Sampling Theorem

Spring 2009
Ammar Abu-Hudrouss Islamic
University Gaza

Continuous Versus Digital

Analogue electronic systems are continuous
x(t)

y(t)

Analogue System

Electronic System are increasingly digitalized

x(t)

A/D

x(n)

y(n)
Digital System

D/A

y(t)

Signals are converted to numbers, processed, and converted back

Digital Signal Processing

Slide 2

Sampling Theorem
Use A-to-D converters to turn x(t) into numbers x[n]
Take a sample every sampling period Ts uniform sampling

Digital Signal Processing

Slide 3

Advantages of Digital over Analogue

Advantages

Flexibility (simply changing program)

Accuracy
Storage
Ability to apply highly sophisticated algorithms.

Disadvantages

It has certain limitations (very fast sample rate is needed when

the bandwidth of signal is very large)
It has a larger time delay compared to the analogue.

Digital Signal Processing

Slide 4

Classification of signals

Mono-channel versus Multi-channel

One Dimensional versus Multidimensional

Continues time versus Discrete time

Continuous values and Discrete Valued

Deterministic versus random

Digital Signal Processing

Slide 5

Periodic Continuous Signal

We will take sinusoidal signals for example. Continuous sinusoidal
signal has the form

x(t ) A cost
The signal can be characterised by three parameters
A: Amplitude, frequency in radian and : phase
The period is defined as

1 2

Digital Signal Processing

Slide 6

Periodic Continuous Signal

In analogue signal, increasing the frequency will always lead to
increase the rate of the oscillation.

7
Digital Signal Processing
Slide 7

Periodic Discrete Signal

Discrete sinusoidal signal has the form

x ( n ) A cos n
1) Discrete time sinusoid is periodic only if its frequency in hertz ( f =
/ 2) is a rational number
From the definition of a periodic discrete signal

x(n) x(n N )
cos(2fn ) cos(2fn 2fN )
This is only true if

2 fN 2 k
k
f
N

k 0 , 1 , 2 ,......
8

Digital Signal Processing

Slide 8

Periodic Discrete Signal

2) Discrete time sinusoid whose radian frequencies are separated
by integer multiples of 2 are identical
To prove this, we start from the signal

x(n) A cosn

A cos(( 2 )n ) A cos(n ) x(n)

As a result, all the following signals are identical

xk (n) A cos( k n ) k 2k k 0,1,2,......

3) All signal in the range - <= < are unique.
So the range of the discrete frequency f is [-0.5 0.5]

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Digital Signal Processing
Slide 9

10
Digital Signal Processing
Slide 10

Analogue to Digital Conversion

xa(t)

x(n)

Sampler
Analog
Signal

xq(n)

Quantizer

Discrete-time
Signal

Coder

Quantized
Signal

101101

Digital
Signal

1) Sampling: Conversion of analogue signal into a discrete

signal by taking sample at every Ts s.
2) Quantization: Conversion of discrete signal into discrete
signals with discrete values. (the value of each sample is
represented by a value selected from a finite set of
possible value)
3) Coding: is process of assigning each quantization level a
unique binary code of b bits.
Digital Signal Processing
Slide 11

Sampling of Analog Signal

We will focus on uniform sampling where

X(n) = xa(nTs)
- < n <

Fs = 1/Ts is the sampling rate given in sample per second

As we can see the discrete signal is achieved by replacing the

continuous variable t by nTs.

Consider the analog signal Xa(t) = A cos(2Ft + )

The sampled signal is
Xa(nT) = A cos(2FnTs + )
X(n) = A cos(2fn + )
The digital frequency = analog freq. X sampling time
f = FTs

Digital Signal Processing

Slide 12

Sampling of Analog Signal

But from previous discussion , for the analoge frequency

-< F < or -< <

And for the digital frequency

-0.5 < f < 0.5 or - < <

From the above argument the infinite analog frequency is

mapped into finite digital frequency.

This mapping is one-to-on as long as the resultant digital

frequency is between the limits of [-0.5 o.5]

Digital Signal Processing

Slide 13

Sampling of Analog Signal

Which leads that

-1/2< FTs <1/2 or - < Ts <
OR
-1/(2Ts) < F < 1/(2Ts) or - /Ts < < /Ts
Hence that highest possible analoge frequency is
Fmax = Fs/2 = 1/(2Ts) and < Fs = /Ts

Digital Signal Processing

Slide 14

Sampling of Analog Signal

Example
Consider the two analog sinusoidal signals
X1(t) = cos 2(10)t and X2(t) = cos 2(50)t
Both are sampled with sampling rate Fs = 40, find the corresponding
discrete sequences

X1(n) = cos 2(10/40)t = cos (n/2)

X2(t) = cos 2(50/40)t = cos (5n/2) = cos (n/2)

1Hz and a 6Hz sinewave are sampled at a rate of 5Hz.

Digital Signal Processing

Slide 15

Sampling of Analog Signal

All sinusoids with frequency

Fk = F0 + k Fs, k= 1,2,3,
Leads to unique signal if sampled at Fs Hz.

proof
xa(t) = cos (2 Fk t + ) = cos (2 (F0 + k Fs )t +)
x(n) = xa(nTs) = cos (2 (F0 + k Fs )/Fs t +)
= cos (2 F0/Fs n + 2 k n +)
= cos (2 F0/Fs n +)

Digital Signal Processing

Slide 16

Sampling Theorem

Sampling Theorem
A continuous-time signal x(t) with frequencies no higher than
fmax (Hz) can be reconstructed EXACTLY from its samples x[n] =
x(nTs), if the samples are taken at a rate fs = 1/Ts that is
greater than 2fmax.
Consider a band-limited signal x(t) with Fourier Transform X()

Digital Signal Processing

Slide 17

Sampling Theorem

Sampling x(t) is equivalent to multiply it by train of impulses

Digital Signal Processing

Slide 18

Sampling Theorem

In mathematical terms

x (n) x (t ) s (t )

x (n) x(t ) (t nTs )

n

Converting into Fourier transform

1
X ( ) X * ( ns )
Ts n
1
X ( )
X ( ns )
Ts n
Digital Signal Processing
Slide 19

Sampling Theorem

By graphical representation in the frequency domain

Digital Signal Processing

Slide 20

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Sampling Theorem

Therefore, to reconstruct the original signal x(t), we can use an

ideal lowpass filter on the sampled spectrum

This is only possible if the shaded parts do not overlap. This

means that fs must be more than TWICE that of B.

Digital Signal Processing

Slide 21

Sampling Theorem
Example
x(t) and its Fourier representation is shown in the Figure.
If we sample x(t) at fs = 20,10,5

1) fs = 20
x(t) can be easily
recovered by LPF

Digital Signal Processing

Slide 22

11

Sampling Theorem

2) fs = 10
x(t) can be recovered
by sharp LPF

3) fs = 5
x(t) can not be
recovered
Compare fs with 2B in each case

Digital Signal Processing

Slide 23

Anti-aliasing Filter

To avoid corruption of signal after sampling, one must ensure

that the signal being sampled at fs is band-limited to a
frequency B, where B < fs/2.
Consider this signal spectrum:

After sampling:

After reconstruction:

Digital Signal Processing

Slide 24

12

Anti-aliasing Filter

Apply a lowpass filter before sampling:

x(t)

Anti-aliasing
filter

x'(t)

y(n)
Sampler

Now reconstruction can be done without distortion or corruption

to lower frequencies:

Digital Signal Processing

Slide 25

Homework
Students are encouraged to solve the following questions from
the main textbook
1.2, 1.3, 1.7, 1.8, 1.9, 1.11 and 1.15

Digital Signal Processing

Slide 26

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