Experiment Applications on Digital Filters
8 Low-pass and high-pass FIR and IIR Filters
PREPARED BY:
Prof. Ammar M. Abu-Hudrouss
Eng. MuhammadHashim I. Jabr
EXPECTED LEARNING OUTCOME:
Our aim is to become familiar with:
• Applications on digital filter design
• Filter designer tool via MATLAB
Part I: Application No. 1
Consider the following analogue filter which has the following H(s) given below:
𝟏𝟏. 𝟕𝟕𝟕𝟕𝟕𝟕𝟕𝟕𝟕𝟕𝒔𝒔𝟑𝟑
𝑯𝑯(𝒔𝒔) = 𝟐𝟐
(𝒔𝒔 + 𝟏𝟏𝟏𝟏𝟏𝟏𝟏𝟏. 𝟗𝟗𝒔𝒔 + 𝟓𝟓𝟓𝟓𝟓𝟓𝟓𝟓𝟓𝟓𝟓𝟓𝟓𝟓)(𝒔𝒔 + 𝟑𝟑𝟑𝟑𝟑𝟑𝟑𝟑)
By using MATLAB software, solve the following:
1. Find the filter poles and plot the pole-zero plot of the filter.
2. Plot the magnitude and phase responses of the filter 𝐻𝐻(Ω)
3. If the bilinear transformation is considered, find the sampling time if we want to
convert the filter to a digital filter with 𝜔𝜔𝑐𝑐 = 0.2π
𝟐𝟐 𝛚𝛚
Theoretically, the relationship between Ω and ω is: 𝛀𝛀 = 𝐓𝐓 ∗ 𝐭𝐭𝐭𝐭𝐭𝐭( 𝟐𝟐 )
Since, Ω represents the imaginary part of the poles of the analogue system,
then Ω = 2236.3 rad/sec
Given that, the digital frequency ω = 0.2 π rad/sec
Therefore T = 290.5868589 µs
1
4. Transform the filter into a digital filter using the bilinear transformation.
5. Plot the filter magnitude phase spectra, and the zero-pole plot.
NOTE: From the magnitude spectrum of H (ω) it is most likely the filter
works as a HIGH-PASS filter. Also, it is noticeable that the digital filter is
converted in which all its corresponding poles lay inside the unity circle
therefore, it is STABLE.
6. Find another two examples of low-pass filter and bandpass filters by searching
different literature and repeat the previous steps; you may assume the sampling
time of your choice.
Part II: Application No. 2
Consider using the filter designer tool in MATLAB to do the following below:
1. Design a low-pass filter with a cutoff frequency of 2.5 KHz and stop-band
frequency of 5 KHz with attenuation of 1 dB and 80 dB.
Consider a minimum order FIR filter with Kaiser window.
2. Construct a discrete signal from 𝐱𝐱 𝐚𝐚 (𝐭𝐭) = 𝟐𝟐𝟐𝟐𝟐𝟐𝟐𝟐 𝟏𝟏𝟏𝟏𝟏𝟏𝟏𝟏𝟏𝟏𝟏𝟏 + 𝐬𝐬𝐬𝐬𝐬𝐬 𝟏𝟏𝟏𝟏𝟏𝟏𝟏𝟏𝟏𝟏𝟏𝟏𝟏𝟏
sampled at a sampling rate of 48000 sample/s, with a total duration of 3 seconds.
Apply the filter designed above in step (1) to this discrete signal, and plot the
signal’s magnitude spectrum before and after filtering.
Convert the signal into audio file before and after filtering. What is the time delay
associated with the filter (you can see that from the time domain signal).
The delay time is approximately 0.1125ms.
3. Use windows recording tool to record two audio files with your voice (3 seconds
each).
The first recording for “... ﺳﯿﻦ ﺳﺎﻛﻨﺔ- ْﺲ
ْ ” ْﺳ ْﺴﺴand the other one "... " ْإﻣ ْﻤ ْﻤ ْﻢ – ﻣﯿﻢ ﺳﺎﻛﻨﺔ
Find the zero-crossing bandwidth for each letter. Apply the designed filter for
each file. Sketch the magnitude spectrum for the input and output files.
4. Design a digital low-pass Chebyshev Filter (Type I) with the same specifications
in step (1).
5. Repeat step (2) and (3) for the Chebsyshev Filter
The delay time is approximately 0.06125ms
2
6. Compare between the FIR filter and the Chebyshev Filter (in term of delay, order,
complexity, performance, ….)
Basis of Comparison FIR Filter IIR Filter
Stand for Finite Impulse Infinite Impulse
Response Response
Nature Non-recursive Recursive
Computational Efficiency Less Comparatively more
Usage Difficult Quite easy
Feedback Absent Present
Stability More Less
Requirement to generate Present and past Present and past samples
current output samples of input. of input along with past
output.
Delay offered High Comparatively lower
Transfer function Only zeros are present. Both poles and zeros are
present.
Memory requirement More Less
Sensitivity Less Comparatively more
Resolution offered at low Less More
frequencies
Controllability Easy Quite Difficult
7. What is the group delay of the filter and why it is important for filters to have
constant envelop group delay> plot the group delay for both Filters.
Group delay: is a measurement of the time taken by the modulated signal
to get through the system. Group Delay is measured in seconds. All
frequency components of a signal are delayed when they pass through a
filter.
Therefore, Group delay in a filter is the time delay of the signal through
the device under test as a function of frequency. For an ideal filter, the
phase will be linear and the group delay would be constant. However, in
the real-world group delay distortions occur, as signals at different
frequencies take different amounts of time to pass through a filter.
And it is important for the filters to have a constant envelope group
delay so that their phase shift varies linearly with the frequency, called:
linear-phase filters.
