Module 8: Numerical Relaying I: Fundamentals: Sampling Theorem

Module 8 : Numerical Relaying I : Fundamentals
Lecture 28 : Sampling Theorem

Objectives
In this lecture, you will review the following concepts from signal processing:
Role of DSP in relaying.
Sampling theorem.
Role of anti-iliasing filter.
28.1 Why Digital Signal Processing?

Digital relaying involves digital processing of one or more analog signals. It involves following three
steps:
1.
Conversion of analog signal to digital form.
2.
Processing of digital form.
3.
Boolean decision to trip or not to trip.

Usually in DSP, after processing information in discrete domain, it has to be converted back to analog
domain. However, for us the step- 3 does not involve conversion of processed signals back to analog
form.
In the previous lecture, we have discussed the step - 1 in detail. In this lecture we discuss the next step.
At this point, a worthwhile observation is that direct analog signal processing is conceptually much
simpler. However, advantages of digital processing far outweigh analog processing. Some of the
advantages of digital processing are as follows:
Operation of digital circuits do not depend on precise values of digital signals. As a result, a digital circuit
is less
sensitive to tolerances of component values.
A digital circuit has little sensitivity to temperature, aging and other external parameters.
In terms of economics of volume, a digital circuit can be reproduced easily in volume quantities. With
VLSI circuits, it is
possible to integrate highly sophisticated and complex digital signal processing systems on a single chip.
In DSP, accuracy of computation can be increased by increasing word length. With the availability of
floating point
arithmetic in digital signal processors, dynamic ranges of signal and coefficients can be increased.
A signal processor can process many signals, reducing processing cost per signal.
Digital implementation allows the realization of certain characteristics not possible with analog
implementation; such
as polygon in R-X plane for distance relaying.
Digital signals can be stored indefinitely without loss of accuracy.
There are also some disadvantages with DSP. One of them is that DSP contains active devices. Active
devices are less reliable than passive components. Passive components consume less power than active
devices. However, advantages of digital relays (i.e. relaying using digital signal processing) are far more
significant than the disadvantages. In what follows, we discuss digital signal processing for relaying.
28.2 Sampling
Consider a continuous time domain sinusoid signal as, x(t) = sin(2 0 t) .The sine wave has frequency
0 e.g. 50 Hz. Let the waveform x(t) be sampled at a rate of s samples/sec, i.e. with time period ts = 1/
s sec. Let the sampling process start at time
. The samples of this sequence are given by

(1)
(2)
(3)
(4)
(5)
Because of periodicity of sine wave, it is not possible to distinguish two samples with a phase difference
equal to 2m , where
is an integer. Therefore,
(6)
(7)
If we choose
to be an integer multiple of
above equation transforms into the following:
i.e.
an integer and substitute t s = 1/ s, the

(8)
Note that in equation( 7)

value of .
has to be varied from sample to sample, so as to maintain fixed
28.2 Sampling
Consider a continuous time domain sinusoid signal as, x(t) = sin(2 0 t) .The sine wave has frequency
0 e.g. 50 Hz. Let the waveform x(t) be sampled at a rate of s samples/sec, i.e. with time period ts = 1/
sec. Let the sampling process start at time
.Then the first
successive samples have the
values;
(1)
(2)
(3)
(4)
(5)
Because of periodicity of sine wave, it is not possible to distinguish two samples with a phase difference
equal to 2m , where
is an integer. Therefore,
(6)
(7)
If we choose
to be an integer multiple of
above equation transforms into the following:
i.e.
an integer and substitute t s = 1/ s, the

(8)
Note that in equation( 7)

value of .
has to be varied from sample to sample, so as to maintain fixed
28.2 Sampling (contd..)

The equation (8), leads to a very interesting result viz.
When sampling at a rate of s samples/sec. if k is any positive or negative integer, we cannot distinguish
between sampled values of sine wave of 0 Hz and a sine wave of (f 0 +kf s) Hz.
Fig 28.2 shows a 7kHz signal
being sampled at 6khz with 0
= 7, s= 6 and k = -1, we
reach the conclusion that we
cannot
distinguish
between
signal of 7kHz and 1kHz with
sampling frequency of 6kHz.
This effect of 7kHz signal taking
an alias of 1kHz signal is called
aliasing. In this case, a high
frequency signal has taken an
alias of a low frequency signal.

An another example, consider a 50Hz signal sampled at 50Hz (see in fig 28.3). It can be seen that
signal is aliased to dc signal.
Similarly, sampling a 50 Hz signal at 51 Hz will alias it to 1 Hz.
Fig 28.4 shows a signal with frequency content between B Hz . Such signals are said to be band limited
signals. Note that because
and
magnitude
component of a real life signals have typically an even symmetry around dc signal. By the observation
made in the previous slide that a signal of 0 Hz can be aliased to ( 0 s ) Hz { = 1, 2,--- } , it
follows that post sampling in frequency domain, we will see repeating lobes (replicas) of original signal,
each lobe being displaced by s Hz. In other words, after sampling we cannot distinguish the signal lobe
from other replicated lobes.
An interesting analog can be drawn by considering a room having many mirrors each reflecting image
from one to another. It is seen that if a person is standing in such a room, another observer cannot
distinguish him from his image. The difficulty can be resolved if the observer has an idea of location or
coordinates of the real person. In the same manner, we can identify the original lobe from replicated
lobes if we have an idea of the frequency content of original signal. In fig 28.5, notice that lobes are
distinctly separated because s > 2B Hz . On the other hand, if s = 2B Hz , then as seen in fig 28.6,
lobes will just touch each other. If however, s < 2B Hz, then lobes will overlap (fig 28.7) and this will
lead to distortion of replicated frequency spectrum. Thus, it is necessary that s the sampling frequency
should atleast equal to 2B Hz.
Thus qualitatively, we can classify sampling frequency into three categories.

1.
Sampling at a rate
2.
Sampling at a rate
3.
Sampling at a rate
1.
When sampling frequency, s< 2B, then there is an overlapping effect around frequency s/2, known as
the folding frequency. As a consequence of superposition, the frequency domain information is
distorted. Thus, we should choose s > 2B. This important result is a part of sampling theorem stated
below in two equivalent ways.
A band limited signal of finite energy, which has no frequency component higher than
Hz, is completely
described
by specifying the values of the signal at instants of time separated by
2.
seconds.
A band limited signal of finite energy which has no frequency component higher than
completely
per sec.
recovered from the knowledge of its samples taken at a rate of
The sampling rate of 2B samples per sec is known as Nyquist rate.
Hz may be
In practice, even a band limited signal will contain noise. Noise reflects as high frequency component in
Hz, we cannot
the overall spectrum, (fig 28.8). Thus, even if we sample the signal at a rate say
reconstruct the correct frequency domain information. Noise is aliased to lower frequency. It distorts the
frequency domain information by superposing an alias of noise on the original signal. To avoid this, in
practice it is necessary to pass the continuous signal first through an analog low pass filter. Such a filter
is known as anti-aliasing filter. Fig 28.9 illustrates this concept.
28.2 Sampling
Review Questions
1.
Derive the sampling theorem.
2.
A 40 kHz signal is sampled at 49 kHz. What is the minimum frequency to which this signal will be aliased.
3.
For the signal in fig 28.2, suggest appropriate sampling frequency.
Recap
In this lecture we have learnt the following:

Role of DSP in relaying.
Sampling theorem.
Role of anti-aliasing filter.

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Module 8 : Numerical Relaying I : Fundamentals

Lecture 28 : Sampling Theorem

Role of anti-iliasing filter.

28.1 Why Digital Signal Processing?

Conversion of analog signal to digital form.

Processing of digital form.

Boolean decision to trip or not to trip.

. The samples of this sequence are given by

an integer and substitute t s = 1/ s, the

Note that in equation( 7)

has to be varied from sample to sample, so as to maintain fixed

sec. Let the sampling process start at time

.Then the first

successive samples have the

an integer and substitute t s = 1/ s, the

Note that in equation( 7)

has to be varied from sample to sample, so as to maintain fixed

28.2 Sampling (contd..)

28.2 Sampling (contd..)

Similarly, sampling a 50 Hz signal at 51 Hz will alias it to 1 Hz.

28.2 Sampling (contd..)

28.2 Sampling (contd..)

Thus qualitatively, we can classify sampling frequency into three categories.

28.2 Sampling (contd..)

28.2 Sampling (contd..)

Derive the sampling theorem.

For the signal in fig 28.2, suggest appropriate sampling frequency.

In this lecture we have learnt the following:

Role of anti-aliasing filter.

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