SS Signals and Systems PRESENTATION

Signals and Systems
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
Preamble I
Slide 3
Syllabus
1
Continuous-time and discrete-time signals: classication
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: denition, properties,
frequency response of LTI Introduction to ltering.
5
Sampling: impulse train sampling, sampling theorem,
reconstruction of signals, effects of under-sampling.
6
Z-transforms: denitions, properties, analysis of LTI
systems, transfer functions.
7
Communication systems: Amplitude modulation (AM),
demodulation
Grading
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.
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.
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,
E.W. Kamen and B.S. Heck,
Fundamentals of Signals and Systems using MATLAB,
L.B. Jackson,
Signals, Systems, and Transforms,
Addison-Wesley, 1991.
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.
Part II
Signals and Systems
Signals and Systems I
Slide 9
Denition
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 owing 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 II
Slide 10
Denition
An energy signal is a signal whose total energy is nite, 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 nite-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 III
Slide 11
Denition
A power signal is a signal whose average power is nite, i.e.
those signals for which P
< . Such a signal must have

innite total energy.
Example
The constant signal x[n] = 4 is a power signal. It has innite
energy, but average power P
= 16.
There are signals for which neither P
nor E
are nite. 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 IV
Slide 12
2
Time reversal: The time reversed version of the signal x(t )
is obtained by a reection 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(t +), where and are given
numbers. A systematic approach to doing this is to rst delay or
advance x(t ) in accordance with the value of , and then to
perform time scaling and/or time time reversal on the resulting
signal in accordance with the value of . The delayed or
advanced signal is linearly stretched if [[ < 1, linearly
compressed if [[ > 1, and reversed in time if < 0.
Signals and Systems V
Slide 13
Denition
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 nite
number of distinct values, then we call this signal a digital
signal.
Signals and Systems VI
Slide 14
Denition
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.
Denition
Deterministic signals are those signals whose values are
completely specied 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 VII
Slide 15
Denition
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 VIII
Slide 16
The even part of the signal x(t ) is
Ex(t ) =
1
2
[x(t ) + x(t )]
The odd part of the signal x(t ) is
Ox(t ) =
1
2
[x(t ) x(t )]
Similarly,
Ex[n] =
1
2
[x[n] + x[n]]
The odd part of the signal x[n] is
Ox[n] =
1
2
[x[n] x[n]]
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 X
Slide 18
Denition
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 XI
Slide 19
Denition
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.
Denition
The unit step function is dened as
u(t )
1 t > 0
0 t < 0
Signals and Systems XII
Slide 20
Denition
Dirac delta function, often referred to as the unit impulse(or
delta) function, is a signal that has innite height and
innitesimal width. The Dirac delta function (x) is dened by
w(x)(x) dx = w(0)
where w(x) is any function that is continuous at x = 0.
Properties of impulse function properties
1
A(t ) = A(t )
2
A(t ) = A
Signals and Systems XIII
Slide 21
3
A(t ) = 0 for t = 0
4
A(t t
0
) + B(t t
0
) = (A + B)(t t
0
)
5
[y(t )][A(t t
0
)]Ay(t
0
)(t t
0
)
Denition
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 XIV
Slide 22
4
Feedback interconnection
Denition
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.
Denition
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 XV
Slide 23
Example
The system specied 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 XVI
Slide 24
Denition
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 rst 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 XVII
Slide 25
Denition
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.
Denition
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 XVIII
Slide 26
Example
Consider the continuous-time system dened 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 XIX
Slide 27
Denition
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 dening 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 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.
Part III
Linear Time-Invariant Systems
Linear Time-Invariant Systems I
Slide 30
Denition
If a system is linear and also time-invariant, then it is called a
linear time-invariant (LTI) system.
Denition
A system is BIBO stable if and only if every bounded input
results in a bounded output. For a xed, linear, system a
necessary and sufcient condition for BIBO stability is
[h(t )[ dt <
where h(t ) is the impulse response of the system.
Linear Time-Invariant Systems II
Slide 31
Denition
The impulse response h(t ) of a continuous-time LTI system is
dened to be the response of the system when the input is (t ).
Denition
The convolution integral or superposition integral of two
continuous-time signals x(t ) and h(t ) is
y(t ) = x(t ) h(t ) =
x()h(t ) d
The properties of convolution are:
1
the commutative property. This property is expressed by
x(t ) h(t ) = h(t ) x(t )
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
)
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
.
Denition
The step response s(t ) of a continuous-time LTI system is
dened 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.
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,
[h()[ d <
Denition
The impulse response h[n] of a discrete-time LTI system is
dened to be the response of the system when the input is [n].
Linear Time-Invariant Systems VI
Slide 35
Denition
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]
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 xed at some value.
3
The product x[k]h[n k] is summed over all k to produce a
single output sample y[n].
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][ <
Part IV
Fourier Series
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 2nf
0
t
n = 0, 1, 2,
The series coefcients are related to v(t ) by
c
n
=
1
T
0
T
0
v(t )e
j 2nf
0
t
dt
Since the coefcients are complex quantities in general,
they can be expressed in the polar form
c
n
= [c
n
[e
j arg c
n
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
jk
0
t
0
3
Frequency shifting
x(t )e
jM
0
t
a
kM
4
Conjugation
x
(t ) a
k
Fourier Series III
Slide 41
5
Time reversal
x(t ) a
k
6
Time scaling
x(t ), > 0
k
7
Periodic convolution
T
x()y(t ) d Ta
k
b
k
8
Multiplication
x(t )y(t )
+
l =
a
l
b
kl
9
Differentiation
dx(t )
dt
jk
0
a
k
= jk
2
T
a
k
Fourier Series IV
Slide 42
10
Integration
x(t ) dt
1
jk
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

Fourier Series V
Slide 43
13
Parsevals relation
1
T
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 satised
1
Over any period, x(t ) must be absolutely integrable, that is,
T
[x(t )[ dt <
2
In any nite interval of time, x(t ) is of bounded variation;
that is, there are more than a nite number of maxima and
minima during any single period of the signal.
Fourier Series VI
Slide 44
3
In any nite interval of time, there are only a nite number
of discontinuities. Furthermore, each of these
discontinuities is nite.
Discrete-time Fourier series pair
x[n] =
k=<N>
a
k
e
jk
0
n
=
k=<N>
a
k
e
jk(2/N)n
a
k
=
1
N
n=<N>
x[n]e
jk
0
n
=
1
N
n=<N>
x[n]e
jk(2/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
jk
0
n
0
Fourier Series VII
Slide 45
3
Frequency shifting
x[n]e
jM
0
n
a
kM
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
Fourier Series VIII
Slide 46
8
Multiplication
x[n]y[n]
l =<N>
a
l
b
kl
9
First difference
x[n] x[n 1] (1 e
jk
0
)a
k
10
Running sum
n
k=
x[k]
1
1 e
jk
0
a
k
Fourier Series IX
Slide 47
11
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
Parsevals relation
1
N
n=<N>
[x[n][
2
=
k=<N>
[a
k
[
2
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
jk
0
)e
jk
0
t
where a
k
are the Fourier series coefcients of x(t ).
Similarly, for a period signal
y[n] =
k=<N>
a
k
H(e
jk
0
)e
jk
0
n
Part V
Fourier Transform
Fourier Transform I
Slide 50
Fourier transform
X(f ) = F [x(t )] =
x(t )e
j 2ft
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 ) =
+
k=
2a
k
( k
0
)
where a
k
are the Fourier series coefcients of x(t ).
Convergence of Fourier Transform: For a signal to have
Fourier transform, the following conditions developed by
P.L. Dirichlet must be satised
Fourier Transform II
Slide 51
1
T
[x(t )[ dt <
2
3
Fourier Transform III
Slide 52
The Fourier transform for periodic signals
X(j ) =
+
k=
2a
k
( k
0
)
Example
The signal x(t ) = sin
0
t has nonzero Fourier series
coefcients a
1
= 1/2j and a
1
= 1/2j . Therefore
X(j ) = /j (
0
) /j ( +
0
)
Fourier transform pairs: Some of the common Fourier
transform pairs are as follows:
Fourier Transform IV
Slide 53
1
Rectangular
sincf
2
Triangular
sinc
2
f
3
Gaussian
e
(bt )
2
(1/b)e
(f /b)
2
4
Symmetric exponential
e
b|t |

2b
b
2
+ (2f )
2
5
Sinc
sinc2Wt
1
2W
f
2W

Fourier Transform V
Slide 54
6
Sinc squared
sinc
2
2Wt
1
2W
f
2W
7
Constant
1 (f )
8
Phasor
e
j (
c
t +)
e
j
(f f
c
)
9
Sinusoid
cos(
c
t +)
1
2
e
j
(f f
c
) + e
j
(f + f
c
)
10
Impulse
(t t
d
) e
j t
d
Fourier Transform VI
Slide 55
11
Sampling
k=
(t kT
s
) f
s
n=
(f nf
s
)
12
Signum
sgn t 1/j f
13
Step
u(t )
1
j 2f
+
1
2
(f )
Fourier transform properties: The Fourier transform
properties are as follows
1
Linearity
ax(t ) + by(t ) aX(j ) + bY(j )
Fourier Transform VII
Slide 56
2
Time shifting
x(t t
0
) e
j t
0
X(j )
3
Frequency shifting
x(t )e
j
0
t
X(j (
0
))
4
Conjugation
x
(t ) X
(j )
5
Time reversal
x(t ) X(j )
6
Time and frequency scaling
x(t )
1
[[
X
7
Convolution
x(t ) y(t ) X(j )Y(j )
Fourier Transform VIII
Slide 57
8
Multiplication
x(t )y(t )
1
2
X(j ) Y(j )
9
Differentiation in time
dx(t )
dt
j X(j )
10
Integration
x(t ) dt
1
j
X(j ) +X(0)()
11
Differentiation in frequency
tx(t ) j
d
d
X(j )
Fourier Transform IX
Slide 58
12
X(j ) = X
(j )
'X(j ) = 'X(j )
X(j ) = X(j )
[X(j )[ = [X(j )[
argX(j ) = argX(j )
13
Symmetry
x(t ) real and even X(j ) real and even
x(t ) real and odd X(j ) imaginary and odd
14
Even and odd decomposition for real signals
x
e
(t ) 'X(j )
x
o
(t ) j X(j )
Fourier Transform X
Slide 59
15
Parsevals relation
[x(t )[
2
dt =
1
2
[X(j )[
2
d
Convergence of Fourier transform: The sufcient
conditions for the convergence are:
1
T
[x(t )[ dt <
2
3
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 .
Part VI
Discrete-Time Fourier Transform
Discrete-Time Fourier Transform I
Slide 62
Synthesis equation
x[n] =
1
2
2
X(e
j
)e
j n
d
Analysis equation
X(e
j
) =
+
n=
x[n]e
j n
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(2/N)n
has the Fourier transform
X(e
j
) =
+
k=
2a
k

2k
N
Example
The DTFT of x[n] = cos
0
n is
X(e
j
) = (
0
) + ( +
0
)
Discrete-Time Fourier Transform III
Slide 64
Basic properties of the DTFT are the following:
1
Periodicity
X(e
j (+2)
) = X(e
j
)
2
Linearity
ax
1
[n] + bx
2
[n] a
1
X
1
(e
j
) + bX
2
(e
j
)
3
Time shifting
x[n n
0
] e
j n
0
X(e
j
)
4
Frequency shifting
e
j
0
n
x[n] X(e
j (
0
)
)
5
Conjugation
x
[n] X
(e
j
)
where denotes the complex conjugate.
Discrete-Time Fourier Transform IV
Slide 65
6
Time reversal
x[n] X(e
j
)
7
Time expansion
x
(k)
[n] =
x[n/k], if n = multiple of k
0, if n = multiple of k
X(e
j
)
8
Convolution
x[n] y[n] X(e
j
)Y(e
j
)
9
Multiplication
x[n]y[n]
1
2
2
X(e
j
)Y(e
j ()
) d
10
Differencing in time
x[n] x[n 1] (1 e
j
)X(e
j
)
Discrete-Time Fourier Transform V
Slide 66
11
Accumulation
n
k=
x[n]
1
1 e
j
X(e
j
) +X(e
j 0
)
+
k=
( 2k)
12
Differentiation in frequency
nx[n]
dX(e
j
)
d
13
X(j ) = X
(e
j
)
'X(e
j
) = 'X(e
j
)
X(e
j
) = X(e
j
)
[X(e
j
)[ = [X(e
j
)[
argX(e
j
) = argX(e
j
)
Discrete-Time Fourier Transform VI
Slide 67
14
Symmetry for real, even signals
x[n] real and even X(e
j
) real and even
15
Symmetry for real, odd signals
x[n] real and odd X(e
j
) purely imaginary and odd
16
even-odd decomposition
x
e
[n] 'X(e
j
)
x
o
[n] j X(e
j
)
17
Parsevals relation
+
n=
[x[n][
2
=
1
2
2
[X(e
j
)[
2
d
Discrete-Time Fourier Transform VII
Slide 68
Basic discrete-time Fourier transform pairs
1
k=<N>
a
k
e
j (2/N)n
2
+
k=
a
k

2k
N
2
e
j
0
n
2
+
l =
(
0
2l )
3
cos
0
n
+
l =
(
0
2l ) +( +
0
2l )
Discrete-Time Fourier Transform VIII
Slide 69
4
sin
0
n

j
+
l =
(
0
2l ) ( +
0
2l )
5
x[n] = 1 2
+
l =
( 2l )
6
periodic square wave with period N
x[n] =
1, [n[ N
1
0, N
1
< [n[ N/2
2
+
k=
a
k

2k
N
7
+
k=
[n kN]
2
N
+
k=

2k
N

Discrete-Time Fourier Transform IX
Slide 70
8
a
n
u[n], [a[ < 1
1
1 ae
j
9
x[n] =
1, [n[ N
1
0, [n[ > N
1
sin[(N
1
+
1
2
)]
sin(/2)
10
sinWn
n
=
W
sinc
Wn
X() =
1, 0 [[ W
0, W < [[
0 < W < , X() periodic with period 2.
11
[n] 1
Discrete-Time Fourier Transform X
Slide 71
12
u[n]
1
1 e
j
+
+
k=
( 2k)
13
[n n
0
] e
j n
0
14
(n + 1)a
n
u[n], [a[ < 1
1
(1 ae
j
)
2
15
(n + r 1)!
n!(r 1)!
a
n
u[n], [a[ < 1
1
(1 ae
j
)
r
Discrete-Time Fourier Transform XI
Slide 72
Example
Find the convolution of h[n] =
n
u[n] and x[n]
n
u[n].
H(e
j
) =
1
1 e
j
X(e
j
) =
1
1 e
j
So
Y(e
j
) =
1
(1 e
j
)(1 e
j
)
By PFE and taking the inverse transform
y[n] =
1

[
n+1
u[n]
n+1
u[n]]
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
) =
2
1
3
4
e
j
+
1
8
e
j 2
By performing PFE on H(e
j
) and by inspection
h[n] = 4
1
2
n
u[n] 2
1
4
n
u[n]
Part VII
Sampling
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
M
.
2
Sampling frequency
s
of x
s
(t ) must be greater than 2
M
,
i.e.
s
> 2
M
.
The second condition is also known as Nyquist criterion.
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.
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=
(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)(t nT)
From the multiplication property
X
p
(j ) =
1
2
[X(j )] P(j )
Sampling III
Slide 77
As
P(j ) =
2
T
+
k=
( k
s
)
Since convolution with an impulse simply shifts a signal, it
follows that
X
p
(j ) =
1
T
+
k=
X(j ( k
s
))
That is, X
p
(j ) is a periodic function of consisting of
superposition of shifted replicas of X(j ), scaled by 1/T.
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)(t nT)
Hence,
x
r
(t ) =
+
n=
x(nT)h(t nT)
For an ideal LPF
h(t ) =

c
T sin(
c
t )
c
t
Sampling V
Slide 79
So
x
r
(t ) =
+
n=
x(nT)
c
T
sin(
c
(t nT))
c
(t nT)
Part VIII
Discrete Fourier Transform
Discrete Fourier Transform I
Slide 81
Discrete Fourier transform: Let x[n] be a nite-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 dened by
X[k] =
N1
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 (2/N)
Discrete Fourier Transform II
Slide 82
The inverse DFT (IDFT) is given by
x[n] =
1
N
N1
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 2/N
3
Frequency shifting
W
kn
0
N
x[n] X[k k
0
]
modN
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] =
N1
i =0
x
1
[i ]x
2
[n i ]
modN
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] =
N1
i =1
X
1
[i ]X
2
[k i ]
modN
9
Parsevals relation
N1
n=0
[x[n][
2
=
1
N
n=0
N 1[X[k][
2
Part IX
The z-Transform
The z-Transform I
Slide 86
Denition
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
where r is the magnitude of z and is the angle of z.
The z-Transform II
Slide 87
Denition
The range of values of the complex variable z for which the
z-transform converges is called the region of
convergence (ROC).
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[
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
The z-Transform V
Slide 90
Example
The z-transform of x[n] =
1
3
n
sin
4
n
u[n] is
X(z) =
1
3
2
z
(z
1
3
e
j /4
)(z
1
3
e
j /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.
The z-Transform VI
Slide 91
3
If x[n] is of nite duration, then the ROC is the entire
z-plane, except possibly z = 0 and/or z = .
Example
The ROC of [n] 1 consists of the entire z-plane. The ROC of
[n 1] z
1
consists of the entire z-plane, including z = but
excluding z = 0. The ROC of [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 nite 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.
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 innity.
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 = .
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
2j
X(z)z
n1
dz
where

denotes integration around a counterclockwise
closed circular contour centered at the origin and with
radius r .
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.
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]
The z-Transform XI
Slide 96
Example
X(z) =
3
5
6
z
1
(1
1
4
z
1
)(1
1
3
z
1
)
,
1
4
< [z[ <
1
3
x[n] =
1
4
n
u[n] 2
1
3
n
u[n 1]
The z-Transform XII
Slide 97
Example
X(z) =
3
5
6
z
1
(1
1
4
z
1
)(1
1
3
z
1
)
, [z[ <
1
4
x[n] =
1
4
n
u[n 1] 2
1
3
n
u[n 1]
The z-Transform XIII
Slide 98
Example
X(z) = 4z
2
+ 2 + 3z
1
. 0 < [z[ <
From the power-series denition of the z-transform, we can
determine the inverse transform of X(z) by inspection:
x[n] = 4[n + 2] + 2[n] + 3[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
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
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
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
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
[n] 1, ROC: All z
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
[nm] z
m
, ROC: All z except 0 (if m > 0) or (if m < 0)
5
n
u[n]
1
1 z
1
, ROC: [z[ > [[
6
n
u[n 1]
1
1 z
1
, ROC: [z[ < [[
The z-Transform XIX
Slide 104
7
n
n
u[n]
z
1
(1 z
1
)
2
, ROC: [z[ > [[
8
n
n
u[n 1]
z
1
(1 z
1
)
2
, ROC: [z[ < [[
9
[cos
0
n]u[n]
1 [cos
0
]z
1
1 [2 cos
0
]z
1
+ z
2
, ROC: [z[ > 1
10
[sin
0
n]u[n]
[sin
0
]z
1
1 [2 cos
0
]z
1
+ z
2
, ROC: [z[ > 1
The z-Transform XX
Slide 105
11
[r
n
cos
0
n]u[n]
1 [r cos
0
]z
1
1 [2r cos
0
]z
1
+ r
2
z
2
, ROC: [z[ > r
12
[r
n
sin
0
n]u[n]
[r sin
0
]z
1
1 [2r cos
0
]z
1
+ r
2
z
2
, ROC: [z[ > r
Denition
A discrete-time LTI system is causal if and only if the ROC of its
system function is the exterior of a circle, including innity.
The z-Transform XXI
Slide 106
Denition
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
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.
The z-Transform XXIII
Slide 108
Denition
An LTI system is stable if and only if the ROC of its system
function H(z) includes the unit circle, [z[ = 1.
The z-Transform XXIV
Slide 109
Example
H(z) =
1
1
1
2
z
1
+
1
1 2z
1
, 1/2 < [z[ < 2
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.
The z-Transform XXV
Slide 110
Example
H(z) =
1
1
1
2
z
1
+
1
1 2z
1
, [z[ < 1/2
h[n] =
1
2
n
+ 2
n
u[n 1]
Since h[n] is neither stable nor causal.
The z-Transform XXVI
Slide 111
Denition
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.
The z-Transform XXVII
Slide 112
For system characterized by linear constant-coefcient
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.
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-coefcient 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
Part X
Laplace Transform
Laplace Tranform I
Slide 115
The basic inverse Laplace transform equation is
x(t ) =
1
2j
+j
j
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 = . This line is
parallel to the j -axis. Furthermore, we can choose any
such line in the ROCi.e., we can choose any value such
that X( + j ) converges.
Some common Laplace transform pairs are as follows
1
(t ) 1
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(
0
t )u(t )

0
s
2
+
2
0
Laplace Tranform III
Slide 117
7
cos(
0
t )u(t )
s
s
2
+
2
0
8
t sin(
0
t )u(t )
2
0
s
(s
2
+
2
0
)
2
9
t cos(
0
t )u(t )
(s
2
2
0
)
(s
2
+
2
0
)
2
10
e
at
sin(
0
t )u(t )

0
(s + a)
2
+
2
0
11
e
at
cos(
0
t )u(t )
s + a
(s + a)
2
+
2
0
Laplace Tranform IV
Slide 118
The Laplace transform of a general continuous-time signal
x(t ) is denes as
X(s)
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 = + j , with and 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
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
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
x() d
1
s
X(s), ROC contains R 's > 0
10
Initial- and nal-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
s0
sX(s).
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 -axis in
the s-plane.
2
For rational Laplace transforms, the ROC does not contain
any poles.
3
If x(t ) is of nite 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 =
0
is in the ROC,
then all values of s for which 's >
0
will also be in the
ROC.
5
If x(t ) is left sided, and if the line 's =
0
is in the ROC,
then all values of s for which 's <
0
will also be in the
ROC.
6
If x(t ) is two sided, and if the line 's =
0
is in the ROC,
then the ROC will consist of a strip in the s-plane that
includes the line 's =
0
.
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 innity. 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.
Part XI
Amplitude (Linear) Modulation
Double Sideband-Suppressed Carrier
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(2f
c
t )
or
s(t ) = 'g(t ) exp(j 2f
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.
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 bers.
Carrier Communication uses modulation.
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
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.
Double Sideband (DSB) IV
Slide 127
Figure: Transmitter.
Double Sideband (DSB) V
Slide 128
Figure: receiver.
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
c
t
1
2
[M( +
c
) + M(
c
)]
That is, the DSB modulation translates or shifts the
frequency spectrum to the left and the right by
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
c
.
Upper sideband (USB) is a portion of DSB spectrum which
lies above
c
.
Double Sideband (DSB) VII
Slide 130
DSB signal does not contain a discrete component of the
carrier frequency
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.
Double Sideband (DSB) VIII
Slide 131
1
Multiplier Modulator: Here modulation is achieved directly
by multiplying m(t ) by cos
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 ampliers. Here the basic components are a
logarithmic and an antilogarithmic amplier with outputs
proportional to the log and antilog of their inputs,
respectively. Using two logarithmic ampliers, we generate
and add the logarithms of the two signals to be multiplied.
The sum is then applied to an antilogarithmic amplier to
obtain the desired product. That is,
AB = antilog[log(AB)]
= antilog[logA + logB]
It is rather difcult to maintain linearity in this kind of
amplier, and they tend to be rather expensive.
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
c
. For example with square pulse train
w(t ) =
1
2
+
2
cos
c
t
1
3
cos 3
c
t +
1
5
cos 5
c
t
or with a square waveform

w(t ) =
4
cos
c
t
1
3
cos 3
c
t +
1
5
cos 5
c
t

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
c
t from
c
to some other frequency
I
. This can
be done by multiplying m(t ) cos
c
t by 2 cos
mix
t , where
mix
=
c
+
I
for up-conversion or
mix
=
c
I
for
down-conversion and then bandpass-ltering the product.
Double Sideband (DSB) XI
Slide 134
Figure: Block diagram of product modulator.
Double Sideband (DSB) XII
Slide 135
Figure: DSB waveforms.
Double Sideband (DSB) XIII
Slide 136
Figure: Spectrum of baseband signal.
Double Sideband (DSB) XIV
Slide 137
Figure: Spectrum of DSB-SC modulated wave.
Double Sideband (DSB) XV
Slide 138
Figure: Coherent detector for demodulating DSB-SC modulated
wave.
Double Sideband (DSB) XVI
Slide 139
Figure: Spectrum of a product modulator output with a DSB-SC
modulated wave as input.
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(2f
a
t ) + cos(2f
b
t )]
[cos(2f
a
t +
1
) + cos(2f
b
t +
1
)]
[cos(2f
a
t +
2
) + cos(2f
b
t +
2
)]
[cos(2f
a
t +
3
) + cos(2f
b
t +
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 ltering to remove the unwanted components.
Double Sideband (DSB) XVIII
Slide 141
(a) Determine the conditions that the phase angles
1
,
2
,
3
and
1
,
2
,
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.
Part XII
Ordinary Amplitude Modulation
Amplitude Modulation (AM) I
Slide 143
AM = DSB-SC + Carrier.
Time domain expression
AM
(t ) = Acos
c
t + m(t ) cos
c
t
= [A + m(t )] cos
c
t
Spectrum
AM
(t )
1
2
[M(+
c
)+M(
c
)]+A[(+
c
)+(
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.
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
c
t
satises 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
=
m
p
A
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 > 1 or A < m
p
.
Denition
Power efciency
=
useful power
total power
=
P
s
P
c
+ P
s
where P
c
the carrier power, and P
s
is the power in sidebands.
Amplitude Modulation (AM) IV
Slide 146
Example
For tone modulation m(t ) = Acos
m
t , P
s
=
(A)
2
2
, and
P
c
=
A
2
2
. Hence
=

2
2 +
2
100%
With maximum modulation index, that is = 1, = 33.3%,
which means that only 33.3% of the power is used to transmit
the sidebands.
Amplitude Modulation (AM) V
Slide 147
Denition
The percentage of positive modulation on an AM signal is
(A
max
A
c
)/(A
c
) 100 = max[m(t )] 100
Denition
The percentage of negative modulation is
(A
c
A
min
)/(A
c
) 100 = min[m(t )] 100
Amplitude Modulation (AM) VI
Slide 148
Denition
The overall modulation percentage is
(A
max
A
min
)/(2A
c
) 100
or
(max[m(t )] min[m(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.
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
c
t + m(t )]w(t )
where
w(t ) =
1
2
+
2
cos
c
t
1
3
cos 3
c
t +
1
5
cos 5
c
t

Amplitude Modulation (AM) VIII
Slide 150
Hence
v
bb
(t ) =
c
2
cos
c
t +
2
m(t ) cos
c
. .. .
AM
+ other terms
. .. .
suppressed by bandpass lter
Demodulation of AM signals
1
Rectier Detector:
v
R
= [A + m(t )] cos
c
t w(t )
where
w(t ) =
1
2
+
2
cos
c
t
1
3
cos 3
c
t +
1
5
cos 5
c
t
Hence
v
R
=
1
[A + m(t )] + other terms of higher frequencies

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/
c
but
should be small compared to 1/2B, where
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 innite
percentage modulation because there is no carrier line
component.
The modulation efciency 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).
Amplitude Modulation (AM) X
Slide 152
Figure: AM waveforms (a) Message.
Amplitude Modulation (AM) XI
Slide 153
Figure: AM waveforms (b) AM wave with < 1.
Amplitude Modulation (AM) XII
Slide 154
Amplitude Modulation (AM) XIII
Slide 155
Figure: AM waveforms (c) AM wave with > 1.
Amplitude Modulation (AM) XIV
Slide 156
Figure: Spectrum of baseband signal.
Amplitude Modulation (AM) XV
Slide 157
Figure: Spectrum of AM wave.
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
QAM
(t ) = m
1
(t ) cos
c
t + m
2
(t ) sin
c
t
demodulation
2
QAM
(t ) cos
c
t = 2[m
1
(t ) cos
c
t + m
2
(t ) sin
c
t ] cos
c
t
= m
1
(t ) + m
1
(t ) cos 2
c
t + m
2
(t ) sin 2
c
t
. .. .
suppressed by lowpass ltering
Amplitude Modulation (AM) XVII
Slide 159
Phase error results in crosstalk
2
QAM
(t ) cos(
c
t +)
Amplitude Modulation (AM) XVIII
Slide 160
Figure: Transmitter.
Amplitude Modulation (AM) XIX
Slide 161
Figure: Receiver.
Part XIII
Single Sideband Modulation
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() = M
+
() + M
()
m(t ) = m
+
(t ) + m
(t )
Single Sideband (SSB) II
Slide 164
[M
+
()[ and [M
()[ are not even functions of .

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
+
() = M()u()
=
1
2
M()[1 + sgn()]
=
1
2
M() +
1
2
M()sgn()
Single Sideband (SSB) III
Slide 165
It can be seen that
jm
h
(t ) M()sgn()
Hence
M
h
() = jM()sgn()
From the table of Fourier transform pairs
1/t j sgn()
Therefore
m
h
(t ) = m(t ) 1/t
or
m
h
(t ) =
1
m()
t
d
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() = j sgn()
It follows that [H()[ = 1 and
h
() = /2 for > 0 and
/2 for < 0. Thus, if we delay the phase of every
component of m(t ) by /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
m
t , the Hilbert
transform is
m
h
(t ) = cos
m
t

2
= sin
m
t
Single Sideband (SSB) V
Slide 167
USB Spectrum
USB
() = M
+
(
c
) + M
( +
c
)
USB signal
USB
(t ) = m
+
(t )e
j
c
t
+ m
(t )e
j
c
t
Substituting for m
+
(t ) and m
(t )
SSB
(t ) = m(t ) cos
c
t m
h
(t ) sin
c
t
where the minus sign applies to USB and the plus signa
applies to LSB.
Generation of SSB signals
Single Sideband (SSB) VI
Slide 168
1
Selective-Filtering Method: A DSB-SC signal is passed
through a bandpass lter to eliminate the undesired
sideband. This method is used in speech processing.
2
Phase-Shift Method:
SSB
(t ) = m(t ) cos
c
t m
h
(t ) sin
c
t
Demodulation of SSB signals
1
Synchronous Method:
SSB
(t ) is multiplied by cos
c
t and
is then passed through a low-pass lter.
Single Sideband (SSB) VII
Slide 169
2
Envelope Detection Method:
SSB+C
= Acos
c
t + [m(t ) cos
c
t + m
h
(t ) sin
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
Single Sideband (SSB) VIII
Slide 170
Using binomial expansion and discarding higher order
terms
E(t ) A
1 +
m(t )
A
= A + m(t )
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.
Single Sideband (SSB) X
Slide 172
Figure: (b) Spectrum of corresponding SSB signal containing the
upper sideband.
Single Sideband (SSB) XI
Slide 173
Problem
Consider the modulated waveform
s(t ) = A
c
cos(2f
c
t ) + m(t ) cos(2f
c
t )

m(t ) sin(2f
c
t )
which represents a carrier plus an SSB signal, with m(t )
denoting the message and

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
Single Sideband (SSB) XII
Slide 174
Figure: Modulation steps in an FDM system.
Part XIV
Vestigial Sideband Modulation
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 rectier) detector.
Vestigial Sideband (VSB) II
Slide 177
If the vestigial shaping lter that produces VSB from DSB
is H
i
(), then the resulting VSB signal spectrum is
VSB
() = [M( +
c
) + M(
c
)]H
i
()
This shaping lter 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
VSB
(t ) by 2 cos
c
t is
e(t ) = 2
VSB
(t ) cos
c
t [
VSB
( +
c
) +
VSB
(
c
)]
Vestigial Sideband (VSB) III
Slide 178
The signal e(t ) is further passed through a low-pass
equalizer lter of transfer function H
0
() whose output is
required to be m(t ). Hence,
M() = [
VSB
( +
c
) +
VSB
(
c
)]
By substitution of
VSB
() in this equation and eliminating
the spectra 2
c
by a low-pass lter H
0
(), we obtain
H
0
() =
1
H
i
( +
c
) + H
i
(
c
)
[[ 2B
Use of VSB in broadcast television: The baseband video
signal of television occupies an enormous bandwidth of
4.5 MHz.
Vestigial Sideband (VSB) IV
Slide 179
The vestigial shaping lter H
i
() 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).
Vestigial Sideband (VSB) V
Slide 180
Figure: Filtering scheme for the generation of VSB modulated wave.
Vestigial Sideband (VSB) VI
Slide 181
Figure: Magnitude response of VSB lter; only the positive-frequency
portion is shown.
Vestigial Sideband (VSB) VII
Slide 182
Problem
The single tone modulating signal m(t ) = A
m
cos(2f
m
t ) is
used to generate the VSB signal
s(t ) =
1
2
aA
m
A
c
cos[2(f
c
+f
m
)t ]+
1
2
A
m
A
c
(1a) cos[2(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(2f
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?
Part XV
Angle (Exponential) Modulation
Angle (Exponential) Modulation I
Slide 184
generalized sinusoidal signal (t )
(t ) = Acos (t )
where (t ) is the generalized angle and is a function of t .
(t ) = Acos(
c
t +
0
)
instantaneous frequency and phase
i
(t ) =
d
dt
(t ) =
i
() d
Angle (Exponential) Modulation II
Slide 185
Example
The instantaneous frequency in hertz of
cos 200t cos(5 sin2t ) + sin 200t sin(5 sin2t ) is found by
noticing that the phase is (t ) = 200t 5 sin2t . Hence,
i
= 200 10 cos 2t .
angle modulation possibilities: phase modulation (PM) and
frequency modulation (FM).
In PM, the angle (t ) is varied linearly with m(t ):
(t ) =
c
t +
0
+ k
p
m(t )
where k
p
is a constant and
c
is the carrier frequency.
Assuming
0
= 0, without loss of generality
(t ) =
c
t + k
p
m(t )
Angle (Exponential) Modulation III
Slide 186
PM signal
PM
(t ) = Acos[
c
t + k
p
m(t )]
Instantaneous frequency
i
(t ) in PM
i
(t ) =
d
dt
=
c
+ k
p

m(t )
Hence, in PM, the instantaneous frequency
i
varies
linearly with the derivative of the modulating signal.
In FM the instantaneous frequency
i
is varied linearly with
the modulating signal.
i
(t ) =
c
+ k
f
m(t )
where k
f
is a constant.
Angle (Exponential) Modulation IV
Slide 187
In FM
(t ) =
[
c
+ k
f
m()] d
=
c
t + k
f
m() d
FM signal
FM
(t ) = Acos
c
t + k
f
m() d

Angle (Exponential) Modulation V
Slide 188
The generalized angle-modulated carrier
EM
can be
expressed as
EM
(t ) = Acos[
c
t +(t )]
= Acos
c
t +
m()h(t ) d
FM results if h(t ) = k
p
(t ), and PM results if h(t ) = k
f
u(t )
Power of an angle-modulated wave is A
2
/2.
Angle (Exponential) Modulation VI
Slide 189
Figure: Illustrative AM, FM, and PM waveforms.
Angle (Exponential) Modulation VII
Slide 190
Narrow-band FM
FM
(t ) A[cos
c
t k
f
a(t ) sin
c
t ]
Narrow-band PM
PM
(t ) A[cos
c
t k
p
m(t ) sin
c
t ]
Carsons rule for FM
B
FM
= 2B( + 1)
where B is the message BW, and the deviation ratio is
=
f
B
Angle (Exponential) Modulation VIII
Slide 191
where the frequency deviation f is
f =
k
f
m
p
2
where k
f
and m
p
are the FM modulator sensitivity and the
peak of the message, respectively.
Carsons rule for PM
B
PM
= 2(f + B)
where the frequency deviation f is
f =
k
p
m
p
2
where k
p
and m
p
are the PM modulator sensitivity and the
peak of the derivative of the message, respectively.
Angle (Exponential) Modulation IX
Slide 192
Example
For
x
c
(t ) = 10 cos[(10
8
)t + 5 sin2(10
3
)t ]
the phase (t ) =
c
t +(t ). Hence, (t ) = 5 sin2(10
3
)t and
(t ) = 5(2)(10
3
) cos 2(10
3
)t . Therefore, [(t )[
max
= 5 rad,
and = [
(t )[
max
= 5(2)(10
3
) rad/s or f = 5 kHz.
Example
Given the angle-modulated signal
x
c
(t ) = 10 cos(210
8
t + 200 cos 210
3
t ), the instantaneous
frequency is
i
= 2(10
8
) 4(10
5
) sin 2(10
3
)t . So
= 4(10
5
),
m
= 2(10
3
), and =

m
= 200. The BW is
W
B
= 2( + 1)
m
= 8.04(10
5
) rad/s
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[
c
t +(t )]
then
y(t ) =
a
2
2
+ a
1
cos[
c
t +(t )] +
a
2
2
cos[2
c
t + 2(t )]
Angle (Exponential) Modulation XI
Slide 194
A similar nonlinearity in AM not only causes unwanted
modulation with carrier frequencies n
c
but also causes
distortion of the desired signal. For instance, if a DSB-SC
signal m(t ) cos
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
c
t +
b
4
m
3
(t ) cos 3
c
t
Passing this signal through a bandpass lter yields
[am(t ) + (3b/4)m
3
(t )] cos
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.
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
FM
(t ) to an ideal
differentiator, the output is

FM
(t ) = A[
c
+ k
f
m(t )] sin
c
t + k
f
m()d()
The signal
FM
(t ) is both amplitude and frequency
modulated, the envelope being A[
c
+ k
f
m(t )]. The
message can be obtained by envelope detection of
FM
(t ).
Angle (Exponential) Modulation XIII
Slide 196
Figure: (a) Scheme for generating an FM wave by using a phase
modulator.
Angle (Exponential) Modulation XIV
Slide 197
Figure: (b) Scheme for generating a PM wave by using a frequency
modulator.
Angle (Exponential) Modulation XV
Slide 198
Figure: Narrowband FM.
Angle (Exponential) Modulation XVI
Slide 199
Figure: Plots of Bessel functions of the rst kind for varying order.
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 xed frequency and varying amplitude. Only the spectra for
positive frequencies are shown.
Angle (Exponential) Modulation XVIII
Slide 201
Angle (Exponential) Modulation XIX
Slide 202
Angle (Exponential) Modulation XX
Slide 203
of varying frequency and xed amplitude. Only the spectra for
Angle (Exponential) Modulation XXI
Slide 204
Angle (Exponential) Modulation XXII
Slide 205
Angle (Exponential) Modulation XXIII
Slide 206
Figure: Block diagram of the indirect method of generating a
wideband FM signal.
Angle (Exponential) Modulation XXIV
Slide 207
Figure: Block diagram of frequency multiplier.
Angle (Exponential) Modulation XXV
Slide 208
Problem
Consider a narrow-band FM signal approximately dened by
s(t ) A
c
cos(2f
c
t ) A
c
sin(2f
c
t ) sin(2f
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 , assuming that is
restricted to the interval 0 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 ,
assuming that is restricted to the interval 0 0.3.
Angle (Exponential) Modulation XXVI
Slide 209
(c) By expanding the angle
i
(t ) of the narrow-band FM signal
s(t ) in the form of a power series, and restricting the
modulation index to a maximum value of 0.3 radians,
show that
i
(t ) 2f
c
t + sin(2f
m
t )

3
3
sin
3
(2f
m
t )
What is the value of the harmonic distortion for = 0.3?
Problem
The sinusoidal modulating wave
m(t ) = A
m
cos(2f
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
.
Angle (Exponential) Modulation XXVII
Slide 210
(a) Determine the spectrum of the resulting phase-modulated
signal, assuming that the maximum phase deviation
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
p
.
This modulated signal is applied to an ideal band-pass lter
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 lter
output as functions of time.
Angle (Exponential) Modulation XXVIII
Slide 211
Problem
An FM signal with modulation index = 1 is transmitted
through and ideal band-pass lter 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 lter output.
Problem
Consider a wide-band PM signal produced by a sinusoidal
modulating wave A
m
cos(2f
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
.
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 rst 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?

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