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Ministry of Higher Education and Scientific Research

Al-Mansour University College

Computer Engineering Department

Digital Communications and Coding

Third Year

By
Msc. Ahmed Saeed

2022-2023
Digital Communications and Coding The Lecturer: Ahmed
Digital Communications and Coding Saeed

Chapter 2

1. Sampling Theorem

The trend in the design of new communication systems has been toward

increasing the use of digital techniques. Digital communications offer several important

advantages compared to analog communications.

To transmit analog signals, such as voice and video signals, by digital means, the

signal has to be converted to a digital signal. While an analog signal is continuous in

both time and amplitude, a digital signal is discrete in both time and amplitude. The first

step to convert analog signals (a signal of continuous time) in to digital form (a signal of

discrete time) is done by sampling.

Sampling is the process in which a continuous-time signal is sampled by

measuring its amplitude at discrete instants. The sampling process is sometimes referred

to as pulse amplitude modulation (PAM). We need to remember, that the result is still an

analog signal with nonintegral values.


The value of the signal is measured at certain intervals in time. Each measurement

is referred to as a sample. The analog signal is sampled every 𝑇𝑠, where 𝑇𝑠 is the

sample interval or sample period and its inverse is called the sampling rate or sampling
1
frequency 𝑇𝑠 =
2𝑓𝑚

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Digital Communications and Coding The Lecturer: Ahmed
Digital Communications and Coding Saeed

Fig. (1): (a) the x (t ) signal


(b) 𝑥𝑠(t) , sampled version of x (t )

In general, to preserve the full information in the signal, it is necessary to sample

at twice the maximum frequency of the signal to avoid losing information that is in

the signal. This is known as the Nyquist rate ƒ𝑛 (or minimum sampling rate ƒ𝑠 𝑚i𝑛 = ƒ𝑛

= 2ƒ𝑚). The sampling theorem states that a signal can be exactly reproduced if it is

sampled at a frequency ƒ𝑠, where ƒ𝑠 is greater than twice the maximum frequency

ƒ𝑚 in the signal ƒ𝑠 ≥ 2ƒ𝑚.

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Digital Communications and Coding The Lecturer: Ahmed
Digital Communications and Coding Saeed

Fig. (2): Spectra of x (t) and 𝑥𝑠(t) (a) x (f) (b) 𝑥𝑠 (f), 𝑓𝑠 > ƒ𝑚
(c) 𝑥𝑠 (f), 𝑓𝑠 = ƒ𝑚 (d) 𝑥𝑠 (f), 𝑓𝑠 < ƒ𝑚

Every pulse train is described by three parameters:


1. The pulse amplitude Ap
2. The pulse cycle Tp, which is also called the pulse frame.
3. The pulse duration Ʈ, also called the pulse width.
The general characteristics of a pulse train are displayed in Fig. (3).

Fig. (3): pulse train parameters.


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Digital Communications and Coding The Lecturer: Ahmed
Digital Communications and Coding Saeed 1
The pulse frequency can be derived from the pulse cycle using the relation fp= .
𝑇𝑝

The ratio between the pulse width Ʈ and the pulse period Tp determines the pulse’s duty
factor Tp. Therefore, a pulse train can also be classified by the set of parameters Ap; fp
and Tp.
While proving sampling theorem we considered that fs = 2fm. consider the case of
ƒ𝑠< 2ƒ𝑚 then the phenomenon is called aliasing.

Effects of aliasing:

1. Since high and low frequencies interfere with each other, distortion is
generated.
2. The data is lost and it cannot be recovered.

Different ways to avoid aliasing:

Aliasing can be avoided by two methods:


1. Sampling rate ƒ𝑠 ≥ 2ƒ𝑚.
When the sampling rate is made higher than 2ƒ𝑚, then the spectrums will not
overlap and there will be sufficient gap between the individual spectrums.
2. Band limiting the signal.
The Sampling rate is, ƒ𝑠 ≥ 2ƒ𝑚. Ideally speaking there should be no aliasing. But
there can be few components higher than 2fm. these components create aliasing.
Hence a low pass filter is used before sampling the signals as shown in fig. (4). Thus,
the output of low pass filter is strictly band limited and there are no frequency
components higher than fm. then there will be no aliasing.

Fig. (4):

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Digital Communications and Coding The Lecturer: Ahmed
Digital Communications and Coding Saeed
2. Pulse Modulation
In pulse modulation some parameter of a pulse train is varied in accordance with
the massage signal. Two families of pulse modulation may be distinguished: analog pulse
modulation and digital pulse modulation.
In analog pulse modulation, a periodic pulse train is used as the carrier wave, and
some characteristics features of each pulse (e.g. Amplitude, Position, and Width) is varied
in a continuous manner in accordance with the corresponding sample value of the
message signal. Thus, in analog pulse modulation, information is transmitted basically in
analog form, but the transmission takes place at discrete times.
In digital pulse modulation, on the other hand, the massage signal isrepresented in a
form that is discrete in both time and amplitude; thereby permitting its transmission in
digital form as a sequence of coded pulses.

The above two techniques can be further classified as,

Pulse modulation

Analog Digital

1. pulse amplitude modulation (PAM) 1. Pulse code modulation (PCM)


2. pulse position modulation (PPM) 2. Delta modulation (DM)
3. pulse width modulation (PWM) 3. Adaptive delta modulation (ADM)
4. Differential pulse code modulation
(DPCM)

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Digital Communications and Coding The Lecturer: Ahmed
Digital Communications and Coding Saeed
2.1 Pulse Amplitude Modulation (PAM)

PAM is the simplest and most basic form of analog pulse modulation. In PAM the
amplitude of regularly spaced pulses is varied in proportion to the corresponding
sample values of a continuous message signal, the pulses can be of a rectangular form
or other appropriate shape.

Fig. (5) (a) Modulating signal. (b) PAM signal.

PAM as defined here is somewhat similar to the sampling with a rectangular


pulse train where the message signal is multiplied by a periodic train of rectangular
pulses. However, in the sampling with a rectangular pulse train the top of each
modulated rectangular pulse varies with the message signal, whereas in PAM it is
maintained flat.
There are two operations involved in the generation of the PAM signal, these are
instantaneous sampling of the message signal every T s second, where the sampling
1
rate 𝑓𝑠 = is chosen in accordance with the sampling theorem and lengthening the
𝑇𝑠

duration of each sample so obtained to some constant value (𝜏).

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Digital Communications and Coding The Lecturer: Ahmed
Digital Communications and Coding Saeed
Multiplexing techniques
Multiplexing is the transmission of information (either voice or data) from more
than one source to more than one destination on the same transmission medium. The
most two common methods used are frequency division multiplexing (FDM) and time
division multiplexing (TDM).

A. Frequency Division Multiplexer (FDM)


In FDM multiple sources that originally occupied the same frequency spectrum
are each converted to a different frequency band and transmitted simultaneously over
a single transmission medium. FDM is an analog multiplexing scheme. Figure below
shows the frequency-time plane.

Fig. (6): Frequency Division Multiplexing (FDM) and Time Division


Multiplexing (TDM)

If two input signals to a mixer are sinusoids with frequencies f A and fB, the
mixing or multiplication will yield new sum and difference frequencies at fA+B and
fA-B. Equation below describes the effect of the mixer.
1
cosAcosB= [cos(𝐴 + 𝐵) + cos(𝐴 − 𝐵)]
2

A simple FDM example with three translated voice channels is shown in Fig. (7).

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Digital Communications and Coding The Lecturer: Ahmed
Digital Communications and Coding Saeed

Fig. (7): Multiplexing of three channels using (FDM)

B. Time Division Multiplexing (TDM)


The use of fairly short pulse widths in PAM signals leaves sufficient space
between samples for insertion of pulses from sampled signals. The method of
combining several sampled signals in a definite time sequence is called time division
multiplexing (TDM). We shall discuss the principle of TDM here with particular
reference to PAM although the principles apply as well to other types of pulse
modulation.
Suppose we wish to time multiplexed two signals using PAM. Let us assume that
both input signal f1(t) and f2(t) are low pass, and band limited to 3KHz. The sampling
theorem states that each must be sampled at a rate no less than 6KHz. This requires a
12KHz minimum clock rate for the two channel system. Fig. (8) shows the block
diagram of PAM/TDM system.

Fig. (8): PAM/TDM system.

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Digital Communications and Coding The Lecturer: Ahmed
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Fig. (9): How data flow in TDM.

The time multiplex PAM output might appear something like that shown in Fig. (10).

Fig. (10): PAM/TDM output signal

The time spacing between adjacent samples in the time multiplex signal waveform
𝑇𝑠
(Tx), can be defined as 𝑇𝑥=
𝑛
where
Ts equal to sampling rate, and
n equal to number of input signals.

To prevent any irretrievable loss of information in the composite waveform then


1
requires that bandwidth Bx of LPF must satisfy the criterion 𝐵𝑥 ≥ .
2𝑇𝑥
At the receiver the composite time multiplexed and filtered waveform must be
resampled and separated into the appropriate channel. One the pulses are separated;
the normal sampling considerations applies and the analog reconstruction of signals
can be obtained by LPF.

With TDM system, transmission from multiple sources occurs on the same
transmission medium but not at the same time. Transmission from various sources is
interleaved in time domain.

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Digital Communications and Coding The Lecturer: Ahmed
Digital Communications and Coding Saeed
The multiplexing operation consists of providing each source with an opportunity
to occupy one or more slots.
The demultiplexing operation consists of desloting the information anddelivering
the data to the intended sink.
The communication switches (S1…SM) have synchronized so that the massage
corresponding to signal (1), for example, appears on the channel (1) output, and so
on. Time is segmented in to intervals called frames. Each frame is further partitioned
in to assignable user time slots. The simplest TDM scheme called fixed-assignment
TDM. In fixed assignment TDM scheme, the entire slot has no data to send during a
particular frame, that slot is wasted.

Fig. (11): Example of Fixed TDM

Another more efficient scheme is the dynamic assignment TDM of the slots
rather than fixed assignment. Figures below show the dynamic assignment TDM
system.

Fig. (12): Example of Dynamic TDM


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Digital Communications and Coding The Lecturer: Ahmed
Digital Communications and Coding Saeed
Example 1:
Channel 1 of two channels PAM system handles 8KHz signal. Channel 2 handles 10
KHz signals. The two channels are sampled at equal intervals of time using very
narrow pulses at the lowest frequency that is theoretical adequate. The sampled
signals are time multiplexed and passed through a LPF before transmission.

(1) What is the minimum clock frequency of the PAM system.

(2) What is the minimum cut off frequency of LPF used before transmission that
will preserve the amplitude information on the output pulses.

(3) What would be the minimum bandwidth if these channels were frequency
multiplexed, using AM technique and SSB technique.

Solution
(1)
𝑓𝑠1 = 2*𝑓𝑚1
𝑓𝑠1 = 2*8 =16 KHZ
𝑓𝑠2 = 2*𝑓𝑚2
𝑓𝑠2 = 2*10 =20 KHZ
In order to sample channel 2 adequately
𝑓𝑠 = 𝑓𝑠2 =20 KHZ
The n = 2
∴ the minimum clock = n* 𝑓𝑠 = 2*20 = 40 KHZ
(2)
1 1
𝑇𝑠= = = 50𝜇 sec
𝑓𝑠 20 𝐾𝐻𝑍

The n = 2
𝑇 50
𝑇 𝑥= 𝑠 = = 25𝜇 𝑠𝑒𝑐
𝑛 2
1
𝐵𝑥 ≥
2𝑇𝑥

∴𝐵𝑥 = 20 KHZ

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Digital Communications and Coding The Lecturer: Ahmed
Digital Communications and Coding Saeed
(3)
For AM
min B.W = 2(𝑓𝑚1+ 𝑓𝑚2) = 2(8+10) = 36𝐾𝐻𝑧
For SSB
min B.W = (𝑓𝑚1+ 𝑓𝑚2) = (8+10) = 18𝐾𝐻𝑧

H.W
Two low pass signals, each band limited 4KHz, are to be time multiplexed into a
single channel using PAM. Each signal is impulse sampled at a rate 10KHz. The time
multiplexed signal waveform is filtered by an ideal LPF before transmission.
(1) What is minimum clock frequency of the system.
(2) What is the minimum cut off frequency of the LPF.
(3) In the receiver side, determine the minimum and maximum acceptable bandwidth
of the LPF used in retrieving the analog signal.

Ans. (1) 20KHz (2) 10KHz (3) 4KHz, 5KHz.

2.2 Pulse Width Modulation (PWM) and Pulse Position Modulation


(PPM).
One type of pulse timing modulation uses constant amplitude pulses whose width
is proportional to the value of message signal at the sampling instants. This type is
designated as pulse width modulation (PWM) or pulse duration modulation (PDM) is
also called.

PAM, PWM and PPM waveforms for a given message signal are shown in the
figure in Fig. (13):

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Fig. (13): PAM, PWM and PPM waveforms

As we can observe, the amplitude and the frequency of the PWM wave remain
constant. Only the width changes. That is why the information is contained in the
width variation. As the noise is normally additive noise, it changes the amplitude of
the PWM signal. At the receiver, it is possible to remove these unwanted amplitude
variations very easily by means of a limiter circuits. As the information is contained in
the width variation, it is unaffected by the amplitude variations introduced by the
noise. Thus, the PWM system is more immune to noise than the PAM signal.

Another possibility is to keep both the amplitude and the width of the pulses
constant but vary the pulse position in proportion to the value of message signal at
sampling instant. This is designated as pulse position modulation (PPM).
In PPM, the amplitude and width of the pulses is kept constant but the position of
each pulse is varied in accordance with the amplitudes of the sampled values of the
modulating signal. The position of the pulses is changed with respect to the position of
reference pulses. The PPM pulses can be derived from the PWM pulses as shown in
Fig. (13).

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Digital Communications and Coding The Lecturer: Ahmed
Digital Communications and Coding Saeed
Generation of PWM & PPM

The block diagram of a PWM signal generator is shown in fig. (14). This circuit
can also be used for the generation of PPM signal.

Fig. (14): PWM and PPM generation.

A sawtooth generator generates a sawtooth signal of frequency fs, and this


sawtooth signal in this case is used as a sampling signal. It is applied to the inverting
terminal of a comparator. The modulating signal x (t) is applied to the non-inverting
terminal of the same comparator. The comparator output will remain high as long as
the instantaneous amplitude of x (t) is higher than that of the ramp signal. This gives
rise to a PWM signal at the comparator output as shown in fig. (14).

Here, it may be noted that the leading edges of the PWM waveform coincide
with the falling edges of the ramp signal. Thus, the leading edges of PWM signal are
always generated at fixed time instants. However, the occurrence of its trailing edges
will be dependent on the instantaneous amplitude of x(t). Therefore, this PWM signal
is said to be trail edge modulated PWM.

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Digital Communications and Coding The Lecturer: Ahmed
Digital Communications and Coding Saeed

Fig. (14): PWM and PPM Waveform.

The PWM pulses obtained at the comparator output are applied to a monostable
multivibrator. The monostable is negative edge triggered. Hence, corresponding to
each trailing edge of PWM signal, the monostable output goes high. It remains high
for a fixed time decided by its own RC components. Thus, as the trailing edges of the
PWM signal keep shifting in proportion with the modulating signal x(t), the PPM
pulses also keep shifting, as shown in fig. (14).

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