Minia University

Faculty of Engineering Electrical Engineering Department

Electronic and communication Section

Lecture 6

Digital Communication

Dr. Eng. Gerges Mansour Salama

Tel: 01284240601 E-mail: gergesmansour@yahoo.com
2 Sampling theory and pulse modulation

INTRODUCTION

We have a large number of continuous-time signals, but we prefer processing of discrete-time

signals. For this purpose we should be able to convert a continuous-time signal into discrete-
time signal.

This problem is solved by a fundamental mathematical tool known as sampling theorem.

Sampling theorem may be viewed as a bridge between continuous-time signals and discrete-
time signals.

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 THE SAMPLING THEOREM
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Sampling of the signals is the fundamental operation in signal-processing.

The number of samples to be taken depends on maximum signal frequency present in the signal.
Different types of samples are also taken like ideal samples, natural samples and flattop samples.

The statement of sampling theorem can be given in two parts as:

A band limited signal of finite energy, which has no frequency-component higher than
Hz, is completely described by its sample values at uniform intervals less than or equal to
second apart.
A band limited signal of finite energy, which has no frequency components higher than f m
Hz, may be completely recovered from the knowledge of its samples taken at the rate of 2fm
samples per second.

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 PROOF OF SAMPLING THEOREM
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 PROOF OF SAMPLING THEOREM
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Sampling of x(t) at a rate of fs Hz (fs samples per second) may be achieved by multiplying x(t) by
an impulse train δTs(t). The impulse train δTs(t) consists of unit impulses repeating periodically
every Ts seconds, where Ts = 1/fs .

sampled signal may be written as

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Now if have to reconstruct x(t) from g(t), we must be able to recover X(w) from G(w). This is
possible if there is no overlap between successive cycles of G(w).
This requires

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 Important Points about Sampling Theorem
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The spectrum G(w) will repeat periodically without overlapping if

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 NYQUIST RATE AND NYQUIST INTERVAL
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Nyquist Rate
When the sampling rate becomes exactly equal to 2 fm samples per second, then it is called
Nyquist rate. Nyquist rate is also called the minimum sampling rate. It is given by
Nyquist Interval
Similarly, maximum sampling interval is called Nyquist interval. It is given by

Sampled spectrum at Nyquist rate.

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EXAMPLE
An analog signal is expressed by the equation
x(t) = 3 cos50πt + 10 sin 300π- cos l00πt. Calculate the Nyquist rate for this signal.
Solution:
The given signal is expressed as

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EXAMPLE

Solution:

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EXAMPLE

Determine:
(i) Minimum sampling rate i.e., Nyquist rate required to avoid aliasing.
(ii) If sampling frequency fs = 400 Hz. What is the discrete-time signal x[n] or x[nTs] obtained
after sampling?
(iii) If sampling frequency fs = 150 Hz. What is the discrete-time signal x[n] or x[nTs] obtained
after sampling?

Solution:
(i) The highest frequency component of continuous-time signal is f =100 Hz. Hence minimum
sampling rate required to avoid aliasing is called Nyquist rate and is given as

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EXAMPLE

Solution:

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 RECONSTRUCTION FILTER (LOW-PASS FILTER)
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The low-pass filter is used to recover original signal from its samples.
This is also known as interpolation filter.

A low-pass filter is that type of filter which passes

only low-frequencies up to a specified cut-off frequency
and rejects all other frequencies above cut-off frequency

Ideal low-pass filter (LPF).

the amplitude response decreases slowly to

become zero. This means that there is a
transition band in case of practical filter.

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the use of practical low-pass filter in reconstruction of original signal from its sample.

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 SIGNAL RECONSTRUCTION:
THE INTERPOLATION FORMULA
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a signal x(t) bandlimited to fm Hz can be reconstructed (interpolated) completely from its

samples. This is achieved by passing the sampled signal through an ideal low-pass filter
(LPF) of cut-off frequency fm Hz.

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 EFFECT OF UNDER SAMPLING : ALIASING

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the signal is under-slampled in this case (fs < 2 f m) and some amount of aliasing is
produced in this under-sampling process. In fact, aliasing is the phenomenon in which a high
frequency component in the frequency-spectrum of the signal takes identity of a lower-
frequency component in the spectrum of the sampled signal,

it is obvious that because of the overlap due to aliasing phenomenon, it is not possible to
recover original signal x(t) from sampled signal g(t) by low-pass filtering since the spectral
components in the overlap regions add and hence the signal is distorted.

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Since any information signal contains a large number of frequencies, so, to decide a
sampling frequency is always a problem

Therefore, a signal is first passed through a low-pass filter. This low-pass filter blocks all the
frequencies which are above fm Hz.

This process is known as band- limiting of the original signal x(t). This low-pass filter is called
prelias filter because it is used to prevent aliasing effect. After bandlimiting, it becomes easy to
decide sampling frequency since the maximum frequency is fixed at fm Hz

to avoid aliasing:

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 SAMPLING TECHNIQUES
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Basically, there are three types of sampling techniques as under:

Instantaneous sampling Natural sampling Flat-top sampling

Ideal Sampling or Instantaneous Sampling or Impulse Sampling

x(t) is the input signal (b) Impulse train

(a) Baseband signal

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Natural sampling

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Flat-top sampling

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(a) Baseband signal x(t)

(b) Instantaneously sample single
(c) Constant pulse width function
(d) Flat top sampled signal

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APERTUR E EFFECT

it may be observed that by using flat top samples an amplitude distortion is introduced in
the reconstructed signal x(t) from g(t). In fact, the high frequency roll off of H(f) acts like a
low-pass filter and thus .attenuates the upper portion of message signal spectrum. These high
frequencies of x(t) are affected. This type of effect is known as aperture
effect.

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Example

This signal is I sampled at the Nyquist rate with a periodic

train of rectangular pulses of duration 50/3 milliseconds.
Determine the spectrum of the sampled signal for
frequencies upto 50 Hz giving relevant expression.
Solution
it may be observed that the signal is bandlimited to 10 Hz.

The spectrum of the flat top sampled signal is given as

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Example

A flat top sampling system samples a signal of maximum frequency 1Hz with 2.5 Hz sampling
frequency. The duration of the pulse is 0.2 seconds. Compute the amplitude distortion due to
aperture effect at the highest signal frequency. Also determine the equalization
characteristic.
Solution Given that sampling frequency

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COMPARISON OF VARIOUS PULSE ANALOG MODULATION METHODS

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Lecture 6 Ch. 3 Dr. Eng. Gerges M. Salama

 Minia University
Faculty of Engineering Electrical Engineering Department
Electronic and communication Section

The End
Lecture 6

Thank you for attention

Digital Communication

Dr. Eng. Gerges Mansour Salama

Tel: 01284240601 E-mail: gergesmansour@yahoo.com