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Digital Control System - Lecture 2

This document discusses digital control systems and some key concepts related to sampling. It describes how digital control systems use digital signals and computers to carry out processes. It also discusses discrete and continuous systems, sampling, anti-aliasing filters, analog to digital conversion, difference equations, z-transforms, and state space models in discrete time systems.

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adityarajput2928
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© © All Rights Reserved
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Download as PDF, TXT or read online on Scribd
61 views

Digital Control System - Lecture 2

This document discusses digital control systems and some key concepts related to sampling. It describes how digital control systems use digital signals and computers to carry out processes. It also discusses discrete and continuous systems, sampling, anti-aliasing filters, analog to digital conversion, difference equations, z-transforms, and state space models in discrete time systems.

Uploaded by

adityarajput2928
Copyright
© © All Rights Reserved
Available Formats
Download as PDF, TXT or read online on Scribd
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Paper Name: Digital Control System

Paper Code: PE EE 601A

Soumyadip Jana,
Assistant Professor,
Department of Electrical Engineering, TMSL
 Digital Control System uses digital signals and digital computers
to carry out a process.

 A digital control system may contain both discrete and


continuous portions.

 A discrete system works on the discrete input-output signal.

 Hence a digital control system model can be viewed from


different perspectives including control algorithm, computer
program, conversion between analog and digital domains,
system performance etc.

 One of the most important aspects is the sampling process level.


 A discrete signal is a signal which contains only two
discrete states such as ‘ON’ and ‘OFF’
 A general Block Diagram of a digital control system is
shown in figure;
 The analog feedback signal coming from the sensor is
usually of low frequencies. It may often include high
frequency ‘noise’. Such noise signals are too fast for
control system to correct.

 Low pass filtering is often needed to allow good


control performance. The Anti-aliasing filter serves
this purpose.

 The analog signal after anti-aliasing processing is


converted into digital form by the A/D conversion
system.
 The conversion system usually consists of an A/D
converter preceded by a sample and hold (S/H) device.

 The A/D converter converts a voltage or current


amplitude at its input into a binary code representing a
quantized amplitude value closest to the amplitude of
the input.

 However , the conversion is not instantaneous. Input


signal variation, during the conversion time of the A/D
converter, can lead to erroneous results.
 For this reason high performance A/D conversion
systems include a S/H device, which keeps the input to
the A/D converter, constant during its conversion
time.
 Sampling is a process by which a continuous time
signal is converted into a sequence of numbers at
discrete time intervals. It is a fundamental property of
digital control system because of the discrete nature of
operation.
 Figure shows the structure and operation of a finite
pulse width sampler where (a) represents the basic
block diagram and (b) illustrates the function of the
same. T is the sampling period and p is the sample
duration.

The term discrete time system denoted a system in


which all its signals are in a digital coded form. Most
practical systems are hybrid in nature i.e. contains both
analog and digital components.
 A sampler is represented by a switch.

 A sampler converts a continuous time signal f(t) into a


sequence of pulses occurring at the sampling instants
0, T, 2T,………where T is the sampling period. The
switch is closed or opened instantaneously to get an
output at the end of the sampler. This process is
repeated after every time ‘T’.
 If the sampler considered is ideal one then the samples
will be the instantaneous values of the signal f(t) at the
instant the switch is closed. But in practice the closing and
reopening of the switch is not instantaneous instead it
takes a small duration Δt between closing and reopening
the switch.
 The sampling duration p approaches ‘o’ i.e. its operation is
instantaneous.
 Thus for a practical sampler the output available from
the sampler is a pulse of duration Δt. The continuous
time signal f(t) is thus sampled as a pulse of duration
Δt at a regular interval of sampling time ‘T’. The
sampled signal obtained at the output of the sampler
is denoted by f*(t) and is called a starred function.

f*(t)=f(t) · δT(t)

Where δT(t) represents a train of unit impulses.


 Therefore a practical sampler is an impulse modulator
with the continuous time signal f(t) as the modulating
signal and the train of unit impulses δT(t) as the carrier.
f*(t)=f(0)·δ(t)+ f(T)·δ(t-T)+ f(2T)·δ(t-2T)+……….
=
Where K represents the discrete time instants and T is
the sampling period.
 Aliasing is an effect of the sampling that causes different
signals to become indistinguishable. Due to aliasing, the
signal reconstructed from samples may become different
than the original continuous signal. This can drastically
deteriorate the performance if proper care is not taken.

One frequency appearing as another

 It is termed as appearance of high frequency signals as low


frequency signal that distracts the desired signal.

 The effect may be removed by using an anti-aliasing filter,


where the high frequency components are removed from
the input signal.
 The effect may be removed by using an anti-aliasing filter,
where the high frequency components are removed from
the input signal.
 It is basically a LPF with a cut off frequency equal to half
of the sample frequency used by the sampler while
sampling.
Thus f0 ≥ 2fc
 Where f0 = Sampling frequency &
fc = Cut off frequency
 To eliminate the aliasing effect in the sampling normally a
low pass filter, a filter that passes low frequencies but
attenuates the high frequencies, is added before the
sampler and the A/D converter, this filter acts as an anti-
aliasing filter by attenuating the high frequencies, it
prevents the aliasing components from being sampled.
 The minimum sampling frequency of a signal that it will
not distort its underlying information, should be double
the frequency of its highest frequency component.

 Reference

http://195.134.76.37/applets/AppletNyquist/Appl_Nyquist
2.html
 Any sampling frequency (f0) less than twice the input
signal frequency/ cut off frequency (fc) will cause aliasing
effect.

 When sampling frequency (f0) equals twice the input


signal frequency (fc), that sampling rate is called Nyquist
Rate .

 If the sampling frequency (f0) equals twice the input


signal frequency (fc), then such a condition is called
Nyquist Criteria for Sampling.
 Sampling Theorem: Since all computer controlled systems
operate at discrete times only, it is important to know the
condition under which a signal can be retrieved from its
values at discrete points. Nyquist explored the key issue
and Shannon gave the complete solution which is known
as Shannon’s sampling theorem.

 Difference Equations: Difference equations replaced the


differential equations in continuous time theory.
Derivatives and integrals are evaluated numerically by
approximating them with differences and sums.
 Z Transform: Z-transform replaced the role of Laplace
transform in continuous domain.

 State Space Theory: The discrete time representation of


state models are obtained by considering the systems only
at sampling points.

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