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Chapter 1

Introduction:
Signal Sampling
and Reconstruction
EE4L005 DIGITAL CONTROL SYSTEMS
N. C. Sahoo
Introduction
• Digital computer is an integral component in all digital
control systems. When digital computer is used as
controller for a physical system, the process of signal
conversion is essential so that the digital and analog
components can be interfaced in the same system.
• For instance, the output of an analog device must first
undergo an analog-to-digital (A/D) conversion before
they can be processed by a digital controller.
• Similarly, the coded signal from a digital controller
must be decoded by a digital-to-analog (D/A)
converter before processing by an analog device.
• Sample-and-Hold (S/H) device is interfaced very
often.
Sampled-Data Control Systems

General structure of a Multivariable Control System


• Plant is generally a continuous-time system whose
physical characteristics are beyond control. The
feedback of output information results in a closed-
loop system. In open-loop, the controller operates on
some pre-set pattern without feedback of outputs.
• The plant accepts continuous-time signals as inputs
and gives out continuous-time signals as outputs.

• If the controller elements are such that the controller


produces continuous-time control signals from
continuous-time input signals (analog controller),
then the overall control system is a continuous-time
system, wherein the signal at every point is a
continuous function of time.

• A digital controller, in which a digital computer forms


the heart, is nowadays very common because of
advances in technology.
 The notable advantages of digital controller are:

 Increased flexibility of control programs,


 Excellent decision-making or logic capability of
digital systems,
 Better reliability,
 Less effect due to noise and disturbance,
 Less cost and weight.

• The program which characterizes a digital controller


can be modified to accommodate design changes or
adaptive performances, without any variations in the
hardware.
• Digital controllers have the inherent characteristics
to accept the data in the form of short duration
pulses, i.e., sampled or discrete-time signals,
and producing similar kind of control signal.

A typical sampled-data control system


w(t) = disturbance to the plant; v(t) = sensor
measurement noise
• In digital control system, digital words arrive with the
same fixed time period T, called the sampling
period.
• In practice, some advanced control systems may
have varying sample periods and/or different
periods in different feedback paths.
• Typically, there is a clock as part of the digital logic
which supplies pulses every T seconds and the A/D
converters send digital words to the computer each
time the pulse arrives.
• Digital words are m(kT) {written as m(k)}.
Numerically coded output u(k) of digital computer is
decoded into a continuous-time signal u(t) by D/A
converter, which controls the plant.
• The overall system is hybrid, in which the signal is in
a sampled form in the digital controller and in a
continuous form in the rest of the system.
• A system of this kind is referred to as a sampled-
data control system.

Sample-and-Hold Operation:
• A sampler is a device that converts an analog signal
into a train of amplitude-modulated pulses.
• A hold device simply maintains the value of the
pulse for a prescribed time duration.
• Mostly, sample and hold functions are performed by
a single device, known as sample-and-hold (S/H).
• The S/H operates by storing input signal voltage as
charge on a small, high quality capacitor. The
actual design of an S/H element usually consists of
a capacitor, an electronic switch, and operational
amplifiers as shown.
• OPAMPs are needed for isolation. The capacitor
and switch cannot be connected directly to the
analog circuitry because of capacitor’s effect on the
driving waveform.
• When the switch is closed, the capacitor rapidly
charges to Vin andVout is equal to Vin approximately.

• When the switch opens, the capacitor retains its


charge so that the output holds at Vin

• If the input voltage changes rapidly while the switch


is closed, the capacitor can follow this voltage
because the charging time constant is very small.
• If the switch is suddenly opened, the capacitor
voltage represents a sample of the input voltage at
the instant the switch was opened. The capacitor
then holds this sample until the switch is again
closed and a new sample taken.

• Although the S/H is available as one unit, it is helpful


to treat the sampling and holding operations
separately for analysis, as shown.

The sampling process is equivalent to multiplying the


signal r(t) with a periodic pulse train pT ,Δ (t ) to produce
the sampled signal r(t) pT ,Δ (t ) .

• The pulse/sampling duration is  and the sampling


period is T.
Sampling and holding operations
D/A and A/D Converters:

Operations performed by (a) DAC (b) ADC

• The D/A converter may be regarded as a device


which consists of a decoder and a S/H unit
• The decoder decodes the digital word into a number.
The transfer function of the decoder is simply a
constant gain (ideally unity).
• Similarly, the A/D converter performs the operations
of S/H, quantization, and encoding when a signal is
to be converted from analog to digital.
• The sampled signal is held till the conversion is
complete. The holding operation thus reduces the
effect of signal variation during conversion.
• If the resolution of ADC is very high, the nonlinear
effect of quantizer can be neglected and since the
decoder and encoder transfer functions are constant
gains, the DAC and ADC essentially reduce to S/H
units for analytical studies of digital system.

• Thus, for analytical studies, the sampled-data control


system can be represented by block diagram as
shown {assuming w(t) = v(t) = 0}.
Mathematical Model of S/H Operation
• Let us first consider the hold operation.

Zero-Order Hold:
Problem Formulation:
• Given the sequence u(0), u(T), …, u(kT), …, we
have to construct u(t), t  0.
• This data reconstruction may be regarded as an
extrapolation process, since the continuous-time
signal is to be constructed from the information
available only at past sampling instants.
• The most common method used for this is
polynomial extrapolation using Taylor’s series.
• By Taylor’s series expansion of u(t) around t = kT ,
u(kT )
uk (t )  u (kT )  u (kT )(t  kT )  (t  kT )  
2
2!
du(t ) d 2 u (t )
where u (kT )  (kT ) 
and u 2
dt t kT dt t  kT

u k (t ) is the corrected version of u(t) for the k-th


sampling period, i.e.,
uk (t )  u (t ) for kT  t  (k  1)T

u k (t ) will be used to represent the output of the data


hold. Since u(t) enters the data hold only in sampled
form, the derivative values are not known.
• However, the derivatives may be approximated by
backward difference:
1
u (kT )  u (kT )  u[( k  1)T ]
T
1
u(kT )  u (kT )  u[( k  1)T ]
T
1
 2 u (kT )  2u[( k  1)T ]  u[( k  2)T ] and so on.
T
• If only the first term in the series expansion is used,
the data hold is called a zero-order hold (ZOH).
• Here, it is assumed that the function u(t) is
approximately constant within the sampling interval at
a value equal to that of the function at the preceding
sampling instant.
• Therefore, for a ZOH,
uk (t )  u (kT ) for kT  t  (k  1)T
• If first two terms in the series expansion are used,
the data hold is called a first-order hold.
• It is obvious that higher the order of the derivative to
be approximated, the larger will be the number of
delay pulses required. The time-delay adversely
affects the stability of feedback control systems.
• Furthermore, a high-order extrapolation also requires
complex circuitry and results in a high data
construction cost. The ZOH is the simplest and most
widely used data hold device.
Sample and ZOH:

ZOH Input ZOH Output


• ZOH clamps the output signal to a value equal to
that of the input signal at the sampling instant.
• If the sampling duration  is very much smaller than
the sampling period T and the smallest time-
constant of the input signal e(t), then the output of
the sampler can be approximated by a sequence of
flat topped pulses.
• Sampling operation is thus described by:

e(t ) pT , (t )   e(kT ) p (t  kT )
k 0
• ZOH output pulse appearing at the k-th sampling
instant can be expressed as
e(kT )[u s (t  kT )  u s (t  k  1T )]
where u s (t ) is the unit-step function.
• ZOH output can be written as:

e (t )   e(kT )[u s (t  kT )  u s (t  k  1T )]
k 0

Taking the Laplace transform,


  skT   s ( k 1)T 
E ( s)   e(kT )  
k 0  s 
1   sT     skT 
=     e(kT ) 
 s   k  0

where  (= 2.71828) is the Euler number {a


different symbol is used so as to distinguish it
from the input signal e(t)}.
• In this Eq., the first factor is independent of e(t).
Thus, the operation can be represented as shown

The function E * ( s ) , called the starred transform,


is defined as:

E * ( s)   e(kT )  skT
k 0

• It should be noted that E * ( s ) is not present in


physical system, but appears in mathematical
modeling.
• This model of sampler does not model a physical
sampler and the block does not model a physical
data hold. However, the combination does
accurately model a sampler/ZOH device.

Representation of Sampler and ZOH


• It is observed that the output of the sampler is a
function of e(kT) only at t = kT, k = 0, 1, 2, … .
• Thus, the operation symbolized by the sampler
cannot be represented by a transfer function
because many different input signals can result in
the same output signal E * ( s ).
• This property of the sampler complicates the analysis
of sampled-data systems.
Ideal Sampler:
• Inverse Laplace transform of E * ( s ) is:

e* (t )   1[ E * ( s)]  e(0) (t )  e(T ) (t  T )  e(2T ) (t  2T )  



  e(kT ) (t  kT )
k 0
where  (t ) is the unit-impulse function.

• Then e* (t ) can be viewed as the train of impulse


functions whose weights are equal to the values of
the signal at the sampling instants.
• This means that the sampler that appears in this
model is an “impulse modulator” with the carrier
signal 
 T (t )    (t  kT )
k 0
and modulating signal e(t).
• The schematic representation of the modulation
process and the impulse train is as shown.
• It is again emphasized that this definition of the
sampling process together with the ZOH transfer
function
1    sT
Gho ( s ) 
s
yields the correct mathematical description of the
sampler/ZOH operation.
Example: Determine E * ( s ) for e(t )  u s (t ) .

Solution: For the unit step signal,


e(kT )  1, k  0, 1, 2, .

E ( s)   e(kT ) skT  e(0)  e(T ) sT e(2T ) 2 sT  
*
k 0

1   sT
 2 sT

1
 ,   sT
1
1   sT

Evaluation of E * ( s ) :
• This type of evaluation of E * ( s ) has limited usefulness
in analysis because it is expressed as an infinite
series.
• However, for many useful time functions, E * ( s ) can be
expressed in closed form.

• In addition, there are also other forms of ( s ) that


E *

are useful.
We know that the inverse Laplace transform of E * ( s ) is:
e* (t )  e(t ) T (t )
*
If Laplace transform of e (t ) is done by complex
*
convolution integral, two more expressions of ( s )
E
can be derived. These two expressions are:

 1 
E ( s)   residues of E ( )
*
T ( s  ) 
at poles  1   
of E (  )
1  e ( 0)
E ( s )   E ( s  jn s ) 
*
T n 2
where  s is the radian sampling frequency, i. e.,
 s  2 T
Example: Determine E * ( s ) given that
1
E (s) 
( s  1)( s  2)
Solution:
1 1
E ( ) 
1   T ( s  ) (  1)(  2)(1   T ( s  ) )
Then,
(  1)
E (s) 
*
T ( s  )

(  1)(  2)(1   )  1
(  2)
(  1)(  2)(1   T ( s  ) )  2

1 1
 
1   T ( s 1) 1   T ( s 2)
Properties of E * ( s ) :
Two important s-plane properties are given below.
Property 1: E * ( s ) is periodic in s with period j s .

kT ( s  jms )
Proof: E ( s  jm s )   e(kT )
*
k 0

Since  sT  2 , and from Euler’s identity


 jkmTs
  1, m is an integer. Thus,

E ( s  jm s )   e(kT ) kTs  E * ( s)
*
k 0
Property 2:
If E (s ) has a pole at s  s1 , then E * ( s ) must have poles
at s  s1  jm s , m  0,  1,  2, .
Proof:
Assuming e(t) to be continuous at all sampling instants,

1
E * ( s)   E ( s  jns ) 
T n
1  E ( s)  E ( s  js )  E (s  j 2s )   E (s  js )  
 
T  E ( s  j 2s )  
If E (s ) has a pole at s  s1 , then the above expression
will contribute a pole at s  s1  jm s ,where m is an
integer.
It is important to note that no equivalent statement
can be made concerning the zeros of E * ( s ) ; that is
the zero locations of E (s ) do not uniquely determine
the zero locations of E * ( s ) .

However, the zero locations are periodic with period j s


; as noted in property 1.
• The primary strip in the s-plane is defined as the
strip for which
 s / 2    s / 2
• It should be noted that if the pole-zero locations are
known for E ( s ) in the primary strip, then all the
pole-zero locations in the entire s-plane are also
known.
Example: Let us consider two signals as:

e1 (t )  cos 1t e2 (t )  cos 31t

• These two signals have the same starred transform.


• Both the signals have the same starred transform,
since the two signals have the same value at each
sampling instant. It is to be noted that  s  41
s s
E1 ( s)  (cos 1t )  2 
s  1 ( s  j1 )( s  j1 )
2

Thus, E1 ( s ) has a pole at s  j1 .


s s
E2 ( s)  (cos 31t )  2 
s  91 ( s  j 31 )( s  j 31 )
2

Thus, E 2 ( s ) has a pole at s   j 31  j (1   s ) .


The other pole of E1 ( s ) occurs at s   j1 , and the
other pole of E 2 ( s ) occurs at s  j 31  j (1   s ) .
*
Frequency Spectrum of E ( s ) :
Fourier Transform of sampled signal:

   
  

 e (t )  E ( j )  e(t )   (t  kT )  e(t )   (t  kT )
* *
 k 0   k  
since e(t )  0 for t  0 .
Using complex convolution property of Fourier
transform,
1   
E ( j ) 
*
E ( j ) *    (t  kT )
2 k  
   
   (t  kT )   s   ( j  jn s )
k   n

1
 E ( j ) 
*
E ( j ) * s   ( j  jns )
2 n 

E ( j ) *  ( j  jn s )  E ( j  jn s )

1
 E ( j )   E ( j  jns )
*

T n

Hence, the effect of ideal sampling is to replicate the


original spectrum centered at integral multiples of  s .
• An ideal filter is a filter with unity gain in the
passband and zero gain in outside the passband. Of
course, such a filter is not physically realizable.

• From Fig. (b), an ideal low-pass filter can completely


recover E ( j ) ,i.e., e(t), if the bandwidth of the filter
were  s 2 ,for the case that the highest frequency
present in E ( j ) is less than  s 2 . This is
essentially the Shannon’s Sampling Theorem.
Shannon’s Sampling Theorem:
A function of time e(t ) which contains no frequency
components greater than f 0 Hz is uniquely determined
by the values of e(t ) at any set of sampling points
spaced 1 (2 f 0 ) seconds apart.
• Suppose, in Fig. (b), that  s is decreased until the
highest frequency components present in E ( j ) are
greater than  s 2 . Then, E ( j ) has the amplitude
spectrum shown in Fig. (c). For this case, no filtering
scheme (ideal or realizable) will recover e(t ) .

• Thus, in choosing the sampling rate for a control


system, the sampling frequency should be greater
than twice the highest-frequency component of
significant amplitude of the signal being sampled.

• It should be noted that the ideal sampler is not a


physical device, and thus the frequency spectrum,
as shown, is not the spectrum of a signal appearing
in a physical system.
• But, these ideas are extended to signals that
appear at the S/H device outputs (sampler-and-
ZOH). Let us obtain frequency response of ZOH.
 jT  jT
1  1  j (T 2 )  j (T 2 )
Gho ( j )    
j j
 j (T 2)    j (T 2)  2  j (T 2)
 
 2j  
sin(T 2)  j (T / 2 ) sin(  s )  j ( s )
T  T 
T 2   s

sin(  s )
Thus, Gho ( j )  T
  s
  0 , sin(  s )  0
Gho ( j )     
s  , sin(  s )  0

Frequency response of ZOH


• The frequency response of ZOH must be interpreted
properly. First, it must be remembered that the data
hold must be preceded by an ideal sampler. The idea
is illustrated below with Figures in the next slide.
• Now, suppose that a sinusoid of frequency 1 is
applied to the sampler, where 1   s 2 .
e(t )  cos 1t
 E ( j )   (  1 )   (  1 ) ---Fig. (a)

• The output of the sampler contains the frequencies


in time domain represented by the impulse
functions in the frequency domain. ----- Fig. (b)
• Thus, the frequency response of the ZOH may be
used to determine the amplitude spectrum of data-
hold output signal. ---- Fig. (c)
• Note that the output signal amplitude spectrum will
be the same as that shown in Fig. (c) if the input
signal frequency is   1  n s , n  any integer.

• Hence, any frequencies    s 2 will reflect into


the frequency range 0     s 2 .
• This effect is called frequency foldover, or
frequency aliasing. These reflected frequencies will
be interpreted by the system as low-frequency
information in e(t ) , which generally cannot be
tolerated.
• The frequency aliasing can be prevented either by
increasing  s or by placing an analog anti-aliasing
filter in front of the sampler.
Input and output curves for a first-order hold (FOH)

x(t) = Input, y(t) = Output

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