LINEAR SYSTEM

AND
SAMPLING PROCESS
UPON COMPLETION OF THIS CHAPTER,
STUDENTS SHOULD BE ABLE T0:
COMPREHEND WHAT IS LINEAR SYSTEM AND
LINEAR TIME INVARIANT
KNOW THE SAMPLING PHENOMENA
DEVELOP THE MATHEMATICAL MODEL OF THE
SAMPLING


Linearity of the System

Given a system x(n) y(n), if Kx(n)Ky(n) and K
1
x
1
(n)+K
2

x
2
(n) Ky
1
(n) + K
2
y
2
(n), then the system is linear, provided
y(n) is the zero state response due to an input signal x(n) or
the zero input response due to initial states.

The zero state response is the response due to input only,
and the zero
input response is due to the initial states (initial conditions)
only. input response is due to the initial states (initial
conditions) only.


Time-Invariant System

Let us represent the solution to this equation as the
output y(n) when an input x(n) is applied. Such a system is
said to be time-invariant if the output is y(n N) when the
input is x(n-N), which means that if the input sequence is
delayed by N samples, the output also is delayed by N
samples.
For this reason, a time-invariant discrete-time system is
also called a shift-invariant system.

WHAT IS TIME-INVARIANT ?
Linear Time Invariant System
(in continuous time, t)

input u(t)
output y(t)
state x(t)

system described by impulse response function h(t)
Y(t) = h(t) * u(t) where * denotes the convolution operation
Taking Laplace Transforms of signals to get U(s), Y(s) and H(s),
Convolution (in time domain) involves integrals
Calculation in Laplace ("s") or Fourier ("frequency") domain, involving
multiplication, is much easier


Sampling phenomena

Quantizing
The AD-converter (analog-digital) converts an analog signal y
a(t
), which can
be a voltage signal from a temperature or speed sensor, to a digital signal
y
d
(t
k
) in the form of a number to be used in operations in the computer.


Sampling phenomena
The digital signal is represented internally in the computer by a number of
bits. One bit is the smallest information storage in a computer. A bit has
two possible values which typically are denoted 0 (zero) and 1 (one).
Assume that y
a
(t) has values in the range [Y min ,Y max ] and that the AD-
converter represents y
a
in the given range using n bits. Then y
a
is
converted to a digital signal in the form of a set of bits:
Each bit b
i
has value 0 or 1. These bits are interpreted as weights or
coecients in a number with base 2:


b
0
is the LSB (least significant bit), while b
n1
is the MSB (most significant
bit).


Sampling phenomena
Let us see how the special values Y min and Y max are represented in
the
computer.
Assume that n = 12, which is typical in AD-converters. y
d
= Y min
is then represented by:





where sub index 2 means number base 2.
Sampling phenomena
y
d
= Y max is represented by:






The resolution q, which is the interval represented by LSB, is:



For a 12-bit AD-converter with Y
max
= 10V and Y
min
= 0V, the resolution is:



Variations smaller than the resolution may not be detected at all by the
AD-converter.
Figure shows an example of an analog signal and the corresponding
quantized signal for n = 12 bits and for n = 4 bits in the AD-converter. The low
resolution is clear with n = 4.


Figure 3.:
Analog signal and the
corresponding quantized
signal for n = 12 bits (up
to 15s) and for n = 4 bits
(after 15 s) in the
AD-converter.


Aliasing

A particular phenomenon may occur when a continuous-time signal is
sampled.
Frequency components (i.e. sinusoidal signal components) in the analog
signal may appear as a low-frequent sinusoid in the digital signal.
This phenomenon is called aliasing , and it appears if the sampling
frequency is too small compared to the frequency of the sampled signal.
In order to reconstruct a continuous signal from the sampled sequence,
the sampling frequency must verify the condition (Nyquist's theorem):


The frequency f
s
=2f
N
is a theoretical limit; in practice, a higher
sampling frequency must be chosen.
, f
N
is the maximum frequency to be transmit 2
s N
f f >
Figure 4: A continuous-time sinusoid y(t) = sin 2tt and the sampled
sinusoids for two dierent sampling intervals
The discrete signal has a dierent (lower) frequency than the continuous-
time signal, see Figure 4. Thus, there is aliasing.
Example
1. Sampling interval h = 0.2s corresponding to sampling frequency


The discrete signal has the same frequency as the continuous-time signal.
Thus, there is is no aliasing.

2. Sampling interval h = 0.8s corresponding to sampling frequency


3. From the above example ,what is the frequency of the resulting signal, f
dis
?
Figure 6 shows how to find f
dis
. The result is

Aliased frequency components may cause problems in control applications.
One way of reducing the aliasing problem is to pass the continuous-time signal
through a lowpass filter before sampling.
This anti-aliasing filter should be an analog filter which may be implemented
using electronic components as resistors, capacitors and operational amplifiers.
Sampled-Data Control Systems
The closed-loop system is to track the aircraft shown
automatically.
( ) ( ) ( )
A R
e t t t u u =
is the yaw-axis pointing angle of the antenna,
and O
A
(t) is the angle to the aircraft.
( )
R
t u
( )
R
t u
In general, it is undesirable to apply a signal in sampled form,
such as a train of narrow rectangular pulses.
Therefore, a data- reconstruction
device, called a data hold, is
inserted into the system directly
following the sampler.
The simplest datareconstruction device is the zero-order hold.
Operation of a sampler zero-order hold combination is described by the
signals shown in following Figure
The signal can be expressed as:



( ) e t
( ) (0)[ ( ) ( )] ( )[ ( ) ( 2 )]
(2 )[ ( 2 ) ( 3 )] ............
e t e u t u t T e T u t T u t T
e T u t T u t T
= +
+ +
Where is the unit-step function. The Laplace transform of is ,
given by:
( ) u t ( ) e t ( ) E t
2 2 3
2
0
1
( ) (0) ( ) (2 ) ............
1
(0) ( ) (2 ) ...
1
( )
Ts Ts Ts Ts Ts
Ts
Ts Ts
Ts
nTs
n
e e e e e
E s e e T e T
s s s s s s
e
e e T e e T e
s
e
e nT e
s

=
( ( (
= + + +
( ( (

(
( = + + +
(


( (
=
( (

The first factor in this expression reperesent

the function of the input signal e(t) and the
sampling period T
This part considered to be a
transfer function
So that
THE IDEAL SAMPLER
The inverse Laplace transform of E* (s) is:






can be expressed as:

In this form it can be readily seen
that is the carrier in the
modulation process, and e(t) is the
modulating signal.

0
( ) ( ) ( 2 ) .............
( ) ( )
T T T
T
n
t t T t T
t t nT
o o o
o o

=
+ + +
=

*
( ) e t
*
( ) ( ) ( ) ( ) ( ) ( ) ( ) .............
(0) ( ) ( ) ( ) .............
e t e t t e t t e t t T
e t e T t T
o o o
o o
= = + +
= + +
( )
T
t o
e(0) e(T) e(2T) e(3T)
e
*
(t)
0 T 2T 3T t
The output signal of an ideal sampler is defined as the signal
whose Laplace transform is

*
0
( ) ( )
nTs
n
E s e nT e

=
=

Representations of the ideal sampler.
note that the definition of the sampling operation together with
the zero-order-hold transfer function defined by:
1
( )
Ts
e
G s
s

=
Example:
Example:
For example we had a peak at 10 rads/sec and our sample time was 0.1
seconds,
Sampling frequency in rad/s,

The amplitude of frequency response




2 2
62.83 /
(0.1)
S
rad s
T
t t
e = = =
2
S
u
e
t
~
10
2
62.83
2
10
. : (2 ) 1 / 57.3
62.83
S
x
rad s
u
e
t
u
t
u t
=
| |
=
|
\ .
| |
= = =
|
\ .
Laplace transfer function of ZOH
The equivalent Laplace transfer function of the ZOH block :
1
( )
Ts
ZOH
e
H s
s

=
To analyse the frequency
response of an s-domain
function we simply replace the s
by jw.
The ZOH becomes: