University of Kentucky

EE 422G - Signals and Systems Laboratory

Lab 2 FIR Filters

Objectives:
 Use filter design and analysis tools to create FIR filters based on general filter
specification.
 Create a simulation with Simulink.

1. Background

Digital filters are used in a wide variety of signal processing applications, such as
spectrum analysis, digital image processing, and pattern recognition. Digital filters
eliminate a number of problems associated with their classical analog counterparts and
thus are often used in place of analog filters. Digital filters belong to the class of discrete-
time LTI (linear time invariant) systems, which are characterized by the properties of
causality, recursibility, and stability. They can be characterized in the time domain by
their unit-impulse response and in the transform domain by their transfer function. A
unit-impulse response sequence of a causal LTI system could be of either finite or infinite
duration and this property determines their classification into either finite impulse
response (FIR) or infinite impulse response (IIR) system. To illustrate this, we consider
the most general case of a discrete time LTI system with the input sequence denoted by
x(kT) and the resulting output sequence y(kT) given by:

If at least one denominator coefficient a v is not zero, then system is recursive; its
impulse response is of infinite duration (IIR system). If all denominator coefficients are
zero, the corresponding system is non-recursive (FIR system); its impulse response is of
finite duration and the transfer function H(z) is a simple polynomial. Commonly, b is
called the  th forward filter coefficient and a v the v th feedback or reverse filter
coefficient.

The design of digital filters involves the following basic steps:

 Determine the desired response. The desired response can be specified in the
frequency domain in terms of the desired magnitude response and/or the
desired phase response. In can also be specified in term of a desired impulse
response.
 Select a class of filters (for example, linear-phase FIR filters or IIR filters) to
approximate the desired response.
  Select the best member in the filter class (i.e. determination of order and
coefficient values).
 Implement the best filter using a general purpose computer, a DSP, or a
custom hardware chip.
 Analyze the filter performance to determine whether the filter satisfies all the
given criteria.

In this lab you will design FIR filter using 2 methods – windowing and the Parks-
McClellan algorithm.

2. Pre-Lab

1. For the window functions listed below, write a script to plot each window
function on the same graph (use different line styles for each window function).
Create another graph and plot its DFT magnitude on a linear scale, and finally
create another graph and plot its DFT magnitude on a dB scale.
a. boxcar

b. triangular

c. hamming

Comment on how the general window shape (steepness of taper) affects the
spectral magnitude (impact on width of main lode and height of sidelobes)

2. Repeat number 1 for a Kaiser window of length 128 and with  values of 4, 6, and
9.
3. For a sampling rate of 48kHz design an order-40 low-pass filter having cut-off
frequency 10kHz by windowing method. In your design, use Hamming window
as the windowing function (see help on fir1). Use the freqz command to plot the
filter’s magnitude response (in dB) and use the filter command to plot the impulse
response.

4. For a sampling rate of 48kHz, design an order-40 low-pass filter having transition
band between 9kHz and 11kHz using the Parks-McClellan FIR filter design
algorithm (see help on firpm). Use the freqz command to plot the filter’s
magnitude response (in dB) and use the filter command to plot the impulse
response.

3. In-Lab exercise

(Assume sampling frequency of 44.1kHz, unless otherwise specified)

1. Design a 127th order linear-phase FIR low-pass filter with a cut-off at 6kHz using
the windowing method (fir1). Design a filter using each of the following
windows:

a. rectangular (boxcar)
b. triangular (triang)
c. Hamming (hamming)

On a single graph plot the all the impulse responses, on another graph plot all the
magnitude responses, and then in individual graphs plot the zero-pole locations of
the 3 filters. In the discussion section compare impact of the window on the filter
characteristics. Refer to the window plots of the prelab and make a general
statement about window characteristics and their impact on the filter
characteristics.

2. Repeat Exercise 1 for a 15th order filter. Comment on the impact of filter order.

3. Repeat Exercise 1 for a length-32 linear-phase FIR low-pass filter using a Kaiser
window with  = 4, 6, and 9 and pass-band cut off of 6kHz. Plot the impulse
response, magnitude response, and zero-pole locations. Compare the
characteristics of the magnitude response with each other and with the filters from
the first exercise. Discuss how the trade-off between transition bandwidth and
ripple varies with  ?

4. Design an optimal 23-point low-pass FIR filter using the Parks-McClellan

algorithm (firpm). Use a pass-band cutoff of 5.5kHz and a stop-band cutoff of
6.5kHz. Plot the impulse response, magnitude response, and zero-pole locations.
Compare the characteristics of the magnitude response to the other designs and
comment on differences.

5. Generate a Square wave signal of 400 Hz and determine an approximate

bandwidth for this signal (frequency range containing significant energy) by
plotting its spectrum by using the FFT command. Use the Parks-McClellan
algorithm (firpm) to design a 51-order FIR low-pass filter, such that the
frequencies starting with the 3rd (i.e. at 1200Hz) harmonic are suppressed by 40
dB. Plot the magnitude response of the filter and also the filtered output. Did it
achieve your specification? Use soundsc to play the sound before and after
filtering. Describe the differences that you hear.

6. Repeat Exercise 5 with an order of 201.

7. Generate the signal x using the following Matlab code:

Fs=8e3; %Specify Sampling Frequency

Ts=1/Fs; %Sampling period.
Ns=512; %Nr of time samples to be plotted.
 t=[0:Ts:Ts*(Ns-1)]; %Make time array that contains Ns elements
%t = [0, Ts, 2Ts, 3Ts,..., (Ns-1)Ts]
f1=500;
f2=1800;
f3=2000;
f4=3200;

x1=sin(2*pi*f1*t); %create sampled sinusoids at different

frequencies
x2=sin(2*pi*f2*t);
x3=sin(2*pi*f3*t);
x4=sin(2*pi*f4*t);

x=x1+x2+x3+x4; %Calculate samples for a 4-tone input signal

With the FIR windowing method, design a 16 order Bandstop filter with stop-band
between 1500 Hz and 2500 Hz using hamming window. Plot the magnitude of the
frequency response of the filter. Is this what you expected? Apply the filter to the input
signal (x), and plot the spectrum magnitude of the input and output signal on the same
figure. Is this output as expected? For spectrum plot the follow code is helpful:

Nfft = 2*length(x);
xfftmag=(abs(fft(x,Nfft))); %Compute spectrum of input signal.
xfftmagh=xfftmag(1:length(xfftmag)/2);
%Plot only the first half of FFT, since second half is mirror image
%the first half represents the positive frequencies from
%0 to Fs/2, the Nyquist sampling limit.
f=[1:length(xfftmagh)]*Fs/Ns; %Make freq array that varies from
%0 Hz to Fs/2 Hz.
plot(f,xfftmagh,’:k’); %Plot frequency spectrum of input signal

%( Now you can hold the figure and repeat for the output signal,
% use a different line style)

8. Using Simulink create the block diagram shown in Figure 1.

Figure 1. Simulink block diagram for FIR filter design

Generate a quadratic bidirectional sweep from 10Hz-12kHz, such that it will make 2
round trips in 10 seconds. Set the sample time to 1/24000. Set the sample time to
1/24000 in the zero order hold parameters. Buffer the inputs of the Spectrum scope.
Set the buffer size to 128 samples, buffer overlap to 64, the FFT length to 256, and
the spectral averages to 1. Use the fir1 command to create coefficients for a 5 th order
low-pass filter with a cut-off at 5kHz. Double click on the digital filter block to
indicate an FIR filter and enter the coefficients. Run the simulation and observe the
spectrum of the output and input signal while the frequency is swept over time. Also
observe the signal amplitude in the time scope and describe what you observe. In
the discussion section explain how this is either consistent or not with what was
expected.