 74 30 82 87 24
S d d 60.19 55.39 58.57 54.36 49.54 52.73
1 NLMS 21.4760 19.0434 21.2014 19.3392 16.8175 1 9 .0 1 70 4 LMS
69 03 61 20 21 70
2 BLMS 30.4492 28.1233 30.2363 27.5242 25.2229 2 7 .3 1 42 ADJL 60.31 55.49 58.68 54.64 49.78 53.00
3 DLMS 30.6626 28.3136 30.3797 27.7334 25.4109 2 7 .4 5 43 5
MS 76 06 50 86 91 29
4 LMS 30.4492 28.1233 30.2363 27.5242 25.2229 2 7 .3 1 42 BLM 60.19 55.39 58.57 54.36 49.54 52.73
5 ADJLMS 30.5135 28.1882 30.2844 27.6906 25.3954 2 7 .4 5 27 6
SFFT 69 03 61 20 21 70
6 BLMSFFT 30.4492 28.1233 30.2363 27.5242 25.2229 2 7 .3 1 42 FILT
7 FILTXLMS 30.4989 28.1757 30.2749 27.6631 25.3715 2 7 .4 3 48 60.29 55.47 58.67 54.61 49.76 52.97
7 XLM
8 S D 22.1895 16.6230 19.6334 19.7969 14.3913 1 7 .3 1 64 82 60 05 20 09 50
S
9 S E 30.4341 19.9908 21.8461 27.5187 17.5675 1 9 .3 4 00 43.82 31.79 37.11 39.07 27.02 32.37
8 SD
1 0 S S 20.5605 11.6222 14.5021 17.6581 10.5830 1 4 .8 9 68 88 65 95 25 79 31

63
Performance Evaluation of Different Adaptive Filtering Algorithms for Reduction of Heart Sound from Lung Sound

59.19 38.72 41.56 53.33 33.68 36.45 7 FILT 0.001 0.003 0.002 0.003 0.006 0.004
9 SE
00 75 61 58 91 07 XLM 7 5 1 2 7 0
40.11 22.44 25.57 34.17 20.66 26.15 S
10 SS
62 73 99 11 00 50 8 SD 0.011 0.053 0.025 0.019 0.091 0.043
60.29 55.49 58.67 54.54 49.73 52.93 0 1 1 0 9 3
11 RLS
22 22 78 62 98 38 9 SE 0.001 0.023 0.015 0.003 0.042 0.027
QRD 25.03 20.63 23.82 25.01 20.59 23.79 9 9 0 7 7 1
12
RLS 01 95 36 13 49 48 10 SS 0.016 0.155 0.094 0.033 0.191 0.088
HSW 84.86 85.85 88.82 78.84 79.80 82.76 9 7 6 5 3 5
13
RLS 42 79 86 73 28 23 11 RLS 0.001 0.003 0.002 0.003 0.006 0.004
SWR 63.37 64.37 67.33 57.41 58.37 61.31 7 5 1 2 7 1
14
LS 68 11 23 98 62 73 12 QRD 0.095 0.191 0.115 0.096 0.192 0.116
60.29 55.49 58.67 54.54 49.73 52.93 RLS 9 7 8 1 7 2
15 HRLS
22 22 78 62 98 38 13 HSW 0.000 0.000 0.000 0.000 0.000 0.000
39.06 34.22 37.43 34.69 29.92 33.13 RLS 1 1 1 2 2 1
16 FTF
55 58 13 66 10 67 14 SWR 0.001 0.001 0.000 0.002 0.002 0.001
SWF 61.98 62.88 65.99 55.89 56.85 59.79 LS 2 2 8 3 5 5
17
TF 84 08 04 64 30 01 15 HRLS 0.001 0.003 0.002 0.003 0.006 0.004
38.02 16.80 35.92 34.72 14.96 32.33 7 5 1 2 7 1
18 AP
92 80 28 72 14 64 16 FTF 0.019 0.040 0.024 0.031 0.065 0.039
APR 38.02 16.80 35.92 34.72 14.96 32.33 1 1 2 5 9 6
19
U 92 80 28 72 14 64 17 SWF 0.001 0.001 0.000 0.002 0.003 0.001
Does Does Does TF 4 5 9 7 0 8
39.41 33.19 37.05
20 BAP not not not 18 AP 0.021 0.049 0.028 0.031 0.077 0.043
45 13 73
work work work 5 8 8 4 7 5
60.19 55.39 58.57 54.36 49.54 52.73 19 APR 0.021 0.049 0.028 0.031 0.077 0.043
21 FDAF
69 03 61 20 21 70 U 5 8 8 4 7 5
Does Does Does 20 BAP Does Does Does 0.018 0.045 0.025
PBFD 54.36 49.54 52.73
22 not not not not not not 3 2 2
AF 20 21 70
work work work work work work
Does Does Does 21 FDAF 0.001 0.003 0.002 0.003 0.006 0.004
PBUF 48.66 43.82 47.03
23 not not not 7 5 1 3 9 2
DAF 42 77 29
work work work 22 PBFD Does Does Does 0.003 0.006 0.004
Does Does Does AF not not not 3 9 2
TDA 28.97 20.37 23.46
24 not not not work work work
FDCT 63 17 83
work work work 23 PBUF Does Does Does 0.006 0.013 0.008
TDA 24.08 20.59 23.77 28.95 20.31 23.41 DAF not not not 3 3 0
25
FDFT 68 25 45 76 36 19 work work work
UFD 54.36 49.54 52.73 51.08 46.25 49.45 24 TDA Does Does Does 0.182 0.197 0.120
26
AF 20 21 70 87 45 95 FDCT not not not 4 7 6
Does Does Does work work work
54.66 49.85 53.05
27 GAL not not not 25 TDA 0.170 0.192 0.116 0.182 0.199 0.121
74 87 42
work work work FDFT 0 8 5 8 1 4
Does Does Does 26 UFD 0.003 0.006 0.004 0.004 0.010 0.006
27.43 22.88 26.06
28 LSL not not not AF 3 9 2 8 0 1
46 45 61
work work work 27 GAL Does Does Does 0.003 0.006 0.004
Does Does Does not not not 2 6 0
QRD 27.43 22.88 26.06
29 not not not work work work
LSL 46 45 61
work work work 28 LSL Does Does Does 0.072 0.148 0.089
not not not 7 1 5
work work work
Table 1.3: Normalized root mean square error for 29 QRD Does Does Does 0.072 0.148 0.089
LSL not not not 7 1 5
different lung sounds work work work
S.NO FILT L=1 L=2
. ER
LEN
GTH
ADA Bron Bron Vesic Bron Bron Vesic
PTIV chial chov ular chial chov ular
E Lung esicu Lung Lung esicu Lung
ALG Soun lar Soun Soun lar Soun
ORI d Lung d d Lung d
THM Soun Soun
S d d
1 NLM 0.013 0.029 0.017 0.021 0.049 0.028
S 1 2 2 4 7 6
2 BLM 0.001 0.003 0.002 0.003 0.006 0.004
S 7 5 1 3 9 2
3 DLM 0.001 0.003 0.002 0.003 0.006 0.004
S 6 4 1 2 7 1
4 LMS 0.001 0.003 0.002 0.003 0.006 0.004
7 5 1 3 9 2
5 ADJL 0.001 0.003 0.002 0.003 0.006 0.004 Fig.1.2: Result of bronchial lung sound corrupted by
MS 7 5 1 2 7 0 heart sound and filtered by hswrls adaptive filtering
6 BLM 0.001 0.003 0.002 0.003 0.006 0.004 algorithm at filter length, L=1.
SFFT 7 5 1 3 9 2

64
 International Journal of Engineering and Advanced Technology (IJEAT)
ISSN: 2249 – 8958, Volume-1, Issue-3, February 2012

Fig. 1.3: Result of bronchovesicular lung sound

corrupted by heart sound and filtered by hswrls adaptive
filtering algorithm at filter length L=1. Fig.1.7: Comparison between normalized root mean
square error for different adaptive filtering algorithm for
bronchial lung sound.

Fig.1.4: Result of vesicular lung sound corrupted by

heart sound and filtered by hswrls adaptive filtering
algorithm at filter length L=1.
Fig. 1.8: Comparison between signal to noise ratio for
different adaptive filtering algorithm for
bronchovesicular lung sound.

Fig. 1.5: Comparison between signal to noise ratio for

different adaptive filtering algorithm for bronchial lung
sound. Fig. 1.9: Comparison between peak signal to noise ratio
for different adaptive filtering algorithm for
bronchovesicular lung sound.

Fig. 1.6: Comparison between peak signal to noise ratio

for different adaptive filtering algorithm for bronchial Fig. 1.10: Comparison between normalized root mean
lung sound. square error for different adaptive filtering algorithm for
bronchovesicular lung sound.

65
Performance Evaluation of Different Adaptive Filtering Algorithms for Reduction of Heart Sound from Lung Sound
Fig. 1.11: Comparison between signal to noise ratio
for different adaptive filtering algorithm for vesicular
lung sound.
Fig.1.12: Comparison between peak signal to noise
ratio for different adaptive filtering algorithm for
vesicular lung sound.
Fig. 1.13: Comparison between normalized root
mean square error for different adaptive filtering
algorithm for vesicular lung sound.
on odd filter length because we take even number of input
samples i.e. 8000 in our proposed work they are pbfdaf
and pbufdaf.
We also observed from table that some adaptive
filtering algorithms give same result they are blms, lms
and blmsfft in LMS algorithm, rls and hrls in RLS
algorithm, ap and apru in affine projection algorithm, fdaf
66
and pbfdaf for even filter length i.e. for L=2, 4, 6, lsl and
qrdlsl in lattice algorithm.
IV DISCUSSION
Adaptive algorithms have been extensively
studied in the past few dacades and have been widely
used in many areas including biomedical, image,
communication etc. and many other applications.
From the table we observed that the hswrls adaptive
filtering algorithm gives the maximum values of the
parameters for filter length 1 to 6. It is observed that
the hswrls gives the maximum SNR, PSNR and
minimum NRMSE for the three lung sounds.
We observed from the table that some
adaptive filter algorithms can not work on filter
length, s L=1, they are bap, tdafdct, gal, lsl, qrdlsl.
Again we observed from table that some adaptive
filtering algorithms can not work
ACKNOLEDGMENT
Mr. R. M. Potdar , Sr. Associate Professor (ETC) Bhilai
Institute of Technology Durg, Chhattisgarh, one of the
authors is indebted to give permission for sending this
paper to the journal. Mrs. Vinni Sharma Associate
Professor (ETC) Bhilai Institute of Technology Durg,
Chhattisgarh also thankful for giving me permission to
send the paper for publication. We would also like to
thanks our colleagues for supporting us.
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– 1235, September 2007.
[3] J. Gnitecki, Z. Moussavi, H. Pasterkamp, “ Recursive Least Square
Adaptive Noise Cancellation Filtering for Heart Sound Reduction
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[5] Yang- Sheng Lu, Wen- Hui Liu, Guang- Xia Qin, “ Removal of The
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[6] M. Varanini, M. Emdin, F. Allegri, M. Raciti, F. Conforti, A.
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[7] Mohamed A. Abdulmagid, Dean J. Krusienski, Siddharth Pal, and
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 International Journal of Engineering and Advanced Technology (IJEAT)
ISSN: 2249 – 8958, Volume-1, Issue-3, February 2012

Sr. Associate Professor Mr. R. M.

Potdar, received B.E. Electronics,
M.Tech. (Hons) Instrumentation &
Control. He is having a total teaching
experiences of 14 years and industrial
experiences of 5 years.His interests
are in Image Processing, Neural
Network & Fuzzy Logic System
Design. His specialization subjects are
Adaptive Control System, Optimal
Control System, Control System
Design, Satellite communication and
Optical Communication. He has
published the papers in 6 international
journals and 2 national journals and also attended 1 international
conference and 11 national conferences. Also he is having Life
Membership of Indian Society of Technical Education, India (ISTE).

Dr Anup Mishra Professor & Head

Electrical & Electronics Department
Bhilai Institute Of Technology, Durg,
Chhattisgarh. He Has 15 Years of
teaching experience, his area of
specialization is Network Analysis,
Digital Electronics, Power Electronics,
Signals & Systems, Material Science. He
has published two national and two
papers in international journals.

Associate Professor Mrs. Vinni

Sharma, received B.E. E&TC, M.Tech.
Instrumentation & Control. She is having
a total teaching experiences of 9.2 years
and industrial experiences of 6 months.
She has published the papers in 4 national
journals and also attended 3 national
conferences. Also she is having Life
Membership of Indian Society of
Technical Education, India (ISTE).

Ms. Tripti Roy, received B.E.

Electrical, & pursuing M.Tech.
Instrumentation & Control from Bhilai
Institute of Technology(BIT), Durg,
Chhattisgarh, India. Her interest is in
Biomedical Technology, Power
System, Protection and Switchgear,
High Voltage.

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