Elliptic High Pass Filter

PA M A N TA S A N N G L U N G S O D N G M AY N I L A
DESIGN OF A 4TH ORDER INFINITE IMPULSE RESPONSE (IIR)

ELLIPTIC FILTER WITH HIGH-PASS CHARACTERISTICS
A Case Study Presented to the Faculty of

College of Engineering and Technology
Pamantasan ng Lungsod ng Maynila
In Partial Fulfillment of the Course

ECE411 Signals, Spectra, and Signal Processing
ALBANIA, Joshua G.
CABUHAYAN, Judith B.
CARTER, Kenway V.
ESPINOSA, Alpha Mae G.
REYES, Angela Mariz V.
Bachelor of Science in Electronics and Communications Engineering
1
October 2016
2
ACKNOWLEDGEMENTS
First and foremost, we give all the gratitude and praises to God for providing us
everything that weve needed to complete this research project.
We would also like to express appreciation and thanks to our professor, Engr. Reynaldo
Ted L. Peas II, for the knowledge that he had passed on to his students, the researchers.
Finally, the researchers have profound gratitude towards their families who have
garnered a lot of patience, motivation, and support through every busy day.
3
TABLE OF CONTENTS
i. Acknowledgements
ii.Table of Contents
iii. List of Figures
iv. List of Tables
I. The Problem and its Background
A. Introduction ......... 6
B. Background of the Study ......... 7
C. Statement of the Problem ........ 7
D. Objectives ........ 8
E. Scope and Limitations ......... 8
F. Definition of Terms ......... 8
II. Review of Related Literature and Studies
A. Foreign Literature and Studies ....... 10
B. Synthesis of Related Literature to the Case Study ..... 15
III. Methodology
A. Procedures .......... 16
1. Algorithm ......... 16
2. Flowchart ......... 18
B. Experiment Set-up ......... 19
1. System Block and Schematic Diagrams ...... 19
2. Characterization of Input Data ..... 19
3. Data Processing and Filter Implementation ..... 19
4
a. Mathematical Processing ....... 19
b. Simulation .......... 23
IV. Data and Computation
A. Results ........ 24
B. Validation.28
V. Summary of Findings, Conclusions and Recommendations
A. Summary of Findings ......... 30
B. Conclusions ......... 30
C. Recommendations ....... 31
Bibliography ........... 32
Appendices ......... 33
About The Proponents ........ 42
5
LIST OF FIGURES
Figure 1 An efficient realization of IIR
Figure 2 A 3rd Order High-Pass Elliptical Filter
Figure 3 Designed 4th Order High-Pass Elliptical Filter
Figure 4 Magnitude and Phase Response of Elliptic High-Pass Filter
Figure 5 Pole-Zero Plot of Elliptic High-Pass Filter
Figure 6 Filtered Star Wars Theme
Figure 7 Filtered Silento Watch Me
Figure 8 Filtered Imperial March
Figure 9 Filtered I Dont Like It I Love It
Figure 10 Filtered Invisible
Figure 11 Magnitude and Phase Response of Elliptic Low-Pass Filter
Figure 12 Pole-Zero Plot of Elliptic Low-Pass Filter
Figure 13 Original I dont like it, I love it
Figure 14 Filtered I dont like it, I love it
Figure 15 Original Imperial March
Figure 16 Filtered Imperial March
Figure 17 Original Invisible by Hunter Hayes
Figure 18 Filtered Invisible by Hunter Hayes
Figure 19 Original Silento Watch Me
Figure 20 Filtered Silento Watch Me
Figure 21 Original Star Wars Theme
Figure 22 Filtered Star Wars Theme
6
Figure 23 Direct Form I
Figure 24 Direct Form II
LIST OF TABLES
Table 1 Frequency Specification
Table 2 Parameters and Values of the Designed Filter
7
Chapter I
The Problem and its Background
A. Introduction
Filters are systems that are mainly created for the purpose of removing unwanted
frequencies, such as noise, from a certain signal. They generally function by only allowing, and
sometimes amplifying, certain values of an input signals frequency to pass through, and by
disallowing the passage of or removing the rest of the signals unwanted frequency. In an in-
depth perspective, filters work by altering the amplitude and/or phase characteristics of a signal
with respect to frequency. They are an important part of modern signal processing as they are
very effective in suppressing interference and, as technology prevails in advancement, the
application of filters on devices is ever-growing.
Digital filters are a very important part of DSP. In fact, their extraordinary
performance is one of the key reasons that DSP has become so popular. As mentioned before,
filters have two uses: signal separation and signal restoration. Signal separation is needed when a
signal has been contaminated with interference, noise, or other signals. [6]
Today, digital filters are more commonly used than analog filters because of
certain advantages such as high accuracy, flexibility and adaptability, and the difficulty of
creating digital filters is easier compared to that of analog filters. Digital filters can be divided
according to the characteristics of their impulse response function. The two types are the infinite
impulse response (IIR) filter, and the finite impulse response (FIR) filter. IIR filters have an
infinite duration of their impulse response, while FIR filters impulse response is only limited to a
certain duration.
Although there are various types and classifications of filters, the focus of this
research is the creation of an infinite impulse response (IIR) elliptic filter with high-pass
characteristics. The problems in designing the said filter will also be highlighted in this research.
8
B. Background of the Study
The most complex part of creating a digital filter, just like in many things, is the
designing process. Infinite impulse response (IIR) filters, unlike finite impulse response (FIR)
filters, have a feedback, or a recursive part of a filter, and are, therefore, known as recursive
digital filters. For this reason, IIR filters have much better frequency response than FIR filters of
the same order. But, unlike FIR filters, their phase characteristic is not linear which can cause a
problem to the systems which need phase linearity. So, it is not preferable to use IIR filters in
digital signal processing when the phase is of the essence. On the other hand, when the linear
phase characteristic is not important, the use of IIR filters is an excellent solution.
Elliptic filters, also known as Cauer or Zolotarev filters, achieve the smallest filter
order for the same specifications, or, the narrowest transition width for the same filter order, as
compared to other filter types. On the negative side, they have the most nonlinear phase response
over their passband. As the ripple in the stopband approaches zero, the filter becomes a type
I Chebyshev filter. As the ripple in the passband approaches zero, the filter becomes a type
II Chebyshev filter and finally, as both ripple values approach zero, the filter becomes a
Butterworth filter. The researchers aim to create a fully functioning elliptic filter with high-pass
characteristics.
C. Statement of the Problem
The researchers, in doing this study, wanted to design and simulate an elliptic high-
pass filter with minimal complexity, and thus arrived on the following questions:
General
1. How can a working elliptic filter with high-pass characteristics be achieved?
Specific
1. What are the magnitude and frequency specifications of the input audio signals that will
be filtered?
9
2. What specifications should be set in order to obtain the digital high-pass transfer function
of the designed filter?
3. What will be the output after filtering the input audio signals?
D. Objectives
Using Matrix Laboratory (MATLAB), the researchers intended to design and

simulate a working elliptic filter with high-pass characteristics using some audio input signals,
specifically aiming the following:
1. To determine the magnitude and frequency specifications of the input audio signals that
will be filtered.
2. To obtain the digital high-pass transfer function of the designed filter with the
specifications set by the researcher.
3. To acquire and analyze the output after filtering the input audio signals.
E. Scope and Limitations
The main scope of the study is the construction of an elliptic high-pass infinite impulse
response (IIR) filter. The researchers limited the following specifications: a p =2.6 dB ,
a s=32dB ,W p=1000 Hz ,W s=1200 Hz , f c =1100 Hz and which will be used to
generate the filter. There are five audio files that will serve as inputs to be filtered.
F. Definition of Terms
The following relevant terminologies were used in the study and were listed
accordingly:
Attenuation the reduction in the level of power, current, or voltage.

Cut-off frequency is the frequency either above or below which the power output of a
filter has fallen to a given proportion of the power in the passband.
Elliptic filter also known as a Cauer filter, is a signal processing filter with equalized
ripple behavior in both the passband and stopband.
10
High-pass filter is a type of filter that passes signals with a frequency higher than
certain cut-off frequency and attenuates signals with frequencies lower than the cut-off
frequency.
Infinite Impulse Response (IIR) a property applying to many linear time-invariant
systems; systems that are distinguished by having an impulse response which does not
become exactly zero past a certain point, but continues indefinitely.
Jacobian Elliptic Functions are a set of basic elliptic functions and auxiliary theta
functions that are useful analogies to the functions of trigonometry, as indicated by the
notation sn for sin.
Passband is the range of frequencies or wavelengths that can pass through a filter.
Ripple factor a measure of effectiveness of a power supply filter in reducing the ripple
voltage; ratio of the ripple voltage to the dc output voltage.
Roll-off rate the rate of decrease in the gain above or below the critical frequencies of
a filter.
Selectivity factor a measure that compares the signal strength received against that of a
similar signal on another frequency; provides some immunity to covering interference.
Stopband is a band of frequencies, between specified limits, through a filter does not
allow signals to pass, or the attenuation is above the required stopband attenuation level.
Chapter II
11
Review of Related Literature and Studies
This chapter provides materials such as articles and studies that are relevant to the
study being conducted.
A. Foreign Literature and Studies
Infinite Impulse Response (IIR) FILTER
IIR filter is used much in application such as high speed and low-power
communication transceivers systems. IIR pursue following properties that is the width of the
pass-band, stop-band, limited ripple at pass-band and limited ripple at stop-band.
IIR filters can be usually implemented using structures having feedback (recursive
structures). The present and the past input samples can be described by the following equation,
[3]
N M
y ( m ) = ak y (mk )+ bk x ( mk ) Equation 2.1
k=1 k=0
Figure 1. An efficient realization of IIR [3]
12
IIR digital filters are characterized by a real rational transfer function of z -1 or,
equivalently by a constant coefficient difference equation. From the difference equation
representation, it can be seen that the realization of the causal IIR digital filters requires some
form of feedback. [3]
DIFFERENCE BETWEEN FIR & IIR
The main difference between IIR filters and FIR filters is that an IIR filter is more
compact in that it can habitually achieve a prescribed frequency response with a smaller number
of coefficients than an FIR filter. An IIR filter can become unstable, whereas an FIR filter is
always stable. [1]
IIR filters have many advantages as follows [4]:
i. Less number of arithmetic operations are required in IIR filter.

ii. There are shorter time delays in these filters.
iii. IIR Filters have similarities with the analog filters.
iv. Lesser number of side lobes in the stop band.
v. They are more susceptible to noises.
ELLIPTIC FILTER
Elliptic filters, also known as Zolotarev filters, achieve the smallest filter order for
the same specifications, or, the narrowest transition width for the same filter order, as compared
to other filter types. On the negative side, they have the most nonlinear phase response over their
passband. The following table compares the basic filter types with regard to filter order and phase
response [5]:
13
Elliptical filter can also be called as Cauer filters. Elliptic filters are equiripple in
both the passband and stopband. They generally meet filter necessities with the lowest order of
any supported filter type. For a given filter order, elliptic filters minimize transition width of the
passband ripple and stopband ripple.
The magnitude response of a low pass elliptic filter as a function of angular

frequency is given by [3],
1
H ( j )= Equation2.2
2 2
1+ R (

P
)
Where,
Rk is the nth order elliptic rational function.
is the cut off frequency.
is the ripple factor.
is the selectivity factor.
JACOBIAN ELLIPTIC FUNCTIONS
Jacobian elliptic functions are a fascinating subject with many applications. Here,
wegive some definitions and discuss some of the properties that are relevant in filter design .The
elliptic function w = sn(z, k) is defined indirectly through the elliptic integral[5]:
w
d dt
z= = , w=sin
0 1k sin 0 (1t )(1k 2 t 2)
2 2 2
Equation 2.3
where the second integral was obtained from the first by the change of variables t
= sin and w = sin. The parameter k is called the elliptic modulus and is assumed to be a real
number in the interval 0 k 1. Thus, writing = (z, k), the function sn is defined as[5]:
14
w = sn(z, k)= sin (z, k) Equation 2.4

The three related elliptic functions, cn, dn, cd, are defined by:
w = cn(z, k)= cos (z, k) Equation 2.5
w = dn(z, k)= 1k 2 sn 2 (z , k ) Equation 2.6
w = cd(z, k) = (cn(z,k))/(dn(z,k)) Equation 2.7
In filter design, only the functions sn and cd are needed. In the limits k = 0 and k = 1, we obtain
the trigonometric and hyperbolic functions, respectively: [5]
sn(z, 0)= sin z , sn(z, 1)= tanh z Equation 2.8
cn(z, 0)= cos z , cn(z, 1)= sech z Equation 2.9
dn(z, 0)= 1 , dn(z, 1)= sech z Equation 2.10
cd(z, 0)= cos z , cd(z, 1)= 1 Equation2.11
THIRD ORDER HIGH PASS ELLIPTIC FILTER
It is necessary to choose a suitable frequency range in order to design basic types

of filters like High pass filters. Table 1 indicates the frequency specification for designing various
types of IIR filter. The sampling rate of the speech signal is 8000 and the number of bits per
sample is 16.[3]
15
Table 1. Frequency Specification
Filter Type Frequency (Hz)

Low pass fp=2000Hz, fs=3000Hz
High pass fp=3000Hz, fs=2000Hz
Band pass f1=1500, f2=2000,
f3=1000,f4=2500
Band stop f1=1000,
f2=3000,f3=1500,f4=2500
In this paper, three types of infinite impulse response filter i.e. Butterworth, Chebyshev
type I and Elliptical filter have been discussed theoretically and experimentally. Butterworth,
Chebyshev type I and elliptic low pass, high pass, band pass and band stop filter have been designed
in this paper using MATLAB Software. The impulse responses, magnitude responses, phase
responses of Butterworth, Chebyshev type I and Elliptical filter for filtering the speech signal have
been observed in this paper. Analyzing the Speech signal, its sampling rate and spectrum response
have also been found.[3]
16
Figure 2. A 3rd Order

High-Pass Elliptical Filter
Figure 2 represents the basic characteristics of a 3 rd order Elliptic High Pass Filter. Its
magnitude response in dB is normally negative. The phase response is very non-linear. Elliptic
filters offer steepest roll-off characteristics but are equiripple in both the pass- and stopbands.
Elliptic filter has a shorter transition region than the Chebyshev type I filter .The main reason is
that it allows ripple in both the stop band and pass band. It is the addition of zeros in the stop
band that causes ripples in the stop band. Elliptical filters have better performance for speech
signal analysis i.e. voice signal can be smoothly heard. [3]
C. Synthesis of Related Literature to the Studies
The foreign literature introduces the definition and advantages of an IIR filter.
In addition, this includes the elliptic filter description, parameter specifications, its
characteristics and advantages. The given transfer function became very helpful to the
researchers. Thus, the proponents have found the materials useful in the study being
17
conducted. The third order high-pass elliptic filter example serves as a guide in obtaining the
desired filter design specifications.
18
19
CHAPTER III
Methodology
A. Procedures
Tools and Equipment
In designing the infinite impulse response elliptic filter, the researchers made use of
the following software: Matrix Laboratory and Audacity. Audacity was used to clip the mp3
files down to the desired time duration and sampling frequency. Matrix Laboratory was used
to compute for the parameters needed, and to simulate the resulting filter design.
General Procedures
In general, the resulting filter design will accept an audio signal as the input. The
system will then filter the input and produce a modified audio signal, with respect to the
specifications, as its output.
1. Algorithm
a) Select the type and the characteristic of the filter to be designed. The researchers
chose to design a high-pass elliptic filter.
b) Choose the Passband Gain, Stopband Gain, Passband Frequency, Stopband
Frequency, Cut-off Frequency, and the specifications of the input audio signal. Set
the passband gain at -2.6dB, stopband gain at -32dB, passband frequency at 1000Hz,
stopband frequency at 1200Hz, and cut-off frequency at 1100Hz. Five different audio
files in mp3 format with sampling frequency of 44100Hz and bitrate of 320kbps.
c) Compute for the Filter Order using eq.no.5.
Cei(rt) Cei ( 1k 2 )
ne = Equation 3.5
Cei( 1rt 2) Cei (k )
d) Compute for the poles Real and Imaginary part.

e) Find the locations of the Zeros (purely imaginary).
f) Generate the Low-pass Transfer Function. Obtain the magnitude response of the
generated transfer function and verify if it is a low-pass filter.
g) Convert the Analog Low-pass Filter to a Digital Low-pass Filter with the use of
Bilinear Transformation.
20
M k
1+ z1 N M .
(
K =1
1
1+ z 1
1z
k
Z )
N
1+ pk 1
( 1
1 pk
Z
)
K=1
M Equation 3.25
( 1z k )
1 NM . K=1
N
.
( 1p k )
K=1
H ( z ) =H 0
h) Transform from digital low-pass filter to digital high-pass filter using the eq.no.26.
k k
1 b x (nk )

m
y(n)= 1 a y ( nk ) +
k k
Equation 3.26
k=0

n

k=1
i) Obtain the Magnitude and Phase Response of the designed filter with the use of
MATLAB R2013a.
j) Compare the results or responses of the designed filter to the 3 rd elliptic high-pass
filter to set as a validation.
k) If the desired output with respect to the specifications set by the researchers are
satisfied, convert the audio files into audio signals.
l) Filter the converted audio signals using the designed filter.
m) Observe the output signal.
2. Flowchart
21
22
The researchers have designed a system that will accept input parameters namely, the
passband frequency, stopband frequency, passband gain, stopband gain, and cut-off
frequency, in order to compute for the parameters needed, which are the filter order, poles,
and zeros, to obtain the normalized transfer function. By using bilinear transformation, the
resulting low-pass filter created will be converted to a high-pass filter. The magnitude and
phase response will determine how the system will modify the input audio signal.
B. Experiment Set-up
1. System Blocks and Schematic Diagrams
Design an Elliptic Input audio files and

Filter the audio signals
using the designedconvert
High-Pass Filter filter it into audio
signals
2. Characterization of the Input Data

The input data are five audio files that are in mp3 format. They have the same
sampling frequency of 44100 Hz and bit rate of 320 kbps. The audio time duration for the
input data are as follows: Watch me Whip- Silento 5 seconds, Invisible-Hunter Hayes
6 seconds, Star Wars Opening Sound Track 6seconds, I dont like it, I love it
4seconds and Imperial March-Star Wars OST 4 seconds.
Using MATLABs command audioread, the researchers were able to read and store
the audio data as y.
3. Data Processing and Filter Implementation
a. Mathematical Processing
The first step in the mathematical computation is computing for the ratio rt with the
use of equation 3.1.
23
wp
rt= Equation 3.1
Ws
Where:
w p is the passband frequency
W s is the stopband frequency
The next steps are the computation of the kernel k and the ripple factor with
the use of equation 3.2 and equation 3.3.
k =(10
0.1a pass 0.1astop
1)/(10 1) Equation 3.2
Where:
A pass is the passband gain
A stop is the stopband attenuation
= 100.1 A 1 pass
Equation 3.3
Where:
A pass is the passband gain
The next step is the computation of the minimum order required for the designed
filter with the use of equation 3.4.
Cei(rt) Cei ( 1k 2 )
ne =
Cei( 1rt ) Cei (k )
2
Equation3.4
Where:
Cei is the complete elliptic integral of the first kind defined as:
/2

2( )
Cei ( k ) =u , k = (1k 2 sin2 x)1 /2 dx
0
Equation 3.5
Afterwards calculate the variable (Vo) needed for the determination of the pole and
zero locations with the use of equation 3.6.
24
1 1
Cei(rt) s c ( , k )
V 0=
n cei(kn)
Equation 3.6
Where:
1
Sc (v, k) is the inverse jacobi elliptic function.
x
dt
Arcsc ( x , k )= Equation 3.7
0 ( 1+t ) (1+ kn t )
2 2 2
Where kn=k
In order to compute for the parameters needed in obtaining the transfer function, it is
necessary to be familiar with the Jacobi elliptic functions which are defined as:
sn ( u , k )=sin ( ) Equation 3.8
cn ( u , k )=cos() Equation 3.9
sc ( u , k )=tan ( ) Equation3.10
d
dn ( u , k )= Equation 3.11
du
The next step is the computation of the real and imaginary part of the poles using the
parameters and functions mentioned earlier, the complete formula for obtaining the poles are
given in equation 3.12 and equation 3.13 respectively.
cn[f (m, rt )] dn[ f ( m ) , rt ] sn ( v0 , 1r t 2 ) cn (v 0 , 1r t 2)

m=
1d n2 [f ( m ) , rt ] s n2 (V 0 , 1r t 2 )
Equation 3.12
sn[f ( m ) , rt] dn(v 0 , 1r t )

2
m = Equation 3.13
1d n 2 [f ( m ) , rt ] s n2 (V 0 , 1r t 2 )
25
Where:
Cei(rt) (2 m+1) N
f ( m )=
n
, m=0,1 .
2 ( )
1(n even)
Equation 3.14
Cei(rt) (2 m+2) N 1
f ( m )=
n
, m=0,1 .
2 (
1(n odd))
Equation 3.15
In the case of odd-order approximations, the computation of the location of the first-
order denominator pole on the negative real axis is at
sn (v 0 , 1r t ) cn(v 0 , 1r t )
2 2
r= Equation3.16
1s n2 (V 0 , 1r t 2 )
The next steps are the computation of the real and imaginary parts of the location of
the zeroes that will be purely imaginary on the j axis, with the use of equation 3.17 and
equation 3.18.
zm=0.0 Equation3.17
1
zm= Equation 3.18
rt sn [f ( m) , rt ]
The next step is the computation of the variables that will be used to compute for the
analog approximation function with the use of equation 3.19 up to equation 3.22.
B 1 m=2 m Equation 3.19
2 2
B 2 m= m +m Equation 3.20
26
A 1 m=2 zm=0.0 Equation 3.21
A 2 m= 2zm+ 2zm=2zm Equation 3.22
Where:
m= 0,1 (n/2)-1 (even n)
m= 0,1 [(n-1)/2]-1 (odd n)
Then, determine the analog transfer function using equation 3.23.
For even-order approximations:

( 100.05 a ) ( B 2 m ) ( s2 + A1 m s+ A 2 m )
pass
M M
H e ,n ( S )= 2
Equation 3.23
( A 2 m ) ( s + B1 m s+ B2 m )
M M
For odd-order approximations:
r ( B2 m ) ( s 2+ A 1m s + A 2 m )
M M
he ,n ( S )= Equation 3.24
(s+ r ) ( A 2 m ) ( s 2+ B1 m s+ B2 m )
M M
Afterwards, transform the given analog lowpass filter into the appropriate digital filter by
bilinear transformation with the use of equation 3.25.
27
M k
1+ z1 N M .
(
K =1
1
1+ z 1
1z
k
Z ) Equation 3
N
1+ pk 1
( 1
1 pk
Z
)
K=1
M .25
( 1z k )
1 N M . K=1
N
.
( 1 pk )
K =1
H ( z )=H 0
Lastly, transform the digital lowpass filter to a digital highpass filter using the equation 3.26.
1 k bk x (nk )

m
y(n)= 1 a y ( nk ) +
k k
Equation 3 .26.
k=0

n

k=1
b. Simulation
The function created by the proponents will make use of the inputted values of the
passband frequency (Wp), stopband frequency (Ws), passband gain (Apass), stopband
gain (Astop), and cut-off frequency (Fc) to calculate the parameters necessary for
obtaining the normalized transfer function of the filter such as the filter order, the real and
imaginary parts of the poles, and the locations of the zeros. By making use of the bilinear
transformation, the normalized transfer function will be converted from the s-domain to
the z-domain, making it easier to convert the resulting low-pass filter to the desired high-
pass filter.
The frequency response of the filter, both magnitude and phase response, will be
shown in a graph to allow the user to check if the designed elliptic high-pass filter is
28
satisfactory. If it is deemed so, the audio signals can now be put through the system for
filtering and be played for comparison with the unfiltered audio files.
Some of following MATLABs built-in function are used for simulation: ellipj for
computing the Jacobi elliptic functions, ellipticK for computing the complete elliptic
integral of the first kind, audioread for reading audio files, and soundsc for scaling data
and playing as sound. The rest are pure mathematical equations and MATLABs basic
functions. The proponents set the following specifications: passband frequency, 1000Hz;
stopband frequency, 1200Hz; passband gain, -2.6dB; stopband gain, -32dB; and cut-off
frequency, 1100Hz.
CHAPTER IV
Data and Computation
A. Results
Upon using the proposed specifications in performing all necessary computations the
researchers obtained the following results:
The equation of poles and zeroes from eq. 3.13-3.14 and 3.17-3.18.
29
1 2 3 4
HEz = 0.44410.8400 z + 1.174 z 0.8400 z +0.4441 z
1 2 3 4
HEp = 3.7938+2.9243 z +5.2123 z +2.5517 z +1.5180 z
Finally, the generated elliptic high-pass transfer function from eq. 3.19-3.26.
1 2 3 4
0.44410.8400 z + 1.174 z 0.8400 z +0.4441 z
H(z) = 1 2 3 4
3.7938+ 2.9243 z +5.2123 z +2.5517 z +1.5180 z
The following Figures 3-4 representing the phase /magnitude responses and pole-zero plot of
the elliptic high-pass filter are produced using the transfer function in MATLAB and are shown
respectively:
Figure 3. Magnitude and Phase Response of Elliptic High-Pass Filter
30
Figure 4. Pole-Zero Plot of Elliptic High-Pass Filter
For the succeeding Figures 5-9, the input audio files are used and processed using the final
filter design in MATLAB, producing the output audio signals as shown next:
1. Star Wars Opening Sound Track
Star Wars Opening Sound Track has a sample rate of 44100Hz and its duration is 6
seconds. Its bit rate is 320kbps.
Legend:
Blue Waveform Input

Signal Frequency
Green Waveform
Filtered Signal
Frequency
31
Figure 5. Filtered Star Wars Theme
2. Watch me Whip
Watch me Whip- Silento has a sample rate of 44100Hz and its duration is 5 seconds. Its bit
rateis 320 kbps.
Legend:
Blue Waveform Input

Signal Frequency
Green Waveform
Filtered Signal
Frequency
Figure 6. Filtered Silento - Watch Me
3. Imperial March-Star Wars OST

Imperial March-Star Wars OST has a sample rate of 44100Hz and its duration is 4 seconds.
Its bit rate is 320kbps.
Legend:
Blue Waveform Input

Signal Frequency
32
Green Waveform
Filtered Signal
Frequency
Figure 7. Filtered Imperial March

4. I dont like it, I love it- Flo Rida
I dont like it, I love it has a sample rate of 44100Hz and its duration is 4 seconds. Its bit
rate is 320kbps.
Legend:
Blue Waveform Input

Signal Frequency
Green Waveform
Filtered Signal
Frequency
Figure 8. Filtered I Dont Like It I Love It
5. Invisible-Hunter Hayes
33
Invisible-Hunter Hayes has a sample rate of 44100Hz and its duration is 6 seconds. Its bit
rate is 320kbps.
Legend:
Blue Waveform Input

Signal Frequency
Green Waveform
Filtered Signal
Frequency
Figure 9. Filtered Invisible

Figures 6-10 show the comparison of the original audio signals from the filtered audio
signals. The blue region indicates the input signal and the green region indicates the output signal. It
can be observed that the filter mostly affects the part of the signal beyond 2kHz with lower
amplitude, since it is designed to have high-pass characteristics. The succeeding table represents the
correlation between the input audio signal and the filtered audio signal.
Table 2. Correlation Coefficient of the input audio signal and the filtered audio signal
Audio File Name Correlation Coefficient
Imperial March 0.5684 B.
Silento - Watch Me Whip 0.1185 B.
Invisible by Hunter Hayes 0.3701
I Dont Like It I Love It 0.6064 B.
Star Wars Theme 0.6633 B.
Validation
34
Figure 10. Designed 4th Order High-Pass Elliptical Filter
Referring to
Fig.2 and Fig.10 above, its shown that the pole-zero plot of the designed
filter has the same orientation of poles and zeroes with the basic filter. Poles are located within
the left side of the unit circle, whereas zeroes are located on the unit circle.
The magnitude and phase response of the designed filter have lower gains compared to
the sample 3rd order filter. Even though both have nonlinear responses that cause distortion, it can
be observed that the basic filter has a more stable phase response.
Then by referring to Figures 11 and 12, it can be noticed that the low-pass characteristics
of the designed filter is very similar with that of the final high-pass filter design in Figure 10,
except that the orientations seem reversed for the magnitude and phase responses and also for the
locations of poles and zeroes on the pole-zero plots.
The low-pass magnitude and phase response, and its pole-zero locations are shown in the
following respectively:
35
Figure 11. Magnitude and Phase Response of Elliptic Low-Pass Filter
Figure 12. Pole-Zero Plot of Elliptic Low-Pass Filter
By using Audacity, the researchers were able to obtain the graphs of the input audio
frequency and the output audio frequency. The following graphs show the huge differences
between the original input and the filtered output audio signals. The x-axis shows the
frequency in Hertz and the y-axis shows the magnitude in decibels.
36
Figure 13. Original I dont like it, I love it
Figure 14. Filtered I dont like it, I love it
37
Figure 15. Original Imperial March
Figure 16. Filtered Imperial March
38
Figure 17. Original Invisible by Hunter Hayes
Figure 18. Filtered Invisible by Hunter Hayes
39
Figure 19. Original Silento Watch Me
Figure 20. Filtered Silento Watch Me
Figure 21. Original Star Wars Theme
40
Figure 22. Filtered Star Wars Theme
41
CHAPTER V
Summary of Findings, Conclusions and Recommendations
A. Summary of Findings
Five audio files with different durations are converted into audio signals. After
filtering, the results show that some parts of input audio signal became attenuated. This is
because the researchers designed an elliptic high-pass filter, which accepts high frequencies
within cut-off frequencies.
Considering elliptic characteristics and specifications: a p =2.6 dB ,

a s=32dB ,W p=1000 Hz ,W s=1200 Hz ,f c =1100 Hz , the researchers obtained
the elliptic high-pass filter transfer function:
0.44410.8400 z1+ 1.174 z2 0.8400 z 3 +0.4441 z4

H(z) =
3.7938+ 2.9243 z1 +5.2123 z 2 +2.5517 z3+1.5180 z4
Input Process
Five audio files namely: Star Wars Opening Sound Track, Watch me Whip, Imperial
March-Star Wars OST, I dont like it, I love it and Invisible-Hunter Hayes, are converted
into audio signals. The time duration of each sounds is six, five, four, four and six seconds
respectively. Using Matrix Laboratory Software, the researchers used the audio signals as the
inputs in their designed filter.
Output Process
After filtering, the results show that some parts of input audio signal became
attenuated. These parts are those with low frequency values. The output will now give the
resulting signal after it has passed through the designed filter.
42
B. Conclusions
Filters are systems that may have different applications depending on their
characteristics. Even though there are a lot of classifications of filters, choosing a specific one
may be done by just setting the filters specifications.
General
1. After a series of mathematical processes, a normalized transfer function of a low-pass

filter can be obtained. Verify using the MATLAB command freqz if the obtained
function exhibits the characteristics of a low-pass filter. Afterwards, by using bilinear
transformation, the digital low-pass normalized transfer function will be computed.
The researchers then transformed the digital low-pass normalized transfer function to
a digital high-pass normalized transfer function.
Specific
1. The magnitude and frequency specifications of the input audio signals are as follows:
I dont like it, I love it at -22.2dB and 2395Hz; Imperial March at -51.1dB and
10538Hz; Star Wars Theme at -39.8dB and 8405Hz; Silento Watch Me at -33.4dB
and 3748Hz; and Invisible by Hunter Hayes at -39.4dB and 4697Hz.
2. The prototype filter created by the proponents is an elliptic digital filter with high-
pass characteristics and the following specifications: passband frequency of 1000Hz,
stopband frequency of 1200Hz, passband frequency of -2.6dB, and a stopband gain of
-32dB. The normalized transfer function of the elliptic high-pass filter is given as
0.44410.8400 z1+ 1.174 z2 0.8400 z 3 +0.4441 z4

1 2 3 4
3.7938+ 2.9243 z +5.2123 z +2.5517 z +1.5180 z
3. The output audio signals, as shown in the validation, are seen to take the shape of the
frequency response of the designed filter. It can be observed that some parts of the
original audio signal have been attenuated after the input passed through the elliptic
filter. These parts appear below the cut-off frequency so it is evident that the designed
filter contains high pass characteristics.
43
C. Recommendations
The proponents would like to recommend future researchers to consider the

duration, sampling frequency and bit rate as dependent variables, aiming to make these
specifications higher to determine if there is significant effect on the output filter response. Other
types of audio inputs may also be used to test the effectiveness of the filter. Future researchers
who will decide to make a filter should have understood basic concepts used in digital signal
processing and are able to operate the programming language with ease. Critical thinking, logical
reasoning and patience is required in order to finish the designed filter.
BIBLIOGRAPHY
[1] J.S. Chitode, Digital Signal Processing, Technical Publication, Pune, ISBN: 9788184314243.
[2] L. D. Paarmann, Design and Analysis of Analog Filters: A Signal Processing Perspective,
2007.
[3] P. Podder, et al., Design and Implementation of Butterworth, Chebyshev-I and Elliptic Filter for
Speech Signal Analysis, Dept. of ECE of Khulna University of Engg., Bangladesh, (0975
8887) Volume 98 No.7, July 2014.
[4] R. Singh and S. K. Arya, Determining Optimum coefficients of IIR Digital Filter using
Analog to Digital Mapping, International Journal of Advancements in Computer Science and
Information Technology, Vol. 01, No. 01, pp.19-23, September 2011.
44
[5] S.J. Orfanidis, Lecture Notes on Elliptic Filter Design, Department of Electrical & Computer
Engineering, Rutgers University, November 2006.
[6] S.W. Smith, The Scientist and Engineers Guide to Digital Signal Processing, 2 nd Edition, San
Diego, CA: California Technical Publishing, 1999, pp. 261-263.
45
APPENDICES
Detailed Computation
1. Frequency Ratio Variable:
pass 1000
rt= = =0.8333
stop 1200
2. Kernel:

0.1a pass
10 1 1.8197
k= = = 0.0339
10
0.1a
1 stop
1584.9
3. Computation of the ripple factor:
= 100.1 A 1 = 0.9054
pass
4. Computation of the minimum order required:

Elliptic Integrals:
Cei( rt) = 2.0673
1k n2 ) = 4.7722
Cei
Cei ( 1rt 2 ) = 1.7172
Cei( kn) = 1.5712
Formula in getting the order:
Cei (rt)Cei ( 1k n 2)
n= = 3.6564
Cei( 1rt 2 ) Cei(kn)
N=round(n)=4
5. Calculate the variable (Vo) needed for the determination of the pole and zero locations
1
S c (v, k) is the inverse jacobi elliptic function.
0.9054
dt
Arcsn ( 0.9054,0 .0339 )= = 0.9532
0 ( 1t ) (1k
2 2 2
t )
Cei( rt) s c1 (1 , kn) 2.06730.9532

V 0= = = 0.3135
n cei(kn) 41.5712
46
6. Computation of the real part of the poles
cn[ f (m, rt )] dn [f ( m) , rt ] sn(v 0 , 1r t 2) cn (v 0 , 1r t 2)

m=F m=
1d n2 [f ( m ) , rt ] s n2 (V 0 , 1r t 2 )
(0.8767)(0.9162)(0.3069)(0.9517)
F 1= 2 2 = -0.2548
1(0.9162) (0.3069)
(0.2902)(0.6034)( 0.3069)(0.9517)
F2 = = -0.0530
1(0.6034 )2 (0.3069)2
sn[f ( m ) ,rt ] dn( v 0 , 1r t )

2
m =W m=
1d n 2 [f ( m ) , rt ] s n2 (V 0 , 1r t 2 )
(0.8767)(0.9855)
W 1= = 0.5147
1(0.9162)2 ( 0.3069)2
(0.9570)(0.9855)
W 2= 2 2 = 0.9766
1(0.9162) ( 0.3069)
7. Computation of the real and imaginary parts of the location of the zeroes that will be purely
imaginary on the j axis
zm=Fz 1=Fz 2=0.0
1
zm= = Wzm
rt sn [f ( m) , rt ]
1
Wz 1= = 2.4949
0.83330.4810
1
Wz 2= = 1.2540
0.83330.9570
8. Computation of the variables that will be used to compute for the analog approximation function
B 1 m=2 m
B 2 m= 2m +2m
A 1 m=2 zm=0.0
47
2 2 2
A 2 m= zm+ zm=zm
B 11=2 ( 0.514 7 )=0.5096
B 12=2 ( 0.9766 )=0.1059
B 21=0.25482 + 0.514 72 = 0.3299
B 22=0.05302 + 0.97662 = 0.9565
A12 = A11=0.0
A 21= 2.49492= 6.2244
A22=1.25402= 1.5724
9. Determination of the analog transfer function using the generalized formula
( 100.05 a ) ( b2 m ) ( s 2+ A 1 m s+a 2m )
pass
M M
H e ,n ( S )=
( a2 m ) ( s + B1 m s+ b2 m )
2
M M
H(S)= (0.029171*(s^2 + 5.6658)*(s^2 + 1.4858))/((s^2 + 0.51478*s + 0.34505)*(s^2 + 0.099331*s +

0.96004))
10. Transform the given analog filter into the appropriate digital filter by bilinear transformation
48
M k
1+ z1 N M .
(
K =1
1
1+ z 1
1z
k
Z )
N
1+ pk 1
( 1
1 pk
Z
)
K=1
M
nm

K =1
( 1z k )
1 . N
.
( 1 pk )
K =1
H ( z ) =H 0
( 1( 1.00002.4949i ) ) ( 1 (1.0000+ 2.4949i ) ) ( 1( 1.00001.2540i ) )

1(1.0000+1.2540 i)

*

( 0.0239 )( 18.5843 )
H ( z )= .
3.7938
( 1(0.7232+ 0.6907 i ) )(1(0.72320.6907i ) )(1(0.2225+0.9749 i ))(1(0.22250.9749 i ))

.
(1( 0.3643+0.5596 i ))(1( 0.36430.5596i ) )(1( 0.0211+ 0. 9470i ) )(1( 0.02110.9470 i ))
11. High-Pass to Low-Pass Transformation Function
0.44410.8400 z1+1.1741 z20.8400 z3+0.4441 z4

H(z)=
3.7938+2.9243 z1+ 5.2123 z2+ 2.5517 z 3+1.5180 z 4
MATLAB Programming Codes
function[HEz,HEp] = elliptichighpass(Wp,Ws,Apass,Astop)
Fc=1100;
rt=Wp/Ws;
a=10^(-0.1.*Apass);
b=10^(-0.1.*Astop);
kn=sqrt(a/b);
49
c=rt^2;
d=1- rt^2;
e=kn^2;
f=1-kn^2;
CEIc=ellipticK(c);
CEId=ellipticK(d);
CEIe=ellipticK(e);
CEIf=ellipticK(f);
n=CEIc*CEIf/(CEIe*CEId);
N=round(n);
E=sqrt(a-1);
F=1/E;
K=kn';
fun=@(x) 1./sqrt((1+x.^2).*(1+(K.^2).*x.^2));
sc=integral(fun,0,F);
Vo=CEIc*sc/(N*CEIe);
m=0:1:(N/2)-1;
%necessary parameters in computing the poles and zeros%
f1=CEIc*1/N;
f2=CEIc*3/N;
[SN1,CN1,DN1]=ellipj(f1,c);
[SN2,CN2,DN2]=ellipj(f2,c);
[SN3,CN3,DN3]=ellipj(Vo,d);
%pole's real%
F1=-1.*(CN1.*DN1.*SN3.*CN3)/(1-((DN1.^2).*SN3^2));
F2=-1.*(CN2.*DN2.*SN3.*CN3)/(1-((DN2.^2).*SN3^2));
%pole's imaginary%
W1=(SN1.*DN3)/(1-((DN1^2).*(SN3^2)));
W2=(SN2.*DN3)/(1-((DN2^2).*(SN3^2)));
50
%location of the zeros (purely imaginary)%

Fz1=0;
Wz1=1/(rt.*SN1);
Fz2=0;
Wz2=1/(rt.*SN2);
%to form quadratics components of the transfer function%
B11=-2.*F1;
B12=-2.*F2 ;
B21=(F1)^2+(W1)^2;
B22=(F2)^2+(W2)^2;
A11=0;
A12=0;
A21=Wz1^2;
A22=Wz2^2;
hA1=[1 A11 A21];
hA2=[1 A12 A22];
hB1=[1 B11 B21];
hB2=[1 B12 B22];
%necessary pi notations to get the normalized transfer function%
proB2=B21*B22;
proA2=A21*A22;
Cons=(10^(0.05*Apass));
He=Cons*proB2/(proA2);
%roots%
zerosS1=roots(hA1);
zerosS2=roots(hA2);
polesS1=roots(hB1);
polesS2=roots(hB2);
%bilinear transformation%
51
M1=1-zerosS1(1);
M2=1-zerosS1(2);
M3=1-zerosS2(1);
M4=1-zerosS2(2);
N1=1-polesS1(1);
N2=1-polesS1(2);
N3=1-polesS2(1);
N4=1-polesS2(2);
m1=((1+zerosS1(1))/(1-zerosS1(1)));
m2=((1+zerosS1(2))/(1-zerosS1(2)));
m3=((1+zerosS2(1))/(1-zerosS2(1)));
m4=((1+zerosS2(2))/(1-zerosS2(2)));
n1=((1+polesS1(1))/(1-polesS1(1)));
n2=((1+polesS1(2))/(1-polesS1(2)));
n3=((1+polesS2(1))/(1-polesS2(1)));
n4=((1+polesS2(2))/(1-polesS2(2)));
MZ=M1*M2*M3*M4;
NP=N1*N2*N3*N4;
mz=poly([m1 m2 m3 m4]);
np=poly([n1 n2 n3 n4]);
Ellipz1=[1 -1*mz(2) mz(3) -1*mz(4) mz(5)];
Ellipp1=[1 -1*np(2) np(3) -1*np(4) np(5)];
%zeros and poles of the elliptic transfer function&
HEz=He*MZ*Ellipz1;
HEp=NP*Ellipp1;
hold on
w=-pi:.01:pi;
freqz(HEz,HEp,w)
zplane(roots(HEz),roots(HEp))
52
%process of inputing the .mp3 file%

x=audioread('C:\Users\angela\Desktop\Flo_Rida_feat._Robin_Thicke_and_Verdine_
White_-_I_Dont_Like_It_I_Love_It[5].mp3');
y=filter(HEz,HEp,x);
soundsc(real(y),44100)
xfft=fft(x);
yfft=fft(y);
plot((1:(length(x)/2)).*8000./(length(x)/2),20*log10(abs(xfft(1:
(floor(length(xfft)/2))))));
hold all;
plot((1:(length(y)/2)).*8000./(length(y)/2),20*log10(abs(yfft(1:
(floor(length(yfft)/2)))))),ylabel('|| Blue - Input Signal || Green - Filtered Signal
||'),title('I Dont Like It I Love It.mp3');
end
Table 3. Parameters and Values of the Designed Filter
Parameters Values
Wp 1000
Ws 1200
Ap -2.6dB
As -32dB
Fc 1100
53
Ratio 0.8333
rt 0.8333
Kn 0.0339
n 3.6564
N 4
E 0.9054
Vo 0.3135
m 0,1
f1 0.5168
f2 1.5504
[SN1,CN1,DN1] [0.4810, 0.8767, 0.9162]
[SN2,CN2,DN2] [0.9570, 0.2902, 0.6034]
[SN3,CN3,DN3] [0.3069, 0.9517, 0.9855]
F1 -0.2548
F2 -0.0530
W1 0.5147
W2 0.9766
Fz1 0
Fz2 0
Wz1 2.4949
Wz2 1.2540
hA1 [1.0000 0 6.2244]
hA2 [1.0000 0 1.5724]
hB1 [1.0000 0.5096 0.3299]
hB2 [1.0000 0.1059 0.9565]
proB2 0.3155
54
proA2 9.7874
He
zerosS1 0.0000 + 2.4949i

0.0000 - 2.4949i
zerosS2 0.0000 + 1.2540i

0.0000 - 1.2540i
polesS1 -0.2548 + 0.5147i

-0.2548 - 0.5147i
polesS2 -0.0530 + 0.9766i

-0.0530 - 0.9766i
mz [1.0000 1.8914 2.6437 1.8914 1.0000]
np [1.0000 -0.7708 1.3739 -0.6726 0.4001]
HEz [0.4441 -0.8400 1.1741 -0.8400 0.4441]
HEp [3.7938 + 0.0000i 2.9243 + 0.0000i 5.2123 +

0.0000i 2.5517 + 0.0000i 1.5180 + 0.0000i]
55
Figure 23. Direct Form I
56
Figure 24. Direct Form II
ABOUT THE PROPONENTS
Joshua G. Albania
Joshua loves to read young adult sci-fi novels and is introvert,
enjoying the company of just a few. He aspires to be an engineer
whom is more inclined to dealing with sounds or acoustics, for he
enjoys much indulging himself in music.
57
Judith B. Cabuhayan
Judith is a cheerful friend who tries to keep everyone around her

motivated. She is currently the Auditor of the PLM Electronics
Engineering Student Society.
Kenway V. Carter
Kenway is an active student. He is presently the Public
Relations Officer of the PLM Engineering Mathematical
Society and a member of the Board of Directors of IECEP-
MSC.
Alpha Mae G. Espinosa

Amae is a gentle person. She loves to make her
friends happy and she has a good sense of humor.
58
Angela Mariz V. Reyes

Angela is a very industrious student who always tries
her best in everything that she does, which results with her
succeeding in a lot of things. She is the Secretary of the PLM
Electronics Engineering Student Society
59

Elliptic High Pass Filter

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PA M A N TA S A N N G L U N G S O D N G M AY N I L A

DESIGN OF A 4TH ORDER INFINITE IMPULSE RESPONSE (IIR)

A Case Study Presented to the Faculty of

In Partial Fulfillment of the Course

Bachelor of Science in Electronics and Communications Engineering

iii. List of Figures

iv. List of Tables

I. The Problem and its Background

B. Background of the Study ......... 7

C. Statement of the Problem ........ 7

E. Scope and Limitations ......... 8

F. Definition of Terms ......... 8

II. Review of Related Literature and Studies

A. Foreign Literature and Studies ....... 10

B. Synthesis of Related Literature to the Case Study ..... 15

B. Experiment Set-up ......... 19

1. System Block and Schematic Diagrams ...... 19

2. Characterization of Input Data ..... 19

3. Data Processing and Filter Implementation ..... 19

a. Mathematical Processing ....... 19

IV. Data and Computation

V. Summary of Findings, Conclusions and Recommendations

A. Summary of Findings ......... 30

About The Proponents ........ 42

Figure 1 An efficient realization of IIR

Figure 2 A 3rd Order High-Pass Elliptical Filter

Figure 3 Designed 4th Order High-Pass Elliptical Filter

Figure 4 Magnitude and Phase Response of Elliptic High-Pass Filter

Figure 5 Pole-Zero Plot of Elliptic High-Pass Filter

Figure 6 Filtered Star Wars Theme

Figure 7 Filtered Silento Watch Me

Figure 8 Filtered Imperial March

Figure 9 Filtered I Dont Like It I Love It

Figure 10 Filtered Invisible

Figure 11 Magnitude and Phase Response of Elliptic Low-Pass Filter

Figure 12 Pole-Zero Plot of Elliptic Low-Pass Filter

Figure 13 Original I dont like it, I love it

Figure 14 Filtered I dont like it, I love it

Figure 15 Original Imperial March

Figure 16 Filtered Imperial March

Figure 17 Original Invisible by Hunter Hayes

Figure 18 Filtered Invisible by Hunter Hayes

Figure 19 Original Silento Watch Me

Figure 20 Filtered Silento Watch Me

Figure 21 Original Star Wars Theme

Figure 22 Filtered Star Wars Theme

Figure 23 Direct Form I

Figure 24 Direct Form II

Table 1 Frequency Specification

Table 2 Parameters and Values of the Designed Filter

The Problem and its Background

B. Background of the Study

C. Statement of the Problem

1. How can a working elliptic filter with high-pass characteristics be achieved?

Using Matrix Laboratory (MATLAB), the researchers intended to design and

E. Scope and Limitations

Attenuation the reduction in the level of power, current, or voltage.

Review of Related Literature and Studies

A. Foreign Literature and Studies

Infinite Impulse Response (IIR) FILTER

Figure 1. An efficient realization of IIR [3]

DIFFERENCE BETWEEN FIR & IIR