Analysis of Signals and Systems

Sampling
Dr. David Antonio-Torres

Objectives
To introduce the concept of sampling and the definition of the
sampling theorem

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Signals and Systems

Contents
Introduction
Periodic sampling
The sampling theorem
Interpolation
The effects of aliasing

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Signals and Systems

Introduction
The analog-to-digital conversion (ADC) is the preceding stage
for the processing of continuous-time signals in the discretetime domain
Once the continuous-time signal is available in discrete time,
the signal can be processed by a digital signal processor
(DSP)
Analog-to-Digital Conversion
xa(t)

Continuous-Time
Signal

Sampler

x[n]

Discrete-Time
Signal

Quantizer

x q[n]

Quantized
Signal

Coder

1011

Digital
Signal

Among the benefits of digital signal processing are: cheaper,

more tolerant to noise and reprogrammable
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Signals and Systems

Periodic Sampling
Periodic sampling consists in the multiplication of a
continuous-time signal with the comb function

p t

t nT

xp t
X p j

X p

x nT t nT

1
X j P j
2

1
j
T

X j k

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Signals and Systems

Periodic Sampling
In periodic sampling, the
following is assumed
x(t) is a band-limited signal
with maximum frequency of
M
The condition S>2M is
met to avoid overlapping
between the replicas of the
frequency spectrum of x(t); if
overlapping is allowed, x(t)
could not be recovered

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Signals and Systems

Sampling Theorem
The conditions just discussed constitute the sampling
theorem:
Let x(t) be a band-limited signal with x(j)=0, for ||
>M. Then x(t) is uniquely determined by its samples
x(nT) for n=0, 1, 2, , if S > 2M where S = 2/T
Given these samples represented by x(nT), x(t) can be
recovered when its samples are processed through an
ideal low-pass filter with gain T and cut-off frequency in
the interval M < C < (S M )
The frequency 2M, which must be exceeded by the
sampling frequency S is commonly referred to as the
Nyquist rate (M is also referred to as the Nyquist
frequency)
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Signals and Systems

Sampling Theorem
The sampling theorem can be further explained by the
diagrams and waveforms below

Definition of the System

Signal Processing
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Signals and Systems

Sampling with Zero-Order Hold

The practical implementation of periodic sampling is
commonly conducted with a zero-order hold (or sample-andhold, S/H)
With a zero-order hold, the amplitude of the sample is
sustained for T seconds

x(t)

Zero-Order xp(t)
Hold

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Signals and Systems

Sampling with Zero-Order Hold

The sampling with zero-order hold
can
be
modeled
as
the
multiplication of the continuoustime signal with the periodic train
of
unit
impulses,
serially
connected to a continuous-time
system
whose
unit-impulse
response is the shifted unit pulse
p(t)

x(t)

xp(t)

h0(t)
1

x o(t)
0

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Signals and Systems

Interpolation
Interpolation is the process of recovering a signal from their
samples; the signal has been sampled by periodic sampling
Interpolation is necessary when the signal needs to be
recovered in its continuous-time form
Three interpolation methods are discussed next
Band-limited
Zero-order
First-order

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Signals and Systems

Band-Limited Interpolation
This is the interpolation implemented by the ideal LPF
xr t x p t h t
xr t

x nT h t nT

h t T

c
t
sinc c

c s 2

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Signals and Systems

Zero-Order Interpolation
This is the interpolation implemented by the zero-order hold,
in which the samples are interpolated by horizontal lines
It can be demonstrated that the Fourier transform of the zeroorder hold is the multiplication of a complex exponential and a
sinc function

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Signals and Systems

First-Order Interpolation
For this interpolation, the
output of the multiplier is
connected to an LTI system
whose unit-impulse response
is the tri function
It can be demonstrated that the
Fourier transform of the tri
function is the sinc square

p(t)

x(t)

xp(t)

h(t)

xp(t)

x r(t)

-T

-T

h(t)

xr(t)

2T

-T

2T
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Signals and Systems

Aliasing
One of the conditions of the sampling theorem is that the
sampling frequency is highly enough to accommodate the
replicas of the spectrum of the signal without overlapping
If overlapping is allowed, the ideal LPF will not be able to
recover the original signal, but a different signal
Because of this erroneous reconstruction, the overlapping is
also referred to as aliasing
Aliasing is demonstrated with a single cosine wave in the
following slides
The reconstruction of the signal will be conducted with an
ideal LPF with a cut-off frequency of c = s/2

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Signals and Systems

Aliasing
Sampling without aliasing
X ( j )

x (t )=cos ( 0 t )

s
0 = ;
6
s

s 0

s
2

s
2

0 s
2

s
( s 0 )

s
( s 0 )

x r (t)=cos (0 t)=x (t )

2 s
0 =
;
6

x r (t )=cos(0 t )=x(t )

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Signals and Systems

Aliasing
Sampling with aliasing

( s 0 )

( s 0 )

s 0
2

s
2

s
4 s
0 =
;
6

xr (t )=cos( s 0)t x (t )

0 s

5 s
0 =
;
6

xr (t )=cos( s 0)tx (t)

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Signals and Systems

Conclusions
Sampling is the first step of the analog-to-digital conversion
(ADC); ADC is the process of converting a continuous-time
signal into a digital signal
Sampling produces discrete-time signals
The conditions to recover a sampled signal from its samples
are established by the sampling theorem
The recovery of a signal from its samples is referred to as
interpolation; we studied three types of interpolation: bandlimited, zero-order and first-order
When the sampling frequency does not comply with the
sampling theorem, the recovery of a signal different from the
original one, aliasing, occurs
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Signals and Systems