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Performance of Wiener Filter and Adaptive Filter for Noise Cancellation in

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Article in International Journal of Computer Applications · July 2014

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 International Journal of Computer Applications (0975 – 8887)
Volume 97– No.15, July 2014

Performance of Wiener Filter and Adaptive Filter for

Noise Cancellation in Real-Time Environment

G.V.P.Chandra Sekhar B. Ananda Krishna, Ph.D M. Kamaraju, Ph.D

Yadav Professor, Department of ECE Professor, Head of Department
Student, M.Tech, DECS Gudlavalleru Engineering ECE
Gudlavalleru Engineering College Gudlavalleru Engineering
College Gudlavalleru-521356, Krishna College
Gudlavalleru-521356, Krishna District Gudlavalleru-521356, Krishna
District Andhra Pradesh, India District
Andhra Pradesh, India Andhra Pradesh, India

ABSTRACT adapting nature according to the input signal. Noise is the

Form the past several decades’ noise cancellation in speech signal that mixes up with the desired signal and corrupts it in
signal gains researchers’ attention. Several techniques were any type of communication system. In signal processing
developed for noise cancellation among them optimal wiener adaptive filters are the alternate method for recovering desired
filter can be the one of the most fundamental approach for speech from the noise. Several algorithms have been proposed
noise cancellation. Later on adaptive filter was introduced to in earlier days to detect the desired signal. Least mean square
attain better performance. This paper shows the capacity of (LMS) algorithm was the most efficient method in terms of
wiener filter and adaptive filter for removal of noise by computation and storage requirements but it has low
estimating the signal by means of removing the noise signal convergence speed. The paper consists of objective, existing
form the corrupted signal.Wiener filter plays a central role in work, proposed work, parameters to be compared,
wide range of applications such as linear prediction, echo filtering,need for adaptive filtering, adaptive algorithms, real
cancellation, signal restoration, channel equalization and time environment, simulation results, conclusion and future
system identification. In this paper the performance of wiener scope and references.
filter and adaptive filter for removal of noise in the presence
of real time environment are compared. In the existing papers
2. OBJECTIVE
the authors have proposed the theory of wiener filter and The main objective of the project is to remove the real time
adaptive filter algorithms in real time environment like noise that is present within the signal and compare the
recorded speech. So this is paper is going to take the part of performance of wiener filter and adaptive filter for noise
the existing paper and going to perform the noise cancellation. cancellation in real time environment.
In order to measure the performance step size is the main
factor for the convergence speed and mean square error.
3. EXISTING WORK
The authors Yen-Hsiang Chen et al [1] proposed, “An
Wiener filter provides better performance for noise
Automotive Application of Real-time Adaptive Wiener Filter
cancellation but it requires large no. of computations i.e.,
for Non-Stationary Noise Cancellation in a Car Environment”.
complexity and cost of the system is going to increase, so
adaptive filter is the alternate approach for removal of noise In this paper, a real-time adaptive wiener filter with two micro
with moderate complexity and cost. The simulation result phones is implemented to reduce noisy speech when noise
clearly shows that wiener filter gives the better performance signals and desired speech are incoming simultaneously. This
but due to high cost adaptive filter is the choice of many paper also gives an analysis of different matrix sizes of the
applications. This paper is going to discuss about wiener filter wiener filter so as to enable the possibility of real-time
theory, wiener filter problem, solution to optimal filtering, implementation. The performance of the proposed design is
adaptive filtering, adaptive algorithm, study of wiener filter measured by as much as 20 dB noise reduction, and the
and adaptive filter for noise reduction etc. proposed adaptive wiener matrix update speed achieves a 28.6
ms/frame, with a matrix size of 200.
Keywords
Wiener filter, optimum linear filters, Adaptive filter,Noise The authors H.kaur et al [2] proposed, “Performance and
cancellation,Performance comparison, Convergence speed, Convergence Analysis of LMS Algorithm”.
Minimum mean squared error, Least mean square In this paper, Instead of computing as suggested by
1. INTRODUCTION Wiener-Hopf equation, in LMS the coefficients are adjusted
from sample to sample in such a way as to minimize the MSE.
Wiener filters are the class of optimal linear filters which
The LMS algorithm is based on steepest descent algorithm.
involves linear estimation of the desired signal by comparing
This paper also gives the analysis of LMS algorithm for
the noise signal with the desired signal corrupted by
different input samples and different number of iterations. The
noise.Wiener proposed an optimal solution to the problem of
simplicity of the LMS algorithm and ease of implementation
noise cancellation but it has some practical limitations.
make it first choice in many applications. The convergence
Adaptive filters are the digital filters with an impulse
rate is low for this algorithm.
response, or transfer function that can be modified time to
time to match with the characteristics of the desired system. The authors Talwar.R et al [3] proposed “Performance
Adaptive filters are used in many applications because of its

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 International Journal of Computer Applications (0975 – 8887)
Volume 97– No.15, July 2014

comparison of adaptive algorithms for noise cancellation”. resist both desire signal and noise signal. The filter length of
the adaptive system is inherently tied to many of the other
In this paper the authors have compared the performance of performance measures. The length of the filter specifies how
different adaptive algorithms like LMS, NLMS, and RLS accurately a given system can be modeled by the adaptive
algorithms for sinusoidal input. Performance is compared for filter. In addition, the filter length affects the convergence
different step size and different number of iterations. Three rate, by increasing or decreasing computation time, it can
adaptive filter algorithms have been compared by simulation affect the stability of the system, at certain step sizes, and it
to achieve high convergence rate and minimum mean square affects the minimum MSE.
error with noise and different values of µ. Every algorithm
works on different methods for noise cancellation and 6. FILTERS
improves system performance. Filters are the basic elements in the signal processing system.
The authors G.V.P.Chandra Sekhar Yadav et al [4] proposed Filter is a device used to suppress the unwanted signal i.e.
“Study of different adaptive filter algorithms for noise noise from the desired signal as shown in figure 2and there
cancellation in real time environment”. are several techniques are used for filtering. Usual method of
estimating the signal corrupted by noise is to pass to it
In this paper the authors have proposed the performance of through the filter that tend to suppress the noise and leaving
wiener filter and different adaptive filter algorithms like LMS, the desired signal this is so called direct filtering. In general
NLMS and RLS algorithms for noise cancellation in real time filters are of two types. Fixed filters and adaptive filters
environment like recorded speech as the input and different
noise signals are added to it and then desired signal is In fixed filters the frequency response or filter coefficients of
estimated by using the adaptive algorithms. the filters are fixed and it requires prior knowledge of the
input signals. If the signal and noise characteristics are known
4. PROPOSED WORK beforehand then it is easy to design the filter that passes the
In the existing work, the authors G.V.P.Chandra Sekhar frequencies contained in the signal and rejects the frequency
Yadav and Dr. B. Ananda Krishna [4] proposed the band occupied by noise.
performance of wiener filter and adaptive filter algorithms An adaptive filter on the other hand requires less or no
like LMS, NLMS and RLS algorithms in real time knowledge about the signal and noise. In adaptive filters the
environment. Now, this paper is going to work on the part of filter coefficients are adjustable or modifiable.
the existing work like wiener filter and adaptive filter
algorithm i.e. LMS algorithm in real time environment by
taking input as recorded speech with different background
noises to compare their performance and the performance is
analyzed with different lengths and parameters of the input as
shown in figure 1.

Fig 2: Basic filter

6.1.1 Basic wiener filter theory
Wiener filter is a filtering system that is an optimal solution to
the statistical filtering problem [1].In the statistical approach
to the solution of the linear filtering problem we have to
assume the characteristics of the desired signal to be estimated
and noise to be removed. So the filter has to be designed in
such a way that the output of filter minimizes the effect of the
noise from the noisy data which is given as input to the filter
[11].
The meaningful approach to the problem of linear filtering is
Fig 1: Block diagram of proposed work by minimizing the mean square value of error signal which is
described as the difference between the desired signal and
5. PARAMETERS filter output. The mean square error (MSE), ξ, is obtained by
Performance of the systems can be measured with respect to taking the “expectation” of the squareof the error signal (ek).
several parameters like Minimum mean square error,
convergence speed, computational complexity, stability, For stationary inputs the solution to this problem is commonly
robustness, signal to noise ratio and filter length. Minimum a wiener filter. Many adaptive algorithms can be viewed as
mean square error denotes how the system will adapt to a approximations of the discrete wiener filter. The signals
given solution. Small MSE indicates that the system is and which are noise and noisy signals respectively are
accurately modified or adapted to a solution for the system. applied to the filter as shown in figure 3. The noisy signal
Large MSE indicates the system can’t be adapted according to consists of the desired data and noise, then wiener filter gives
the requirement. Step size is the major factor for convergence the output as the estimate of the noise signal which is then
speed. Convergence rate indicates the rate at which the filter subtracted from the noisy data to yield error signal.
converges to its resultant state.Usually faster convergence rate
is the desired characteristic of the adaptive filter.
Computational complexity is important particularly in real
time adaptive applications. Robustness is how the system can

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 International Journal of Computer Applications (0975 – 8887)
Volume 97– No.15, July 2014

Fig 3: Wiener filter

Fig 4: Mean square error surface

Where xk and w are the input signal vector and weight vector Wiener filter is formulated to map an input signal to the
respectively of the filter and are given by output that is as close to the desired signal as possible.
A signal is completely recoverable from noise when the
spectra of the signal and noise don’t overlap each other. If the
Wiener filter is represented by coefficient vector . Wiener signal and noise occupies different parts of the frequency
filter coefficients are calculated to minimize the mean square spectrum, they can be separated by using either low pass or
value between the filter output and desired signal. Error signal high pass filter. If the signal and noise has overlap spectra, in
is obtained by taking the difference between the noisy signal this case it is not possible to completely separate the signal
and estimation of the noise signal, after that mean square error from noise but the effects of noise can be reduced by using
(MSE) is calculated by taking expectation to square of the wiener filter.
error signal ek2.
The choice of wiener filter order affects:
The ability of the filter to remove the distortion and reduce the
Performance surface is used to determine the solution to the noise.
statistical filtering problem i.e. it finds the optimum value of The computational complexity of the filter and
the filter coefficients where the mean square error should be The numerical stability of the wiener solution.
minimized [5].
And also the wiener filter has a limited practical usefulness
6.1.2 Performance surface because of the following reasons.
The mean square error of the wiener filter is a quadratic It requires the auto correlation matrix R and the cross
function of filter coefficient and has single minimum value as correlation vector P both of which are not known.
mentioned above. A plot of mean square erroragainst the filter It involves matrix inversion, which is time consuming.
coefficient is bowl shaped surface with a unique bottom If the signals are non-stationary then both R and P will change
point as shown in figure 4. It is known as the mean square with time and so will have to be computed repeatedly.
error surface and is Non-negative. The gradient of the mean
square error surface is given by For real time applications, a way of obtaining on a
sample by sample basis is required. Adaptive algorithms are
used to achieve this without having to compute R and P
Where called as toeplitz matrix andis the N-length explicitly or performing a matrix inversion. The solution to
cross correlation vector which is symmetric Wiener-Hopf equation is the steepest descent algorithm [6].
is the auto correlation matrix.
6.1.3 Adaptive filters
The least mean square error point corresponds to the Adaptive filter is essentially a digital filer with self-adjusting
minimum error power. At this operating point the mean characteristics. It adapts automatically to changes in its input
square error surface has zero gradient value and filter weight signals [6]. Adaptive filters have the capability of adaptively
vector has its optimum value . tracking the signal under non stationary conditions.
In the basic form, the wiener theory assumes that signals are
stationary. However if the coefficients are periodically
regulated for every N samples then the filter adapts itself to
the characteristics of the signal become an adaptive filter.
This is known as the wiener-Hopf equation or solution [1]. In Adaptive filter provides alternate solution to the formulation
adaptive filter it adapts or adjusts the filter of wiener filter problem [8].
weights according to suitable
algorithm, to find the optimum value of the weight vector on Adaptive filters are used in the following cases:-
the mean square error surface. When it is necessary for the filter characteristics to be
variable, adapted to changing conditions.
When there is spectral overlap between signal and noise.
If the band occupied by the noise is unknown or varies with
time.

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 International Journal of Computer Applications (0975 – 8887)
Volume 97– No.15, July 2014

The use of conventional filters in the above cases would lead 7. ADAPTIVE ALGORITHMS
to unacceptable distortion of the desired signal. Adaptive algorithms are used to adjust the coefficients of the
digital filter. Such that the error signals is minimized
according to some criterion.
The most commonly used adaptive algorithms are least mean
square (LMS), normalized least mean square (NLMS) because
of their robustness and simplicity and also have recursive least
square (RLS) algorithm for noise cancellation.

7.1.1 Least mean square (LMS) algorithm

One of the most successful adaptive algorithms is the LMS
algorithm. Instead of computing in one go as suggested
by wiener–Hopf equation, in the LMS the coefficients are
adjusted from sample to sample in such or way as to minimize
the MSE [2].
Fig 5: Adaptive filter
This amounts to descending along the surface of the figure 4
Fig 5 shows the basic adaptive filter with input signal and towards its bottom.
desired signal as inputs and one output signal with adaptive
An adaptive filter is required to find a solution for its tap-
algorithm to adapt changes in the input signal.
weight vector that satisfies the normal equation. An
6.1.4 Adaptive filter as noise canceller alternative procedure is to use the method of steepest descent,
An adaptive filter consists of two distinct parts: a digital filter which is one of the oldest methods of optimization.
with adjustable coefficients, and an adaptive algorithm which The standardLMS algorithm performs the following
is used to adjust or modify the coefficients of the filter. operations to update the coefficients of an adaptive filter:
Adaptive noise canceller has two inputs namely primary input Calculates the output signal from the adaptive filter.
and reference input . The primary input receives the
signal form the signal source that is corrupted by noise Calculates the error signalby using the following equation
which is uncorrelated with the signal and the reference input
receives noise from the noise source which is uncorrelated
with the signal in primary input while correlate with noise in
the primary input.
Updates the filter coefficients by using the following
The noise passes through the filter to produce an output equation:
which is close to the estimate of the primary input noise .
The noise estimate is then subtracted from the corrupted Where μ is the step size of the adaptive filter, is the filter
signal to produce an estimate of the signal as the output of coefficients vector, and is the filter input vector.
adaptive noise canceller.
The LMS algorithm is based on the steepest descent algorithm
where the weight vector is updated from sample to sample is
shown in figure 7 is as follows

Where and are the weight and the true gradient vectors
respectively at the sampling instant. µ controls the size of
incremental correction applied to the adaptive filter from one
iteration to the next.
The steepest descent algorithm in the above equation still
requires knowledge of R and P. since is obtained by
evaluating the equation.

Fig 6: Block diagram of adaptive filter as noise canceller

The LMS algorithm is a practical method of obtaining
The main objective in noise cancelling is to produce an estimates of the filter weights in real time without matrix
optimum estimate of the noise in the contaminated signals and inversion in the equation or the direct
hence an optimum estimates of the desired signal [5]. computation of the auto correlation and cross correlation.
This is achieved by using error or estimated signal in a
feedback arrangement to adjust the digital filer coefficients, In the LMS algorithm, instantaneous estimates are used for .
via a suitable adaptive algorithm, to minimize the noise in Thus
as shown in fig 6.
The output signal serves two purposes:
1) As an estimate of the desired signal and
2) As an error signal which is used to adjust the filter
coefficients.

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 International Journal of Computer Applications (0975 – 8887)
Volume 97– No.15, July 2014

Where 8. REAL TIME ENVIRONMENT

Replace the value of in steepest descent algorithm then The proposed algorithm is going to work in real time
environment like recorded speech used as input signal and this
input signal is going to mixes with the unwanted noise which
is also in recorded form and the wiener filter and adaptive
algorithms are designed in such a way that these unwanted
noise is removed at the end of the final iteration.
Mat lab is the software tool that is going to work on different
signal processing tools.
The instruction used in mat lab to record the real time speech
signal is given as follows
Wavrecord it Record sound using PC-based audio input
device.
Syntax:
Wavrecord (n, fs, 'dtype') uses the data type specified by the
string 'dtype' to record the sound.
Fig 7: Adaptive filter using LMS algorithm The string 'dtype' can be one of the following:
Clearly, the LMS algorithm above doesn’t require prior 'double' (default value), 16 bits/sample
knowledge of the signal statistics (that is the correlations R
and P), but instead uses their instantaneous estimates as 'Single',16 bits/sample
shown above. 'int16', 16 bits/sample
The weights obtained by the LMS algorithm are only 'uint8', 8 bits/sample
estimates, but these estimates improve gradually with time as
weights are adjusted and the filter learns the characteristics of 0.6
Input signal

the signals. Eventually, the weights converge [9].

0.4
The condition of convergence is
0.2
Where is the maximum Eigen value of the input data
covariance matrix.
0
Amplitude

In general Wk never reaches the theoretical optimum (the

wiener solution), but fluctuates about it as shown in figure 8. -0.2

-0.4

-0.6

-0.8
0 100 200 300 400 500 600 700 800 900 1000
No.Of.Iterations

Fig 9: Recorded speech signal

Noise signal
0.15

0.1

0.05

0
Amplitude

-0.05

-0.1

Fig 8: An illustration of the variations in the filter weights

-0.15
The simplicity of the LMS algorithm and ease of
implementation, make it the algorithm of first choice in many -0.2
0 100 200 300 400 500 600 700 800 900 1000
real-time systems [10]. The LMS algorithm requires No.Of.Iterations
approximately 2N+1 multiplications and 2N+1 addition for Fig 10: Recorded noise signal
each new set of input and output samples [7].
The real time signal that is recorded by using micro phone is
In general any signal processing operations involves large no shown in figure 9. This signal is going to mix with the
of multiply and accumulate arithmetic operations, because of recorded unwanted noise signal shown in figure 10 and the
direct implementation LMS algorithm is more attractive.

20
 International Journal of Computer Applications (0975 – 8887)
Volume 97– No.15, July 2014

resultant of the mixing is shown in figure 11. And the mixed Output Of LMS algorithm for Sinusoidal Input
2
noise is eliminated as possible asby using different algorithms
at the end of the final iteration. 1.5

Noisy Signal
1
0.6

0.5
0.4

Amplitude
0
0.2
-0.5
0
Amplitude

-1

-0.2
-1.5

-0.4 -2
0 100 200 300 400 500 600 700 800 900 1000
No.Of.Iterations
-0.6
Fig 13: LMS algorithm output for sinusoidal input
-0.8
0 100 200 300 400 500 600 700 800 900 1000 MSE Of Wiener filter for Sinusoidal Input
-2
No.Of.Iterations 10

Fig 11: Noisy signal -4

10

9. SIMULATION RESULTS
The result analysis from the outputs of wiener filter and LMS
Squared Error Value
-6
10
algorithm shows that Wiener algorithm provides better
performance but it has high degree of Complexity.
-8
10
From the figure 12 it is clear that wiener filter gives good
performance but it has high computational complexity.
Wiener filter has to compute R and P explicitly but LMS -10
10
algorithm is suited for real time applications where sample by
sample bias is done. In wiener filter the computational
-12
complexity is also high because of auto correlation and cross 10
0 100 200 300 400 500 600 700 800 900 1000
correlation so adaptive filters are designed for moderate No.Of.Iterations

computations. Fig 14: MSE for wiener filter for sinusoidal input

1.5
Output Of Wiener filter for Sinusiodal Input Figure 14 and 15 shows the Mean Square Error for sinusoidal
input for wiener filter and LMS algorithm respectively.
Figures show that LMS algorithm has Minimum Mean Square
1
Error which yields better results. MSE is decreasing function
for LMS algorithm where as in wiener filter MSE is a random
0.5 function it decreases first and then increases hence LMS
algorithm is suitable for better performance and low
Amplitude

0 complexity.
MSE Of LMS algorithm for Sinusoidal Input
2
-0.5 10

0
10
-1

-2
10
-1.5
Squared Error Value

0 100 200 300 400 500 600 700 800 900 1000
-4
No.Of.Iterations 10

Fig 12: wiener filter output for sinusoidal input -6

10
Figure 12 and 13 shows the outputs of wiener filter and LMS
algorithm for sinusoidal input. From the figures it is clear that -8
10
wiener filter will provide better results than LMS algorithm
-10
but it requires more complexity. 10

Whereas LMS algorithm gives better performance for more -12

10
0 100 200 300 400 500 600 700 800 900 1000
no of iterations.The output of LMS algorithm is effective No.Of.Iterations
when there are large numbers of iterations and also it has low
complexity. Fig 15: MSE for LMS algorithm for sinusoidal input

21
 International Journal of Computer Applications (0975 – 8887)
Volume 97– No.15, July 2014

Output Of Wiener filter for realtime Input up to some iteration and after some point MSE is going to
0.6
decrease. This is the good scenario but it is going to decrease
up to some other iteration and will suddenly increase at final
0.4
iteration.
0.2 MSE of LMS algorithm for realtime input
-4
10

0
Amplitude

-6
10
-0.2

Squared Error Value

-0.4 -8
10

-0.6
-10
10
-0.8
0 100 200 300 400 500 600 700 800 900 1000
No.Of.Iterations
-12
10
Fig 16: wiener filter output for real time input
Output Of LMS algorithm for realtime Input -14
0.6 10
0 100 200 300 400 500 600 700 800 900 1000
No.Of.Iterations
0.4
Fig 19: MSE for LMS algorithm in real time environment
0.2 Fig 19 shows that the mean square error for LMS algorithm.
Figure shows that MSE is an increasing function for this
0 reason the output of LMS algorithm is somewhat distorted.
Amplitude

On comparing it with the previous figure both are quite

-0.2
opposite to each other. Though it provides some distorted
results than wiener filter it is preferred because of low
-0.4
complexity and convergence rate.
-0.6 Considering all these parameters wiener filter and LMS
algorithms are compared with respect to speed, convergence
-0.8
0 100 200 300 400 500 600 700 800 900 1000 rate, complexity and mean square error as shown in table 1.
No.Of.Iterations
Table 1: Comparison of wiener and LMS algorithm
Fig 17: LMS algorithm output for real time input
Algorithm Speed Converge Complexity MSE
Figure 16 and 17 shows the outputs of wiener filter and LMS
nce rate
algorithm for real time recorded speech. Wiener filter gives
Wiener Low - High Low
approximately desired signal as output but it has to compute
filter
autocorrelation, cross correlation and matrix inversion to yield
LMS High Low Low High
results whereasLMS algorithm provides alternate solution to
optimal filter coefficients problem and removes noise better
than wiener filter. 10. CONCLUSION AND FUTURE
MSE is used to analyze the performance by minimizing it. SCOPE
Mean square error should be as minimum as possible for Different algorithms have been introduced by several authors
obtaining good performance of any system. to remove noise within the signal and compared their
MSE of Wiener filter for realtime Input
performance with respect to different parameters. Out of
-1
10 which wiener filter is the ancient method for noise
cancellation but it requires high complexity. So adaptive
-2
10 filters are introduced to reduce the complexity and
-3
computational speed. As the authors tried to reduce the mean
10 square error the complexity and stability of the systems will
Squared Error Value

-4
increases.From the simulation results it is clear that LMS
10
algorithm is best suited because of its low cost and complexity
-5
than wiener filter.LMS algorithm is the best algorithm but it
10
has low convergence speed. This project compares only LMS
-6 algorithm and wiener filters for real time input. As a future
10
work the same work is going to extend in real time
10
-7 environment with different background noise and compare
their performance with other adaptive algorithms i.e., NLMS
10
-8 and RLS to achieve high convergence rate. The performance
0 100 200 300 400 500 600 700 800 900 1000
No.Of.Iterations
is analyzed with different parameters and lengths of input
signal.
Fig 18: MSE for wiener filter in real time environment
Figure 18 shows the mean square error for wiener filter. From 11. REFERENCES
the figure we can analyze that the mean square error increases [1] Yen-Hsiang chen, Shanq-Jang Ruan, Tom Qi, ”An

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Volume 97– No.15, July 2014

Automotive Application of real time adaptive wiener saddle River, NJ: Prentice –Hall, 2002.
filter for noise cancellation in a car environment,”
IEEE,2012,4673-2193. [7] J.Gorriz and J.Ramrez, “A Novel LMS Algorithm
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[8] K. A. Lee, W. S. Gan, and S. M. Kuo, Subband Adaptive
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trends in communication, C2SPCA 2013.
[9] C.Gabriela and M.Sarachin, “Echo Cancellation Using
[4] G.V.P.Chandra Sekhar Yadav and Dr. B. Ananda LMS Algorithm,” U.P.B Sci Bull., Series C, vol. 71, no.
Krishna, “Study of different adaptive filter algorithms for 4, 2009.
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[5] B.Widrow and S.D.Stearns, Adaptive Signal Processing,
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[6] S.Haykin, Adaptive Filter Theory, Fourth edition, Upper University, Taipei, Taiwan, ROC.

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