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PRINCIPLES OF
COMMUNICATIONS
SKEE 3533
SECTION 06
Dr Noor Asniza Murad
Asniza@fke.utm.my

Adapted from latest version by NAS

Journal
Which journal?: Journal of Electromagnetic
Waves and Applications (JEMWA)

Topic
4Pulse
Why choosing this journal?:
Modulation
- Covers the Antenna
theory and applications,
biological effect.
- Succinct papers within 10 pages
Introduction
- Sample and guideline
for authors sample
Sampling Theorem

Introduction
o In the early 90s, telecommunication networks is changing towards digital
world.

o With the rapid advancement in the fields of VLSI and microprocessor, several
telecommunication components such as transmission line and switching has
been using digital signals in their operation.
o Therefore, information signals must be changed to digital form so that it
can be transmitted through this network.
o Several techniques requiring full coding of the original signal will be used:
Pulse Code Modulation (PCM)
Differential PCM (DPCM)
Adaptive Differential PCM (ADPCM)
Delta Modulation (DM)
Adaptive Delta Modulation (ADM)
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Digital Modulation
Advantages :
Immunity to noise
Easy storage and processing:
MP, DSP, RAM, ROM, Computer
Regeneration
Easy to measure
Enables encryption
Data from several sources can be integrated and transmitted using
the same digital communication system
Error correction detection can be utilized
Disadvantages :
Requires a bigger bandwidth
Analog signal need to be changed to digital first
Not compatible to analog system
Need synchronization

Voice :

Analog : 4 kHz
Digit : 2 x 4 kHz x 8 bit = 64 kb/s
BWmin 32 kHz
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TRANSMISSION METHOD FOR ANALOG & DIGITAL SIGNALS

Analog
input
Analog input

Analog channel
Baseband
Modulator

Analog
channel

Analog
output
De-Modulator

encoder

Digital
channel

decoder

Digital input

Modem

Analog
channel

Modem

Analog input

ADC &
encoder

Digital
channel

Decoder &
DAC

Digital input

Analog input

ADC &
encoder

Modem Analog Modem

channel

Analog output

Digital output

Digital output

DAC &
decoder

Analog output

Analog output

Digital Modulation
(Model of Digital Comm.)
The model of Digital System is shown below:

Input
Signal

Source
Encoder

Channel
Encoder

Digital
Modulation

TRANSMITTER

Output
Signal

Source
Decoder

Channel
Decoder

Trans. Medium
or Channel

Digital
Demodulation

RECEIVER
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Types of Pulse Modulation

sampling analog
signal

Transport the pulses

from source to
destination over a
channel
A device to perform this is called:

Pulse
Modulation
consists of:

PWM (Pulse Width

Modulation)
Vm

Convert into
discrete pulses

PAM (Pulse Amplitude Modulation)

VPAM Vm

PPM (Pulse Position

Modulation)
d (pulse delay) Vm

Pulse Modulation is a
process of:

PCM (Pulse Code

Modulation)

i) ADC (Analog-to-Digital Converter)

ii) DAC (Digital-to-Analog Converter).
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PAM
(Pulse Amplitude Modulation)
Definition: The amplitudes of regularly spaced pulses vary with instantaneous sample
values of a continuous message signal
It is used to describe the conversion of analog signal to pulse-type signal in
which the amplitude of the pulse denotes the analog information.
In addition, it is a series of pulses in which the amplitude of each pulse
represents the amplitude of the information signal at a given time.

PAM (Pulse Amplitude Modulation)

VPAM Vm

PWM
(Pulse Width Modulation)
Definition: The width/duration of pulses are varied in proportional to the Amplitude of
the analog signal at the time the signal is sampled.
PWM is also called a pulse duration modulation (PDM) or pulse length modulation
(PLM).
PWM (Pulse Width Modulation)
Vm

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PPM
(Pulse Position Modulation)
Definition: The relative position of regularly spaced pulses is proportional with the amplitude
of the modulating signal
It is a series of pulses in which the timing of each pulse represents the amplitude
of the information signal at a given time.
PPM (Pulse Position Modulation)
d (pulse delay) Vm

The highest amplitude of the sample produces a pulse to the far right, and
the lowest amplitude of the sample produces a pulse to the far left
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Pulse Modulation

Less susceptible to
noise

Less susceptible to
noise compared to PAM

Easily effected by noise

12

Pulse Modulation: Comparison

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Sampling Theorem
m(t)

ms(t)

m(t)

ms(t)

Ts

Digital signal
s(t)

s(t)

t
A process of periodically sampling the continually changing analog input voltage
and convert it to a series of constant amplitude pulses

Sampling Theorem: The minimum sampling rate is equal to twice the highest audio/message
signal frequency

fs 2 fm

Nyquist theorem

If fs 2fm : foldover distortion or aliasing occurs

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Sampling Theorem
m(t)
t

m(t)

ms(t)

ms(t)
s(t)

Digital signal
s(t)

Ts
t

Fourier series for impulse train :

1
st 1 2 cos st 2 cos 2st 2 cos 3st .....
Ts
1
2
where
T

s
2f s
s
fs
Ts
Therefore :

Nyquist theorem
states that:

fs 2 fm
f N 2 fm

ms t mt st
1
mt 2mt cosst 2mt cos 2st 2mt cos 3st .....
Ts
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Frequency domain

Time domain

M ()

m(t )

Analog signal
0

S ( )

s(t )

6Ts

m 0 m

Ts

6Ts

2
Ts

digital signal
(sample pulse)

M s ()

ms (t )

1
Ts

Sampled signal
6Ts

Ts

6Ts

s m
s

m 0 m

s m

s m

s m

The sampled spectrum shows that the components at 2fs, 3fs are the
replications of the main sampled spectrum
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Reconstruction
RX

TX

M ()

ms(t)

m(t)
m 0 m

M r ()
1

h(t)

mr(t)

m 0 m

Low pass filter

Pulse signal
s(t)

H ()

Output at this point

Ts

M s ()
1
Ts

s m
s

m 0 m

s m

s m

s m

0 n

 Sampling process shown previously uses an ideal pulse signal

However, it is quite difficult to generate an ideal pulse signal practically
The usual pulse signal generated is as shown below:
s(t)

t
Ts

- pulse width
Ts pulse period

A 2 A
2 nt
s(t )

cn kos

Ts
Ts n1
Ts
diwhere
mana
n
sin
Ts
n
cn
sinc
n
Ts
Ts

2
s(t )

Ts Ts

n
2nt
sinc
cos

Ts
Ts
n 1

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Mathematical analysis
ms(t)

m(t)
t

m(t)

ms(t)

s(t)
Pulse signal
s(t)
Fourier series for pulse signal, s(t) :

Therefore, the sampled signal:

2
s(t )

Ts Ts

n
2nt
sinc
cos

Ts
Ts
n 1

ms (t ) m(t )s(t )

2
ms (t ) m(t ).
Ts Ts

2nt

cn cos

Ts
n 1

m(t ) m(t )2
ms (t )

Ts
Ts

2nt
cn cos

Ts
n 1

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Mathematical analysis: continue

For n = 1, 2 , 3 ..

m(t ) m(t )2
2t m(t )2
4t
ms (t )

c1 cos

c2 cos

Ts
Ts
Ts
Ts
Ts
m(t )2
6t
c3 cos
....
Ts
Ts
The above expression shows that the frequency components of the
sampled signal is at fs , 2fs and 3fs . Components 2fs and 3fs is a replica
of the spectrum of the sampled signal.

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ms (t )

m(t ) m(t )2
2t m(t )2
4t

c1 cos

c2 cos

Ts
Ts
Ts
Ts
Ts
m(t )2
6t
c3 cos
....
Ts
Ts

The spectrum of the sampled signal has sidebands fs fm , 2fs fm ,

3fs fm and so on.

ms(f)

Spectrum of the sampled signal

f
0

fs-fm

fs

fs+fm

2fs-fm 2 2f
s
s

2fs+fm

3fs-fm3f

3fs

3fs+fm

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The choice of sampling frequency, fs must follow the sampling theorem to overcome
the problem of aliasing and loss of information
(a) Sampling frequency=> fs1 < 2fm (max)
ms(f)

Aliasing

Shannon sampling
theorem=> fs 2fm
f

fm

fs1

2fs1

3fs1

A bandlimited signal that has

a maximum frequency, fmax
can be regenerated from the
sampled signal if it is sampled
at a rate of at least 2fmax .

(b) Sampling frequency=> fs2 > 2fm (max)

ms(f)

f
fm

fs2

2fs2

3fs2

Nyquist frequency
fs = 2fm= fN

BW = 2fm(max)

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fs > 2fm

fs = 2fm

fs < 2fm

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Detection of sampled signal

By using LPF to the sampled signal, ms(t)
ms(t)

LPF

m(t)

Cut-off frequency , fo for LPF must be within the range: fm fo fs - fm

o Even though the sampled signal can be detected easily at fs = 2fm ,
but usually the fs is set fs > 2fm . The main reason is to have a guardband .

o Therefore, the maximum frequency that can be processed by the sampled data
using sampling frequency, fs (without aliasing) is:
o

fs
1
fm
2 2T s

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Sampling methods: PAM

ms(t)

Natural Sampling

t
Ideal Sampling

Flat-top Sampling

In every sampling methods, the pulse amplitude is directly proportional to the

amplitude of the information signal
Practically, an ideal sampling is difficult to generate
However, by using an ideal and natural sampling, noise can be eliminated, which is
not the case for flat-top sampling
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m(t)
Information signal
t

s(t)
Pulse signal

t
Ts

ms(t)

ms(t)

Ts

t
Ts

Natural Sampling

t
Ts

Flat-top Sampling
Sampled signal (PAM)

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Mathematical analysis: PAM

From the sampling process, the sampled signal:

m(t ) m(t )2
ms (t )

Ts
Ts
where :

If:

Therefore

Taking:

cn cos
n 1

2nt
Ts

n
cn sinc
Ts

Ts

therefore

sinc

m(t ) m(t )2
ms (t )

Ts
Ts

n
1
Ts

2nt
cos

Ts
n 1

mt 1 cos m t
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replace

mt 1 cos m t

into

m(t ) m(t )2
ms (t )

Ts
Ts

2
ms t 1 cos m t 1 cos m t
Ts
Ts
1 cos m t

Ts

1 cos m t

Ts

cos
n 1

2nt
Ts

cos n t
n 1

2 cos n t
n 1

2
cos
n

s
Ts n1

1 cos m t 1 2 cos s t 2 cos 2 s t 2 cos 3 s t ....
Ts

1 cos m t 2 cos s t 2 cos s t cos m t

Ts 2 cos 2 s t 2 cos 2 s t cos m t ...

1 cos m t

1 cos m t 2 cos s t cos s m t cos s m t

Ts 2 cos 2 s t cos2 s m t cos2 s m t .........
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It can be shown that the output sampled signal is the same as the output PAM
signal when :
T
s

=> ms(t) = VPAM

that is, the pulse width is much smaller compared to the pulse period Ts .

(a) PAM generation v (t)

m

Voltage
translator

vPAM(t)
vd(t)

(b) PAM detection

vPAM(t)

LPF

vm(t)

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Pulse Width Modulation (PWM)

(pulse width) follows the instantaneous value of the information
signal vm(t) :

vm (t )

o o cosmt

o represents the width that is fixed

o 1 cosmt

The equation shows that the pulse

width, of the output signal PWM
varies according to the instantaneous
value of the information signal.

according to the minimum value of

the information signal

Replacing inside the general equation of the sampled signal:

vPWM PWM

Ts

1 2 cos st 2 cos 2st ...

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vPWM PWM

o 1 cos mt
Ts

1 2 cos st 2 cos 2st ...

vPWM

o 1 2 cosst 2 cos 2 st cosmt 2 cos st cosmt

2 cos 2 st cosmt ...

Ts

vPWM

o 1 2 cos st 2 cos 2 st cosmt cos( s m )t

cos( s m )t cos(2 s m )t cos(2 s m )t...

Ts

Generation of PWM signal is by changing the value of sample signal of

the PAM signal into a specific period

PWM generation using voltage

to time converter

vPAM(t)

555 timer

PWM detection using LPF

vPWM(t)

LPF

vPWM(t)

vm(t)
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Pulse Position Modulation (PPM)

(pulse width) follows the instantaneous value of the information

Td vm (t )

signal vm(t) :

T To To cos mt

To represents the period that is

fixed according to the minimum
value of the information signal

T To 1 cos mt

The equation shows that the pulse

period, T of the output signal PPM
varies according to the instantaneous
value of the information signal.

vPPM

Ts

1 2 cos st 2 cos 2st ...

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PPM signal is generated from the PWM signal

At the receiver, the PPM signal must be demodulated to PWM then is

fed to the LPF for the information signal
(a) PPM generation

vPPM(t)

Discrminator
d/dt

Continuity/
directional

Pulse
Generator

vPWM(t)

(b) PWM detection using LPF

vPPM(t)

Integrator &
Clock pulse

LPF

vm(t)

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Example 1:
Determine the Nyquist sampling rate for the baseband signal as follows:
(a) Audio signal 50Hz fm 20kHz
(b) Audio signal 10Hz fm 15kHz
(c) What is the new BW of the sampled signal in (b) if this signal is
sampled 35% higher than the Nyquist frequency??

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Solution:
(a) Audio signal 50Hz fm 20kHz
Sampling theorem states that fs < 2fm (max)
Therefore fs = 2 (20kHz) = 40 kHz
(b) Audio signal 10Hz fm 15kHz
Sampling theorem states that fs < 2fm (max)
Therefore fs = 2 (15kHz) = 30 kHz
(c) From (b) fs =fN = 30 kHz
The new BW = 30kHz+35%(fN) = 1.35 (30kHz) = 40.5kHz

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Example 2:
State two effects when the sampling rate is increased?
i)
ii)

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EXAMPLE
Fig. 1 shows the idealized spectrum of a message signal. The signal is sampled at
a rate equal to 10kHz, 20kHz and 30kHz using uniform pulses. For each case:

i. sketch the spectrum of the resulting signal

ii. comment on your results in (i)
iii. if each of the sampled signals is passed through ideal LPF with the BW of
10kHz, with the help of the spectrum, explain the output

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