International Journal of Advanced Engineering Research and Technology (IJAERT), ISSN: 23488190

th
st
ICRTIET-2014 Conference Proceeding, 30 -31 August 2014

KAISER WINDOW BASED 21 TAP FIR FILTER FOR AUDIO

APPLICATIONS
1

Saseendran T K 2Rajesh Mehra

M E scholar Department of ECE, NITTTR Chandigarh. India
2
Associate Professor Department of ECE, NITTTR Chandigarh India
1

ABSTRACT
Digital filtering is one of the main fundamental aspect of
Digital signal processing. In this paper low-pass FIR
filter is implemented using an efficient adjustable
window function based on Kaiser window technique. In
this adjustable window function for a fixed length the
bandwidth of main lobe and side lobe amplitude can be
varied by changing the value of . The drawback of
fixed main lobe width and the side lobe amplitude can be
overcome with the help of the Kaiser window. In this
work, firstly audio signal was recorded in wave format
after that designed FIR filter applied to this recorded
audio signal. Signal comparison between the original
audio and filtered audio signal shows that the high
frequency component of speech signal are significantly
removed by using this FIR low-pass filter. The
performance of designed filter is analyzed and compared
by using three different values of using Mat lab.
Keywords- Kaiser window, FIR filter, FFT, DSP Lowpass filter.

1. INTRODUCTION
Digital signal processing (DSP) is the mathematical
manipulation of an information signal to modify or
improve it in some way. It is characterized by the
representation of discrete time, discrete frequency, or
other discrete domain signals by a sequence of numbers
or symbols and the processing of these signals. The goal
of DSP is usually to measure, filter and/or compress
continuous real-world analog signals. The first step is
usually to convert the signal from an analog to a digital
form, by sampling and then digitizing it using an analogto-digital converter (ADC), which turns the analog
signal into a stream of numbers. This process is more
complex than analog processing and has a discrete value
range, the application of computational power to digital
signal processing allows for many advantages over
analog processing in many applications, such as error
detection and correction in transmission as well as data
compression, [1].

Filters are a basic component of all signal processing

and telecommunication systems. Filters are widely
employed in signal processing and communication
systems in applications such as channel equalization,
noise reduction, radar, audio processing, video
processing, biomedical signal processing, and analysis of
economic and financial data.. Digital filters are divided
into two categories, including Finite Impulse Response
(FIR) and Infinite Impulse Response (IIR), [2].
Digital filter plays an important role in digital signal
processing applications. Digital filters are widely used in
digital signal processing applications, such as digital
signal filtering, noise filtering, signal frequency analysis,
speech and audio compression, biomedical signal
processing and image enhancement etc. A digital filter is
a system which passes some desired signals more than
others to reduce or enhance certain aspects of that signal.
It can be used to pass the signals according to the
specified frequency pass-band and reject the frequency
other than the pass-band specification.
The basic filter types can be classified into four
categories: low-pass, high-pass, band-pass, and bandstop. On the basis of impulse response, there are two
fundamental types of digital filters: Infinite Impulse
Response (IIR) filters, and Finite Impulse Response
(FIR) filters,[1]. Finite Impulse Response digital filter
has strictly exact linear phase, relatively easy to design,
highly stable, computationally intensive, less sensitive to
finite word-length effects, arbitrary amplitude-frequency
characteristic and real-time stable signal processing
requirements etc. Thus, it is widely used in different
digital signal processing applications, [1] [2].
FIR filter is described by differential equation. The
output signal is a convolution of an input signal and the
impulse response of the filter.

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International Journal of Advanced Engineering Research and Technology (IJAERT), ISSN: 23488190
th
st
ICRTIET-2014 Conference Proceeding, 30 -31 August 2014

x(n) is the input signal and. h(n) is the impulse response

of fir filter.
The transfer function of a causal FIR filter is obtained by
taking the z-transform of impulse response of FIR filter .

Fourier transform to the ideal frequency characteristics

of digital filter. Then this unit sample response must be
truncated at some point, this process is equivalent to
multiplying it by a finite length window function. After
truncation and windowing, an FFT is used to generate
the corresponding frequency response of FIR filter. The
frequency response can also be modified by choosing
different window functions.

Most Common type filters include a low-pass filter,

which pass through the frequencies below their cutoff
frequencies, and progressively attenuates frequencies
above the cutoff frequency of a signal according to
desired requirements. There are many straightforward
techniques for designing FIR digital filters to meet
arbitrary frequency and phase response specifications,
such as window design method or frequency sampling
techniques. The Window method is the most popular and
effective method because this method is simple
convenient, fast and easy to understand. The main
advantage of this design technique is that the impulse
response coefficient can be obtained in closed form
without the need for solving complex optimization
problems, [3] [4].

In the study of Fourier transform of these different

Fixed window functions, for the fixed length the
Rectangular window provides smallest main lobe width
but the highest peak of side lobe among them, So
Rectangular window is not widely used in digital signal
processing applications. The Hanning and Hamming
window provides good side lobe attenuation compare to
rectangular window, so these windows are commonly
used in different DSP applications. For higher side lobe
attenuation Blackman window is used but the Blackman
window has a wider main lobe width compare to
Hanning and Hamming window. The Kaiser window is a
kind of adjustable window function which provides
independent control of the main lobe width and ripple
ratio. But the Kaiser window has the disadvantage of
higher computational complexity due to the use of
Bessel functions in the calculation of the window
coefficients, [8].

Window functions can be divided into two categories:

Fixed and Adjustable window functions. Mostly used
fixed window functions are: Rectangular window,
Hanning window, Hamming window and Blackman
window. The Kaiser window is a kind of adjustable
window function. Here these different widows are used
for the Digital FIR filter designing and spectral
performance analysis, [4]-[7]. FIR filter design using a
new window function is given in [8]. The Performance
Enhancement Study of FIR Filters Based on adjustable
Window Function is given in [9]. A novel window
function yielding suppressed main lobe width and
minimum side lobe peak is described in [10]. In this
paper an efficient adjustable window function based on
Kaiser Function is used for designing an FIR filter.

3. KAISER WINDOW TECHNIQUE

1. Desired magnitude response of the ideal filter is given
by the equation [ 4]

(3)
2. Ideal Impulse response hd(n) of filter was obtained by
applying inverse Fourier transform to the desired ideal
frequency response Hd(ejw) of digital filter [4].
(4)

2. FIR DESIGN METHODS

For designing the digital FIR filters at first, the desired
filter responses are characterized, and their coefficient
values are calculated for a causal FIR filter. There are
different methods to find the coefficients of digital filter
from frequency specifications. They are Fourier series
method, window method, Frequency sampling method
and the optimal filter design method. A simple and
efficient way to design an FIR filter is window method.
In the Window Design Method, The unit impulse
response of ideal filter was obtained by applying inverse

4. The window function w(n) is selected according to

requirements of transition bandwidth and stop-band
attenuation. In this paper an efficient window
function based on Kaiser window function given in
equation (5) is used for designing FIR filter

Divya Jyoti College of Engineering & Technology, Modinagar, Ghaziabad (U.P.), India

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International Journal of Advanced Engineering Research and Technology (IJAERT), ISSN: 23488190
th
st
ICRTIET-2014 Conference Proceeding, 30 -31 August 2014

Where is the adjustable parameter and I0(x) modified

zeroth - order Bessel function, which can be expressed in
power series form

I0(x) = 1 + [(x/2)r/r!]2
r=1

(6)

which is seen to be positive for all real value of u .In

practice it is sufficient to keep only the first 20 terms in
the simulation of equation (6) . The parameter control
the minimum attention s = -20 log10(s) in the stop band
of the windowed filter response. Formulas for estimating
and the filter order N = 2M,for specified s and
normalized transition band width have been
developed by Kaiser. The filter order N is estimated
using the formula. [2].

Firstly an audio signal is recorded in wave format and

load into Mat lab by using the following command,[5].
[x fa] =
wavread ('audio.wav') x is the sample audio
signal and fa is the frequency of audio signal. The
frequency of the audio signal fa is 8192 Hz. Suppose we
want to remove the high-frequency components above
frequency 8192 Hz of the audio signal. The cutoff
frequency of the low-pass FIR filter is selected fc =
10800 Hz. and fs=40800Hz.The summation term is 21
and the value of can be change to 0.5, 3.5 and 8.5. The
cutoff frequency and value of of the filter can be
changed according to desired specifications. The below
figure shows the original audio signal and its analyzed
wave forms such as filter order and the cutoff frequency
can also be changed according to given specifications.
20
0

s 8
2.285()

(7)

magnitude in dB

The is the normalized transition band width the pass

band ripple p is approximately equal to s. For most
practical purpose the summation up to 21 term is
sufficient .I0(x) is listed for different values of x in table
1, [9].

x
0.0
0.1
0.2
0.4
0.6
0.8
1.0

I0(x)
1.0000
1.0025
1.0100
1.0404
1.0921
1.1665
1.2661

x
1.2
1.4
1.6
1.8
2.0
2.2
2-4

I0(x)
1.3938
1.5534
1.7500
1.9696
2.2796
2.6292
3.0492

x
2.6
2.8
3.0
3.2
3.6
3.8
4.0

I0(x)
3.5532
4.1574
4.3306
5.7472
8.0278
9.5169
11.302

-60
-80
-100
-120
-140

0.1

0.2

0.3

0.4
0.5
0.6
normalised frequensy

0.7

0.8

0.9

Figure 1 frequency response low pass filter using Kaiser

window for different values of (.0.5,3.5 and 8.5)
Time domain

Frequency domain
30

20
10

0.8

Amplitude

Table 1. The zeroth-order Bessel function I0(x) for

different values of x

-40

Magnitude (dB)

N =

-20

0.6

0.4

0
-10
-20
-30

0.2
-40
0

-50
5

10
Samples

15

20

0.2
0.4
0.6
0.8
Normalized Frequency ( rad/sample)

Figure 2 Kaiser window in time and frequency domain

for =0.5

5. DESIGN SIMULATIONS
FIR filters using the window method are widely used in
different audio signal processing such as audio signal
filtering noise reduction, frequency boosting etc. This
type of digital filter is used to modify the frequency
response of an audio signal according to desired speech
processing application. Here we use Kaiser window for
different values of (for 0.5,3.5 and 8.5).The result
shows the different values of as shown in figure 1.The
figure 2 shows the time and frequency response of the
filter using Kaiser window.

Figure 3 Frequency spectrum of original audio signal

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International Journal of Advanced Engineering Research and Technology (IJAERT), ISSN: 23488190
th
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ICRTIET-2014 Conference Proceeding, 30 -31 August 2014

2. Sanjit K. Mitra, Digital Signal Processing: A

computer-base approach, Tata McGraw-Hill,pp 527566, 2001
3. Oppenheim, R. Schafer, and J. Buck, Discrete-Time
Signal Processing second edition, Prentice-Ha, pp 354372 1999.
Figure 4 Frequency spectrum of analyzed audio signal
( =0.5)

Figure 5 Frequency spectrum of analyzed audio signal

( =3..5)

4. Saurabh Singh Rajput, Implementation of FIR Filter

using Adjustable Window Function and Its Application
in Speech Signal Processing, International Journal for
electrical, Electronic and mechanical control Volume 1
Issue 1 November 2012, pp 41-44
5. Sonika Gupta, Aman Panghal Performance
Performance Analysis of FIR Filter Design by Using
Rectangular, Hanning and Hamming Windows
Methods International Journal of Advanced Research in
Computer Science and Software Engineering Volume 2,
Issue 6, June 2012. pp 273-277.
6. Prof.GopalS.Gawande, Dr.K.B.Khanchandani, T.P.
Marode performance analysis of fir digital filter design
techniques International Journal of computing &
Corporate research volume 2 issue 1 January 2012. pp
123-126

Figure 6 Frequency spectrum of analyzed audio signal

( =8..5)

7. S. M. Shamsul Alam , Md. Tariq Hasan Performance

Analysis of FIR Filter Design by Using Optimal,
Blackman Window and Frequency Sampling Methods.
International Journal of Electrical & Computer Sciences
IJECS-IJENS Vol: 10 No:01.pp 116 -120

6. CONCLUSION
Digital filter plays a very important role in different
digital signal processing applications. Digital filters can
also play a major role in audio signal processing
applications. In this paper, Kaiser window based 21
terms FIR filter designed and analyzed the audio signal
for different values of . The simulation result shows
that in the FIR filter = 8.5 is better performance than
other values of . This filter can also be used for other
audio signal processing applications. In digital filters
different parameters such as filter order and cut off
frequency can also be changed according to given
specifications.
REFERENCES
1. J.G.Proakis and D.G.Manolakis, Digital Signal
Processing Principles, Algorithms and Applications
Third edition Prentice-Hall pp 342-425, 2002

8. Mahrokh G. Shayesteh and Mahdi MottaghiKashtiban FIR filter design using a new window
function 1/09,2009 IEEE.
9. P. Remeshbabu. Digital signal processing.2nd edn
Scitech publications Chennai.pp 600 -645,
10. S.Salivahanan, A.Vallavaraj,C. Gnanapriya, Digital
Signal Processing, Tata McGrawHill, pp 309-409, 2006.
Authors
Saseendran.T. K received Bachelors of
Technology degree in Electronics and
Communication
Engineering
from
College of Engineering, Trivandrum,
KU
Kerala, India., He is pursuing
Masters of Engineering degree in
Electronics and Communication Engineering from
National Institute of Technical Teachers Training and

Divya Jyoti College of Engineering & Technology, Modinagar, Ghaziabad (U.P.), India

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International Journal of Advanced Engineering Research and Technology (IJAERT), ISSN: 23488190
th
st
ICRTIET-2014 Conference Proceeding, 30 -31 August 2014

Research, Punjab University, Chandigarh, India. His

current research interests are in Digital Signal
Processing and Embedded System Design
Rajesh Mehra received the Bachelors
of Technology degree in Electronics
and Communication Engineering from
National Institute of Technology,
Jalandhar, India in 1994, and the
Masters of Engineering degree in
Electronics and Communication Engineering from
National Institute of Technical Teachers Training &
Research, Punjab University Chandigarh, India in 2008.
He is pursuing Doctor of Philosophy degree in
Electronics and Communication Engineering from
National Institute of Technical Teachers Training &
Research, Punjab University, Chandigarh, India.
He is an Associate Professor with the Department of
Electronics & Communication Engineering,, National
Institute of Technical Teachers Training & Research,
Ministry of Human Resource Development, member of
IEEE and ISTE Chandigarh, India. His current research
and teaching interests are in Signal, and
Communications Processing, Very Large Scale
Integration Design. He has authored more than 175
research publications including more than 100 in
Journals Mr. Mehra is member r of IEEE and ISTE.

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