Time, Frequency Analysis of Signals

Introduction
Fast development in the area of Digital signal
processing (DSP) is due to
1. Rapid development in digital computer technology
and IC fabrication has led to
Cheaper realization of DSP systems
Usage of DSP systems from non Real time to RT systems

2. DSP systems can perform complex SP functions

Difficult and expensive to be performed by analog systems

3. Development of digital storage systems allow storage

of digital signals without deterioration

Introduction
3. DSP systems are more reliable than analog signal
processing systems
The precision achieved with digital HW and SW is much better
compared to their analog counter parts

4. DSP systems allows programmable operations

Signal processing functions can be more easily modified in
software than in hardware
This improves the flexibility in system design

DSP is not suitable for all signal processing problems

For RT processing of very wide BW signals, analog/ optical
processing is still the solution

Signal?
A signal represents a physical quantity or variable
It contains information about the nature or state of a
physical system
It is represented mathematically as a function of one or
more independent variables
Example: s1(t) = 20t2 represent a signal with time t as the
independent variable
s(x, y)= 3x2 + 2xy + 10y2 is a signal of two
independent variables x and y
A picture is a brightness function of two spatial variables
By default time is taken as the independent variable

Signals are processed to extract information

4

Signal Processing Systems

Signal Processing Systems

The Block diagram of Digital signal processing
system consists of three elements
Interfaces between Analog signal and Digital signal
These interfaces are A/D Converter and D/A Converter
A/D Converter is a series of three elements:
Sampler, Quantizer and Coder

The value of a signal can be a

Real quantity:
s1(t) = A sin3t
Complex quantity: s2(t) = Aej3t = A cos3t + j A sin3t
Vector quantity (generated by multiple sources):
s(t) = [s1(t) s2(t) s3(t)]T
6

Classification of signals
Continuous time signal

Discrete time signal

Classification of signals
Analog and Digital signals
A Digital signal has discrete values for both Time and
Amplitude
They can take only a finite number of distinct amplitude
values
Quantization: Process of Amplitude discretization

Analog signals is one for which both time and

amplitude take a continuum of values

Even and Odd signals

Even :

x ( n) = x ( n)

Odd :

x ( n) = x ( n)

Any signal can be represented as

Even component + Odd component
x(n) = xe (n) + xo (n)
1
[x(n) + x(n)]
2
1
xo (n) = [x(n) x( n)]
2

xe (n) =

Even and Odd signals

Even and Odd signals

Periodic and Aperiodic signals

A Continuous time signal is periodic, if x(t + mT) = x(t)
for all values of t and integer values m
The smallest value of T for which this relation holds is its
Fundamental period, represented as T0
A CT signal is Aperiodic, if it is not periodic

A Discrete time signal is periodic, if x(n+mN) = x(n) for

all n and integer values m
The smallest value of N for which this relation holds is its
Fundamental period, represented as N0

12

Concept of frequency
Frequency is the number of cycles/s of a periodic signal
Mathematically we have negative frequencies too

A signal in complex exponential form is

xa (t ) = Ae j (t + ) = A[cos(t + ) + j sin(t + )]

A j (t + )
A[cos(t + )] = e
+ e j (t + )
2

Sum of two equal amplitude complex conjugate phasors

Angular frequencies rad/s CW and ACW rotations
ACW Positive frequency; CW Negative frequency
Thus, we have frequencies in the range - < F <
13

Discrete-Time sinusoidal signals

A discrete-time sinusoidal signal is expressed as
x(n) = A cos(n + ) = A cos(2fn + ),

< n<

Sample number, n, is an integer variable

A is the amplitude, is the phase in radians
is the frequency in radians per sample
f has the dimension of cycles/sample

14

Properties of Discrete-time sinusoids

1. A discrete-time sinusoid is periodic only if its
frequency is a rational number
Periodicity implies x(n + N) = x(n) for all n
The smallest value of N for which this relation is
satisfied is the Fundamental period
A sinusoid with frequency f0 is periodic if
cos [2f0(n + N) + ] = cos [2f0n + ]
0 (= 2f0) is the frequency of the sinusoid

15

Properties of Discrete-time sinusoids

Above relation is satisfied only if 2f0N = 2k
Or f0 = k/N - A rational number
There should not be a common factor between k and N
N is the Fundamental period of the sinusoid

A small change in frequency can result in a large change

in Fundamental period
For f1 = 31/60, N1 = 60
For f2 = 30/60 N2 = 2

16

Properties of Discrete-time sinusoids

2. DT sinusoids with frequencies separated by an
integer multiple of 2 are identical

cos[(0 + 2 )n + ] = cos[0 n + 2n + ] = cos[0 n + ]

Sinusoidal sequences xk (n) are identical

xk (n) = cos[k n + ], k = 0, 1, 2,

k = 0 + 2k ,
0
Sinusoids with frequency | | > is identical to
that with frequency | | <
A sinusoid with frequency | | > is called an alias of a
corresponding sinusoid with frequency | | <
17

Properties of Discrete-time sinusoids

The ranges OR f are unique
All frequencies with ||> Or |f|>

are aliases

Frequency range for discrete-time sinusoids is finite

with duration 2
Frequencies in any interval of 2 constitute all the existing
discrete-time sinusoids
Usually the range is used for all analysis

Negative frequencies for Discrete-time signals

A ( jn + )
x(n) = A cos(n + ) = e
+ e ( jn + )
2
The concept of Negative frequencies
18

Harmonically related Complex Exponentials

A DT complex exponential is periodic, if its frequency f 0 is a
rational number : Let f 0 = 1/N
A set of harmonically related complex exponentials is
sk (n) = e j 2kf 0 n = e

sk + N (n) = e j 2 ( k + N ) f 0 n =

j 2kn
N ;

j 2kn
e j 2ne N

= e

k = 0,1,2,...
j 2kn
N

= sk (n)

The set sk (n) has only N distinct periodic complex exponentials

All members of sk (n) have a common period of N samples
19

Harmonically related Complex Exponentials

N complex exponentials

sk (n) = e

j 2kn
N ;

k = 0,1,2,..., ( N 1)

1
is a harmonically related set with Fundamental frequency f 0 =
N
x(n) can be written as a linear combination
x ( n) =

N 1

N 1

k =0

k =0

ck sk (n) = ck e

j 2kn
N

This is the Fourier series representation of a DT sequence

{ck }are the Fourier coefficients
20

DISCRETE-TIME SIGNALS AND

SEQUENCES
21

Discrete-time Sequences
Unit-sample sequence
Unit sample is the discretetime impulse
It has the role of unit impulse
in CT signals and systems
It is also called impulse

Discrete-time Sequences
Unit-step sequence

Shifted unit-step sequence

Discrete-time Sequences
Unit sample is related to Unit step by
Unit step : u (n) =

(n) = u (n) u (n 1)

(k )

k =

Sequence p (n) = a3 (n + 3) + a1 (n 1) + a2 (n 2) + a8 (n 8)
In general,

x ( n) =

x(k ) (n k )

k =

Manipulations of DT signals
Transformation of the independent variable, time
y(n) = x(n k), k is an integer
Delay for positive value of k:
Shift to the right
Advance for negative value of k: Shift to the left

Example: Show the graphical representation of

a) Signals x(n-3) and x(n+2) for the given x(n)
b) Signals x(-n) and x(-n+2) for the given x(n)

25

Manipulations of DT signals

Manipulations of DT signals

Manipulations of DT signals
Down-sampling
Time scaling where y(n) = x(Dn): D is an integer

Example:
Show the graphical representation of the signal x(2n)
for the given signal x(n)

Amplitude scaling by a constant A

Multiply the value of every signal sample by A
y(n) = A x(n)

28

Manipulations of DT signals

Discrete-time system
A system is an operator T[ ]
It maps an input sequence x(n) In to an output
sequence y(n)
y(n) = T[x(n)]
This operator is a mathematical expression or rule
x(n)

T[ ]

Example of a system:

y(n)

Accumulator
30

Discrete-time system
Input - output relationship of the Accumulator is
y (n) =

n 1

k =

k =

x(k ) = x(k ) + x(n) = y(n 1) + x(n)

Current output = Current input + Previous output

y (n 1) is the initial condition to the Accumulator

It is the effect on the system from all previous inputs
If the Accumulator had no prior excitation, y (n 1) = 0
In this case the system is initially Relaxed
Every system is assumed to be Relaxed at n =
31

Linear systems
A linear system is defined by superposition principle
Let y1 (n) = T [x1 (n)] & y2 ( n) = T [x2 (n)]
The system is linear if
T [ax1 ( n) + bx2 ( n)] = aT [x1 (n)] + b[x2 (n)]
= ay1 (n) + by2 (n)
a and b are arbitrary constants

32

Linear shift invariant systems

y (n)

= T[x(n)] = T x(k ) (n k )

k =
=

x(k )T [ (n k )]

k =

where

x(k )hk (n)

k =

hk (n) = T[ (n k )]

hk (n) depends on both n & k; hence not useful in computing y (n)

To circumvent this, we put the constraint of Shift invariance
Shift invariance h(n k ) = T[ (n k )]
k is an integer (positive or negative)
n is related to time; Shift invariance is Time invariance
33

Linear shift invariant systems

With Shift invariance y (n) =

x(k )T [ (n k )]

k =

x ( k ) h ( n)

k =

x ( k ) h( n k )

k =

This is Convolution sum y (n) = x(n) h(n) = h(n) x(n)

An LSI system is fully characterized by its Unit sample
response and is represented by its convolution sum

34

Linear shift invariant systems

Cascaded LSI systems
All the following configurations are identical
x(n)

h1(n)

h2(n)

y(n)

x(n)

h2(n)

h1(n)

y(n)

x(n)

h1(n)*h2(n)

y(n)

Linear shift invariant systems

LSI systems in parallel
The following configurations are identical
h1(n)
x(n)

y(n)

h2(n)

x(n)

h1(n) + h2(n)

y(n)

36

Linear shift invariant systems

Problem :
Consider a system with Unit - sample response
a ,
n0

h( n ) = a n u ( n)
=
0,
n<0
Find the response of the system to the input
n

x ( n ) = u ( n) u ( n N )

Linear shift invariant systems

x(k) and h(k) are shown below
1

x(k)
0

N-1

1
a<1

h(k)

0
38

Linear shift invariant systems

y(n) = nth value of the output sequence
= product x(k)h(n-k)
y(n) = 0 for n<0
The products x(k)h(n-k) are zero, since x(k) & h(n-k) have no
non-zero overlapping samples

For 0n<N, x(k) & h(n-k) have non-zero overlapping

samples for 0kn
Then

y ( n) =

ank =

k =0

( n +1)
1

a
n

1 a

0n< N

39

Linear shift invariant systems

For n N , x(k ) & h(n k ) have non - zero overlapping
samples for 0 k N 1.
Then

y ( n) =

N 1

nk

k =0

y(n)

= a

n 1 a

1 a

nN

a<1

N-1

40

Stability and Causality

Other important factors in system design are
Stability (BIBO stability)
Every bounded input produces a bounded output

Causality
Output of the system do not depend on future inputs

Stability :

Condition is

S=

h( k ) <

Let the input be bounded for all n : x(n) M

y (n) =

h( k ) x( n k )

M h(k ) = MS < ; Bounded

41

Stability and Causality

If S = , output will be unbounded for some bounded input
Eg : -

Let the input be

x ( n) = {

h*( n )
h( n)
0

h( n) 0
h(n)=0

For this bounded input, output is unbounded for n = 0

y ( 0) =

k =

x ( k ) h( k ) =

h( k )

h( k )

= S =

Causality :
Output at n = n0 depends only on the input for n n0
42

Stability and Causality

y ( n) =
=

h( k ) x( n k )

k =

x( k ) h( n k )

k =0

=
=
=

h( k ) x( n k )

k =0
n

x ( k ) h( n k )

k =0
n

h( k ) x( n k )

k =0

h(n) can have values only in the range 0 to n

An LSI system is causal only if its h(n) = 0 for n < 0
A causal sequence is one for which x(n) = 0, for n < 0
43

Stability and Causality

h( n) = a nu ( n)

Example of Causality and Stability :

This system is causal since h(n) = 0, for n < 0

S=

h( k ) =

k =

a =

k =

k =0

The system is stable only if S <

1
1 a

For a < 1,

S=

For a 1,

the series diverges

Thus, for stability

a <1
44

Linear Constant coefficient difference equations

An N th order LCCDE for a system
N

a k y (n k )

k =0

br x(n r )

r =0

x(n) and y (n) are input and output of the system

The system is assumed causal, unless otherwise stated
Evaluate h(n) for the 1st order LCCDE,

y (n) y (n 1) = x(n)

We can evaluate h(n) for the following two conditions

1. Causal system :

h(n) = 0, for n < 0

2. Non - causal system : h(n) = 0, for n > 0

45

Linear Constant coefficient difference equations

When x(n) = (n), y (n) = h(n),
Condition 1 :

h(n) = ah(n 1) + (n)

h(0) = ah(1) + 1

= 1,

h(n) = ah(n 1) + 0 = a n
Condition 2 :

h(1) = ah(0) + 0

=a

= a nu ( n)

h(n 1) = (1 / a)[h(n) (n)] OR h(n) = (1 / a )[h(n + 1) (n + 1)]

h(0) = (1 / a )[h(1) (1)] = 0,

h(1) = (1 / a)[h(0) (0)] = a 1

h( n) = (1 / a )[h( n + 1) ( n + 1)] = a n = a nu ( n 1)
46

Linear Constant coefficient difference equations

This LCCDE do not have a unique solution
h(n)= anu(n) - Causal lter that is Stable if a<1
h(n)= -anu(-n-1): Non-causal lter that is Stable if a >1
This difficulty is overcome with the additional
constraint of causality
Unless stated otherwise, a system that satisfies a LCCDE also
satisfy the condition for it to be an LSI system

47

FIR and IIR systems

LSI systems can have two types of Unit sample responses
1. Finite duration Impulse Response (FIR) system
Has a difference equation with N = 0
0

Then,

a k y (n k )

k =0

1
y ( n) =
a0

br x(n r )

r =0
bn
= 0a0

& h( n) {
br x(n r )
r = 0

y (n), is a Convolution sum

n = 0,1,...M
Otherwise

2. Infinite duration Impulse Response (IIR) system

Has a difference equation with N > 0
48

FREQUENCY DOMAIN REPRESENTATION

OF DISCRETE-TIME SIGNALS & SYSTEMS
49

Frequency domain representation

Output of an LSI system is a convolution sum
Steady-state response of an LSI system to a sinusoidal
input is a sinusoid of the same frequency as the input
Amplitude & phase of the output depends on the system

This property makes representation of signals in terms

of sinusoids useful in the study of linear systems
Fourier representation is an integral part of LSI systems

Input sequence can be x(n) = ejn, -<n<

Radian frequency =2f

50

Frequency domain representation

y ( n) =

h( k ) x ( n k ) =

k =

= e
H (e

)=

h ( k )e

k =
jn
j

H (e

j ( n k )

=e

jn

jk
h
(
k
)
e

k =
j

= x ( n) H (e

j k
h
(
k
)
e
also denoted as H ( )

k =

H (e j ) is the Frequency response of the system

Change to amplitude & phase as a function of
H (e

) = H R (e

) + jH I (e

) = H (e

)e

j arg H ( e j )

51

Frequency domain representation

Let
Then

A j j 0 n
(e e
+ e j e j 0 n )
2

j 0
j 0 A j j 0 n
y ( n ) = x ( n ) H (e ) = H (e ) (e e
+ e j e j 0 n )
2

x(n) = A cos(0 n + ) =

= A H (e j0 ) cos(0n + + )
H (e j0 ) is magnitude & = arg H (e j0 ) is phase response at 0
H (e

)=

j
jk
h
(
k
)
e
is
Fourier
series
representa
tion
of
H
(
e
)

k =

Fourier coefficients are the unit sample response, h(k )

52

Frequency domain representation

H (e j ) is periodic with period 2 , because e j ( + 2 ) k = e jk
1
j jn
Taking the invese, h(n) =
H
(
e
)e d

2
Example : Evaluate frequency response of the system
h( k ) =
H ( e j ) =

N 1

e jk

0 n N 1
elsewhere

1,
0,

1 e

j N

sin

j ( N 1)

2 e
2
=
j

e
k =0
sin
2
Magnitude and phase of can be plotted for the specific value of N
=

53

Frequency domain representation

x(n)
X (e j )
FT

Let the input sequence x(n) :

X (e

)=

x(k ) e jk

k =

Then

y ( n ) = H (e

This implies

FT

x(n) =

1
j jn
X
(
e
)e d

1
j
j j n
) x ( n) =
H
(
e
)
X
(
e
)e d

2
1
j j n
(
)e d
=
Y
e

Y ( e j ) = H ( e j ) X ( e j )

Convolution in time domain is

Multiplication in the frequency domain
54

Frequency domain representation

Example : Ideal Discrete time Low pass filter
Frequency response specifications for this filter is
H (e j ) =

1,
0,

co
co <

We can compute h(n) as

1 co jn
sin co n
h( n) =
e d
=

co
2
n
h(n) is not causal since it has got values for n < 0
An ideal LPF is not realizable

55

SAMPLING OF CONTINUOUS-TIME (CT)

SIGNALS
56

Sampling of Analog signals

DT signals are derived from CT signals by sampling
Periodic or Uniform sampling is generally used
Periodic sampling is described by the relation
x(n) = xa(nT),
- < n <
Sampling period T: Time interval between successive samples
Sampling rate Fs = 1/T
With periodic sampling,
t = nT = n/Fs

Relationship between and

Let the analog periodic signal be
xa(t) = A cos(t + ) = A cos(2Ft + )
57

Sampling of Analog signals

On periodic sampling at 1/T samples/ s
xa(t)

= xa(nT) = x(n)
= A cos(2FnT + ) = A cos(2nF/Fs + )
= A cos(2fn + ) = A cos(n + )

f = F/Fs is termed Normalized frequency

This is in Cycles/ sample
= T is in Radians/ sample

58

Sampling of Analog signals

Periodic sampling of a CT signal xa(t) = A cos(2F0t + )
produces DT signal x(n) = A cos(2f0n + ); f0 = F0/Fs
This implies mapping of the large frequency range for F &
to small frequency ranges f (=F/Fs) & (= T)
The ranges of f & are
- < f <
And
- < <

By sampling, the highest values of F & becomes

Fmax = 1/2T And max = /T
This reduction in range introduces the ambiguity termed
Aliasing

59

Sampling of Analog signals

Example of aliasing error
Sampling of CT signals x1(t) = cos 2(10)t and
x2(t) = cos 2(50)t at a sampling rate of Fs=40
Corresponding DT signals are
x1(n) = cos 2(10/40)n = cos (/2)n
x2(t) = cos 2(50/40)n = cos (5/2)n = cos (/2)n

After sampling x1(n) = x2(n)

They become indistinguishable as to whether the samples
correspond to x1(t) or x2(t)
At Fs=40, F2(=50Hz) is an Alias of frequency F1(= 10Hz)
In general F1 + 40k, k = 1, 2, are Aliases of F1
60

Sampling of Analog signals

With no aliasing, the relationship between f0 and F0 is
one to one
It is possible to reconstruct xa(t) from the samples x(n)

There is Aliasing when Fk = F0 + kFs, k = 1, 2, ,

Relationship between x(n) and xa(t) is not one to one
x(n) can represent an infinite number of CT sinusoids xa(t)

Relationship between CT and DT sinusoids is shown in

the figure below

61

Sampling of Analog signals

62

Sampling of Analog signals

Example: For the analog signal xa(t) = 3cos 100t
a) Minimum sampling rate required to avoid aliasing
F = 50 Hz. Hence minimum value of Fs = 2*F = 100 samples/s

b) If the signal is sampled at 200 Hz, what is the DT

signal obtained after sampling?
x(n) = 3 cos (100/200)n = 3 cos (/2)n

c) If the signal is sampled at 75 Hz, what is the DT signal

obtained after sampling?
x(n) = 3cos (100/75)n = 3 cos (4/3)n
= 3 cos (2 - 2/3)n = 3 cos (2/3)n

63

Sampling of Analog signals

d) What is the frequency 0 < F < Fs/2 of a sinusoid that
yields samples identical to those obtained in (c) above
In the above part of the problem f = 1/3 and Fs = 75 Hz
F = fFs = (1/3)75 = 25 Hz
The sinusoidal signal ya(t) = 3 cos 2Ft = 3 cos 50t yields
identical samples
For the sampling rate Fs = 75 Hz,
F = 50Hz is an alias of F = 25 Hz

64

Sampling Theorem
Sampling rate of analog signal should be Fs 2Fmax
To determine Fs, we need only a general information on
the frequency content of the signal
Major speech signals fall below 3 kHz
Frequency content of TV signals is up to 5 MHz

An analog signal can be represented as the sum of N

sinusoids of different amplitudes, frequency and phase
N

xa (t ) = Ai cos(2Fi t + i )
i =1

65

Sampling Theorem
Before sampling, the signal is Low pass filtered to limit
the highest frequency content to Fmax = Fs/2
This is the Anti aliasing filter

Frequency component of xa(t) is mapped into a DT

sinusoid with frequency
- fi = Fi/Fs
OR
- i = 2fi
When Fs 2Fmax, all frequency components in xa(t) are mapped
to the DT frequency components in the fundamental interval
All frequency components in xa(t) get represented in sampled
form without ambiguity

66

Sampling Theorem
When Fs 2Fmax, we can reconstruct the analog
signal without any distortion due to aliasing
Sampling Theorem gives the ideal interpolation formula
for this reconstruction

Sampling Theorem : If the highest frequency contained

in an analog signal xa (t ) is B and the signal is sampled at
a rate > 2 B, then xa (t ) can be exactly recovered from its
sin 2Bt
samples using the interpolation function g (t ) =
2Bt
67

Sampling Theorem
n
Samples of xa (t ) = xa (nT ) = xa
Fs
Convolution of xa (nT ) and g (t ) is the reconstructed signal xa (t )
n

sin 2B t

n
2B
n

xa (t ) = xa g t = xa

n
Fs n = 2 B

n = Fs
2B t

2B
where Fs = 2 B;
(2 B = FN , the Nyquist rate)
Reconstruction requires infinite number of samples x(t ) and g (t )
The reconstruction formula has only theoretical importance
In practice, we use other interpolation techniques
68

Sampling Theorem
Deduction of the Interpolation function
1
xa (t ) =
2

X a ( j ) e

1
xa (nT ) = x(n) =
2
1
=
2
1
=
2

jt

d X a ( j) = xa (t ) e jt dt
FT

X a ( j) e jnT d

r =

( 2 r +1) / T

( 2 r 1) / T

/T

r =

/T

X a ( j) e jnT d

2r jnT j 2rn

X a j + j
d
e e
T

69

Sampling Theorem
Interchanging the order of integration & summation
x ( n)

1
=
2

2r jnT

X
j

+
j
d
e
a
/ T r
T

1
=
2

1
2r jn

Xa j + j
e d
T r
T
T
=
( substituted = T )

/T

But from Discrete time FT

Comparing the expressions

1
x ( n) =
2

X (e j ) e jn d

1
2r

X (e ) = X a j + j

T r = T
T
j

70

Sampling Theorem
Expressing in terms of analog frequency
X (e

jT

1
2r

) = X a j + j

T r =
T

Plots of X a ( j ) and X (e j ) are shown below

0
0T
>
= >
If T is large
OR
2 T
2
2
This results in overlap of CT transform
Upper frequencies in X a ( j ) get reflected to lower frequencies
in X (e j ) & take the identity of low frequencies :

Aliasing
71

Sampling Theorem
1 Xa(j)

FT of CT signal
-0/2

j
1/T X(e )

0/2

FT of the DT signal (large T)

-2

-0T/2

-2

0T/2

1/T
-4

-
-0T/2

X(ej)

0T/2

FT of the DT signal (small T)

2

72

Sampling Theorem
0
0T
<
= <
OR
T
2
2
2
This corresponds to sampling rate of at least twice the

There is no overlap if

highest frequency cmponent of X a ( j ) : Nyquist rate

Now

X (e

) = X (e

jT

) = X a ( j ),

T
T
T

The relation between xa (t ) and its FT X a ( j) is

1
xa (t ) =
2

X a ( j) e jt d

73

Sampling Theorem
0
When
< , the range of integration can be modified
2
T
1 /T
1 /T
jt
jT
jt
xa (t ) =
X
(
j

)
e
d

=
TX
(
e
)
e
d
a

/
T

/
T

2
2
By definition,

X (e

) = X (e

jT

)=

xa (nT )e jnT

n =

Substituting for X (e jT )
T /T
jnT jt
xa (t ) =
xa (nT )e
e d

/
T

2
n =

74

Sampling Theorem
Interchanging the order of integration and Summation
xa (t )

T / T j(t nT )

x
(
nT
)
e
d

a
2 / T

n =
sin

n =

xa (nT )

The interpolation function is g (t ) =

(t nT )

sin

Substituting T =

1
1
=
;
Fs 2 B

(t nT )

(t )

(t )
g (t ) =

sin 2Bt
2Bt
75

Sampling Theorem
Examples:
1. What is the Nyquist rate for the signal
xa(t) = 3cos 50t + 10sin 300t - cos 100t ?
Fmax = 150 Hz,

FN = 2* Fmax = 300 samples/s

When 10 sin 300t is sampled at 300 Hz, we get the

sample values as 10 sin n
All samples will have the same value, which could be zero
To avoid this situation analog signal is sampled at a rate higher
than the Nyquist rate
76

Sampling Theorem
2. If we sample the following analog signal at 5 kHz
xa(t) = 3 cos2000t + 5 sin6000t + 10 cos12000t
What is the DT signal obtained after sampling?
Nyquist rate for this signal: FN = 2* Fmax = 12kHz
On sampling this signal at 5 kHz,
x(n) = 3 cos 2(1/5)n + 5 sin 2 (3/5)n + 10 cos 2 (6/5)n
= 3 cos 2(1/5)n + 5 sin 2 (1 - 2/5)n + 10 cos 2 (1 + 1/5)n
= 3 cos 2(1/5)n - 5 sin 2 (2/5)n + 10 cos 2 (1/5)n
= 13 cos 2(1/5)n - 5 sin 2 (2/5)n

77

Sampling Theorem
3. What is the analog signal ya(t) that we can reconstruct
from the above samples, using ideal interpolation?
Frequency components at 1 kHz and 2 kHz only are present in
the sampled signal
Analog signal reconstructed using ideal interpolation is
ya(t) = 13 cos 2000t - 5 sin 4000t
ya(t) is different from the original analog signal xa(t)
The distortion is the result of aliasing effect - Problem with
low sampling rate

78