This document is meant as a tutorial for the firsttime user. In this
tutorial we walk the user through the different features of the applet.
In doing this, we try to reveal all of the features of the
applet. There are a small number of undesirable
"hidden" features (a.k.a. bugs) remaining. While we are
working hard to correct these problems, we release this version of the
software in hopes that it will be beneficial to its users.
Now, on with the show...
Two Options are available:
A
full
description of all parts of the applet or
Step by step instructions for each algorithm showing how to
use each algorithm:
Butterworth,
Chebyshev,
Bessel,
Kaiser,
Parametric Equalizer,
AllPass,
User defined filter.

Starting up
The first thing the user needs to do is to
start the applet.
You could also download the
source code
from our website and compile it yourself. At this point in the
tutorial we will assume that you can access the applet and that
you have started it.

Setting parameters
Once the applet has loaded, you will see the following screen:
The applet can be split into three rows. The first row holds
the Menu bar.
The second row has two columns. The first column
is a Control panel where the parameters to design the filter or
parameters to change the time scale of the plots are
entered.
The second column in the second row is a Display panel
that displays the filter coefficients after the filter is
computed.
The third row consists of magnitude and phase plots of the
filter computed.
The top section of the applet is the Menu bar. Using the
menu bar, the user can choose a design algorithm, change the
time scale in the plots, compute a filter, or reset the applet to
its default settings.
To begin designing a filter, click on the "Algorithms" menu and
choose an option. Once a filter design had been chosen, the
parameters for the filter will be displayed on the Control
Panel. Enter the parameters as shown in the Control panel. Then
to compute the filter, click on the Compute option under the
"Go" menu.
Other features that can be used in this applet are a tool to
modify filter coefficients and the ability to zoom both the plots.
Filter Coefficients can be modified by choosing the Modify
option under "Parameters" menu. This will display a panel in the
Control panel that will look like as shown below.
Enter the coefficient number and the value for that
coefficient. Numerator or denominator coefficients can be
changed by proper choice. The coefficients can also be changed in
pairs using the Change Pairs option. After changing the
coefficients, click on the Modify button in the bottom of the
panel to update the value of the coefficients.
The time scales on the magnitude response plot can
be changed choosing the Magnitude option under the "Parameters"
menu. After this is done, the control panel will look like
Enter the parameters as required on the textfields and click on the
"Update" button. There will be corresponding change in the Magnitude
plot.
The phase plot also can be changed the same way as described for the
magnitude plot except that the user needs to choose "Phase" option
under "Parameters" menu.
The applet has an extensive error messaging system. This will
notify you if there is an error in the computation of a
particular filter or if you have entered an invalid value for
parameters.
There error window looks like the following. By selecting
OK, you will close the window and be able to correct
the problem and try again.

Using the Butterworth filter algorithm
When using the Butterworth filter algorithm, the main screen will
look like as shown below
The control panel displays parameters like sample frequency (Hz), the
lower passband frequency (Hz), upper passband frequency
(Hz), transition bandwidth (Hz), stopband attenuation (dB),
and the filter order. If the filter order is set to 0,
the filter order used will be the optimal order as computed
by the algorithm. The user can also set his own filter order by
entering a nonnegative, nonzero value in the filter order textfield.
Using the Chebyshev filter algorithm
When using the Chebyshev filter algorithm, the main screen will
look like as shown below
The control panel displays parameters like sample frequency (Hz), the
lower passband frequency (Hz), upper passband frequency
(Hz), transition bandwidth (Hz), stopband attenuation (dB),
and the filter order. If the filter order is set to 0,
the filter order used will be the optimal order as computed
by the algorithm. The user can also set his own filter order by
entering a nonnegative, nonzero value in the filter order
textfield. Note the ripples in the passband that did not exist
in the Butterworth filter. Also, for the same specifications as
that of Butterworth filter, Chebyshev filter gives a lower
filter order.
Using the Bessel filter algorithm
When using the Bessel filter algorithm, the main screen will
look like as shown below
The control panel displays parameters like sample frequency (Hz), the
lower passband frequency (Hz), upper passband frequency
(Hz), transition bandwidth (Hz), stopband attenuation (dB),
and the filter order. If the filter order is set to 0,
the filter order used will be the optimal order as computed
by the algorithm. The user can also set his own filter order by
entering a nonnegative, nonzero value in the filter order
textfield. One important thing to note in the Bessel filter
computation is that the filter order cannot exceed twenty five. If the
user enters a filter order greater than twenty five, an error will be
shown.
Using the Kaiser filter algorithm
When using the Kaiser filter algorithm, the main screen will
look like the following.
The user is able to select the sample frequency (Hz), the
lower critical frequency (Hz), the upper critical frequency
(Hz), the attenuation (dB), the transitional bandwidth (Hz),
and the filter order. If the filter order is set to 0,
the filter order used will be the optimal order as computed
by the algorithm. Also, the filter odd needs to be odd else an
error message will be shown.

Using a parametric equalizer
When using the Parametric equalizer, the main screen will
look like the following.
The user can set the Sample Frequency(Hz), Center
Frequency(Hz), Gain at Center Frequency(dB) and
Bandwidth(Hz). The parametric equalizer is a second order
filter that can boost or cut a particular frequency by the
given gain. If the gain value is positive, then that particular
center frequency is boosted. If the gain value is negative, then
the center frequency is attenuated by the value of gain. The
sharpness of the boost or cut is controlled by the bandwidth.
If the gain is positive, then the parametric equalizer is
a Resonator whose gain can be varied. If the gain is negative,
then the parametric equalizer is a Notch filter.

Using a AllPass filter
When using an allpass filter, the main screen will
look as shown below.
The user can set the Sample Frequency(Hz), Center
Frequency(Hz) and Bandwidth(Hz). An allpass filter is one
which has constant magnitude response for all frequencies but
the phase response varies based on the Sample Frequency(Hz),
Center Frequency(Hz) and Bandwidth(Hz). While an allpass filter
can have any number of poles and zeros, the allpass filter
designed in this applet is a second order filter.

Defining your own filter coefficients
You can also define a filter by entering any coefficients
you desire and compute the magnitude and phase. The main
screen for this looks like the following.
The screen above shows a filter having three numerator
coefficients and two denominator coefficients.
When entering filter coefficients, you can determine the
coefficient number and value. You can also add coefficients
to the numerator, denominator, or both simultaneously. It is
also possible to add the coefficients in pairs. After entering
each coefficient click on the 'Add' button to update the
coefficient's list. Once you have entered all the desired
coefficients, select the Compute option in the 'Go' menu.
This is the end of the tutorial. Begin the learning experience.
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