2018 International Conference on Computing, Mathematics and Engineering Technologies – iCoMET 2018
Analysis of Signal Noise Reduction by Using Filters
Hina Magsi
Department Of Electrical Engineering
Sukkur Institute of Business Administration, University
Sukkur, Pakistan
hina.mece17@iba-suk.edu.pk
Faheem Akhtar Chachar
Department Of Electrical Engineering
Sukkur Institute of Business Administration, University
Sukkur, Pakistan
faheem.akhtar@iba-suk.edu.pk
Abstract: This paper designs and compares three different
filtering methods to reduce the effects of noise in medical health
applications such as an electrocardiogram (ECG). The low pass
filter excludes the noise at a low-level, moving average filter takes
average values of the signal, and Finite Impulse Response (FIR)
removes the high frequency components from the ECG and gives
the low frequency component which is the desired information
signal. Simulation results reveal that FIR filter performs better by
reducing the attenuation in the ECG signal than low pass filter and
moving average filters and is appropriate for the medical health
applications.
Keywords: noise, attenuation, sampling, filter, low-pass, moving
average, FIR.
HEC Islamabad and Sukkur IBA, University, Sukkur, Sindh, Pakistan.
978-1-5386-1370-2/18/$31.00 ©2018 IEEE
Ali Hassan Sodhro
Department Of Electrical Engineering
Sukkur Institute of Business Administration, University
Sukkur, and DISP LAB, University Lumiere Lyon2, Lyon,
France
ali.hassan@iba-suk.edu.pk, alihassan.sodrho@uni-lyon2.fr
Saeed A.Khan Abro
Department of Electrical Engineering
Sukkur Institute of Business Administration, University
Sukkur, Pakistan
saeed.abro@iba-suk.edu.pk
I. INTRODUCTION
One of the most challenging problems in digital signal
processing is to receive the information signal without any loss.
It is important to reduce the attenuation produces by random
noise and improve the performance of the signal. As we know
that digital signal is not natural phenomena, it is generated by
converting the analog signal into a digital signal. The analog
signal is uniformly sampled at a sampling rate/Nyquist rate
fs  2  f max , after that sampled signal is quantized and
encoded to get the final digital signal. When the signal
transmits through the channel, it faces some random
disturbances due to environment or during analog to digital
conversion process which is normally distributed at the time
termed as AWGN (additive white Gaussian noise). This noise
added to the original information signal and produces the errors
in the information. The noise distorts the information signal
and lowers its quality. Hence, the received contains a mixture
of information and news. The main task is to extract the
original information from a signal that is the most difficult. The
statistics of the noise corrupting a signal are unknown in many
situations and changes with time. Moreover, the power of noise
may be greater than the power of the desired signal being
transmitted [8]. In this paper the filters are proposed to reduce
the attenuation level of signal and get the desired signal. The
filter is a device which shapes the signal waveform in a desired
manner. The main purpose of filters in digital signal processing
is to reduce the noise which improves the performance of the
signal and to extract the desired information from the signal.
There are many different types of filters that are used to reduce
the effects of noise. Like a low pass filter which takes the
lower frequencies and rejects the higher frequencies, moving
average filter that takes average values at the time and the FIR
filter which allows low frequency components. These three
types of filters have been proposed to improve the accuracy
and efficiency of the information signal.
The main contribution of this paper is to design the low-
pass, FIR and moving average filters for noise reduction in
ECG signal.
 Remaining of the paper is organized as follows. Section II,
presents related work, Section III, discusses the proposed filters
used in this paper. Section IV proposes experimental results.
Section V presents the comparison of three filters and paper is
concluded in section VI.
II. LITERATURE REVIEW
Many researchers have proposed different algorithms for
noise reduction and extracting original information signal. The
usage of filters has big volume of literature. In this paper, some
of the work has been presented.
Kai Siedenburg and Monika Dorfler [1] propose the audio
denoising by thresholding time-frequency. The general
framework of time-frequency soft-thresholding has been
presented to improve denoising quality. It had been also shown
that simple, non-iterated operators perform better as compared
to cutting edge method when evaluating SNR.
Pragati Agrawal and Jayendra Singh Verma [3] develop
different types of linear and nonlinear filters for noise reduction
from corrupted images. The purpose of using these filters is to
provide better performance and eliminating the impulse noise
or Gaussian noise and remove errors from highly corrupted
images. Radhika Bhagat and Ramandeep Kaur [2], present the
audio filtering using extended high pass filter. There have been
several techniques of audio filtering like spectral subtraction,
Dolby noise reduction, use of low-pass and high pass filter,
FIR and IIR filtering. The different formulas and difference
equations have been designed for efficient implementation of
time varying filter applications. Manjeet Singh and Er. Naresh
Kumar Garg [4] design the digital filters for removal of noise
signal. The time domain and frequency domain representation
of the signal have been performed using fast Fourier transform
technique. The Butterworth filter is used to reduce the noise
from signals with different frequency and ripple factor. Er.
Mannu Singla and Mr. Harpal Singh [5] present the noise
reduction using Butter worth, Chebyshev and Elliptical Filters.
The noise reduction system is dependent on the application. In
some cases, the requirement is to increase the overall speech
quality. Adaptive algorithms are designed to analyze the
waveform of the background no neural noise, then based on the
specific algorithm generate a signal that will either phase shift
or inverts the polarity of the original signal.
Students of Artificial Intelligence Research Unit, faculty of
engineering Malaysia [6] propose and study the methods and
techniques used for noise reduction in audio applications. The
best filter is the Butterworth IIR filter that takes less memory
and gives a flat magnitude response. The best method applied
to it is a Hidden Markov Model. For noise filtering, adaptive
filter is the best choice and it can be used with different method
to improve the performance of the audio signals.
Students of Red Sea University [9] propose the
performance of median filter based on window sizes to remove
Salt and pepper noise in RGB images. The salt, pepper noise is
the type of impulse noise generated when images transmit
through the internet or when analog to digital conversion
occurs. They showed the results which were taken in
MATLAB that when cascading the median filter at the high
level of window sizes it has a good performance in RGB
images at low noise densities and in high noise densities, but
with higher windows sizes it introduced the blurring effect of
the noise.
A group of students from Australia and Bangladesh [8]
presents the performance and analysis the noise reduction
through an adaptive filter using the NLMS algorithm. The
system of adaptive filter depends on the effects of step size,
number of filter coefficients, number of samples and input
noise level by considering a speech signal. The parameters
individually showed best performance having optimum values.
IEEE members Andersen and Marc [7] propose adaptive
time-frequency analysis scheme using an asymmetric window.
This technique is suitable for audio noise reduction in the low
delay of 0 to 4ms and less computational complexity. The
application is the real time sound devices. The reduction in the
delay is obtained by applying a FIR filter to the signal as a gain
and computational complexity reduced by using FFT
technique.
Tahir, Iqbal & Abdul Samee [10] design the removal of
noise using various filters in MRI (Magnetic Resonance
Imaging) brain image. When capturing MRI images, these are
affected by different types of noise. They used different filters
and conclude that to improve the performance of MRI image
depends on the type of the filtering technique which is used.
Median filter gives better result in the MRI brain image as
compared to others.
Salih proposed the effects of low pass filter in noise
reduction of audio signal. He taken the sound sample in wave
format and passed that signal through low pass filter and
analysis the noisy audio signal by taking different frequency
and ripple factor [11]. The filter implements difference
equations. These equations and their function in terms of low
pass filter are explained. He concludes that low pass filter
reduces the noise upto greater extent when it’s implemented.
Djurovic Presented the removal of salt and pepper noise
from digital images. He gives his results for block matching 3D
filtering (BM3D) method to the decision based adaptive
median filter [12]. He further observed that the results are good
and reduced noise level upto 2dB in both grayscale and color
images.
Sonia et al [13] proposed bilateral filter and minimum mean
square error filter for noise reduction. It includes two main
steps one is generation of reference image from noisy image to
extracts the patches and other is to apply bilateral filter to each
patches. This has benefit that it do not affects the image
original appearance. The algorithm has been tested at low,
medium and high noise density to check the performance of
color images. This paper concludes that bilateral filter is best
for removal of salt and pepper noise from images.
Priyanka et al proposed the reduction noise in remote
sensing image by using Kalman filter and recovered the
original image from corrupted image [14]. The Kalman filter is
compared with Wiener filter. The remote sensing image faced
Gaussian noise, salt and pepper noise. To reduce this noise
Kalman filter and wiener filter is used. Kalman filter give
better response and good efficiency in reduction of noise as This signal is sampled at the Nyquist rate which is twice of Original Signal

compared to Wiener filter. maximum frequency (fs=2xfmax=20z). Then the signal is

1 Filtered Signal
120
0.9

Biswas et al designed the technique to improve the 0.8 100

efficiency of Kawahara filter to reduce the noise. The Gabor 0.7

80
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Kuwahara filter has been proposed which decreases the noise

Amplitude (V)
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without harming the edges of information. This filter is

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60
cosine signal
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integrated with Gabor transformation [15]. This model gives 0.5

0.3 40

Amplitude(m)
0

the better results without losing information at edges and gives -0.5
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best quality of images by reducing noise level.
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Sampling
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III. PROPOSED FILTERS FOR NOISE REDUCTION IN ECG -0.5

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SIGNAL 1
Time(sec)
Quantized signal

Many researchers have worked and proposed different

0.5

Amplitude(m)
0

Fig. Encoding with noise and Filtered Signal

exact

types of filters for noise reduction in the ECG signal. The -0.5

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0 20 40 60 80 100
quantized
error

120

results have been taken in MATLAB. As we know that 1

Time(sec)
encoding

information signal is the analog signal continuous in time. To 0.8

0.6

Amplitude(m)
minimize the signal loss the uniform sampling must be taken.
0.4

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We must make sure that quantization error will be minimal.

0
0 500 1000 1500 2000 2500 3000 3500 4000 4500
Time(sec)

This digital signal is now sent through the channel. When the
signal received at the receiver it contains some portion of
AWGN (additive white noise Gaussian noise) which is
combined with the original signal. To extract the original
information from this signal is difficult. The paper proposed
the three different types of filters, i.e. Low pass filter, moving
average filter and FIR filter which reduce the noise from the
signal and original information can be extracted. The low pass
filter is simple and easy to implement. The second filter used is
the moving average filter which is also the simplest and easiest
digital filter. The third filter used is the FIR (finite Impulse
Response) filter which removes higher frequency components
from the signal and gives the output as the low frequency
components which is the required original signal.
IV. SIMULATION RESULTS
The Fig: 1 represent the analog to digital conversion. The
information signal is taken at a frequency of 10Hz.
Fig. 2 Original Signal with noise
quantized at 3 bits and 120 samples. The green points in the
graph of quantization show the quantization error. Finally, the
cosine signal is encoded. The conversion is analyzed without
considering any type of noise. A noiseless channel is assumed,
which means the ideal condition is followed while converting
an analog signal to digital one.
A. SIGNAL WITH NOISE
As in practical condition the channel faces the noise
(random signal). Now, the noisy channel has been taken by
producing AWGN (Additive White Gaussian Noise). The
graph shows the original signal and noise signal. As in the
graph it can be observed that how noise signal disturbs the
original analog signal. When the signal is sampled at the
Nyquist rate. The samples produce at the signal plus noise
which gives inaccurate sampling. After quantization, the levels
of quantization interrupt by the noise and the quantization error
increases. The encoded signal is also disturbing due to the
noise. The task is to minimize this noise to recover the original
information.

B. LOW PASS FILTER 2

cosine signal with noise

original signal
1 noise signal

1) REMOVE NOISE FROM ENCODING

Amplitude(m)

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0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Time(sec)
sampling with noise
2
original signal
1 noisy signal
sampling signal
Amplitude(m)

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0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Time(sec)
Quantized signal with noise
1

0.5
Amplitude(m)

-0.5 quantized
error
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0 20 40 60 80 100 120
Time(sec)
Encodng with noise
1

0.8

0.6
Amplitude(m)

0.4

0.2

0
0 500 1000 1500 2000 2500 3000 3500 4000 4500
Time(sec)

Fig. 1 Block diagram of Digital Signal

Fig. 3 Signal with noise

 To reduce the noise from encoding. The digital signal When the digital signal passes through the moving average
passes through a low pass filter. A low-pass filter is a filter filter the noise of the signal reduced up to 95%. The filtered
that passes signals with a frequency lower than a certain cutoff signal looks like the digital signal.
frequency and attenuates signals with frequencies higher than 2) REMOVE NOISE FROM QUANTIZATION
the cutoff frequency. By using the low pass filter, it reduces In quantization, the quantization error is minimized
the noise from the signal. and noise signal also decreases. The filtered signal gives
2) REMOVE NOISE FROM QUANTIZATION the approximately original analog signal. The information Original Signal

The noise from quantization is reduced when the signal can be extracted from this signal. 0.8
1
quantized signal
error
0.8
filtered signal

passes through a low pass filter. This filter reduces the

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0.4
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quantization error in the noisy quantized signal. The original 0.2

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information is recovered with minimum noise.

Original Signal Filtered Signal 0
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0.03
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0 0 20 40 60 80 100 120 0 20 40 60 80 100 120
0.02 Time (s) Time (s)

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0
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-0.01 Fig. 11 Quantized Signal with noise and Filtered Signal
-1.5
0 0.1 0.2 0.3 0.4 0.5
Time (s)
0.6 0.7 0.8 0.9 1
-0.02
0 5 10
Time (s)
15 20 25 Fig. Quantized Signal with noise and Filtered Signal

Fig. Quantized Signal with noise and Filtered Signal

3) REMOVE NOISE FROM SAMPLING
3) REMOVE NOISE FROM SAMPLING
When the distorted sampling signal passes through the
The noise is added to the analog signal and the signal is
moving average filter it reduces its noise, but still there are
distorted signal. By taking the samples of this signal it gives
some components of noise in the original signal and we can’t
samples of the signal and the addition of noise. When we want Original Signal
filtered signal

extract the original information.

0.1
1.5

to recover the original signal, it gives information plus noise 1

0.08

0.06

so this signal is passed through a low pass filter which 0.5

0.04

0.02

Amplitude (V)
minimizes its noise so that the original information can be
Amplitude (V)

0 0
Original Signal

extracted.
1 Filtered Signal -0.02
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quantized signal
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error -0.04

0.6 1 -0.06
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Time (s) Time (s)

0
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Fig. Sampling with noise and Filtered Signal

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0 20 40 60 80 100 120 0 20 40 60 80 100 120
Time (s) Time (s)
D. FIR FILTER
The FIR low pass filter removes high frequency signal
Fig. Sampling with noise and Filtered Signal components (noise) from the input signal and its output gives
the low frequency components (original data).
C. MOVING AVERAGE FILTER
1) REMOVE NOISE FROM ENCODING
The moving average filter takes M samples of input at a time When noisy encoded signal passes through the FIR filter,
and takes the average of those M-samples and produces a then noise will be filtered-out from the encoded signal the
single output point. It is a very simple LPF (Low Pass Filter) original information signal can be achieved in its accurate Original Signal

structure to filter unwanted noisy component from the original form.

1
A m p litu d e (V )

data.
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1) REMOVE NOISE FROM ENCODING Original Signal
1
filtered signal
1

0.9
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0 500 1000 1500 2000 2500 3000 3500 4000
0.8
0.8 Time (s)
0.7 Lowpass Filtered Signal
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Amplitude (V)

1
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y

0.4 0.4 0.5

0.3 0.3 0
0.2 0.2
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0.1 0.1 0 500 1000 1500 2000 2500 3000 3500 4000 4500
Time (s)
0 0
0 500 1000 1500 2000 2500 3000 3500 4000 0 500 1000 1500 2000 2500 3000 3500 4000
Time (s) x

Fig. Encoding with noise and Filtered Signal

Fig. Encoding with noise and Filtered Signal
 Original Signal
1
2) REMOVE NOISE FROM QUANTIZATION
quantized signal Besides, these filters are stable with finite input and output.
A mplitude (V )
0.5 error
Due to this reason, FIR filters outperform the low pass and
0
moving filters. Whereas, the low pass and moving average
-0.5
filter remove the small amount of noise components of the
-1
0 50 100 150
Time (s)
200 250 300 350 signal. Moreover, FIR filter is the digitizing and reduce the
1
Lowpass Filtered Signal noise up to a large extent from the signal.
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VI. CONCLUSION & FUTURE WORK
-0.5 This paper presents filter design and noise reduction
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0 50 100 150 200 250 300 350
comparison in medical health application such as, ECG. As
Time (s) filters are the type of device which removes the noise from the
signal and gives the original information as the output. It is
observed that FIR digital filter outperforms the low pass and
moving average filters. Besides, FIR filter gives the linear
phase response (i.e., distortion free), while low pass and
The quantization error is minimized when the distorted
moving average filters are simple and easy to implement but
quantized signal passed through the FIR low pass filter. The
they remove the high frequency components up to small
original data can be extracted from the filtered signal.
amount. The FIR filters give the information about constants
3) REMOVE NOISE FROM SAMPLING
delays, pass band and stop band ripples, so can be considered
The FIR low pass filter removes the noise components
as a potential candidate for ECG. In the near future other
from the distorted sampling signal. It removes the high
filters such as Kalman and adaptive filters will be developed
frequency components from the signal and passes low
Original Signal for noise reduction in medical applications.
frequency2 components which are our original analog signal.
A m plitude (V )

The information
1 can be extracted from this filtered signal. ACKNOWLEDGMENT
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This work is funded by HEC Pakistan under the START-
UP RESEARCH GRANT PROGRAM (SRGP) #21-
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0 0.1 0.2 0.3 0.4 0.5
Time (s)
0.6 0.7 0.8 0.9 1 1465/SRGP/R&D/HEC/2016, and Sukkur IBA University,
Lowpass Filtered Signal Sukkur, Sindh, Pakistan.
1

0.5

0
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-0.5 [1] Kai Siedenburg & Monika Dorfler,”Audio Denoising by generalized
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