Comparison Between Different Methods of Noise Cancellation

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Abstract
This paper presents performance
comparison between different methods
of noise cancellation. Reduction of noise
from speech signals plays a vital role in
modern communication systems.
Adaptive filtering is a powerful
technique for signal detection because of
the random pattern of the noise and the
non-deterministic sources of the
interference. It is preferable to design a
filter that is adapted to the background
noise to remove interference from
observations. Wavelet Transform
Adaptive Filtering is modern
mathematical approach for achieving
and analyzing spectral components of
signals. In this paper different
algorithms (e.g. LMS, RLS,
Thresholding etc.) are derived for noise
cancellation and comparison made
between them in Fourier as well as
Wavelet Transform Domain.
Keywords: Wavelet transform,
Adaptive Filtering, Least Mean Square
Algorithm

1. Introduction
Wavelet Transform
In the past, Fourier Transforms
were used to project a signal onto the
subspace of orthogonal vectors, but
Fourier Transforms have a undesired
property when analyzing certain signals.
Due to the uncertainty principle, Fourier

transform cannot accurately present the
signal in both time and frequency domain,
particularly in the case of real time system
realization. The wavelet transform have an
advantage in this respect. Wavelet
transform have some desirable and useful
properties for analyzing real time signals,
and they present a flexible tool for multi-
resolution analysis of continuous time
signals. Their implementation is fairly
easy and straight forward. In wavelet
transform, the signal from time domain is
changed to a weighted sum of translates
and dilates of a mother wavelet. Wavelet
transform can be classified as Continuous
Wavelet Transform (CWT) and Discrete
Wavelet Transform (DWT). Due to the
digital nature of the experiments, we are
primarily concerned with DWT. A
continuous time signal can be presented in
wavelet domain as:

x( t) = J
]k
] k
( t)
,kz

= J
]k
]k
( t) + c
0k
0k
( t)
kz
kz
]=0

where, Z is set of integers,
]k
=
2
]
2
( 2
]
t k) is an orthonormal basis
derived from the scaling function ( t) (for
the subspace ( I
]
I
]+1
) and the wavelet
function
]k
( t) = 2
]
2
( 2
]
t k)
constitutes an orthonormal basis (for the
subspace w
]
I
]+1
I
]
) derived from
( t) . Also, d
jk
and c
ok
are wavelet
Comparison between Different Methods of Noise Cancellation
Archana Sharma
1
1
Assistant Professor, Sunderdeep College of Engineering & Technology, Ghaziabad 201001 UP,
Dr. Ranjit Singh
2

2
Professor, Department of Electronics & Communication Engineering, Ajay Kumar Garg Engineering College,
Ghaziabad 201009 UP and
Vikas Sharma
3

3
Senior Software Engineer, Aircent, Electronic City, Plot 31, Sector 18, Gurgaon 122015, Haryana
JOURNAL OF COMPUTING, VOLUME 4, ISSUE 5, MAY 2012, ISSN 2151-9617
https://sites.google.com/site/journalofcomputing
WWW.JOURNALOFCOMPUTING.ORG 23

+
y(k)
e(k)
d(k)
-
u(k)
coefficients used to represent the original
time domain input vector X(n) in the
wavelet domain.
The values of k and j determine
the time domain shift and scale of the
mother wavelet, respectively. It is through
these two parameters that the different
resolution levels, and therefore different
subspaces are created. In practice, the
infinite sums are truncated and replaced by

x( t) = J
]k
]k
( t)
k
]
]=0

which presents a description of a signal in
a terms of the bandpass filters whose
bandwidth and center frequency is
increased by the factor of 2
j
.
In the discrete form, the signal x(t)
is sampled to obtain x(k), and the wavelet
transform projects this signal onto the
wavelet subspace W
j
given as-

x
]
( k) = J
]k
]k
( n) + c
0k
ok
( n)
kz
kz

The wavelet coefficients
approximation in discrete domain can be
expressed as-
J
]k
= x( n)
]k
( n)
n

c
ok
= x( n)
n

ok
( n)
where d
jk
and c
0k
are the discrete wavelet
coefficients. These expressions for d
jk
and
c
0k
present the discrete wavelet transforms.
The choice of the type of wavelet
determines
]k
( n) and
0k
( n) .

LMS Algorithm

The least-mean-square (LMS)
algorithm is a widely used algorithm. It is
named by its originators, Widrow and Hoff
(1960). A significant feature of the LMS
algorithm is its simplicity. Moreover, it
does not require measurement of the
pertinent correlation functions, nor does it
require matrix inversion. Indeed, it is the
simplicity of the LMS algorithm that has
made it the standard against which other
adaptive filtering algorithms are
benchmarked.
The LMS algorithm is a linear
adaptive filtering algorithm that consists of
two basic processes: A filtering process,
which involves computing the output of a
transversal filter produced by a set of tap
inputs, and generating an estimation error
by comparing this output to a desired
response. An adaptive process, which
involves the automatic adjustment of the
tap weights of the filter in accordance with
the estimation error. Thus, the combination
of these two processes working together
constitutes a feedback loop around the
LMS algorithm, as illustrated in the block
diagram :

Least mean squares (LMS)
algorithm is a class of adaptive filter used
to mimic a desired filter by finding the
filter coefficients that relate to producing
the least mean squares of the error signal.
Hence our goal is to minimize the mean
square error between the adaptive filter
output and desired signal.

E
2
| c( k) | = E
2
| J( k) y( k) |

Adaptive Weight-
Control Algorithm
Transversal Filter


w
H
(n-1)u(n)
+
u(n)
d(n)
(n)
-
It is a stochastic gradient descent
method in that the filter is only adapted
based on the error at the current time.

RLS Algorithm

In this algorithm we use the
method of least squares to develop a
recursive algorithm for the design of
adaptive transversal filters such that, given
the least-squares estimate of the tap-weight
vector of the filter at iteration n-1, we may
compute the updated estimate of this
vector at iteration n upon the arrival of
new data. We refer to the resulting
algorithm as the recursive-least-squares
(RLS) algorithm.
An important feature of the RLS
algorithm is that it utilizes information
contained in the input data, extending back
to the instant of time when the algorithm is
initiated.
The desired recursive equation for
updating the tap-weight vector:
w( n) = w( n 1 ) + k( n) [ J
( n)
u
H
( n) w( n 1 ) ]
= w( n 1 ) + k( n)
( n)
Where ( n) is the a priori estimation error
defined by

( n) = J( n) u
1
( n) w
( n 1 )

= J( n) w
H
( n 1 ) u( n)
the inner product w
H
( n 1 ) u( n)
represents an estimate of the desired
response J( n) , based on the old least-
squares estimate of the tap-weight vector
that was made at time n 1.
and, k( n) is gain vector, defined by

k( n) =
-1
( n) u( n)

where ( n) is correlation matrix.
The a priori estimation error ( n)
is, in general, different from the a
posteriori estimation error

c( n) = J( n) w
H
( n) u( n)

Representation of RLS algorithm in block
diagram is shown

As we discussed previously that it
utilizes information contained in the input
data, when the algorithm is initiated, the
resulting rate of convergence is therefore
typically an order of magnitude faster than
the simple LMS algorithm. This
improvement in performance, however, is
achieved at the expense of a large increase
in computational complexity.

Threshold De-Noising

De-noising using linear filters is
not efficient for functions with
discontinuities. This is due to the linear
nature of this process, which prevents to
efficiently estimate discontinuities.
Wavelet approximation using thresholding
allows an adaptive representation of signal
discontinuities. We will thus use wavelet
thresholding to perform a non-linear de-
noising of 1D signals.
Thresholding is done by the
following definition:
J
]k
= _
J
]k
, |J
]k
| z
0 , |J
]k
| < z

where z 0 is threshold value/parameter.

Transversal Filter
w(n-1)

Adaptive Weight-
Control Algorithm

It is of two types: Soft
Thresholding and Hard Thresholding. The
hard thresholding method keeps some
coefficients fixed and sets others to 0; in
contrast the soft thresholding method
either shrinks or sets them to 0.
The noisy signal is basically of the
following form:
s( n) = ( n) + oc( n)
where time n is equally spaced.
In the simplest model, suppose that
e(n) is a Gaussian white noise N(0,1) and
the noise level o is supposed to be equal to
1. The de-noising objective is to suppress
the noise part of the signal s and to recover
f.
The de-noising procedure proceeds
in three steps:
1. Decomposition. Choose a wavelet,
and choose a level N. Compute the
wavelet decomposition of the
signal s at level N.
2. Detail coefficients thresholding.
For each level from 1 to N, select a
threshold and apply soft
thresholding to the detail
coefficients.
3. Reconstruction. Compute wavelet
reconstruction based on the
original approximation coefficients
of level N and the modified detail
coefficients of levels from 1 to N.
2. Experimental Setup /
Methodology
We have implemented adaptive filters
in MATLAB version R2010a. To generate
wavelet transform, db1 wavelet family
is used. cA and cD are approximate and
detailed coefficients of wavelet
respectively. For adaptive filtering,
predefined function adaptfilt in
MATLAB is used. It creates adaptive
object based on algorithm used (e.g.
adaptfilt.lms/adaptfilt.rls).
We create individual script for each
algorithm for wavelet transform domain
and also for Fourier transform domain.
The main difference between wavelet
transform domain and Fourier transform
domain is that wavelets are localized in
both time and frequency domain whereas
the standard Fourier transform is only
localized in frequency domain.
To observe the performance of
different methods of noise cancellation
over signal and voice, sinusoidal signal
and sound file (in wav format) are
processed through each Matlab script one
by one.
After processing of signals, filtering
performance and quality of output of
filters are observed and analyzed.
3. Results and Conclusion
The behavior of LMS adaptive filter,
RLS adaptive filter and Threshold filter
are compared with each other. The
performance of filters in wavelet transform
domain is also compared with performance
in Fourier transform domain. Resulting
outputs are as shown:

Fig 1: Original signal desired to obtained

Fig 2: Random Noise added to the original signal

Fig 3: Distorted (Noisy) Signal

Fig 4: Output of RLS adaptive filter in DWT domain

Fig 5: Output of LMS adaptive filter in DWT domain

Fig 6: Output of Threshold filter in Wavelet domain

As shown in above screen shots,
among the outputs of three algorithm:
LMS, RLS and Threshold, most reliable
algorithm is thresholding. But it is only for
the case of low noise addition. As we
derived, and also by name, threshold
algorithm is basically an approximation.
When the added noise is large in
magnitude, this algorithm is not much
efficient as in the case of low noise
addition. In case of high noise level, output
of filter deviates from desired signal.
Among the adaptive filters: LMS
and RLS, as shown in output that
performance of RLS algorithm is
improved than LMS algorithm, however, it
costs in the form of increased
computational complexity.
These algorithms are analyzed in
Fourier transform also. When sound files
are processed in fourier transform domain,
the observed performance is not much
better as in the case of wavelet transform
domain.
4. References
1. A. Abbate, C. M. DeCusatis and P. K.
Das, Wavelets and Subbands:
Fundamentals and Applications, 1st
edition, Birkuser, Boston, MA, 2002.
2. N. Erdol and F. Basbug, Wavelet
transform based adaptive filters:
Analysis and new results, IEEE
Trans. Signal Processing, vol. 44,
September 1996.
3. Simon Haykin, Adaptive Filter
Theory, 3
rd
Edition, Prentice Hall,
2003.
4. G. Evangelista, C. W. Barnes,
Discrete-Time Wavelet Transforms
and Their Generalizations, Proc.
ISCAS-90, New Orleans, 1990.
5. Raghuveer and Bopardikar, Wavelet
Transforms: Introduction to Theory
and Applications, Addison - Wesley,
1998.
6. Donoho, D.L.; I.M. Johnstone, "Ideal De-
noising in An Orthonormal Basis Chosen
from a Library of Bases," C.R.A.S. Paris,
t. 319, Ser. I, 1994.
7. Gerald Kaiser, A Friendly Guide to
Wavelets, Birkhauser,1994.
8. Donoho, D.L. (1995), "De-noising by
Soft-Thresholding," IEEE Trans. on Inf.
Theory, 41, 3.
9. Srinath Hosur and Ahmed H. Tewfik,
Fellow, IEEE, Wavelet Transform
Domain Adaptive FIR Filtering, vol. 45,
No. 3, March 1997.
10. Milos Doroslovacki and Hong Fan,
Wavelet-Based Adaptive Filtering,
Department of ECE, University of
Cincinnati, Cincinnati, OH 45221-0030,
1993.


Archna Sharma is currently pursuing MTech in Electronics & Communications from Ajay
Kumar Garg Engineering College, Ghaziabad. Graduated in Electronics & Instrumentation
from I.E.T, Lucknow with 78% (Honors) in 2003.
Since Aug 2010, working as Assistant Professor in Sunderdeep College of Engineering &
Technology Ghaziabad. Earlier, she worked as senior lecture in IMS Engineering College
Ghaziabad.
She has abiding passion for teaching. During her teaching career spanning nine years, she
taught a number of courses to all the BTech-year students. Takes active part in
administrative and professional activities including establishment of Digital Integrated
Circuit Lab, Network Lab, Electronics Lab I, Electronics Lab II etc.

Dr Ranjit Singh obtained B.Tech. M.Tech. and Ph.D. degrees from the Indian Institute of
Technology, Kanpur in 1969, 1971 and 1976 respectively. He specialized in the area of
Electronic circuits and devices.
Published large number of technical papers in IETE journals in addition to in-depth
technology-reviews covering emerging trends in Communications and information
technology.

Since last three and half years, he is teaching at Ajay Kumar Garg Engineering College
where, he is a Professor in the Department of Electronics & Communications Engineering.
He has abiding passion for teaching and research. Currently guiding MTech. and PhD
scholars besides supervising BTech projects. He is Life Fellow of the IETE and attended
international conferences held in France, Singapore, USA, Hong Kong and Nepal. Daily
practices advanced meditation.

Vikas Sharma obtained MTech in Computer Science with distinction from Birla Institute of
Technology, Mesra Ranchi in May 2008. Earlier completed M.C.A. from S.I.S.T, UPTU,
Lucknow in May 2004.

Currently working as Senior Software Engineer for Wireless & Convergence SBU of
Product Engineering & Services BU of ARICENT Group, Gurgaon with present focus on
Protocol mediation platform for 3G and LTE based networks.

Possesses domain knowledge in the Telecom areas of SNMP, GTPv.1 GPRS (2.5G) and
GTPv.2 LTE (4G). Good Programming and analytical skills with good exposure to
application development. Capable of working in large projects with fast-paced deadlines and
deliverables.


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