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Digital Filter Implementation Using MATLAB
http://asic-soc.blogspot.com/2009/01/digital-filter-implementation-using.html
MATLAB SOFTWARE
ABOUT MATLAB
The name MATLAB stands for matrix laboratory. MATLAB is a high-performance technical
computing. It integrates computation, visualization, and programming in an easy-to-use environment
where problems and solutions are expressed in familiar mathematical notation. Typical uses include:
Math and computation
Algorithm development
Modeling, simulation, and prototyping
Data analysis, exploration, and visualization
Scientific and engineering graphics
Application development, including graphical user interface building

MATLAB is an interactive system whose basic data element is an array that does not require
dimensioning. This allows you to solve many technical computing problems, especially those with
matrix and vector formulations, in a fraction of the time it would take to write a program in a scalar
non-interactive language such as C or Fortran.
MATLAB features a family of application-specific solutions called toolboxes. Very important to most
users of MATLAB, toolboxes allow you to learn and apply specialized technology. Toolboxes are
comprehensive collections of MATLAB functions (M-files) that extend the MATLAB environment to
solve particular classes of problems. Areas in which toolboxes are available include signal processing,
control systems, neural networks, fuzzy logic, wavelets, simulation, and many others.
THE MATLAB SYSTEM
The MATLAB system consists of five main parts:
Development Environment: This is the set of tools and facilities that help you use MATLAB
functions and files. Many of these tools are graphical user interfaces. It includes the MATLAB desktop
and Command Window, a command history, and browsers for viewing help, the workspace, files, and
the search path.
The MATLAB Mathematical Function Library: This is a vast collection of computational algorithms
ranging from elementary functions like sum, sine, cosine, and complex arithmetic, to more
sophisticated functions like matrix inverse, matrix eigenvalues, Bessel functions, and fast Fourier
transforms.
The MATLAB Language: This is a high-level matrix/array language with control flow statements,
functions, data structures, input/output, and object-oriented programming features. It allows both
"programming in the small" to rapidly create quickly and dirty throwaway programs, and
"programming in the large" to create complete large and complex application programs.
Handle Graphics: This is the MATLAB graphics system. It includes high-level commands for two-
dimensional and three-dimensional data visualization, image processing, animation, and presentation
graphics. It also includes low-level commands that allow you to fully customize the appearance of
graphics as well as to build complete graphical user interfaces on your MATLAB applications.
The MATLAB Application Program Interface (API)
This is a library that allows you to write C and Fortran programs that interact with MATLAB. It includes
facilities for calling routines from MATLAB (dynamic linking), calling MATLAB as a computational
engine, and for reading and writing MAT-files.
MATLAB WORKSPACE
The MATLAB workspace consists of the set of variables (named arrays) built up during a MATLAB
session and stored in memory. You add variables to the workspace by using functions, running M-
files, and loading saved workspaces. For example, if you type
t = 0:pi/4:2*pi;
y = sin(t);
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The workspace includes two variables, y and t, each having nine values.
ABOUT SIMULINK
Simulink is a software package for modeling, simulating, and analyzing dynamical systems. It
supports linear and nonlinear systems, modeled in continuous time, sampled time, or a hybrid of the
two. Systems can also be multirate, i.e., have different parts that are sampled or updated at different
rates.
For modeling, Simulink provides a Graphical User Interface (GUI) for building models as block
diagrams, using click-and-drag mouse operations. With this interface, you can draw the models just
as you would with pencil and paper. This is a far cry from previous simulation packages that require
you to formulate differential equations and difference equations in a language or program. Simulink
includes a comprehensive block library of sinks, urges, linear and nonlinear components, and
connectors. You can also customize and create your own blocks.
Models are hierarchical, so you can build models using both top-down and bottom-up approaches.
You can view the system at a high level, then double-click on blocks to go down through the levels to
see increasing levels of model detail. This approach provides insight into how a model is organized
and how its parts interact. After you define a model, you can simulate it, using a choice of integration
methods, either from the Simulink menus or by entering commands in MATLAB's command window.
The menus are particularly convenient for interactive work, while the command-line approach is
very useful for running a batch of simulations (for example, if you are doing Monte Carlo simulations
or want to sweep a parameter across a range of values). Using scopes and other display blocks, you
can see the simulation results while the simulation is running. In addition, you can change parameters
and immediately see what happens. The simulation results can be put in the MATLAB workspace for
post processing and visualization.
Model analysis tools include linearization and trimming tools, which can be accessed from the
MATLAB command line, plus the many tools in MATLAB and its application toolboxes. And because
MATLAB and Simulink are integrated, you can simulate, analyze, and revise your models in either
environment at any point
DESIGN AND STUDY OF FILTERS
DESIGN STEPS FOLLOWED
MATLAB offers varieties of toolboxes using that we can easily design the required digital filter and
can observe its phase and magnitude characteristics; construct realization structure of the designed
filter; analyze working of the filter.
To design a filter and analyze it we followed below mentioned steps:
1. Using Filter Designer And Analyzer window we designed our required filter. To open this
window type fdatool in Command Window and press Enter. The obtained filter coefficients (i.e.
numerator and denominator coefficients) are noted.
2. Using Filter Realization Wizard window we have constructed the filter structure by inputting
filter coefficients and selecting the appropriate form. To open this window type dspfwiz on Command
Window and press enter key. Using Launch Pad also we can open this window. Select DSP Blockset
from the Launch Pad and click on the +ve mark corresponding to it. A drop down list opens up. From
this list select Filter Realization Wizard and double click on it. This opens that window. The constructed
structure appears as a subsystem model. A double click on the subsystem block opens the filter
structure in Simulink window, which can be simulated.
3. Using Simulink block library function generator, oscilloscope and MUX are connected to the filter
structure. Using Simulink debugger the structure is simulated and the results are observed on the
oscilloscope.
IIR FILTER DESIGN ISSUES
We restrict our discussion to the design and analysis of IIR filters only, even though FIR filters
provide linear phase throughout the frequency range. In application of our project phase angle
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variation is of no importance; if phase angle variation is also important then automatically discussion
orients towards FIR filters.
Before directly going into the analysis of IIR filters let us have a glance over Shannons sampling
theorem.
A signal containing maximum frequency f1Hz may be completely represented by regularly spaced
samples, provided the sampling rate fs is at least 2f1 sample per second.
i.e. fs=2f1 Nyquist sampling rate
If signal is sampled at less than 2f1 rate, aliasing error occurs. Signal is then represented with
distortion, which depends on the degree of aliasing. To avoid such distortions use antialiasing filter, a
low pass filter with cutoff frequency at f1 (or fs/2).
Because of above reason designing any digital filter at higher frequency side becomes difficult owing
to higher sampling rate and its generation. Hence we need to restrict ourselves to lower frequency
side. Further discussion on these issues will be elaborated in respective filter design studies to forth
come.
IIR Low pass And High pass Filter Design Issues
The ideal response of low pass and high pass filters is as shown in Figure 1
Practically such sharp roll off is not achievable. Using MATLAB we can easily design these filters
and simulate it. Even at higher frequencies (like 40 KHz) simulink work satisfactorily. The filter
nicely exhibits passband and stopband action.
In most of the applications lowpass/highpass filters are used at lower frequencies with increased
order, so that sharp roll-off is achieved.

Figure 1
IIR Bandpass Filter Design and Its Implementation
Ideally bandpass filter should have a perfect passband as shown in the Figure 2
But practically such a sharp cutoff at passband edges is not possible. Therefore practically we aim
towards a response, which has as much sharper role-off as possible, so that channel selection is
performed noiselessly. Chebyshev and elliptic filters provide very good response in this regard. Hence
we select Chebyshev type of filter for our study and analysis as it provides satisfactory bandpass
characteristics.

Figure 2
Chebyshev Type-II Filter Design
Filter parameters are as shown below
Order = 2; Sampling frequency Fs = 100 KHz; Fstop1 = 39 KHz; Fstop2 = 41 KHz; Astop= 60 dB
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Input the above parameters to the respective places of Filter Design & Analysis Tool. After
completing this click on Design Filter button. This creates the required filter .To observe the
response (step or frequency or impulse) click on the respective icon of the toolbar. Toolbar also has
icons to observe pole-zero plot, group delay, filter coefficients etc. Thus the obtained filter coefficients
are as follows:
Numerator (b) Denominator (a)
0.000062910740701 1.000000000000000
0.000000000000000 1.621131129004091
-0.000062910740701 0.999874178518599
For the above parameters magnitude response can be observed by clicking on corresponding icon
of the toolbar. The corresponding frequency response is as shown below in Figure 3.

Figure 3
For the same above filter parameters MATLAB program is written in a M-file to create chebyshev
type-II bandpass filter.
The program is as follows.
File name:cheby.m
clc;
clear all;
close all;

As=60; %stopband attenuation
wp1=3.8e+004; %lower cutoff frequency
wp2=4.2e+004; %higher cutoff frequency
n=1; %order
fs=100000; %sampling frequency
wn=[wp1 wp2]/(fs/2); %Nyquist frequency
[b,a] = cheby2(n,As,wn); %compute filter coefficients
[h,f]=freqz(b,a,fs/2); %compute frequency response
mag=20*log10(abs(h)); %get magnitude response
subplot (2,1,1);
plot(f*(fs/2)/pi, mag);grid; %plot response with grid lines
ang=angle(h); %get phase response
subplot(2,1,2);
plot(f*(fs/2)/pi,ang);grid; %plot response with grid lines

Run this program either by pressing F5 key on keyboard or select Run from debug menu. The
obtained magnitude and phase response is as shown below in Figure 4
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Figure 4
For both of the above filter designs, the corresponding filter realization structure of Direct Form 2
is as shown in Figure 5. This can be created using Filter Realization Wizard by inputting filter
coefficients.
From the frequency response curves (Figure 4) of the above filter we can analyze the filter.
Thus the Chebyshev type-2 filter designed with n=1(implies order 2n=2) and pass band =2 KHz
(implies 1 KHz from 40 KHz) has very sharp (like a pulse) passband and also not exactly centered
at 40 KHz.Owing to this shift of passband either towards the left or right side of frequency domain,
the center frequency may get attenuated or noise signals may creep through the filter resulting in a
distorted output. Another possibility is suppose by some means frequency of input signal itself varies
from its central value then filter may pass unwanted signals and it may attenuate the original signal
itself creating a noisy output. This may due to the inconsistency of function generator. We cant say
even crystal oscillator can produce exactly 40 KHz signal. It may vary a little (100 Hz) due to
variation of temperature, pressure, applied voltage etc.
Figure 5
Therefore we need a filter, which has a passband, so that even though center frequency varies
from its value, the signal is reproduced faithfully at the output and all other unwanted signal
frequencies are completely attenuated.
We can meet the requirements by changing the filter parameters like order, pass band, edge
frequency or stop band attenuation. But it is found that keeping order n=1 even though we increase
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pass band range and decrease stop band attenuation, dont yield satisfactory result instead it just
attenuated pass band itself.
Therefore to improve the filter performance the only left option is to increase the filter order and
this is what done in most of the practical cases. We adjust the pass band frequency so that around
500 to 800 Hz pass band with minimum attenuation is obtained.
Using Filter Design & Analysis Tool we can do this and by noting down the filter coefficients
we can realize filter structure.
CHEBYSHEV TYPE-II FILTER DESIGN WITH INCREASED ORDER
Filter parameter: n = 4 (2n = 4=> n =2)
Fs = 100000Hz.
Fstop1 = 38000Hz.
Fstop2 = 41800Hz.
Astop = 40dB.
Let the file name given to this filter be filter40.fda
Figure 6
The obtained magnitude response is as shown in Figure 6 and the corresponding filter coefficients
are as follows,
Numerator (b) Denominator (a)
b(0)=0.010045097978992. a(0)=1.000000000000000.
b(1)=0.031677960303578. a(1)=3.205603560877597.
b(2)=0.044659670697279. a(2)=4.521600439016961.
b(3)=0.031677960303578. a(3)=3.129988499837994.
b(4)=0.010045097978992. a(4)=0.953386226509250.
EFFECT OF TRUNCATION OF COEFFICIENT ON FILTER RESPONSE
Digital signal processing algorithms are realized either with special purpose digital hardware or as
programs for a general-purpose digital computer. In both cases the numbers and coefficients are
stored in finite-length registers. Therefore, coefficients and numbers must be quantized by truncations
or rounding before they can be stored.
The following errors arise due to quantization of numbers,
1. Input quantisation error.
2. Product quantisation error.
3. Coeficiente quantisation error.

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4. The conversion of a continuous time input signal into digital value produces an error, which is
known as input quantization error arises due to the representation of the input signal by a fixed
number of digits in the A/D conversion process.
5. Product quantization errors arise at the output of a multiplier. Multiplication of a and b bit with
ab bit coefficient results a product having 2b bits. Since ab bit register is used, the multiplier
output must be rounded or truncated to b bits, which produces an error.
6. The filter coefficients are computed to infinite precision in theory. If they are quantized, the
frequency response of the resulting filter may differ from the desired response and sometimes the
filter may fail to meet the desired specifications. If the poles of the desired filter are close to the
unit circle, then those of the filter with quantized coefficients may lie just outside the unit circle,
leading to the instability.
The other errors arising from quantization are round off noise and limit cycle oscillations.
It can be understood from the above points that quantization error due to A/D conversion process
is difficult to minimize below certain limits in any sophisticated processor. Using higher sampling rate
we can minimize this error.
Using processors with higher word length registers can minimize product quantisation errors. As
length of the operating registers become more and more error becomes less and less. If floating point
arithmetic is supported in processor then this error can be eliminated to a very large extent.
Now let us analyze the filter40.fda named filter for different truncated values.
In this file, the coefficients obtained are 64-bit length (i.e. 16 decimal number excluding decimal
point). Notice that these are the coefficients, which are obtained by designing the filter in Filter
design & Analysis tool.
To find out what is the effect of truncating the coefficient data, we use M-file program cheby.m.
Here we directly feed the filter coefficients to the freqz command as shown below:

clc;
clear all;
close all;
b=[b(0) b(1) b(2) b(3) b(4)]; %fill numerator coefficients
a=[a(0) a(1) a(2) a(3) a(4)]; %fill denominator coefficients
[h, f]=freqz (b,a,fs/2);
If you dont put semicolon at the end of the line where numerator and denominator coefficients are
written and you run the program, you can find that at the command window, these coefficients are
rounded up to 4
th
decimal point. The software itself automatically does the truncation of data. The
resultant frequency response by this truncated data is same as that of obtained with non-truncated
data with hardly noticeable differences.
The rounding up of numerator and denominator to third decimal point causes below response:

Numerator (b) denominator (a)
0.010 1.000
0.032 3.206
0.045 4.522
0.032 3.130
0.010 0.953

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Figure 7
The rounding up of numerator and denominator to second decimal point causes heavy variation in
pass band attenuation as shown in Figure 8.
Numerator (b) Denominator (a)
1. 1.00
2. 3.21
3. 4.52
4. 3.13
1. 0.95

Figure 8
Rounding up of filter coefficient to the first decimal point abruptly changes frequency characteristics
of the filter.
From the above discussion it follows that truncating the filter coefficients to 4
th
decimal point yields
good acceptable frequency characteristics and hence can be implemented in hardware.
The filter structure for the above filter is as shown below in Figure 9.

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Figure 9
SIMULATING THE FILTER STRUCTURE
To simulate the above filter structure, it has to be modified a little bit.
First open Simulink Library Browser. Select and drag the Sine Wave block from the source
library to the window where the structure is created. Similarly Scope is dragged from the Sinks
library and the Mux is dragged from the Signals & Systems library. Substitute input by sine Wave
block, output by Mux and Scope block. By double clicking on each block we can get the properties
of each block. For example: double click on sine wave block. This will open a window wherein we
can write frequency of the signal, signal voltage, sampling time etc. In the same way mux and scope
can be configured.
After connecting all these blocks click on start icon on the toolbar to start simulation. Options are
there to pause the simulation, stop the simulation etc. We can also see the simulation status on
simulink debugger window. Double click on the scope block. This opens the oscilloscope where we
can observe the input and output waveforms.
The study and analysis of simulation of the above filter structure as well as many other filter
structures (like Cheby type-1, 2 Butterworth lowpass/highpass) are carried out successfully.
One thing that is particularly noticeable is that for higher frequencies simulation of filter structure
takes more time compared to simulation of filter structure at lower frequencies. Effect of truncation
of coeffients can also be observed on the Scope and also works as the study has been done in earlier
sections.
References
1. Digital Signal Processing By Sanjit K.Mitra
2. Digital Signal Processing By P.Ramesh Babu
3. Digital Filters By T.J.Terrel And E.T.Powner
4. BASIC Digital Signal Processing By Gordon B. Lockart And Barry M.G.Cheetham
5. Digital Signal Processing By Alan V.Oppenheim And Ronald W.Schafer
6. DSP Microprocessors: Advances and Automotive Applications By Subra Ganeshan And Dr.Gopal
Arvamudhan
7. www.mathworks.com