CO-520-B

Signals and Systems Lab

Fall Semester 2024

Course Signals and Systems Lab – CO-520-B

Instructors - Uwe Pagel, Res.I Room 37 Tel.: +49 421 200 3114
- upagel (at) constructor.university
- Prof. Dr.Ing. Werner Henkel Tel.: +49 421 200 3157
- whenkel (at) constructor.university

Website - http://uwp-cu-lab.my-board.org/

August 28, 2024

Contents

I General remarks on the course 3

1 Experiments and Schedule 4

2 Grading and Attendance 5

2.1 Grading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
2.2 Cheating & Copying . . . . . . . . . . . . . . . . . . . . . . . . . . . 5

3 Lab Guidelines 6
3.1 Prelab . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
3.2 Lab Report . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
3.3 Supplies . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
3.4 Safety . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6

4 Manual Guideline 7
4.1 Circuit Diagrams . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
4.2 Values in Circuit Diagrams . . . . . . . . . . . . . . . . . . . . . . . . 9
4.3 Reading before the first Lab Session . . . . . . . . . . . . . . . . . . . 9

II Experiments 10
5 Experiment 1 : RLC-Circuits - Transient Response 11
5.1 Introduction to the experiment . . . . . . . . . . . . . . . . . . . . . 11
5.2 Execution Transient response of RLC-Circuits . . . . . . . . . . . . . 21
5.3 Evaluation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21

6 Experiment 2 : RLC-Circuits - Frequency Response 23

6.1 Introduction to the experiment . . . . . . . . . . . . . . . . . . . . . 23
6.2 Handling of the function generator and the oscilloscope . . . . . . . . 31
6.3 References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32
6.4 Prelab RLC Circuits - Frequency response . . . . . . . . . . . . . . . 33
6.5 Execution RLC Circuits - Frequency response . . . . . . . . . . . . . 34

7 Experiment 3 :
Fourier Series and Fourier Transform 35
7.1 Introduction to the experiment . . . . . . . . . . . . . . . . . . . . . 35
7.2 Prelab Fourier Series and fourier Transform . . . . . . . . . . . . . . 47
7.3 Execution Fourier Series and fourier Transform . . . . . . . . . . . . 49
7.4 Evaluation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 50

1
8 Experiment 4 : Sampling 51
8.1 Introduction to the experiment . . . . . . . . . . . . . . . . . . . . . 51
8.2 Prelab Sampling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61
8.3 Execution Sampling . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63

9 Experiment 5 : AM Modulation 65
9.1 Introduction to AM and FM experiments . . . . . . . . . . . . . . . . 65
9.2 Amplitude Modulation . . . . . . . . . . . . . . . . . . . . . . . . . . 66
9.3 Prelab AM Modulation . . . . . . . . . . . . . . . . . . . . . . . . . . 75
9.4 Execution AM Modulation . . . . . . . . . . . . . . . . . . . . . . . . 76
9.5 Evaluation AM modulation . . . . . . . . . . . . . . . . . . . . . . . 77

10 Theory 6 : FM Modulation 79
10.1 Frequency Modulation . . . . . . . . . . . . . . . . . . . . . . . . . . 79
10.2 References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 83
10.3 Prelab FM Modulation . . . . . . . . . . . . . . . . . . . . . . . . . . 84

III Additional Information 85

A Appendix 86
A.1 Hardcopy from oscilloscope screen . . . . . . . . . . . . . . . . . . . . 86
A.2 Books and other Tools . . . . . . . . . . . . . . . . . . . . . . . . . . 86

2
 Part I

General remarks on the course

3
1. Experiments and Schedule
1. 1.Day - 8:15 to 11:00 & Wed., Date TBD.
Introduction to the Lab
RLC-Circuits - Transient Response
Mandatory report!

2. 2.Day - 8:15 to 11:00 & Wed., Date TBD.

RLC-Circuits - Frequency Response
Only description of execution, data, and prelab!

3. 3.Day - 8:15 to 11:00 & Wed., Date TBD.

Fourier Series and Fourier Transform
Mandatory report!

4. 4.Day - 8:15 to 11:00 & Wed., Date TBD.

Sampling
Only description of execution, data, and prelab!

5. 5.Day - 8:15 to 11:00 & Wed., Date TBD.

AM Modulation
Mandatory report!

6. For experiment ’Theory 6 : FM Modulation’ you only have to deliver the

prelab together with experiment 5.

4
2. Grading and Attendance
2.1 Grading
1. The lab is a part of the module CO-520 and counts 30%. The grade is in %
and composed of the submitted lab reports and prelabs.

2. Distribution of the grades:

Action Involved Grade

5 Prelabs (Exp.2-6) each 8% Individual 40%
3 Lab Reports each 20% Individual 60%

3. Attendance to the course is mandatory. Missing an experiment without valid

excuse will subtract 1/3 from the grade.

2.2 Cheating & Copying

In case of cheating or plagiarism (marked citations are allowed but no complete
copies from a source) we will follow ’The Code of Academic Integrity’. The
report will be counted as not submitted 0%.
Note that there can be more consequences of a disciplinary nature de-
pending on the circumstances.

5
3. Lab Guidelines
3.1 Prelab
The experiments must be prepared by each student before they are carried out. The
manual provides a theoretical overview. In the prelab section, questions are asked
which must be prepared in writing before the date. Without the Prelab or at least a
good knowledge of the experiment, the student may be excluded from participation.
The prelab becomes the theory part of the respective report. The prelabs of the
experiments on which no report is written must also be submitted!

3.2 Lab Report

It is mandatory to write three lab reports.The general structure of a report is known
from the first year. Now it should become like this:

ˆ Cover Sheet - as before

ˆ Introduction - prelab belonging to the experiment

ˆ Experimental Set-up and Results - as before

ˆ Evaluation
Conclusion - is now a combined summary including error discussion

ˆ References - as before

ˆ Data from this week’s second experiment.

ˆ Additional: The prelab from this week’s second experiment!!

The submission of the report should be by email to [email protected].

Of course in case of computer problems hand written reports are also accepted.
Deadline is the Sunday after the experiment at 23:59. If you miss it, the experiment
will be downgraded or even reduced to 0%!!! As already stated before lab reports
are individual work.

3.3 Supplies
All equipment, cabling and component you need should be in your work area. If
you cannot find it, ask your lab instructor or teaching assistant, do not take it from
another group. Before leaving the lab, put everything back, where you found it!
Please bring your laptop so that you can record the readout the oscilloscope.

3.4 Safety
Recall the Safety Session from first semester!

6
4. Manual Guideline
The manual and the course website contain all the necessary information about the
laboratory. In addition, the manual contains a description of all experiments. Each
experiment is divided into the section ”Objective” and one (or more) subsection(s)
with ”Preparation”, ”Execution” and ”Evaluation”.

The Objective Section should give an introduction to the problem. In some

cases it also contains theory not completely covered in the lecture.

The Preparation Section describes the electrical setup.

The Execution Section is a detailed description on what to do and how and

what to measure.

The Evaluation Section should deepen the understanding of the topic. There
are questions about the experiment. You should solve these with help of the taken
data and compare the results to theory.

Before starting the experiment, read the complete description and try to understand
the problem. If something is not clear, read again and/or ask the TA/instructor.
Follow the preparations carefully to have the proper setup(s) and not to destroy
any component. Be sure to record -ALL- required data. All group members should
document the experiment! If data is missing you will have problems evaluating the
experiment!!

4.1 Circuit Diagrams

Next is an overview about the used symbols in circuit diagrams.

Connections

connected not connected

wire
wires wires

Connection are usually made using 1 or 0.5m flexible lab wires to connect the setup
to an instrument or voltage source and short solid copper wires on the breadboard.
In most of our experiments we consider these connections as ideal, i.e. a wire is a
real short with no ’Impedance’. In the following semesters you will see that this is
not true.

7
 Instruments

+ +
A ammeter V voltmeter

Since we use multimeters this symbol tells you how to connect and configure the
instrument. Take care of the polarity. Be careful, in worst case you blow it!!!

Voltage/Current Sources

+ +
V ~

fixed variable real ideal AC source pulse

real ideal current source signal generator generator
voltage source

These are the symbols used in the manual. If you check the web and look into
different books there are also other symbols in use!

Lumped Circuit Elements

+ +

variable electrolytic
resistor capacitor inductor
resistor capacitor

There is a different symbol for every lumped circuit element. Depending which
standard is used (DIN or IEC).

Semiconductors

NPN PNP N-channel P-channel diode zener diode

Transistor JFET

Same as with the symbols before you may find different representations for every
component!

8
4.2 Values in Circuit Diagrams
As you will see in the lab, we use resistors with colored rings. These rings represent
numbers or a multiplier. Most of the resistors have five rings. Three digits for the
value, one multiplier for the dimension, and one for the tolerance. In the circuit
diagrams we have a similar scheme. There are three digits and a dimension. The
letter of the dimension also acts as the comma i.e.:

1R00, 10R0, 100R for 1 Ω, 10 Ω, 100 Ω (= Value ∗ 100 )

1K20, 10K0, 100K for 1.2 KΩ, 10 KΩ, 100 KΩ (= Value ∗ 103 )
1M00, 10M0 for 1 MΩ, 10 MΩ (= Value ∗ 106 )

The numbering for capacitors in the circuit diagram is similar. Only the dimension
differs. Instead R, K, M (Ω, KΩ, MΩ) we have µ, n, or p (µF, nF, pF) (i.e. 1n5
means 1.5nF). The value is printed as number on the component.

4.3 Reading before the first Lab Session

As preparation for the first lab session read the description of the workbench, es-
pecially the parts about the power supply and the multimeter. You will find the
document on the course Web page in ’GeneralEELab I & II Files’
’Instruments used for the Experiments’.

9
 Part II

Experiments

10
5. Experiment 1 : RLC-Circuits - Tran-
sient Response
5.1 Introduction to the experiment
5.1.1 Objectives of the experiment
The aim of the first experiment is to investigate the transient response of second
order systems. A typical second order system in electrical engineering is an RLC
circuit. As part of the prelab, the transient behavior of such systems is investigated
using Matlab. Various RLC circuit configurations are implemented and tested as
part of the experiment. The experimental and simulated results are compared and
the differences are discussed.

5.1.2 Introduction
Second-order systems are very common in nature. They are named second-order
systems, as the highest power of derivative in the differential equation describing
the system is two. In electrical engineering, circuits consisting of two energy storage
elements, capacitors and inductors, for example RLC circuits, can be described
as second-order electrical system. These circuits are frequently used to select or
attenuate particular frequency ranges, as in tuning a radio or rejecting noise from
the AC power lines. The handout is divided into two parts. Throughout the first
part of the handout the following topics will be discussed.

1. The equation describing a second-order system in its general form.

2. The complete solution for a second-order differential equation (D.E.) repre-

senting the complete response of the system. The complete solution consists
of two responses.

(a) The transient response,

..depending on the circuit parameters the circuit operates under,
i. Over-damped condition
ii. Critically damped condition
iii. Under-damped condition
(b) The steady-state response,
.. due to a constant input signal (DC source).

The second part of the handout describes the practical part of the experiment. It
will explain how to develop a second-order differential equation describing a series
RLC circuit configuration and how the differential equation can be solved in order
to have a complete solution consisting of the transient response and steady-state
response.

11
5.1.3 Differential equations describing second-order systems
Circuits with two energy storage elements as the RLC circuits are described by a
second-order ordinary D.E.
d2 y(t) dy(t)
a2 + a 1 + a0 y(t) = x(t) (5.1)
dt2 dt
where,
y(t) is the response of the system to an applied input x(t).
a0 , a1 and a2 are the system parameters.
In the context of the response of second-order systems, it is more useful to rewrite
Eq. (5.1) in the form of a linear constant coefficient non-homogeneous differential
equation
d2 y(t) dy(t)
+ 2ζω n + ωn2 y(t) = Kωn2 x(t) (5.2)
dt2 dt
Thus, the system parameters become
r
a0
ωn = (5.3)
a2
a1
ζ= √ (5.4)
2 a0 a2
1
K= (5.5)
a0
where,
ωn is the natural frequency.
ζ is the damping ratio.
K is the gain of the system.

5.1.4 The complete solution for a second-order D.E.

The complete solution of the second-order non-homogeneous differential equation,
Eq. (5.2), is given by the following sum
y = yh + yf (5.6)
The solution consists of two terms, where yh is the homogeneous solution of the
second-order non-homogeneous differential equation, Eq. (5.2), where the input
x(t) = 0 . This solution satisfies the initial condition of the system. Therefore,
the homogenous solution of the second-order non-homogeneous differential equation
describes the transient response of the system. The term yf is the forced solution of
the second-order non-homogeneous differential equation, Eq. (5.2), where an exter-
nal input signal x(t) ̸= 0 is applied to the system. Therefore, the forced solution of
the second-order non-homogeneous differential equation describes the steady-state
response of the system. In the following we will derive in details the homogenous
solution and the forced solution for a constant input signal (DC source).

12
a. The homogeneous solution
The homogeneous solution of the second-order non-homogeneous differential equa-
tion can be found by rewriting Eq. (5.2), where the applied input x(t) is equal to
0, so that the non-homogeneous D.E is reduced to a homogeneous equation with
constant coefficients
d2 y(t) dy(t)
2
+ 2ζωn + ωn2 y(t) = 0 (5.7)
dt dt
This equation has a solution of the form

y(t) = Ceλt (5.8)

By substituting y(t) from Eq. (5.8) in Eq. (5.7), we get

Ceλt (λ2 + 2ζωn λ + ωn2 ) = 0 (5.9)

From Eq. (5.9), we get

λ2 + 2ζωn λ + ωn2 = 0 (5.10)

The above equation is called the characteristic equation. To find the homogeneous
solution, we need to solve the characteristic equation. The characteristic equation
has two roots (solutions)
p
λ1 = −ζωn + ωn ζ 2 − 1 (5.11a)
p
λ2 = −ζωn − ωn ζ 2 − 1 (5.11b)

By substituting in Eq. (5.8), we get

p
y1 (t) = C1 exp ((−ζ + ζ 2 − 1)ωn t) (5.12a)
p
y2 (t) = C2 exp ((−ζ − ζ 2 − 1)ωn t) (5.12b)

Each root λ1 and λ2 contributes a term to the homogeneous solution. The homoge-
neous solution is

y h = C 1 e λ1 t + C 2 e λ2 t (5.13)

where,

C1 and C2 are unknown coefficients determined by the initial conditions.

λ1 and λ2 are unknown constants determined by the coefficients of the
D.E. (depends on the circuit parameters R, L and C).

The time-dependent response of the circuit depends upon the relative values of the
damping ratio ζ and the undamped natural frequency ωn (radians/sec). According
to the relative values of ζ and ωn , we can classify the transient response into three
cases:

13
1. Under-damped case : 0 < ζ < 1, λ1 and λ2 are complex numbers
2. Critically damped case : ζ = 1, λ1 and λ2 are real and equal
3. Over-damped case : ζ > 1, λ1 and λ2 are real and unequal
Now, let us describe each case separately.
1. Under-damped Case: 0 < ζ < 1
For 0 < ζ < 1 the homogeneous solution of the second-order homogeneous
differential equation exhibits a damped oscillatory behavior.
p p
y(t) = exp (−ζωn t)(C1 cos (ωn (1 − ζ 2 ) t) + C2 sin (ωn (1 − ζ 2 ) t)) (5.14)

where,
C1 and C2 are unknown coefficients derived from the initial conditions.
On defining ωd , the damped natural frequency is given as
p
ωd = ωn 1 − ζ 2 (5.15)

Eq. (5.14) can be written as

y(t) = exp (−ζωn t)(C1 cos (ωd t) + C2 sin (ωd t)) (5.16)

Thus, the under-damped response is an exponentially damped sinusoid whose

rate of decay depends on the factor ζ. The terms ± exp (−ζωn t) define what
is called the envelope of the response. A step response of an under-damped
system is shown in Fig. 5.1. The oscillations of decreasing amplitude, exhibited
by the waveform are called ringing.

V 2
T
d

Envelope
 exp( nt )

 exp( nt )
Envelope

Figure 5.1: Under-damped 2nd order homogenous D.E.

2. Critically damped Case: ζ = 1

When the damping ratio is equal to one, the general solution is of the form
y(t) = C1 exp (−ζωn t) + C2 t exp (−ζωn t) (5.17)
where, C1 and C2 are unknown coefficients derived from the initial conditions.
Thus for a critically damped system, the response is not oscillatory. It ap-
proaches equilibrium as quickly as possible.

14
 3. Over-damped Case: ζ > 1
When the damping ratio is greater than one, the general solution to the ho-
mogenous equation is
p p
y(t) = C1 exp ((−ζ + ζ 2 − 1)ωn t) + C2 exp ((−ζ − ζ 2 − 1)ωn t) (5.18)

where, C1 and C2 are unknown coefficients derived from the initial conditions.
This indicated that the response is the sum of two decaying exponentials.
The total solution of a second-order system for the previously discussed cases can
be summarized as shown in Fig. 5.2 for ζ = 0.1, 0.3, 1, 2 and 3.

Overshoot
 = 0.1
 = 0.3
Ringing

y(t) 1

=3
=2
=1
0 1 2 3 4 5
n t -3
x 10

Figure 5.2: Solution of a second-order homogenous system

Figure 5.2 shows the normalized step response, where A is the amplitude of the
constant input signal. Generally, the aim of normalizing (scaling) is to compare the
output (the response) to a reference value. In our case, the reference value is the
step function and A represents the input voltage (DC source).
Note: When ζ = 0, the response becomes undamped and oscillations continue
indefinitely at frequency ωn .

b. The forced solution

The response to a forcing function will be of the same form as the forcing function.
The forced solution of the non-homogeneous second-order differential equation is
usually given by a weighted sum of the input signal x(t) and its first and second
derivatives. If the input x(t) is constant, then the forced response yf is constant as
well. If x(t) is sinusoidal, then yf is sinusoidal.
However, we will not discuss the forced solutions of second-order differential equa-
tions as part of this lab. Here we will deal only with constant input signals (DC
sources). The necessary steps for determining the steady-state response of RLC
circuits with DC sources will be described later in the handout.

15
5.1.5 Solving a second-order differential system
The behavior of a series RLC circuit shown in Fig. 5.3 can be determined from a
simple circuit analysis.
VR VL

R L

Vin C VC Vout

Figure 5.3: Second order system based on a serial RLC circuit

The input voltage is the sum of the output voltage and voltage drops across the
inductor and resistor
Vin = VR + VL + Vout (5.19)
The current i is related to the current flowing through the capacitor
dVout
i = iC = C (5.20)
dt
The voltage drop across the resistor is given by
dVout
VR = iR = RC (5.21)
dt
the voltage drop across the inductor is
di d dVout d2 Vout
VL = L = L (C ) = LC (5.22)
dt dt dt dt2
Substituting Eq. (5.21) and (5.22) into (5.19) yields
d2 Vout dVout
LC 2
+ RC + Vout = Vin (5.23)
dt dt
Thus, from Eq. (5.3), (5.4) and (5.5), the undamped natural frequency (radians/sec),
the damping ratio and the gain of the circuit are:
1
ωn = √ (5.24)
LC
r
R C
ζ= (5.25)
2 L
K=1 (5.26)
Example:
Let’s assume that the components in Fig. 5.3 have the values R = 50Ω, C = 1µF
and L = 50mH. By substituting in Eq. (5.24) and (5.25)
1
ωn = √ = 4472rad/sec
LC
r
R C
ζ= = 0.1 (under-damped case)
2 L

16
5.1.6 Definitions and Practical Hints
Step response

The step response is the response of a system upon applying an input signal in the
form of a step function.

The steady-state value

The steady-state value is the magnitude of the voltage, or current, after the system
has reached stability.

Ringing

Ringing is the oscillation phenomenon that occurs if the system is under-damped.

Overshoot

An overshoot is observed if the transient signal exceeds the final steady state value.
The overshoot is often represented by a percentage of the final value of the step
response. The percentage overshoot is

Vmax − VSteadyState
Percentage overshoot = ∗ 100%
VSteadyState

The undamped natural frequency, ωn

It is the frequency of oscillation of the system without damping.

The Step Response

For the step response of an under-damped system shown in Fig. 5.4, the transient
response specifications are:
• Peak time, Tp
It is the time required for the response to reach the peak of the overshoot.
• Rise time, Tr
It is the time required for the step response to rise from 10% to 90% of its
final value for critical and over-damped cases, and from 0% to 100% for under-
damped cases.
• Settling time, Ts
It is the time required for the step response to settle within a certain percentage
of its final value. The percentage can be chosen to be 2% or 5%.
Note: Not all these specifications apply to all cases of system response. For example,
for an over-damped system, the terms ringing, peak time and maximum overshoot
do not apply.

17
 1.6

1.4
max. Overshoot
1.2

1
y(t)
0.8 2% or 5%
of the final value
0.6

0.4

0.2

Tr 1 2 3 4 5
Tp t -3
x 10
Ts

Figure 5.4: Under-damped second-order system

5.1.7 Initial conditions of switched circuits

The switched circuit is a circuit with one or more switches that open or close at a
certain point in time. We are interested in the change of the current and voltage
of energy storage elements (L and C) after the switch changed from open to close
or vice versa. So if the time of the switch is t = 0, we want to determine the
current through the inductor and voltage across the capacitor at t = 0− and t = 0+
immediately before and after the switching. These values together with the sources
determine the behavior of the circuit for t > 0. Before continuing with a solved
example two important notes should be mentioned which are:
a. The current through an inductor cannot change instantaneously, whereas the
voltage drop across an inductor can change instantaneously.

b. The voltage drop across a capacitor cannot change instantaneously, whereas

the current flow through a capacitor can change instantaneously.

Example:
Consider the circuit shown in Fig. 5.5. The switch has been closed and steady
state conditions were reached. In order to find vC (0− ) and iL (0− ) the capacitor is
replaced by an open circuit and the inductor by a short circuit as shown in Fig. 5.6.
It can be easily calculated that
Vin
iL (0− ) = (5.27)
R1 + R2
R2
vC (0− ) = Vin (5.28)
R1 + R2

18
Example:
Consider the circuit shown in figure (5). The switch has been closed and steady state
conditions were reached. In order to find vC(0-)and iL(0-) the capacitor is replaced by an
Fig. 5: Switched circuit
open circuit and the inductor by a short circuit as shown in figure (6).

Fig. 5: Switched circuit

Figure 5.5: Switched circuit FigureFig. 6: Steady-state conditions
5.6: Steady-state conditions

It can be easily calculated that

V
i L (0 − ) =cannot
Since the current through an inductor in
change instantaneously and the voltage (27)
R1 + R 2
across a capacitor cannot change instantaneously, R
therefore,
v C (0 − ) = Vin 2
(28)
Vin R1 + R 2
iL (0− ) = iL (0+ ) = (5.29)
Since the current through an inductor cannot change instantaneously and the vol
R1 + R2
across a capacitor cannot change instantaneously, therefore,
Fig. 6: Steady-state conditions
R Vin
vC (0− ) = vC (0+ ) = Vin
2 i L (0 − ) = i L ( 0 + ) = (5.30) (29)
R +R R 1 + R2
1 2
It can be easily calculated that
Vin
i L (0 − ) = 5.1.8 The Complete Response
R1 + R 2
(27)
13

v C (0 − ) = Vin The
R 2 necessary steps to determine the complete response of a second-order system
(28)
1 + R2
Rbased on RLC network with DC sources are:
Since the current through an inductor cannot change instantaneously and the voltage
across a capacitorˆcannot For transient response therefore,
change instantaneously,
V
i L (0 − ) = i L ( 0 + ) = in
(29)
R 1 + R 2 1. Using Ohm’s law, KVL, and/or KCL, obtain a second
order differential
nonhomogeneous equation. Another way is to obtain two first-order dif-
ferential equations, and then combine them to a second order differential
13
non-homogeneous equation.
2. Solve the homogeneous equation corresponding to the obtained second
order differential non-homogeneous equation.
3. Obtain the complete solution by adding the forced solution to the ho-
mogeneous solution. The complete solution still contains unknown coef-
ficients C1 and C2 .
4. Use the initial conditions to determine the value of C1 and C2 .

ˆ For DC steady state response

1. Replace all capacitances with open circuits.

2. Replace all inductances with short circuits.

ˆ Solve the remaining circuit.

19
5.1.9 References
1. A. V. Oppenheim, A. S. Willsky, S. H. Nawab, ”Signals and Systems”, Prentice
Hall, Second Edition (1997)

2. Sarma, M.S., ”Introduction to Electrical Engineering”, Oxford University Press,

2001.

3. R.A. DeCarlo, P-M. Lin, Linear Circuit Analysis, Oxford press, 2nd edition.

4. Allan R. Hambley ,”Electrical Engineering: Principles and Applications”,

Prentice Hall,Second Edition.

20
5.2 Execution Transient response of RLC-Circuits
5.2.1 Problem : Design of an RLC circuit
Implement the RLC circuit shown below on the breadboard.
R-Decade
100Ohm 10mH

Vpp = 1V
Voff = 0.5V
6n8F
f = 100Hz
Ri = 50Ohm

1. Set the function generator to produce a 100 Hz square wave with an ampli-
tude of 0.5 V and an offset of 0.5 V. Check with the oscilloscope if the signal
modulates between 0 V and 1 V. Set the R-decade to 100 Ω. Connect the
oscilloscope in parallel to the capacitor.
2. Measure the damped frequency fd . The frequency fd can be determined by
measuring the time or frequency of the exponentially damped sinusoidal. Take
a hardcopy of one signal period and one focusing on the ringing phenomenon.
3. Calculate the damped radian frequency ωd . In your calculation, consider the
internal resistance of the function generator to be 50 Ω. Compare the calcu-
lated value with the measured value in step (2). If they are consistent, proceed
with the next steps.
4. Calculate the resistance so that the circuit is critically damped. Display the
signal and take a hardcopy.
5. Check if the practical signal is critically damped. Vary the the R-decade value
and take a hardcopy of the final result.
6. Set the R-decade to 30k Ω, so that the circuit is over-damped. Display the
transient voltage across the capacitor and take a hardcopy.

5.3 Evaluation
1. Use the circuit from the experiment and obtain the differential equation for
the voltage vc (t) across the capacitor when R = 100 Ω, identify the damping
nature of the circuit and determine the values for the coefficients C1 and C2 .
2. Plot the voltage vc (t) using Matlab.
3. Calculate the resistor value to obtain a critically damped case and obtain the
corresponding equation describing the voltage vc (t) including the values for C1
and C2 . Plot the voltage vc (t) using Matlab.
4. Compare the experimental results obtained in the lab with the calculations.
Provide a detailed explanation if the experimental results deviate. Discuss the
origin of the deviation.

21
5. Solve the following problem:
The switch in the circuit below is closed at t = 0.

S i R1 = 25 Ohm R2 = 56 Ohm

u_1 i_c i_L

t=0

U=16.2V u_2 C=2uF L=20mH

(a) Obtain the differential equation for the current iL (t).

(b) Identify the damping nature of the circuit and determine the values for
the coefficients C1 and C2 .
Show the formula for the complete response!
(c) Plot the current iL (t) using Matlab.

22
6. Experiment 2 : RLC-Circuits - Frequency
Response
6.1 Introduction to the experiment
6.1.1 Objective of the experiment
The goal of the experiment is to study the frequency response of RLC circuits
and their application as analog filters and resonators. As part of the prelab, the
frequency response of different RLC circuit configurations will be studied using
Matlab. As part of the experimental procedure, different RLC circuit configurations
will be implement and tested. The experimental and the simulation results will be
compared and the differences will be discussed.

Introduction
In the experiment ”RLC Transient Response”, we studied the time dependent re-
sponse of RLC circuits including the transient response and the steady-state response
for a constant DC input signal. In the second experiment, we will only explore the
steady-state response of an RLC circuit for a periodic sinusoidal input signal. As
the frequency of the periodic sinusoidal input signal changes the circuit response
changes, that is why the second experiment is called ”RLC frequency response”.
In electronics, resonating circuits are often used to select or attenuate particular
frequency ranges, as in tuning a radio. In its easiest form, a resonator can be
realized using a resistor, inductor, and a capacitor. Therefore, the circuit consists
of at least two different energy storage devices.

Series and parallel RLC configurations

In this experiment, we will in particular study the series and the parallel configura-
tion of the RLC resonating circuits shown in Fig. 6.1 and (6.2).
First, we will write down the impedance of the series RLC circuit in Fig. 6.1.
1
ZS = RS + jωL + (6.1)
jωC
The admittance of the parallel RLC circuit in Fig. 6.2 can be expressed in an analog
way
1
YP = GP + jωC + (6.2)
jωL
The absolute value of the impedance and the admittance is given by
s  2
2 1
|ZS | = RS + ωL − (6.3a)
ωC

23
 I

I
C Vc
C L RP
U
V L VL
Ic IL IR

RS VR

Figure 6.1: Series resonator based Figure 6.2: Parallel resonator based on
on a serial RLC circuit. a parallel RLC circuit.

s  2
1
|YP | = G2P + ωC − (6.3b)
ωL
The phase of the complex impedance and admittance can be expressed by
 
ωL − 1/ωC
φ = arctan (6.4a)
RS

 
ωC − 1/ωL
φ = arctan (6.4b)
GP
Based on the phasor plot of the series and parallel resonators in Fig. 6.3 and Fig. 6.4,
the amplitude and the phase of the complex impedance and admittance can be
determined for a given frequency.
In general, the impedance and the admittance can be written as

Z(ω) = R + jX(ω) (6.5a)

Y (ω) = G + jB(ω) (6.5b)

In the phasor plot, the tip of the vector corresponds to the impedance Z(ω) or the
admittance Y (ω) of the circuit. The following information can be extracted from
the phasor plot.
ˆ The circuit is in resonance if the oscillation parameter is maximized. In the
case of a series resonator the oscillation parameter is the current, whereas for
a parallel resonator the oscillation parameter is the voltage.
ˆ The oscillation parameter is maximized and the resonance point is reached,
when the phasor intersects with the real axis of the graph. In this case the
frequency ω is getting equal to the resonance frequency ω0 .

24
 • •TheThe oscillation
oscillation parameter
parameter is maximized
is maximized andandthethe resonance
resonance point
point is reached,
is reached, when
when
the phasor intersects with the real axis of the graph. In this case the frequency ω is
the phasor intersects with the real axis of the graph. In this case the frequency ω is
getting
getting equal
equal to the
to the resonance
resonance frequency
frequency ω0.ω0.

Fig.Fig.6.3:
Figure 3: 3:Phasor
Phasordiagram
Phasor diagramof
diagram of of aseries
aa se- seriesFig.Fig.
Figure4: 4: Phasor
Phasor
6.4: diagram
diagram
Phasor diagramof of a parallel
aof parallel
a par-
resonator
resonator based
based on ona
ries resonator based on a RLC cir- a
RLCRLC circuit.
circuit. In In allel resonator based on a RLC cir-In In
resonator
resonator based
based on on
a a
RLCRLC circuit.
circuit.
this
this case,
case, the the impedance
impedance of of
a a series
series this case,
this thethe admittance of of a parallel
cuit. In this case, the impedance of cuit.case,
In this admittance
case, a parallel
the admittance of
resonator
resonator is plotted
is plotted [3].[3]. resonator is plotted
resonator is plotted [3]. [3].
a series resonator is plotted [3]. a parallel resonator is plotted [3].

The frequency ω0 is the frequency associated with the resonance point of the circuit.
The frequency
Consequently,
The ω0 ωis0 the
the impedance
frequency is the frequency
results
frequency associated
in associated with
with thethe resonance
resonance point
point of of
thethe circuit.
circuit.
Consequently,
Consequently, thethe impedance
impedance results
results in in
Im(Z(ω = ω0 )) = 0 (6.6)
Z ((ωZ (=ωω=0 ω
Im(Im )) 0=))0= 0 (6)(6)
and the admittance results in
andand
thethe admittance
admittance results
results in in
Im(Y (ω = ω0 )) = 0 (6.7)
Im(Im (Y (=ωω=0 ω)) 0=))0= 0
Y (ω (7)(7)
In both cases, the phase gets zero
In both
In both cases,
cases, thethe phase
phase gets
gets zero
zero
φ=0 (6.8)
2222
Furthermore, the following circumstances apply for the resonance points of a series
resonator:

ZS (ω = ω0 )) = Re(ZS (ω)) = RS (6.9)

If ω = ω0 , the impedance is minimized and |ZS (ω0 )| = min.

The parallel resonator exhibits an analog behavior.

YP (ω = ω0 )) = Re(YP (ω)) = GP = 1/RP (6.10)

Due to the opposite characteristics of an inductive reactance (the inductive reac-

tance increases as the frequency is increased) and a capacitive reactance (capacitive
reactance decreases with higher frequencies) the reactance XL equals XC for the
resonance frequency. Consequently, the inductive and the capacitive impedances
compensate each other, so that the reactive power becomes zero, which means that
only effective (real) power is consumed by the circuit.

25
6.1.2 Application of series and parallel RLC circuits
Reactive power compensation

The reactive power is minimized under the following conditions:

ˆ The reactive power gets minimized for a series resonator if the circuit is
driven by a current source. In such a case the input current is constant (|I| =
const.) and the impedance determines the voltage drop across the circuit. As
the impedance is minimized Z(ω = ω0 ) → min. it follows that the voltage
drop is minimized as well |V (ω0 )| = |I| ∗ |Z(ωo )| → min.!

ˆ The reactive power gets minimized for a parallel resonator if the circuit
is driven by a voltage source. In such a case the input voltage is constant
(|V | = const.). As the admittance is minimized Y (ω = ω0 ) → min. it follows
that the current is minimized as well |I(ω0 )| = |V | ∗ |Y (ωo )| → min.!

The reactive power is maximized under the following conditions:

ˆ The reactive power gets maximized for a series resonator if the circuit is
driven by a voltage source. In such a case the input voltage is constant (|V | =
const.) and the impedance determines the current flow. As the impedance is
minimized Z(ω = ω0 ) → min., it follows that the current flow through the
circuit is maximized |I(ω0 )| = |V |/|Z(ωo )| → max.!

ˆ The reactive power gets maximized for a parallel resonator if the circuit is
driven by a current source. In such a case the input current is constant (|I| =
const.). As the admittance is minimized Y (ω = ω0 ) → min. it follows that the
voltage drop across the circuit is maximized as well |V (ω0 )| = |I|/|Y (ωo )| →
max.!

Therefore, resonators can be applied for reactive power compensation. Very often
electrical consumers have an ”inductive character”. This is the case if several in-
ductive consumers like motors, pumps or heaters are in operation. The reactive
power consumption of a load can be reduced or compensated by using a reactive
load, which has the opposite reactive impedance. For example: A capacitive load
can be used to compensate an inductive load. As a consequence the impedance of
the whole system is getting reduced, which means the impedance is getting nearly
real.

Filters

Analog Filters can be constructed based on RLC circuits, which transmit certain
frequencies (resonance frequency) in an optimized fashion. Other frequencies (higher
and lower frequencies) can be attenuated or blocked. By combining RLC circuits
with slightly different resonance frequencies band-pass filters and band-stop filters
can be designed.

26
RLC filter design The frequency response of a RLC circuit can be represented
by a magnitude and a phase diagram. Two magnitude and phase plots for a series
and a parallel resonator are shown in figures 6.5 and 6.6. The upper graphs show
the magnitude diagram and the lower graphs the phase diagram of the series and
the parallel resonator.
The magnitude of the signal is normalized to simplify visualization and facilitate a
comparison of different magnitude plots. Furthermore, the frequency is normalized
in all graphs by the resonance frequency ω0 . We know from our previous discussion
that the current is maximized for a series resonator in resonance, whereas for a
parallel resonator the voltage is maximized. Consequently, Fig. 6.5 exhibits the
normalized current for the series resonator and Fig. 6.6 shows the normalized voltage
for a parallel resonator.
The magnitude and the phase for the two circuits in figures 6.5 and 6.6 were cal-
culated for two different resistors. The dashed lines correspond to the resistors Rs1
and Rp1 , whereas the solid lines correspond to the resistors Rs2 and Rp2 . With
increasing resistance of the resistor Rs the width of the magnitude for the series
resonator is enhanced. The opposite behavior is observed for a parallel resonator.
With increasing resistance of the parallel resistor Rp the width of the magnitude is
reduced. It follows that, Rs1 < Rs2 and Rp1 > Rp2 .
The phase diagram exhibits a corresponding behavior. With increasing series re-
sistance the transition region from −π/2 to π/2 is widened, whereas for increasing
parallel resistance the transition region is getting narrower. Therefore, the series and
the parallel resistance have a distinct influence on the bandwidth of the resonators.

Bandwidth and quality factor The bandwidth is a measure of the frequency

selectivity of a resonating circuit. The bandwidth B of the resonators can directly
be extracted either from the magnitude or the phase plot in Fig. 6.5 and Fig. 6.6.
In the magnitude plot, the bandwidth corresponds to the full width of the√curve at
half maximum, which means that the magnitude is dropped by a factor 2. From
the phase plot, the bandwidth B can be extracted by taking the difference of the
phase between +45◦ ≡ +π/4 and −45◦ ≡ −π/4). Furthermore, the bandwidth can
be determined by mathematical means.

In the following, we will focus on a series resonator In the case of a phase

difference of 45◦ the real part and the imaginary part of the impedance are equal.
It follows,
Im(Z S (ω1 )) = Im(Z S (ω2 )) = RS (6.11)
The equation can be expressed in different terms for negative values of the imaginary
part,
1
|Im(Z(ω1 ))| = − ω1 L = RS (6.12)
ω1 C
So the frequency ω1 can be determined by:
s 
2
RS RS 1
ω1 = − + + (6.13)
2L 2L LC

27
 Voltage
Voltage const.
const. Current
Current const.
const.
1 1 1 1
I/I I/I U / U U / U0
0 0 0
B /ω B1/ω B1/ω B1/ω
1
0.707 0.707 0.707 0.707
RS1 RS1 RS1 RP1
0.5 0.5 0.5 0.5
B /ω B2/ω B2/ω B2/ω
2

RS2 RS2 R
R P2
S2

0 0 0 0
-2 -2 -1 -1 0 0 1 1 2 2 -2 -2 -1 -1 0 0 1 1 2 2
10 10 10 10 10 10 10 10 10 10 10 10 10 10 10 10 10 10 10 10

ω / ω0 ω / ω0 ω / ω0 ω / ω0

1.57 1.57 1.57 1.57

φV - φIφV - φI φV - φIφV - φI
0.79 0.79 0.79 0.79

R R
S1 S1 RS1 RP1
0 0 0 0

-0.79 -0.79 -0.79 -0.79

RS2 RS2 RS2 RP2

-1.57 -1.57 -2 -1 0 1 2
-1.57 -1.57 -2 -1 0 1 2
-2 10 -1 10 0 10 1 10 2 10 -2 10 -1 10 0 10 1 10 2 10
10 10 10 10 10 10 10 10 10 10

ω / ω0 ω / ω0 ω / ω0 ω / ω0

Figure 6.5: Magnitude and phase plot of Figure 6.6: Magnitude and phase plot of
a series resonator (RLC circuit) driven a parallel resonator (RLC circuit) driven
by a voltage source (|V | = const.). The by a current source (|I| = const.). The
normalized magnitude and the phase normalized magnitude and the phase
are shown as a function of the nor- are shown as a function of the normal-
malized frequency. Rs is the series re- ized frequency. Rp is the parallel re-
sistor of the RLC circuit. Rs1 is the sistor of the RLC circuit. Rp1 is the
larger and Rs2 the smaller series resis- larger and Rp2 the smaller series resis-
tor: Rs1 > Rs2 [3]. tor: Rp1 > Rp2 [3].

For the positive imaginary part the following equation applies:

1
|Im(Z(ω2 ))| = ω2 L − = RS (6.14)
ω2 C
So the frequency ω2 can be determined by:
s 
2
RS RS 1
ω2 = + + (6.15)
2L 2L LC

Based on the two resonance frequencies the bandwidth B can be determined as:
RS
B = ω2 − ω1 = (6.16)
L
Besides the bandwidth, the quality-factor (Q-factor) is also an important measure
of the frequency selectivity. For example, filters with high Q-factors are important

28
and necessary for applications in wireless communications to separate or filter out
closely spaced channels/bands. The quality factor is a representation of the width
of the resonance peak (the larger the Q value, the narrower the peak). Hence, there
is a relation between the Q-factor and the bandwidth, where high-Q circuit has a
small bandwidth and low-Q circuit has a large bandwidth.
In terms of energy consumption, the Q-factor is a measure for the ratio between the
reactive power (inductive or capacitive) and the real power of a resonator. In other
word, it is a measure of the energy-storage in relation to the energy dissipation of
the circuit. The Q-factor can be determined by:
QC QL
Q= = (6.17)
PR PR
where,
Q is the Q-factor of the resonator.
QC and QL are the reactive power of the capacitor or inductor.
PR is the real (effective) power.
The quality factor of a series resonator can be expressed by:
X0 ω0
QS = = (6.18)
RS ω2 − ω1
X0 is the reactive resistance of series RLC circuit under resonance conditions.

As the circuit is driven in resonance frequency, the following equation applies:

1
X0 = ω0 L = (6.19)
ω0 C
or it can be expressed by the following terms,
1
ω0 = √ (6.20)
LC
and
r
L
X0 = (6.21)
C
It can be seen from Eq. (6.17)-(6.21) that the resonance frequency is defined by the
inductance and capacitance of the circuit, whereas the series resistance of the circuit
determines the Q-factor.

6.1.3 Practical hints

Phasor diagram
Generally, the phasor diagram represents a complex variable G(jω) by a rotating
phasor as ω varies from zero to infinity. where,

G(jω) = Re[G(jω)] + Im[G(jω)] = R(ω) + jX(ω) = |G(jω)|∡G(jω)

29
 |G(jω)|2 = [R(ω)]2 + [jX(ω)]2
 
X(ω)
∡G(jω) = φ = arctan
R(Ω)
Each point on the curve represents the complex value of the variable G(jω) for a
given frequency. In the phasor diagram provides information about the locus of the
complex variable and the circulation of the phasor as the frequency increases. The
projections of G(jω) on the axis are the real and imaginary components at that
frequency.

Magnitude and phase plot of frequency response

Magnitude and phase plots are used to represent the frequency response of any given
circuit. Both plots are usually shown together and called Bode diagram.
a. Plot of the magnitude of |G(jω)|, indicating the amplitude change imposed on
a sinusoidal input signal as a function of frequency.
b. Plot of the phase angle ∡G(jω) , indicating the phase change imposed on a
sinusoidal input signal as a function of frequency.
Both graphs are plotted against the frequency ω rad/sec.

Lissajou figure
An oscilloscope can be used to determine the phase difference between two periodic
signals. However, an oscilloscope is only able to measure a voltage signal. A current
can be measured by creating a voltage drop over a resistor. The phase difference
can be extracted by applying both of these signals to the input channels of the
oscilloscope.
However, for small phase differences it is difficult to extract the phase difference
directly from the screen of the oscilloscope. As an alternative, a Lissajou figure can
be used to determine the phase difference of two signals. A Lissajou figure is more
accurate if it comes to smaller phase differences. The concept of a Lissajou figure
goes back to the ”old days” when people were using cathode ray tubes (CRT) based
oscilloscopes. The position of a spot on the screen of a CRT oscilloscope is controlled
by the voltages applied to the x and y deflection capacitors of the cathode ray tube.
In normal operation (a voltage is shown as a function of time), a triangular voltage
is applied to the x-deflection capacitor so that the spot moves from the left to the
right side of the screen. In the case of a Lissajou figure we directly apply the second
signal to the x-deflection capacitor.
The x-deflection of the signal can be described by:
Vx (t) = Vpp ∗ sin (ωt) (6.22)
whereas the y-deflection can be described by:
Vy (t) = c ∗ Ipp ∗ sin (ωt + φ) (6.23)
Vpp is the peak amplitude of the voltage, c is a constant factor (resistance) and Ipp
is the peak current.

30
 Signals and Systems Lab, Advanced
Signals
Electrical
and Systems
Engineering
Lab, Advanced
Lab Course
Electrical
I, Fall 2010,
Engineering
Jacobs University
Lab Course Bremen.
I, Fall 2010, Jacobs University Bremen.

In normal operation (a In voltage

normal is
operation
shown as(a avoltage
function
is shown
of time),asa atriangular
function of
voltage
time),isa triangular voltage is
applied to the x-deflection
applied
capacitor
to the x-deflection
so that the spot
capacitor
moves sofrom
that the
the left
spottomoves
the right
from
side
the left to the right side
of the screen. In the case
of theofscreen.
a Lissajou
In the
figure
caseweofdirectly
a Lissajou
apply figure
the second
we directly
signal
apply
to the
the second signal to the
x-deflection capacitor.x-deflection capacitor.

Fig. 7: Extraction Fig.

of 7:the Extraction
phase Fig.of 8:theExtraction
phase Fig.
of the8: Extraction
phase of the phase
difference between difference
two sinusoidal
betweendifference
two sinusoidal
by using a Lissajou
difference
figure.
by using a Lissajou figure.
Figure 6.7: Extraction
signals. of the phase dif-
signals. Figure 6.8: Extraction of the phase dif-
ference between two sinusoidal signals. ference by using a Lissajou figure.
The x-deflection of theThe
signal
x-deflection
can be described
of the signal
by: can be described by:

V (t=
For ωt )= V
X sin (ωcan
0, ⋅ we
P t ) , extract:
V (t ) = V ⋅ sin
X 2(ω∗t )c, ∗ Ipp ∗ sin φ = a
P (22) (22)

Furthermore, we can
whereas the y-deflection determine
whereas
can bethe described 2 ∗by:ymax
y-deflection can be 2 ∗ c ∗ Iby:
=described pp = b.
As a consequence we can deduce the phase difference from the parameters a and b
V (t ) = c ⋅ I ⋅ sin (ωt + ϕV ) (t ) = c ⋅ I ⋅ sin (ωt + ϕ) (23) (23)
by the following simple expression
Y P Y P

VP is the peakamplitude
a  VP isofthe thepeak
voltage,
amplitude
c is a constant
of the voltage,
factor c(resistance)
is a constant andfactor
IP is (resistance)
the and IP is the
φ = arcsin
peak current. peak current. (6.24)
b
For ωt = 0, we can extract: For ωt2=⋅ c0,
⋅ I we (ϕ) =extract:
⋅ sincan
P a. 2 ⋅ c ⋅ I ⋅ sin (ϕ) = a .
P
The Furthermore,
accuracy we ofcan this method
determine
Furthermore, 2 ⋅ yweiscan
= 2distinctly
max ⋅ I = b . 2 higher
⋅ cdetermine
P ⋅ y = 2 ⋅ cthan
max ⋅ I = b . a direct comparison of the
P
signals. This is in particular
As a consequence weAs can
a consequence
truewe
deduce the phase
forcan small
difference
phase
deduce from the phase
differences.
the parameters
differenceafrom
and the
b byparameters a and b by
the following simple expression
the following simple expression

6.2ϕ = arcsin
Handling
a
  ϕ = arcsinof
 a  the function generator and the
  (24) (24)
b b
oscilloscope
The accuracy of this method
The accuracy
is distinctly
of thishigher
method
than
is adistinctly
direct comparison
higher thanofa the
direct
signals.
comparison of the signals.
This is in particular true
This
for is
small
in particular
phase differences.
Throughout the procedure of thetrue
labforthe
small phase differences.
frequency response of several circuits has
to be taken. In order to measure the frequency response of a circuit the sweep mode
of the function generator can be used. In this case the function generator changes
its frequency over time. 29 29
Using and setting up the sweep mode:
a. Settings of the Function generator:
Select the ”Sweep Menu” from the function generator and set:

Sweep frequency from 100 Hz (1: START F)

Sweep frequency to 100 KHz (2: STOP F )
Sweep time to 500 msec (3: SWP TIME )
Sweep mode to logarithmic (4: SWP MODE )

b. Settings of the Oscilloscope:

To see the full sweep at the oscilloscope you have to set the time base
to 500 ms/10 div = 50 ms/ div.
c. Synchronization of the function generator and the oscilloscope:
In order to measure a frequency response with the oscilloscope the function
generator and the oscilloscope have to be synchronized. To do so, the syn-
chronization signal of the signal generator has to be connected with the trig-
ger input of the oscilloscope and the oscilloscope has to be set to ”external
trigger”.

31
 d. Grounding of the function generator and the oscilloscope:
The grounds of the function generator and the oscilloscope have to be con-
nected together throughout all measurements in order to have the correct
output on the oscilloscope screen. You have to construct the circuit in this
way that one of the terminals of the component under test is always on ground
while measuring the voltage across the component.

6.3 References
1. M.S. Sarma, Introduction to Electrical Engineering, Oxford Series in Electrical
and Computer Engineering, 2000.

2. J. Keown, ORCAD PSpice and Circuit Analysis, Prentice Hall Press (2001).

3. F. Hohls, Lab experiments, Resonators, University Hannover, Spring 2003.

4. R.A. DeCarlo, P-M. Lin, Linear Circuit Analysis, Oxford Press, 2nd edition.

32
6.4 Prelab RLC Circuits - Frequency response
6.4.1 Problem: RLC resonator
Given is a series RLC resonant circuit with R = 390Ω, C = 270 nF and L = 10 mH.

1. Name the filter characteristic measured over the different components, com-
ponent combinations.

2. Show the Bode magnitude plot across the resistor, the capacitor, the inductor
and across the capacitor and the inductor together. Use a 5 V amplitude and
vary the frequency starting at 100 Hz up to 100 KHz.
Develop a Matlab script to plot the four characteristic in one graph. Attach
the script to the prelab!

3. Taking the magnitude across the resistance represents a band-pass filter. Cal-
culate the bandwidth and the Q factor of the circuit. Extract the bandwidth
from the Matlab plot and compare.

33
6.5 Execution RLC Circuits - Frequency response
6.5.1 Problem 1 : Characterization of an RLC resonator
Implement a series RLC resonant circuit based on R = 390Ω, C = 270 nF, and
L = 10 mH. Use the function generator as source.

1. Set the function generator to sine, 5 Vpp , and no offset. Use the sweep mode.
Vary the frequency in 500 ms from 100 Hz to 100 kHz. Use log sweep mode.

2. Obtain hardcopies of the voltage across

(a) the resistor VR .

(b) the capacitor VC .
(c) the inductor VL .
(d) both the capacitor and the inductor together VCL .

6.5.2 Problem 2 : Bandwidth and quality factor of RLC

Band Pass
A series RLC resonance circuit may be used to create a band-pass filter. Use the
components from the problem above.

1. Change the circuit in a way that you get a band pass!

2. Find the resonance frequency of the circuit using the oscilloscope.

There are two methods, either using the amplitude or the phase. Observing the
phase is usually the more accurate way. Again you have the choice depending
on how you operate the oscilloscope. For the resonance case best way is using
the XY-Mode and Lissajou figures.

ˆ What is the phase shift at resonance?

ˆ What is the shape of the Lissajou figure at resonace?

Set the Oscilloscope to XY-Mode and change the frequency at the generator
until you get the right display. At resonance switch back to Xt-Mode, adjust
to best resolution and take a hardcopy.

3. Find the upper and lower −3dB frequencies to determine the bandwidth of the
band-pass filter. Use the oscilloscope Measure function to align amplitude
and/ or phase to the right values. Take hardcopies at both cutoff frequencies!

34
7. Experiment 3 :
Fourier Series and Fourier Transform
7.1 Introduction to the experiment
7.1.1 Objectives of the experiment
The goal of the second experiment of the Signals and Systems Lab is to study differ-
ent signals in terms of their Fourier coefficients and to get a deeper understanding
of the Fourier transform. The handout will provide the basic theory and describes
the various variables and concepts involved. A more detailed description of the the-
ory can be found in reference [1]. The prelab and the experimental procedure will
concentrate on the simulation and implementation of the Fast Fourier Transform
(FFT) rather than the detailed mathematical description. The experimental and
the simulation results will be compared and differences will be discussed.

7.1.2 Introduction
A signal can be represented in the time domain or in the frequency domain. The fre-
quency domain representation is also called the spectrum of the signal. The Fourier
analysis is the technique that is used to decompose the signal into its constituent
sinusoidal waves, i.e. any time-varying signal can be constructed by superimposing
sinusoidal waves of appropriate frequency, amplitude, and phase. The knowledge of
the frequency content of a signal can be very useful. For example, the frequency
content of human speech can be filtered, the quality of transmitted signals can be
improved and noise can be removed. The Fourier transform is used to transform a
signal from the time domain to the frequency domain. For certain signals, Fourier
transform can be performed analytically with calculus. For arbitrary signals, the
signal must first be digitized, and a Discrete Fourier Transform (DFT) is performed.
On the other hand, the inverse Fourier transform is used to transform a signal from
the frequency domain to the time domain. The handout is divided into two parts.
The first part introduces Fourier series representation and Fourier transform for
periodic continuous-time signals. The second part describes Fourier series represen-
tation and Fourier transform for periodic discrete-time signals.

35
7.1.3 Part I: Continuous time signals
A. Fourier series representation
A continuous-time periodic signal can be described by the sum of basic signals,
i.e. the sum of sine or cosine waves.

Periodic signals
A signal is defined as periodic, if for some positive value of T , the signal can be
described by Eq. (7.1),

x(t) = x(t + nT ) (7.1)

This must hold for all t. The fundamental period is the minimum positive, nonzero
value of T for which the above equation is satisfied. The value ω0 = 2π/T is referred
to as the fundamental frequency.

Determination of Fourier series coefficients

Given a function x(t), its Fourier series coefficients ck can be obtained by using the
following equation.
Z
1
ck = x(t)e−jkω0 t dt (7.2)
T T

Thus, x(t) can be expressed in terms of ck as follows,

+∞
X
x(t) = ck ejkω0 t (7.3)
k=−∞

On substituting k = 0 into Eq. (7.2), the DC or the constant component of the

signal x(t) is obtained,
Z
1
c0 = x(t)dt (7.4)
T T

So far we expressed the signal x(t) as a sum of superimposed complex exponential

functions. However, a number of other ways can be used to represent x(t). As
example, Eq. (7.3) can be rewritten as,
∞
a0 X h i
x(t) = + Ak cos (kω0 t) + Bk sin (kω0 t) (7.5)
2 k=1

In such a case, each continuous time periodic function can be described by the sum
of superimposing sine and cosine functions. The required Fourier coefficients are a0
(DC component), Ak and Bk .
In the following section, the Fourier series coefficients and the Fourier series for a
continuous time periodic square wave will be obtained analytically.

36
Continuous time periodic square wave
A continuous time periodic square wave is given in Fig. 7.1 with a period T and a
pulse width of 2T1 .

y = x(t)

T 2  T1 T1 T T t
2

Figure 7.1: Continuous periodic square wave

The signal is described as follows:

(
1, |t| ⩽ T1
x(t) = (7.6)
0, T1 < |t| < T /2

The Fourier series coefficients are obtained using,

Z
1
ck = x(t)e−kω0 t dt (7.7)
T T
To calculate the constant c0 , use k = 0 and perform the integration over any interval
of length T ,

1 T /2 1 T /4
Z Z  
1 T 1
c0 = x(t) dt = 1 dt = = (7.8)
T −T /2 T −T /4 T 2 2

Due of the symmetry of x(t) about t = 0, it is suitable to integrate from −T /2 ⩽

t < T /2. The signal is only equal to 1 in the range of −T /4 ⩽ t < T /4, so that
Eq. (7.7) results to,

1 T1
Z Z
1 −jkω0 t
ck = x(t)e dt = x(t)e−jkω0 t dt
T T T −T1
T1
ejkω0 T1 − e−jkω0 T1
 
1 2
=− e−jkω0 t = (7.9)
jkω0 T −T1 kω0 T 2j

Making use of the trigonometric identity,

ejx − e−jx
sin (x) = (7.10)
2j
Eq. (7.9) becomes
2 1
ck = sin (kω0 T1 ) = sin (kω0 T1 ) (7.11)
kω0 T kπ
where, T is the period of the signal and T1 is the width of the periodic pulses.

37
For more demonstration, Matlab was used to plot the scaled Fourier coefficient T ak
given by Eq. (7.11) for k = −50 to k = 50 . The width of the pulses were kept
constant at T 1, and the period of the signal was varied from T = 4T1 , T = 8T1 , and
T = 20T1 as shown in Fig. 7.2, Fig. 7.3 and Fig. 7.4, respectively.

Scaled Fourier Series Coefficients Tak for x(t) Scaled Fourier Series Coefficients Tak for x(t)
0.5 0.25

0.4 0.2

0.3 0.15
Amplitude

Amplitude
0.2 0.1

0.1 0.05

0 0

-0.1 -0.05

-0.2 -0.1
-50 0 50 -50 0 50
Harmonic Number Harmonic Number

Figure 7.2: Scaled Fourier coefficients Figure 7.3: Scaled Fourier coefficients
T ak for T1 fixed and T = 4T1 T ak for T1 fixed and T = 8T1

Scaled Fourier Series Coefficients Tak for x(t)

0.1

0.08

0.06
Amplitude

0.04

0.02

-0.02

-0.04
-50 0 50
Harmonic Number

Figure 7.4: Scaled Fourier coefficients T ak for T1 fixed and T = 20T1

Now, we will continue to describe x(t) in terms of sine and cosine functions as
described by Eq. (7.5). The coefficient a0 becomes a0 = 2c0 . The coefficients Ak
and Bk can be determined by

Ak = 2 ℜe {ck } (7.12)
Bk = −2 ℑm {ck } (7.13)

Using Eq. (7.11), for T = 4T1 .

(
2
2 π k = 1, 3, 5, ...(odd)
Ak = sin (k ) = kπ (7.14)
kπ 2 0 k = 2, 4, 6, ... (even)

38
 Bk = 0 (7.15)

Substituting into Eq. (7.5),

∞
1 2 X π cos (kω0 t)
x(t) = + sin (k ) ∗ (7.16)
2 π k=1,3,5,...
2 k

For more demonstration, Matlab was used to plot the first 3, 50, 500 harmonics of
a square wave given in Eq. (7.16) and the sum of the harmonics. The time axis is
normalized. Time t = 1 corresponds to t = T .

Cosine Oscillations
1.5

1
Amplitude

0.5

-0.5

-1
0 0.2 0.4 0.6 0.8 1
Time t/T

Figure 7.5: The first 3 harmonics Figure 7.6: The first 50 harmonics

Figure 7.7: The first 500 harmonics

A comparison of Fig. 7.6 and Fig. 7.7 indicates that the shape of the reconstructed
square wave can already be recognized after the summation of the first 50 harmon-
ics. However, the reconstructed signal exhibits a lot of ’ringing’ at each step change
in the square wave, i.e. the Fourier series exhibits a peak followed by rapid oscil-
lations. The phenomenon is called Gibbs effect. With increased number of terms,

39
e.g.1000 harmonics, reconstructed signal is getting closer to the original signal. Gen-
erally, this phenomenon is due to the discontinuities in the square wave and many
high frequency components are required to construct the signal accurately. More
information about Gibbs effect can be found in reference [1].

B. Continuous time Fourier transform

The continuous time Fourier transform is a generalization of the Fourier series. It
can be also called Fourier series representation for continuous time aperiodic signals.
It only applies to continuous time aperiodic signals. The Fourier transform of a given
signal x(t) is defined as
Z ∞
X(jω) = x(t)e−jωt dt (7.17)
∞

Using the inverse Fourier Transform the original signal can be obtained using the
following equation,
Z ∞
1
x(t) = X(jω)ejωt dω (7.18)
2π ∞
As an example, we will calculate the Fourier transform of the square pulse shown in
Fig. 7.8.

y = x(t)

τ t

Figure 7.8: Square pulse with a pulse width of τ

The signal in Fig. 7.8 is 0 everywhere except in the range −τ /2 ⩽ t < τ /2, where
the signal is equal to 1. After calculating the Fourier transform we get:
Z τ /2 τ /2
e−jωt − e−jωτ /2
 jωτ /2 
−jωt e
X(jω) = e dt = = (7.19)
−τ /2 jω −τ /2 jω

Again making use of the trigonometric identity (Eq. (7.10)),

sin ( ωτ
 
2
)  ωτ 
X(jω) = τ ∗ ωτ = τ sinc (7.20)
2
2

NOTE

Fourier declared that an aperiodic signal could be viewed as a periodic signal with
an infinite period.
As an example, we studied the Fourier series of a square wave and the Fourier
transform of a rectangular pulse. As the period becomes infinite, the periodic square
wave approaches the Fourier transform of the rectangular pulse.

40
7.1.4 Part II: Discrete time signals
A. Fourier series representation
In this part, we will discuss the Fourier series representation for discrete time signals.
The discussion will closely follow the discussion in the first part.

Periodic signals
A signal is defined as periodic, with period N if,
x[n] = x[n + N ] (7.21)
This must hold for all n. The fundamental period is the smallest positive integer
N for which the above equation holds. The parameter ω0 = 2π
N
is referred to as the
fundamental frequency.

Determination of Fourier series coefficients

Given a function x[n], its Fourier series coefficients ak can be obtained using the
following equation.
1 X
ak = a[n]e−jkω0 n (7.22)
N
n=(N )

Thus x[n] can be expressed in terms of ak as follows,

1 X
x[n] = ak ejkω0 n (7.23)
N
n=(N )

In the following section, the Fourier series coefficients for a discrete time periodic
square wave will be obtained analytically.

Discrete time periodic square wave

Given a discrete time periodic square wave as shown in Fig. 7.9 with a period N
and a pulse width of 2N1 . The signal can be described as follows:
x[n] = 1 for − N1 ⩽ n ⩽ N1 (7.24)

-N -N1 0 N1 N n

Figure 7.9: Discrete periodic square wave with a period of N and a pulse width of
2N1

The Fourier series coefficients can be obtained using

N1
1 X
ak = e−jk(2π/N )n (7.25)
N n=−N
1

41
where, m = n + N1 , so that the Fourier coefficients can be described as

2N 2N1
1 X1 −jk(2π/N )(m−N1 ) 1 X
ak = e = ejk(2π/N )N1 e−jk(2π/N )m (7.26)
N m=0 N m=0

The summation term is a geometric series. On using the equation for the sum of a
geometric series

−1
N
(
X
n N α=1
ak = α = 1−αn
(7.27)
n=0 1−α
α ̸= 1

The following equation is obtained

1 − e−jk2π(2N1 +1)/N
 
1
ak = e−jk(2π/N )N1
N 1 − e−jk(2π/N )

!
1 e−jk(2π/2N ) ejk2π(N1 +1/2)/N − e−jk2π(N1 +1/2)/N
= (7.28)
N e−jk(2π/2N ) (ejk(2π/2N ) − e−jk(2π/2N ) )

The equation can be rewritten as

h i
2πk(N1 +1/2)

1 sin N
N π/k k ̸= 0, ±N, ±2N, ...
ak = sin N ( ) (7.29)
 2N1 +1
N
k = 0, ±N, ±2N, ...

The scaled Fourier series coefficients N ak are plotted in Fig. 7.10 for 2N1 + 1 = 5
and N = 40.

Fourier Series Coefficients Nak for x(n)

0.14

0.12

0.1

0.08
Amplitude

0.06

0.04

0.02

-0.02

-0.04
-80 -60 -40 -20 0 20 40 60 80
Harmonic Number

Figure 7.10: Fourier series coefficients of the discrete periodic square wave

42
 NOTE

There are some important differences between Fourier series representation for con-
tinuous time periodic signals and Fourier series representation for discrete time peri-
odic signals. The Fourier series representation for discrete time periodic signals is a
finite series, while Fourier series representation for continuous time periodic signals
is infinite series. Also, an important property that must be noted is that discrete
Fourier series coefficients are periodic with period N, i.e.

ak = ak+N (7.30)

Compare Fig. 7.4 (T = 20T1 ) and Fig. 7.10 (N = 20N1 ) and you can notice the two
differences clearly.

B. Discrete time Fourier transform

Computers and other digital electronic based systems cannot handle continuous time
signals. These systems can only process discrete data. Therefore, a discrete form
of the Fourier Transform, i.e. a numerical computation of the Fourier transform,
is needed to give us spectral analysis of discrete signals. This transform is called
discrete Fourier transform.
The discrete Fourier transform converts a time domain sequence resulting from
sampling a continuous time signal into an equivalent frequency domain sequence
representing the frequency content of the given signal. Thus, the DFT is given by
the following equation.
N
X −1
X[k] = x[n]e−j2πkn/N k = 0, 1, 2, ..., N − 1 (7.31)
n=0

The Inverse Discrete Fourier Transform (IDFT) performs the reverse operation and
converts a frequency domain sequence into an equivalent time domain sequence
N −1
1 X
x[n] = X[k]e−j2πkn/N n = 0, 1, 2, ..., N − 1 (7.32)
N k=0

The DFT plays an important role in many applications of digital signal processing
including linear filtering, correlation analysis and spectrum analysis.
The computation of the DFT is computationally expensive as the DFT computes
the sequence X[k] of N complex valued numbers given another sequence of data x[n]
of length N . From the equations for the DFT, it can be seen that to compute all
N values N 2 complex multiplications and N 2 − N complex additions are required.
The direct computation using the DFT is inefficient because it does not exploit the
symmetry and periodicity properties. Thus a number of algorithms exist that makes
these computations more efficient. An important algorithm for computer the DFT
is the Fast Fourier Transform (FFT).
The FFT algorithms exploit these two basic properties and make the computation
more efficient.

43
 NOTE
It should be clear that FFT is not an approximation of the DFT. It yield the same
result as the DFT with fewer computations required.

In MATLAB, the FFT is performed using the function (fft). The output of the fft()
function by itself is a vector of complex numbers. The following formula returns
a vector of the magnitudes of each of the frequencie’s contributions to the signal’s
amplitude.
abs(fft data)
y =2∗ (7.33)
length(data)
To achieve the absolute value of the complex magnitude we need the abs() function.
Since the fft() returns the complex amplitudes scaled by the overall length of the
data, we need to divide by length of the data. Finally the equation has to be
multiplied by 2 (because of Euler’s Relation!?). Only the first half of the vector y
contains relevant data, so
 
length(data)
y=y 1: (7.34)
2
The highest frequency that can be perceived in a signal is given by the Nyquist
Frequency:
Fs
fnyqu = where Fs is the sampling frequency (7.35)
2
The frequencies that correspond to the y vector range from 0 Hz to the Nyquist
Frequency can be generated by:

f = linspace (0, fnyqu , length(y)) (7.36)

Finally the plot command ’plot(f, y)’ shows the frequency components from the y
vector. A detailed description of fft() function can be found along with examples in
the Matlab documentation [5].

7.1.5 Definitions and practical hints

The sampling process is the transfer of a continuous time signal into a discrete
time signal or the transfer from the world of analog signal processing to the world of
digital signal processing. In practice, most signal processing is performed on discrete
time signals and not on continuous time signals. This applies despite the fact that
most of the signals encountered in science and engineering are analog in nature. In
the next experiment, the sampling theory and sampling techniques will be discussed
in more details.

Continuous Fourier transform

The Continuous Fourier transform is used to transform a continuous time signal into
the frequency domain. It describes the continuous spectrum of a non-periodic time
signal.

44
Discrete Fourier transform
Is used in the case where both the time and the frequency variables are discrete.

Fast Fourier transform

Is a special algorithm which implements the discrete Fourier transform with con-
siderable savings in computational time. It must be clear that the FFT is not a
different transform from the DFT, but rather just a means of computing the DFT
with a considerable reduction in the number of calculations required.

FFT using the oscilloscope

As part of the experiment, the FFT has to be obtained using the oscilloscope. The
following steps should be carried out. First the time domain signal should be set
properly by the following steps:
a. Press AUTOSET to display the time domain waveform.
b. Position the time domain waveform vertically at center of the screen to get
the ”true” DC value.
c. Position the time domain waveform horizontally so that the signal of interest
is contained in the center eight divisions.
d. Set the YT waveform SEC/DIV timebase to provide the resolution you want
in the FFT waveform. This decides how high the frequency content of the
transform will be. The smaller the time scale used the higher the frequency
content.
Then the FFT can be performed as follows:
a. Locate the FFT button on the front panel of the oscilloscope. On pressing it
the oscilloscope will change into FFT mode and shows the FFT menu.
c. In the menue choose the right source channel. A similar picture like below will
appear dependant on the signal .

1 1. Frequency value at center


position of the screen.

2. Vertical scale in dB/division.

Magnitude is referenced to
0dB, where 0 dB equals
1 VRM S .

3. Horizontal scale in
frequency/division

4. Sample rate in number of

  
samples/sec
2 3 4 5
5. FFT window type.
Figure 7.11: Screen hard copy

45
The Fourier series represents a periodic waveform as an infinite series of harmonically
related sinusoids. Since the Fourier series contains only discrete frequencies, each
sinusoidal component of the waveform is represented by a vertical line on a plot
of the signal magnitude versus frequency. The height of the line represents the
magnitude of the contribution from that particular frequency. The location of the
line along the horizontal axis identifies its frequency.
For all FFT’s in this experiment always use the Hanning window, as it is best suited
among the given options for a periodic signal. A windowing function is basically a
function used to cut out a part of the signal in time domain so that an FFT can be
carried out on it. Thus basically it is multiplication by some kind of rectangular pulse
in time domain, which implies convolution with some kind of a sinc in frequency
domain. Using a windowing function affects the transform but is the only practical
method for obtaining it.
Once the time domain signal has been set up as discussed previously and you are in
the FFT screen. The following steps must be carried out:

a. Bring the region of the frequency domain that you are interested in towards
the middle of the screen.

b. Adjust the FFT zoom button in order to zoom into the transform sufficiently
till you reach the magnitude of frequency that you desire. At this point you
should normally make your hardcopy, as it is where you would be able to see
the transform most clearly.

7.1.6 References
1. A. V. Oppenheim, A. S. Willsky, S. H. Nawab, ”Signals and Systems”, Prentice
Hall, Second Edition (1997).

2. Raymond A. DeCarlo, Pen-Min Lin, ”Linear Circuit Analysis”, Oxford Uni-

versity Press, Second Edition (2001).

3. J. G. Proakis, D. G. Manolakis, ”Digital Signal Processing”, Prentice Hall,

Third Edition (2002).

4. Sarma, M.S., ”Introduction to Electrical Engineering”, Oxford University Press,

2001.

5. MATLAB Documentation.

6. TDS220 manual.

7. TDS200-Series Extension Modules manual.

46
7.2 Prelab Fourier Series and fourier Transform
7.2.1 Problem 1 : Decibels
In the lab, you must be able to express the signal amplitude in Vpp and Vrms , also
you have to know what dBVrms corresponds to.
1. Given x(t) = 5 cos (2π1000t) ,
a. what is the signal amplitude and the Vpp voltage?
b. what is the Vrms value of the provided signal?
c. what is the amplitude of the spectral peak in dBVrms ?
2. For a square wave of 1 Vpp the voltage changes between −0.5 V and 0.5 V,
a. what is the signal amplitude in Vrms ?
b. what is the amplitude in dBVrms ?

7.2.2 Problem 2 : Determination of Fourier series coeffi-

cients
1. Determine the Fourier series coefficients up to the 5th harmonic of the function
f (t) = 4 t2 − 0.5 < t < 0.5

2. Use MatLab to plot the original function and the inverse Fourier transform.
Put both graphs into the same diagram.

7.2.3 Problem 3 : FFT of a Square/Rectangular Wave

Write a Matlab script:
1. Generate a square wave of 1 ms period, 2 Vpp amplitude, no offset, and duty cy-
cle 50% (hint : use ’square’ function). Use 200 kHz as the sampling frequency
for the problem.
2. Plot the square wave in time domain.
3. Obtain the FFT spectrum using Matlab FFT function. Make the FFT length
to be the length of the square wave data vector.
4. Plot the single-sided amplitude spectrum in dBVrms .
5. Plot the spectrum including only the first four harmonics in dBVrms .
Hint: Use the Matlab command ’xlim’.
6. Repeat the previous steps using 20% and 33% duty cycles, respectively. Keep
period and amplitude constant.
7. Discuss the changes for smaller pulse width. Use Eq. (7.11) to prove your
statement.
Hint: Use the subplot command to plot the spectrum magnitude for the
three cases 50%, 33% and 20% duty cycle to ease the comparison.

47
7.2.4 Problem 4 : FFT of a sound sample
The Matlab command ’[y,Fs] = audioread(filename)’ reads a wave file specified by
the string ’filename’, returning the sampled data in ’y’ and the sample rate in ’Fs’
in Hertz.

1. Download the ’Soundfile for Prelab question ’FFT of a sound sample”

from the course webpage.

2. Using Matlab, read the sound file and plot the first 10 ms of the signal.

3. Use the Matlab FFT function to compute the spectrum and plot the single-
sided amplitude spectrum in dBVrms .

4. What are the tones forming this signal?

48
7.3 Execution Fourier Series and fourier Trans-
form
7.3.1 Problem 1 : FFT of Single Tone sinusoidal wave
1. Use the function generator to generate a sinusoidal wave having 500 Hz fre-
quency, 2 Vpp amplitude and no offset. Use the measure function to verify all
properties. Take a hard copy in time domain.
2. Obtain the FFT spectrum using the oscilloscope FFT function. Use the cursor
to measure the properties. Take hard copies of the complete spectra and the
zoomed spectra peak.
3. Generate a sinusoidal wave having 0 dB spectrum peak, 2 KHz frequency, with-
out a dc offset. What is the amplitude value? Use the measure function and
the cursors. Take hard copies of time and frequency domain.

7.3.2 Problem 2 : FFT of square wave

1. Use the function generator to generate a square wave having 1 ms period, 2 Vpp
amplitude, and no offset. Check the properties with the measure function.
Take a hard copy.
2. Obtain the FFT spectrum. Instead of using the time base (sec/div) control
to accurately measure the frequency components, use the FFT zoom control
that provides a zoom factor up to 10 and use the cursors to determine the
amplitudes and the frequency of the fundamental and the first four harmonics.
Take hard copies of the FFT signal.
3. Obtain the FFT spectrum for 20 % duty cycles. Determine the amplitudes of
the fundamental frequency and the first four harmonics. Take hardcopies of
the signal in time and frequency domains.

7.3.3 Problem 2 : FFT of Multiple-Tone sinusoidal wave

1. Combine the signal from the sine output of the auxiliary signal generator and
a 2 Vpp , 10 KHz sinusoidal wave from the Agilent signal generator. Use the
following circuit.
R1
10K0

R2
10K0
+10V sine Uout

Sine 2Vpp Auxiliary Function

10KHz ~ 10V
Generator R3
100K
Gnd square

2. Take a hard copy of the signal in time domain.

3. Take a hard copy of the FFT spectrum of the signal.

49
7.4 Evaluation
7.4.1 Problem 1 : FFT of Single Tone sinusoidal wave
In the lab report:

1. What is the reference value of the oscilloscope for 0dB.

2. Use Matlab to calculate the expected FFT spectra for the parameters given in
part 7.3.1.1. Is the calculated spectra consistent with the measured spectra?

3. Use Matlab to calculate the expected FFT spectra for the parameters given in
part 7.3.1.3. Is the calculated spectra consistent with the measured spectra?

4. Compare the results from Matlab with the measured values. Discuss the dif-
ferences.

7.4.2 Problem 2 : FFT of square wave

In the lab report:

1. For frequency domain measurements, the frequency scale needs to be expanded

in order to accurately measure the frequency components. This could be done
with the time base (sec/div) control. What is the effect of doing this on the
measured bandwidth? Information can be found in reference [6] and [7].

2. Use the hardcopies taken to discuss the effect of changing the duty cycle on
the FFT results.

7.4.3 Problem 3 : FFT of square wave

In the lab report:
Use the hardcopy of the spectrum and discuss the linearity of the FFT.

50
8. Experiment 4 : Sampling
8.1 Introduction to the experiment
8.1.1 Objectives of the experiment
The goal of this experiment is to introduce the basics of sampling and to implement
different sampling circuits.
The foundations of sampling will be discussed on the signals and systems and on
the component level. Different sampling schemes like impulse train sampling, rect-
angular pulses sampling and sample and hold will be introduced. In addition, the
consequence of the different sampling schemes on the reconstructed signals will be
described.
Furthermore, two different types of sampling circuits will be discussed. The first
sampling circuit is the sampling bridge used for high-speed sampling applications,
e.g. as part of a digital oscilloscope. The second sampling circuit is a sampling
circuit based on Metal Oxide Semiconductor Field Effect Transistors (MOSFETs).

8.1.2 Introduction
Most of the signals in nature exist in an analog form. Sampling is the transfer of
a continuous time signal into a discrete time signal. Sampling is the first and a
very important step to provide signals, which can be digitally processed. Therefore,
sampling is the connecting element between the world of analog and digital signal
processing.
The handout is divided into three parts, where the first part of the handout in-
troduces the sampling theory and discusses the difference between sampling and
quantization of signals. The second part will explain different ways of sampling like
impulse train sampling, rectangular pulses sampling and sample and hold schemes.
Finally, different implementations of sampling circuits will be discussed.

8.1.3 Sampling Theory

Sampling step versus quantization step
Sampling is the transfer of a continuous time signal into a discrete time signal.
However, sampling should be clearly distinguished from the quantization of a signal.
Both steps are necessary to carry out an analog-to-digital (A/D) conversion.
Quantization is the transformation of a continuous signal into a discrete signal in
terms of discrete amplitudes or discrete signal levels. The quantization leads to
discrete values for the amplitude, whereas the sampling leads to values of discrete
times. The difference between sampling and quantization is illustrated in Fig. 8.1. It
provides a schematic description of an A/D conversion process. In order to transfer
continuous time signals into a stream of bits, the signal can be first quantized and
afterwards sampled or the signal can be first sampled and quantized afterwards.

51
In both cases, we get a signal that is discrete in terms of time and amplitude. After
the quantization and the sampling the signal is coded, meaning
Signals theLab,
and Systems signal is trans-
Fall 2010, Jacobs University
formed in a stream of bits, which can be processed or transmitted. After carrying

Continuous-time signal

x [t]
(analog signal)
Quantization Sampling

Discrete time or
amplitude signal
x [t]

x [t]
(digital signal)

t t

Sampling Quantization Discrete time and

x [t]

amplitude signal
(digital signal)

t Coding
x [t]

Data stream
(digital signal)

Figure Fig.
8.1: 1:
Schematic illustration
Schematic of of
illustration anan
A/D
A/Dconversion
conversionprocess using
process a sampling,
using a sampling,
quantization and coding steps. quantization and coding steps.

outAfter carrying
certain out certain
processing processing
steps the stepsisthe
digital signal verydigital
often signal
convertedis very
backoften
into converted
an
backsignal.
analog into anUnder
analog signal.
certain Under certainit circumstances,
circumstances, it is possible
is possible to completely to completely
recover the
recover
initial the This
signal. initialvery
signal. This very
important important
property property
follows follows the
the sampling sampling
theorem. theorem.
This
theorem is simple, but very important and useful, because based on the sampling on the
This theorem is simple, but very important and useful, because based
sampling
theorem theorem
we can decidewe can decide
whether whether
a signal can be acompletely
signal can be completely
recovered or not. recovered or
not.
The sampling theorem
The sampling theorem
We assume that x(t) is a band-pass limited signal which has a Fourier Transform
We assume
X(jω) that x ( t ) is
= 0 for frequencies a band-pass
larger limited cut-off
than a maximum signal frequency
which hasωM a .Fourier Transform
The initial
X(jω)=0
signal can be forreconstructed
frequencies from largerthethan a maximum
sampled cut-off
signal if the frequency
sampling frequencyωM.ωSThe
is initial
twosignal
timescan be than
larger reconstructed
the maximum fromcut-off
the sampled signal
frequency if the
of the bandsampling frequency ωS is
pass filter.
two times larger than the maximum cut-off frequency of the band pass filter.
ωS > 2ωM (8.1)
ωs > 2ω M (1)
The sampling frequency ωS can be described by
The sampling
2π frequency ωS can be described by
ωS = (8.2)
2π
T
ωs = (2)
where T Tis the period of an impulse train sampling signal. If the sampling frequency
is where
smallerTthan is the2 period
times theof anmaximum
impulse train sampling
frequency signal.
of the band-pass limited signal,
theIf initial signal cannot
the sampling be completely
frequency is smaller reconstructed
than 2 times afterwards.
the maximumIn thisfrequency
case, the of the
band-pass limited signal, the initial signal cannot be completely reconstructed
afterwards. In this case, the sample52 rate is not high enough and the term
under-sampling or aliasing is used. If the sampling frequency is exactly equal to 2
times the maximum frequency of the band-pass limited signal, then we speak about
 Signals and Systems Lab, Fall 2010, Jacobs University

the Nyquist frequency or the Nyquist rate. The sampling frequency has to be higher
than the Nyquist
sample rate.high
rate is not Otherwise, the signal
enough and cannot
the term be reconstructed.
under-sampling or aliasing is used. If
the sampling
Remark: Samplingfrequency
is notisonlyexactly equal towhen
important 2 times
we the
dealmaximum frequency
with voltages of the
or currents.
band-pass
There limited other
are several signal,areas,
then wewhere
speak about
we havethe Nyquist
to deal frequency or the Nyquist
with sampling. Take for
rate. The sampling frequency has to be higher than the
example the area of digital photography or digital image processing. The Nyquist rate. Otherwise,
transfer of
an the signaltocannot
analog be reconstructed.
a digital picture requires a sampling step. The original picture is
Remark:
sampled for Sampling
example by is not only important
a digital camera andwhenthe we sampling
deal with voltages
signal isorincurrents.
this case
defined by the size of the pixel of your camera in combination with the optics of for
There are several other areas, where we have to deal with sampling. Take your
exampleSo,
camera. theif area of digital about
you complain photography or digitalofimage
the resolution processing.
your digital camera The transfer
you already
made
of anyour experience
analog with picture
to a digital the sampling theorem.
requires a sampling step. The original picture is
sampled for example by a digital camera and the sampling signal is in this case
defined by the size of the pixel of your camera in combination with the optics of
your camera. So, if you complain about the resolution of your digital camera you
already made your experience with the sampling theorem.
Part II:
8.1.4 Sampling Methods
1. Impulse Train sampling (Ideal sampling)
1. Impulse Train sampling (Ideal sampling)
We will first discuss the ideal sampling, where a periodic series of unit impulses is
used
Weas thefirst
will sampling
discuss signal. The
the ideal samplingwhere
sampling, scheme is therefore
a periodic series called
of unitImpulse
impulsesTrain
is
sampling. The schematic sampling procedure is shown in figure (2).
used as the sampling signal. The sampling scheme is therefore called Impulse Train
sampling. The schematic sampling procedure is shown in Fig. 8.2.

Figure
Fig. 8.2: Schematic
2: Schematic illustration
illustration of Impulse
of Impulse TrainTrain sampling
sampling [1]. Top:
[1]. Top: Continuous
Continuous time
time signal, Middle: Impulse train, Bottom: Sampled signal
signal, Middle: Impulse train, Bottom: Sampled signal

53
6
The continuous signal x(t) is sampled by the signals p(t), which represents the
impulse train signal. The periodic impulse train p(t) is described by

∞
X
p(t) = δ(t − nT ) (8.3)
n=−∞

The sampling signal is multiplied with the input signal x(t). After multiplication,
we get the signal xp (t). The index p indicates that an impulse samples the signal.

∞
X
xp (t) = x(t) δ(t − nT ) (8.4)
n=−∞

We can rewrite Eq. (8.4), so that we get the following equation

∞
X
xp (t) = x(nT ) · δ(t − nT ) (8.5)
n=−∞

The sampled signal xp (t) is illustrated in Fig. 8.2. Further information regarding
Impulse Train sampling can be found in chapter 7 of reference [1].
Now, let us discuss how the impulse train sampled signal can be reconstructed as
shown in Fig. 8.3. We still assume that the initial band-pass limited signal was
sampled by an impulse train and that the sampling frequency was higher than 2
times the maximum frequency of the band-pass limited input signal. The output
signal xp (t) in the frequency domain can be described by

∞
X n 
XP (f ) = X(f ) δ f− (8.6)
n=−∞
T

The initial signal is convolved with an impulse train in the frequency domain. It is
assumed that the input signal in the frequency domain corresponds to a triangle.
It can be seen that the signal is reproduced at integer multiples of the sampling
frequency. The output signal Xp (jω) of the sampling circuit is shown in Fig. 8.3c.
The input signal can be recovered if the signal XP (jω) is filtered by a low-pass filter
with a gain of T and a cut-off frequency greater than ωM and less than ωs − ωM to
cut-off the redundant part of the signal as indicated in Fig. 8.3d. The final output
signal, which is a perfect reconstruction of the input signal is shown in Fig. 8.3e.

54
 
p (t )    (t  nT )
n  

x p (t )
x(t ) H ( j ) xr (t )

a 
X ( j ) X p ( j )

 M M   S  M M S 

b  c 
H ( j ) X r ( j )
M  c  (s  M )

 c c   M M 

d  e 

Figure 8.3: Recovery of a continuous-time signal from its samples using an ideal low
pass filter [1].

2. Rectangular Pulses sampling (Real sampling)

So far, we discussed the sampling of a continuous signal by using an impulse train.
However, in practice it is difficult to generate and transmit unit impulses. Therefore,
it is very convenient to generate rectangular pulses. The sampling signal p(t) in
Fig. 8.2 can be therefore described by
∞  
X t − nT
p(t) = rect (8.7)
n=−∞
T0

where T0 the width of the rectangular pulse which is used as the sampling signal.
Consequently, the output signal is given by:
∞  
X t − nT
xp (t) = x(nT ) · rect (8.8)
n=−∞
T0

In practice, it is common to use a first order sample and hold scheme rather than a
Rectangular Pulses sampling scheme.

55
3. First order Sample and Hold

In such a case, the sampling signal is held constant for a certain period. For example,
the sampling signal is held constant until the next sample is taken or the signal is
held constant for a shorter period.

x(t)

p(t)
(a)

T 2T 3T 4T 5T 6T t
=T0

x(t)

p(t)
(b)

T0 T 2T 3T 4T 5T 6T t

Figure 8.4: A hold function (a) the value is held until the next sample is taken. (b)
held for half of the period of the sampling signal [3].

Is the value held for half the sampling period the sampling signal is given by

∞  
X t − T0 /2 − nT
p(t) = rect (8.9)
n=−∞
T0

where T0 = T . Consequently, the output signal is given by

∞  
X t − T0 /2 − nT
xp,SH (t) = x(nT ) · rect (8.10)
n=−∞
T0

The expression can be rewritten by

" ∞
#  
X t − T0 /2
xp,SH (t) = x(nT ) · δ(t − nT ) · rect
n=−∞
T0
" ∞
#    
X t T0
= x(t) δ(t − nT ) · rect ·δ t− (8.11)
n=−∞
T0 2

The sample and hold procedure can be described by an impulse train sampling
step, where the output signal of the impulse train sampling step is convolved with
a rectangular pulse. A schematic implementation of a first order sample and hold
procedure is shown in Fig. 8.5.

56
 where the output signal of the impulse train sampling step is convolved with a
rectangular pulse.
A schematic implementation of a first order sample and hold procedure is shown in
figure (5).

First order sample and hold

 T 
t − 0 
∞ rect  2 
xp,i (t ) = x(t ) ⋅ ∑δ (t − n ⋅ T )  T 0 
n=−∞  
xp,SH (t )
×
x(t )
∞
p(t ) = ∑δ (t − n ⋅ T )
n=−∞
 T0 
 ∞
 t − 
xp,SH (t ) = x(t ) ⋅ ∑δ (t − nT ) ∗ rect 2 
 n=−∞   T0 
 

Fig. 5: Description of a first order sample and hold implementation. The first order
Figure 8.5: Description of a first order sample and hold implementation. The first
sample and hold implementation can be described by an ideal or Impulse-train
order sample and hold implementation can be described by an ideal or Impulse-train
sampled signal, which is convolved with a rectangular signal.
sampled signal, which is convolved with a rectangular signal.
A simple low-pass filter with a constant gain cannot be used to reconstruct the initial
signal x ( t ) from x p,SH ( t ) . The Fourier transform of equation (11) explains why the
initial signal x ( t ) cannot be completely reconstructed after being sampled and held.
A simple low-pass filter with a constant gain cannot be used to reconstruct the
initial signal x(t) from∞xp.SH(t). The Fourier transform of Eq. (8.10) explains why
 n  T 
X p ,SH (fsignal
the initial ) = X(x(t)
f )∗ ∑ ⋅ sin c(πfT0 )reconstructed
δ f −be completely
cannot ⋅ exp − jπf 0 after being sampled and
(12)
held. n = −∞  T   2 

∞ 10
X n 
Xp,SH = X(f ) δ f− · sinc(πf T0 ) · exp (−jπf T0 /2) (8.12)
n=−∞
T

The first two terms of Eq. (8.12) are identical with Eq. (8.6). However, the con-
volution of the impulse train sampled signal in the time domain corresponds to a
multiplication with a sinc function in the frequency domain and the additional shift
in the time domain leads to a phase shift in the frequency domain.
The problem can be solved by the implementation of a filter, which compensates for
the introduced nonlinearities. Therefore, a filter with the following properties can
be used.

exp (jπf T0 /2)

Hr (f ) = (8.13)
sinc(πf T0 /2)

A summary of the different sampling schemes is given in table (8.1).

57
 H r (f ) = ⎝ 2⎠
(13)
⎛ T0 ⎞
sin c⎜ πf ⎟
⎝ 2⎠
A summary of the different sampling schemes is given in table (1).

(a)
∞
x(t x i (t ) = ∑ x (nT ) ⋅ δ (t − n ⋅ T )
) p(t) n = −∞

(b)
∞
⎛ t − n ⋅T ⎞
x(t x p (t ) = x (t ) ⋅ ∑ rect ⎜⎜ ⎟
⎟
) n = −∞ ⎝ T0 ⎠
p(t)

(c)
⎡ ∞
⎤
x(t x p (t ) = ⎢ x(t ) ⋅ ∑ δ (t − nT )⎥
) p(t) ⎣ n = −∞ ⎦
⎛ t ⎞ ⎛ T0 ⎞
∗ rect ⎜⎜ ⎟⎟ ∗ δ ⎜ t − ⎟
⎝ T0 ⎠ ⎝ 2⎠

Table 8.1: Overview of the different

Table 1: Overview sampling
of the different schemes
sampling (a) (a)
schemes impulse train
impulse train sampling,
sampling,
(b) real or rectangular(b)sampling
real or rectangular
and (c) sampling and hold
sample and (c) sample
[3] and hold [3].

8.1.5 Sampling Circuit Applications

11
Several different sampling circuits are known. Very often, the sampling circuit is
the first stage of an A/D conversion circuits. In the following, we will discuss two
implementations of sampling circuits. Depending on the application of the A/D
conversion circuit, the requirements are different. The most important parameters
are the sampling rate and the resolution.
For the A/D conversion of speech signals the sampling rate can be relatively low,
however, the resolution has to be relatively high. In the case of a digital oscilloscope,
the sampling rate has to be high, but the resolution can be low in comparison to other
A/D converters. The following specifications are important for sampling circuits:

1. The transfer function of the sampling circuit should be close to being ideal. If
the sampling circuit is switched on the attenuation of the signal should be as
low as possible, whereas in the switched off state the attenuation should be as
high as possible.

2. The sampling rate should be as high as possible.

3. The propagation delay between the input and the output side should be as
low as possible.

4. The input and the output side should be decoupled.

58
The sampling bridge
The sampling bridge is used for high-speed sampling. An implementation of a sam-
pling bridge is shown in Fig. 8.6. Generally, two reasons exist why sampling bridges
are applied for high-speed sampling. The diodes are typically realized by Gallium
Arsenide rather than silicon. Electrons ”travel” faster inside the Gallium Arsenide
crystal than inside a silicon crystal so that the switching speed of a Gallium ar-
senide diode is higher. Furthermore, the Gallium Arsenide diodes used for sampling
circuits are Schottky diodes rather then pn-diodes. Schottky diodes usually have a
very small equivalent capacitance, which allows fast switching.
Remark: As part of the lab we will use regular silicon pn-diodes to demonstrate
the working principle of a sampling bridge. The general working principle of the
circuits is not affected by this modification.

R1
+

Vs+
1
D1 D2

3 4

Ri D3 D4 V2 R_L +
2
Vs-
V1
R2
V0 ~

Figure 8.6: High speed Sampling circuit based on a four diodes.

The operating principle of the circuit can be described as follows. The input voltage
is given by the voltage V0 . The resistor Ri is the internal resistance of the input
source. The sampling circuit is formed by the four diodes D1 to D4 , the two resistors
R1 and R2 and the sampling signal Vs + and Vs −. At the same time, positive and
negative sampling pulses are applied to the node (1) and (2). The voltage Vs is
high enough so that the diodes are forward biased. Consequently, the current path
between the node (3) and (4) gets conductive and the signal from the input side is
applied to the load resistor on the output side. In the ideal case, the voltage V2 is
equal to V1 , where V1 is the voltage V0 plus the voltage drop across the resistor Ri .
The following aspects have to be considered while designing a sampling bridge. The
amplitude of the sampling signal Vs has to be higher than the voltage V1 . Otherwise,
not all diodes of the bridge are forward biased. Furthermore, the resistors R1 and
R2 have to be in a certain range. If the resistance for R1 and R2 are too high,
the voltage drop across the diodes is not high enough and the diodes are not under
forward bias conditions. If the resistance is too low, the input signal will not be
applied to the load, because the resistance of the sampling circuit is smaller than
the resistance of the load, so that the current will pass through the sampling circuits.
Furthermore, it is obviously clear that the diodes are non-ideal devices. Therefore,
the amplitude of the sampling voltage has to be at least equal to the diffusion voltage
of the diodes.

59
CMOS sampling circuits
For most applications, the sampling rate does not have to be extremely high. In the
field of speech processing, for example, the sampling rate is in the range of several
1000 samples per second (Ksamples/s). In such cases, CMOS circuits are typically
used. The major advantage of CMOS circuits is that the sampling circuit can be
directly combined with other electronic components on the same chip.
A CMOS sampling circuit is shown in Fig. 8.7. The circuit consists of the inverter
and two pass transistors. The circuit in Fig. 8.7 can be used for bidirectional data
transmission as well as for implementing a sampling circuit. An implementation of
the CMOS switch based on the transistor level is shown in Fig. 8.8. The inverter
can be implemented by two MOS field effect transistors.
VDD

Sampling Sampling
Signal Signal

In Out In VDD Out

VSS VSS

Figure 8.7: Sampling circuit with in- Figure 8.8: Sampling circuit on the
verter and two pass transistors. transistor level.

Bidirectional data transfer means that the circuit can be used for a data transfer
from the input to the output side or a data transfer from the output to the input
side. It is realized by two pass transistors. In order to transfer the data from
the input to the output side or vice versa the gate of the NMOS transistor (arrow
towards the gate) is applied to VDD and the gate voltage of the PMOS transistor
(arrow away from the gate) is applied to Vss . A CMOS inverter provides the inverted
sampling signal. The sampling signal is directly connected to the gate of the NMOS
and the inverted sampling signal is applied to the gate of the PMOS transistor.
Consequently, the voltages applied to both of the gates of the pass transistors are
always inverted to each other. In the off-state the gate of the NMOS transistor is
applied to Vss and the gate of the PMOS transistor is on VDD .
It is important to mention that the sampling circuit in Fig. 8.8 has limitations in
terms of the range of operation. Only signals can be sampled or transferred, which
have voltage levels between VDD and VSS .

8.1.6 Reference
1. A.V. Oppenheim, A.S. Willsky, S.H. Nawab, Signals and Systems, 3rd edition,
Prentice Hall Signal Processing Series (1997).
2. Fairchild Semiconductors, CD4016BC data sheet
3. O. Loffeld, Allgemeine Nachrichtentechnik, University Siegen.

60
8.2 Prelab Sampling
8.2.1 Problem 1: The Sampling Theorem
1. Analog signals are usually passed through a low-pass filter prior to sampling.
Why is this necessary?

2. What is the minimum sampling frequency for a pure sine wave input at 3KHz?
Assume that the signal can be completely reconstructed.

3. What is the Nyquist frequency?

4. What are the resulting frequencies for the following input sinusoids 500Hz,
2.5KHz, 5KHz and 5.5KHz if the signals are sampled by a sampling fre-
quency of 5KHz?

5. Mention three frequencies of signal that alias to a 7Hz signal. The signal is
sampled by a constant 30 Hz sampling frequency.

8.2.2 Problem 2: Impulse Train Sampling and Real Sam-

pling
Consider the sampling shema shown in figure (8.2). The input signal x(t) is given
by a sine function, with an amplitude of 5 V peak and a frequency of 50 Hz. The
sampling signal p(t) is represented by a unity impulse train. Use an overall sampling
rate of 100 k samples/s) for the whole problem.

1. Carry out simulations for the following cases:

(a) Under Sampling (use 48 Hz)

(b) Nyquist Sampling
(c) Over Sampling (use 1000 Hz

Use the command subplot to visualize the continuous signal x(t), the sampling
signal p(t) and the result for each of these cases.

2. The signal x(t) should be sampled by a rectangular pulse train. Modify the
sampling function p(t), so that the width of the sampling pulse is 50% of the
sampling period. Carry out simulations for the following cases:

(a) Under Sampling

(b) Nyquist Sampling
(c) Over Sampling

Use the same sampling rates and the same plot setup as before.

61
8.2.3 Problem 3: Sampling using a Sampling bridge
Modify the circuit in figure (8.6) in such a way that a single sampling source can be
used to sample the input signal.

1. Sketch the modified circuit.

2. Explain the operation of the modified circuit.

62
8.3 Execution Sampling
8.3.1 Problem 1: Digital Sampling Oscilloscope
The easiest way to visualize sampling and sampling effects is using an digital os-
cilloscope like the one in the lab! The input signal is sampled and the continuous
time signal is converted into a discrete time signal. Sampling frequency is chosen in
a way that the graph on the screen looks like a continuous line. In fact you have
2500 dots! If the input signal exceeds the Nyquist frequency for the given time base
aliasing happens. This alias will appear as signal on the screen! Below is the table
of sampling rates for a given time base:

1. Demonstrate that the graph on the oscilloscope screen consists of single points.

ˆ Connect the function generator to the oscilloscope. Use f = 300 Hz,

A = 5 Vpp , no offset.
ˆ Set the sampling rate at the oscilloscope to 25 kS/s.
ˆ In the DISPLAY menu of the oscilloscope set DISPLAY → Dots.
Take a hardcopy.
ˆ Switch DISPLAY back to DISPLAY → Vectors. Take a hardcopy.

2. What happens when the input signal exceeds the Nyquist frequency.

ˆ Keep the sampling rate at the oscilloscope at 25 kS/s.

ˆ Set the frequency at the generator to 24900 Hz, 25000 Hz, 25020 Hz , and 25500 Hz.
Use the ’MEASURE’ function to measure the alias frequency!. Take a
hardcopy for every step.

63
8.3.2 Problem 2: Sampling using a sampling bridge
1. Implement a modified sampling circuit using a single sampling source (See
Prelab). Use the square signal of the auxiliary function generator as the sam-
pling signal. Use 1N4148 diodes and 10KΩ resistors for the bridge and a
100 KΩ resistor as load.

2. Connect a DC source to the input. Test the circuit by varying the amplitude
of the input signal between 2.0 V and 3.0 V and check whether the amplitude
of the sampled signals follows the variation.

3. Connect the signal generator to the input. Set the amplitude to Vpp = 1.5 V,
and the offset to V = 2.5 V. Take hardcopies of the input and the sampled
signal at 50 Hz, and 200 Hz.

64
9. Experiment 5 : AM Modulation
9.1 Introduction to AM and FM experiments
9.1.1 Objectives of the experiments
The goal of this experiment of the Signals and Systems Lab is to study different
analog modulation techniques.
In the first part of the experiment amplitude modulation will be investigated.
We will examine the properties of double-sideband (DSB) modulation, double-
sideband suppressed carrier (DSB-SC) modulation, and single-sideband amplitude
(SSB) modulation and their frequency spectra. The techniques used for demodula-
tion will be explained. Practically, the oscilloscope will be used to demonstrate the
impact of the amplitude modulation parameters on the modulated signal in time
and frequency domain. Furthermore, you will build a complete amplitude modu-
lation based system using the function generator as a modulator and the envelope
detector circuit as a demodulator.
In the second part of the experiment frequency modulation will be investigated. The
influence of frequency modulation parameters on the bandwidth will be explained.
Practically, the oscilloscope will be used as a spectrum analyzer to demonstrate the
impact of the frequency modulation parameters on the frequency domain. Further-
more, you will build a simple demodulation circuit consisting of a slope detector.

9.1.2 Introduction
Communication systems play a key role in the modern world in transmitting infor-
mation. A Modulator is a part of all modern day electronic communication systems
such as radio, television, and telephony.
One of the final steps before the transmission of the signal is modulation and one of
the first steps on receiving the signal is demodulation. Modulation is the process of
embedding an information-bearing signal into a second carrier signal while extracting
the information-bearing signal is known as the demodulation process.
One large class of modulation methods relies on the concept of amplitude modula-
tion (AM) in which the signal we wish to transmit is used to modulate the amplitude
of another signal. A very common form of amplitude modulation is sinusoidal am-
plitude modulation in which the information signal is used to vary the amplitude of
a sinusoidal signal. Another important form of AM systems involves the modula-
tion of the amplitude of a pulsed signal, which is called pulse amplitude modulation
(PAM). A wide variety of modulation methods are used in practice. In the hand-
out we will examine some of the most important of these amplitude modulation
techniques.
In frequency modulation, the information-bearing signal or the message signal we
wish to transmit is used to modulate the frequency of another signal that is the
carrier signal rather than amplitude variations in the carrier signal as in case of

65
amplitude modulation. Frequency Modulation is part of a more general class of
modulation schemes known as angle modulation. Angle modulation includes both
phase modulation and frequency modulation. Theories and concepts are similar for
phase modulation and frequency modulation, but we will only refer to frequency
modulation in this lab.

9.1.3 Why to Modulate?

Why do we have to modulate a signal for transmission? Or, why can’t the signal
be sent as it is? Modulation is required for a variety of reasons. There are three
reasons that force us to use modulation.
The first reason is the attenuation of the channel, i.e. air attenuation is high for
low frequency voice signals, but the attenuation is significantly lower for higher
frequencies.
The second reason has to do with the laws of electromagnetic propagation, which
dictate that the size of the radiating element, the antenna, is a significant fraction of
the wavelength of the signal being transmitted. For example, if we want to transmit
a 1 kHz signal by a quarter wave antenna, the size of the antenna would need to be
75 km. On the other hand, if the signal is being transmitted on a high frequency
carrier, say 630 kHz, the corresponding size of the radiating antenna needs to be
only 119 m.
The third reason is the possibility of transmitting different signals simultaneously.
Since the audio signals relevant to humans are between a few Hertz and a few
thousand Hertz, we could only transmit one baseband signal at a time. With si-
multaneous transmission, the signals would overlap and we would not be able to
separate them.

9.2 Amplitude Modulation

9.2.1 Band-limited signal AM modulation
The concept of amplitude modulation is a complex or sinusoidal signal (carrier sig-
nal) has its amplitude modulated by the information-bearing signal (message signal
or modulating signal). In the handout, we will explain the amplitude modulation on
the form where the carrier signal is sinusoidal (i.e. sinusoidal amplitude modulation)

c(t) = Ac cos (2πfc t + θc ) (9.1)

Where, Ac is the amplitude and fc is the frequency of the carrier. For convenience,
we choose θc = 0,so that the carrier signal is
c(t) = Ac cos (2πfc t) (9.2)
Fig. 9.1 and Fig. 9.2 shows the carrier signal in time and frequency domain respec-
tively. The modulating signal x(t), could be music, video, or any other bandlimited
signal. The amplitude of the carrier wave is varied about a mean value, linearly
with the modulating signal thus forming the envelope of the modulated signal that
has essentially the same shape as the modulating signal.

66
 Signals
Signals
and
and
Systems
Systems
Lab,
Lab,
Fall
Fall
2009,
2009,
Jacobs
Jacobs
University
University

Fig.
Fig.1:1:
Figure Time
Time
9.1: domain
domain
Time domain Fig.
Fig.2:9.2:
Figure 2:Frequency
FrequencyDomain
FrequencyDomain
Domain

Mathematically this can be represented as:

Mathematically
Mathematicallythis
thiscan
canbeberepresented
representedas:
as:
y(t) = Ac [1 + kx(t)] cos (2πfc t) (9.3)
y(t)=
y(t)=A’k’
where cA[1+[1+
c is k kx(t)]
the x(t)]
cos(2πf
cos(2πf
transmitter ct)
ct)
sensitivity. Two requirement must be satisfied in (3)
(3)
order
to have a correct amplitude modulated signal that can be modulated properly:
where,k
where,kisisthe thetransmitter
transmittersenstivity.
senstivity.
1. The amplitude of (kx(t)) is always less than unity ,that is, |km(t)| < 1 for all t
Two
Two,requirment
requirment
otherwise must
the mustbebewave
carrier stisfied
stisfiedininorder
becomes ordertotohave
havea acorrect
overmodulated correct
and amplitude
amplitudesignal
the modulated modulated
modulated
signal
signal that
that can
can bebemodulated
modulated
then exhibits envelope distortion.properly:
properly:

1.1.2.The
Theamplitude
amplitude
carrier ofof(kfc
signal (kx(t))
x(t))isis
must always
bealways less
greater lessthan
than than
theunity
unity,that
highest,thatis,is,|k|km(t)|
frequency m(t)|< <1 1forforallallt ,t ,
component
otherwise
W of the message signal x(t), that is fc >> W , otherwise an envelope cannotsignal
otherwise the
thecarrier
carrier wave
wave becomes
becomes overmodulated
overmodulated and
and thethe modulated
modulated signal
then
then
be exhibits
exhibitsenvelope
visualized. envelopedistorsion.
distorsion.
2.2.
A The
Thecarrier
carrier
band-limited signal
signal
signal fc fmust
x(t) c must bebegreater
in time greaterthan
domain isthanthe
shown thehighest
inhighest frequency
Fig. 9.3.frequencycomponent
componentWWofof
the
themessage
messagesignal
signalx(t),x(t),that
thatisisfc fc>>>>W, W,otherwise
otherwiseananenvelopeenvelopecannotcannotbebe
visualized.
visualized.
A Aband-limited
band-limitedsignal
signalx(t)
x(t)inintime
timedomain
domainisisshown
shownininfigure
figure(3)
(3). .

Figure 9.3: A band-limited signal in time domain

A band-limited signal is a signal whose Fourier transform is zero outside a given

Fig.
Fig.3:3:A Aband-limited
band-limitedsignal
signalinintime
timedomain
domain
range of frequency, i.e. X(jω) = 0 for |f | > W , as shown in Fig. 9.4.

A Aband-limited
band-limitedsignal
signalisisa asignal
signalwhose
whoseFourier Fouriertransform
transformisiszero
zerooutside
outsidea agiven
given
range
rangeofoffrequency,
frequency,i.e.
i.e.X(jω)
X(jω)= =0 0forfor|f||f|> >W,
W,asasshown
shownininfigure
figure(4).
(4).

Figure 9.4: A band-limited signal in frequency domain

The carrier signal Fig.

Fig.4:4:A Aband-limited
modulated band-limitedsignal
signalininfrequency
by the band-limited frequencydomain
modulating domain
signal is shown in
Fig. 9.5.
The
Thecarrier
carriersignal
signalmodulated
modulatedbybythe
theband-limited
band-limitedmodulating
modulatingsignal
signalisisshown
showninin
figure
figure(5).
(5). 67

66
 Figure 9.5: Modulated signal in time domain

The Fourier transform of the modulated signal calculated from Eq. (9.3) is given by,

AC kAC
Y (f ) = [δ(f − fC ) + δ(f + fC )] + [X(f − fC ) + X(f + fC )] (9.4)
2 2
Fig. 9.6 illustrates the modulated signal spectrum.

Bandwidth = 2W
Y(f)
AC/2 AC/2
Lower Upper
Sideband Sideband
kAC/2

-fC 0 fC

Figure 9.6: Modulated signal in frequency domain

The spectrum consists of two delta functions weighted by the factor Ac/2 at ±fC
and two copies of the base-band spectrum scaled by the factor kAc/2 and shifted
at ±fC . The sidebands above and below the carrier frequency are called the upper
and lower sidebands.

9.2.2 Single frequency AM modulation

Consider a modulating signal x(t) that consists of a single frequency component

x(t) = Am cos (2πfm t) (9.5)

Fig. 9.8 and Fig. 9.8 shows the modulating signal in time and frequency domain
respectively.
The carrier signal modulated by the band-limited modulating signal is shown in
Fig. 9.9.
Mathematically this can be represented as:
y(t) = Ac [1 + kAm cos (2πfm t)] cos (2πfc t) (9.6)

68
 Figure 9.7: Time domain Figure 9.8: Frequency Domain

Figure 9.9: Modulated signal in time domain

where,

m = kAm (9.7)

’m’ is called the modulation factor/index. To avoid overmodulation ’m’ must be kept
below unity. Using the trigonometric identities on Eq. (9.6), the Fourier transform
of the modulated signal is given by

AC
Y (f ) = [δ(f − fC ) + δ(f + fC )]+
2
mAC
[δ(f − fC − fm ) + δ(f + fC + fm )]+ (9.8)
4
mAC
[δ(f − fC + fm ) + δ(f + fC − fm )]
4

Thus the spectrum of an AM signal consists of delta functions as shown in Fig. 9.10.

Figure 9.10: Modulated signal in frequency domain

69
9.2.3 AM signal power and bandwidth
The transmitted power and the channel bandwidth are two primary communication
resources and should be used efficiently. The AM signal is a voltage function. The
average power delivered to a resistor by the AM signal is compromised of three com-
ponents, carrier power, upper side frequency power and lower side frequency power.
The transmission bandwidth of the AM signal is equal to the difference between the
highest frequency component (fc +W ) and the lowest frequency component (fc −W )
which is exactly twice the message bandwidth W, that is,

BT = 2W (9.9)

Where, W is the maximum frequency contained in the modulating message signal.

9.2.4 How to improve the efficiency of the DSB AM system

A. Double Sideband Suppressed Carrier (DSB-SC) modulation In double
side band suppressed carrier (DSB-SC) modulation the transmitted signal consists
only the upper and lower sidebands as shown in Fig. 9.11.

Bandwidth = 2W
Y(f)
AC/2 AC/2
Lower Upper
Sideband Sideband
kAC/2

-fC 0 fC

Figure 9.11: DSB-SC modulation in frequency domain

Transmitted power is saved through the suppression of the carrier signal, but the
transmission bandwidth is the same as in DSB modulation.

B. Single Sideband (SSB) modulation In single side band (SSB) modulation

the transmitted signal consists only the upper sideband or the lower sideband. It
is an optimum form of modulation, as it requires the minimum transmitted power
and the minimum transmission bandwidth.

9.2.5 Amplitude demodulation

The information signal is recovered through demodulation. There are two commonly
used methods for demodulation, each with its own advantages and disadvantages.
The process referred to as synchronous demodulation, in which the transmitter
and receiver are synchronized in phase, an alternative method referred to as asyn-
chronous demodulation.

70
a. Synchronous Demodulation For the DSB-SC modulation, the modulating
signal x(t) is multiplied by the carrier signal to produce the modulated output signal.

y(t) = x(t) cos (2πfC t) (9.10)

The demodulation process involves multiplication of the modulated signal by the

same carrier signal that was used in the modulator- hence the name synchronous.
We multiply the modulated signal with the carrier signal to obtain the following:

w(t) = y(t) cos (2πfC t) (9.11)

Thus:

w(t) = x(t) cos2 (2πfC t) (9.12)

Using the trigonometric identity:

1
cos2 A = (1 + cos (2A)) (9.13)
2
we get:
1 1
x(t) = x(t) + x(t) cos (4πfC t) (9.14)
2 2
Thus, w(t) consists of the sum of two terms, namely one-half the original signal and
one-half the original signal modulated with a sinusoidal carrier at twice the original
carrier frequency fc . As discussed previously the band-limited signal has significantly
lower frequencies as compared to the carrier signal. Thus, the original signal and the
signal that has double the frequency of the carrier signal are sufficiently separated
in frequency domain to apply a low pass filter to w(t). On applying a low pass filter
the first term, i.e., 1/2x(t) will remain and the second term will be eliminated. This
is graphically represented in Fig. 9.12.

Figure 9.12: Synchronous Demodulation

This process is quite simple,but practically it is difficult to implement due to a the

following problem.

71
For the sinusoidal carrier, let θC and φC denote the phase of the modulating and
the demodulating carriers respectively. The input to the low pass filter is now:

w(t) = x(t) cos (2πfC t + θC ) cos (2πfC t + φC ) (9.15)

On using the trigonometric identities, we get the following equation

1 1
w(t) = x(t) cos (θC − φC ) + cos (4πfC t + θC + φC ) (9.16)
2 2
Thus, the message signal is multiplied by an amplitude factor of 1/2 cos (θC − φC ).
From this equation, we can clearly see that the possibility of recovering the original
information signal strongly depends on the phase difference, i.e. on the difference
between the θC and φC . Thus, if (θC −φC ) is zero, or a multiple of π, then the original
signal is recovered perfectly. If (θC − φC ) is a multiple of π/2 then the information
is lost completely. More greater importance, the phase relation between the two
oscillators must be maintained over time, so that the amplitude factor cos (θC − φC )
does not vary.
This requires careful synchronization between the modulator and the demodula-
tor, which is often difficult, particularly when they are geographically separated, as
in a typical communications system. These effects of lack of synchronization are
also between the frequencies of the carrier signals used in both the modulator and
demodulator.
An alternative method can be used referred to as asynchronous demodulation, where
a slight change to the modulation process makes a much simpler detector possible.

b. Asynchronous Demodulation Asynchronous Demodulation avoids the need

for synchronization between the modulator and demodulator. Suppose that the
modulating signal is always positive by adding a constant (dc offset) to the original
signal. The modulating signal thus becomes

x(t) = x(t) + C (9.17)

This basically means that the signal will be shifted upwards along the y-axis. This
can be seen in Fig. 9.13.

Figure 9.13: Signal shifting

72
The dc value C must chosen in an appropriate manner to shift the entire signal above
the time axis. If we look to the frequency domain, the only effect is the addition of
a delta function at zero frequency with the corresponding magnitude. The signal in
frequency domain is shown in Fig. 9.14.

Figure 9.14: Signal shifting effect in frequency domain

The modulated signal is shown in Fig. 9.15.

Figure 9.15: Modulated signal in time domain

Thus, the envelope of xc (t), can be approximately recovered through the use of a
circuit that tracks these peaks to extract the envelope. Such a circuit is referred
to as an envelope detector or peak detector. A very simple envelope detector can
be made by low-pass filtering a full-wave rectified modulated signal as shown in
Fig. 9.16.

Vin R C Vout

Figure 9.16: Envelope detector circuit

The diode in the circuit allows only the positive part of the cycle to pass and then a
capacitor/resistor combination extracts the shape or envelope of the signal. Suitable
values of R and C should be used for various carrier and modulation frequencies.
To use the envelope detector for demodulation, we require that C be sufficiently
large, so that xc (t) = x(t) + C is positive. Assuming that Am denotes the maximum
amplitude of x(t). For xc (t) = x(t) + C is positive, we require that C > Am .

73
9.2.6 Asynchronous vs. synchronous modulation
The output from the asynchronous modulator has an additional delta component at
fc in the spectrum that is not present by using synchronous modulator. This carrier
component in the output represents inefficiency in the amount of power required to
transmit the modulated signal.
An advantage for the asynchronous modulation is the ability of a simple envelope
detector to follow the input and extract the message signal.
For synchronous modulation complicated demodulator is needed because the oscil-
lator in the demodulator must be synchronized with the oscillator in the modulator,
in phase and frequency.

9.2.7 References
1. A. V. Oppenheim, A. S. Willsky, S. H. Nawab, ”Signals and Systems,” Prentice
Hall, Second Edition 1997.

2. Theodore S. Rappaport, Wireless Communications: Principles and Practice

(2nd Edition)

74
9.3 Prelab AM Modulation
9.3.1 Problem 1: Single frequency Amplitude Modulation
1. Derive an expression describing the modulation index m as a function of the
modulation envelope, (use Amin and Amax !).

2. Derive an expression describing the ratio of the total sideband power to the
total power rP = Ps /Ptot in the modulated wave delivered to a load resistor.
Express the ratio in terms of the modulation index.

3. Calculate the ratio rP assuming a modulation index of 100%.

4. A Carrier

VC (t) = 5 cos (20000πt)

is AM modulated by a signal

Vm (t) = 2 + cos (2000πt)

(Take care, the signal is uof f +as cos(ωt) !!)

Calculate the ratio rP . How would you change the input signals to maximize
the side band to total power ratio?

9.3.2 Problem 2: Amplitude Demodulation

Use MatLab to generate an AM modulated signal with a carrier frequency f =
20 KHz and am = 5 V. Modulate with a sine wave f = 500 Hz. Use a modulation
index m = 50%. Simulate the demodulation of this AM signal with Matlab. Show
the following steps:

1. Plot the modulated signal in time and frequency domain.

2. Design a first and a third order low pass filter (butterworth filter) to demodu-
late the signal. The cut-off frequencies of the filters should be 1 KHz. Plot the
Bode diagram of these filters for a frequency range from 100 Hz to 100 KHz to
verify the function.

3. Rectify the AM modulated signal and apply the 1. order low pass filter to the
rectified signal. Plot the rectified and the demodulated signal.

4. Change the order of the filter from 1. to 3. Plot the demodulated signal.

5. Why is it better to use a higher order filter for the demodulation of the signal?

6. Attache the full Matlab script to the prelab.

75
9.4 Execution AM Modulation
9.4.1 Problem 1: AM modulated Signals in Time Domain
The AM signal is generated by the function generator. Settings of the function
generator:

Signal Shape = Sine

Modulation = AM
Carrier frequency = 20 kHz
Carrier Amplitude = 10 VP P
Modulation Frequency = 500 Hz
Modulation index = 50%

1. Connect the function generator to the oscilloscope. Measure the frequency and
the amplitude properties of the modulated signal and obtain the modulation
index. Take hardcopies.

2. Repeat step 1 by setting the modulation index to 70%. Take hardcopies.

3. Adjust the modulation index to be 120% and observe the effect on the AM
signal. Take a hardcopy.

9.4.2 Problem 2: AM Modulated Signals in Frequency Do-

main
Use the same setup as before. Set the modulation index at the function generator
to 70%. At the oscilloscope display the amplitude modulated signal in frequency
domain. (FFT!). Use the cursors to measure the magnitudes and the frequencies.
Take hardcopies!

9.4.3 Problem 3: Demodulation of a message signal

Use the following circuit:
Amplifier
(Impedance Converter) 1. Order Low Pass
filtered output
Rectifier and
Envelope Detector L1 R2
D1 - 10mH 180R 3.(!) Order Low Pass
1N4148 LM741
filtered output
+

C1 R1 2. Order Low Pass C2

6n8F 22k0 1u0F

Fig. 1: Demodulation circuit

Do not forget that an OP-Amp needs a power supply. Below is the pinout and the
necessary circuit.

76
 ■ Integrator Package
■ Active filter Part Number Temperature Range
N D
■ Function generator UA741C 0°C, +70°C • •
The high gain and wide range of operating voltag- UA741I -40°C, +105°C • •
es provide superior performances in integrator, UA741M -55°C, +125°C • •
summing amplifier and general feedback applica- Example : UA741CN
tions. The internal compensation network (6dB/
N = Dual in Line Package (DIP)
octave) insures stability in closed loop circuits. D = Small Outline Package (SO) - also available in Tape & Reel (DT)

PIN CONNECTIONS (top view)

1 - Offset null 1
1 8 2 - Inverting input
3 - Non-inverting input 10V 100nF 7
V+
4 - VCC-
2 7 5 - Offset null 2
6 - Output UA741
3 6 7 - VCC+
8 - N.C. V-
4 5 10V 100nF 4

November 2001 Fig. 2: LM741 pinout and supply circuit 1/5

Connect the function generator to the input of the demodulating circuit. Use the
following settings:
Signal Shape = Sine
Modulation = AM
Carrier frequency = 20 KHz
Carrier Amplitude = 10 VP P
Modulation Frequency = 500 Hz
Modulation index = 50%
1. Display the AM modulated signal together with the 1. order filter output.
Take a hardcopy.
2. Display the AM modulated signal together with the 3. order filter output.
Take a hardcopy.
3. Measure the amplitude of the demodulated signal at the 3. order output.
4. Take a FFT of the signal at the 3. order filer output. Check if there is still a
20kHz component. Take hardcopies.

9.5 Evaluation AM modulation

9.5.1 Problem 1: AM modulated Signals in Time Domain
1. What is the relation between the modulation index and the relative magnitudes
of the frequency components?
2. Calculated the modulation index using the measurements! Compare to the
index you have used to generate the AM signal.
3. Discuss the effect and the disadvantages of using a modulation index greater
than 100

9.5.2 Problem 2: AM Modulated Signals in Frequency Do-

main
1. How does the spectrum look like in theory? Compare to the experiment!
2. Does the function generator generate a DSB or DSB-SC AM signal?
3. How does changing the carrier frequency affect the AM spectrum?
4. How does changing the message frequency affect the AM spectrum?
5. Determine the modulation index m using the measured values.

77
9.5.3 Problem 3: Demodulation of a message signal
1. Compare the 1. and 3. order filter output signal with the message signal.

2. Compare the measured signals with the MatLab results. What are the differ-
ences between simulation and measurement?

78
10. Theory 6 : FM Modulation
10.1 Frequency Modulation
10.1.1 Objective
This is the second part of the modulation experiment. The influence of frequency
modulation parameters on the bandwidth will be explained. Practically, the os-
cilloscope will be used as a spectrum analyzer to demonstrate the impact of the
frequency modulation parameters on the frequency domain. Furthermore, you will
build a simple demodulation circuit consisting of a slope detector.

10.1.2 Single frequency FM modulation

In frequency modulation, the amplitude of the modulated carrier signal is kept con-
stant while its frequency is varied by the modulating message signal. The basic idea
of frequency modulation is shown in Fig. 10.1. The carrier frequency is controlled
at each instant by the voltage of the modulating signal. The frequency of the mod-
ulated signal is increased if the input signal is positive, whereas the frequency is
reduced if the input signal is negative.
The general definition of frequency modulated signal SF M (t) is given by the formula:
 Z t 
SF M (t) = AC cos (2πfC t + θ(t)) = AC cos 2πfC t + 2πKf m(τ )dτ (10.1)
0
where,
m(τ ) is the modulating signal.
AC is the amplitude of the carrier.
fC is the carrier frequency.
Kf is the frequency deviation constant measured in Hz/V.
For the case of a single frequency sinusoidal modulating signal, m(τ ) = Am cos (2πfm τ )
the frequency modulated signal SF M (t) will be expressed as:
Kf Am
SF M (t) = AC cos (2πfc t + sin (2πfm t)) (10.2)
fm
The factor Kf Am is called the frequency deviation. It is defined as the maximum
frequency shift away from fc . The frequency modulation index βf , is expressed as:
Kf A m ∆f
βf = = (10.3)
fm fm
So, the SF M (t) signal can be represented as:
SF M (t) = AC cos (2πfC t + βf sin (2πfm t)) (10.4)
Frequency modulated signals are classified into two categories based on the value of
βf .

79
 Figure 10.1: frequency modulation process

a. Narrow Band Frequency Modulation (NBFM) For small values of the

frequency modulation index (βf << 1), we have Narrow Band Frequency Modula-
tion (NBFM). In this case, the frequency modulated signal SF M (t) becomes:
SF M (t) = AC cos (2πfC t) − AC βf sin (2πfC t) sin (2πfm t) (10.5)
The derivation of Eq. (10.5) can be found in reference [2].

b. Wide Band Frequency Modulation (WBFM) As the modulation index

increases, the signal occupies more bandwidth. In this case the modulation scheme is
called Wide Band Frequency Modulation (WBFM). Therefore, for a single frequency
sinusoidal modulating signal the frequency modulated signal could be written in the
form:
∞
X
SF M (t) = AC Jn (β) cos (2π(fC + nfm )t) (10.6)
n=−∞

The derivation of Eq. (10.6) can be found in reference [2].

10.1.3 FM spectrum
As with amplitude modulation, the modulation process causes sidebands to be pro-
duced at frequencies above and below the carrier. However, for a frequency mod-
ulation based system, there are a lot more, all spaced at multiples of f m from the

80
carrier frequency fC . As a result, the bandwidth needed to accommodate a fre-
quency modulated signal is considerably larger than that for amplitude modulated
signal having the same modulating frequency.
Fig. 10.2 shows the spectrum of a frequency modulated signal for various values
of the modulation index βf . The modulating signal in these examples is a single
frequency sinusoidal signal.

Figure 10.2: Frequency modulation index effect

When a sinusoidal signal such as m(t) = Am cos (2πfm t) is used, the spectrum
contains a carrier component and many number of sidebands located on either side
of the carrier frequency, spread at integer multiples of the modulating frequency fm
(fc ± nfm ), for all positive n (n = 0 is the carrier frequency component).
The only exception is at a very low frequency modulation index, most of the infor-
mation is contained within the range of the first upper and lower sidebands, which
makes the total bandwidth sufficient for transmission about the same as for am-
plitude modulation based system, that is 2fm . With larger frequency modulation
indexes, the number of sidebands increases and we obtain larger bandwidth.
Fortunately, something else is happening which keeps the total bandwidth reason-
able. To get a large frequency modulation index, we need a large frequency deviation
but a small modulation frequency, according to the modulation index definition. The
modulation frequency; however, determines the spacing between sidebands. So, at
high modulation index, we may have many sidebands, but they will be closely spaced,
so the total occupied bandwidth will not be much larger. This is demonstrated in
Fig. 10.3.
Theoretically, the bandwidth of a frequency modulated carrier is infinite. In practice,
however, we find that the frequency modulated signal is effectively limited to a finite
number of significant sideband frequencies within an approximate bandwidth, BT ,
given by Carson’s rule.
Carson bandwidth rule is a rule defining the approximate bandwidth requirements of
communications system components for a carrier signal that is frequency modulated.

81
 Figure 10.3: Same deviation, but different modulation index

In case of a single frequency modulation, the empirical Carson’s rule is given by

BT ∼
= 2fm (βf + 1) (10.7)

For more practical case, an arbitrary modulating signal m(t) is considered and its
highest frequency component is denoted by W . Then, replacing β by D and replacing
fm with W in Eq. (10.7) we get

BT ∼
= 2W (D + 1) (10.8)

D is called the deviation ratio and it is defines as the ratio of the frequency deviation
∆f , which corresponds to the maximum possible amplitude of the modulating signal
m(t), to the highest modulation frequency W .
The maximum frequency deviation depends on the maximum amplitude of the mod-
ulating signal and the sensitivity of the modulator. The sensitivity of the modulator
is called the frequency-deviation constant, Kf . Thus, D is given by the following
formula:
Kf ∗ max |m(t)|
D= (10.9)
W
where, W is the highest modulation frequency, m(t) is the message signal and Kf
is the frequency-deviation constant.
The deviation ratio D plays the same role for arbitrary modulation that the modu-
lation index βf plays for the case of a single sinusoidal modulation. From a practical
viewpoint, Carson’s rule somewhat underestimated the bandwidth.

10.1.4 Single frequency FM demodulation

There are many ways to recover the original information from a frequency mod-
ulated signal. The frequency demodulator should produce an output voltage with

82
instantaneous amplitude that is directly proportional to the instantaneous frequency
of the input frequency modulated signal. Thus, a frequency-to-amplitude converter
circuit is a frequency demodulator.
Various techniques such as slope detection, zero-crossing detection, phase locked
discrimination and quadrature detection are used to demodulate the frequency mod-
ulated signal.
Universally, demodulators use a phase-locked loop (PLL), an extremely useful circuit
that finds its way into all sorts of electronic systems. Briefly, it consists of an
oscillator whose frequency can be varied by means of a voltage (that is, a voltage
controlled oscillator or VCO), and a feedback loop, which results in the frequency
of the oscillator being locked to the frequency of the incoming signal. In the process
the circuit produces a voltage, which is proportional to the variation in the signal
frequency.
In this experiment we will use a simple slope detector do demodulate an FM signal.
A slope detector is essentially a resonator (tank) circuit which is tuned to a frequency
either slightly above or below the fm carrier frequency.

10.1.5 Frequency Modulation vs. Amplitude Modulation

The main advantages of using frequency modulation over amplitude modulation are:

1. Frequency modulated signals have better noise immunity than amplitude mod-
ulated signals since signals are represented as frequency variations rather than
amplitude variations. Frequency modulated signals are less susceptible to at-
mospheric and impulse noise, which tend to cause rapid fluctuations in the
amplitude of the received radio signal.

2. In amplitude modulation, the peak amplitude of the envelope of the carrier is

directly dependent on the amplitude of the modulating signal, which requires
a large dynamic range of the transmitter and the receiver. For frequency mod-
ulation, the envelope carrier is constant. Consequently, the FM transmitter
can always operate at peak power.

There are also some serious disadvantages in frequency modulation.

1. Frequency modulated signal typically requires larger bandwidth.

2. Frequency modulation based systems are much more complicated to analyze

and build than amplitude modulation based systems.

10.2 References
1. A. V. Oppenheim, A. S. Willsky, S. H. Nawab, ”Signals and Systems,” Prentice
Hall, Second Edition 1997.

2. Theodore S. Rappaport, Wireless Communications: Principles and Practice

(2nd Edition)

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10.3 Prelab FM Modulation
10.3.1 Problem 1: Frequency Modulator
A sinusoidal modulation signal,

m(t) = 4 cos (8000πt)

is applied to an FM modulator that has a frequency deviation constant Kf =

10 kHz/ V. Compute the frequency deviation and the frequency modulation index.

10.3.2 Problem 2: FM signal in the frequency domain

Plot a frequency modulated signal in the frequency domain. The signal exhibits
2.5 V peak carrier amplitude, 40 kHz carrier frequency, 5 kHz modulation frequency.
Vary βf between 0.2, 1, and 2. Display the magnitudes in dBrms ! Calculate the
bandwidth using Carlsons rule. Tabulate the peak magnitudes inside the bandwidth
from the three plots.

10.3.3 Problem 3: Frequency demodulation

In Fig. 10.4 a simple slope detector circuit is shown . The circuit can be used as
a FM demodulation circuit. Provide a bode diagram and explain the operation
principle of the circuit.
Hint: The resonance frequency of the circuit is set higher than the carrier frequency
fc of the FM signal.

rectified,
AM shaped
FM-Input FM signal
demodulated
AM signal

Figure 10.4: Schematic circuit of a slope detector.

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 Part III

Additional Information

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A. Appendix
A.1 Hardcopy from oscilloscope screen
For the documentation and evaluation of an experiment it is useful to save the data
from the oscilloscope screen. This is possible with the help of a printer, a computer
or an USB stick.
The TBS series oscilloscope in the laboratory has a USB interface. Insert a USB
stick into the socket on the front panel. To get a hardcopy and the data press the
button with the floppy disk symbol. Do not forget to save the recorded data from
the stick to your computer at the end of the course.

A.2 Books and other Tools

A.2.1 Book
ˆ Sarma

ˆ Floyd

A.2.2 Programs
LTSpice
LTSpice is a powerful and unrestricted circuit simulator for analog circuits. It con-
tains a graphical user interface for entering circuit diagrams and a waveform viewer
for displaying the results. It is freely available for Windows, Mac and with an
emulator also for Linux. Download link is ’from Analog Devices’.

Octave
GNU Octave is a high-level programming language intended for numerical com-
putations. It is typically used for such problems as solving linear and nonlinear
equations, numerical linear algebra, statistical analysis, and for performing other
numerical experiments. It may also be used as a batch-oriented language for auto-
mated data processing. It is freely available for Windows, Mac, and Linux. Octave
has been built with MATLAB compatibility in mind, and shares many features with
MATLAB. Octave is available from ’https://octave.org/’.

KiCad
KiCad is an open-source software suite for creating electronic circuit schematics,
printed circuit boards (PCBs), and associated part descriptions. KiCad supports
an integrated design workflow in which a schematic and corresponding PCB are
designed together, as well as standalone workflows for special uses. KiCad also

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includes several utilities to help with circuit and PCB design, including a PCB
calculator for determining electrical properties of circuit structures, a Gerber viewer
for inspecting manufacturing files, a 3D viewer for visualizing the finished PCB, and
an integrated SPICE simulator for inspecting circuit behavior. It is freely available
for Windows, Mac, and Linux from ’https://www.kicad.org’.

MatLab
MATLAB (an abbreviation of ”MATrix LABoratory”) is a proprietary multi-paradigm
programming language and numeric computing environment developed by Math-
Works. MATLAB allows matrix manipulations, plotting of functions and data,
implementation of algorithms, creation of user interfaces, and interfacing with pro-
grams written in other languages. It is available for Windows, Mac, and Linux. As
registered student you may download it and use the university based license.

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