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SIGNALS & SYSTEMS

(20A04301T)

LECTURE NOTES

II -B.TECH & I-SEM

Prepared by:
Dr. S. Muni Rathnam, Professor
Department of Electronics and Communication Engineering

VEMU INSTITUTE OF TECHNOLOGY

(Approved By AICTE, New Delhi and Affiliated to JNTUA, Ananthapuramu)

Accredited By NAAC, NBA( EEE, ECE & CSE) & ISO: 9001-2015 Certified Institution

Near Pakala, P.Kothakota, Chittoor- Tirupathi Highway

Chittoor, Andhra Pradesh-517 112

Web Site: www.vemu.org

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Course Code L T P C
SIGNALS AND SYSTEMS
20A04301T 3 0 0 3
Pre-requisite Mathematics - I
Semester III
Course Objectives:
 To introduce students to the basic idea of signal and system analysis and its characterization in time and frequency domains.
 To present Fourier tools through the analogy between vectors and signals
 To teach concept of sampling and reconstruction of signals.
 To analyze characteristics of linear systems in time and frequency domains.
 To understand Laplace and z-transforms as mathematical tool to analyze continuous and discrete-time signals and systems.
Course Outcomes (CO): After completion of the course, the student can able to
CO-1: Explain the mathematical description & representation of continuous time & Discrete time Signals and Systems. Perform the
mathematical operations on Signals and Systems
CO-2: Represent the periodic signals using Fourier Series and Analyze the spectral characteristics & Sampling theorem for
continuous time signals using Fourier Transform.
CO-3: Analyze the spectral characteristics of continuous time signals using Laplace Transform (LT) and Apply same to continuous
time systems for finding the response & Analyzing Stability.
CO-4: Analyze the filter characteristics and Physical realization of LTI System and also explain about PSD & ESD
CO-5: Apply Z-Transform (ZT) & Discrete Time Fourier Transform (DTFT) to analyze the spectral characteristics of discrete time
signals. Find the response and Analyze the stability of discrete time system by applying Z-Transform (ZT).
Unit – I: Signals and Systems
Signals & Systems: Basic definitions and classification of Signals and Systems (Continuous time and discrete time), operations on
signals, Concepts of Convolution and Correlation of signals, Analogy between vectors and signals-Orthogonality, mean square
error
Unit – II: Fourier Series and Fourier Transform
Fourier series: Trigonometric & Exponential, Properties of Fourier series, concept of discrete spectrum, Illustrative Problems.
Continuous Time Fourier Transform: Definition, Computation and properties of Fourier transform for different types of signals
and systems, Inverse Fourier transform. Statement and proof of sampling theorem of low pass signals, Illustrative Problems.
Unit – II: Fourier Series and Fourier Transform
Fourier series: Trigonometric & Exponential, Properties of Fourier series, concept of discrete spectrum, Illustrative Problems.
Continuous Time Fourier Transform: Definition, Computation and properties of Fourier transform for different types of signals
and systems, Inverse Fourier transform. Statement and proof of sampling theorem of low pass signals, Illustrative Problems.
Unit – IV: Signal Transmission through LTI systems
Signal Transmission through Linear Systems: Linear system, impulse response, Response of a linear system for different input
signals, linear time-invariant (LTI) system, linear time variant (LTV) system, Transfer function of a LTI system. Filter
characteristics of linear systems. Distortion less transmission through a system, Signal bandwidth, System bandwidth, Ideal LPF,
HPF and BPF characteristics, Causality and Paley-Wiener criterion for physical realization, Relationship between bandwidth and
rise time, Energy and Power spectral densities, Illustrative Problems
Unit – V: DTFT & Z-Transform
Discrete Time Fourier Transform: Definition, Computation and properties of Discrete Time Fourier transform for different types
of signals and systems.
Z–Transform: Definition, ROC, Properties, Poles and Zeros in Z-plane, The inverse Z-Transform, System analysis, Transfer
function, BIBO stability, System Response to standard signals, Solution of difference equations with initial conditions. Illustrative
Problems.
Textbooks:
1.A.V. Oppenheim, A.S. Willsky and S.H. Nawab, “Signals and Systems”, 2nd Edition, PHI, 2009.
2. Simon Haykin and Van Veen, “Signals & Systems”, 2nd Edition, Wiley, 2005.
Reference Books:
1. BP Lathi, “Principles of Linear Systems and Signals”, 2nd Edition, Oxford University Press, 015.
2. Matthew Sadiku and Warsame H. Ali, “Signals and Systems A primer with MATLAB”, CRC Press, 2016.
3. Hwei Hsu, “Schaum's Outline of Signals and Systems”, 4thEdition, TMH, 2019

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UNIT-I
SIGNALS & SYSTEMS

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UNIT-1

SIGNALS & SYSTEMS

Signal :A signal describes a time varying physical phenomenon which is intended to convey
information. (or) Signal is a function of time or any other variable of interest. (or) Signal is a
function of one or more independent variables, which contain some information.

Example: voice signal, video signal, signals on telephone wires, EEG, ECG etc.

Signals may be of continuous time or discrete time signals.

System :System is a device or combination of devices, which can operate on signals and
produces corresponding response. Input to a system is called as excitation and output from it is
called as response. (or) System is a combination of sub units which will interact with each other
to achieve a common interest.

For one or more inputs, the system can have one or more outputs.

Example: Communication System

Elementary Signals or Basic Signals:

Unit Step Function

Unit step function is denoted by u(t). It is defined as u(t) = 1 when t ≥ 0 and

0 when t < 0

• It is used as best testsignal.

• Area under unit step function isunity.

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Unit Impulse Function

0; �≠ 0
Impulsefunctionisdenotedbyδ(t).anditisdefinedasδ(t)={∞; �= 0 }
∞
∫−(�)��= 1
∞

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Ramp Signal

Ramp signal is denoted by r(t), and it is defined as r(t) =

Area under unit ramp is unity.

Parabolic Signal

Parabolic signal can be defined as x(t)=

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Signum Function

Signum function is denoted as sgn(t). It is defined as sgn(t) =

sgn(t) = 2u(t) – 1

Exponential Signal

Exponentialsignalisin theformof x(t) =eαt

.The shape of exponential can be defined by α

Casei:if α= 0→x(t) = e0= 1

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Caseii:if α< 0i.e.-vethenx(t) =e−αt

. The shape is called decaying exponential.

Caseiii:if α> 0i.e.+vethen x(t)=eαt

. The shape is called raising exponential.

Rectangular Signal

Let it be denoted as x(t) and it is defined as

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Triangular Signal

Let it be denoted as x(t)

Sinusoidal Signal

Sinusoidalsignalisin theformof x(t)=Acos( w0±ϕ)orAsin(w0±ϕ)

Where T0 = 2π/w0

Classification of Signals:

Signals are classified into the following categories:

• Continuous Time and Discrete TimeSignals

• Deterministic and Non-deterministicSignals
• Even and Odd Signals
• Periodic and AperiodicSignals
• Energy and PowerSignals
• Real and Imaginary Signals

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Continuous Time and Discrete Time Signals

A signal is said to be continuous when it is defined for all instants of time.

A signal is said to be discrete when it is defined at only discrete instants of time/

Deterministic and Non-deterministic Signals

A signal is said to be deterministic if there is no uncertainty with respect to its value at

any instant of time. Or, signals which can be defined exactly by a mathematical formula are
known as deterministicsignals.

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A signal is said to be non-deterministic if there is uncertainty with respect to its value at

some instant of time. Non-deterministic signals are random in nature hence they are called
random signals. Random signals cannot be described by a mathematical equation. They are
modelled in probabilistic terms.

Even and Odd Signals

A signal is said to be even when it satisfies the condition x(t) = x(-t)

Example 1: t2, t4… cost etc.

Let x(t) = t2

x(-t) = (-t)2 = t2 =x(t)

∴ t2is evenfunction

Example 2: As shown in the following diagram, rectangle function x(t) = x(-t) so it is also even
function.

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A signal is said to be odd when it satisfies the condition x(t) = -x(-t)

Example: t, t3 ... And sin t

Let x(t) = sin t

x(-t) = sin(-t) = -sin t = -x(t)

∴ sin t is odd function.

Any function ƒ(t) can be expressed as the sum of its even function ƒe(t) and odd function ƒo(t).

ƒ(t ) = ƒe(t ) + ƒ0(t )

where

ƒe(t ) = ½[ƒ(t ) +ƒ(-t )]

Periodic and Aperiodic Signals

A signal is said to be periodic if it satisfies the condition x(t) = x(t + T) or x(n) = x(n + N).

Where

T = fundamental time period,

1/T = f = fundamentalfrequency.

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The above signal will repeat for every time interval T0 hence it is periodic with period T0.

Energy and Power Signals

A signal is said to be energy signal when it has finite energy.

A signal is said to be power signal when it has finite power.

NOTE:A signal cannot be both, energy and power simultaneously. Also, a signal may be neither
energy nor power signal.

Power of energy signal = 0

Energy of power signal = ∞

Real and Imaginary Signals

A signal is said to be real when it satisfies the condition x(t) = x*(t)

A signal is said to be odd when it satisfies the condition x(t) = -x*(t)
Example:
If x(t)= 3 then x*(t)=3*=3 here x(t) is a real signal.
If x(t)= 3j then x*(t)=3j* = -3j = -x(t) hence x(t) is a odd signal.
Note: For a real signal, imaginary part should be zero. Similarly for an imaginary signal, real
part should be zero.

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Basic operations on Signals:

There are two variable parameters in general:

1. Amplitude
2. Time

(1) The following operation can be performed withamplitude:

Amplitude Scaling

C x(t) is a amplitude scaled version of x(t) whose amplitude is scaled by a factor C.

Addition

Addition of two signals is nothing but addition of their corresponding amplitudes. This
can be best explained by using the following example:

As seen from the previous diagram,

-10 < t < -3 amplitude of z(t) = x1(t) + x2(t) = 0 + 2 = 2

-3 < t < 3 amplitude of z(t) = x1(t) + x2(t) = 1 + 2 = 3

3 < t < 10 amplitude of z(t) = x1(t) + x2(t) = 0 + 2 = 2

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Subtraction

subtraction of two signals is nothing but subtraction of their corresponding amplitudes.

This can be best explained by the following example:

As seen from the diagram above,

-10 < t < -3 amplitude of z (t) = x1(t) - x2(t) = 0 - 2 = -2

-3 < t < 3 amplitude of z (t) = x1(t) - x2(t) = 1 - 2 = -1

3 < t < 10 amplitude of z (t) = x1(t) - x2(t) = 0 - 2 = -2

Multiplication

Multiplication of two signals is nothing but multiplication of their corresponding

amplitudes. This can be best explained by the following example:

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As seen from the diagram above,

-10 < t < -3 amplitude of z (t) = x1(t) ×x2(t) = 0 ×2 = 0

-3 < t < 3 amplitude of z (t) = x1(t) - x2(t) = 1 ×2 = 2
3 < t < 10 amplitude of z (t) = x1(t) - x2(t) = 0 × 2 = 0

(2) The following operations can be performed withtime:

Time Shifting

x(t ±t0) is time shifted version of the signal x(t).

x (t + t0) →negative shift
x (t - t0) →positive shift

Time Scaling

x(At) is time scaled version of the signal x(t). where A is always positive.

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|A| > 1 → Compression of the signal

|A| < 1 → Expansion of the signal

Note: u(at) = u(t) time scaling is not applicable for unit step function.

Time Reversal

x(-t) is the time reversal of the signal x(t).

Classification of Systems:

Systems are classified into the following categories:

• Liner and Non-linerSystems

• Time Variant and Time InvariantSystems
• Liner Time variant and Liner Time invariantsystems
• Static and DynamicSystems
• Causal and Non-causalSystems
• Invertible and Non-InvertibleSystems
• Stable and UnstableSystems

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Linear and Non-linear Systems

A system is said to be linear when it satisfies superposition and homogenate principles.

Consider two systems with inputs as x1(t), x2(t), and outputs as y1(t), y2(t) respectively. Then,
according to the superposition and homogenate principles,

T [a1 x1(t) + a2 x2(t)] = a1 T[x1(t)] + a2 T[x2(t)]

∴ T [a1 x1(t) + a2 x2(t)] = a1y1(t) + a2 y2(t)

From the above expression, is clear that response of overall system is equal to response of
individual system.

Example:

y(t) = x2(t)

Solution:

y1 (t) = T[x1(t)] = x21(t)

y2 (t) = T[x2(t)] = x22(t)

T [a1 x1(t) + a2 x2(t)] = [ a1 x1(t) + a2 x2(t)]2

Which is not equal to a1y1(t) + a2 y2(t). Hence the system is said to be non linear.

Time Variant and Time Invariant Systems

A system is said to be time variant if its input and output characteristics vary with time.
Otherwise, the system is considered as time invariant.

The condition for time invariant system is:

y (n , t) = y(n-t)

The condition for time variant system is:

y (n , t) ≠ y(n-t)

Where y (n , t) = T[x(n-t)] = inputchange

y (n-t) = output change

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Example:

y(n) = x(-n)

y(n, t) = T[x(n-t)] = x(-n-t)

y(n-t) = x(-(n-t)) = x(-n + t)

∴ y(n, t) ≠ y(n-t). Hence, the system is time variant.

Liner Time variant (LTV) and Liner Time Invariant (LTI) Systems

If a system is both liner and time variant, then it is called liner time variant (LTV) system.

If a system is both liner and time Invariant then that system is called liner time invariant (LTI)
system.

Static and Dynamic Systems

Static system is memory-less whereas dynamic system is a memory system.

Example 1: y(t) = 2 x(t)

For present value t=0, the system output is y(0) = 2x(0). Here, the output is only dependent upon
present input. Hence the system is memory less or static.

Example 2: y(t) = 2 x(t) + 3 x(t-3)

For present value t=0, the system output is y(0) = 2x(0) + 3x(-3).

Here x(-3) is past value for the present input for which the system requires memory to get this
output. Hence, the system is a dynamic system.

Causal and Non-Causal Systems

A system is said to be causal if its output depends upon present and past inputs, and does not
depend upon future input.

For non causal system, the output depends upon future inputs also.

Example 1: y(n) = 2 x(t) + 3 x(t-3)

For present value t=1, the system output is y(1) = 2x(1) + 3x(-2).

Here, the system output only depends upon present and past inputs. Hence, the system is causal.

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Example 2: y(n) = 2 x(t) + 3 x(t-3) + 6x(t + 3)

For present value t=1, the system output is y(1) = 2x(1) + 3x(-2) + 6x(4) Here, the system output
depends upon future input. Hence the system is non-causal system.

Invertible and Non-Invertible systems

A system is said to invertible if the input of the system appears at the output.

Y(S) = X(S) H1(S) H2(S)

= X(S) H1(S) · 1(H1(S))

Since H2(S) = 1/( H1(S) )

∴ Y(S) = X(S)

→ y(t) = x(t)
Hence, the system is invertible.

If y(t) ≠ x(t), then the system is said to be non-invertible.

Stable and Unstable Systems

The system is said to be stable only when the output is bounded for bounded input. For a
bounded input, if the output is unbounded in the system then it is said to be unstable.

Note: For a bounded signal, amplitude is finite.

Example 1: y (t) = x2(t)

Let the input is u(t) (unit step bounded input) then the output y(t) = u2(t) = u(t) = bounded
output.

Hence, the system is stable.

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Example 2: y (t) = ∫x(t)dt

Let the input is u (t) (unit step bounded input) then the output y(t) = ∫u(t)dt= ramp signal
(unbounded because amplitude of ramp is not finite it goes to infinite when t → infinite).

Hence, the system is unstable.

Analogy Between Vectors and Signals:

There is a perfect analogy between vectors and signals.

Vector

A vector contains magnitude and direction. The name of the vector is denoted by bold
face type and their magnitude is denoted by light face type.

Example: V is a vector with magnitude V. Consider two vectors V1 and V2 as shown in the
following diagram. Let the component of V1 along with V2is given by C12V2. The component of
a vector V1along with the vector V2 can obtained by taking a perpendicular from the end of V1 to
the vector V2 as shown indiagram:

The vector V1can be expressed in terms of vector V2

V1= C12V2 + Ve

Where Ve is the error vector.

But this is not the only way of expressing vector V1 in terms of V2. The alternate possibilities
are:

V1=C1V2+Ve1

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V2=C2V2+Ve2

The error signal is minimum for large component value. If C12=0, then two signals are said to be
orthogonal.

Dot Product of Two Vectors

V1 . V2 = V1.V2cosθ

θ = Angle between V1 and V2

V1. V2 =V2.V1

From the diagram, components of V1a long V2 = C 12 V2

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Signal

The concept of orthogonality can be applied to signals. Let us consider two signals f1(t) and f2(t).
Similar to vectors, you can approximate f1(t) in terms of f2(t) as

f1(t) = C12 f2(t) + fe(t) for (t1< t < t2)

⇒ fe(t) = f1(t) – C12 f2(t)

One possible way of minimizing the error is integrating over the interval t1 to t2.

However, this step also does not reduce the error to appreciable extent. This can be corrected by
taking the square of error function.

Where ε is the mean square value of error signal. The value of C12 which minimizes the error,
you need to calculate dε/dC12=0

Derivative of the terms which do not have C12 term are zero.

Put C12 = 0 to get condition for orthogonality.

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Orthogonal Vector Space

A complete set of orthogonal vectors is referred to as orthogonal vector space. Consider a three
dimensional vector space as shown below:

Consider a vector A at a point (X1, Y1, Z1). Consider three unit vectors (VX, VY, VZ) in the
direction of X, Y, Z axis respectively. Since these unit vectors are mutually orthogonal, it satisfies
that

We can write above conditions as

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The vector A can be represented in terms of its components and unit vectors as

Any vectors in this three dimensional space can be represented in terms of these three unit
vectors only.

If you consider n dimensional space, then any vector A in that space can be represented as

As the magnitude of unit vectors is unity for any vector A

The component of A along x axis =A.VX

ThecomponentofAalongYaxis=A.VY The
component of A along Z axis =A.VZ

Similarly, for n dimensional space, the component of A along some G axis

=A.VG....................... (3)
Substitute equation 2 in equation 3.

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Orthogonal Signal Space

Let us consider a set of n mutually orthogonal functions x1(t), x2(t)... xn(t) over the
interval t1 to t2. As these functions are orthogonal to each other, any two signals xj(t), xk(t) have
to satisfy the orthogonality condition.i.e.

Let a function f(t), it can be approximated with this orthogonal signal space by adding the
components along mutually orthogonal signals i.e.

The component which minimizes the mean square error can be found by

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All terms that do not contain Ck is zero. i.e. in summation, r=k term remains and all other terms
are zero.

Mean Square Error:

The average of square of error function fe(t) is called as mean square error. It is denoted by ε
(epsilon).

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The above equation is used to evaluate the mean square error.

Closed and Complete Set of Orthogonal Functions:

Let us consider a set of n mutually orthogonal functions x1(t), x2(t)...xn(t) over the interval
t1 to t2. This is called as closed and complete set when there exist no function f(t) satisfying the
condition

If this function is satisfying the equation

For k=1,2,.. then f(t) is said to be orthogonal to each and every function of orthogonal set.
This set is incomplete without f(t). It becomes closed and complete set when f(t) is included.
f(t) can be approximated with this orthogonal set by adding the components along mutually
orthogonal signals i.e.

Orthogonality in Complex Functions:

If f1(t) and f2(t) are two complex functions, then f1(t) can be expressed in terms of f2(t) as

f1(t)=C12f2(t).. withnegligible error

Where f2*(t) is the complex conjugate of f 2(t)

If f1(t) and f2(t) are orthogonal then C12 = 0

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The above equation represents orthogonality condition in complex functions.

Fourier series:

To represent any periodic signal x(t), Fourier developed an expression called Fourier
series. This is in terms of an infinite sum of sines and cosines or exponentials. Fourier series uses
orthoganality condition.

Jean Baptiste Joseph Fourier,aFrench mathematician and a physicist; was born in

Auxerre, France. He initialized Fourier series, Fourier transforms and their applications to
problems of heat transfer and vibrations. The Fourier series, Fourier transforms and Fourier's
Law are named in hishonour.

Fourier Series Representation of Continuous Time Periodic Signals

A signal is said to be periodic if it satisfies the condition x (t) = x (t + T) or x (n) = x (n + N).

Where T = fundamental time period,

ω0= fundamental frequency = 2π/T

There are two basic periodic signals:

x(t)=cosω0t(sinusoidal)
&x(t)=ejω0t(complex exponential)
These two signals are periodic with period T=2π/ω0

. A set of harmonically related complex exponentials can be represented as {ϕk(t)}

All these signals are periodic with period T

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According to orthogonal signal space approximation of a function x (t) with n, mutually

orthogonal functions is given by

Where ak= Fourier coefficient = coefficient of approximation.

This signal x(t) is also periodic with period T.
Equation 2 represents Fourier series representation of periodic signal x(t).
The term k = 0 is constant.
The term k=±1 having fundamental frequency ω0 , is called as 1st harmonics.
The term k=±2 having fundamental frequency 2ω0 , is called as 2nd harmonics, and so
on... The term k=±n having fundamental frequency nω0, is called as nth harmonics.

Deriving Fourier Coefficient

We know that

Multiplye−jnω0tonbothsides.Then

Consider integral on both sides.

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by Euler's formula,

Hence in equation 2, the integral is zero for all values of k except at k = n. Put k = n in
equation 2.

Replace n by k

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Properties of Fourier series:

Linearity Property

Time Shifting Property

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Frequency Shifting Property

Time Reversal Property

Time Scaling Property

Differentiation and Integration Properties

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Multiplication and Convolution Properties

Conjugate and Conjugate Symmetry Properties

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Trigonometric Fourier Series (TFS)

sinnω0tand sinmω0t are orthogonal over the interval (t0,t0+2πω0). So sinω0t,sin2ω0tforms

an orthogonal set. This set is not complete without {cosnω0t } because this cosine set is also
orthogonal to sine set. So to complete this set we must include both cosine and sine terms. Now
the complete orthogonal set contains all cosine and sine terms i.e. {sinnω0t,cosnω0t} where n=0,
1,2...

The above equation represents trigonometric Fourier series representation of x(t).

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Exponential Fourier Series (EFS):

Consider a set of complex exponential functions

which is orthogonal over the interval (t0,t0+T). Where T=2π/ω0 . This is a complete set so it is
possible to represent any function f(t) as shown below

Equation 1 represents exponential Fourier series representation of a signal f(t) over the interval
(t0, t0+T). The Fourier coefficient is given as

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Relation Between Trigonometric and Exponential Fourier Series:

Consider a periodic signal x(t), the TFS & EFS representations are given below respectively

Compare equation 1 and 2.

Similarly,

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Problems
1. Acontinuous-timesignalx(t)isshowninthefollowingfigure.Sketchandlabeleach of the
following signals.

( a ) x(t-2); ( b)x(2t); ( c)x(t/2); (d) x ( - t)

Sol:

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2. Determine whether the following signals are energy signals, power signals,or
neither.

(d) we know that energy of a signalis

And by using

we obtain

Thus, x [ n ] is a power signal.

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(e) By the definition of power ofsignal

3. Determine whether or not each of the following signals is periodic. If a signalis

periodic, determine its fundamental period.

Sol:

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1. Determine the even and odd components of the followingsignals:

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UNIT– II

CONTINUOUS TIME
FOURIER TRANSFORM

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UNIT - II

CONTINUOUS TIME FOURIER TRANSFORM

INTRODUCTION:

The main drawback of Fourier series is, it is only applicable to periodic signals. There
aresome naturally produced signals such as nonperiodic or aperiodic, which we cannot represent
using Fourier series. To overcome this shortcoming, Fourier developed a mathematical model to
transform signals between time (or spatial) domain to frequency domain & vice versa, which is
called 'Fouriertransform'.

Fourier transform has many applications in physics and engineering such as analysis of
LTI systems, RADAR, astronomy, signal processing etc.

Deriving Fourier transform from Fourier series:

Consider a periodic signal f(t) with period T. The complex Fourier series representation
of f(t) is given as

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In the limit as T→∞,Δf approaches differential df, kΔf becomes a continuous variable f, and
summation becomes integration

Fourier transform of asignal

Inverse Fourier Transformis

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Fourier Transform of Basic Functions:

Let us go through Fourier Transform of basic functions:

FT of GATE Function

FT of Impulse Function:

FT of Unit Step Function:

FT of Exponentials:

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FT of SignumFunction :

Conditions for Existence of Fourier Transform:

Any function f(t) can be represented by using Fourier transform only when the function
satisfies Dirichlet’s conditions. i.e.

• The function f(t) has finite number of maxima andminima.

• There must be finite number of discontinuities in the signal f(t),in the given interval of
time.
• It must be absolutely integrablein the given interval of timei.e.

Properties of Fourier Transform:

Here are the properties of Fourier Transform:

Linearity Property:

Then linearity property states that

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Time Shifting Property:

Then Time shifting property states that

Frequency Shifting Property:

Then frequency shifting property states that

Time Reversal Property:

Then Time reversal property states that

Time Scaling Property:

Then Time scaling property states that

Differentiation and Integration Properties:

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Then Differentiation property states that

and integration property states that

Multiplication and Convolution Properties:

Then multiplication property states that

and convolution property states that

Sampling Theorem and its Importance:

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Statement of Sampling Theorem:

A band limited signal can be reconstructed exactly if it is sampled at a rate atleast twice
the maximum frequency component in it."

The following figure shows a signal g(t) that is bandlimited.

Figure1: Spectrum of band limited signal g(t)

The maximum frequency component of g(t) is fm. To recover the signal g(t) exactly from its
samples it has to be sampled at a rate fs ≥ 2fm.
The minimum required sampling rate fs = 2fm is called “Nyquist rate”.
Proof:
Let g(t) be a bandlimited signal whose bandwidth is fm (ωm =2πfm).

Figure 2: (a) Original signal g(t) (b) SpectrumG(ω)

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Figure 3:

Let gs(t) be the sampled signal. Its Fourier Transform Gs(ω) is given by

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Aliasing:
Aliasing is a phenomenon where the high frequency components of the sampled signal
interfere with each other because of inadequate sampling ωs<ωm

Aliasing leads to distortion in recovered signal. This is the reason why sampling
frequency should be atleast twice the bandwidth of the signal.

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Oversampling:
In practice signal are oversampled, where fs is signi_cantly higher than Nyquist rate to
avoid aliasing.

Problems

1. Find the Fourier transform of the rectangular pulse signal x(t) definedby

Sol: By definition of Fourier transform

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Hence we obtain

The following figure shows the Fourier transform of the given signal x(t)

Figure: Fourier transform of the given signal

2. Find the Fourier transform of the following signalx(t)

Sol: Signal x(t) can be rewritten as

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Hence, we get

The Fourier transform X(w) of x(t) is shown in the following figures

(a) (b)
Fig: (a) Signal x(t) (b) Fourier transform X(w) of x(t)

4. Find the Fourier transform of the periodic impulsetrain

Fig: Train of impulses

Sol: Given signal can be written as

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Thus, the Fourier transform of a unit impulse train is also a similar impulse train. The following
figure shows the Fourier transform of a unit impulse train

Figure: Fourier transform of the given signal

5. Find the Fourier transform of the signumfunction

Sol: Signum function is definedas

The signum function, sgn(t), can be expressed as

Sgn(t)= 2u(t)-1

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We know that

Fig: Signum function

Let

Then applying the differentiation property , we have

Note that sgn(t) is an odd function, and therefore its Fourier transform is a pure imaginary
function of w

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Properties of Fourier Transform:

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Fourier Transform of Basic Functions:

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DISCRETE TIME FOURIER TRANSFORM

Discrete Time Fourier Transforms (DTFT)

Here we take the exponential signals to be where is a real number.The

representation is motivated by the Harmonic analysis, but instead of following the historical
development of the representation we give directly the defining equation.Let be

discrete time signal such that that is sequence is absolutely summable.

Thesequence can be represented by a Fourier integral of theform.

(1)
Where

(2)

Equation (1) and (2) give the Fourier representation of the signal. Equation (1) is referred
as synthesis equation or the inverse discrete time Fourier transform (IDTFT) and equation (2)is
Fourier transform in the analysis equation. Fourier transform of a signal in general is a complex
valued function, we can write

where is magnitude and is the phase of. We also use the term Fourier
spectrumorsimply,thespectrumtoreferto.Thus iscalledthemagnitudespectrumand is
called the phase spectrum. From equation (2) we can see that is a periodic function
with period i.e.. We can interpret (1) as Fourier coefficients in the representation of a
periodic function. In the Fourier series analysis our attention is on the periodic function, here we
are concerned with the representation of the signal. So the roles of the two equation are
interchanged compared to the Fourier series analysis of periodicsignals.

Now we show that if we put equation (2) in equation (1) we indeed get the signal.
Let

where we have substituted from (2) into equation (1) and called the result as.
Since we have used n as index on the left hand side we have used m as the index variable forthe

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sum defining the Fourier transform. Under ourassumptionthat sequence isabsolutely

summable we can interchange the order of integration and summation.Thus

Example: Let

Fourier transform of this sequence will exist if it is absolutely summable. We have

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Fourier transform of Periodic Signals

For a periodic discrete-time signal,

its Fourier transform of this signal is periodic in w with period 2∏ , and is given

Now consider a periodic sequence x[n] with period N and with the Fourier series representation

The Fourier transform is

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Properties of the Discrete Time Fourier Transform:

Let and be two signal, then their DTFT is denoted by and. Thenotation

is used to say that left hand side is the signal x[n] whose DTFT is is given at right hand
side.

1. Periodicity of theDTFT:

2. Linearity of theDTFT:

3. Time Shifting and FrequencyShifting:

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4. Conjugation and Conjugate Symmetry:

From this, it follows that ReX(e jw)is an even function of w and ImX(e jw)is
an odd function of w . Similarly, the magnitude of X(e jw) is an even function and the
phase angle is
an odd function. Furthermore,

5. Differencing andAccumulation

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The impulse train on the right-hand side reflects the dc or average value that can result from
summation.

For example, the Fourier transform of the unit step x[n] u[n] can be obtained by using
the accumulation property.

6. TimeReversal

7. Time Expansion

For continuous-time signal, we have

For discrete-time signals, however, a should be an integer. Let us define a signal with k a
positive integer,

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For k  1, the signal is spread out and slowed down in time, while its Fourier transform is
compressed.

Example: Consider the sequence x[n] displayed in the figure (a) below. This sequence can be
related to the simpler sequence y[n] as shown in (b).

As can be seen from the figure below, y[n] is a rectangular pulse with 2 1 N  , its Fourier
transform is given by

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Using the time-expansion property, we then obtain

8. Differentiation inFrequency

The right-hand side of the above equation is the Fourier transform of jnx[n] .Therefore,
multiplying both sides by j , we see that

9. Parseval’sRelation

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Properties of the Discrete Time Fourier Transform:

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Basic Discrete Time Fourier Transform Pairs:

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UNIT– III

SIGNAL TRANSMISSION
THROUGH LINEAR SYSTEMS

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UNIT – III
SIGNAL TRANSMISSION THROUGH LINEAR SYSTEMS

Linear Systems:

A system is said to be linear when it satisfies superposition and homogenate principles.

Consider two systems with inputs as x1(t), x2(t), and outputs as y1(t), y2(t) respectively. Then,
according to the superposition and homogenate principles,

T [a1 x1(t) + a2 x2(t)] = a1 T[x1(t)] + a2 T[x2(t)]

∴ T [a1 x1(t) + a2 x2(t)] = a1y1(t) + a2 y2(t)

From the above expression, is clear that response of overall system is equal to response of
individual system.

Example:

y(t) = 2x(t)

Solution:

y1 (t) = T[x1(t)] = 2x1(t)

y2 (t) = T[x2(t)] = 2x2(t)

T [a1 x1(t) + a2 x2(t)] = 2[ a1 x1(t) + a2 x2(t)]

Which is equal to a1y1(t) + a2 y2(t). Hence the system is said to be linear.

Impulse Response:

The impulse response of a system is its response to the input δ(t) when the system is
initially at rest. The impulse response is usually denoted h(t). In other words, if the input to an
initially at rest system is δ(t) then the output is named h(t).

δ(t) h(t)
system

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Liner Time variant (LTV) and Liner Time Invariant (LTI) Systems

If a system is both liner and time variant, then it is called liner time variant (LTV) system.

If a system is both liner and time Invariant then that system is called liner time invariant (LTI)
system.
Response of a continuous-time LTI system and the convolution integral

(i) Impulse Response:

The impulse response h(t) of a continuous-time LTI system (represented by T) is defined to
be the response of the system when the input is δ(t), that is,

h(t)=T{δ(t)}------------ (1)
(ii) Response to an ArbitraryInput:

The input x(t) can be expressed as

--------(2)
Since the system is linear, the response y(t of the system to an arbitrary input x( t ) can be
expressed as

--------(3)

Since the system is time-invariant, we have

--------(4)
Substituting Eq. (4) into Eq. (3), we obtain

-------(5)
Equation (5) indicates that a continuous-time LTI system is completely characterized by its impulse
response h( t).
(iii) ConvolutionIntegral:
Equation (5) defines the convolution of two continuous-time signals x ( t ) and h(t) denoted
by

-------(6)

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Equation (6) is commonly called the convolution integral. Thus, we have the fundamental
result that the output of any continuous-time LTI system is the convolution of the input x ( t ) with
the impulse response h(t) of the system. The following figure illustrates the definition of the impulse
response h(t) and the relationship of Eq.(6).

Fig.: Continuous-time LTl system.

(iv) Properties of the Convolution Integral:

The convolution integral has the following properties.

(v) StepResponse:
The step response s(t) of a continuous-time LTI system (represented by T) is defined to
be the response of the system when the input is u(t); that is,
S(t)= T{u(t)}
In many applications, the step response s(t) is also a useful characterization of the system.
The step response s(t) can be easily determined by,

Thus, the step response s(t) can be obtained by integrating the impulse response h(t).
Differentiating the above equation with respect to t, we get

Thus, the impulse response h(t) can be determined by differentiating the step response s(t).

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Distortion less transmission through a system:

Transmission is said to be distortion-less if the input and output have identical wave
shapes. i.e., in distortion-less transmission, the input x(t) and output y(t) satisfy the condition:

y (t) = Kx(t - td)

Where td = delay time and

k = constant.

Take Fourier transform on both sides

FT[ y (t)] = FT[Kx(t -td)]

= K FT[x(t -td)]

According to time shifting property,

Thus, distortion less transmission of a signal x(t) through a system with impulse response h(t) is
achieved when

|H(ω)|=K and (amplitude response)

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A physical transmission system may have amplitude and phase responses as shown below:

FILTERING
One of the most basic operations in any signal processing system is filtering. Filtering is
the process by which the relative amplitudes of the frequency components in a signal are
changed or perhaps some frequency components are suppressed. As we saw in the preceding
section, for continuous-time LTI systems, the spectrum of the output is that of the input
multiplied by the frequency response of the system. Therefore, an LTI system acts as a filter on
the input signal. Here the word "filter" is used to denote a system that exhibits some sort of
frequency-selectivebehavior.

A. Ideal Frequency-Selective Filters:

An ideal frequency-selective filter is one that exactly passes signals at one set of
frequencies and completely rejects the rest. The band of frequencies passed by the filter is
referred to as the pass band, and the band of frequencies rejected by the filter is called the stop
band.
The most common types of ideal frequency-selective filters are the following.

1. Ideal Low-PassFilter:
An ideal low-pass filter (LPF) is specified by

The frequency wcis called the cutoff frequency.

2. Ideal High-PassFilter:
An ideal high-pass filter (HPF) is specified by

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3. Ideal BandpassFilter:
An ideal bandpass filter (BPF) is specified by

4. Ideal BandstopFilter:
An ideal bandstop filter (BSF) is specified by

The following figures shows the magnitude responses of ideal filters

Fig: Magnitude responses of ideal filters (a) Ideal Low-Pass Filter (b)Ideal High-Pass Filter

© Ideal Bandpass Filter (d) Ideal Bandstop Filter

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UNIT– IV
LAPLACE TRANSFORM

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UNIT – IV
LAPLACE TRANSFORM

THE LAPLACE TRANSFORM:

we know that for a continuous-time LTI system with impulse response h(t), the output y(t)of the
system to the complex exponential input of the form estis

A. Definition:

The function H(s) is referred to as the Laplace transform of h(t). For a general continuous-time
signal x(t), the Laplace transform X(s) is defined as

The variable s is generally complex-valued and is expressed as

Relation between Laplace and Fourier transforms:

Laplace transform of x(t)

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Inverse Laplace Transform:

We know that

Conditions for Existence of Laplace Transform:

Dirichlet's conditions are used to define the existence of Laplace transform. i.e.
The function f has finite number of maxima and minima.
 There must be finite number of discontinuities in the signal f ,in the given interval of
time.
It must be absolutely integrable in the given interval of time. i.e.

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Initial and Final Value Theorems

If the Laplace transform of an unknown function x(t) is known, then it is possible to determine
the initial and the final values of that unknown signal i.e. x(t) at t=0+ and t=∞.

Initial Value Theorem

Statement: If x(t) and its 1st derivative is Laplace transformable, then the initial value of x(t) is
given by

Final Value Theorem

Statement: If x(t) and its 1st derivative is Laplace transformable, then the final value of x(t) is
given by

Properties of Laplace transform:

The properties of Laplace transform are:

Linearity Property

Time Shifting Property

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Frequency Shifting Property

Time Reversal Property

Time Scaling Property

Differentiation and Integration Properties

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Multiplication and Convolution Properties

Region of convergence.

The range variation of σ for which the Laplace transform converges is called region of
convergence.

Properties of ROC of Laplace Transform

 ROC contains strip lines parallel to jω axis in s-plane.

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 If x(t) is absolutely integral and it is of finite duration, then ROC is entire s-plane.

If x(t) is a right sided sequence then ROC : Re{s} >σo.

If x(t) is a left sided sequence then ROC : Re{s} <σo.

 If x(t) is a two sided sequence then ROC is the combination of two regions.

ROC can be explained by making use of examples given below:

Example 1: Find the Laplace transform and ROC of x(t)=e− at u(t)x(t)=e−atu(t)

Example 2: Find the Laplace transform and ROC of x(t)=e at u(−t)x(t)=eatu(−t)

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Example 3: Find the Laplace transform and ROC of x(t)=e −at u(t)+e at u(−t)
x(t)=e−atu(t)+eatu(−t)

Referring to the above diagram, combination region lies from –a to a. Hence,

ROC: −a<Res<a

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Causality and Stability

For a system to be causal, all poles of its transfer function must be right half of s-plane.

A system is said to be stable when all poles of its transfer function lay on the left half of
s-plane.

A system is said to be unstable when at least one pole of its transfer function is shifted to
the right half of s-plane.

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• A system is said to be marginally stable when at least one pole of its transferfunction
lies on the jωaxis ofs-plane

LAPLACE TRANSFORMS OF SOME COMMON SIGNALS

A. Unit Impulse Function δ( t):

B. Unit Step Function u(t):

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Some Laplace Transforms Pairs:

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UNIT – V
Z - TRANSFORM

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UNIT – V
Z-Transform
Analysis of continuous time LTI systems can be done using z-transforms. It is a powerful
mathematical tool to convert differential equations into algebraic equations.

The bilateral (two sided) z-transform of a discrete time signal x(n) is given as

The unilateral (one sided) z-transform of a discrete time signal x(n) is given as

Z-transform may exist for some signals for which Discrete Time Fourier Transform (DTFT) does
not exist.

Concept of Z-Transform and Inverse Z-Transform

Z-transform of a discrete time signal x(n) can be represented with X(Z), and it is defined as

The above equation represents the relation between Fourier transform and Z-transform

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Inverse Z-transform:

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Z-Transform Properties:

Z-Transform has following properties:

Linearity Property:

Time Shifting Property:

Multiplication by Exponential Sequence Property

Time Reversal Property

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Differentiation in Z-Domain OR Multiplication by n Property

Convolution Property

Correlation Property

Initial Value and Final Value Theorems

Initial value and final value theorems of z-transform are defined for causal signal.

Initial Value Theorem

For a causal signal x(n), the initial value theorem states that

This is used to find the initial value of the signal without taking inverse z-transform

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Final Value Theorem

For a causal signal x(n), the final value theorem states that

This is used to find the final value of the signal without taking inverse z-transform

Region of Convergence (ROC) of Z-Transform

The range of variation of z for which z-transform converges is called region of convergence of z-
transform.

Properties of ROC of Z-Transforms

• ROC of z-transform is indicated with circle in z-plane.

• ROC does not contain anypoles.

• If x(n) is a finite duration causal sequence or right sided sequence, then the ROC isentire
z-plane except at z =0.

• If x(n) is a finite duration anti-causal sequence or left sided sequence, then the ROCis
entire z-plane except at z =∞.

• If x(n) is a infinite duration causal sequence, ROC is exterior of the circle with radiusa.
i.e. |z| > a.

• If x(n) is a infinite duration anti-causal sequence, ROC is interior of the circle withradius
a. i.e. |z| < a.

• If x(n) is a finite duration two sided sequence, then the ROC is entire z-plane except atz
= 0 & z = ∞.

The concept of ROC can be explained by the following example:

Example 1: Find z-transform and ROC of a n u[n]+a − nu[−n−1] anu[n]+a−nu[−n−1]

The plot of ROC has two conditions as a > 1 and a < 1, as we do not know a.

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In this case, there is no combination ROC.

Here, the combination of ROC is from a<|z|<1/a

Hence for this problem, z-transform is possible when a < 1.

Causality and Stability

Causality condition for discrete time LTI systems is as follows:

A discrete time LTI system is causal when

• ROC is outside the outermostpole.

• In The transfer function H[Z], the order of numerator cannot be grater than the order of
denominator.

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Stability Condition for Discrete Time LTI Systems

A discrete time LTI system is stable when

• its system function H[Z] include unit circle|z|=1.

• all poles of the transfer function lay inside the unit circle|z|=1.
Z-Transform of Basic Signals

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Some Properties of the Z- Transform:

Inverse Z transform:
Three different methods are:
1. Partial fractionmethod
2. Power seriesmethod
3. Long divisionmethod

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Example: A finite sequence x [ n ] is defined as

Find X(z) and its ROC.

Sol: We know that

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For z not equal to zero or infinity, each term in X(z) will be finite and consequently X(z) will
converge. Note that X ( z ) includes both positive powers of z and negative powers of z. Thus,
from the result we conclude that the ROC of X ( z ) is 0 <lzl< m.

Example: Consider the sequence

Find X ( z ) and plot the poles and zeros of X(z).

Sol:

From the above equation we see that there is a pole of ( N- 1)thorder at z = 0 and a pole at z = a .
Since x[n] is a finite sequence and is zero for n < 0, the ROC is IzI>0. The N roots of the
numerator polynomial are at

The root at k = 0 cancels the pole at z = a. The remaining zeros of X ( z ) are at

The pole-zero plot is shown in the following figure with N=8

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