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Dr Reza Danesfahani

Faculty of Engineering

University of Kashan

Kashan, Iran

28th January 2006

Signals and Systems www.danesfahani.com

Part I

Preamble, Introduction, and Overview

Signals and Systems www.danesfahani.com

Preamble I

Slide 3

Syllabus

1

Continuous-time and discrete-time signals: classiﬁcation

and properties.

2

Basic properties of systems: linearity, time-invariance,

causality, stability.

3

Linear time-invariant (LTI) systems: convolution,

characterization, impulse response.

4

Fourier series and Fourier transform: deﬁnition, properties,

frequency response of LTI Introduction to ﬁltering.

5

Sampling: impulse train sampling, sampling theorem,

reconstruction of signals, effects of under-sampling.

6

Z-transforms: deﬁnitions, properties, analysis of LTI

systems, transfer functions.

7

Communication systems: Amplitude modulation (AM),

demodulation

Grading

Signals and Systems www.danesfahani.com

Preamble II

Slide 4

1

Midterm Exam (20%)

2

Final Exam(80%)

Type of Exam: Closed-book. Students may bring one A4

sheet of formulae to exam.

References

A.V. Oppenheim, A.S. Willsky and S.H. Nawab,

Signals & Systems,

Prentice-Hall International, second edition, 1997.

G.E. Carlson,

Signal and Linear System Analysis.,

John Wiley & Sons, second edition, 1998.

R.E. Ziemer, W.H. Tranter and D.R. Fannin,

Signals & Systems, Continuous and Discrete,

Prentice Hall, fourth edition, 1998.

Signals and Systems www.danesfahani.com

Preamble III

Slide 5

S. Haykin and B.V. Veen,

Signals and Systems,

John Wiley & Sons, 1999.

C.L. Phillips and J.M. Parr,

Signals, Systems, and Transforms,

Prentice Hall, second edition, 1999.

F.J. Taylor,

Principles of Signals and Systems,

McGraw-Hill, 1994.

H.P. Hsu,

Theory and Problems of Signals and Systems,

McGraw Hill, 1995.

Signals and Systems www.danesfahani.com

Preamble IV

Slide 6

J.R. Buck, M.M. Daniel and A.C. Singer,

Computer Explorations in Signals and Systems Using

MATLAB,

Prentice Hall, 1997.

L. Balmer,

Signals and Systems, An Introduction,

Prentice Hall, 1998.

E.W. Kamen and B.S. Heck,

Fundamentals of Signals and Systems using MATLAB,

Prentice Hall, 1998.

L.B. Jackson,

Signals, Systems, and Transforms,

Addison-Wesley, 1991.

Signals and Systems www.danesfahani.com

Preamble V

Slide 7

Z.Z. Karu,

Signals and Systems Made Rediculously Simple,

ZiZi Press, 2001.

S.T. Karris,

Signals and Systems with Matlab Applications,

Orchard Publications, second edition, 2003.

B.P. Lathi,

Signal Processing & Linear Systems,

Oxford University Press, 1998.

S. Haykin,

Communication Systems,

John Wiley and Sons, fourth edition, 2001.

Signals and Systems www.danesfahani.com

Part II

Signals and Systems

Signals and Systems www.danesfahani.com

Signals and Systems I

Slide 9

Deﬁnition

A signal is a function representing a physical quantity or

variable, and typically it contains information about the behavior

or nature or phenomenon. For instance, in a RC circuit the

signal may represent the voltage across the capacitor or the

current ﬂowing in the resistor. Mathematically, a signal is

represented as a function of an independent variable t . Thus, a

signal is denoted by x(t ).

Signals and Systems www.danesfahani.com

Signals and Systems II

Slide 10

Deﬁnition

An energy signal is a signal whose total energy is ﬁnite, i.e.

those signals for which E

∞

< ∞. Such a signal must have zero

average power, since in the continuous time case, for example,

P

∞

= lim

T→∞

E

∞

2T

= 0

Example

An example of a ﬁnite-energy signal is a signal that takes on

the value 1 for 0 ≤ t ≤ 1 and 0 otherwise. In this case E

∞

= 1

and P

∞

= 0.

Signals and Systems www.danesfahani.com

Signals and Systems III

Slide 11

Deﬁnition

A power signal is a signal whose average power is ﬁnite, i.e.

those signals for which P

∞

< ∞. Such a signal must have

inﬁnite total energy.

Example

The constant signal x[n] = 4 is a power signal. It has inﬁnite

energy, but average power P

∞

= 16.

There are signals for which neither P

∞

nor E

∞

are ﬁnite. A

simple example is the signal x(t ) = t .

Transformation of the independent variable

1

Time shift: A signal and its time shifted version are identical

in shape, but are displaced or shifted relative to each other.

Signals and Systems www.danesfahani.com

Signals and Systems IV

Slide 12

2

Time reversal: The time reversed version of the signal x(t )

is obtained by a reﬂection about t = 0.

3

Time scale: the independent variable is scaled.

Example

Suppose that we would like to determine the effect of

transforming the independent variable of a given signal, x(t ), to

obtain a signal of the form x(αt +β), where α and β are given

numbers. A systematic approach to doing this is to ﬁrst delay or

advance x(t ) in accordance with the value of β, and then to

perform time scaling and/or time time reversal on the resulting

signal in accordance with the value of α. The delayed or

advanced signal is linearly stretched if [α[ < 1, linearly

compressed if [α[ > 1, and reversed in time if α < 0.

Signals and Systems www.danesfahani.com

Signals and Systems V

Slide 13

Deﬁnition

If a continuous-time signal x(t ) can take on any value in the

continuous interval (a, b), where a may be −∞and b may be

+∞, then the continuous-time signal x(t ) is called an analog

signal. If a discrete-time signal x[n] can take on only a ﬁnite

number of distinct values, then we call this signal a digital

signal.

Signals and Systems www.danesfahani.com

Signals and Systems VI

Slide 14

Deﬁnition

A signal x(t ) is a real signal if its value is a real number, and a

signal x(t ) is a complex signal if its value is a complex number.

A general complex signal x(t ) is a function of the form

x(t ) = x

1

(t ) + jx

2

(t )

where x

1

(t ) and x

2

(t ) are real signals and j =

√

−1.

Deﬁnition

Deterministic signals are those signals whose values are

completely speciﬁed for any given time. Random signals are

those signals that take random values at any given time and

must be characterized statistically.

Signals and Systems www.danesfahani.com

Signals and Systems VII

Slide 15

Deﬁnition

A signal x(t ) or x[n] is referred to as an even signal if

x(−t ) = x(t )

x[−n] = x[n]

A signal x(t ) or x[n] is referred to as an odd signal if

x(−t ) = −x(t )

x[−n] = −x[n]

Signals and Systems www.danesfahani.com

Signals and Systems VIII

Slide 16

The even part of the signal x(t ) is

E¦x(t )¦ =

1

2

[x(t ) + x(−t )]

The odd part of the signal x(t ) is

O¦x(t )¦ =

1

2

[x(t ) −x(−t )]

Similarly,

E¦x[n]¦ =

1

2

[x[n] + x[−n]]

The odd part of the signal x[n] is

O¦x[n]¦ =

1

2

[x[n] −x[−n]]

Signals and Systems www.danesfahani.com

Signals and Systems IX

Slide 17

Example

The odd and even components of x(t ) = e

jt

are

x

e

(t ) =

1

2

(e

jt

+ e

−jt

)

x

o

(t ) =

1

2

(e

jt

−e

−jt

)

Signals and Systems www.danesfahani.com

Signals and Systems X

Slide 18

Deﬁnition

A continuous-time signal x(t ) is said to be periodic with period

T if there is a positive nonzero value of T for which

x(t ) = x(t + T) all t

Any sequence which is not periodic is called a nonperiodic

(or aperiodic) sequence. The fundamental period T

0

of x(t ) is

the smallest positive value of T for which the above equation

holds.

Signals and Systems www.danesfahani.com

Signals and Systems XI

Slide 19

Deﬁnition

A discrete-time signal x[n] is periodic with period N, where N is

a positive integer, if it is unchanged by a time shift of N, i.e. if

x[n] = x[n + N]

for all values of n. The fundamental period N

0

is the smallest

positive value of N for which the above equation holds.

Deﬁnition

The unit step function is deﬁned as

u(t ) ∆

1 t > 0

0 t < 0

Signals and Systems www.danesfahani.com

Signals and Systems XII

Slide 20

Deﬁnition

Dirac delta function, often referred to as the unit impulse(or

delta) function, is a signal that has inﬁnite height and

inﬁnitesimal width. The Dirac delta function δ(x) is deﬁned by

ˆ

∞

−∞

w(x)δ(x) dx = w(0)

where w(x) is any function that is continuous at x = 0.

Properties of impulse function properties

1

Aδ(−t ) = Aδ(t )

2

ˆ

∞

−∞

Aδ(t ) = A

Signals and Systems www.danesfahani.com

Signals and Systems XIII

Slide 21

3

Aδ(t ) = 0 for t = 0

4

Aδ(t −t

0

) + Bδ(t −t

0

) = (A + B)δ(t −t

0

)

5

[y(t )][Aδ(t −t

0

)]Ay(t

0

)δ(t −t

0

)

Deﬁnition

A system is a mathematical model of a physical process that

relates the input signal to the output signal.

interconnections of systems

1

Series (cascade) interconnection

2

Parallel interconnection

3

series-parallel interconnection

Signals and Systems www.danesfahani.com

Signals and Systems XIV

Slide 22

4

Feedback interconnection

Deﬁnition

If the input and output signals are continuous-time signals, then

the system is called a continuous-time system. If the input and

output signals are discrete-time signal, then the system is

called a discrete-time system.

Deﬁnition

A system is said to be memoryless if the output at any time

depends on only the input at that same time. Otherwise, the

system is said to have memory.

Signals and Systems www.danesfahani.com

Signals and Systems XV

Slide 23

Example

The system speciﬁed by the relationship

y[n] = (2x[n] −x

2

[n])

2

is memoryless. An example of a discrete-time system with

memory is accumulator or summer

y[n] =

n

¸

k=−∞

x[k]

Signals and Systems www.danesfahani.com

Signals and Systems XVI

Slide 24

Deﬁnition

A system is said to be invertible if distinct inputs lead to distinct

outputs. For a discrete-time case, if a system is invertible, then

an inverse system exists that, when cascaded with the original

system, yields an output equal to the input to the ﬁrst system.

Example

An example of an invertible continuous-time system is

y(t ) = 2x(t ) for which the inverse system is w(t ) =

1

2

y(t ). An

example of a noninvertible system is y(t ) = x

2

(t ).

Signals and Systems www.danesfahani.com

Signals and Systems XVII

Slide 25

Deﬁnition

A system is called causal if its output y(t ) at an arbitrary time

t = t

0

depends on only the input x(t ) for t ≤ t

0

. That is, the

output of a causal system at the present time depends on only

the present and/or past values of the input, not on its future

values. A system is called noncausal if it is not causal.

Deﬁnition

A system is called time-invariant if a time shift in the input

signal causes the same time shift in the output signal.

Otherwise, the system is time-varying.

Signals and Systems www.danesfahani.com

Signals and Systems XVIII

Slide 26

Example

Consider the continuous-time system deﬁned by

y(t ) = sin[x(t )]

This system is time invariant.

Example

The following discrete-time system is time-varying.

y[n] = nx[n]

Signals and Systems www.danesfahani.com

Signals and Systems XIX

Slide 27

Deﬁnition

A system is linear if the following two properties hold:

1

additivity. The response to x

1

(t ) + x

2

(t ) is y

1

(t ) + y

2

(t ).

2

homogeneity or scaling.The response to ax

1

(t ) is ay

1

(t ),

where a is any complex constant.

The two properties deﬁning a linear system can be combined

into a single statement:

continuous time:ax

1

(t ) + bx

2

(t ) →ay

1

(t ) + by

2

(t )

discrete time:ax

1

[n] + bx

2

[n] →ay

1

[n] + by

2

[n]

Signals and Systems www.danesfahani.com

Signals and Systems XX

Slide 28

Example

The system whose input x(t ) and output y(t ) are related by

y(t ) = tx(t ) is linear, while the system y(t ) = x

2

(t ) in not linear.

Signals and Systems www.danesfahani.com

Part III

Linear Time-Invariant Systems

Signals and Systems www.danesfahani.com

Linear Time-Invariant Systems I

Slide 30

Deﬁnition

If a system is linear and also time-invariant, then it is called a

linear time-invariant (LTI) system.

Deﬁnition

A system is BIBO stable if and only if every bounded input

results in a bounded output. For a ﬁxed, linear, system a

necessary and sufﬁcient condition for BIBO stability is

ˆ

∞

−∞

[h(t )[ dt < ∞

where h(t ) is the impulse response of the system.

Signals and Systems www.danesfahani.com

Linear Time-Invariant Systems II

Slide 31

Deﬁnition

The impulse response h(t ) of a continuous-time LTI system is

deﬁned to be the response of the system when the input is δ(t ).

Deﬁnition

The convolution integral or superposition integral of two

continuous-time signals x(t ) and h(t ) is

y(t ) = x(t ) ∗ h(t ) =

ˆ

∞

−∞

x(τ)h(t −τ) dτ

The properties of convolution are:

1

the commutative property. This property is expressed by

x(t ) ∗ h(t ) = h(t ) ∗ x(t )

Signals and Systems www.danesfahani.com

Linear Time-Invariant Systems III

Slide 32

2

the distributive property. This property is expressed by

x(t ) ∗ [h

1

(t ) + h

2

(t )] = x(t ) ∗ h

1

(t ) + x(t ) ∗ h

2

(t ).

3

the associative property. This property is expressed by

¦x(t ) ∗ h

1

(t )¦ ∗ h

2

(t ) = x(t ) ∗ ¦h

1

(t ) ∗ h

2

(t )¦

4

the shift property. This property states that if

f

1

(t ) ∗ f

2

(t ) = c(t )

then

f

1

(t ) ∗ f

2

(t −T) = c(t −T)

f

1

(t −T) ∗ f

2

(t ) = c(t −T)

and

f

1

(t −T

1

) ∗ f

2

(t −T

2

) = c(t −T

1

−T

2

)

Signals and Systems www.danesfahani.com

Linear Time-Invariant Systems IV

Slide 33

5

The width property. This property states that if the

durations (widths) of f

1

(t ) and f

2

(t ) are T

1

and T

2

,

respectively, then the duration (width) of f

1

(t ) ∗ f

2

(t ) is

T

1

+ T

2

.

Deﬁnition

The step response s(t ) of a continuous-time LTI system is

deﬁned to be the response of the system when the input is a

step function u(t ).

Properties of continuous-time LTI systems:

1

system with or without memory. For a causal

continuous-time LTI system

h(t ) = 0 t < 0

2

causality.

Signals and Systems www.danesfahani.com

Linear Time-Invariant Systems V

Slide 34

3

stability. A continuous-time LTI system is BIBO stable if its

impulse response is absolutely integrable, that is,

ˆ

∞

−∞

[h(τ)[ dτ < ∞

Deﬁnition

The impulse response h[n] of a discrete-time LTI system is

deﬁned to be the response of the system when the input is δ[n].

Signals and Systems www.danesfahani.com

Linear Time-Invariant Systems VI

Slide 35

Deﬁnition

The convolution sum or superposition sum of the sequences

x[n] and h[n] is expressed by

y[n] =

∞

¸

k=−∞

x[k]h[n −k] = x[n] ∗ h[n]

properties of the convolution sum

1

Commutative

x[n] ∗ h[n] = h[n] ∗ x[n]

2

Associative

¦x[n] ∗ h

1

[n]¦ ∗ h

2

[n] = x[n] ∗ ¦h

1

[n] ∗ h

2

[n]¦

Signals and Systems www.danesfahani.com

Linear Time-Invariant Systems VII

Slide 36

3

Distributive

x[n] ∗ ¦h

1

[n] + h

2

[n]¦ = x[n] ∗ h

1

[n] + x[n] ∗ h

2

[n]

Convolution sum operation

y[n] = h[n] ∗ x[n] =

∞

¸

k=−∞

h[k]x[n −k]

1

The impulse response h[k] is time-reversed to obtain h[−k]

and then shifted by n to form h[n −k] = h[−(k −n)] which

is a function of k with parameter n.

2

Two sequences x[k] and h[n −k] are multiplied together for

all values of k with n ﬁxed at some value.

3

The product x[k]h[n −k] is summed over all k to produce a

single output sample y[n].

Signals and Systems www.danesfahani.com

Linear Time-Invariant Systems VIII

Slide 37

4

Steps 1 to 3 are repeated as n varies over −∞to ∞to

produce the entire output y[n].

System with or without memory

Causality. The causality condition for a discrete-time LTI

system is

h[n] = 0 n < 0

Stability. A DT LTI system is BIBO stable if its impulse

response is absolutely summable, that is,

∞

¸

k=−∞

[h[k][ < ∞

Signals and Systems www.danesfahani.com

Part IV

Fourier Series

Signals and Systems www.danesfahani.com

Fourier Series I

Slide 39

Fourier series: The exponential Fourier series of a power

signal v(t ) with period T

0

= 1/f

0

is

v(t ) =

∞

¸

−∞

c

n

e

j 2πnf

0

t

n = 0, 1, 2,

The series coefﬁcients are related to v(t ) by

c

n

=

1

T

0

ˆ

T

0

v(t )e

−j 2πnf

0

t

dt

Since the coefﬁcients are complex quantities in general,

they can be expressed in the polar form

c

n

= [c

n

[e

j arg c

n

Signals and Systems www.danesfahani.com

Fourier Series II

Slide 40

where arg c

n

stands for the angle of c

n

.

The plot of [c

n

[ as a function of f represents the amplitude

spectrum, and the plot of arg c

n

represents the phase

spectrum.

Properties of continuous-time Fourier series:

1

Linearity

Ax(t ) + By(t ) ↔Aa

k

+ Bb

k

2

Time shifting

x(t −t

0

) ↔a

k

e

−jkω

0

t

0

3

Frequency shifting

x(t )e

jMω

0

t

↔a

k−M

4

Conjugation

x

∗

(t ) ↔a

∗

−k

Signals and Systems www.danesfahani.com

Fourier Series III

Slide 41

5

Time reversal

x(−t ) ↔a

−k

6

Time scaling

x(αt ), α > 0 ↔α

k

7

Periodic convolution

ˆ

T

x(τ)y(t −τ) dτ ↔Ta

k

b

k

8

Multiplication

x(t )y(t ) ↔

+∞

¸

l =−∞

a

l

b

k−l

9

Differentiation

dx(t )

dt

↔jkω

0

a

k

= jk

2π

T

a

k

Signals and Systems www.danesfahani.com

Fourier Series IV

Slide 42

10

Integration

ˆ

t

−∞

x(t ) dt ↔

1

jkω

0

a

k

11

Conjugate Symmetry for real signals

a

k

= a

∗

−k

'¦a

k

¦ = '¦a

k

¦

·¦a

k

¦ = −·¦a

k

¦

[a

k

[ = [a

−k

[

arga

k

= −arga

−k

12

Even and odd decomposition

x

e

(t ) ↔'¦a

k

¦

x

o

(t ) ↔j ·¦a

k

¦

Signals and Systems www.danesfahani.com

Fourier Series V

Slide 43

13

Parseval’s relation

1

T

ˆ

T

[x(t )[

2

dt =

+∞

¸

k=−∞

[a

k

[

2

Convergence of the Fourier series: For a signal to have

Fourier series, the following conditions develiped by P.L.

Dirichlet must be satisﬁed

1

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

ˆ

T

[x(t )[ dt < ∞

2

In any ﬁnite interval of time, x(t ) is of bounded variation;

that is, there are more than a ﬁnite number of maxima and

minima during any single period of the signal.

Signals and Systems www.danesfahani.com

Fourier Series VI

Slide 44

3

In any ﬁnite interval of time, there are only a ﬁnite number

of discontinuities. Furthermore, each of these

discontinuities is ﬁnite.

Discrete-time Fourier series pair

x[n] =

¸

k=<N>

a

k

e

jkω

0

n

=

¸

k=<N>

a

k

e

jk(2π/N)n

a

k

=

1

N

¸

n=<N>

x[n]e

−jkω

0

n

=

1

N

¸

n=<N>

x[n]e

−jk(2π/N)n

Properties of discrete-time Fourier series:

1

Linearity

Ax[n] + By[n] ↔Aa

k

+ Bb

k

2

Time shifting

x[n −n

0

] ↔a

k

e

−jkω

0

n

0

Signals and Systems www.danesfahani.com

Fourier Series VII

Slide 45

3

Frequency shifting

x[n]e

jMω

0

n

↔a

k−M

4

Conjugation

x

∗

[n] ↔a

∗

−k

5

Time reversal

x[−n] ↔a

−k

6

Time scaling

x

(m)

[n] =

x[n/m], if n is a multiple of m

0, otherwise

↔

1

m

a

k

7

Periodic convolution

¸

r =<N>

x[r ]y[n −r ] ↔Na

k

b

k

Signals and Systems www.danesfahani.com

Fourier Series VIII

Slide 46

8

Multiplication

x[n]y[n] ↔

¸

l =<N>

a

l

b

k−l

9

First difference

x[n] −x[n −1] ↔(1 −e

−jkω

0

)a

k

10

Running sum

n

¸

k=−∞

x[k] ↔

1

1 −e

−jkω

0

a

k

Signals and Systems www.danesfahani.com

Fourier Series IX

Slide 47

11

Conjugate Symmetry for real signals

a

k

= a

∗

−k

'¦a

k

¦ = '¦a

k

¦

·¦a

k

¦ = −·¦a

k

¦

[a

k

[ = [a

−k

[

arga

k

= −arga

−k

12

Even and odd decomposition

x

e

[n] ↔'¦a

k

¦

x

o

[n] ↔j ·¦a

k

¦

13

Parseval’s relation

1

N

¸

n=<N>

[x[n][

2

=

¸

k=<N>

[a

k

[

2

Signals and Systems www.danesfahani.com

Fourier Series X

Slide 48

Fourier series and LTI systems The output of an LTI

system with a period input signal x(t ) is

y(t ) =

∞

¸

k=−∞

a

k

H(e

jkω

0

)e

jkω

0

t

where a

k

are the Fourier series coefﬁcients of x(t ).

Similarly, for a period signal

y[n] =

¸

k=<N>

a

k

H(e

jkω

0

)e

jkω

0

n

Signals and Systems www.danesfahani.com

Part V

Fourier Transform

Signals and Systems www.danesfahani.com

Fourier Transform I

Slide 50

Fourier transform

X(f ) = F [x(t )] =

ˆ

∞

−∞

x(t )e

−j 2πft

dt

The Fourier transform for periodic signals is of the form of

a linear combination of impulses equally spaced in

frequency, that is,

X(j ω) =

+∞

¸

k=−∞

2πa

k

δ(ω −kω

0

)

where a

k

are the Fourier series coefﬁcients of x(t ).

Convergence of Fourier Transform: For a signal to have

Fourier transform, the following conditions developed by

P.L. Dirichlet must be satisﬁed

Signals and Systems www.danesfahani.com

Fourier Transform II

Slide 51

1

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

ˆ

T

[x(t )[ dt < ∞

2

In any ﬁnite interval of time, x(t ) is of bounded variation;

that is, there are 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. Furthermore, each of these

discontinuities is ﬁnite.

Signals and Systems www.danesfahani.com

Fourier Transform III

Slide 52

The Fourier transform for periodic signals

X(j ω) =

+∞

¸

k=−∞

2πa

k

δ(ω −kω

0

)

Example

The signal x(t ) = sinω

0

t has nonzero Fourier series

coefﬁcients a

1

= 1/2j and a

−1

= −1/2j . Therefore

X(j ω) = π/j δ(ω −ω

0

) −π/j δ(ω +ω

0

)

Fourier transform pairs: Some of the common Fourier

transform pairs are as follows:

Signals and Systems www.danesfahani.com

Fourier Transform IV

Slide 53

1

Rectangular

¸

t

τ

↔τsincf τ

2

Triangular

t

τ

↔τsinc

2

f τ

3

Gaussian

e

−π(bt )

2

↔(1/b)e

−π(f /b)

2

4

Symmetric exponential

e

−b|t |

↔

2b

b

2

+ (2πf )

2

5

Sinc

sinc2Wt ↔

1

2W

¸

f

2W

**Signals and Systems www.danesfahani.com
**

Fourier Transform V

Slide 54

6

Sinc squared

sinc

2

2Wt ↔

1

2W

f

2W

7

Constant

1 ↔δ(f )

8

Phasor

e

j (ω

c

t +φ)

↔e

j φ

δ(f −f

c

)

9

Sinusoid

cos(ω

c

t +φ) ↔

1

2

e

j φ

δ(f −f

c

) + e

−j φ

δ(f + f

c

)

10

Impulse

δ(t −t

d

) ↔e

−j ωt

d

Signals and Systems www.danesfahani.com

Fourier Transform VI

Slide 55

11

Sampling

∞

¸

k=−∞

δ(t −kT

s

) ↔f

s

∞

¸

n=−∞

δ(f −nf

s

)

12

Signum

sgn t ↔1/j πf

13

Step

u(t ) ↔

1

j 2πf

+

1

2

δ(f )

Fourier transform properties: The Fourier transform

properties are as follows

1

Linearity

ax(t ) + by(t ) ↔aX(j ω) + bY(j ω)

Signals and Systems www.danesfahani.com

Fourier Transform VII

Slide 56

2

Time shifting

x(t −t

0

) ↔e

−j ωt

0

X(j ω)

3

Frequency shifting

x(t )e

j ω

0

t

↔X(j (ω −ω

0

))

4

Conjugation

x

∗

(t ) ↔X

∗

(−j ω)

5

Time reversal

x(−t ) ↔X(−j ω)

6

Time and frequency scaling

x(αt ) ↔

1

[α[

X

j ω

α

7

Convolution

x(t ) ∗ y(t ) ↔X(j ω)Y(j ω)

Signals and Systems www.danesfahani.com

Fourier Transform VIII

Slide 57

8

Multiplication

x(t )y(t ) ↔

1

2π

X(j ω) ∗ Y(j ω)

9

Differentiation in time

dx(t )

dt

↔j ωX(j ω)

10

Integration

ˆ

t

−∞

x(t ) dt ↔

1

j ω

X(j ω) +πX(0)δ(ω)

11

Differentiation in frequency

tx(t ) ↔j

d

dω

X(j ω)

Signals and Systems www.danesfahani.com

Fourier Transform IX

Slide 58

12

Conjugate Symmetry for real signals

X(j ω) = X

∗

(−j ω)

'¦X(j ω)¦ = '¦X(−j ω)¦

·¦X(j ω)¦ = −·¦X(−j ω)¦

[X(j ω)[ = [X(−j ω)[

argX(j ω) = −argX(−j ω)

13

Symmetry

x(t ) real and even ↔X(j ω) real and even

x(t ) real and odd ↔X(j ω) imaginary and odd

14

Even and odd decomposition for real signals

x

e

(t ) ↔'¦X(j ω)¦

x

o

(t ) ↔j ·¦X(j ω)¦

Signals and Systems www.danesfahani.com

Fourier Transform X

Slide 59

15

Parseval’s relation

ˆ

+∞

−∞

[x(t )[

2

dt =

1

2π

ˆ

+∞

−∞

[X(j ω)[

2

dω

Convergence of Fourier transform: The sufﬁcient

conditions for the convergence are:

1

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

ˆ

T

[x(t )[ dt < ∞

2

In any ﬁnite interval of time, x(t ) is of bounded variation;

that is, there are 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. Furthermore, each of these

discontinuities is ﬁnite.

Signals and Systems www.danesfahani.com

Fourier Transform XI

Slide 60

Duality theorem: This theorem states that if v(t ) and V(f )

constitute a known transform pair, and if there exists a time

function z(t ) related to the function V(f ) by

z(t ) = V(f )

then

F[z(t )] = v(−f )

where v(−f ) equals v(t ) with t = −f .

Signals and Systems www.danesfahani.com

Part VI

Discrete-Time Fourier Transform

Signals and Systems www.danesfahani.com

Discrete-Time Fourier Transform I

Slide 62

Synthesis equation

x[n] =

1

2π

ˆ

2π

X(e

j ω

)e

j ωn

dω

Analysis equation

X(e

j ω

) =

+∞

¸

n=−∞

x[n]e

−j ωn

Signals and Systems www.danesfahani.com

Discrete-Time Fourier Transform II

Slide 63

A periodic sequence x[n] with period N and with the

Fourier series representation

x[n] =

¸

k=<N>

a

k

e

jk(2π/N)n

has the Fourier transform

X(e

j ω

) =

+∞

¸

k=−∞

2πa

k

δ

ω −

2πk

N

Example

The DTFT of x[n] = cos ω

0

n is

X(e

j ω

) = πδ (ω −ω

0

) +πδ (ω +ω

0

)

Signals and Systems www.danesfahani.com

Discrete-Time Fourier Transform III

Slide 64

Basic properties of the DTFT are the following:

1

Periodicity

X(e

j (ω+2π)

) = X(e

j ω

)

2

Linearity

ax

1

[n] + bx

2

[n] ↔a

1

X

1

(e

j ω

) + bX

2

(e

j ω

)

3

Time shifting

x[n −n

0

] ↔e

−j ωn

0

X(e

j ω

)

4

Frequency shifting

e

j ω

0

n

x[n] ↔X(e

j (ω−ω

0

)

)

5

Conjugation

x

∗

[n] ↔X

∗

(e

−j ω

)

where ∗ denotes the complex conjugate.

Signals and Systems www.danesfahani.com

Discrete-Time Fourier Transform IV

Slide 65

6

Time reversal

x[−n] ↔X(e

−j ω

)

7

Time expansion

x

(k)

[n] =

x[n/k], if n = multiple of k

0, if n = multiple of k

↔X(e

j ω

)

8

Convolution

x[n] ∗ y[n] ↔X(e

−j ω

)Y(e

−j ω

)

9

Multiplication

x[n]y[n] ↔

1

2π

ˆ

2π

X(e

j θ

)Y(e

j (ω−θ)

) dθ

10

Differencing in time

x[n] −x[n −1] ↔(1 −e

−j ω

)X(e

j ω

)

Signals and Systems www.danesfahani.com

Discrete-Time Fourier Transform V

Slide 66

11

Accumulation

n

¸

k=−∞

x[n] ↔

1

1 −e

−j ω

X(e

j ω

) +πX(e

j 0

)

+∞

¸

k=−∞

δ(ω −2πk)

12

Differentiation in frequency

nx[n] ↔

dX(e

j ω

)

dω

13

Conjugate Symmetry for real signals

X(j ω) = X

∗

(e

−j ω

)

'¦X(e

j ω

)¦ = '¦X(e

−j ω

)¦

·¦X(e

j ω

)¦ = −·¦X(e

−j ω

)¦

[X(e

j ω

)[ = [X(e

−j ω

)[

argX(e

j ω

) = −argX(e

−j ω

)

Signals and Systems www.danesfahani.com

Discrete-Time Fourier Transform VI

Slide 67

14

Symmetry for real, even signals

x[n] real and even ↔X(e

j ω

) real and even

15

Symmetry for real, odd signals

x[n] real and odd ↔X(e

j ω

) purely imaginary and odd

16

even-odd decomposition

x

e

[n] ↔'¦X(e

j ω

)¦

x

o

[n] ↔j ·¦X(e

j ω

)¦

17

Parseval’s relation

+∞

¸

n=−∞

[x[n][

2

=

1

2π

ˆ

2π

[X(e

j ω

)[

2

dω

Signals and Systems www.danesfahani.com

Discrete-Time Fourier Transform VII

Slide 68

Basic discrete-time Fourier transform pairs

1

¸

k=<N>

a

k

e

j (2π/N)n

↔2π

+∞

¸

k=−∞

a

k

δ

ω −

2πk

N

2

e

j ω

0

n

↔2π

+∞

¸

l =−∞

δ(ω −ω

0

−2πl )

3

cos ω

0

n ↔π

+∞

¸

l =−∞

¦δ(ω −ω

0

−2πl ) +δ(ω +ω

0

−2πl )¦

Signals and Systems www.danesfahani.com

Discrete-Time Fourier Transform VIII

Slide 69

4

sinω

0

n ↔

π

j

+∞

¸

l =−∞

¦δ(ω −ω

0

−2πl ) −δ(ω +ω

0

−2πl )¦

5

x[n] = 1 ↔2π

+∞

¸

l =−∞

δ(ω −2πl )

6

periodic square wave with period N

x[n] =

1, [n[ ≤ N

1

0, N

1

< [n[ ≤ N/2

↔2π

+∞

¸

k=−∞

a

k

δ

ω −

2πk

N

7

+∞

¸

k=−∞

δ[n −kN] ↔

2π

N

+∞

¸

k=−∞

δ

ω −

2πk

N

**Signals and Systems www.danesfahani.com
**

Discrete-Time Fourier Transform IX

Slide 70

8

a

n

u[n], [a[ < 1 ↔

1

1 −ae

−j ω

9

x[n] =

1, [n[ ≤ N

1

0, [n[ > N

1

↔

sin[ω(N

1

+

1

2

)]

sin(ω/2)

10

sinWn

πn

=

W

π

sinc

Wn

π

↔X(ω) =

1, 0 ≤ [ω[ ≤ W

0, W < [ω[ ≤ π

0 < W < π, X(ω) periodic with period 2π.

11

δ[n] ↔1

Signals and Systems www.danesfahani.com

Discrete-Time Fourier Transform X

Slide 71

12

u[n] ↔

1

1 −e

−j ω

+

+∞

¸

k=−∞

πδ(ω −2πk)

13

δ[n −n

0

] ↔e

−j ωn

0

14

(n + 1)a

n

u[n], [a[ < 1 ↔

1

(1 −ae

−j ω

)

2

15

(n + r −1)!

n!(r −1)!

a

n

u[n], [a[ < 1 ↔

1

(1 −ae

−j ω

)

r

Signals and Systems www.danesfahani.com

Discrete-Time Fourier Transform XI

Slide 72

Example

Find the convolution of h[n] = α

n

u[n] and x[n]β

n

u[n].

H(e

j ω

) =

1

1 −αe

−j ω

X(e

j ω

) =

1

1 −βe

−j ω

So

Y(e

j ω

) =

1

(1 −αe

−j ω

)(1 −βe

−j ω

)

By PFE and taking the inverse transform

y[n] =

1

α −β

[α

n+1

u[n] −β

n+1

u[n]]

Signals and Systems www.danesfahani.com

Discrete-Time Fourier Transform XII

Slide 73

Example

Find the impulse response of the LTI system characterized by

y[n] −

3

4

y[n −1] +

1

8

y[n −2] = 2x[n]

H(e

j ω

) =

2

1 −

3

4

e

−j ω

+

1

8

e

−j 2ω

By performing PFE on H(e

j ω

) and by inspection

h[n] = 4

1

2

n

u[n] −2

1

4

n

u[n]

Signals and Systems www.danesfahani.com

Part VII

Sampling

Signals and Systems www.danesfahani.com

Sampling I

Slide 75

Sampling theorem: A continuous-time signal x(t ) can be

uniquely reconstructed from its samples x

s

(t ) with two

conditions:

1

x(t ) must be band-limited with a maximum frequency ω

M

.

2

Sampling frequency ω

s

of x

s

(t ) must be greater than 2ω

M

,

i.e. ω

s

> 2ω

M

.

The second condition is also known as Nyquist criterion.

ω

s

is referred to as Nyquist frequency, i.e., the smallest

possible sampling frequency in order to recover the original

analog signal from its samples.

Sampling of a continuous-time signal x(t ) can be done by

obtaining its values at periodic times x(kT), where T is the

sampling period.

Signals and Systems www.danesfahani.com

Sampling II

Slide 76

Sampling is done by multiplying x(t ) by a train of impulses

with a period T

x

p

(t ) = x(t )p(t )

where

p(t ) =

+∞

¸

n=−∞

δ(t −nT)

x

p

(t ) is an impulse train with the amplitude of the impulses

equal to the samples of x(t ) at intervals spaces by T; that

is,

x

p

(t ) =

+∞

¸

n=−∞

x(nT)δ(t −nT)

From the multiplication property

X

p

(j ω) =

1

2π

[X(j ω)] ∗ P(j ω)

Signals and Systems www.danesfahani.com

Sampling III

Slide 77

As

P(j ω) =

2π

T

+∞

¸

k=−∞

δ(ω −kω

s

)

Since convolution with an impulse simply shifts a signal, it

follows that

X

p

(j ω) =

1

T

+∞

¸

k=−∞

X(j (ω −kω

s

))

That is, X

p

(j ω) is a periodic function of ω consisting of

superposition of shifted replicas of X(j ω), scaled by 1/T.

Signals and Systems www.danesfahani.com

Sampling IV

Slide 78

Reconstruction of a signal from its samples using

interpolation

x

r

(t ) = x

p

(t ) ∗ h(t )

where

x

p

(t ) =

+∞

¸

n=−∞

x(nT)δ(t −nT)

Hence,

x

r

(t ) =

+∞

¸

n=−∞

x(nT)h(t −nT)

For an ideal LPF

h(t ) =

ω

c

T sin(ω

c

t )

πω

c

t

Signals and Systems www.danesfahani.com

Sampling V

Slide 79

So

x

r

(t ) =

+∞

¸

n=−∞

x(nT)

ω

c

T

π

sin(ω

c

(t −nT))

ω

c

(t −nT)

Signals and Systems www.danesfahani.com

Part VIII

Discrete Fourier Transform

Signals and Systems www.danesfahani.com

Discrete Fourier Transform I

Slide 81

Discrete Fourier transform: Let x[n] be a ﬁnite-length

sequence of length N, that is,

x[n] = 0 outside the range 0 ≤ n ≤ N −1

The DFT of x[n], denoted as X[k], is deﬁned by

X[k] =

N−1

¸

n=0

x[k]W

kn

N

k = 0, 1, , N −1

where W

N

is the N

th

root of unity given by

W

N

= e

−j (2π/N)

Signals and Systems www.danesfahani.com

Discrete Fourier Transform II

Slide 82

The inverse DFT (IDFT) is given by

x[n] =

1

N

N−1

¸

n=0

X[k]W

−kn

N

n = 0, 1, , N −1

Properties of the DFT: Basic properties of the DFT are the

following:

1

Linearity

a

1

x

1

[n] + a

2

x

2

[n] ↔a

1

X

1

[k] + a

2

X

2

[k]

2

Time shifting

x[n −n

0

]

modN

↔W

kn

0

N

X[k] W

N

= e

−j 2π/N

3

Frequency shifting

W

−kn

0

N

x[n] ↔X[k −k

0

]

modN

Signals and Systems www.danesfahani.com

Discrete Fourier Transform III

Slide 83

4

Conjugation

x

∗

[n] ↔X

∗

[−k]

modN

where ∗ denotes the complex conjugate.

5

Time reversal

x[−n]

modN

↔X[−k]

modN

6

Duality

X[n] ↔Nx[−k]

modN

7

Circular convolution

x

1

[n] ⊗x

2

[n] ↔X

1

[k]X

2

[k]

where

x

1

[n] ⊗x

2

[n] =

N−1

¸

i =0

x

1

[i ]x

2

[n −i ]

modN

Signals and Systems www.danesfahani.com

Discrete Fourier Transform IV

Slide 84

8

Multiplication

x

1

[n]x

2

[n] ↔

1

N

X

1

[k] ⊗X

2

[k]

where

X

1

[k] ⊗X

2

[k] =

N−1

¸

i =1

X

1

[i ]X

2

[k −i ]

modN

9

Parseval’s relation

N−1

¸

n=0

[x[n][

2

=

1

N

¸

n=0

N −1[X[k][

2

Signals and Systems www.danesfahani.com

Part IX

The z-Transform

Signals and Systems www.danesfahani.com

The z-Transform I

Slide 86

Deﬁnition

For a general discrete-time signal x[n], the bilateral (or

two-sided) z-transform X(z) is

X(z) =

∞

¸

n=−∞

x[n]z

−n

The variable z is generally complex-valued and is expressed in

polar form as

z = re

j Ω

where r is the magnitude of z and Ω is the angle of z.

Signals and Systems www.danesfahani.com

The z-Transform II

Slide 87

Deﬁnition

The range of values of the complex variable z for which the

z-transform converges is called the region of

convergence (ROC).

Signals and Systems www.danesfahani.com

The z-Transform III

Slide 88

Example

The z-transform of x[n] = a

n

u[n] is

X(z) =

∞

¸

n=0

(az

−1

)

n

The ROC is the range of values of z for which [z[ > [a[.

Therefore,

X(z) =

1

1 −az

−1

=

z

z −a

, [z[ > [a[

Signals and Systems www.danesfahani.com

The z-Transform IV

Slide 89

Example

The z-transform of x[n] = −a

n

u[−n −1] is

X(z) =

1

1 −az

−1

=

z

z −a

, [z[ < [a[

Example

The z-transform of x[n] = 7

1

3

u[n] −6

1

2

u[n] is

X(z) =

7

1 −

1

3

z

−1

−

6

1 −

1

2

z

−1

, [z[ >

1

2

Signals and Systems www.danesfahani.com

The z-Transform V

Slide 90

Example

The z-transform of x[n] =

1

3

n

sin

π

4

n

u[n] is

X(z) =

1

3

√

2

z

(z −

1

3

e

j π/4

)(z −

1

3

e

−j π/4

)

, [z[ > 1/3

The properties of the region of convergence (ROC) for the

z-tansform are as follows:

1

The ROC of X(z) consists of a ring in the z-plane centered

about the origin.

2

The ROC does not contain any poles.

Signals and Systems www.danesfahani.com

The z-Transform VI

Slide 91

3

If x[n] is of ﬁnite duration, then the ROC is the entire

z-plane, except possibly z = 0 and/or z = ∞.

Example

The ROC of δ[n] ↔1 consists of the entire z-plane. The ROC of

δ[n −1] ↔z

−1

consists of the entire z-plane, including z = ∞but

excluding z = 0. The ROC of δ[n + 1] ↔z consists of the entire

z-plane (including z = 0).

4

If x[n] is a right-sided sequence, and if the circle [z[ = r

0

is

in the ROC, then all ﬁnite values of z = 0 for which [z[ > r

0

will also be in the ROC.

5

If x[n] is a left-sided sequence, and if the circe [z[ = r

0

is in

the ROC, then all values of z for which 0 < [z[ < r

0

will be in

the ROC.

Signals and Systems www.danesfahani.com

The z-Transform VII

Slide 92

6

If x[n] is two-sided, and if the circle [z[ = r

0

is in the ROC,

then the ROC will consist of a ring in the z-plane that

includes the circle [z[ = r

0

.

Example

The z-transform of x[n] = b

|n|

, b > 0 is

X(z) =

1

1 −bz

−1

−

1

1 −b

−1

z

−1

, b < [z[ <

1

b

7

If the z-transform X(z) of x[n] is rational, then its ROC is

bounded by the poles or extends to inﬁnity.

8

If the z-transform X(z) of x[n] is rational, and if x[n] is right

sided, then the ROC is the region in the z-plane outside the

outermost pole – i.e., outside the circle of radius equal to

the largest magnitude of the poles of X(z). Furthermore, if

x[n] is causal (i.e., if it is right sided and equal to 0 for

n < 0), then the ROC also include z = ∞.

Signals and Systems www.danesfahani.com

The z-Transform VIII

Slide 93

9

If the z-transform X(z) of x[n] is rational, and if x[n] is left

sided, then the ROC is the region in the z-plane inside the

innermost nonzero pole – i.e., inside the circle of radius

equal to the smallest magnitude of the poles of X(z) other

than any at z = 0 and extending inward to and possible

including z = 0. In particular, if x[n] is anticausal (i.e., if it is

left sided and equal to 0 for n > 0), then the ROC also

includes z = 0.

Inverse z-transform:

1

Inversion formula

x[n] =

1

2πj

‰

X(z)z

n−1

dz

where

denotes integration around a counterclockwise

closed circular contour centered at the origin and with

radius r .

Signals and Systems www.danesfahani.com

The z-Transform IX

Slide 94

2

Use of table of z-transform pairs. In this method, X(z) is

expressed as a sum

X(z) = X

1

(z) + + X

n

(z)

where X

1

(z) + + X

n

(z) are functions with know inverse

transforms x

1

[n] + + x

n

[n].

3

Power series expanding.

4

Partial fraction expansion. This method provides the most

generally useful inverse z-transform, especially when X(z)

is a rational function of z.

Signals and Systems www.danesfahani.com

The z-Transform X

Slide 95

Example

Consider the z-transform

X(z) =

3 −

5

6

z

−1

(1 −

1

4

z

−1

)(1 −

1

3

z

−1

)

, [z[ >

1

3

The inverse z-transform is

x[n] =

1

4

n

u[n] + 2

1

3

n

u[n]

Signals and Systems www.danesfahani.com

The z-Transform XI

Slide 96

Example

Consider the z-transform

X(z) =

3 −

5

6

z

−1

(1 −

1

4

z

−1

)(1 −

1

3

z

−1

)

,

1

4

< [z[ <

1

3

The inverse z-transform is

x[n] =

1

4

n

u[n] −2

1

3

n

u[−n −1]

Signals and Systems www.danesfahani.com

The z-Transform XII

Slide 97

Example

Consider the z-transform

X(z) =

3 −

5

6

z

−1

(1 −

1

4

z

−1

)(1 −

1

3

z

−1

)

, [z[ <

1

4

The inverse z-transform is

x[n] = −

1

4

n

u[−n −1] −2

1

3

n

u[−n −1]

Signals and Systems www.danesfahani.com

The z-Transform XIII

Slide 98

Example

Consider the z-transform

X(z) = 4z

2

+ 2 + 3z

−1

. 0 < [z[ < ∞

From the power-series deﬁnition of the z-transform, we can

determine the inverse transform of X(z) by inspection:

x[n] = 4δ[n + 2] + 2δ[n] + 3δ[n −1]

Properties of the z-transform are as follows:

1

Linearity

ax

1

[n] + bx

2

[n] ↔aX

1

(z) + bX

2

(z), ROC: R

1

∩ R

2

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The z-Transform XIV

Slide 99

2

Time shifting

x[n −n

0

] ↔z

−n

0

X(z)

ROC: R, except for the possible addition or deletion of the region.

3

scaling in the z-domain

z

n

0

x[n] ↔X

z

z

0

, ROC : [z

0

[R

4

Time reversal

x[−n] ↔X(z

−1

), ROC:

1

R

5

Time expansion

x

k

[n] ↔X(z

k

), ROC : R

1/k

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The z-Transform XV

Slide 100

6

Conjugation

x

∗

[n] ↔X

∗

(z

∗

), ROC: R

7

Differentiation in the z-domain

nx[n] ↔−z

dX(z)

dz

, ROC: R

Signals and Systems www.danesfahani.com

The z-Transform XVI

Slide 101

Example

Consider the following z-transform

X(z) =

az

−1

(1 −az

−1

)

2

, [z[ > [a[

From

a

n

u[n] ↔

1

1 −az

−1

, [z[ > [a[

Hence

na

n

u[n] ↔−z

d

dz

1

1 −az

−1

=

az

−1

(1 −az

−1

)

2

, [z[ > [a[

8

Convolution

x

1

[n] ∗ x

2

[n] ↔X

1

(z)X

2

(z), ROC: R

1

∩ R

2

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The z-Transform XVII

Slide 102

9

First difference

x[n] −x[n −1] ↔(1 −z

−1

)X(z), ROC: R ∩ ¦[z[ > 0¦

10

Accumulation

n

¸

k=−∞

x[k] ↔(1 −z

−1

)

−1

X(z), ROC: R ∩ ¦[z[ > 1¦

11

Initial value theorem. If x[n] = 0 for n < 0, then

x[0] = lim

z→∞

X(z)

Some common z-transform pairs are as follows:

1

δ[n] ↔1, ROC: All z

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The z-Transform XVIII

Slide 103

2

u[n] ↔

1

1 −z

−1

, ROC: [z[ > 1

3

−u[−n −1] ↔

1

1 −z

−1

, ROC: [z[ < 1

4

δ[n−m] ↔z

−m

, ROC: All z except 0 (if m > 0) or ∞(if m < 0)

5

α

n

u[n] ↔

1

1 −αz

−1

, ROC: [z[ > [α[

6

−α

n

u[−n −1] ↔

1

1 −αz

−1

, ROC: [z[ < [α[

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The z-Transform XIX

Slide 104

7

nα

n

u[n] ↔

αz

−1

(1 −αz

−1

)

2

, ROC: [z[ > [α[

8

−nα

n

u[−n −1] ↔

αz

−1

(1 −αz

−1

)

2

, ROC: [z[ < [α[

9

[cos ω

0

n]u[n] ↔

1 −[cos ω

0

]z

−1

1 −[2 cos ω

0

]z

−1

+ z

−2

, ROC: [z[ > 1

10

[sinω

0

n]u[n] ↔

[sinω

0

]z

−1

1 −[2 cos ω

0

]z

−1

+ z

−2

, ROC: [z[ > 1

Signals and Systems www.danesfahani.com

The z-Transform XX

Slide 105

11

[r

n

cos ω

0

n]u[n] ↔

1 −[r cos ω

0

]z

−1

1 −[2r cos ω

0

]z

−1

+ r

2

z

−2

, ROC: [z[ > r

12

[r

n

sinω

0

n]u[n] ↔

[r sinω

0

]z

−1

1 −[2r cos ω

0

]z

−1

+ r

2

z

−2

, ROC: [z[ > r

Deﬁnition

A discrete-time LTI system is causal if and only if the ROC of its

system function is the exterior of a circle, including inﬁnity.

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The z-Transform XXI

Slide 106

Deﬁnition

A discrete-time LTI system with rational system function H(z) is

causal if and only if (a) the ROC is the exterior of a circle

outside the outermost pole; and (b) with H(z) expressed as a

ratio of polynomials in z, the order of the numerator cannot be

greater than the order of the denominator.

Example

The system with the following system function is not causal,

because the numerator of H(z) is of higher order than the

denominator.

H(z) =

z

3

−2z

2

+ z

z

2

+

1

4

z +

1

8

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The z-Transform XXII

Slide 107

Example

Consider a system with system function

H(z) =

1

1 −

1

2

z

−1

+

1

1 −2z

−1

, [z[ > 2

The impulse response of this system is

h[n] =

¸

1

2

n

+ 2

n

u[n]

Since h[n] = 0 for n < 0, we conclude that the system is causal.

Signals and Systems www.danesfahani.com

The z-Transform XXIII

Slide 108

Deﬁnition

An LTI system is stable if and only if the ROC of its system

function H(z) includes the unit circle, [z[ = 1.

Signals and Systems www.danesfahani.com

The z-Transform XXIV

Slide 109

Example

Consider a system with system function

H(z) =

1

1 −

1

2

z

−1

+

1

1 −2z

−1

, 1/2 < [z[ < 2

The impulse response of this system is

h[n] =

1

2

n

u[n] −2

n

u[−n −1]

Since h[n] is absolutely summable, we conclude that the

system is stable. Further, it is seen that the system is

noncausal.

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The z-Transform XXV

Slide 110

Example

Consider a system with system function

H(z) =

1

1 −

1

2

z

−1

+

1

1 −2z

−1

, [z[ < 1/2

The impulse response of this system is

h[n] = −

¸

1

2

n

+ 2

n

u[−n −1]

Since h[n] is neither stable nor causal.

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The z-Transform XXVI

Slide 111

Deﬁnition

A causal LTI system with rational system function H(z) is stable

if and only if all of the poles of H(z) lie inside the unit circle –

i.e., they must all have magnitude smaller than 1.

Example

Consider a causal system with system function

H(z) =

1

1 −az

−1

which has a pole at z = a. For this system to be stable, its pole

must be inside the unit circle, i.e., we must have [a[ < 1.

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The z-Transform XXVII

Slide 112

For system characterized by linear constant-coefﬁcient

difference equation, the properties of the z-transform

provide a particularly convenient procedure for obtaining

the system function, frequency response, or time-domain

response of the system.

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The z-Transform XXVIII

Slide 113

Example

Consider an LTI system for which the input x[n] and output y[n]

satisfy the linear constant-coefﬁcient difference equation

y[n] −

1

2

y[n −1] = x[n] +

1

3

x[n −1]

Applying the z-trasform

H(z) =

Y(z)

X(z)

=

1 +

1

3

z

−1

1 −

1

2

z

−1

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Part X

Laplace Transform

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Laplace Tranform I

Slide 115

The basic inverse Laplace transform equation is

x(t ) =

1

2πj

ˆ

σ+j ω

σ−j ω

X(s)e

st

ds

This equation states that x(t ) can be represented as a

weighted integral of complex exponentials. The contour of

integration in this equation is the straight line in the s-plane

corresponding to all points s satisfying 's = σ. This line is

parallel to the j ω-axis. Furthermore, we can choose any

such line in the ROC–i.e., we can choose any value σ such

that X(σ + j ω) converges.

Some common Laplace transform pairs are as follows

1

δ(t ) ↔1

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Laplace Tranform II

Slide 116

2

u(t ) ↔

1

s

3

t

n

n!

u(t ) ↔

1

s

n+1

4

e

−at

u(t ) ↔

1

s + a

5

t

n

e

−at

n!

u(t ) ↔

1

(s + a)

n+1

6

sin(ω

0

t )u(t ) ↔

ω

0

s

2

+ω

2

0

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Laplace Tranform III

Slide 117

7

cos(ω

0

t )u(t ) ↔

s

s

2

+ω

2

0

8

t sin(ω

0

t )u(t ) ↔

2ω

0

s

(s

2

+ω

2

0

)

2

9

t cos(ω

0

t )u(t ) ↔

(s

2

−ω

2

0

)

(s

2

+ω

2

0

)

2

10

e

−at

sin(ω

0

t )u(t ) ↔

ω

0

(s + a)

2

+ω

2

0

11

e

−at

cos(ω

0

t )u(t ) ↔

s + a

(s + a)

2

+ω

2

0

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Laplace Tranform IV

Slide 118

The Laplace transform of a general continuous-time signal

x(t ) is deﬁnes as

X(s) ∆

ˆ

+∞

−∞

x(t )e

−st

dt

The independent variable s corresponds to a complex

variable in the exponent of e

−st

. The complex variable s

can be written as s = σ + j ω, with σ and ω the real and

imaginary parts, respectively.

Properties of the Laplace transform are as follows:

1

Linearity

ax

1

(t ) + bx

2

(t ) ↔aX

1

(s) + bX

2

(s), ROC contains R

1

∩ R

2

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Laplace Tranform V

Slide 119

2

Time shifting

x(t −t

0

) ↔e

st

0

X(s), ROC = R

3

Shifting in the s-domain

e

s

0

t

x(t ) ↔X(s −s

0

), ROC = R +'¦s

0

¦

4

Time scaling

x(at ) ↔

1

[a[

X

s

a

, ROC =

R

a

5

Conjugation

x

∗

(t ) ↔X

∗

(s

∗

), ROC=R

6

Convolution property

x

1

(t ) ∗ x

2

(t ) ↔X

1

(s)X

2

(s), R

1

∩ R

2

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Laplace Tranform VI

Slide 120

7

Differentiation in the time domain

x(t )

dt

↔sX(s), ROC = R

8

Differentiation in the s-domain

−tx(t ) ↔

dX(s)

ds

, ROC = R

9

Integration in the time domain

ˆ

t

−∞

x(τ) dτ ↔

1

s

X(s), ROC contains R ∩ ¦'¦s¦¦ > 0

10

Initial- and ﬁnal-value theorems state that if x(t ) = 0 for

t < 0 and x(t ) contains no impulses or higher-order

singularities at t = 0, then x(0

+

) = lim

x→∞

sX(s) and

lim

t →∞

x(t ) = lim

s→0

sX(s).

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Laplace Tranform VII

Slide 121

The properties of the region of convergence (ROC) for

Laplace transform are as follows:

1

The ROC of X(s) consists of strips parallel to the j ω-axis in

the s-plane.

2

For rational Laplace transforms, the ROC does not contain

any poles.

3

If x(t ) is of ﬁnite duration and is absolutely integrable, then

the ROC is the entire s-plane.

4

If x(t ) is right sided, and if the line '¦s¦ = σ

0

is in the ROC,

then all values of s for which '¦s¦ > σ

0

will also be in the

ROC.

5

If x(t ) is left sided, and if the line '¦s¦ = σ

0

is in the ROC,

then all values of s for which '¦s¦ < σ

0

will also be in the

ROC.

6

If x(t ) is two sided, and if the line '¦s¦ = σ

0

is in the ROC,

then the ROC will consist of a strip in the s-plane that

includes the line '¦s¦ = σ

0

.

Signals and Systems www.danesfahani.com

Laplace Tranform VIII

Slide 122

7

If the Laplace transform X(s) of x(t ) is rational, then its

ROC is bounded by poles or extends to inﬁnity. In addition,

no poles of X(s) are contained in the ROC.

8

If the Laplace transform X(s) of x(t ) is rational, then if x(t )

is right sided, the ROC is the region in the s-plane to the

right of the rightmost pole. If x(t ) is left sided, the ROC is

the region in the s-plane to the left of the leftmost pole.

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Part XI

Amplitude (Linear) Modulation

Double Sideband-Suppressed Carrier

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Double Sideband (DSB) I

Slide 124

Most of the common signalling techniques consist of

modulating an analog or digital baseband signal onto a

carrier.

All can be represented by the generic bandpass form

s(t ) = g(t ) cos(2πf

c

t )

or

s(t ) = '¦g(t ) exp(j 2πf

c

t )¦

where f

c

is the carrier frequency.

The function g(t ) is referred to as the envelope of the

modulated signal. The desired type of modulating signal is

obtained by selecting the appropriate modulation function

g[m(t )] where m(t ) is the analog or digital baseband

message signal.

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Double Sideband (DSB) II

Slide 125

We always assume that m(t ) is real but the function g(t ) is

complex or real. Either the frequency, phase or amplitude

of the carrier is varied in proportion to the baseband signal

m(t ).

Modulation is a process that causes a shift in the range of

frequencies in a signal.

The term baseband is used to designate the band of

frequencies of the signal delivered by the source or the

input transducer.

In telephony, the baseband is 0 to 3.5 kHz.

In television, the baseband is 0 to 4.3 MHz.

Baseband communication does not use modulation.

Example: Long-distance PCM over optical ﬁbers.

Carrier Communication uses modulation.

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Double Sideband (DSB) III

Slide 126

Example: Long-haul communication over a radio link.

One of the basic parameters (amplitude, frequency, or

phase) of a sinusoidal carrier of high frequency ω

c

is varied

in proportion to m(t ). This results in AM, FM, or PM,

respectively.

FM and PM belong to the class of modulation known as

angle modulation.

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Double Sideband (DSB) IV

Slide 127

Figure: Transmitter.

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Double Sideband (DSB) V

Slide 128

Figure: receiver.

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Double Sideband (DSB) VI

Slide 129

The carrier amplitude A is made directly proportional to the

modulating signal m(t ). The DSB signal is

m(t ) cos ω

c

t ↔

1

2

[M(ω +ω

c

) + M(ω −ω

c

)]

That is, the DSB modulation translates or shifts the

frequency spectrum to the left and the right by ω

c

.

If the bandwidth of m(t ) is B Hz, then the bandwidth of the

modulated signal is 2B Hz.

Lower sideband (LSB) is a portion of DSB spectrum which

lies below ω

c

.

Upper sideband (USB) is a portion of DSB spectrum which

lies above ω

c

.

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Double Sideband (DSB) VII

Slide 130

DSB signal does not contain a discrete component of the

carrier frequency ω

c

. For this reason DSB is called

DSB-SC.

Demodulation, or detection, is the process of recovering

the signal from the modulated signal.

in Synchronous or coherent detection the local oscillator

has the same frequency and phase as the carrier used for

modulation.

Modulation can be obtained by multiplier modulators,

nonlinear modulators, and switching modulators.

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Double Sideband (DSB) VIII

Slide 131

1

Multiplier Modulator: Here modulation is achieved directly

by multiplying m(t ) by cos ω

c

t using an analog multiplier

whose output is proportional to the product of two input

signal. Another way to multiply two signals is through

logarithmic ampliﬁers. Here the basic components are a

logarithmic and an antilogarithmic ampliﬁer with outputs

proportional to the log and antilog of their inputs,

respectively. Using two logarithmic ampliﬁers, we generate

and add the logarithms of the two signals to be multiplied.

The sum is then applied to an antilogarithmic ampliﬁer to

obtain the desired product. That is,

AB = antilog[log(AB)]

= antilog[logA + logB]

It is rather difﬁcult to maintain linearity in this kind of

ampliﬁer, and they tend to be rather expensive.

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Double Sideband (DSB) IX

Slide 132

2

Nonlinear Modulator: Modulation can be achieved by using

nonlinear devices, such as a semiconductor diode or a

transistor.

3

Switching Modulator: The multiplication operation required

for modulation can be replaced by a simpler switching

operation in which a modulated signal can be obtained by

multiplying m(t ) not only by a pure sinusoidal but by any

periodic signal w(t ) of the fundamental radian frequency

ω

c

. For example with square pulse train

w(t ) =

1

2

+

2

π

cos ω

c

t −

1

3

cos 3ω

c

t +

1

5

cos 5ω

c

t −

**or with a square waveform
**

w(t ) =

4

π

cos ω

c

t −

1

3

cos 3ω

c

t +

1

5

cos 5ω

c

t −

**Signals and Systems www.danesfahani.com
**

Double Sideband (DSB) X

Slide 133

Frequency mixing or frequency conversion is used to

change the carrier frequency of a modulated signal

m(t ) cos ω

c

t from ω

c

to some other frequency ω

I

. This can

be done by multiplying m(t ) cos ω

c

t by 2 cos ω

mix

t , where

ω

mix

= ω

c

+ω

I

for up-conversion or ω

mix

= ω

c

−ω

I

for

down-conversion and then bandpass-ﬁltering the product.

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Double Sideband (DSB) XI

Slide 134

Figure: Block diagram of product modulator.

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Double Sideband (DSB) XII

Slide 135

Figure: DSB waveforms.

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Double Sideband (DSB) XIII

Slide 136

Figure: Spectrum of baseband signal.

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Double Sideband (DSB) XIV

Slide 137

Figure: Spectrum of DSB-SC modulated wave.

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Double Sideband (DSB) XV

Slide 138

Figure: Coherent detector for demodulating DSB-SC modulated

wave.

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Double Sideband (DSB) XVI

Slide 139

Figure: Spectrum of a product modulator output with a DSB-SC

modulated wave as input.

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Double Sideband (DSB) XVII

Slide 140

Problem

Consider a multiplex system in which four input signals m

1

(t ),

m

2

(t ), m

3

(t ), and m

4

(t ) are respectively multiplied by the

carrier waves

[cos(2πf

a

t ) + cos(2πf

b

t )]

[cos(2πf

a

t +α

1

) + cos(2πf

b

t +β

1

)]

[cos(2πf

a

t +α

2

) + cos(2πf

b

t +β

2

)]

[cos(2πf

a

t +α

3

) + cos(2πf

b

t +β

3

)]

and the resulting DSB-SC signals are summed and then

transmitted over a common channel. In the receiver,

demodulation is achieved by multiplying the sum of the

DSB-SC signals by the four carrier waves separately and then

using ﬁltering to remove the unwanted components.

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Double Sideband (DSB) XVIII

Slide 141

(a) Determine the conditions that the phase angles α

1

, α

2

, α

3

and β

1

, β

2

, β

3

must satisfy in order that the output of the

kth demodulator is m

k

(t ), where k = 1, 2, 3, 4.

(b) Determine the minimum separation of carrier frequencies

f

a

and f

b

in relation to the bandwidth of the input signals so

as to ensure a satisfactory operation of the system.

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Part XII

Amplitude (Linear) Modulation

Ordinary Amplitude Modulation

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Amplitude Modulation (AM) I

Slide 143

AM = DSB-SC + Carrier.

Time domain expression

ϕ

AM

(t ) = Acos ω

c

t + m(t ) cos ω

c

t

= [A + m(t )] cos ω

c

t

Spectrum

ϕ

AM

(t ) ↔

1

2

[M(ω+ω

c

)+M(ω−ω

c

)]+πA[δ(ω+ω

c

)+δ(ω−ω

c

)]

That is, the AM spectrum is simply a translated version of

the modulated signal (both positive and negative with

power equally distributed between the two) plus delta

functions at the carrier line spectral component.

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Amplitude Modulation (AM) II

Slide 144

Larger power at the transmitter, which makes it rather

expensive.

Less expensive receivers.

The envelope of AM has the information about the

message m(t ) only if the AM signal [A + m(t )] cos ω

c

t

satisﬁes the condition A + m(t ) ≥ 0 for all t . This means

that

A ≥ m

p

where m

p

is the peak amplitude (positive or negative) of

m(t ).

Modulation index

µ =

m

p

A

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Amplitude Modulation (AM) III

Slide 145

Because A ≥ m

p

and because there is no upper bound on

A, it follows that

0 ≤ 1

as the required condition for the viability of demodulation of

AM by an envelope detector.

Overmodulation results if µ > 1 or A < m

p

.

Deﬁnition

Power efﬁciency

η =

useful power

total power

=

P

s

P

c

+ P

s

where P

c

the carrier power, and P

s

is the power in sidebands.

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Amplitude Modulation (AM) IV

Slide 146

Example

For tone modulation m(t ) = µAcos ω

m

t , P

s

=

(µA)

2

2

, and

P

c

=

A

2

2

. Hence

η =

µ

2

2 +µ

2

100%

With maximum modulation index, that is µ = 1, η = 33.3%,

which means that only 33.3% of the power is used to transmit

the sidebands.

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Amplitude Modulation (AM) V

Slide 147

Deﬁnition

The percentage of positive modulation on an AM signal is

(A

max

−A

c

)/(A

c

) 100 = max[µm(t )] 100

Deﬁnition

The percentage of negative modulation is

(A

c

−A

min

)/(A

c

) 100 = min[µm(t )] 100

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Amplitude Modulation (AM) VI

Slide 148

Deﬁnition

The overall modulation percentage is

(A

max

−A

min

)/(2A

c

) 100

or

(max[µm(t )] −min[µm(t )])/(2) 100

So, the positive modulation refers to the maximum amount

the amplitude of the modulated carrier is increased over

the maximum amplitude of the unmodulated carrier.

And, the negative modulation refers to the maximum

amount the amplitude of the modulated carrier is

decreased in comparison with the unmodulated carrier.

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Amplitude Modulation (AM) VII

Slide 149

Note that for any message signal which is symmetric (at

least in terms of maximum and minimum values) about the

x axis, the percentage of positive and negative modulation

(and therefore the overall modulation) will always be equal.

The percentage of modulation can be greater than 100%,

in which case A

min

has a negative value

Generation of AM signal

v

bb

(t ) = [c cos ω

c

t + m(t )]w(t )

where

w(t ) =

¸

1

2

+

2

π

cos ω

c

t −

1

3

cos 3ω

c

t +

1

5

cos 5ω

c

t −

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**

Amplitude Modulation (AM) VIII

Slide 150

Hence

v

bb

(t ) =

c

2

cos ω

c

t +

2

π

m(t ) cos ω

c

. .. .

AM

+ other terms

. .. .

suppressed by bandpass ﬁlter

Demodulation of AM signals

1

Rectiﬁer Detector:

v

R

= ¦[A + m(t )] cos ω

c

t ¦w(t )

where

w(t ) =

¸

1

2

+

2

π

cos ω

c

t −

1

3

cos 3ω

c

t +

1

5

cos 5ω

c

t −

Hence

v

R

=

1

π

[A + m(t )] + other terms of higher frequencies

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Amplitude Modulation (AM) IX

Slide 151

2

Envelope Detector: In an envelope detector, the output of

the detector follows the envelope of the modulated signal.

The time constant RC must be large compared to 1/ω

c

but

should be small compared to 1/2πB, where ω

c

and B and

radian frequency of carrier and bandwidth of the message

to be recovered.

Compared with an AM signal, a DSBSC signal has inﬁnite

percentage modulation because there is no carrier line

component.

The modulation efﬁciency of a DSBSC signal is 100%

since no power is wasted in a discrete carrier.

However, a product detector is always required for a

DSBSC signal (more expensive than an envelope

detector).

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Amplitude Modulation (AM) X

Slide 152

Figure: AM waveforms (a) Message.

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Amplitude Modulation (AM) XI

Slide 153

Figure: AM waveforms (b) AM wave with µ < 1.

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Amplitude Modulation (AM) XII

Slide 154

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Amplitude Modulation (AM) XIII

Slide 155

Figure: AM waveforms (c) AM wave with µ > 1.

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Amplitude Modulation (AM) XIV

Slide 156

Figure: Spectrum of baseband signal.

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Amplitude Modulation (AM) XV

Slide 157

Figure: Spectrum of AM wave.

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Amplitude Modulation (AM) XVI

Slide 158

AKA quadrature multiplexing

The DSB signals occupy twice the bandwidth required for

the baseband. This disadvantage can be overcome by

transmitting two DSB signals using carriers of the same

frequency but in phase quadrature.

Modulation

ϕ

QAM

(t ) = m

1

(t ) cos ω

c

t + m

2

(t ) sinω

c

t

demodulation

2ϕ

QAM

(t ) cos ω

c

t = 2[m

1

(t ) cos ω

c

t + m

2

(t ) sinω

c

t ] cos ω

c

t

= m

1

(t ) + m

1

(t ) cos 2ω

c

t + m

2

(t ) sin 2ω

c

t

. .. .

suppressed by lowpass ﬁltering

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Amplitude Modulation (AM) XVII

Slide 159

Phase error results in crosstalk

2ϕ

QAM

(t ) cos(ω

c

t +θ)

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Amplitude Modulation (AM) XVIII

Slide 160

Figure: Transmitter.

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Amplitude Modulation (AM) XIX

Slide 161

Figure: Receiver.

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Part XIII

Amplitude (Linear) Modulation

Single Sideband Modulation

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Single Sideband (SSB) I

Slide 163

The DSB spectrum has two sidebands; the upper

sideband (USB) and the lower sideband (LSB), both

containing the complete information of the baseband

signal. A scheme in which only one sideband is transmitted

is known as sideband (SSB) transmission, which requires

only one-half the bandwidth of the DSB signal.

Time-domain representation

M(ω) = M

+

(ω) + M

−

(ω)

m(t ) = m

+

(t ) + m

−

(t )

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Single Sideband (SSB) II

Slide 164

[M

+

(ω)[ and [M

−

(ω)[ are not even functions of ω.

Therefore, m

+

(t ) and m

−

(t ) cannot be real; they are

complex.

m

+

(t ) =

1

2

[m(t ) + jm

h

(t )]

and

m

−

(t ) =

1

2

[m(t ) −jm

h

(t )]

To determine m

h

(t ), we note that

M

+

(ω) = M(ω)u(ω)

=

1

2

M(ω)[1 + sgn(ω)]

=

1

2

M(ω) +

1

2

M(ω)sgn(ω)

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Single Sideband (SSB) III

Slide 165

It can be seen that

jm

h

(t ) ↔M(ω)sgn(ω)

Hence

M

h

(ω) = −jM(ω)sgn(ω)

From the table of Fourier transform pairs

1/πt ↔−j sgn(ω)

Therefore

m

h

(t ) = m(t ) ∗ 1/πt

or

m

h

(t ) =

1

π

ˆ

∞

−∞

m(α)

t −α

dα

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Single Sideband (SSB) IV

Slide 166

which is known as the Hilbert transform of m(t ). The

Hilbert transformer has the following frequency response

H(ω) = −j sgn(ω)

It follows that [H(ω)[ = 1 and θ

h

(ω) = −π/2 for ω > 0 and

π/2 for ω < 0. Thus, if we delay the phase of every

component of m(t ) by π/2 (without changing its amplitude),

the resulting signal is m

h

(t ), the Hilbert transform of m(t ).

Example

For the simple case of a tone modulation, that is, when the

modulating signal is a sinusoid m(t ) = cos ω

m

t , the Hilbert

transform is

m

h

(t ) = cos

ω

m

t −

π

2

= sinω

m

t

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Single Sideband (SSB) V

Slide 167

USB Spectrum

Φ

USB

(ω) = M

+

(ω −ω

c

) + M

−

(ω +ω

c

)

USB signal

φ

USB

(t ) = m

+

(t )e

j ω

c

t

+ m

−

(t )e

−j ω

c

t

Substituting for m

+

(t ) and m

−

(t )

φ

SSB

(t ) = m(t ) cos ω

c

t ∓m

h

(t ) sinω

c

t

where the minus sign applies to USB and the plus signa

applies to LSB.

Generation of SSB signals

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Single Sideband (SSB) VI

Slide 168

1

Selective-Filtering Method: A DSB-SC signal is passed

through a bandpass ﬁlter to eliminate the undesired

sideband. This method is used in speech processing.

2

Phase-Shift Method:

φ

SSB

(t ) = m(t ) cos ω

c

t ∓m

h

(t ) sinω

c

t

Demodulation of SSB signals

1

Synchronous Method: φ

SSB

(t ) is multiplied by cos ω

c

t and

is then passed through a low-pass ﬁlter.

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Single Sideband (SSB) VII

Slide 169

2

Envelope Detection Method:

φ

SSB+C

= Acos ω

c

t + [m(t ) cos ω

c

t + m

h

(t ) sinω

c

t ]

where the envelope E(t ) is given by

E(t ) = ¦[A + m(t )]

2

+ m

2

h

(t )]¦

1/2

= A

¸

1 +

2m(t )

A

+

m

2

(t )

A

2

+

m

2

h

(t )

A

2

1/2

With A [m(t )[ and A [m

h

(t )[

E(t ) · A

¸

1 +

2m(t )

A

1/2

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Single Sideband (SSB) VIII

Slide 170

Using binomial expansion and discarding higher order

terms

E(t ) · A

¸

1 +

m(t )

A

= A + m(t )

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Single Sideband (SSB) IX

Slide 171

Figure: (a) Spectrum of a message signal m(t) with an energy gap of

width 2fa centered on the origin.

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Single Sideband (SSB) X

Slide 172

Figure: (b) Spectrum of corresponding SSB signal containing the

upper sideband.

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Single Sideband (SSB) XI

Slide 173

Problem

Consider the modulated waveform

s(t ) = A

c

cos(2πf

c

t ) + m(t ) cos(2πf

c

t ) −

ˆ

m(t ) sin(2πf

c

t )

which represents a carrier plus an SSB signal, with m(t )

denoting the message and

ˆ

m(t ) its Hilbert transform.

Determine the conditions for which an ideal envelope detector,

with s(t ) as input, would produce a good approximation to the

message signal m(t ).

CCITT standard

group: 12 channels

supergroup: 5 groups

mastergroup: 10 supergroups

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Single Sideband (SSB) XII

Slide 174

Figure: Modulation steps in an FDM system.

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Part XIV

Amplitude (Linear) Modulation

Vestigial Sideband Modulation

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Vestigial Sideband (VSB) I

Slide 176

AKA asymmetric sideband system is a compromise

between DSB and SSB.

VSB inherits the advantages of DSB and SSB but avoids

their disadvantages at a small cost.

VSB signals are relatively easy to generate, and, at the

same time, their bandwidth is only (typically 25%) greater

than that of SSB signal.

In VBS, instead of rejecting one sideband completely (as in

SSB), a gradual cutoff of one sideband, is accepted.

Demodulation Methods:

1

Synchronous detection.

2

Envelope (or rectiﬁer) detector.

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Vestigial Sideband (VSB) II

Slide 177

If the vestigial shaping ﬁlter that produces VSB from DSB

is H

i

(ω), then the resulting VSB signal spectrum is

Φ

VSB

(ω) = [M(ω +ω

c

) + M(ω −ω

c

)]H

i

(ω)

This shaping ﬁlter allows the transmission of one sideband,

but suppresses the other sideband, not completely, but

gradually.

The product e(t ) of multiplying the incoming VSB signal

φ

VSB

(t ) by 2 cos ω

c

t is

e(t ) = 2Φ

VSB

(t ) cos ω

c

t ↔[Φ

VSB

(ω +ω

c

) + Φ

VSB

(ω −ω

c

)]

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Vestigial Sideband (VSB) III

Slide 178

The signal e(t ) is further passed through a low-pass

equalizer ﬁlter of transfer function H

0

(ω) whose output is

required to be m(t ). Hence,

M(ω) = [Φ

VSB

(ω +ω

c

) + Φ

VSB

(ω −ω

c

)]

By substitution of Φ

VSB

(ω) in this equation and eliminating

the spectra ±2ω

c

by a low-pass ﬁlter H

0

(ω), we obtain

H

0

(ω) =

1

H

i

(ω +ω

c

) + H

i

(ω −ω

c

)

[ω[ ≤ 2πB

Use of VSB in broadcast television: The baseband video

signal of television occupies an enormous bandwidth of

4.5 MHz.

Signals and Systems www.danesfahani.com

Vestigial Sideband (VSB) IV

Slide 179

The vestigial shaping ﬁlter H

i

(ω) cuts off the lower

sideband of the DSB spectrum of a television signal

gradually at 0.75 MHz to 1.25 MHz below the carrier

frequency f

c

. The resulting VSB spectrum bandwidth is

6 MHz.

Linearity of amplitude modulation: Principle of

superposition applies in all amplitude modulation

schemes (AM, DSB, SSB, VSB).

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Vestigial Sideband (VSB) V

Slide 180

Figure: Filtering scheme for the generation of VSB modulated wave.

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Vestigial Sideband (VSB) VI

Slide 181

Figure: Magnitude response of VSB ﬁlter; only the positive-frequency

portion is shown.

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Vestigial Sideband (VSB) VII

Slide 182

Problem

The single tone modulating signal m(t ) = A

m

cos(2πf

m

t ) is

used to generate the VSB signal

s(t ) =

1

2

aA

m

A

c

cos[2π(f

c

+f

m

)t ]+

1

2

A

m

A

c

(1−a) cos[2π(f

c

−f

m

)t ]

where a is a constant, less than unity, representating the

attenuation of the upper side frequency.

(a) Find the quadrature component of the VSB signal s(t ).

(b) The VSB signal, plus the carrier A

c

cos(2πf

c

t ), is passed

through an envelope detector. Determine the distortion

produced by the quadrature component.

(c) What is the value of constant a for which this distortion

reaches its worst possible condition?

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Part XV

Angle (Exponential) Modulation

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Angle (Exponential) Modulation I

Slide 184

generalized sinusoidal signal φ(t )

φ(t ) = Acos θ(t )

where θ(t ) is the generalized angle and is a function of t .

φ(t ) = Acos(ω

c

t +θ

0

)

instantaneous frequency and phase

ω

i

(t ) =

dθ

dt

θ(t ) =

ˆ

t

−∞

ω

i

(α) dα

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Angle (Exponential) Modulation II

Slide 185

Example

The instantaneous frequency in hertz of

cos 200πt cos(5 sin2πt ) + sin 200πt sin(5 sin2πt ) is found by

noticing that the phase is θ(t ) = 200πt −5 sin2πt . Hence,

ω

i

= 200π −10π cos 2πt .

angle modulation possibilities: phase modulation (PM) and

frequency modulation (FM).

In PM, the angle θ(t ) is varied linearly with m(t ):

θ(t ) = ω

c

t +θ

0

+ k

p

m(t )

where k

p

is a constant and ω

c

is the carrier frequency.

Assuming θ

0

= 0, without loss of generality

θ(t ) = ω

c

t + k

p

m(t )

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Angle (Exponential) Modulation III

Slide 186

PM signal

φ

PM

(t ) = Acos[ω

c

t + k

p

m(t )]

Instantaneous frequency ω

i

(t ) in PM

ω

i

(t ) =

dθ

dt

= ω

c

+ k

p

˙

m(t )

Hence, in PM, the instantaneous frequency ω

i

varies

linearly with the derivative of the modulating signal.

In FM the instantaneous frequency ω

i

is varied linearly with

the modulating signal.

ω

i

(t ) = ω

c

+ k

f

m(t )

where k

f

is a constant.

Signals and Systems www.danesfahani.com

Angle (Exponential) Modulation IV

Slide 187

In FM

θ(t ) =

ˆ

t

−∞

[ω

c

+ k

f

m(α)] dα

= ω

c

t + k

f

ˆ

t

−∞

m(α) dα

FM signal

φ

FM

(t ) = Acos

¸

ω

c

t + k

f

ˆ

t

−∞

m(α) dα

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**

Angle (Exponential) Modulation V

Slide 188

The generalized angle-modulated carrier φ

EM

can be

expressed as

φ

EM

(t ) = Acos[ω

c

t +ψ(t )]

= Acos

¸

ω

c

t +

ˆ

t

−∞

m(α)h(t −α) dα

FM results if h(t ) = k

p

δ(t ), and PM results if h(t ) = k

f

u(t )

Power of an angle-modulated wave is A

2

/2.

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Angle (Exponential) Modulation VI

Slide 189

Figure: Illustrative AM, FM, and PM waveforms.

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Angle (Exponential) Modulation VII

Slide 190

Narrow-band FM

φ

FM

(t ) · A[cos ω

c

t −k

f

a(t ) sinω

c

t ]

Narrow-band PM

φ

PM

(t ) · A[cos ω

c

t −k

p

m(t ) sinω

c

t ]

Carson’s rule for FM

B

FM

= 2B(β + 1)

where B is the message BW, and the deviation ratio β is

β =

∆f

B

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Angle (Exponential) Modulation VIII

Slide 191

where the frequency deviation ∆f is

∆f =

k

f

m

p

2π

where k

f

and m

p

are the FM modulator sensitivity and the

peak of the message, respectively.

Carson’s rule for PM

B

PM

= 2(∆f + B)

where the frequency deviation ∆f is

∆f =

k

p

m

p

2π

where k

p

and m

p

are the PM modulator sensitivity and the

peak of the derivative of the message, respectively.

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Angle (Exponential) Modulation IX

Slide 192

Example

For

x

c

(t ) = 10 cos[(10

8

)πt + 5 sin2π(10

3

)t ]

the phase θ(t ) = ω

c

t +φ(t ). Hence, φ(t ) = 5 sin2π(10

3

)t and

φ

(t ) = 5(2π)(10

3

) cos 2π(10

3

)t . Therefore, [φ(t )[

max

= 5 rad,

and ∆ω = [φ

(t )[

max

= 5(2π)(10

3

) rad/s or ∆f = 5 kHz.

Example

Given the angle-modulated signal

x

c

(t ) = 10 cos(2π10

8

t + 200 cos 2π10

3

t ), the instantaneous

frequency is ω

i

= 2π(10

8

) −4π(10

5

) sin 2π(10

3

)t . So

∆ω = 4π(10

5

), ω

m

= 2π(10

3

), and β =

∆ω

ω

m

= 200. The BW is

W

B

= 2(β + 1)ω

m

= 8.04π(10

5

) rad/s

Signals and Systems www.danesfahani.com

Angle (Exponential) Modulation X

Slide 193

Immunity of angle modulation to nonlinearities. For

example, consider a second-order nonlinear device whose

input x(t ) and output y(t ) are related by

y(t )a

1

x(t ) + a

2

x

2

(t )

If

x(t ) = cos[ω

c

t +ψ(t )]

then

y(t ) =

a

2

2

+ a

1

cos[ω

c

t +ψ(t )] +

a

2

2

cos[2ω

c

t + 2ψ(t )]

Signals and Systems www.danesfahani.com

Angle (Exponential) Modulation XI

Slide 194

A similar nonlinearity in AM not only causes unwanted

modulation with carrier frequencies nω

c

but also causes

distortion of the desired signal. For instance, if a DSB-SC

signal m(t ) cos ω

c

t passes through a nonlinearity

y(t ) = ax(t ) + bx

3

(t ), the output is

y(t ) =

¸

am(t ) +

3b

4

m

3

(t )

cos ω

c

t +

b

4

m

3

(t ) cos 3ω

c

t

Passing this signal through a bandpass ﬁlter yields

[am(t ) + (3b/4)m

3

(t )] cos ω

c

t . The distortion component

(3b/4)m

3

(t ) is present along with the desired signal am(t ).

In telephone systems, several channels are multiplexed

using SSB signals. The multiplexed signal is frequency

modulated and transmitted over a microwave radio relay

system with many links in tandem.

Signals and Systems www.danesfahani.com

Angle (Exponential) Modulation XII

Slide 195

Generation of FM waves by indirect method of Armstrong:

In this method the NBFM is converted to WBFM by using

frequency multipliers. Thus, if we want a 12-fold increase

in the frequency deviation, we can use a 12th-order

nonlinear device or two second-order and one third-order

device in cascade.

Demodulation of FM: If we apply ϕ

FM

(t ) to an ideal

differentiator, the output is

˙ ϕ

FM

(t ) = A[ω

c

+ k

f

m(t )] sin

¸

ω

c

t + k

f

ˆ

t

−∞

m(α)d(α)

The signal ˙ ϕ

FM

(t ) is both amplitude and frequency

modulated, the envelope being A[ω

c

+ k

f

m(t )]. The

message can be obtained by envelope detection of ˙ ϕ

FM

(t ).

Signals and Systems www.danesfahani.com

Angle (Exponential) Modulation XIII

Slide 196

Figure: (a) Scheme for generating an FM wave by using a phase

modulator.

Signals and Systems www.danesfahani.com

Angle (Exponential) Modulation XIV

Slide 197

Figure: (b) Scheme for generating a PM wave by using a frequency

modulator.

Signals and Systems www.danesfahani.com

Angle (Exponential) Modulation XV

Slide 198

Figure: Narrowband FM.

Signals and Systems www.danesfahani.com

Angle (Exponential) Modulation XVI

Slide 199

Figure: Plots of Bessel functions of the ﬁrst kind for varying order.

Signals and Systems www.danesfahani.com

Angle (Exponential) Modulation XVII

Slide 200

Figure: Discrete amplitude spectra of an FM signal, normalized with

respect to the carrier amplitude, for the case of sinusoidal modulation

of ﬁxed frequency and varying amplitude. Only the spectra for

positive frequencies are shown.

Signals and Systems www.danesfahani.com

Angle (Exponential) Modulation XVIII

Slide 201

Figure: Discrete amplitude spectra of an FM signal, normalized with

respect to the carrier amplitude, for the case of sinusoidal modulation

of ﬁxed frequency and varying amplitude. Only the spectra for

positive frequencies are shown.

Signals and Systems www.danesfahani.com

Angle (Exponential) Modulation XIX

Slide 202

Figure: Discrete amplitude spectra of an FM signal, normalized with

respect to the carrier amplitude, for the case of sinusoidal modulation

of ﬁxed frequency and varying amplitude. Only the spectra for

positive frequencies are shown.

Signals and Systems www.danesfahani.com

Angle (Exponential) Modulation XX

Slide 203

Figure: Discrete amplitude spectra of an FM signal, normalized with

respect to the carrier amplitude, for the case of sinusoidal modulation

of varying frequency and ﬁxed amplitude. Only the spectra for

positive frequencies are shown.

Signals and Systems www.danesfahani.com

Angle (Exponential) Modulation XXI

Slide 204

Figure: Discrete amplitude spectra of an FM signal, normalized with

respect to the carrier amplitude, for the case of sinusoidal modulation

of varying frequency and ﬁxed amplitude. Only the spectra for

positive frequencies are shown.

Signals and Systems www.danesfahani.com

Angle (Exponential) Modulation XXII

Slide 205

Figure: Discrete amplitude spectra of an FM signal, normalized with

respect to the carrier amplitude, for the case of sinusoidal modulation

of varying frequency and ﬁxed amplitude. Only the spectra for

positive frequencies are shown.

Signals and Systems www.danesfahani.com

Angle (Exponential) Modulation XXIII

Slide 206

Figure: Block diagram of the indirect method of generating a

wideband FM signal.

Signals and Systems www.danesfahani.com

Angle (Exponential) Modulation XXIV

Slide 207

Figure: Block diagram of frequency multiplier.

Signals and Systems www.danesfahani.com

Angle (Exponential) Modulation XXV

Slide 208

Problem

Consider a narrow-band FM signal approximately deﬁned by

s(t ) · A

c

cos(2πf

c

t ) −βA

c

sin(2πf

c

t ) sin(2πf

m

t )

(a) Determine the envelope of this modulated signal. What is

the ratio of the maximum to the minimum value of this

envelope? Plot this ratio versus β, assuming that β is

restricted to the interval 0 ≤ β ≤ 0.3.

(b) Determine the average power of the narrow-band FM

signal, expressed as a percentage of the average power of

the unmodulated carrier wave. Plot this result versus β,

assuming that β is restricted to the interval 0 ≤ β ≤ 0.3.

Signals and Systems www.danesfahani.com

Angle (Exponential) Modulation XXVI

Slide 209

(c) By expanding the angle θ

i

(t ) of the narrow-band FM signal

s(t ) in the form of a power series, and restricting the

modulation index β to a maximum value of 0.3 radians,

show that

θ

i

(t ) · 2πf

c

t +β sin(2πf

m

t ) −

β

3

3

sin

3

(2πf

m

t )

What is the value of the harmonic distortion for β = 0.3?

Problem

The sinusoidal modulating wave

m(t ) = A

m

cos(2πf

m

t )

is applied to a phase modulator with phase sensitivity k

p

. The

unmodulated carrier wave have frequency f

c

and amplitude A

c

.

Signals and Systems www.danesfahani.com

Angle (Exponential) Modulation XXVII

Slide 210

(a) Determine the spectrum of the resulting phase-modulated

signal, assuming that the maximum phase deviation

β

p

= k

p

A

m

does not exceed 0.3 radians.

(b) Not included.

Problem

Suppose that the phase-modulated signal of the

previous problem

has an arbitrary value for the maximum phase deviation β

p

.

This modulated signal is applied to an ideal band-pass ﬁlter

with mid-band frequency f

c

and a passband externding from

f

c

−1.5f

m

to f

c

+ 1.5f

m

. Determine the envelope, phase, and

instantaneous frequency of the modulated signal at the ﬁlter

output as functions of time.

Signals and Systems www.danesfahani.com

Angle (Exponential) Modulation XXVIII

Slide 211

Problem

An FM signal with modulation index β = 1 is transmitted

through and ideal band-pass ﬁlter with mid-band frequency f

c

and bandwidth 5f

m

, where f

c

is the carrier frequency and f

m

is

the frequency of the sinusoidal modulating wave. Determine

the amplitude spectrum of the ﬁlter output.

Problem

Consider a wide-band PM signal produced by a sinusoidal

modulating wave A

m

cos(2πf

m

t ), using a modulator with a

phase sensitivity equal to k

p

radian per volt.

(a) Show that if the maximum phase deviation of the PM signal

is large compared with one radian, the bandwidth of the

PM signal varies linearly with the modulation frequency f

m

.

Signals and Systems www.danesfahani.com

Angle (Exponential) Modulation XXIX

Slide 212

(b) Compare this characteristic of a wide-band PM signal with

that of wide-band FM signal.

Problem

An FM signal with a frequency deviation of 10 kHz at a

modulation frequency of 5 kHz is applied to two frequency

multipliers connected in cascade. The ﬁrst multiplier doubles

the frequency and the second multiplier triples the frequency.

Determine the frequency deviation and the modulation index of

the FM signal obtained at the second multiplier output. What is

the frequency separation of the adjacent side frequencies of

this FM signal?

Signals and Systems www.danesfahani.com

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Signals and Systems by Dr Reza Danesfahani, Faculty of Engineering University of Kashan Kashan, Iran

A property of MVG_OMALLOOR

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Signals and Systems by Dr Reza Danesfahani, Faculty of Engineering University of Kashan Kashan, Iran

SS

Signals and Systems by Dr Reza Danesfahani, Faculty of Engineering University of Kashan Kashan, Iran

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