Roberts (2003) Fundamentals of Signals and Systems

EE2D1 – Fundamentals of Signals and Systems (Maths) – Course Overview
Clive Roberts
Course Overview
Introduction
During this course we will be studying the mathematics required for engineering
transforms and analysis in communications and control.
We will study:
• Fourier Analysis;
• Laplace Transforms;
• z-Transforms;
• Statistics and Probability.
Key Texts
I list two key texts:
• E Kreyszig, Advanced Engineering Mathematics, Wiley, 1999, ISBN 0-471-
33328-X
• G James, Advanced Modern Engineering Mathematics, Prentice Hall, 1999,
ISBN 0-201-59621-0
However there are a large number of texts available in the library in these areas. It is
better to pick a text that you find easy to follow and understand. The notes supplied
each week will be comprehensive, however a text will provide further information
and extra examples. The notes and other useful information will also be available at
www.eee.bham.ac.uk/robertc
Tutorials
Tutorials (staffed by PGTAs) will take place on Thursday at 11am during weeks 3, 5,
7, 9. You will be expected to hand in work to be marked prior to these tutorials by
Tuesday at 11am prior to the tutorial.
Support Maths
Optional support maths (staffed by PGTAs) will be available on Wednesday at 2pm in
NG15 during weeks 3, 6, 8, 10. This will provide an opportunity for those that are
having some difficulty to go through the material again in a non- lecture environment
and ask questions having been through the material.
Reading / Self-Study Week

There will be no lectures for this course during week 9 (only a tutorial). This is to
provide you with some time to review the course material, catch up with any
outstanding work, revise for the class test and commence the coursework exercise.
Class Test
The class test will be held during the tutorial session of week 11. This will account
for 10% of the course’s final mark.
Coursework
The coursework, given in week 8, will account for 20% of the course’s final mark.
Examination
During the summer examination period you will sit a 1½ hour paper.
EE2D1 – Fundamentals of Signals and Systems (Maths) – Course Overview
Clive Roberts
Timetable
Week 1
Tuesday at 2pm – Course Overview Lecture – LT1
Week 2
Monday at 11am – Introduction to Fourier Analysis Lecture – LT1
Wednesday at 11am – Fourier Series Examples and Applications Lecture – LT1
Week 3
Monday at 11am – Introduction to Fourier Transforms Lecture – LT1
Wednesday at 11am – Fourier Transforms Examples and Applications Lecture – LT1
Wednesday at 2pm – Fourier Support Maths – NG15
Thursday at 11am – Fourier Tutorials – Seminar Rooms
Week 4
Monday at 11am – Further Fourier Analysis Lecture – LT1
Wednesday at 11am – Introduction to Laplace Transforms Lecture – LT1
Week 5
Monday at 11am – Laplace Transforms Examples and Applications Lecture – LT1
Wednesday at 11am – Laplace Transform Application to DEs Lecture – LT1
Thursday at 11am – Laplace 1 Tutorials – Seminar Rooms
Week 6
Monday at 11am – Laplace Transforms Examples and Applications Lecture – LT1
Wednesday at 11am – Further Laplace Transforms Lecture – LT1
Wednesday at 2pm – Laplace Support Maths – NG15
Week 7
Monday at 11am – Intro. to Discrete Systems and z-Transforms Lecture – LT1
Wednesday at 11am – z-Transforms Exa mples and Applications Lecture – LT1
Thursday at 11am – Laplace 2 Tutorials – Seminar Rooms
Week 8
Monday at 11am – MatLab Lecture – LT1
Wednesday at 11am – Coursework Lecture – LT1
Wednesday at 2pm – z-Transform Support Maths – NG15
Week 9 (Reading / Self-study Week)

Thursday at 11am – z-Transorm Tutorials – Seminar Rooms
Week 10
Monday at 11am – Further z-Transforms Lecture – LT1
Wednesday at 11am – Introduction to Probability and Statistics Lecture – LT1
Wednesday at 2pm – Support Maths – NG15
Week 11
Monday at 11am – Statistics Examples and Applications Lecture – LT1
Wednesday at 11am – Probability Examples and Applications Lecture – LT1
Thursday at 11am – Class Test – Seminar Rooms
EE2D1 – Signals and Systems – Lecture Notes 1
Clive Roberts
Fourier Analysis
Introduction
Fourier Analysis is a family of mathematical techniques, all based on decomposing
signals into sinusoids. Fourier Analysis is named after Jean Baptiste Joseph Fourier –
a French mathematician and physicist. Fourier was interested in heat propagation,
and presented a paper in 1807 on the use of sinusoids to represent temperature
distributions. The paper contained a novel claim that any continuous periodic signal
could be represented as the sum of properly chosen sinusoidal waves.
One of the reviewers of the paper, Lagrange, voted against the publication of the
paper insisting that the technique could not be used to represent signals with corners
(discontinuous slopes such as square waves). It was not until after Lagrange’s death
that the work was officially published – some time during the French Revolution, so
Fourier probably had better things to think about!!
Well, who was right? A good question... Lagrange was right in his assertion that a
summation of sinusoids can not exactly form a signal with corner; however, you can
get exceptionally close. So close, in fact, that the difference between the actual and
reconstructed waveform has zero energy – close enough for us!! This phenomenon is
nowadays referred to as Gibb’s Effect – more about that later.
Fourier Analysis can be broken into four categories:

1. Periodic – Continuous
The examples here include: sine waves, square waves and any other waveform
that happens to repeat itself in a regular pattern continuously (± ∞ ). The
transform for this type of signal is referred to as Fourier Series.
2. Aperiodic – Continuous
This includes, for example, decaying exponentials and the Gaussian curve.
These signals extend to both positive and negative infinity without repeating
themselves in a periodic pattern. This type of transform is simply known as
the Fourier Transform.
3. Periodic – Discrete
These are discrete signals that repeat themselves in a periodic fashion from
negative to positive infinity. This class of transform is known as the Discrete
Fourier Transform or DFT.
4. Aperiodic – Discrete
These signals are only defined at discrete points between positive and negative
infinity and do not repeat themselves in a periodic manner. This type of
transform is called the Discrete Time Fourier Transform.
You may wonder why sinusoids are used rather than, say, square waves or saw tooth
waves. There are an infinite number of ways to decompose a signal, so it is possible;
however, our aim is to end up with something that is easier to deal with than the
original signal. The component sine and cosine waves are simpler than the original
signal because they have a property known as sinusoidal fidelity - if the input to a
linear system is sinusoidal then the output will also be a sinusoidal at exactly the same
frequency as the input (i.e. only a change in amplitude and phase may be seen). In
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electronics a circuit is often considered by its frequency response, which produce
graphs of how a circuit’s gain and phase vary with frequency.
Figure 1 - The four categories of Fourier Anaylsis

Other important properties of sine and cosine waves which we should be familiar with
are:
• Sines and cosines do not have any jumps of kinks in them;
• Sines and cosines can be differentiated or integrated as many times as you like
and they don’t develop any discontinuities;
• When we differentiate or integrate sines and cosines they stay the same shape.
Periodic Waveforms (revision)

In order to undertake a study of Fourier series we therefore need a good understanding
of periodic functions, especially sine and cosine functions. Some of the important
definitions and properties are listed below:
φ
• The function f (t ) = A sin (ωt + φ ) = A sin ω  t +  is a sine wave of amplitude
 ω
ω 2π
A, angular frequency ω , frequency , period T = and phase angle φ .
2π ω
φ
The time displacement is define to be . As shown in Figure 2. Similar
ω
remarks can be made about the function Acos(ωt + φ ) and together the sine
and cosine functions form a class of functions known as sinusoids or
harmonics.
• It is particularly important that you understand how to integrate these

functions:
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cos nωt sin nωt

o ∫ sin nωt.dt = − nω
+c ∫ cos nωt = nω
+c
for n = ±1, ±2, ….
• Sometimes a function occurs as the sum of a number of different sine and

cosine components such as:
o f (t ) = 2 sin ω1t + 0.8 sin 2ω 1t + 0.7 sin 4ω 1t
Note in particular that the angular frequencies of all the components are
integer multiples of angular frequency ω 1 . The component with the lowest
frequency (or largest period) is 2π sin ω1t . The quantity ω 1 is called the
fundamental angular frequency and this component is called the fundamental
or first harmonic. The component with the angular frequency 2ω 1 is called
the second harmonic and so on. A consequence of this is that the resulting
function, f(t), is periodic and has the same frequency as the fundamental.
Some harmonics may be missing, for example, above the third harmonic is
missing. In power systems application a common value for ω 1 maybe 100 π
as this corresponds to a frequency of 50 Hz.
Fourier Series
The Fourier series approximates to a function f(t) by using a trigonometric polynomial
of degree N as follows:
N
a
o f (t ) ≈ 0 + ∑ [ a n cos( nt ) + bn sin( nt )]
2 n=1
Assuming that f(t) is continuous on the interval − π ≤ t ≤ π , the coefficients a n and

bn can be computed by the formulas:
1 π
π −∫π
o a0 = f (t ).dt
for the constant term and:

1π 1 π
o a n = ∫ f (t ) cos( nt ).dt bn = ∫ f (t ) sin( nt ).dt

π −π π −π
Example 1
Consider the periodic function:
0, − π < t < 0,
o f (t ) = 
 t, 0 < t <π
with period 2π as shown in Figure 5.
Using the equation for a 0 we get:

π
1 1π π
o a0 =
π ∫−π f (t).dt = π ∫0 t.dt = 2
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a0 π
which gives a constant term of = .
2 4
The coefficients of the cosine terms are computed using the equation for an .
1 π
o a n = ∫ f (t ) cos( nt ).dt
π −π
1π
π ∫0
o an = t cos( nt ).dt
π
t 
 n sin( nt ) + n 2 cos( nt )  = πn 2 [cos( nπ ) − 1]
1 1 1
o an =
π 0
Similarly the sine terms are computed as:

1 π
o bn = ∫ f (t ) sin( nt ).dt
π −π
1π
o bn = ∫ t sin( nt ).dt
π 0
π
 t 
 − n cos( nt ) + n 2 sin( nt )  = − n [cos( nπ )]
1 1 1
o bn =
π 0
The series approximation can be rewritten using the identity relationship

cos( nπ ) = ( −1) n and by knowing that an = 0 when n is an even integer. The result is:
N
a
o f (t ) ≈ 0 + ∑ [ a n cos( nt ) + bn sin( nt )]
2 n=1
which expands as:
π 2 2
o f (t ) ≈ − cos( t ) − cos( 3t ) − .......
4 π 9π
1 1
+ sin( t ) − sin( 2t ) + sin( 3t ) + ........
2 3
Figure 2 shows the plots for the Fourier approximation for Example 1.
One to try before the next lecture…..

o f (t ) = {1 + t , − π < t < π
with period 2π .
Sketch f(t) and calculate a concise expression for the Fourier Series.
 N
2 
Answer =  f ( t ) ≈ 1 − ∑ (−1) n sin nt 
 n =1 n 
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4
N=5
3 f(t)
2
f(t)
-1
-4 -3 -2 -1 0 1 2 3 4
t
4
N=20
3 f(t)
2
f(t)
-1
-4 -3 -2 -1 0 1 2 3 4
t
Figure 2 - Fourier series approximations for Example 1

Clive Roberts
Fourier Series (cont.)

Now we have tried a couple of examples, there are a few things we should be aware
of.
a0
Firstly, the term represents the average value (or dc component) of the waveform.
2
Secondly, the period of integration that we have use up until now has been 2π , but
T T
we can use the Fourier Series over any complete period, i.e. t = − to t = . If we
2 2
think carefully about the period of integration, it may make the process simpler. The
Fourier Series equations then become:
a N
  2nπt   2nπt 
o f (t ) ≈ 0 + ∑ a n cos  + bn sin  
2 n =1   T   T 
Assuming that f(t) is continuous on the interval − T ≤ t ≤ T , the coefficients an

2 2
and bn can be computed by the formulas:
T
2
2
o a0 =
T ∫ f (t ).dt
−T
2
for the constant term and:
T T
2 2  2nπt  2 2  2nπt 
an = ∫
T −T
f (t ) cos
 T 
.dt bn = ∫
T −T
f (t ) sin 
 T 
.dt
2 2
Thirdly, the Fourier Series of even and odd functions can be computed with
significantly less effort than that needed for functions without such symmetry. The
properties that define an even or odd function are:
• An even function has a graph that is symmetric with respect to the vertical axis
(t=0) and therefore satisfies the equation f (t ) = f (−t ) .
For the even function, f e (t ) , a range of integration that is symmetrical about
the vertical axis, where t = 0, gives:
π π
o ∫
−π
f e (t ).dt = 2 ∫ f e (t ).dt
0
Based on these results, if f(t) is an even function:

a N
f (t ) ≈ 0 + ∑ [a n cos(nt )]
2 n =1
• An odd function is symmetric with respect to the origin and satisfies the
equation − f (t ) = f (−t ) .
The integration over symmetrical range about the vertical axis for an odd
function f o (t ) is zero:
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π
o ∫f
−π
o (t ).dt = 0
Based on these results, if f(t) is an odd function:

a N
o f (t ) ≈ 0 + ∑ [bn sin( nt )]
2 n =1
• A rotated function is symmetric about the zero axis, and the wave shape of
alternate half-cycles is identical but reversed in sign and satisfies
f (t ) = − f (−t + T ) . There is no constant term when rotational symmetry
2
exists, as the dc component is zero. If the function is also odd, only odd-
harmonic sine terms will appear in the Fourier Series. An even function with
rotational symmetry will have a Fourier Series consisting of only odd
harmonic cosine terms.
Example 2 (an odd example)

o f (t ) = {1 + t , − π < t < π
with period 2π . This is an odd function as − f (t ) = f (−t ) ; therefore there will be no
an terms.
a 
The dc component of the waveform  0  is:
 2
a0 (π + 1) + (−π + 1)
o = =1
2 2
Now we can work out the sine components:

π
1
o bn = ∫ f (t ) sin( nt ).dt
π −π
π
1
o bn =
π ∫ (t + 1) sin(nt ).dt
−π
π
1  1+ t 1  2
o bn = − cos(nt ) + 2 sin( nt ) = − [cos(nπ )]
π n n  −π n
The series approximation can be rewritten using the identity relationship
cos(nπ ) = (−1) n and by knowing that an = 0 when n is an even integer. The result is:
a N
o f (t ) ≈ 0 + ∑ [bn sin( nt )]
2 n =1
which we can write concisely as:
N
 − 2(−1) 2 
o f (t ) ≈ 1 + ∑  sin( nt ) 
n =1  n 
or expand as:
2
o f (t ) ≈ 1 + 2 sin t − sin 2t + sin 3t.......
3
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Example 3 (an even example)
0, − π < t < − π
 2
o f (t ) = 4, − π < t < π
2 2
 −π < t < π
0,
2
with period 2π . This is an even function as f (t ) = f (−t ) therefore there will be no
bn terms.
a 
By inspection, the dc component of the waveform  0  is:
 2
a0
o =2
2
Now we can work out the cosine components:

π
2
1
o an =
π ∫ f (t ) cos(nt ).dt
−π
2
π
2
1
o an =
π ∫ 4 cos(nt ).dt
−π
2
π
4 1  2 8   nπ 
o a n =  sin( nt ) = sin  
π n  −π 2 nπ   2 
The result is:
a0 N
o + ∑ [a n cos(nt )]
f (t ) ≈
2 n =1
which we can write expanded as:
8 8 8
o f (t ) ≈ 2 + cos t − cos 3t + cos 5t.......
π 3π 5π
Half-range Series
Sometimes an engineering function is not periodic but is only defined over a finite
interval. In cases like this Fourier analysis can still be used. In these cases we
attempt to make the waveform periodic in some manner. For example, we may reflect
the function in the vertical axis and then repeat in periodically so that the result is a
periodic even function. We have performed what is called a periodic extension of the
T
given function. Within the period of interest of t = 0 to t = , nothing has changed
2
but a periodic function has been constructed. We can find the Fourier Series of this
periodic function and within the period of interest this will converge to the required
function. What happens outside the period of interest is not important. In addition,
the periodic function is now even, so the Fourier Series will contain no sine terms.
An alternative method would also be to reflect the waveform in both the vertical and t
axes to gain an odd periodic function.
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Whichever method we choose, the resulting Fourier Series only gives a representation
T
of the original function in the interval 0 < t < and as such is termed as a half-range
2
Fourier Series. Similarly we refer to half-range sine series for odd series containing
only sine terms and half-range cosine series for even series containing only cosine
terms.
Clive Roberts
Tutorial Sheet 1
1. Find the Fourier Series representation of the function:
− 4, − π < t < 0
o f (t ) =  period 2π
 4, 0<t <π

o f (t ) = { t 2 + πt , − π < t < 0 period 2π

2(t + 1), − 1 < t < 0
o f (t ) =  period 2
 0, 0 < t <1

t 2 , − π < t < 0
o f (t ) =  period 2π
 0, 0 < t < π

− A, − T < t < 0
o f (t ) =  2 period T
 A, 0 < t <T
2
6. Find the half-range cosine series representing the function:

o f (t ) = {sin t 0 < t < π
7. Graph an appropriate periodic extension of:

{
o f (t ) = e t 0 < t < 1
and find its half-range cosine series.
8. Find the half-range sine series representing the function:

o f (t ) = {2 − t 0 < t < 2
9. Find the average power developed across a 2Ω resistor by a current signal

with a period 2π given by:
− t , − π < t < 0
o i (t ) =  period 2π
 0, 0<t <π
Clive Roberts
Applications of the Fourier Series

Parseval’s Theorem
If the function f(t) is periodic with period T and has Fourier coefficients of an and bn
the Parseval’s theorem states:
( )
π 2
1 a N
o f 2 (t ) = ∫ ( f (t )) 2 .dt = 0 + ∑ a n + bn
2 2
π −π 2 n =1
This is frequently used in power calculations.
Example 4
Find the average power developed across a 1Ω resistor by a voltage signal with a
period 2π given by:
sin 2t cos 3t
o v(t ) = cos t − +
3 2
Here, v(t) is already expressed as a Fourier Series with a1 = 1, a3 = 1 and b2 = − 1 .

2 3
2
All other Fourier coefficients are 0. The instantaneous power is v and hence the
R
average power over one period is given by: :
π
1
o v(t ) 2 = ∫ ( f (t )) 2 .dt
π −π
2 2
 1  1
o v(t ) = 1 +  −  +   = 1.36V
2 2
 3  2
2
v(t )
o Pav = R = 0.68W
2
Frequency Spectrum
In general, the Fourier Series of a periodic function with period T seconds contains
the fundamental sinusoid and numerous harmonics, some of which may be zero. The
plot of the magnitude of the frequency components is called the amplitude, or
1
frequency spectrum. The frequency components are spaced f 0 = hertz apart. On a
T
graph the spectrum is a series of points (or lines) and is called a discrete spectrum. A
discrete Fourier spectrum is characteristic of all periodic functions.
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Figure 1 - Discrete Fourier Spectrums of common waveforms
Complex Notation
An alternative notation for Fourier Series involving complex number is often used in
practical engineering applications. For this we use the Euler relationship:
o e ± jϑ = cos ϑ ± j sin ϑ
from which we can obtain expressions for cos ϑ and sin ϑ :

e jϑ + e − jϑ e jϑ − e − jϑ
o cos ϑ = sin ϑ =
2 2j
which enables us to rewrite our general approximation of the Fourier Series as:
a N
  2nπt   2nπt 
o f (t ) ≈ 0 + ∑ a n cos  + bn sin  
2 n =1   T   T 
a 0 N  e 
j 2 nπt − j 2 nπt j 2 nπt − j 2 nπt
+e −e
T T T
T
e
o f (t ) ≈ + ∑ an + bn 
2 n =1  2 2j 
 
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a 0 N  a n − jbn − j 2 nπt T a n + jbn − j 2 nπt T 

o f (t ) ≈ +∑ e + e
2 n =1  2 2 

which we can write as

N j 2 nπt
o f (t ) ≈ ∑ cn e
n=− N
T
where
a n − jbn a + jbn
o cn = c−n = n = 1,2,....
2 2
a0
and c0 = . It can be shown that the Fourier coefficients, c n , are the given by:
2
T
1 2 − j 2 nπt
o cn = ∫
T −T
f (t ).e T
.dt
2
Example 5
Find the complex Fourier series representation of the function with period T defined
by:
1, − T < t < T
o f (t ) =  4 4
0, otherwise
We find
T
1 4 − j 2 nπt T
T −T∫
o cn = e .dt
4
T
 − j 2 nπt T  4
1 e 
o cn =
T  − j 2nπt 
 T  −T 4
− 1  − jnπ 2 jnπ
o cn = e − e 2 
j 2nπ  
jnπ − jnπ
1  e 2 − e 2 

o cn =
nπ  2j 
 
1 nπ − j 2 nπ t
o cn = sin e T
nπ 2
therefore:
N
1 nπ − j 2 nπt T
o f (t ) = ∑
n = − N nπ
sin
2
e
Frequency Response of a Linear System

If a sine wave of amplitude Ai is applied to a linear system then the amplitude Ao of
the output is given by:
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o Ao = G ( jω ) Ai
and the phase shift, φ , is given by:

o φ = ∠G ( jω )
Note that Ao and φ are dependent upon ω . It is important to note that G ( jω ) is a

frequency dependent function.
It is now possible to analyse the effect of applying a generalised periodic waveform to

a linear system. The first stage is to calculate the Fourier components of the input
waveform. The amplitude and phase shift of each of the output components is then
calculated using G ( jω ) . Finally, the output components are added to obtain the
output waveform.
Example 6
Consider the low-pass filter in the diagram below. Using Kirchoff’s voltage law and
Ohm’s law we obtain:
o vi = iR + vo
Vin C Vout
For the capacitor:

i
o vo =
jωC
Eliminating i gives:
o vi = v o jωRC + vo = vo (1 + jωRC )
vo Ao 1
o = = G ( jω ) =
vi Ai 1 + jωRC
It is convenient to convert G ( jω ) to a polar form.

1∠0
o G ( jω ) =
1 + (ωRC ) 2 ∠ tan −1 ωRC
1
o G ( jω ) = ∠ tan −1 ωRC
1 + (ωRC ) 2
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Therefore:
1
o G ( jω ) =
1 + (ωRC ) 2
o ∠G ( jω ) = − tan −1 ωRC
Note that the circuit is a low-pass filter; it allows low frequencies to pass easily and
rejects high frequencies. The cut-off point of the filter, that is the at which significant
frequency attenuation begins to occur, can be varied by changing the values of R and
C. The quantity RC is usually known as the time constant for the system, τ . When
τ = 0.3 the equations of gain and phase shift reduce to:
1
o G ( jω ) =
1 + 0.09ω 2
o ∠G ( jω ) = − tan −1 0.3ω
If we examine the response of the system to a given input:

 1, 0<t <π
o f (t ) =  period is 2π
− 1, − π < t < 0
a 
The waveform is a rotated function so will not contain any dc component,  0  = 0 ,
 2
and will only contain odd harmonic sine components. Therefore calculating the
Fourier Series reduces to:
N
o f (t ) ≈ ∑ bn sin nt
n =1
and
π
1
o bn =
π ∫ f (t ) sin nt.dt
−π
therefore:
π
1 
0
o bn =  ∫ − sin nt.dt + ∫ − sin nt.dt 

π  −π 0 
π
1   cos nt   − cos nt  
0
o bn =  +
π   n  −π  n  0 
1
o bn = [(cos 0 − cos(− πn )) + (− cos πn − − cos 0)]
nπ
1
o bn = (2 − 2 cos πn ) = 2 (1 − cos πn)
nπ πn
The values of the first few (odd) coefficients are:

2 4
o b1 = (1 − cos π ) =
π π
2 4
o b3 = (1 − cos 3π ) =
3π 3π
Clive Roberts
2 4
o b5 = (1 − cos 5π ) =
5π 5π
The next stage is to evaluate the gain and phase changes of the Fourier components.
n=1
o ω1 = 1
1
o G ( jω 1 ) = = 0.96
1 + 0.09 × 1
o ∠G ( jω1 ) = − tan −1 0.3 = −16.7°
n=3
o ω3 = 3
1
o G ( jω 3 ) = = 0.74
1 + 0.09 × 9
o ∠G ( jω 3 ) = − tan −1 0.9 = −42.0°
n=5
o ω5 = 5
1
o G ( jω 5 ) = = 0.55
1 + 0.09 × 25
o ∠G ( jω 3 ) = − tan −1 1.5 = −56.3°
The output of the system can therefore be calculated as:

3.84 2.96 2.2
o v0 = sin (π − 16.7°) + sin(3π − 42.0°) + sin(5π − 56.3°)
π 3π 5π
Figure 2 - Input and Output of the low-pass filter

Clive Roberts
Fourier Transforms
Introduction
From the work to date we have seen that almost any periodic signal can be
represented as a linear combinations of sine and cosine waves of various frequencies
and amplitudes. All frequencies are integer multiples of the fundamental frequency.
However, many waveforms are not periodic, examples include pulse signals and noise
signals.
The Fourier techniques can still be useful by the applications of the Fourier
Transform. The Fourier Transform is used in communications engineering and signal
processing. For example, it can be used to analyse the processes of modulation,
which involves superimposing an audio signal on to a carrier signal, and
demodulation, which involves removing the carrier signal to leave the audio signal.
Under certain conditions it can be shown that a non-periodic function f(t), can be
expressed not as the sum of sine and cosine waves bus as an integral:
∞
o f (t ) = ∫ ( A(ω ) cos ωt + B (ω ) sin ωt )dω

0
where:
∞ ∞
1 1
o A(ω ) =
π ∫
−∞
f (t ) cos ωt .dt B(ω ) =
π ∫ f (t ) sin ωt.dt
−∞
Provided that:
1. f(t) and f’(t) are piecewise continuous in every finite interval;
∞
2.
−∞
∫ f (t ) dt exists.
At a point of discontinuity of f(t) the integral representation converges to the average

value of the right hand and left hand limits. As with Fourier Series, and equivalent
complex representation exists which is more commonly used:
∞
1
o f (t ) = ∫
2π − ∞
F (ω ).e jωt .dω
where:
∞
∫ f (t ).e
− j ωt
o F (ω ) = .dt
−∞
These equations form what is referred to as a Fourier Transform Pair. The Fourier
Transform of f(t) is F (ω ) which is sometimes written as ℑ{ f (t )} . Similarly f(t) is
the Inverse Fourier Transform of F (ω ) , usually denoted by ℑ −1 {F (ω )}.
The Fourier Transform of f(t) is defined to be:

∞
o ℑ{ f ( t )} = F (ω ) = ∫ f (t).e
− jω t
dt
−∞
Clive Roberts
There is no universal convention concerning the definition of these integrals and a
number of variants are still correct. Therefore, it is important to be aware of possible
differences when consulting text books.
Example 1
Find the Fourier Transform of the function f (t ) = u (t )e −t , where u(t) is the unit step
1 0 < t < ∞
function i.e.  .
0 − ∞ < t < 0
∞
∫ f (t)e
− jω t
o F (ω ) = dt
−∞
∞
o F (ω ) = ∫ e −t e − jωt dt
0
∞
o F (ω ) = ∫ e −(1+ jω ) t dt
0
∞
 e −(1+ jω )t 
o F (ω ) =  
 − (1 + jω )  0
1
o F (ω ) = as e −(1+ jω ) t → 0 as t → ∞
1 + jω
Some Prope rties of the Fourier Transform
Figure 1 - Homogeneity of the Fourier Transform

Clive Roberts
Figure 2 - Additivity of the Fourier Transform

The time and frequency domains are alternative ways of representing signals. The
Fourier Transform is the mathematical relationship between these two representations.
If a signal is modified in one domain, it will also be changed in the other domain,
although usually not in the same way. Mathematical operations, such as addition,
scaling and shifting, also have a matching operation in the opposite domain. These
relationships are called Properties of the Fourier Transform, how a mathematical
change in one domain results in a mathematical change in the other domain.
The Fourier Transform is linear, that is, it possesses the properties of homogeneity
and additivity.
Homogeneity means that a change in amplitude in one domain produces an identical

change in amplitude in the other domain, i.e. when the amplitude of a time domain
waveform is changed, the amplitude of the sine and cosine waves making up that
waveform must also change by an equal amount. This is shown in Figure 1, and
represented mathematically as:
o ℑ{ fk } = kℑ( f )
Clive Roberts
Additivity of the Fourier Transform means that the addition in one domain
corresponds to addition in the other domain. An example of this is shown in Figure 2.
Since the two time domain signals add to produce a new time domain signal, the two
corresponding spectra add to produce a new spectrum. This is represented
mathematically as:
o ℑ{ f + g } = ℑ{ f }+ ℑ{g}
Example 2
We saw in Example 1 that:
o {
ℑ u (t ) e − t =} 1
1 + jω
Also:
∞
o { }
ℑ u (t )e − 2t = ∫ u (t ).e −2t e − jωt .dt
−∞
∞
o {
ℑ u (t ) e − 2t
}= ∫ e − (2 + j ω ) t
.dt
0
∞
 e − ( 2+ jω ) t 
o {
ℑ u (t ) e − 2t
} = 
 − (2 + jω )  0
o {
ℑ u (t ) e − 2t } =
1
2 + jω
Therefore:
o {
ℑ u (t ) e − t + u (t ) e − 2t = } 1
+
1
1 + jω 2 + jω
2 + j ω + 1 + jω
o {
ℑ u (t ) e − t + u (t ) e − 2t } =
(1 + jω )( 2 + jω )
3 + 2 jω
o {
ℑ u (t ) e − t + u (t ) e − 2t } =
(1 + jω )( 2 + jω )
Figure 3 shows how the phase is affected when the time domain waveform is shifted
to the left or right. This is referred to as the First Shift Theorem. The magnitude has
not been included, as it is not changed by the shift. In Figures 3a to 3d the waveform
is gradually shifted from having a peak at 128 samples to have it at sample 0.
The time domain waveform in Figure 3 is symmetrical around a vertical axis, i.e. the
left and right sides are mirror images of each other. Signals with this type of
symmetry are referred to as having linear phase, because the phase of their frequency
spectrum is a straight line. Signals that don’t have the left-right symmetry are called
nonlinear phase, and have phase that are something other than a straight line.
When the time domain waveform is shifted to the right, the phase remains a straight
line, but experiences a decrease in slope. When the time domain is shifted to the left,
there is an increase in the slope. This is the main property you need to remember from
this section; a shift in the time domain corresponds to changing the slope of the phase.
Clive Roberts
This is represented mathematically by the First Shift Theorem:

{ }
o ℑ e jat f (t ) = F (ω − a) where a is a constant
Clive Roberts
Example 3
Show that the Fourier transform of:
3 − 2 < t < 2
o f (t ) = 
0 otherwise
6 sin 2ω
is given by F (ω ) = . Then use the shift theorem to find the Fourier
ω
Transform of e − jt f (t ) .
2
2
 e − jω t 
o F (ω ) = 3 ∫ e − jω t
dt = 3 
−2  − j ω  −2
 e −2 jω − e 2 jω 
o F (ω ) = 3 
 − jω 
 e 2 jω − e −2 jω  e jϑ − e − jϑ
o F (ω ) = 6  from Euler’s sin ϑ =
 2 jω  2j
6
o F (ω ) = sin 2ω
ω
We have ℑ{ f ( t )} = F (ω ) = sin 2ω . Using the first shift theorem with a = -1 we
6
ω
obtain:
o ℑ{e − jt f ( t )} = F (ω + 1) =
6
sin 2(ω + 1)
ω +1
Clive Roberts
Fourier Transforms (continued)
First and Second Shift Theorems

We can represent the First Shift Theorem (frequency shift) mathematically as:
{ }
o ℑ e jat f (t ) = F (ω − a) where a is a constant
We can represent the Second Shift Theorem (time shift) mathematically as:
o ℑ{ f ( t − a )} = e − jat F (ω ) where a is a constant
Spectra
In the Fourier analysis of periodic waveforms we stated that although a waveform
physically exists in the time domain it can be regarded as comprising of components
with a variety of temporal frequencies. The amplitude and the phase of these
components are obtained through Fourier coefficients an and bn . This is known as a
frequency domain description. Plots of amplitude versus frequency and phase versus
frequency are together known as the spectrum of a waveform. Periodic function have
discrete or line spectra as we have previously seen. Only a discrete set of frequencies
is required to synthesize a periodic waveform. However, when analyzing non-
periodic waveforms via the Fourier Transform we find that a continuous range of
frequencies is required. Instead of discrete spectra we have a continuous spectra. The
modulus of the Fourier Transform, F (ω ) , gives the spectrum amplitude, while it
argument arg( F (ω )) describes the spectrum’s phase.
Amplitude Modulation
Amplitude modulation is a technique that allows audio signals to be transmitted as
electromagnetic radio waves. The maximum frequency of audio signals is typically
10 kHz. If these signals were to be transmitted directly then it would be necessary to
use a very large antenna. This can be seen by calculating the wavelength of an
electromagnetic wave of frequency 10 kHz using the formula:
o c = fλ
where c is the velocity of an electromagnetic wave is a vacuum (3 × 10 8 ms-1 ), f is the

frequency of the wave and λ is its wavelength. λ therefore is equal to 30,000 m. It
can be shown that an antenna must have dimensions of at least one-quarter of the
wavelength of the signal being transmitted if it is to be reasonably efficient. Clearly a
very large antenna would be needed to transmit a 10 kHz signal directly. The solution
is to have a carrier signal of a much higher frequency than the audio signal which is
usually termed the modulation signal. This allows the antenna to be a reasonable size
as a higher frequency signal has a lower wavelength. The arrangement for mixing the
two signals is shown in Figure 1.
If an amplitude modulated signal is given by:

o φ (t ) = x (t ) cos ω c t
where ω c is the angular frequency of the carrier signal and x(t) is the modulation
signal. Using Euler’s Identities φ (t ) can be expressed as:
Clive Roberts
Antenna
x (t ) X φ (t ) = x( t ) cos ωc t
Audio Signal Modulated Signal
cosω c t
Carrier Signal
Figure 1 - Block diagram of amplitude modulation
e jωc t + e − jω ct
o φ (t ) = x (t )
2
Taking the Fourier Transform and using the first shift theorem gives (this represents a
time shift):
 ( e jω c t + e − jω ct ) 
o ℑ{φ (t )} = Φ(ω ) = ℑx (t ) 
 2 
 e jω ct x (t )   e − jω ct x(t ) 
o ℑ{φ (t )} = Φ (ω ) = ℑ  + ℑ 
 2   2 
o ℑ{φ (t )} = Φ(ω ) = ( X (ω − ω c ) + X (ω + ω c ))
1
2
where X (ω ) = ℑ{x (t )}, the frequency spectrum of the modulation signal.
Figure 2a shows an audio signal with a DC bias such that the signal always has a
positive value. Figure 2b shows that its frequency spectrum is composed of
frequencies from 300 Hz to 3 kHz, the range needed for voice communication, plus a
spike for the DC component. All other frequencies have been removed by analog
filtering. Figures 2c and 2d show the carrier signal, a pure sinusoid of much higher
frequency than the audio signal. In the time domain, amplitude modulation consists
of multiplying the audio signal by the carrier wave. As shown in 2e, this results in an
oscillatory waveform that has an instantaneous amplitude proportional to the original
audio signal. The envelope of the carrier wave is equal to the modulating signal. This
signal can be routed to an antenna, converted into a radio wave, and then detected by
a receiving antenna. This results in a signal identical to 2e being generated in the
radio receiver's electronics. A detector or demodulator circuit is then used to convert
the waveform in 2e back into the waveform in 2a.
Since the time domain signals are multiplied, the corresponding frequency spectra are
convolved. That is, 2f is found by convolving 2b and 2d. Since the spectrum of the
carrier is a shifted delta function, the spectrum of the modulated signal is equal to the
audio spectrum shifted to the frequency of the carrier. This results in a modulated
Clive Roberts
Figure 2 - The Amplitude Spectrum of the he Amplitude Modulated Signal with ω c = 3π
Figure 3 - Amplitude Modulation
spectrum composed of three components: a carrier signa;, an upper sideband, and a

lower sideband.
These correspond to the three parts of the original audio signal: the DC component,
the positive frequencies between 0.3 and 3 kHz, and the negative frequencies between
-0.3 and -3 kHz, respectively. Even though the negative frequencies in the original
audio signal are somewhat elusive and abstract, the resulting frequencies in the lower
sideband are as real as you could want them to be.
Clive Roberts
For example, consider the frequency spectrum for television transmission. A standard
TV signal has a frequency spectrum from DC to 6MHz. By using these frequency-
shifting techniques, 82 of these 6 MHz wide channels are stacked end-to-end. For
instance, channel 3 is from 60 to 66MHz, channel 4 is from 66 to 72 MHz, channel 83
is from 884 to 890 MHz, etc. The television receiver moves the desired channel back
to the DC to 6MHz band for display on the screen. This scheme is called frequency
Domain multiplexing.
Convolution and Correlation

Convolution is an important technique in signal and image processing. It provides a
means of calculating the response (output) of a system to an arbitrary input signal if
the impulse response is known. The impulse response is the response of the system to
an impulse function. Convolving the input signal and the impulse response results in
the response to the arbitrary system. Correlation is a second important technique. It
can be used to determine the time delay between a transmitted signal and a received
signal as they might occur in radar or sonar detection equipment.
Convolution and the Convolution Theorem

It f(t) and g(t) are two real piecewise continuous functions, their convolution, which is
denoted by f * g, is defined as:
∞
o f ∗g = ∫ f (λ ) g (t − λ )dλ
−∞
f * g is itself a function of t, so we sometimes will write (f * g)(t). Convolution is an
integral with respect to λ (a dummy variable).
The convolution theorem states that convolution in the time domain corresponds to
multiplication in the frequency domain:
o If ℑ{ f ( t )} = F (ω ) and ℑ{g (t )} = G(ω ) then…
o ℑ{ f ∗ g} = F (ω )G(ω )
This theorem gives us a technique for calculating the convolution of two function
using the Fourier Transform, since:
o f ∗ g = ℑ −1 {F (ω ) G(ω )}
So it is possible to find the convolution of f(t) and g(t) by:

1. finding
g the corresponding Fourier Transforms, F (ω ) and G (ω ) ;
2. multiplying these together to form F (ω ) G (ω ) ;
3. finding the inverse Fourier Transform which gives us f * g.
Correlation and Correlation Theorem

If f(t) and g(t) are two piecewise continuous function their correlation, which is
denoted as f ê g, is defined as:
∞
o fêg= ∫ f (λ ) g (λ − t )dλ
−∞
This formula is the same as that for convolution except that the function g is not
folded; that is we have g ( λ − t ) here rather than g ( t − λ ) .
Clive Roberts
Tutorial Sheet 2
1. Find the Fourier Transform of the function:
0 t < −π

o f (t ) = 1 −π ≤ t ≤ π
0 π <t


0 t < −2π
1 − 2π ≤ t < −π
 2
o f (t ) =  1 −π ≤ t ≤ π
1 π < t ≤ 2π
 2
 0 2π < t

0 t < −π
1 − π ≤ t < −π 2
 2

o f (t ) =  1 −π ≤ t ≤ π
2 2
1 π < t ≤π
 2 2
 0 π <t

0 t < −π 2

o f (t ) =  t −π 2 ≤ t ≤π 2
 π <t
0 2

 0 t < −π 2

o f (t ) = t + π 2 −π 2 ≤ t ≤ π 2
 π <t
 0 2

e − t t > 0
o f (t ) =  t
e t≤0
Clive Roberts
Laplace Transforms
The Laplace transform is a well-established mathematical technique for
solving differential equations. It is named in honor of the great French
mathematician, Pierre Simon De Laplace (1749-1827). Like all transforms, the
Laplace transform changes one signal into another according to some fixed set of
rules or equations. As illustrated in Figure 1, the Laplace transform changes a signal
in the time domain into a signal in the s-domain, also called the s-plane. The time
domain signal is continuous, extends to both positive and negative infinity, and may
be either periodic or aperiodic. The Laplace transform allows the time domain to be
complex.
Figure 1 - The Laplace Transform

As shown in Figure 1, the s-domain is a complex plane (i.e. there are real numbers
along the horizontal axis and imaginary numbers along the vertical axis). The distance
along the real axis is expressed by the variable, σ . Likewise, the imaginary axis uses
the variable, ω , the natural frequency. This coordinate system allows the location of
any point to be specified by providing values for σ and ω . Using complex notation,
each location is represented by the complex variable, s, where: s = σ + jω . Just as
with the Fourier transform, signals in the s-domain are represented by capital letters.
Clive Roberts
For example, a time domain signal, x (t ) , is transformed into an s-domain signal,
X (s ) , or alternatively, X (σ , ω ) . The s-plane is continuous, and extends to infinity in
all four directions.
Just as the Fourier transform analyses signals in terms of sinusoids, the

Laplace transform analyses signals in terms of sinusoids and exponentials. From
a mathematical standpoint, this makes the Fourier transform a subset of the
more elaborate Laplace transform. Figure 1 shows a graphical description of how the
s-domain is related to the time domain. To find the values along a vertical line in the
s-plane (the values at a particular σ ), the time domain signal is first multiplied by the
exponential curve: e −σt . The left half of the s-plane multiplies the time domain with
exponentials that increase with time (σ < 0), while in the right half the exponentials
decrease with time ( σ > 0). Next take the complex Fourier transform of the
exponentially weighted signal. The resulting spectrum is placed along a vertical line
in the s-plane, with the top half of the s-plane containing the positive frequencies and
the bottom half containing the negative frequencies. Take special note that the values
on the y-axis of the s-plane ( σ = 0 ) are exactly equal to the Fourier transform of the
time domain signal.
As discussed previously the Fourier Transform can be given as:

∞
∫ f (t ).e
− j ωt
o F (ω ) = .dt
−∞
This can be expanded into the Laplace transform by first multiplying the time domain
signal by the exponential term:
∞
∫ [ f (t ).e ].e
−σ t − j ωt
o F ( s ) = F (σ , ω ) = .dt
−∞
The location in the complex plane can be represented by the complex variable, s,
where s = σ + jω . This allows the equation to be reduced to:
∞
∫ x(t ).e
−st
o F (s) = .dt
−∞
Figure 2 shows three pairs of points in the s-plane: A and A’, B and B’, and C and C’.
Since each of these pairs has specific values for σ and ± ω , there are two waveforms
associated with each pair: cos(ωt ).e −σt and sin( ωt ).e −σt . For instance, points A and
A’ are at a location of σ = 1.5 and ω = ±40 , and therefore correspond to the
waveforms: cos( 40t ) e −1.5t and sin( 40t )e −1. 5t .
Figure 3 shows an example of a time domain waveform, its frequency spectrum and
its s-domain representation. The example time domain signal is a rectangular pulse of
width two and height one. As we have previously seen earlier in the course the
complex Fourier transform of this signal is a sinc function in the real part, and zero in
the imaginary part. The s-domain is an undulating two-dimensional signal.
Clive Roberts
Figure 2 - Waveforms associated with the s-domain
Figure 3 -Time, frequency and s-domains

Clive Roberts
An analogy of how the Laplace transform is used in signal processing is…. Imagine
you are traveling by train at night between two cities. Your map indicates that the
path is very straight, but the night is so dark you cannot see any of the surrounding
countryside. With nothing better to do, you notice an altimeter on the wall of the
carriage and decide to keep track of the elevation changes along the route. Being
bored after a few hours, you strike up a conversation with the conductor: “Interesting
terrain,” you say. “It seems we are generally increasing in elevation, but there are a
few interesting irregularities that I have observed”. Ignoring the conductor's obvious
disinterest (and the scared look in his eyes!!), you continue: “Near the start of our
journey, we passed through some sort of abrupt rise, followed by an equally abrupt
descent. Later we encountered a shallow depression”. Thinking you might be
dangerous or demented, the conductor decides to respond: “Yes, I guess that is true.
Our destination is located at the base of a large mountain range, accounting for the
general increase in elevation. However, along the way we pass on the outskirts of a
large mountain and through the center of a valley”. Now, think about how you
understand the relationship between elevation and distance along the train route,
compared to that of the conductor. Since you have directly measured the elevation
along the way, you can rightly claim that you know everything about the relationship.
In comparison, the conductor knows this same complete information, but in a simpler
and more intuitive form: the location of the hills and valleys that cause the dips and
humps along the path. While your description of the signal might consist of
thousands of individual measurements, the conductor's description of the signal will
contain only a few parameters.
To show how this is analogous to signal processing, imagine that we are trying to
understand the characteristics of some electric circuit. To aid in our investigation, we
carefully measure the impulse response and/or the frequency response. However, this
does not mean that you know the information in the simplest way. In particular, you
understand the frequency response as a set of values that change with freque ncy. Just
as in our train analogy, the frequency response can be more easily understood in terms
of the terrain surrounding the frequency response. That is, by the characteristics of
the s-plane. With the train analogy in mind, look back at Figure 3. How does the
shape of this s-domain aid in understanding the frequency response? It doesn't! The
s-plane in this example provides no insight into why the frequency domain behaves as
it does. This is because the Laplace transform is designed to analyze a specific
class of time domain signals: impulse responses that consist of sinusoids
and exponentials. If the Laplace transform is taken of some other waveform the
resulting s-domain is meaningless. Systems that belong to this class are
extremely common in science and engineering. This is because sinusoids and
exponentials are solutions to differential equations
One of the things we know about the impulse response is that it will be composed of
sinusoids and exponentials, because these are the solutions to the differential
equations that govern the system. Try as we might, we will never find this type of
system having an impulse response that is, for example, a square pulse or triangular
waveform. Also, the impulse response will be infinite in length. That is, it has nonzero
values that extend from t = 0 to t = ∞ . This is because sine and cosine waves have a
constant amplitude, and exponentials decay toward zero without ever actually
reaching it. If the system we are investigating is stable, the amplitude of the impulse
response will become smaller as time increases, reaching a value of zero at t = ∞ .
Clive Roberts
There is also the possibility that the system is unstable, for example, an amplifier that
spontaneously oscillates due to an excessive amount of feedback. In this case, the
impulse response will increase in amplitude as time increases, becoming infinitely
large. Even the smallest disturbance to this system will produce an
unbounded output.
The center column in Figure 5 shows the impulse response of an RLC notch filter. It
contains an impulse at t = 0, followed by an exponentially decaying sinusoid. As
illustrated in (a) through (e), we will probe this impulse response with various
exponentially decaying sinusoids. Each of these probing waveforms is characterized
by two parameters: ω , that determines the sinusoidal frequency, and σ , that
determines the decay rate. In other words, each probing waveform corresponds to a
different location in the s-plane, as shown by the s-plane diagram in Figure 4. The
impulse response is probed by multiplying it with these waveforms, and then
integrating the result from t = −∞ to t = ∞ . This action is shown in the right column.
Our goal is to find combinations of σ and ω that exactly cancel the impulse
response being investigated. This cancellation can occur in two forms: the area
under the curve can be either zero, or just barely infinite. All other results
are uninteresting and can be ignored. Locations in the s-plane that produce a
zero cancellation are called zeros of the system. Likewise, locations that produce the
“just barely infinite” type of cancellation are called poles. Poles and zeros are
analogous to the mountains and valleys in our train analogy, representing the terrain
“around” the frequency response.
Figure 4 - Pole-Zero Example

Clive Roberts
Figure 5 - Probing the Impulse Response

Clive Roberts
Laplace Transforms (cont.)

The Laplace Transform has the same properties to the Fourier Transform, namely:
Additivity
o l( f + g ) = l( f ) + l ( g )
To find the Laplace Transform of a sum of function, we sum the Laplace Transforms
of the individual functions.
Homogeneity:
o l( kf ) = kl( f )
If we multiply a function by a constant, k, then the corresponding transform is also

multiplied by k.
Example 1
Find the Laplace Transform of the following:
o 5t 2 − 2e t
o { } { }
l 5t 2 − 2e t = 5l t 2 − 2l e t { }
o { }
l 5t 2 − 2e t =
10
−
2
s 3
s −1
First Shift Theorem:

{ }
o l e − at f (t ) = F ( s + a )
We obtain F ( s + a) by replacing every s in F(s) by s + a. The variable s has been

shifted by an amount a.
Example 2
Use the first shift theorem to calculate the Laplace Transform of f (t ) = e −3t t sin 5t .
o l{t sin 5t } = 2
10s
from the tables
( s + 25) 2
o { }
l e −3t f (t ) = F ( s + 3) the first shift theorem
10( s + 3)
l{e −3t f (t )}=
( )
o
( s + 3)2 + 25
2
10( s + 3)
o { }
l e −3t f (t ) =
(
s 2 + 6 s + 34
2
)
Second Shift Theorem:
o l{u (t − d ) f ( t − d )} = e − sd F ( s )
Clive Roberts
The function, u (t − d ) f ( t − d ) , is obtained by moving u (t ) f (t ) to the right (delaying)
by an amount d.
Example 3
Given l{ f (t )} = , find l{u (t − 2) f (t − 2)}
2s
s +9
Use the second shift theorem with d = 2.

2 se −2 s
o l{( u( t − 2) f (t − 2)} =
s+9
Laplace Transforms of Derivatives and Integrals

Later we will use the Laplace Transform to solve differential equations. In order to
do this we need to be able to find the Laplace Transform of derivatives of functions.
Let f(t) be a function of t, and f’ and f’’ the first and second derivatives of f. The
Laplace Transform of f(t) is F(s). Then:
o l{ f '} = sF ( s ) − f (0 )
o l{ f ' '} = s 2 F ( s) − sf (0) − f ' ( 0)
where f (0) and f ' (0) are the initial values of f and f’. The general case for the
Laplace Transform to an nth derivative is thus:
{ }
o l f ( n) = s n F ( s ) − s n−1 f ( 0) − s n −2 f ' (0) − ... − f ( n−1) ( 0)
Another useful and important result is:

t  1
o l ∫ f (t ).dt  = F ( s)
0  s
Example 4
Given x(0) = 2 and x’(0) = -1, calculate the expressions for the Laplace Transform of
2 x ' '−3x'+ x .
o l{x '} = sX ( s ) − x (0) = sX ( s ) − 2
o l{x ' '} = s 2 X ( s ) − sx (0) − x ' ( 0) = s 2 X ( s ) − 2 s + 1
Therefore:
o l{2 x' '−3x '+ x} = 2( s 2 X ( s ) − 2 s + 1) − 3( sX ( s ) − 2) + X ( s)
o l{2 x' '−3x '+ x} = ( 2s 2 − 3s + 1) X ( s ) − 4 s + 8
where the Laplace Transform of x(t) is X(s).
Example 5
The voltage, v(t), across a capacitor of capacitance C is given by:
1 t
o v (t ) = ∫ i (t ).dt
C0
Clive Roberts
Taking the Laplace Transform gives:
1
o V (s) = I (s)
Cs
where the Laplace Transform of v(t) is V(s) and the Laplace Transform of i(t) is I(s).
Inverse Laplace Transforms

Before we attempt to solve differential equations using the Laplace Transform we
must consider the inverse Laplace Transform. We now must consider the problem of
finding a function f(t), having been given the Laplace Transform, F(s). The Laplace
Transform tables and the linearity properties of the Laplace Transform help us to do
this.
Example 6
16
Find the inverse Laplace Transform of
s3
−1 16 
l  3  = 8t
2
o
s 
from the Laplace Transforms Tables and knowledge of homogeneity
The inverse Laplace Transform of a fraction is often best found by expressing it as its
partial fractions, and finding the inverse transform of these.
Example 7
4s − 1
Find the inverse Laplace Transform of .
s2 − s
4s − 1 4s −1 A B
o = = +
s − s s( s − 1) s s − 1
2
To solve:
o 4 s − 1 = A( s − 1) + Bs
substituting for s = 1 gives:
o B = 3 and A = 1
therefore:
o l −1   + l −1 
1 3 
 = 1 + 3e
t
 s  s − 1
We can also solve the inverse Laplace Transform using complex numbers. We use
this case when all the factors in the denominator are linear (i.e. no quadratic factors).
Example 8
s+3
Find the inverse Laplace Transform of .
s + 6s + 13
2
First we factorise the denominator:

− b ± b 2 − 4 ac − 6 ± 36 − 4(13)
o s= =
2a 2
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− 6 ± − 16
o s= = −3 ± 2 j
2
The denominator can then be factorised as (s – a)(s – b) where a = −3 + 2 j and
b = −3 − 2 j .
s+3 s+3 A B
o = = +
s + 6s + 13 ( s − a )( s − b) s − a s − b
2
To solve:
o s + 3 = A ( s − b) + B ( s − a )
substituting for s = -3 + 2j gives:
o 2 j = A( −3 + 2 j − b) = A(4 j )
1 1
o A = and B =
2 2
therefore:
o l −1 {F ( s)} = l −1 
1 1  1 −1  1 
+ l  
2  s− a  2  s −b 
e at + e bt
o l −1 {F ( s)} =
2
o l −1 {F ( s)} = ( e ( −3+ 2 j) t + e (− 3− 2 j )t )
1
2
o l −1 {F ( s)} = e − 3t ( e 2 jt + e −2 jt )
1
2
o l −1 {F ( s)} = e − 3t (cos 2t + j sin 2t + cos 2t − j sin 2t )
1
2
o l {F ( s )} = e −3t cos 2t
−1
Clive Roberts
Solving Differential Equations using the Laplace

Transforms
In engineering, differential equations are commonly used to model dynamic systems.
These are systems that change with time. Examples include electronic circuits with
time-dependent currents and voltages, a chemical production line in which pressures,
tank levels, flow rates, etc. vary with time and a semiconductor device in which hole
and electron densities change with time.
Up until now we have considered how to find the Laplace Transform of a function of
time and how to find the inverse Laplace Transform. We now are going to apply this
to finding the particular solution to differential equations. The initial conditions are
automatically satisfied when solving an equation using the Laplace Transform. They
are contained in the transform of the derivative terms.
The Laplace Transform of the equation is found. This transforms the differential
equation into an algebraic equation. The transform of the dependent variable is found
and then the inverse transform is calculated to give the required particular solution.
Example 1
Solve the differential equation:
dx
o +x=0 x ( 0) = 3
dy
using the Laplace Transform.
The Laplace Transform of each term is found:

 dx 
o l( x) = X ( s ), l  = s X ( s) − x( 0)
 dy 
 dx 
o l  = l( x ' ) = s X ( s) − x( 0) = sX ( s ) − 3
 dy 
Therefore
o sX ( s ) − 3 + X ( s) = 0
o sX ( s ) + X ( s ) = 3
o ( s + 1) X ( s) = 3
3
o X (s) =
s +1
Taking the Laplace Transform using the Laplace Transform tables gives:
o x (t ) = 3e −t
Clive Roberts
Tutorial Sheet 3
1. Solve the following equation using Laplace Transforms:
o y ' '+ y '−2 y = −2 y (0) = 2, y ' ( 0) = 1

o x ' '+2 x '−2 x = e − t x (0) = x ' (0) = 0

o x ' '−5x '+6 x = 6t − 4 x (0) = 1, x' ( 0) = 2
4. Consider the circuit in Figure 1. Before the switch is closed, the capacitor has
a voltage V across it. If the switch is closed at t = 0 a current will flow in the
circuit and the voltage, v, across the capacitor decays with time. Derive a
differential equation for the circuit, and hence (using the Laplace Transform)
−t
prove that v = v( 0)e RC
for t ≥ 0 .
5. Suppose an electronic thermometer is used to measure the temperature of an

oven. The sensing element does not respond instantly to changes in the oven
temperature because it takes a long time for the element to heat up or cool
down. Provided the electronic circuitry does not introduce further time delays
then the differential equation that models the thermometer is given by:
dv
o τ m + vm = v 0
dt
where v m = measured temperature, v o = oven temperature, τ = time constant of
the sensor. For convenience the temperature is measured relative to the
ambient room temperature, which forms a base line for temperature
measurement.
If the sensing element of an electronic thermometer has a time constant of 2

seconds. If the temperature of the oven increases linearly at the rate of 3 o Cs-1
starting from an ambient room temperature of 20 o C at t = 0, calculate the
response of the thermometer to the changing oven temperature.
Figure 1
Clive Roberts
Laplace Transforms in Control Engineering
Transfer Functions
It is possible to obtain a mathematical model of an engineering system that consists of
one or more differential equations. We have seen how the solution to differential
equations can be found using the Laplace Transform - this leads to the concept of a
transfer function. Consider the differential equation assuming it models an
engineering system:
dx( t )
o + x( t ) = f (t ) x (0) = x0 = 0
dt
The f(t) represents the input to the system and x(t) represents the output, or response
of the system to the input f(t). Taking the Laplace Transform of the differential
equation gives:
o sX ( s) − x0 + X ( s ) = F ( s)
o (1 + x) X ( s ) = F ( s )
therefore:
X (s) 1
o = G ( s) =
F ( s) 1+ s
X ( s)
As we know from our studies in Systems Engineering, is called the transfer
F ( s)
function. It is the ratio of the Laplace Transform of the output to the Laplace
Transform of the input. It is usually denoted as G(s).
The transfer function provides an algebraic relationship between the input and the
output of an engineering system, thus it allow the analysis of dynamic systems based
on differential equations. The example above assumes zero initial conditions. This is
necessary to form the transfer function, without this assumption the relationship
between the input and the output would have been more complicated. The
relationship would vary depending on how much energy was stored in the system at
t = 0 . By assuming zero initial condition, the transfer function depends purely on the
system characteristics. If the effect of the combination of a systems input and initial
conditions is required, it is necessary to carry out a full solution of the differential
equations as previously considered. In practice however, there are many cases where
the simplified treatment provided by the transfer function is perfectly acceptable.
Transport Lag
Transport Lag is a term used to describe the time delay that may be present in certain
engineering systems. An example would be a conveyor feeding a furnace with coal
supplied by a hopper. Varying the opening at the base of the hopper will vary the
amount of fuel supplied to the furnace, but there is a time delay before this changed
quantity of fuel reaches the furnace. The time delay depends on the speed and length
of the conveyor. Mathematically the function describing variation in the quantity of
fuel entering the furnace is a time-shifted version of the function describing the
variation in the quantity of fuel placed on the conveyor. The refore, let:
o u (t ) q (t ) = quantity of fuel placed on the conveyor, where u(t) is the
unit step.
o u (t − d ) q( t − d ) = quantity of fuel entering the furnace
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o d = time delay introduced by the conveyor
o l = length of conveyor (m)
o v = speed of the conveyor (ms-1 )
The input to the conveyor is u (t )q (t ) . The output from the conveyor is

l
u (t − d ) q (t − q ) . If the conveyor is moving at a constant speed then v = and so
d
l
d = . The transfer function that models the conveyor belt can be obtained by using
v
the second shift theorem.
o Q d ( s) = l{u( t − d ) q(t − d )} = e − sd l{q (t )} = e − sd Q ( s )
Transport lags can cause difficulty when trying to control a system because of the
delay between taking a control action and its effect being felt. In this example,
increasing the quantity of fuel on the conveyor does not lead to an immediate increase
in fuel entering the furnace. The difficulty of controlling the furnace temperature
increases as the transport lag introduced by conveyor increases.
Coal
Conveyor Velocity, v
Conveyor Length, l
Furnace
Figure 1 – Coal is fed into the furnace via the conveyor belt
Poles, zeros and the s-plane
Most transfer functions for engineering systems can be written as rational functions
(i.e. as the ratio of two polynomials in s, with a constant factor, K).
P( s )
o G( s ) = K
Q( s )
P(s) is of order m, and Q(s) is of order n; for a practical system m < n. Hence G(s)
may be written as:
K ( s − z1 )( s − z 2 )...( s − z m )
o G( s ) =
( s − p1 )( s − p2 )...( s − p n )
The values of s that make G(s) zero are known as the system zeros and correspond to
the roots of P(s) = 0, that is s = z1 , z2 ,…., zm. The values of s that make G(s) infinite
are known as the system poles and correspond to the roots of Q(s) = 0, that is s = p1 ,
p2 , …., pn . Poles may be real or complex. Comple x poles always occur in complex
conjugate pairs whenever the polynomial Q(s) has real coefficients.
Engineers find it useful to plot these poles and zeros on an s-plane diagram. A
complex plane plot is used with a real axis label of σ and an imaginary axis label of
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j ω . Poles are marked as crosses and zeros are marked as small circles. Figure 2
shows an s-plane plot for the transfer function:
3( s − 3)( s + 2)
o G( s ) =
( s + 1)( s + 1 + 3 j )( s + 1 − 3 j )
The benefit of this approach is that it allows the character of a linear system to be
determined by examining the s-plane plot. The transient response of the system can
be easily visualised by the number and position of the system poles and zeros.
Pole-Zero Map
3
1
Imag Axis
-1
-2
-3
-2 -1.5 -1 -0.5 0 0.5 1 1.5 2 2.5 3
Real Axis
Figure 2 – Poles and zeros plotted for the transfer function: G ( s) = 3( s − 3)( s + 2)
( s + 1)( s + 1 + 3 j )( s + 1 − 3 j )
1
If X ( s ) = where the pole p1 is given by a + bj, then
s − p1
o x (t ) = e p1t = e at e jbt = e at (cos bt + j sin bt )
The real part of the pole, a, gives rise to an exponential term and the imaginary part,
b, gives rise to an oscillatory term. If a < 0 the response, x(t), will decrease to zero as
t →∞.
Given a system with transfer function G(s), input signal R(s) and output signal C(s),
then:
C (s)
o = G( s )
R( s )
o C ( s ) = G ( s ) R( s )
and so:
K ( s − z1 )( s − z 2 )...( s − z m )
o C ( s) = .R( s )
( s − p1 )( s − p 2 )...( s − pn )
The poles and zeros of the system are independent of the input that is applied. All
that R(s) contributes to the expression for C(s) is extra poles and zeros.
1
When R( s ) = (i.e. a unit step input):
s
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K ( s − z1 )( s − z 2 )...( s − z m )
o C ( s) =
( s − p1 )( s − p2 )...( s − p n ) s
A1 A2 An B
o C ( s) = + + .... + + 1
s − p1 s − p 2 s − pn s
Taking the inverse Laplace Transform gives us:
o c (t ) = A1e p1t + A2e p2t + .... + An e pn t + B1
If the system is stable then p1 , p2 , …., pn will have negative real parts and their
contribution to c(t) will vanish as t → ∞ .
The response caused by the system poles is often called a transient response because
it decreases with time for a stable system. The component of the transient response
due to a particular pole is independent of the system input and is determined by the
nature of the system poles. It is now possible to derive a series of rules relating the
transient response of the system to the position of the system poles in the s plane.
Rule 1
The poles may be either real or complex, but for a particular pole it is necessary for
the real part to be negative if the transient caused by the pole is to decay with time.
Otherwise the transient response will increase with time and the system will be
unstable. Poles of a linear system must all lie in the left half of the s-plane for
stability.
Rule 2
The further a plot is to the left of the imaginary axis, the faster its transient decays.
This is because its transient contains a larger negative exponential term. For example,
e-5t decays faster than e-2t . The poles near to the imaginary axis are termed the
dominant poles as their transients take longer to decay. It is quite common for
engineers to ignore the effects of poles that are more than five or six time further
away from the imaginary axis than the dominant poles.
Rule 3
For a real system, poles with imaginary components occur as complex conjugate
pairs. The transient resulting from this pair of pole has the form of a sinusoidal term
multiplying an exponential term. For a pair of stable poles, (i.e. with a negative real
part – see Rule 1), the transient can be sketched by drawing a sinusoid confined with a
decaying exponential envelope.
A complex system may have many poles and zeros but by plotting them on the s-
plane the engineer begins to get a feel for the character of the system. The form of the
transients relating to particular poles or pairs of poles can be obtained using the above
rules. The magnitude of the transients, that is the values of A 1 , A2, …., An depends on
the system zeros.
It can be shown that having a zero near a pole reduces the magnitude of the transient
relating to that pole. Engineers often deliberately introduce zeros into a system to
reduce the effect of unwanted poles. If a zero coincides with the pole, it cancels it and
the transient corresponding to that pole is eliminated.
Clive Roberts
Difference Equations and the z-Transform
Introduction
The Laplace Transform has been shown to be useful for the solution of differential
equations, construction of transfer functions and control theory. Laplace methods are
used when the variables are considered to be continuous. The z transform plays a
similar role for discrete systems. We will consider the z transform and its uses in
solving linear constant coefficient difference equations.
Difference Equations
Difference equations are the discrete equivalent of differential equations. The
terminology is similar and the methods of solution have several aspects in common.
Difference equations arise whenever an independent variable can have only discrete
values. They are a growing are of importance in engineering due to their usefulness
when dealing with discrete-time systems such as microprocessors.
Consider a simple difference equation:

o x[n + 1] − x[ n] = 10
The dependent variable is x, while the independent variable is n.
A solution to the difference equation is obtained by knowing the dependent variable

for each value of interest of the independent variable. Therefore, the solution takes
the form of a sequence. There are often many different sequences that satisfy a
difference equation. The general solution embraces all of these and all possible
solutions can be obtained from it.
Example 1
Show x[n ] = A2 n , where A is a constant, is a solution of:
o x[n + 1] − 2 x[ n] = 0
Therefore:
o x[n ] = A2 n
o x[n + 1] = A2 n+1 = 2 A2 n
o x[n + 1] − 2 x[ n] = 2 A2 n − 2 A2 n = 0
If in addition we are given a condition, perhaps x[0] = 3 , the constant A can be

equated. If x[n ] = A2 n , then x[0] = A2 0 = A , and therefore A = 3 . The solution is
thus x[ n] = 3(2 n ) . This is the specific solution and satisfies both the difference
equation and the given condition.
The order is the difference between the highest and lowest arguments of the
dependent variable. The equation:
o 3x[ n + 2] − x[ n + 1] − 7 x[ n] = n
is second order because the difference between n + 2 and n is 2.
o x[n + 1] − x[ n − 1] = 7 x[ n − 2]
Clive Roberts
is third order because the difference between n + 1 and n − 2 is 3. As with
differential equations, in general the higher the order the more difficult the difference
equation is to solve.
In some difference equations the dependent variable occurs only once. These are
considered to be of zero order. We refer to these as non-recursive difference
equations because calculations of the value of the dependent variable does not require
knowledge of the previous values. In addition, difference equations of order ≥ 1 are
referred to as recursive difference equations because their solution requires
knowledge of previous values of the dependent variable. The difference equation:
o x[n ] = n 2 + n + 1
has zero order and so is a non-recursive difference equation.
Sometimes an equation or expression can be written in different ways. We need to be

able to rewrite equations so that comparisons can be made. When general solutions to
equations are found usually the equation is initially written in a standard form,
therefore there may be a need to rewrite equations.
Example 2
Rewrite the equation so that the highest argument of the dependent variable is n + 1 .
o x[n + 3] − x[ n + 2] = 2n x[ 2] = 7
The highest argument in the equation is n + 3 ; this is reduced by 2 to n + 1 . To do
this n is replace by n − 2 .
o x[n + 1] − x[ n] = 2(n − 2) x[ 2] = 7
Block Diagram Representation of Difference Equations

Many engineering systems can be modelled by means of difference equations. It is
possible to represent a difference equation using a block diagram.
Difference equations operate on discrete time data and therefore a continuous signal is
sampled before use. In the most common form of sampling, a sample is taken at
regular intervals, T. We will refer to the sequence that has been obtained by sampling
a continuous waveform at intervals T as x[n].
Several components are used in the block diagram. The delay block (Figure 1) has the
effect of delaying the sequence by one sample interval (i.e. a delay of T).
x[n] T x[n-1]
Figure 1 - A delay block
Another block diagram element represents the addition of a constant to a sequence. A

sequence can be scaled by a constant also. Both of these elements are shown in
Figure 2.
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x[n] x[n]
+ x[n]+a x kx[n]
a k
Figure 2 - Adding a constant to a sequence and scaling a sequence by a constant
Example 3
A simple example of a discrete time filter is described by the difference equation:
o y[n ] − ay[ n − 1] = x[ n]
where x[n] is the input sequence, y[n ] is the output sequence and a is a constant. If
a is positive then the filter behaves as a low-pass filter which rejects high frequencies
but allows low frequencies to pass. If a is negative then the filter behaves as a high-
pass filter. A block diagram for the filter is shown in Figure 3. This is a recursive
filter because calculations of y[n ] requires knowledge of previous values of the
output sequence. The block diagram contains a feedback path – this is a feature of
recursive difference equations.
x[n] ∑ y[n]
x T
-ay[n-1]
y[n-1]
-a
Figure 3 - A discrete time filter
Example 4 - Numerical Solutions of Difference Equations

Given:
o x[n + 1] − x[ n] = n x[0] = 1
determine x[1], x[2] and x[3].
n =1
o x[1] − x[ 0] = 0
∴ x[1] = 1
n=2
o x[ 2] − x[1] = 1
∴ x[ 2] = 2
Clive Roberts
n =3
o x[3] − x[ 2] = 2
∴ x[3] = 4
Example 5
From Example 3 the solution for a low-pass discrete-time filter is:
o y[n ] − ay[ n − 1] = x[ n]
If a is positive then the filter is a low-pass filter. Choosing a = 0.5 we will examine
the response of the filter to a unit step input applied at n = 0, i.e.
0 n<0
o x[n ] = 
1 otherwise
Assuming y[n ] = 0 for n ≤ −1.
o y[0] = 0.5 y[ −1] + x[0]
o y[0] = 0.5(0) + 1
o y[0] = 1
Similarly:
o y[1] = 0.5[ y − 1] + x[1]
o y[0] = 0.5(1) + 1
o y[0] = 1.5
and so on.
Clive Roberts
z-Transforms
The sequence f [ k ], k ∈ N may have been created by sampling a continuous signal.
We define its z-transform to be:
∞
F ( z ) = Z {f [k ]} = ∑ f [ k ] z
−k
o
k =0
We can see that the z-transform creates an infinite series, thus:
o Z { f [k ]} = f [0] +
f [1] f [2] f [ 3]
+ 2 + 3 + ....
z z z
In most engineering applications it is not necessary to work with the infinite series as
it is often possible to express this in a closed form. The closed form is generally valid
for values of z within a region known as the radius of absolute convergence as will
become apparent during some of the examples given.
Example 1
1 k = 0
Find the z-transform of the sequence defined by f [k ] = 
0 k ≠ 0
∞
Z { f [k ]} = ∑ f [ k ] z
−k
o
k =0
Z { f [k ]} = f [0] +
f [1] f [2] f [ 3]
o + 2 + 3 + ....
z z z
Z { f [k ]} = 1 + + 2 + 3 + ....
0 0 0
o
z z z
o F ( z) = 1
This is sometimes called the Kronecker delta sequence, δ [k ] .
Example 2
Find the z-transform of the sequence defined by f [k ] = k , k ∈ N .
∞
Z { f [k ]} = ∑ kz
−k
o
k =0
Z { f [k ]} = 0 +
1 2 3
o + 2 + 3 + ....
z z z
1 2 3 
o Z { f [k ]} = 1 + + 2 + ....
z z z 
−2
If we use the binomial theorem to express 1 −  as an infinite series, we find that:
1
 z
z
o F ( z) =
( z − 1) 2
Note that the process of taking the z-transform converts the sequence f[k] into the
continuous function F(z).
Clive Roberts
Sampling a continuous signal
Most of the signals encountered in the real world are continuous in time. This means
they have a signal level for every value of time over a particular time interval of
interest. An example would be the temperature of a room. This type of signal can be
modelled using a continuous mathematical function in which for each value of t there
is a continuous signal level, f(t). Many engineering system have signals which are
only of interest at particular points (which are often equally spaced and separated by
time T). These are referred to as discrete signals. They are modelled by a
mathematical function that is only defined at certain points in time.
If we have a continuous signal, f(t), defined for t ≥ 0 which we sample (measure) at

intervals of time T we obtain a sequence of sample values of f(t), i.e. f[0], f[1], f[2], …
It can be shown that sampling a continuous signal does not lose the essence of the
signal provided that the sampling rate is sufficiently high. It is often convenient to
represent a discrete signal as a series of weighted impulses. The strength of each
impulse is the level of the signal at the corresponding point in time; using:
∞
o f * (t ) = ∑ f [ k ]δ (t − kT )
k =0
the * indicates that f(t) has been sampled. This is a useful mathematical way of
representing a discrete signal as the properties of the impulse function lend
themselves to a value that exists for a short interval of time. Note, no sampling
method has zero sampling time, but provided the sampling time is much smaller that
the sampling interval, then this is a valid mathematical model of a discrete system.
We can apply the z-transform directly to a continuous function, f(t), if we regard the
function as having been sampled at discrete intervals of time.
The Relationship Between the z Transform and the Laplace Transform

We have defined the z-transform independently of any other transforms. However,
there is a close relationship between the z-transform and the Laplace transform, the z-
transform is regarded as the discrete equivalent of the Laplace transform.
If the continuous signal f(t) is sampled at intervals of time, T, we obtain a sequence of

sampled values f [ k ], k ∈ N . As we already know this sequence can be regarded as a
train of impulses:
∞
o f * (t ) = ∑ f [ k ]δ (t − kT )
k =0
Taking the Laplace transform, we have:
∞ ∞
o l{ f * (t )} = ∫ e − st ∑ f [ k ]δ (t − kT )dt
0 k =0
∞ ∞
o l{ f * (t )} = ∑ f [ k ]∫ e − stδ (t − kT ) dt
k=0 0
assuming that it is permissible to interchange the order of summation and integration.
∞
l{ f * (t )} = ∑ f [ k ]e
− skT
o
k =0
Now making the variable z = esT
Clive Roberts
∞
l{ f * (t )} = ∑ f [ k ] z
−k
o
k =0
which is the definition of the z-transform.
Example 3
1
Consider the function f (t ) = u (t )e −t which has the Laplace transform F ( s) = .
s +1
Suppose f(t) is sampled at intervals T to give the sequence f [k ] = e − kT , for k = 1, 2,
3…
a) Find the z-transform of f[k]
b) Make the change of variable z = e sT to obtain F * ( s)
c) Show that provided the sample interval T is sufficiently small, TF*(s)
approximates the Laplace transform F(s)
Clive Roberts
Coursework Assignment
During the course we have studied the use of Fourier Series, Fourier Transform,
Laplace Transform and z-Transform. As we have discussed, these tools are very
useful in many engineering applications.
You should write a 2500-4000 word report summarising qualitatively the principle of
each of the techniques, typical application areas and variants of the techniques we
have considered. In addition, for each technique you should detail a real life
engineering application you have found through research (library, web, research
papers) explaining why the chosen technique is used and how the work has been
implemented.
Your report should be submitted to reception by Monday 27th January 2003. Late
submissions will be penalised at 2% per day.
Clive Roberts
z-Transforms (cont.)
Because of the relationship between the Laplace and z transforms many of the
properties are similar.
Linearity
If f[k] and g[k] are two sequences, then:
o Z { f [ k ] + g[ k ]} = Z { f [ k ]} + Z {g[ k ]}
First Shift Theorem

If f[k] is a sequence and F(z) is its transform, then:
o Z { f [ k + i ]} = z i F ( z ) − ( z i f [0] + z i−1 f [1] + .... + zf [i − 1])
In particular, if i = 1:
o Z { f [ k + 1]} = zF ( z ) − zf [ 0]
If i = 2:
o Z { f [ k + 2]} = z 2 F ( z ) − z 2 f [0] − zf [1]
Second Shift Theorem

The function f(t)u(t) is defined by:
 f (t ) t ≥ 0
o f (t )u (t ) = 
 0 t<0
where u(t) is the unit step function. The function f (t − iT ) u(t − iT ) , where i is a
positive integer, represents as shift to the right (a time delay) of i sample intervals.
The second shift theorem states:

o Z { f [ k − i ]u[ k − i ]} = z − i F ( z )
Inversion of the z-transform

Just as it is necessary to invert Laplace transforms we need to be able to invert z-
transforms, and as before we make use of the tables, partial fractions and shift
theorems.
Example
Find the sequence whose z-transform is:
2z 2 − z
o F ( z) =
( z − 5)( z + 4)
The first stage is to divide the expression for F(z) by z:

F (z) 2z −1
o =
z ( z − 5)( z + 4)
We now split the RHS into partial fractions using the standard techniques, giving:
F (z) 1 1
o = +
z z−5 z+4
Clive Roberts
Multiplying back through by z gives:
z z
o F ( z) = +
z −5 z + 4
Which we now invert using the tables giving us f [k ] = 5 k + ( −4) k
The z-transform and Difference Equations

As we have previously discussed, the z-transform is used to solve difference
equations.
Example
Solve the difference equation y[k + 1] − 3 y[k ] = 0 , y[0] = 4 .
Taking the z transform of both sides of the equation gives us:

o Z {y[ k + 1] − 3 y[ k ]} = Z {0} = 0
Since Z{0} = 0. Using the properties of linearity we find:

o Z {y[ k + 1]}− 3Z {y[k ]} = Z {0} = 0
Using the first shift theorem on the first of the terms on the LHS we obtain:
o zZ {y[k ]} − 4 z − 3Z {y[ k ]} = 0
Writing Z {y[ k ]} = Y ( z ) , this becomes:

o ( z − 3)Y ( z ) = 4 z
so that:
4z
o Y ( z) =
z −3
The function on the RHS is the z-transform of the required solution. Inverting this
using the tables we find y[k ] = 4( 3) k .
Tutorial Sheet 4
1. Use z-transforms to solve x[k + 1] − 3 x[ k ] = −6 , x[0] = 1
Answer - x[k ] = 3 − 2( 3k )
2. Use z-transforms to solve 2 x[ k + 1] − x[ k ] = 2 k , x[0] = 2

2k ( 0.5) k
Answer - +5
3 3
3. Use z-transforms to solve x[k + 1] + x[k ] = 2k + 1 , x[0] = 0

Answer - k
4. Use z-transforms to solve x[k + 2] − 8 x[ k + 1] + 16 x[k ] = 0 , x[0] = 10 , x[1] = 20

Answer - 10(4 k ) − 5( k 4 k )
Clive Roberts
Probability and Statistics

Introduction to Probability
Consider a machine which manufactures electronic components. These must meet a
certain specification. The quality control department regularly samples the
components. Suppose, on average, 92 out of 100 components meet the specification.
Imagine that a component is selected at random and let A be the outcome that a
component meets the specification; let B be the outcome that a component does not
meet the specification. Therefore we say the probability of A occurring is
100 = 0.92 and the probability of B occurring is 100 = 0 .08 . The probability is
92 8
therefore considered to be a measure of the likelihood of the occurrence of a particular
outcome.
We should note that the probability of all possible outcomes is 1. The process of
selecting a component is called a trial. The possible outcomes are called events. In
the above example there are two possible events, A and B.
If we attempt to define probability in a more formal way… Let E be an event. We

wish to obtain the probability that E will occur, i.e. P(E), when a trial takes place. In
order to do this we repeat the trial a large number of times, n. We count the number
of times that an event E occurs (denoted by m). We can conclude that:
m
o P( E ) =
n
The larger the number of trials that take place, the more confident we are of our
estimate of the probability of E occurring. For example, consider the trial of tossing a
coin. If we wish to calculate the probability of a head occurring then measuring the
result of 1000 tosses of the coin is likely to give a more accurate estimate that
measuring the results from 10 tosses.
Example 1
Machine A and B make components, which are then placed on a conveyor belt. Of
those made by machine A, 93% are acceptable. Of those made by machine B, 95%
are acceptable. Machine A makes 60% of the components and machine B makes the
rest. Find the probability that a component selected at random from the conveyor belt
is:
a) made by machine A
b) made by machine A and acceptable
c) made by machine B and acceptable
d) made by machine B and unacceptable
Clive Roberts
Mutually Exclusive Events
If we have a machine that manufactures car components. Suppose each components
falls into one of the following categorie s:
o top quality;
o OK;
o not very good;
o rubbish.
After many samples have been taken and tested, it is found that under certain
circumstances the probability that a component falls into a categories is as follows:
Category Probability
top quality 0.18
OK 0.65
not very good 0.12
rubbish 0.05
The four categories cover all possibilities and so the probability must sum to 1.
Example 2
Using the data in the table calculate the probability that a component selected at
random is either standard or top quality.
In this example the events A and B could not possibly occur together. A component is
either top quality or OK but cannot be both. We therefore define A and B as being
mutually exclusive. If Ei and Ej are mutually exclusive we denote this by:
o Ei ∩ E j = ∅
where ∅ represents an empty set, i.e. we are saying that Ei ∩ E j is an impossible

event and can never occur.
Complementary Events
Consider an electronic circuit. Either the circuit work or it doesn’t. Let A be the
event that the circuit works correctly, and B be the event that the circuit does not work
correctly. Either A or B must happen when the circuit is switched on, and one
excludes the other. The events A and B are said to be complementary.
Two events are considered to be complementary if they are mutually exclusive and in
a single trial where either A or B must happen. Therefore:
o P( A) + P( B ) = 1
and
o P( A ∩ B ) = 0
Instead of using the events A and B we tend to refer to the events as A and A .
If the probability that a component last 3 years or less is the event D and this has the
probability P(D) = 0.08. We can say:
o D: the component last 3 years or less;
Clive Roberts
o D : the component last more than 3 years.
D and D are complementary events so:

o P( D ) + P( D ) = 1
o P( D ) = 1 − P ( D ) = 1 − 0.08 = 0.92
Multiplication Law
If two machines, M and N, both manufacture components. Of the components made
by machine M , 92% are of an acceptable standard and 8% are rejected. For machine
N, only 80% are of an acceptable standard and 20% are rejected.
If all the components are manufactured by machine M the P(E) = 0.92, and if all the
components are manufacture by machine N the P(E) = 0.8 (where E is the event that a
component is of an acceptable standard).
If half the components are manufactured by machine M and half by machine N then
P(E) = 0.86. Why??
This leads to the idea of conditional probability. If we define events:

o A: the component is manufactured by machine M;
o B: the component is manufactured by machine N.
Then the probability that a component is of an acceptable standard, given it is

manufactured by machine M, is written as P(E|A). This is interpreted as the
conditional probability of E given A. Similarly P(E|B) is the probability of E
happening, given B has already happened.
Consider events A and B for which are mutually exclusive ( A ∩ B = ∅ ). Suppose we

know that event A has occurred and we need to find the probability that B occurs, i.e.
P(B|A). Knowing that event A has occurred we can restrict our attention to set A.
Event B will occur if any outcome is in A ∩ B . Therefore:
P( A ∩ B)
o P( B | A) =
P( A)
The multiplication law therefore states:
o P( A ∩ B) = P ( B | A).P( A)
Example 3
A manufacturer studies the reliability of a certain component so that suitable
guarantees can be given: 83% of components remain reliable for at least 5 years; 92%
remain reliable for at least 3 years. What is the probability that a component which
has remained reliable for 3 years will remain reliable for 5 years?
Clive Roberts
For independent events E1 and E2 :
o P( E1 | E2 ) = P( E1 )
o P( E 2 | E1 ) = P( E 2 )
o P( E1 ∩ E2 ) = P ( E1 ) P( E 2 )
The concept of independence maybe applied to more than two events. Three or more
events are independent if every pair is independent.
A compound event may comprise several independent events. The multiplication law
is extended in an obvious way:
o P( E1 ∩ E2 ∩ E 3 ) = P ( E1 ) P( E 2 ) P( E 3 )
o P( E1 ∩ E2 ∩ E3 ∩ E4 ) = P ( E1 ) P( E 2 ) P( E 3 ) P ( E4 )
Example 4
The probability that a component is fault is 0.04. Two components are picked at
random. Calculate the probability that:
a) both components are faulty;
b) both components are not faulty;
c) one of the components is faulty;
d) one of the components is not faulty;
e) at least one of the components is faulty;
f) at least one of the components is not faulty.
Clive Roberts
Example Class Questions

Question 1
The life spans of 5000 electrical components are measured to assess their reliability.
The life span (L) is recorded and the results are shown in the table below. Find the
probability that a randomly selected component will last for:
a) more than 3 years?
b) between 3 and 5 years?
c) less than 4 years?
Lifespan of components Number

L>5 500
4< L≤5 2250
3< L ≤ 4 1850
L≤3 400
Question 2
Machines M and N manufacture a component. The probability that the component is
of an acceptable standard is 0.95 when manufactured by machine M and 0.84 when
manufactured by machine N. Machine M supplies 65% of components; machine N
supplies 35%. A component is picked at random.
a) What is the probability that the component is of an acceptable
standard?
b) What is the probability that a component is of an acceptable standard
and is made by machine M?
c) What is the probability that the component is of an acceptable standard
given it is made by machine M?
d) What is the probability that the component was made by machine M?
e) What is the probability that the component was made by machine M
given it is of an acceptable standard?
f) The component is not of an acceptable standard. What is the
probability that it was made by machine N?
Question 3
Machine 1 manufactures an electronic chip, A, of which 90% are acceptable.
Machine 2 manufactures an electronic chip, B, of which 83% are acceptable. Two
chips are picked at random (one of each kind). Find the probability that they are both
acceptable.
Question 4
Machines 1, 2 and 3 manufacture resistors A, B and C respectively. The probabilities
of their respective acceptabilities are 0.9, 0.93 and 0.8. One of each resistor is
selected at random.
a) Find the probability that they are all acceptable;
b) Find the probability that at least one resistor is acceptable.

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EE2D1 – Fundamentals of Signals and Systems (Maths) – Course Overview

Reading / Self-Study Week

Week 9 (Reading / Self-study Week)

Fourier Analysis can be broken into four categories:

Figure 1 - The four categories of Fourier Anaylsis

Periodic Waveforms (revision)

• It is particularly important that you understand how to integrate these

cos nωt sin nωt

• Sometimes a function occurs as the sum of a number of different sine and

Assuming that f(t) is continuous on the interval − π ≤ t ≤ π , the coefficients a n and

for the constant term and:

o a n = ∫ f (t ) cos( nt ).dt bn = ∫ f (t ) sin( nt ).dt

Using the equation for a 0 we get:

Similarly the sine terms are computed as:

The series approximation can be rewritten using the identity relationship

One to try before the next lecture…..

Figure 2 - Fourier series approximations for Example 1

Fourier Series (cont.)

Assuming that f(t) is continuous on the interval − T ≤ t ≤ T , the coefficients an

Based on these results, if f(t) is an even function:

Based on these results, if f(t) is an odd function:

Example 2 (an odd example)

Now we can work out the sine components:

Now we can work out the cosine components:

2. Find the Fourier Series representation of the function:

3. Find the Fourier Series representation of the function:

4. Find the Fourier Series representation of the function:

5. Find the Fourier Series representation of the function:

6. Find the half-range cosine series representing the function:

7. Graph an appropriate periodic extension of:

8. Find the half-range sine series representing the function:

9. Find the average power developed across a 2Ω resistor by a current signal

Applications of the Fourier Series

This is frequently used in power calculations.

Here, v(t) is already expressed as a Fourier Series with a1 = 1, a3 = 1 and b2 = − 1 .

Figure 1 - Discrete Fourier Spectrums of common waveforms

from which we can obtain expressions for cos ϑ and sin ϑ :

a 0 N  a n − jbn − j 2 nπt T a n + jbn − j 2 nπt T 

which we can write as

Frequency Response of a Linear System

and the phase shift, φ , is given by:

Note that Ao and φ are dependent upon ω . It is important to note that G ( jω ) is a

It is now possible to analyse the effect of applying a generalised periodic waveform to

For the capacitor:

It is convenient to convert G ( jω ) to a polar form.

If we examine the response of the system to a given input:

The values of the first few (odd) coefficients are:

The output of the system can therefore be calculated as:

Figure 2 - Input and Output of the low-pass filter

o f (t ) = ∫ ( A(ω ) cos ωt + B (ω ) sin ωt )dω

At a point of discontinuity of f(t) the integral representation converges to the average

The Fourier Transform of f(t) is defined to be:

Some Prope rties of the Fourier Transform

Figure 1 - Homogeneity of the Fourier Transform

Figure 2 - Additivity of the Fourier Transform

Homogeneity means that a change in amplitude in one domain produces an identical

This is represented mathematically by the First Shift Theorem:

Fourier Transforms (continued)

First and Second Shift Theorems

where c is the velocity of an electromagnetic wave is a vacuum (3 × 10 8 ms-1 ), f is the

If an amplitude modulated signal is given by:

Figure 1 - Block diagram of amplitude modulation