Digital Signal Processing: Course Code: Credit Hours:3 Prerequisite:30107341

Digital Signal Processing
Course Code : 30102545

Credit Hours :3
Prerequisite :30107341
Instructor Information

Name Dr.Mohammad khrisat
Office No. B17 F4
Tel (Ext)
E-mail mkhrisat@bau.edu.jo
Office Hours 11:00-12:00 STT
Class Times Building Day Start Time End Time Room No.
Online Sunday 10:00 AM 11:00PM Online
Tuesday
Thursday
Course description Introduction to DSP. Discrete signals
and systems. Z-transform. Modeling and implementation of
discrete time systems. Time and frequency domain analysis of
signal processors. Design and analysis of finite impulse
response filters (FIR). Design and analysis of infinite impulse
response filters (IIR). Digital filter networks. Digital equalizers.
The DFT and FFT algorithms. DSP algorithms and applications
Course Title: Digital Signal Processing

Credit Hour(3-0)
Textbook: Textbook Title
Image of the textbook Cover

COURSE OBJECTIVES
The main objectives of the course are:
Introduce and discuss digital signal processing systems
Study of discrete time signals and systems
Manipulate discrete time signals and systems using transform methods
Develop the capabilities of students to perform signals and systems analysis using
computers
Analysis and design of digital IIR and FIR filters.
COURSE SYLLABUS
Week Course Topic

Week 1 Introduction to Digital Signal & Image Processing
Week 2 Analog to Digital & Digital to Analog Conversion
Week 3 Digital Signals
Week 4 Difference Equation & Diagrams
Week 5 Convolution & Correlation
Week 6 The z-Transforms & Stability
Week 7 Discrete Fourier Transform & Frequency Response
Week 8 Mid Term Exam
Week 9 Discrete Fourier Analysis & Signal Spectrum
Week 10 Finite Impulse Response Filter Design
Week 11 Infinite Impulse Response Filter Design
Week 12 Hardware and Software for Digital Signal Processors
Week 13 Adaptive Filters & Applications
Week 14 Multi-rate Signal Processing
Week 15 Fundamentals of Image Processing
Digital Signal Processing
Lecture-13
Magnitude & Phase Reponses
(Linear & Logarithmic Scales)
8
Properties of Magnitude & Phase Reponses
• The magnitude and phase responses are periodic, and they repeat
every 2𝜋 radians.
• This periodicity holds for all DTFTs.
• Both magnitude and phase responses are continuous functions

(they have values for every value of frequency).
• The magnitude response is an even function. The portion to the

left of the zero is a perfect mirror image of what lies to the right of
zero, that is,
• This is true for all magnitude responses.

9
Properties of Magnitude & Phase Reponses
• The phase response is an odd function.
• This is true for all phase responses.
• Normally, there is no need to record the part of the magnitude and phase
responses to the left of Ω = 0: This potion can be deduced from the plot for Ω
> 0.
• Furthermore, there is no need to consider values of Ω above.
• The frequency responses of the filters are usually plotted for values 0 ≤ Ω ≤ 𝜋.
• The advantage of using decibels is that a that an extremely large range of

gains , from very large to very small, can be conveniently plotted on same
graph. 10
Frequency Response & Filter Shape
The filter shape for a digital filter is furnished by its magnitude response.
LOW PASS FILTER
11
HIGH PASS FILTER
12
BAND PASS FILTER
13
BAND STOP FILTER
14
•Example-7:
Sketch the frequency response of the system.
Solution
Computing the frequency response with a step size of = 𝜋/4
15
16
Magnitude Response Phase Response
17
•Example-8:
Find and sketch the frequency response of the filter
whose impulse response is
h[n] = δ[n] – 1.25 δ[n-2]
Plot phases in degrees and linear gains.
Solution
The frequency response is given by
Magnitude and phase responses are calculated in the table.
18
19
20
•Example-9:
the difference equation describing a digital filter is
y[n] = 1.5y[n-1] – 0.85y[n-2] + x[n]
Find and sketch the frequency response. Plot phases in degrees
and gains in decibels.
Solution
The frequency response is given by
Magnitude and phase responses are calculated in the table.
21
22
23
• A filter is considered to pass signals where the gain is at least

0.707 of its maximum value.
• This is equivalent to a gain of 3 dB below maximum.
• In other words, when the maximum gain of a filter is 1, or 0 dB,

the edge of the pass band occurs where the gain is 0.707, or -3
dB.
• The shape of a digital filter can be designed independent of

sampling frequency.
• The sampling frequency chosen will, however, determine the

range of input frequency over which filter will act. 24
• When sampling rate is known, the frequency axis can be

plotted in terms of analog frequency 𝑓 in Hertz instead of
digital frequency in Ω radians.
• It is often easier to understand filter behavior in this way.
• As we know that
• With this conversion, nothing in frequency response diagram

changes except labelling of the horizontal axis.
• But the choice of sampling frequency has an enormous

impact on the effect of the filter. 25
Example-10: A filter with frequency response shown in the

following figure is used in a digital filter with a 12 kHz sampling
frequency.
a) Plot the frequency response against analog frequency in Hz

instead of digital frequency in radians.
b) What is the bandwidth of the low pass filter in Hz.
c) What is the bandwidth if the sampling frequency is changed to

30 kHz.
26
27
a) Plot the frequency response against analog frequency in Hz instead

of digital frequency in radians.
Digital Frequencies: 0 to π radians
Analog Frequencies: 0 to Fs/2 Hz
b)
• The bandwidth is the range of frequencies passed by the filter.
• For a low pass filter, this corresponds to the frequency where the gain
drops by 3 dB from its low frequency value.
• For the magnitude response in figure (on previous slide), this occurs
at a frequency of 1800 Hz.
c) If the sampling frequency were changed to 30khz, frequencies of

interest would range from 0 to 15 kHz, and bandwidth would
become 4500 Hz. 28
Filter’s Shape from Poles & Zeros
• When the poles and zeros of a filter are known, the filter’s
shape can be estimated without computing the frequency
response first.
• For a digital frequency Ω, the smaller the distance between ejΩ

and a zero, the smaller the gain at that frequency.
• Similarly, the smaller the distance between ejΩ and a pole, the
greater the gain at that frequency.
29
• Thus a filter with a single pole at z = 0.99 has its largest gains
at low frequencies Ω, which describes a low pass filter.
• Conversely, a filter with a single pole at z = -0.99 has its largest

gains at high frequencies Ω, which describes a high pass filter.
• The closer the zeros and poles are to the unit circle, the more
selective the filter, that is, the steeper the transition between
the pass band and stop band regions.
30
• Point which is moving on unit circle according to the different

values of frequency is
ejΩ = cosΩ +jsinΩ
Or in ordered pair form (cosΩ, sinΩ)
• As the frequency Ω increases from 0 to π radians, the point

(cosΩ, sinΩ) will travel through unit circle from (1,0) to (-1,0).
• The pole P is defined as a complex number

α + jβ
Or in ordered pair form (α , β)
31
• The filter’s shape can be computed as
• Where the distance between the point ejθ and pole P and can
be computed as
І ejΩ – P І = І (cosΩ +jsinΩ) – (α + jβ) І
32
• For a digital frequency Ω between 0 and π radians, the closer

ejΩ is to a pole of a filter, and further e jΩ is to a zero, the larger
the magnitude tends to be.
• A pole close to the unit circle leads to a filter shape with an

extremely large magnitude at some frequency.
• A zero close to the unit circle leads to a filter shape with a small
magnitude at some frequency.
• Extremely large and small magnitudes tend to signal filters that

are very selective.
33
•
Example-11: Deduce the filter shape describe by the transfer function.
Solution
This filter has a single pole at z = 0.45, with no zeros, and a unity gain.
• The distance between the point ejΩ and

pole is smaller near Ω = 0, and largest at Ω = π radians.
• Since the magnitude of the filter is reciprocal to the this

distance, the magnitude is largest near Ω=0, and smallest at Ω = π rad.
34
35
Filter Shape
36
•Example-12:
Deduce the filter shape describe by the difference
equation.
Solution
The transfer function of the filter is
• This filter has three poles at z = 0

• Zeros at z = j and z = -j
37
• Since all poles are at origin the distances between the point
ejΩ and the poles are the same for all values of Ω.
• Thus only the zero positions affect the overall filter shape.
• A magnitude of zero will be obtained for Ω = π/2 radians.
• The filter have a band stop effect as shown in the figure.
38
•Example-13:
Compare the filter shape for the following filters.
Solution
• None of the filters have zeros.

• The first filter has the poles , with magnitude of 0.95.
• The second filter has the poles , magnitude of 0.9.
• The third filter has the poles , magnitude of 0.8.
39
• As the figure shows, the value of Ω for which the poles are nearest the unit
circle is the same for all three filters.
• Therefore the magnitude response should all give a spike at this value of Ω.
• The poles of first filter (a) are closest to the unit circle, therefore the spike for
this filter should be largest.
Three Pole-Zero Plots Superimposed
40

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Digital Signal Processing

Course Code : 30102545

Course Title: Digital Signal Processing

Image of the textbook Cover

Week Course Topic

• This periodicity holds for all DTFTs.

• Both magnitude and phase responses are continuous functions

• The magnitude response is an even function. The portion to the

• This is true for all magnitude responses.

• The phase response is an odd function.

• This is true for all phase responses.

• Furthermore, there is no need to consider values of Ω above.

• The advantage of using decibels is that a that an extremely large range of

LOW PASS FILTER

HIGH PASS FILTER

BAND PASS FILTER

BAND STOP FILTER

Computing the frequency response with a step size of = 𝜋/4

Magnitude Response Phase Response

The frequency response is given by

Magnitude and phase responses are calculated in the table.

Magnitude Response Phase Response

The frequency response is given by

Magnitude and phase responses are calculated in the table.

Magnitude Response Phase Response

• A filter is considered to pass signals where the gain is at least

• This is equivalent to a gain of 3 dB below maximum.

• In other words, when the maximum gain of a filter is 1, or 0 dB,

• The shape of a digital filter can be designed independent of

• The sampling frequency chosen will, however, determine the

• When sampling rate is known, the frequency axis can be

• It is often easier to understand filter behavior in this way.

• With this conversion, nothing in frequency response diagram

• But the choice of sampling frequency has an enormous

Example-10: A filter with frequency response shown in the

a) Plot the frequency response against analog frequency in Hz

b) What is the bandwidth of the low pass filter in Hz.

c) What is the bandwidth if the sampling frequency is changed to

a) Plot the frequency response against analog frequency in Hz instead

c) If the sampling frequency were changed to 30khz, frequencies of

• For a digital frequency Ω, the smaller the distance between ejΩ

• Conversely, a filter with a single pole at z = -0.99 has its largest

• Point which is moving on unit circle according to the different

• As the frequency Ω increases from 0 to π radians, the point

• The pole P is defined as a complex number

• The filter’s shape can be computed as

• For a digital frequency Ω between 0 and π radians, the closer

• A pole close to the unit circle leads to a filter shape with an

• Extremely large and small magnitudes tend to signal filters that

• The distance between the point ejΩ and

• Since the magnitude of the filter is reciprocal to the this

• This filter has three poles at z = 0

• None of the filters have zeros.

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