UNIVERSITY OF GONDAR

INSTITUTE OF TECHNOLOGY

DEPARTMENT OF ELECTRICAL ENGINEERING

Time and Frequency Domain Analysis of Signal and Systems

Lecture Note

By Belaineh Eskezia

March-10-2023
 Signal and Systems Analysis Lecture Note prepared for Exit Exams

Contents
Chapter 1 Fundamentals of Signals ............................................................................................................ 1

1.1 Introduction to Signals and Systems: .................................................................................................. 1

1.1.1 What is a Signal ? ......................................................................................................................... 1

1.1.2 What is a System ? ........................................................................................................................ 1

1.2 General signal characteristics: ............................................................................................................. 1

1.3 Classification of signals ....................................................................................................................... 2

1.3.1 Continuous-Time and Discrete-Time Signals .............................................................................. 2

1.3.2 Analog and Digital Signals ........................................................................................................... 3

1.3.3 Periodic and Aperiodic Signals .................................................................................................... 3

1.3.4 Even and Odd Signals ................................................................................................................... 5

1.3.5 Causal, Noncausal, and Anticausal Signal ................................................................................... 6

1.3.6 Energy and Power Signal.............................................................................................................. 6

1.3.7 Deterministic and Random Signals ............................................................................................ 11

1.4 Basic or Elementary signals............................................................................................................... 11

1.4.1 Exponential Signals .................................................................................................................... 11

1.4.2 Impulse functions........................................................................................................................ 12

1.4.3 The Unit Step Function ............................................................................................................... 14

1.4.4 Ramp function: ........................................................................................................................... 14

1.4.5 The Rectangular Pulse Function ................................................................................................. 15

1.4.6 The Signum Function ................................................................................................................. 15

1.4.7 The Complex Exponential Function ........................................................................................... 15

1.4.8 The Sinc Function ....................................................................................................................... 16

1.5 Basic Operations of Signals ............................................................................................................... 16

1.5.1 Time Shift ................................................................................................................................... 16

1.5.2Time Reversal .............................................................................................................................. 17

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1.5.3 Time Scaling ............................................................................................................................... 17

1.5.4 Decimation and Expansion ......................................................................................................... 18

1.5.5 Combination of Operations......................................................................................................... 18

1.6 Operation performed on dependent variable: .................................................................................... 19

1.6.1 Amplitude scaling ....................................................................................................................... 19

1.6.2 Amplitude addition: .................................................................................................................... 19

1.6.3 Amplitude multiplication: ........................................................................................................... 19

1.6.4 Differentiation............................................................................................................................. 19

1.6.5 Integration: .............................................................................................................................. 19

Chapter 2 Fundamentals of Systems....................................................................................................... 21

2.1 Properties of systems: ........................................................................................................................ 21

2.1.1 Static (Memoryless) and Dynamic (with memory): ................................................................... 21

2.1.2 Stable & unstable system ............................................................................................................ 22

2.1.3 Causal and non-Causal system: .................................................................................................. 22

2.1.4 Time variant and time invariant System ..................................................................................... 22

2.1.5 Linear and non-linear system: .................................................................................................... 23

2.1.6 Invertible and non-invertible system .......................................................................................... 24

2.1.7 Linear –time invariant system (LTI)........................................................................................... 25

2.2 Convolution ....................................................................................................................................... 26

2.2.1 Resolution of a Discrete-time signal into impulses: ................................................................... 26

2.3 Impulse response and convolution sum: ............................................................................................ 27

2.3.1 Impulse response ........................................................................................................................ 27

Response of LTI system to arbitrary inputs: The convolution sum..................................................... 27

Properties of convolution ..................................................................................................................... 28

How to Evaluate Convolution? ............................................................................................................ 28

The Convolution Integral ..................................................................................................................... 29

Some Properties of the Convolution Integral ...................................................................................... 30

Convolution with an Impulse............................................................................................................... 30

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Graphical Convolution......................................................................................................................... 30

Chapter 3 Fourier Series ........................................................................................................................... 35

3.1 Eigenfunctions of an LTI System ...................................................................................................... 35

3.1.1 Continuous-time Case ................................................................................................................. 35

3.1.2 Discrete-time Case ...................................................................................................................... 36

3.1.3 Summary ..................................................................................................................................... 37

3.1.4 Why is eigenfunction important?................................................................................................ 37

3.2 Fourier Series Representation ............................................................................................................ 38

3.2.1 Continuous-time Fourier Series Coefficient ............................................................................... 39

3.2.2 Discrete-time Fourier Series coefficients ................................................................................... 41

3.2.3 How do we use Fourier series representation? ........................................................................... 43

3.2.4 How many Fourier series coefficients are sufficient? ................................................................ 43

3.3 Trigonometric Fourier Series Representation .................................................................................... 44

3.4 Properties of Fourier Series Coefficients ........................................................................................... 46

Chapter 4 Continuous Time Fourier Transform ....................................................................................... 48

4.1 Insight from Fourier Series ................................................................................................................ 48

4.2 Fourier Transform .............................................................................................................................. 49

4.3 Relation to Fourier Series .................................................................................................................. 51

4.4 Examples............................................................................................................................................ 53

4.5 Properties of Fourier Transform ........................................................................................................ 55

4.6 System Analysis using Fourier Transform ........................................................................................ 58

Chapter 5 Discrete-Time Fourier Transform ............................................................................................ 61

5.1 Review on Continuous-time Fourier Transform................................................................................ 61

5.2 Deriving Discrete-time Fourier Transform ........................................................................................ 62

5.3 Why is X(ejω) periodic? ..................................................................................................................... 63

5.4 Properties of Discrete-time Fourier Transform ................................................................................. 64

5.5 Examples............................................................................................................................................ 65

Chapter 6 Sampling Theorem ................................................................................................................... 68

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6.1 Analog to Digital Conversion ............................................................................................................ 68

6.2 Frequency Analysis of A/D Conversion ............................................................................................ 69

6.2.1 How does P(jω) look like? .......................................................................................................... 69

6.2.2 How does Xp(jω) look like? ........................................................................................................ 70

6.2.3 What happens if T becomes larger and larger? .......................................................................... 71

6.2.4 What is the minimum sampling rate such that there is no aliasing? ........................................... 72

6.3 Sampling Theorem............................................................................................................................. 73

6.3.1 Explanation ................................................................................................................................. 73

6.4 Digital to Analog Conversion ............................................................................................................ 74

6.4.1 Given Xp(t) (no aliasing), how do I recover x(t)? ....................................................................... 74

6.4.2 If Xp(t) has aliasing, can I still recover x(t) from xp(t) ? ........................................................... 75

6.4.3 What can I do if my sampling device does not support a very high sampling rate? .................. 76
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Chapter 1 Fundamentals of Signals

1.1 Introduction to Signals and Systems:
1.1.1 What is a Signal ?
 A signal is formally defined as a function of one or more variables that conveys information on the
nature of a physical phenomenon.
 When the function depends on a single variable, the signal is said to be one dimensional.
Eg.; Speech signal (Amplitude varies with respect to time)
 When the function depends on two or more variables, the signal is said to be multidimensional. E.g.;
Image – 2D (Horizontal & vertical coordinates of the images are two dimensional)
1.1.2 What is a System ?
A system is formally defined as an entity that manipulates one or more signals to accomplish a function,
thereby yielding new signals.
The system can be described mathematically by an operator O acting on the function x(t) to provide the
output function y(t)

e.g.; In a communication system the input signal could be a speech signal or computer data. The system
itself is made up of the combination of a transmitter, channel and a receiver. The output signal is an
estimate of the information contain in the original message

The examples of other systems are control systems, biomedical signal processing system, audio system,
remote sensing system, microelectronic mechanical system etc.

1.2 General signal characteristics:

a) Multichannel & multidimensional signals: A signal is described by a function of one or more
independent variables. The value of the function (dependent variable) can be real valued scalar
quantity,a complex valued quantity or perhaps a vector.
Real valued signal, x1 (t) = A sin(3πt)
Complex valued signal x2(t) = Ae j3 πt = A cos(3πt) + jAsin(3πt)

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In some applications, signals are generated by multiple sources or multiple sensors. Such signals can be
represented in vector form and we refer such a vector of signal as a multichannel signal.
E.g.; In electrocardiography, 3-lead & 12-lead electrocardiograms (ECG) are often used, which result in
3-channel & 12-channel signals.
One dimensional: If the signal is a function of a single independent variable, the signal is called 1-D
signal. e.g.; Speech signal

Multidimensional signal: Signals can be functions of more than one variable, e.g., image signals
(2D),Colour image (3D), etc

1.3 Classification of signals

Signals can be classified in several ways. Some important classifications of signals are:
1.3.1 Continuous-Time and Discrete-Time Signals
Continuous-time signals are defined for a continuous of values of the independent variable. In the case of
continuous-time signals, the independent variable t continuous.
Discrete-time signals are defined only at discrete times, and for these signals, the independent variable n
takes on only a discrete set of amplitude values as shown in

Figure (a) Continuous-time signal, (b) discrete-time signal

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1.3.2 Analog and Digital Signals

An analog signal is a continuous-time signal whose amplitude can take any value in a continuous range. A
digital signal is a discrete-time signal that can only have a discrete set of values. The process of
converting a discrete-time signal into a digital signal is referred to as quantization.

1.3.3 Periodic and Aperiodic Signals

A signal x(t) is said to be periodic with period T(a positive nonzero value), if it exhibits periodicity, i.e.,
x(t +T) = x(t), for all values of t .Periodic signal has the property that it is unchanged by a time shift of T
A signal that does not satisfy the above periodicity property is called an aperiodic signal.

Figure (a) Periodic signal, (b) aperiodic signal

Example For each of the following signals, determine whether it is periodic or aperiodic. If periodic, find
the period
I. x(t) = 5 sin(2πt)
II. x(t) = 1 +cos(4t +1)
III. x(t) = e-2t
𝜋
IV. x (t)= 𝑒 𝑗(5𝑡+ 2 )

π
V. x(t) = ej(5t+ 2 )𝑒 −2𝑡

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Proposition Let continuous-time signals x1(t) and x2(t) be periodic signals with fundamental periods T1
and T2, respectively. The signal x(t) that is a linear combination of x1(t) and x2(t) is periodic if and only if
there exist integers m and k such that mT1 =kT2 and

The fundamental period of x(t) is given by mT1 = kT2 provided that the values of m and k are chosen such
that the greatest common divisor (gcd) between m and k is 1.
Example For each of the following signals, determine whether it is periodic or aperiodic. If periodic, find
the period.

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1.3.4 Even and Odd Signals

The continuous-time signal is said to be even when x(t)=x(-t). The continuoustime signal is said to be odd
when x(-t)=-x(t) Odd signals are also known as non symmetrical signals.

Figure (a) Even signal, (b) odd signal

Any signal can be expressed as sum of its even and odd parts as

Some important properties of even and odd signals are:

✓ Multiplication of an even signal by an odd signal produces an odd signal
✓ Multiplication of an even signal by an even signal produces an even signal
✓ Multiplication of an odd signal by an odd signal produces an even signal.

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1.3.5 Causal, Noncausal, and Anticausal Signal

A causal signal is one that has zero values for negative time, i.e., t < 0. A signal is noncausal if it has
nonzero values for both the negative and positive times. An anticausal signal has zero values for positive
time, i.e., t > 0

Figure (a) Causal signal, (b) noncausal signal, (c) anticausal signal

1.3.6 Energy and Power Signal

A signal x(t) with finite energy, which means that amplitude --->0 as time --->ꝏ, is said to be energy signal,
whereas a signal x(t) with finite and nonzero power is said to be power signal. The instantaneous power p(t)
of a signal x(t) can be expressed by
p(t)=x2(t)
The total energy of a continuous-time signal x(t) can be defined as
ꝏ

E = ∫ 𝑥 2 (𝑡)𝑑𝑡
−ꝏ
for a complex valued signal
ꝏ

E = ∫ |𝑥 2 (𝑡)|𝑑𝑡
−ꝏ

Since the power is the time average of energy, the average power is defined as
𝑇
2
1
p = lim ∫ 𝑥 2 (𝑡)𝑑𝑡
𝑇→∞ 𝑇
−𝑇
2

In the case of a discrete-time signal x[n], the integrals in above equations are replaced by corresponding
sums. Thus, the total energy of x[ n] is defined by

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and its average power is defined by

A signal is referred to an energy signal if and only if the total energy is finite .i.e.,
0<E<∞
A signal is referred to an power signal if and only if the average power is finite .i.e.,
0<P<∞
Note: Energy signal has zero time average power and power signal has infinite energy
Example check if x(t) defined below is an energy signal or not

Solution
The energy of the signal is given by

==> it is an Energy signal

Example check the signal given below is power signal or not

Solution . The signal is periodic with period 2. Hence, averaging x2(t) over infinitely large time interval is
the same as averaging over one period, i.e., 2. Thus, the average power P is

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Example Compute energy and power for the following signals, and determine whether each signal is energy
signal, power signal, or neither

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1.3.7 Deterministic and Random Signals

Deterministic signal can be described mathematically as a function of time. A random signal takes
random statistically characterized random values as shown below at any given time. Noise is a common
example of random signal.

Figure (a) Deterministic signal, (b) random signal

1.4 Basic or Elementary signals

Several elementary signals feature prominently in the study of signals and systems. These are
exponential and sinusoidal signals, the step function, the impulse function, and the ramp
function, all of which serve as building blocks for the construction of more complex signals
1.4.1 Exponential Signals
A real exponential signal, in its most general form, is written as
x(t) = Beat,
where both B and a are real parameters. The parameter B is the amplitude of the exponential signal
measured at time t = 0. Depending on whether the other parameter a is positive or negative, we may
identify two special cases:

Fig: Growing exponential, for a > 0 Decaying exponential, f o r a < 0

In discrete time, it is common practice to write a real exponential signal as x[n] = Brn

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Fig: Growing exponential for r > 1 Decaying exponential for 0 < r <1

1.4.2 Impulse functions

The discrete-time version of the unit impulse is defined by

Fig: Discrete time form of unit impulse

The continuous-time version of the unit impulse is defined by the following pair of relations:

Fig: Continuous time form of unit impulse

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Above equation says that the impulse δ (t) is zero everywhere except at the origin. Equation says that
the total area under the unit impulse is unity. The impulse δ (t) is also referred to as the Dirac delta
function.
The delta function can be evolved as the limit of the rectangular pulse

Figure (a) Rectangular pulse, (b) unit impulse

Some Special Properties of the Impulse Function
Sampling property If an arbitrary signal x(t) is multiplied by a shifted impulse function, the product is
given by

implying that multiplication of a continuous-time signal and an impulse function produces an impulse
function, which has an area equal to the value of the continuous-time function at the location of the impulse.
Also, it follows that for t0 = 0,

Shifting property

Scaling property

The unit impulse function can be obtained by taking the derivative of the unit step function as follows:

The unit step function is obtained by integrating the unit impulse function as follows:

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1.4.3 The Unit Step Function

The continuous-time version of the unit- step function is defined by: u(t)

Figure Unit step function

The discrete-time version of the unit- step function is defined by:

1.4.4 Ramp function:

The integral of the step function u(t) is a ramp function of unit slope.
The ramp function is defined as

which can also be written as

Figure ramp function

The discrete-time version of the unit- ramp function is defined by

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figure: Discrete of ramp function

1.4.5 The Rectangular Pulse Function

The continuous-time rectangular pulse function is defined as

figure The rectangular pulse function

1.4.6 The Signum Function

The signum function also called sign function is defined as

Figure The signum function

1.4.7 The Complex Exponential Function

A real exponential function is defined as

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1.4.8 The Sinc Function

The continuous-time sinc function is defined as

Figure The sinc function

1.5 Basic Operations of Signals

Operation performed on independent variable:
1.5.1 Time Shift
For any t and n , time shift is an operation defined as
0 0

If t0 >0, the time shift is known as “delay”. If t0 < 0, the time shift is known as “advance”.
Example. In Fig. given below, the left image shows a continuous-time signal x(t). A time- shifted version
x(t- 2) is shown in the right image.

Figure: An example of time shift.

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1.5.2Time Reversal

Time reversal is defined as for continues and discrete cases as below

which can be interpreted as the “flip over the y-axis”.

Example: the time reversal of x(t) is shown below

Figure: An example of time reversal

1.5.3 Time Scaling

Time scaling is the operation where the time variable t is multiplied by a constant a:

If a > 1, the time scale of the resultant signal is “decimated” (speed up)
If 0 < a < 1, the time scale of the resultant signal is “expanded” (slowed down)

Figure : An example of time scaling.

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1.5.4 Decimation and Expansion

Decimation and expansion are standard discrete-time signal processing operations.
Decimation is defined as yD[n] = x[Mn], for some integers M ,Where, M is the decimation factor.
Expansion is defined as

Where, L is the expansion factor.

Figure : Examples of decimation and expansion for M = 2 and L = 2.

1.5.5 Combination of Operations

Generally, linear operation (in time) on a signal x(t) can be expressed as y(t) = x(at-b).
The recommended method is “Shift, then Scale”
Example: The signal x(t) shown in Figure of sketch x(3t- 5).

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Figure: An example of Shift, then Scale

1.6 Operation performed on dependent variable:

1.6.1 Amplitude scaling
Let x(t) denote a continuous time signal By amplitude scaling, we get y(t) = cx(t)
Where, c is the scaling factor.
Example: An electronic amplifier, a device that performs amplitude scaling.
For discrete time signal y[n] = cx[n]

1.6.2 Amplitude addition:

Let x1(t) and x2(t) is a pair of continuous time signal By adding these two signals, we get y(t) = x1(t) +
x2(t) . eg: An audio mixer
For discrete time signal, y[n] = x1[n] + x2[n]
1.6.3 Amplitude multiplication:
Let x1(t) and x2(t) is a pair of continuous time signal
By multiplying these two signals, we get y(t) = x1(t) x2(t)
eg: An AM radio signal, in which
x1(t) is an audio signal and x2(t) is an sinusoidal carrier wave
For discrete time signal, y[n] = x1[n] x2[n]
1.6.4 Differentiation

y(t)=k𝑑𝑥(𝑡)
𝑑𝑡
𝑑𝑖(𝑡)
Example: Voltage across an inductor L is v(t)=L
𝑑𝑡
1.6.5 Integration:

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y(t) = ∫ 𝑥(𝜏)𝑑𝜏
−∞
Example: Voltage across a capacitor C is
𝑡
1
y(t) = ∫ 𝑖(𝜏)𝑑𝜏
𝐶
−∞

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Chapter 2 Fundamentals of Systems

✓ Systems are used to process signals to allow modification or extraction of additional information
from the signal.
✓ A system may consist of physical components (hardware realization) or an algorithm (operator) that
computes the output signal from the input signal.
✓ A physical system consists of inter-connected components which are characterized
by their input-output relationships

Figure:Continuous-time and discrete-time systems: Here H & T are operators.

2.1 Properties of systems:

2.1.1 Static (Memoryless) and Dynamic (with memory):

Static: A system is static if the output at time t (or n) depends only on the input at time t (or n).
Examples: 1. y(t) = (2x(t)- x2(t))2 is memoryless, because y(t) depends on x(t) only. There is no x(t -1)
or x(t + 1) terms
2. y[n] = x2[n] is memoryless. In fact, this system is passing the input to output directly,
without any processing.
3. Current flowing through a resistor i.e., i(t) = 1R v(t)

Dynamic: A system is said to possess memory if its output signal depends on past or future values of
input.
Example 1. Inductor and capacitor, since the current flowing through the inductor at time „t‟ depends
on
the all past values of the voltage v(t) i.e.,
t
1
i(t) = L ∫ v(τ)dτ
−∞

t
1
v(t) = C
∫ i(τ)dτ
−∞
1
2. The moving average system given by y(n) = (x(n) + x(n − 1) + x(n − 2))
3

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2.1.2 Stable & unstable system

A system is said to be bounded-input, bounded-output (BIBO) stable if and only if every bounded input
results in a bounded output, otherwise it is said to be unstable
If for |x(t)| ≤ Mx < ∞ for all t, output is |y(t)| ≤ My < ∞ for all t; where Mx & My are some finite positive
number
Example: 1. y(t) = x (t-3) is a stable system.
2 y(t) = t x(t) is an unstable system.
3. y[n] = ex[n] is a stable system.
Assume that |x(n)| ≤ Mx < ∞, for all „t‟ y[n] = ex[n] = eMx = finite → Stable
4. y[n] = rn x[n], where r > 1
Assume that |x(n)| ≤ Mx < ∞, for all „t‟, then
|y[n]| = |rn x[n]| = |rn| | x[n]|
as „n‟ →∞ |rn| →∞ ,so y[n]→∞ hence unstable.

2.1.3 Causal and non-Causal system:

Causal: A system is said to be causal if the present value of output signal depends only on the present
or past values of the input signal. A causal system is also known as physical or non-anticipative
system.
Example: 1. The moving average system given by y[n] = (x[n] + x[n − 1] + x[n − 2])/3
2. y(t) = x(t)cos(6t)
Note: i) Any practical system that operates in real time must necessarily be causal.
ii) All static systems are causal.
Non-Causal: A system is said to be non-causal if the present value of output signal depends on one or
more future values of the input signal.
Example:1. The moving average system given by, y[n] = (x[n] + x[n − 1] + x[n + 2])/3

2.1.4 Time variant and time invariant System

Time invariant: A system is time-invariant if a time-shift of the input signal results in the same time-shift
of the output signal.
That is, if x(t) → y(t), then the system is time-invariant if x(t- tO) → y(t- tO), for any tO .

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Figure : Illustration of a time-invariant system.

Example . The system y(t) = sin[x(t)] is time-invariant

Proof. Let us consider a time-shifted signal x1(t) = x(t- tO). Correspondingly, we let y1(t) be
the output of x1(t). Therefore,y1(t) = sin[x1(t)] = sin[x(t- tO)].
Now, we have to check whether y1(t) = y(t- tO). To show this, we note that y(t- tO) = sin[x(t - tO)],
which is the same as y1(t). Therefore, the system is time-invariant.
Time variant: A system is time-variant if its input –output characteristic changes with time.
Example The system y[n] = nx[n] is time-variant.
Proof: Output for a time shifted input is

y[n] | x(n-k) = nx(n-k)

then the same time shifted output is y(n-k) = (n-k)x(n-k)
the above two equations are not same. Hence it is time variant.

2.1.5 Linear and non-linear system:

Linear system: A system is said to be linear if it satisfies two properties i.e.; superposition &
homogeneity.
Superposition: It states that the response of the system to a weighted sum of signals be equal to the
corresponding weighted sum of responses (Outputs of the system) to each of the individual input signal.
For an input x(t) = x1(t), the output y(t) = y1(t)
and input x(t) = x2(t), the output y(t) = y2(t)
then, the system is linear if and only if T [a1x1(t) + a2x2(t)] = a1T [x1(t)] + a2T [x2(t)]
Homogeneity: If the input x(t) is scaled by a constant factor „a‟, then the output y(t) is also scaled by
exactly the same constant factor „a‟.
For an input x(t) →output y(t) and input x1(t) = ax(t)→output y1(t) = ay(t)
Example: The system y(t) = 2πx(t) is linear. To see this, let‟s consider a signal
x(t) = ax1(t) + bx2(t),
where y1(t) = 2πx1(t) and y2(t) = 2πx2(t). Then

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ay1(t) + by2(t) = a(2πx1(t)) + b(2πx2(t))

= 2π [ax1(t) + bx2(t)] = 2πx(t) = y(t).
Example : The system y[n] = (x[2n])2 is not linear. To see this, let‟s consider the signal
x[n] = ax1[n] + bx2[n],
where y1[n] = (x1[2n])2 and y2[n] = (x2 [2n])2. We want to see whether y[n] = ay1[n] + by2[n]. It holds that
ay1[n] + by2[n] = a(x1[2n])2 + b(x2[2n])2 .
However,
y[n] = (x[2n])2 = (ax1[2n] + bx2[2n])2 = a2(x1[2n])2 + b2(x2[2n])2 + 2abx1[n]x2[n].

2.1.6 Invertible and non-invertible system

A system is said to be invertible if the input of the system can be recovered from the output. Let the set of
operations needed to recover the input represents the second system which is connected in cascade with
the given system such that the output signal of the second system is equal to the input applied to the given
system.
Let H be the continuous time system
x(t) input signal to the system
y(t) output signal of the system
Hinv the second continuous time system

The output signal of the second system is given by

Hinv{y(t)} = Hinv{Hx(t)}= HinvH{x(t)}
For the output signal to equal to the original input, we require that

Where „I‟ denotes the identity operator.

The system whose output is equal to the input is an identity system. The operator Hinv must satisfy the
above condition for H to be an invertible system.
Cascading a system, with its inverse system, result in an identity system.

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Example:An inductor is described by the relation

τ
1
y(t) = L
∫ x(τ)dτ
−∞

is an invertible system
because, by rearranging terms, we get
𝑑
𝑥(𝑡) = 𝐿𝑑𝑡 𝑦(𝑡) which is the inversion formula.

Note:i) A system is not invertible unless distinct inputs applied to the system produce distinct outputs.
ii) There must be a one to one mapping between input and output signal for system to be invertible.
Non-invertible System: When several different inputs results in the same output, it is impossible to
obtain the input from output. Such system is called a non-invertible system.
Example: A square-law system described by the input output relation
y(t) = x2(t), is non-invertible,
because distinct inputs x(t) & -x(t) produce the same output y(t) [not distinct output]

2.1.7 Linear –time invariant system (LTI)

Linear time invariant (LTI) systems are good models for many real-life systems, and they have properties
that lead to a very powerful and effective theory for analyzing their behavior. The LTI systems can be
studied through its characteristic function, called the impulse response. Further, any arbitrary input signal
can be decomposed and represented as a weighted sum of unit sample sequences. As a consequence of the
linearity and time invariance properties of the system, the response of the system to any arbitrary input
signal can be expressed in terms of the unit sample response of the system. The general form of the
expression that relates the unit sample response of the system and the arbitrary input signal to the output
signal, called the convolution sum, is also derived.

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2.2 Convolution
2.2.1 Resolution of a Discrete-time signal into impulses:
Any arbitrary sequence x(n) can be represented in terms of delayed and scaled impulse sequence δ(n).
Let x(n) is an infinite sequence as shown in figure below.

Figure:Representing of a signal x[n] using a train of impulses δ[n- k].

The sample x(0) can be obtained by multiplying x(0), the magnitude, with unit impulse δ(n)

Similarly, the sample x(-3) can be obtained as shown in the figure

In the same way we can get the sequence x[n] by summing all the shifted and scaled impulse function

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2.3 Impulse response and convolution sum:

2.3.1 Impulse response:
A discrete-time system performs an operation on an input signal based on predefined criteria to produce a
modified output signal. The input signal x[n] is the system excitation, and y[n] is the system response.
The transform operation is shown in the figure below.

If the input to the system is the unit impulse i.e., x[n] = δ[n], then the output of the system is known as
impulse response represented by h[n] where h[n] = T [δ[n]]
If the input to the system is unit impulse input δ(t), the system output is called the impulse response and
denoted by h(t):

Response of LTI system to arbitrary inputs: The convolution sum

From the above discussion, we get the response of an LTI system to an unit impulse as
the impulse response h[n] i.e.,

As we know the arbitrary input signal is a weighted sum of impulse, the LHS = x[n]
having a response in RHS = y[n] known as convolution summation.

In other words, given a signal x[n] and the impulse response of an LTI system h[n], the convolution
between x[n] and h[n] is defined as

We denote convolution as y[n] = x[n] *h[n].

Equivalent form: Letting m = n - k, we can show that

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Properties of convolution:
The following “standard” properties can be proved easily:
1. Commutative: x[n] * h[n] = h[n] *x[n]
2. Associative: x[n] *(h1[n] *h2[n]) = (x[n] *h1[n]) *h2 [n]
3. Distributive: x[n] *(h1[n] + h2 [n]) = (x(n)* h1[n]) + (x[n] *h2[n])

How to Evaluate Convolution?

To evaluate convolution, there are four basic steps:
1. Fold 3. Multiply
2. Shift 4. Summation
Example:Consider the signal x[n] and the impulse response h[n] shown below.

Let’s compute the output y[n] one by one. First, consider y[0]:

Note that h[-k] is the flipped version of h[k], and ∑∞

𝑘=−∞ 𝑥[𝑘]ℎ[−𝑘] = 1 is the multiply add between

x[k] and h[-k].

To calculate y[l], we flip h[k] to get h[-k], shift h[-k] go get h[l-k], and multiply-add to get
∞

∑ 𝑥[𝑘]ℎ[1 − 𝑘]
𝑘=−∞

Therefore

The calculation is shown in the figure below

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The Convolution Integral

The output of a system for an input expressed as weighted superposition as

Thus, the output y(t) of a linear time-invariant system to an arbitrary input x(t) is obtained in terms of the
unit impulse input δ(t). is referred to as the convolutional integral and is denoted by the symbol * as
∞

y(t) = x(t) ∗ h(t) = ∫ x(τ)h(t − τ)dτ

−∞

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Some Properties of the Convolution Integral

The Commutative Property

The Distributive Property

The Associative Property

Convolution with an Impulse

Convolution with Delayed Input and Delayed Impulse Response

Graphical Convolution
An understanding of graphical interpretation of convolution is very useful in computing the convolution
of more complex signals. The stepwise procedure for graphical convolution is as follows:
Step 1: Make x(τ) fixed.
Step 2: Invert h(τ) about the vertical axis (t = 0) to obtain h(τ).
Step 3: Shift the h(τ) along the τ axis by t0 seconds so that the shifted h(τ) is representing h(t0-τ).
Step 4: The area under the product of x(τ) and h(t0-τ) is y(t0), the value of convolution at t =t0.
Step 5: Repeat steps 3 and 4 for different values of positive and negative to obtain y(t) for all values of t.

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Example Graphically determine the continuous-time convolution of h(t) and x(t) for the following:

Solution

To compute y(t) =x(t) * h(t), first h(τ) is to be obtained by inverting h(τ) about the vertical axis. Then, the
product of x(τ) and h(t τ) is formed, point by point, and this product is integrated to compute y(t). Thus,
the overlap area between the rectangles forming x(τ) and h(t -τ) is y(t). Clearly, y(0) = 0 because there is
no overlap between the rectangles forming x(τ) and h(t -τ) at t ¼=0. For 0 < t < 8, there is overlap
between the rectangles forming x (τ) and h(t -τ). For ≥8 there is no overlap, and hence, y(8) =0. These are
illustrated below with the final result for y(t). The shaded portion represents the overlap area of the
product x(τ) and h(t -τ)

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Exercise Determine the continuous-time convolution of x(t) and h(t) for the following

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Chapter 3 Fourier Series

The objective of this chapter is to identify a family of signals {xk(t)} such that:
1. Every signal in the family passes through any LTI system with only a scale change (or other simply
described change)
xk(t) -→ λkxk(t)
where λk is a scale factor.
2. “Any” signal can be represented as a “linear combination” of signals in their family.

This would allow us to determine the output generated by x(t):

where the scalar ak comes from the definition of linear combination

3.1 Eigenfunctions of an LTI System

To answer the first question, we need the notion of eigenfunction of an LTI system.
Definition For an LTI system, if the output is a scaled version of its input, then the input function is called
an eigenfunction of the system. The scaling factor is called the eigenvalue of the system

3.1.1 Continuous-time Case

Consider an LTI system with impulse response h(t) and input signal x(t):

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The function H(s) is known as the transfer function of the continuous-time LTI system. Note that H(s) is
defined by the impulse response h(t), and is a function in s (independent of t). Therefore, H(s)x(t) can be
regarded as a scalar H(s) multiplied to the function x(t).
From the derivation above, we see that if the input is x(t) = e st, then the output is a scaled version
y(t) = H(s)est.

Therefore, using the definition of eigenfunction, we show that

1. est is an eigenfunction of any continuous-time LTI system, and
2. H(s) is the corresponding eigenvalue
If we specialize to the subclass of periodic complex exponentials of the ejωt , ω ∈ R by setting s = jω, then

H(jω) is called the frequency response of the system.

3.1.2 Discrete-time Case

Next, we consider the discrete-time case:

Suppose that the impulse response is given by h[n] and the input is x[n] = z n , then the output y[n] is

where we defined

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and H(z) is known as the transfer function of the discrete-time LTI system. Similar to the continuous-time
case, this result indicates that
1. zn is an eigenfunction of a discrete-time LTI system, and
2. H(z) is the corresponding eigenvalue

Considering the subclass of periodic complex exponentials e−j(2π/N)n by setting z = e j2π/N , we have

where Ω = 2π/N , and H(ejΩ) is called the frequency response of the system.
3.1.3 Summary
In summary, we have the following observations:

That is, est is an eigenfunction of a CT system, whereas zn is an eigenfunction of a DT system. The

corresponding eigenvalues are H(s) and H(z). If we substitute s = jω and z = ejΩ respectively, then the
eigenfunctions become ejωt and ejΩn ; the eigenvalues become H(jω) and H(ejΩ).
3.1.4 Why is eigenfunction important?
The answer to this question is related to the second objective in the beginning. Let us consider a signal
x(t):

Therefore, the output is

The result implies that if the input is a linear combination of complex exponentials, the output of an LTI
system is also a linear combination of complex exponentials. More generally, if x(t) is an infinite sum of
complex exponentials.

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then the output is again a sum of complex exponentials:

Similarly for discrete-time signals, if

Then

This is an important observation, because as long as we can express a signal x(t) as a linear combination
of eigenfunctions, then the output y(t) can be easily determined by looking at the transfer function (which
is fixed for an LTI system!). Now, the question is : How do we express a signal x(t) as a linear
combination of complex exponentials?

3.2 Fourier Series Representation

Existence of Fourier Series In general, not every signal x(t) can be decomposed as a linear combination
of complex exponentials. However, such decomposition is still possible for an extremely large class of
signals. We want to study one class of signals that allows the decomposition. They are the periodic signals
x(t + T) = x(t)
which satisfy the square integrable condition or Dirichlet conditions

1. Over any period x(t) must be absolutely integrable,that is ,

2. In any finite interval of time x(t) is of bounded variation; that is, there are no more than a finite
number of maxima and minima during any single period of the signal.
3. In any finite interval of time, there are only a finite number of discontinuities. For this class of
signals, we are able to express it as a linear combination of complex exponentials:

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Here, ω0 is the fundamental frequency

ω0 = 2π/T ,
and the coefficients ak are known as the Fourier Series coefficients.
Given a periodic signal x(t) that is square integrable,
how do we determine the Fourier Series coefficients ak? This is answered by the following theorem.

3.2.1 Continuous-time Fourier Series Coefficient

Theorem The continuous-time Fourier series coefficients ak of the signal

is given by

Proof. Let us consider the signal

If we multiply on both sides e-jnω0t , then we have

Integrating both sides from 0 to T yields

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we have

which is equivalent to

Example. Sinusoids
Consider the signal x(t) = 1+1/2 cos 2πt+sin3πt. The period of x(t) is T = 2 [Why?] so the fundamental
frequency is ω0 = 2π/T = π. Recall Euler’s formula ejθ = cosθ+jsinθ, we have

Therefore, the Fourier series coefficients are (just “read off” from this equation!):

Example . Periodic Rectangular Wave

Let us determine the Fourier series coefficients of the following signal

The Fourier series coefficients are (k ≠0):

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Example . Periodic Impulse Train

for any k
3.2.2 Discrete-time Fourier Series coefficients
To construct the discrete-time Fourier series representation, we consider periodic discrete-time signal with
period N
x[n] = x[n + N]

Theorem . The discrete-time Fourier series coefficients ak of the signal

is given by

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Example. Let us consider the following signal shown below. We want to determine the discretetime F.S.
coefficient

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3.2.3 How do we use Fourier series representation?

Fourier series representation says that any periodic square integrable signals (or signals that satisfy
Dirichlet conditions) can be expressed as a linear combination of complex exponentials. Since complex
exponentials are eigenfunctions to LTI systems, the output signal must be a linear combination of
complex exponentials. That is, for any signal x(t) we represent it as

Then, the output signal is given by

Letting bk = H(jkω0)ak, we have

3.2.4 How many Fourier series coefficients are sufficient?

If we define

then xN (t) is an approximation of x(t). As N → ∞, we see that xN (t) → x(t). As an illustration of xN (t) as
N increases, we can see the following figure. Therefore, the number of Fourier series coefficients depends
on the accuracy that we want to achieve. Typically, the number N is chosen such that the residue of the
approximation

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3.3 Trigonometric Fourier Series Representation

If x(t) is an even periodic signal, then

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Example Find the trigonometric Fourier series representation of the periodic signal shown in Figure with
A =3 and period T0=2π.

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3.4 Properties of Fourier Series Coefficients

There are a number of Fourier series properties that we encourage you to read the text. The following is a
quick summary of these properties.

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Chapter 4 Continuous Time Fourier Transform

Let us begin our discussion by reviewing some limitations of Fourier series representation. In Fourier
series analysis, two conditions on the signals are required:
1. The signal must be periodic, i.e., there exist a T > 0 such that x(t + T) = x(t).
2. The signal must be square integrable

, or satisfies the Dirichlet conditions.

In this chapter, we want to extend the idea of Fourier Series representation to aperiodic signals. That is,
we want to relax the first condition to aperiodic signals.

4.1 Insight from Fourier Series

Let’s first consider the following periodic signal

where x(t) = x(t + T). The Fourier Series coefficients of x(t) are (check yourself!)

If we substitute ω = kω0, then

Multiplying T on both sides yields

which is the normalized Fourier Series coefficient. Pictorially below, indicates that the normalized Fourier
series coefficients Tak are bounded by the envelop X(ω) = 2 sin(ωT1) /ω , as illustrated in Fig below

Figure : Fourier Series coefficients of x(t) for some T.

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When T increases, the spacing between consecutive ak reduces. However, the shape of the envelop
function X(ω) = 2sin(ωT1) /ω remains the same. This can be seen in below

Figure : Fourier Series coefficients of x(t) for some T ′ , where T ′ > T

In the limiting case where T → ∞, then the Fourier series coefficients Tak approaches the envelop
function X(ω). This suggests us that if we have an aperiodic signal, we can treat it as a periodic signal
with T → ∞. Then the corresponding Fourier series coefficients approach to the envelop function X(ω).
The envelop function is called the Fourier Transform of the signal x(t). Now, let us study Fourier
Transform more formally

4.2 Fourier Transform

The derivation of Fourier Transform consists of three steps.
Step 1. We assume that an aperiodic signal x(t) has finite duration, i.e., x(t) = 0 for |t| > T/2, for some T.
Since x(t) is aperiodic, we first construct a periodic signal :

Step 2

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Step 3.

as

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The two equations above are known as the Fourier Transform pair.as Analysis Equation (because we are
analyzing the time signal in the Fourier domain) and the latter is called the Synthesis Equation (because
we are gathering the Fourier domain information and reconstruct the time signal).
To summarize we have

Theorem . The Fourier Transform X(jω) of a signal x(t) is given by

and the inverse Fourier Transform is given by

4.3 Relation to Fourier Series

At this point you may wonder: What is the difference between Fourier Series and Fourier Transform? To
answer this question, let us apply Fourier Transform to the following two types of signals.
1. Aperiodic Signal: As we discussed in the derivation of Fourier Transform, the Fourier Transform of
an aperiodic signal is the limiting case (when ω0 → 0) of applying Fourier Series analysis on the
periodically padded version of the aperiodic signal. Fourier Transform can be applied to both periodic
and aperiodic signals, whereas Fourier Series analysis can only be applied to periodic signals.
2. Periodic Signal: If the signal x(t) is periodic, then we do not need to construct x(t)
and set ω0 → 0. In fact, ω0 is fixed by the period of the signal: If the period

Figure : Fourier Transform on aperiodic signals is equivalent to applying Fourier series analysis on the
periodically padded version of the signal, and set the limit of ω0 → 0

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of x(t) is T0, then ω0 = 2π/T0 . Now, since x(t) is periodic, we can apply Fourier Series analysis to x(t) and
get

where ak is the Fourier Series coefficient. If we further apply Fourier Transform then we have

Here, the last equality is established by the fact that inverse Fourier Transfom

Therefore, we showed that the Fourier Transform of a periodic signal is a train of impulses with amplitude
defined by the Fourier Series coefficients (and scaled by a factor of 2π).

Figure Fourier Transform and Fourier Series analysis of a periodic signal:

Both yields a train of impulses. For Fourier Transform, the amplitude is multiplied by a factor of 2π. For
Fourier Series coefficients, the separation between each coefficient is ω0

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4.4 Examples
Example 1

The magnitude and phase can be calculated as

Example 2

Example 3.

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Example 4. Consider the aperiodic signal

The Fourier Transform is

Example 5. Let us determine the CTFT of the unit step function u(t). To do so, we apply CTFT and get

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4.5 Properties of Fourier Transform

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Example

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4.6 System Analysis using Fourier Transform

First Order System
Let us consider the first order system

General Systems
In general, we want to study the system

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Example 1

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Example 2

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Chapter 5 Discrete-Time Fourier Transform

5.1 Review on Continuous-time Fourier Transform
Before we derive the discrete-time Fourier Transform, let us recall the way we constructed continuous-
time Fourier Transform from the continuous-time Fourier Series. In deriving the continuous-time Fourier
Transform, we basically have the following three steps:

By defining

which is known as the continuous-time Fourier Transform, we showed

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5.2 Deriving Discrete-time Fourier Transform

Now, let’s apply the same concept to discrete-time signals. In deriving the discrete time Fourier
Transform, we also have three key steps.

If we define

then

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• Step 3: Putting the above equations to back substitution

Therefore

Figure : As N → ∞, ω0 → 0. So the area becomes infinitesimal small and sum becomes integration.

5.3 Why is X(ejω) periodic?

It is interesting to note that the continuous-time Fourier Transform X(jω) is aperiodic in general, but the
discrete-time Fourier Transform X(ejω) is always periodic. To see this, let us consider the discrete-time
Fourier Transform (we want to check whether X(ejω) = X(e j(ω+2π) )!):

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5.4 Properties of Discrete-time Fourier Transform

Discrete-time Fourier Transform:

Discrete-time Inverse Fourier Transform:

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5.5 Examples

To sketch the magnitude |X(ejω)|, we note that |X(e jω)| = |1 + 2 cos ω|.

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Figure : Magnitude plot of |X(e jω)| in Example 1.

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Exercises Find the DTFT for the following sequences:

(a) x(n) = u(n) – u(n – 5)
(b) x(n) =αn(u(n) – u(n – 8)), |α| < 1
(c) x(n) = n(0.5)n u(n)_x0007_(d) x(n)=|a|nsin(ωn) , |a|<1

Exercises Determine the inverse DTFT of each of the following DTFTs:

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Chapter 6 Sampling Theorem

Sampling theorem plays a crucial role in modern digital signal processing. The theorem concerns about
the minimum sampling rate required to convert a continuous time signal to a digital signal, without loss of
information.

6.1 Analog to Digital Conversion

Consider the following system shown below This system is called an analog to-digital (A/D) conversion
system. The basic idea of A/D conversion is to take a continuous-time signal, and convert it to a discrete-
time signal.

Figure An analog to digital (A/D) conversion system.

Mathematically, if the continuous-time signal is x(t), we can collect a set of samples by multiplying x(t)
with an impulse train p(t):

where T is the period of the impulse train. Multiplying x(t) with p(t) yields

Pictorially, xp(t) is a set of impulses bounded by the envelop x(t) as shown in Figure below

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Figure : An example of A/D conversion.

The output signal xp(t) represents a set of samples of the signal x(t).
We may regard xp(t) as the samples of x(t). Note that xp(t) is still a continuous-time signal!
(We can view xp(t) as a discrete-time signal if we define xp[n] = x(nT). But this is not an important issue
here.)

6.2 Frequency Analysis of A/D Conversion

Having an explanation of the A/D conversion in time domain, we now want to study the A/D conversion
in the frequency domain. (Why? We need it for the development of Sampling Theorem!) So, how do the
frequency responses X(jω), P(jω) and Xp(jω) look like?

6.2.1 How does P(jω) look like?

Let’s start with P(jω). , we know that

This means that the frequency response of the impulse train p(t) is another impulse train. The only
difference is that the period of p(t) is T, whereas the period of P(jω) is 2π/T .

Figure Illustration of X(jω) and P(jω).

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6.2.2 How does Xp(jω) look like?

Next, suppose that the signal x(t) has a frequency response X(jω). We want to know the frequency
response of the output xp(t). From the definition of xp(t), we know know that
xp(t) = x(t)p(t).
Therefore, by the multiplication property of Fourier Transform, we have

Shown in above are the frequency response of X(jω) and P(jω) respectively. To perform the convolution
in frequency domain, we first note that P(jω) is an impulse train. Therefore, convolving X(jω) with P(jω)
is basically producing replicates at every 2π /T . The result is shown in below.

Figure: Convolution between X(jω) and P(jω) yields periodic replicates of X(jω)

Mathematically, the output Xp(jω) is given by

The result is illustrated in Figure

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6.2.3 What happens if T becomes larger and larger?

If T becomes larger and larger (i.e., we take fewer and fewer samples), we know from the definition of
p(t) that the period (in time domain) between two consecutive

Figure Illustration of xp(t) and Xp(jω).

impulses increases (i.e., farther apart). In frequency domain, since

the period 2π/T reduces! In other words, the impulses are more packed in frequency domain when T
increases. Figure below illustrates this idea.

Figure :When T increases, the period in frequency domain reduces.

If we consider Xp(jω), which is a periodic replicate of X(jω) at the impulses given by P(jω), we see that
the separation between replicates reduces. When T hits certain limit, the separation becomes zero; and
beyond that limit, the replicates start to overlap! When the frequency replicates overlap, we say that there
is aliasing.

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Figure : When T is sufficiently large, there will be overlap between consecutive replicates.

Therefore, in order to avoid aliasing, T cannot be too large. If we define the sampling rate to be

then smaller T implies higher ωs. In other words, there is a minimum sampling rate such that no aliasing
occurs.

Figure :Meanings of high sampling rate v.s. low sampling rate

6.2.4 What is the minimum sampling rate such that there is no aliasing?
Here, let us assume that the signal x(t) is band-limited. That is, we assume X(jω) = 0 for all |ω| > W,
where W is known as the band-width. To answer this question, we need the Sampling Theorem.

Figure: Left: A band limited signal (since X(jω) = 0 for all ω > |W|.) Right: A band non-limited signal.

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6.3 Sampling Theorem

Theorem (Sampling Theorem). Let x(t) be a band limited signal with X(jω) = 0 for all |ω| > W. Then the
minimum sampling rate such that no aliasing occurs in Xp(jω) is ωs > 2W
where ωs = 2π/T .

6.3.1 Explanation
Suppose x(t) has bandwidth W. The tightest arrangement that no aliasing occurs is shown below.

Figure: Minimum sampling rate that there is no aliasing

In this case, we see that the sampling rate ωs (= 2π/T ) is ωs = 2W

If T is larger (or ωs is smaller), then 2π/T becomes less than 2W, and aliasing occurs. Therefore, the
minimum sampling rate to ensure no aliasing is ωs > 2W.

Example Suppose there is a signal with maximum frequency 40kHz. What is the minimum sampling rate

Figure : Example: Minimum sampling frequency

Answer : Since ω = 2πf, we know that the max frequency (in rad) is ω = 2π(40 × 103 ) = 80 × 103π (rad).
Therefore, the minimum Sampling rate is: 2 × (80 × 103π), which is 160 × 103π (rad) = 80kHz.

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6.4 Digital to Analog Conversion

In the previous sections, we studied A/D conversion. Now, given a discrete-time signal (assume no
aliasing), we would like to construct the continuous time signal.

6.4.1 Given Xp(t) (no aliasing), how do I recover x(t)?

If no aliasing occurs during the sampling processing (i.e., multiply x(t) with p(t)), then we can apply a
lowpass filter H(jω) to extract the x(t) from xp(t).shows a schematic diagram of how this is performed. To
see how an ideal lowpass filter can extract x(t) from xp(t), we first look at the frequency response of
Xp(jω). Suppose that p(t) has a period of T (so that ωs = 2π/T )

Figure : Schematic diagram of recovering x(t) from xp(t).

The filter H(jω) is assumed to be an ideal lowpass filter Then

As shown in the top left of above figure , Xp(jω) is a periodic replicate of X(jω). Since we assume that
there is no aliasing, the replicate covering the y-axis is identical to X(jω). That is, for |ω| < ωs /2 ,
Xp(jω) = X(jω).
Now, if we apply an ideal lowpass filter

then
Xp(jω)H(jω) = X(jω),
for all ω. Taking the inverse continuous-time Fourier transform, we can obtain x(t).

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Figure : Left: Multiplication between Xp(jω) and the lowpass filter H(jω).
The extracted output Xˆ(jω) is identical to X(jω) if no aliasing occurs. By applying inverse Fourier
transform to Xˆ(jω) we can obtain x(t).

6.4.2 If Xp(t) has aliasing, can I still recover x(t) from xp(t) ?
The answer is NO. If aliasing occurs, then the condition Xp(jω) = X(jω does not hold for all |ω| < ωs /2 .
Consequently, even if we apply the lowpass filter H(jω) to Xp(jω), the result is not X(jω). This can be
seen in Figure below

Figure : If aliasing occurs, we are unable to recover x(t) from xp(t) by using an ideal lowpass filter..

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6.4.3 What can I do if my sampling device does not support a very high sampling rate?
• Method 1: Buy a better sampling device !
• Method 2: Send signals with narrower bandwidth or limit the bandwidth before sending :

Method 3 : use multi rate system with proper sample rating at each stage

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