A report on

Digital Filter Designing

Conducted by
Toalha Mohammad Tosrif Roll: 0302105
MD. Zulfiker Jahirye Roll: 0302011

Supervised by
Ashraf-uz-zaman
Asst. prof. EEE, CUET

Chittagong University of Engineering & Technology

Department of Electrical & Electronic Engineering
Title:
Digital Filter Designing.
Objective:
Our objective is to study different types of Digital Filters and simulate them on
MATLAB.
Literature study:
Digital Signal Processing (DSP) is based on the fact that an analog signal can be digitized
and input to a general purpose digital computer or special purpose digital processor. To
get desired frequency from these digital signals we often use Digital filters.
Digital filters are usually classified by the duration of their impulse response, which can
be either finite or infinite. The method for designing and implementing these two filter
classes differ considerably.
Finite Impulse Response (FIR) filters
Infinite Impulse Response (IIR) filters

Finite impulse response (FIR) filters are digital filters whose response to a unit impulse
(unit sample function) is finite in duration. On the other hand Infinite Impulse Response
(IIR) filters whose response to a unit impulse (unit sample function) is infinite in
duration. FIR and IIR both have advantages and disadvantages and neither is best in all
situations.
FIR Filters:
The main advantage of FIR filter is, it can be easily designed to have constant phase
delay and/or constant group delay. And one of the disadvantages is, an FIR filters
impulse response duration, although finite, may have to be very long to obtain sharp
cutoff characteristics. A digital filter`s impulse response h[n] is related to the frequency
response H(ej) via the DTFT:

------------- 1
For an FIR filter, h[n] is nonzero only for 0 n N. therefore; the limits of the
summation can be changed to yield.

------------- 2
where h[n] is the system`s impulse response. This equation can be calculated directly at
any desired value of . From this equation we can find that the output of FIR Filter is a
linear combination of the present input and the N previous inputs.
There are several methods of FIR filter designing. The Fourier series method is one of
them. This method is based on the fact that the frequency response of a digital filter is
periodic and is therefore representable as a Fourier series. The method can be described
by some steps given bellow.
Step 1. First we have to specify a desired frequency response Hd ().
Step 2. Then have to specify the desired number of filter taps N.
 Step 3. Then have to compute the filter coefficients h[n] for n = 0, 1, 2 N-1
using

--------------- 3
Where m= n-(N-1)/2.
Step 4. And at last using the equation 1 and 2 we can compute the actual
frequency response of the resulting filter. If the performance is not adequate then we can
change N or Hd() and go back to step 3.
There are several types of responses such as, Lowpass, Highpass, Bandpass, Bandstop.
We can select our desired one from these as the frequency response Hd.
Magnitude Response of a Lowpass filter
1.4

1.2

0.8
Magnitude

0.6

0.4

0.2

0
0 5 10 15 20
Frequency (kHz)

One of the other methods of designing of FIR filter is Frequency Sampling Method. In
Fourier Method FIR filters was specified in the continuous-frequency domain and the
discrete-time impulse response coefficient were obtained via the Fourier series. We can
modify this procedure so that the desired frequency response is specified in the discrete
frequency domain and then the inverse Discrete Fourier Transform (DFT) to obtain the
corresponding discrete impulse.
IIR Filters:
The general form for an Infinite Impulse Response (IIR) filter`s output y[k] at time k is
given by

--------------- 4
This equation indicates that the filter`s output is a linear combination of the present input,
the M previous inputs, and the N previous outputs. The corresponding system function is
given by
A direct realization of Eq. 4 is shown in fig-2 using the signal flow graph notation. The
structure shown is known as the Direct form I realization or direct form I structure for IIR
system. Examination of the figure reveals that the system can be viewed as two systems
in cascade, the first system using x [k-M] through x[k] to generate an intermediate signal
that we can call w[k] and the second system using w[k] and y[k-N] through y[k-1] to
generate y[k].

Fig-2: Signal flow graph of direct form I realization for IIR system
Magnitude Response rsponse of Butterworth filtersof order N=4
1.2

0.8
Magnitude

0.6

0.4

0.2

0
0 5 10 15 20
Frequency (kHz)

There are various approaches for designing IIR filters. Such as

Impulse Invariance
Step Invariance
Bilinear Transformation

The most popular technique for the design of IIR digital filters is the Bilinear
Transformation Method. The Bilinear Transformation converts the transfer function for
an analog filter into the system function for a digital filter by making the substitution.
If the analog prototype filter is stable, the bilinear transformation will result in a stable
digital filter.
The steps for bilinear transformation are given bellow.
Step 1: First we have to obtain the transfer function Ha(s) for the desired analog
prototype filter.
Step 2: In the transfer function obtained in step 1, we have to make the
substitution

Where T is the sampling interval of the digital filter. Call the resulting digital system
function H (z).
Step 3: The analog prototype filter`s transfer function Ha(s) will in general, be a
ratio of polynomials in s. Therefore, the system function H(z) obtained in step 2 will, in
general, contain various power of ratio (1-z-1)/(1+z-1) in both the numerator and the
denominator. Multiply both the numerator and denominator by the highest power of 1+z -
1
, and collect terms to obtain H(z) as a ratio of polynomials in z -1 of the form

Step 4: Use the ak and bk obtained in step 3 to realize the filter

Research plan & Methodology:

Our research plan has some steps,
Study: At first we will study on Digital filters. As it is the main part of
DSP so we first start with DSP. We have collected some reference books and studying
them. There are various kinds of digital filters, we are studying them.
Simulation: Then comes the step of simulation. In this step we will simulate
the filters we`ve studied in MATLAB or C. To do this we need to know the programs. We
are studying these programs parallely with the other studies.
Implementation: In this step we`ll implement some of these filters in the DSP kit
that we have In CUET. We need to study the DSP kit for the practical implementation.

Desired result:
At the end of this our study and other works we expect some results
A clear concept on Digital filters.
Computer simulation of various filters.
Implementation of some filters (at least 2) on DSP kit.

Conclusion:
Digital Filter plays vary vital role in Digital Signal Processing. So we need to know their
various responses as far as farther study on DSP in concerned. As a first step this research
may help us. The MATLAB simulation would give us the actual or theoretical response
on the other hand the DSP kit implementation would give us the practical response and
we would compare them both.
We`ve got proper help and book references from our teacher Md. Ashraf-uz-zaman (Asst.
Prof.) and hope that through out our thesis we will get proper help from him and all our
teachers.
References:
Digital Signal Processing
by John G. Proakis
Digital Filter Designer`s
Handbook by C. Britton Rorabaugh