DEP30013 : COMMUNICATION

SYSTEM FUNDAMENTALS

1
 COURSE LEARNING OUTCOME

Upon completion of this course, students should be able to:-

1. apply the concept of electronic communication system by using
appropriate diagram and standard formula. (C3, PLO1)
2. assemble the related communication equipment systematically
in performing the measurement of appropriate signals
parameter. (P4, PLO5)
3. demonstrate the ability to work in a team to complete the
assigned tasks during practical work sessions (A3, PLO9)

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CHAPTER 2 : MODULATION TECHNIQUES

LEARNING OUTCOME
2.1 Remember modulation and demodulation.
2.2 Understand modulation and demodulation
2.3 Remember Analog Modulation
2.4 Understand Analog Modulation
2.5 Understand digital information in communication system
2.6. Understand M-ary Encoding
2.7 Remember pulse modulation

3
CHAPTER 2 : MODULATION TECHNIQUES

LEARNING OUTCOME

2.8 Remember Pulse Code Modulation (PCM)

2.9 Understand Pulse Code Modulation
2.10 Understand Sampling process in PCM
2.11 Apply Nyquist Sampling Theorem
2.12 Understand Quantization process in PCM
2.13 Apply the M-ary, Quantization error and SQR formulas
2.14 Remember the digital modulation techniques
2.15 Apply the digital modulation techniques

4
2.1 Remember modulation and demodulation.
2.2 Understand modulation and demodulation

At the end of this learning session, student

should be able to explain :
- Modulation process
- Demodulation process

5
MODULATION & DEMODULATION

6
MODULATION & DEMODULATION

• Because it is often impractical to propagate low

frequency information signals over standard
transmission media, it is often necessary to modulate the
information signal onto a higher-frequency analog signal
called a carrier signal.

• This is because the modulation will transform the low

frequency baseband information signal into a higher
frequency passband signal. For example low-frequency
audio signal into a high radio-frequency (RF) signal.

• In essence, the high frequency carrier signal is used to

carry the low frequency information signal through the
system.

7
WHAT IS MODULATION?

• DEFINITION: Modulation is a process of

changing one or more properties of the high
frequency analog carrier signal in proportion with
the values of information signal.

• The information (modulating) signal modulates

(mengubah) the carrier signal by changing either
its amplitude, frequency, or phase to produce
modulated signal.

• Modulated signal is the carrier signal that have

been modified by information signal.

8
MODULATION PROCESS

Modulator
Modulated Signal (High Frequency)

Information Signal
(Low frequency) Carrier Signal
(High frequency)

 Modulation is performed in a transmitter by a circuit called

a Modulator.
 The information can be in analog or digital form, and the
modulator can perform either analog or digital
modulator.
 The information signal combines with the carrier in the
modulator to produce a high frequency modulated signal.
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DEMODULATION PROCESS

Modulator Demodulator
Modulated Signal

-
(High Frequency)
Information
Signal (Low Demodulated Signal = info
frequency) Carrier Signal (Low Frequency)
(High frequency)

 Demodulation - the reverse process of modulation. It

is a process extracting the information signal from the
modulated-carrier signal.
 Demodulation is performed in a receiver by a circuit
called Demodulator.
 Demodulated signal = Original Information Signal

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TYPES OF MODULATION
Types of Modulation

Analog Modulation Digital Communication

Amplitude Modulation (AM) Pulse Modulation Digital Modulation

Frequency Modulation (FM)

Amplitude Shift Keying
Pulse Width Modulation
Phase Modulation (PM) (ASK)
(PWM)

Pulse Position Modulation Frequency Shift Keying

(PPM) (FSK)
Differential PCM (DPCM)
Pulse Amplitude Modulation Phase Shift Keying (PSK)
Delta Modulation PCM
(PAM)

Pulse Code Modulation Quadrature Amplitude

Adaptive Delta Modulation PCM (PCM) Modulation (QAM)

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TYPES of MODULATION

• In Analog Modulation, both Information Signal and Carrier

signal are in analog waveform.

• In Digital Modulation, Information Signal is in digital

waveform, while Carrier signal is in analog waveform.

• In Pulse Modulation, Information Signal is in analog

waveform, while Sampling signal is in digital waveform.

• Pulse modulation is necessary to convert the analog signal to

digital signal and vice versa for digital transmission.

• while Digital Modulation is necessary to convert the digital

signal to analog signal and vice versa for digital radio.
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TYPES of MODULATION

A summary of the various modulation technique is shown here;

AM FM PM

vc(t) = Vp sin (2πft + Ɵ)

ASK FSK PSK

QAM
where;
v(ct) = time-varying sine wave of Carrier signal voltage
A = peak amplitude (volts)
f = frequency (hertz, Hz)
θ = phase shift (radians)
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WHY MODULATION IS NECESSARY?

1. It is extremely difficult to radiate low-frequency signals from

an antenna in the form of electromagnetic energy. So, we
need to increase the frequency of information signal by
doing the modulation process.

2. To convert the analog signal to digital signal and vice versa

for matching with communication medium and communication
needs.

3. Information signals often occupy the same frequency band.

If signals from two or more sources are transmitted at the
same time, they would interfere with each other. To avoid
interfering with each other, each station(source) converts its
information to a different frequency band or channel by
modulation process.
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WHY MODULATION IS NECESSARY?

4. To increase the bandwidth of the signal.

5. To multiplex more number of signal.
6. To reduce the antenna height and size.
7. To reduce equipment complexity.

Did you know MODEM is

stand for
Modulator-Demodulator?

15
2.3 Remember Analog Modulation
2.4 Understand Analog Modulation

At the end of this learning session, student

should be able to explain :
- Amplitude Modulation
- Frequency Modulation
- Phase Modulation

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ANALOG MODULATION

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ANALOG MODULATION

In Analog Modulation, both Information Signal and

Carrier signal are in analog waveform.
• Amplitude Modulation (AM) - the amplitude (Vp)
of the analog carrier signal is varied proportional
to the analog information signal.
• Frequency Modulation (FM) - the frequency (f) of
the analog carrier signal is varied proportional to
the analog information signal.
• Phase Modulation (PM) - the phase (Ɵ) of the
analog carrier signal is varied proportional to the
analog information signal.
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1. AMPLITUDE MODULATION (AM)

• Definition: Amplitude Modulation (AM) is the process of

changing the amplitude of analog carrier signal in
proportion with the amplitude of the analog information
signal.

• In AM, the amplitude (V) of the carrier signal is varied

proportional to the information signal. While the frequency
(f) and phase (Ɵ) of carrier signal are remains unchanged.

• The carrier amplitude is simply changed according to the

amplitude of the information signal. When the information
signals amplitude is increased, the carrier signal amplitude
also increased and vice versa.

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1. AMPLITUDE MODULATION (AM)

vc(t) Vc sin 2πfct

vm(t) Vm sin 2πfmt

vAM (t) (VcSin 2πfct)( 1  mSin 2πfmt)

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1. AMPLITUDE MODULATION (AM)

• Figure 2.1 shows the AM generation by using a

non-linear AM modulator.

• The information/modulating signal will

modulate the amplitude of carrier signal to
produce high frequency AM modulated signal
by using AM Modulator circuit.

• The shape of AM modulated signal is called AM

envelope. This “envelope” contains the
information signal.
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1. AMPLITUDE MODULATION (AM)
AMPLITUDE MODULATED
Characteristics: MODULATING SIGNAL SIGNAL (VAM)
(Vm) AM Envelope = info
- Low frequency
- eg: audio signal,
voice
- May contains
AM MODULATOR
single frequency
or multiple
frequency such as
human voice.

Characteristics:
CARRIER SIGNAL
(Vc) - The amplitude of carrier signal is
Characteristics: varied by the modulating signal.
- High frequency - Frequency and phase remain
- eg: microwave frequency constant
- frequency and amplitude fixed. - High frequency
Figure 2.1 : Single- Frequency Amplitude Modulation
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AMPLITUDE MODULATION (AM)

Figure 2.2 : Multi -Frequency Amplitude Modulation

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ANGLE MODULATION (FM & PM)

• Frequency Modulation (FM) and Phase Modulation (PM) are

both forms of Angle Modulation.
• This is because whenever the frequency of a carrier is
varied, the phase is also varied and vice versa.
• Therefore, FM and PM must both occur whenever either
form of angle modulation is performed.
• The difference between FM and PM lies in which property of
the carrier (the frequency or phase) is directly varied by
modulating signal and which property is indirectly varied.
• If frequency is varied directly in accordance with modulating
signal – FM.
• If phase is varied directly in accordance with modulating
signal – PM.
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ANGLE MODULATION (FM & PM)

Carrier Signal

Information Signal

Frequency Modulated
(FM) Signal

Phase Modulated
(PM) Signal

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2. FREQUENCY MODULATION (FM)

• Definition: FM is the process of changing the frequency

of analog carrier signal in proportion with the amplitude
of the analog information signal.

• In FM, the carrier amplitude and phase remains constant

while the carrier frequency is varied by the modulating
signal.

• The amount of carrier frequency changes is proportional

to the amplitude of the information signal. As the
modulating signal amplitude increases, the carrier
frequency increases and vice versa.

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2. FREQUENCY MODULATION (FM)

vc(t) Vc sin 2πfct

vm(t) Vm sin 2πfmt

vFM (t) Vc Cos[2πfct  mf Sin 2fmt ]

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2. FREQUENCY MODULATION (FM)

• This modulation is perform by a circuit called FM

Modulator as shown in Figure 2.3 below.

Figure 2.3: FM Modulator

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3. PHASE MODULATION (PM)

• Definition
PM is the process of changing the phase of
analog carrier signal in proportion with the
amplitude of the information signal
• In PM, the carrier amplitude and frequency
remains constant while the carrier phase is
varied by the modulating signal.
• As the modulating signal amplitude increases,
the carrier phase increases and vice versa.

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3. PHASE MODULATION (PM)

vc(t) Vc sin 2πfct

vm(t) Vm sin 2πfmt

Phase Modulated Signal

vPM (t) Vc Cos[2πfct  mp Sin 2fmt ]
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WAVEFORM & MATHEMATICAL EXPRESSION

vm(t) Vm sin 2πfmt

vc(t) Vc sin 2πfct

vAM (t) (VcSin 2πfct)(1  mSin 2πfmt)

vFM (t) Vc Cos[2πfct  mf Sin 2fmt ]

vPM (t) Vc Cos[2πfct  mpSin 2fmt ]

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 COMPARISON OF FM AND PM

Phase Modulation Frequency Modulation

• the phase of the carrier • the frequency of the carrier
waveform varies with the waveform varies with the
information signal. information signal.
• Equation: • Equation:
m(t) Vc Cos[2πfct  mp Cos2fmt ] m(t) Vc Cos[2πfct  mf Sin2fmt ]

from equation m(t ) Vc cos[2f c t   (t )]

• By varying the phase, θ • By varying the frequency, fc

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2.5 Understand digital information in
communication system
At the end of this learning session, student
should be able to explain :
- The difference between digital radio and digital
transmission.
- Digital communication block diagram
- Advantages of the digital communication
compares to analog modulation.

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DIGITAL COMMUNICATION

34
DIGITAL COMMUNICATION

• The term Digital Communication covers a broad range of

communication techniques including digital transmission
and digital modulation (digital radio).

• Digital transmission is a true digital system where digital

signals are transferred between two or more points in a
communication system.

• Digital signals could be a binary digit (bit 0 and bit 1) or

other form of discrete-level digital pulses.

• With digital transmission, there is NO analog carrier and the

original source information may be in digital or analog form.
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DIGITAL COMMUNICATION

• If the information signal is in analog forms, it must be

converted to digital pulses prior to digital transmission
and converted back to analog form at the receive end.

• The analog signal is converted into digital signal by using

Pulse Code Modulation (PCM) technique.

• Since digital pulses CANNOT be propagated through a

wireless transmission medium (free space) ; therefore, the
digital transmission required physically medium such as a
metallic cable (twisted, coaxial cable) or an fibre optic cable.

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DIGITAL COMMUNICATION

• Digital Modulation (Digital Radio) is a transmittal of

digitally-modulated analog carrier signals between two
or more points in a communication system.

• With digital radio, the information signal and

demodulated signal are in digital form. While the
carrier signal and modulated signal are in analog
form.

• The digital pulses could be originated from computer-

generated data or digital transmission system or
digitally-encoded analog information signal.
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DIGITAL COMMUNICATION

• In digital radio system, digital pulses modulate

the analog carrier signal to produce digitally-
modulated carrier signal in analog form.

• Since the modulated signal is in analog form,

therefore the transmission medium could be a
wireless transmission medium(free space) or
physically facility (metallic or fiber optic cable).

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TYPES of DIGITAL COMMUNICATION

DIGITAL COMMUNICATION SYSTEM

Digital Transmission Digital Radio

Pulse Modulation Digital Modulation

Pulse Width Modulation (PWM) Amplitude Shift Keying (ASK)

Pulse Position Modulation (PPM) Frequency Shift Keying (FSK)

Pulse Amplitude Modulation (PAM) Phase Shift Keying (PSK)

Delta
Pulse Code Modulation (PCM) Modulation
Quadrature Amplitude
Differential PCM Modulation (QAM)
Delta-Sigma (DPCM)
Modulation
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Basic Elements of Digital Communication
System

Information
Source
Source Encoder Channel Encoder Digital Modulator
ASK,FSK,PSK Signal
Analog, digital signal Message Symbol Code Symbol
(Bits) (Bits) Channel
ASK,FSK,PSK Signal

Information
Source Decoder Channel Decoder Digital Demodulator
Sink

Figure 2.4 : Basic Elements of Digital Communication System

Source Encoder
Digital Information
Textual Information
Analog Information Sampler Quantizer Encoder Channel Encoder

Figure 2.5 : Formatting Process

40
Basic Elements of Digital Communication
System

1) Information Source - The source of information can be analog or

digital, e.g. Analog- audio, voice; Digital- teletype signal.
2) Source Encoder – to convert the information signal from source into
digital signals (serial bits) by formatting the signals (refer Figure 2.5)
and compressed that signal. Then, these bits are grouped to form
message symbols. For example: PCM process, Character Encoding
(ASCII code) process.
3) Channel Encoder – is used for error correction coding. It can reduces
the probability of error by introduces some redundancy in the message
symbols and transform it into code symbols(code words).
4) Digital Modulator - to convert the serial bits (digital waveform) into
electric signals (analog waveform) so that we can transmit them on
channel. For example ASK, FSK & PSK Modulation process.
41
Basic Elements of Digital Communication
System

5) Channel - The communication channel is the physical medium that

is used for transmitting signals from transmitter to receiver. In
wireless system, this channel consists of atmosphere. For telephony,
this channel is wired like twisted pair cable & optical fiber.
6) Digital Demodulator – the reverse process of modulation and
converts the electric signals back to the serial bits (code symbols).
7) Channel Decoder - to reconstruct the original serial bits(message
symbols) from the code symbols used by the channel encoder and
the redundancy contained in the received data. Example: Bit Error
Rate (BER) process.
8) Signal Decoder - to convert back the serial bits(message symbols)
into original source information signal.
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ADVANTAGES of DIGITAL COMMUNICATION

The advantages of digital transmission compared to analog

transmission are;
i. Noise Immunity - Digital signals are less susceptible than analog signals
to interference caused by noise.
ii. Reduction of errors - Errors caused by noise and interference can be
detected and corrected systematically.
iii. High security - Digital system more secure than analog system because
the system can be encoded digital data to a unique code (data encryption)
and data can only be understood by the sender and receiver only.
iv. Digital circuit easier to be interfaced compare to analog circuits (because
there are two levels of digital signals only '1 'and '0').
v. Ease of processing and multiplexing.
vi. Inexpensive digital circuitry may be used extensively.
43
APPLICATION OF DIGITAL COMMUNICATION

• ADC – Analog to Digital Converter

• DAC – Digital to Analog Converter
• MODEM – Modulator-Demodulator
• Digital Camera
• Digital Video
• Broadband digital subscriber lines (DSL)
• Telemetry
• Teleconferencing
• Compact Disk (CD)
• Hard Disk Drive
• Personal Communication System (PCS)
• Satellite Communication System
44
2.6 Understand M-ary encoding

At the end of this learning session, student

should be able to explain and solve the problem:
- M-ary encoding

45
M-ARY ENCODING

46
M-ary CODING
• M-ary is a term derived from the word binary.

• M = represents a digit that corresponds to the

number of conditions or levels or combinations
possible for a given number of binary variables
(n).

• For example, a digital signal with 4 possible

conditions (either voltages, levels, frequencies,
phases and so on) is an M-ary system where
M = 4.

47
M-ary CODING
• The number of conditions, M possible with n bits is
expressed mathematically as;
M 2 n
Where, n = number of bits
M = number of conditions, or levels, or
combinations
possible with n bits

• The number of bits,n that necessary to produce a

given number of cxonditions, M is expressed
mathematically as;
n log2 M
48
M-ary CODING
Example 2.1
For example, with n = 1 bit, only 21 = 2 conditions are
possible. With two bits, 22 = 4 conditions are possible.
With three bits, 23 = 8 conditions are possible, and so on.

Example 2.2
A digital signal has two levels. How many bits are needed
per level? Draw that digital signals.

Example 2.3
A digital signal has four levels. How many bits are needed
per level? Draw that digital signals.

49
M-ary CODING
Solution example 2.2
n log 2 M number of levels, M = 2
n log 2(2) number of bits per level, n = 1 bit
log 2
n 1 bit
log 2

50
M-ary CODING
Solution example 2.3
n log 2(4) number of levels, M = 4
log 4 number of bits per level, n = 2 bits
n 2 bits
log 2

51
2.7 Remember Pulse Modulation

At the end of this learning session, student

should be able to explain:
- Pulse Modulation techniques
- Draw the diagram of Pulse Modulation
techniques (PWM, PPM, PAM and PCM)

52
PULSE MODULATION

53
PULSE MODULATION

• In Pulse Modulation, Information Signal is in

analog waveform. While Sampling signal is in
digital waveform. (there is NO carrier signal in
pulse modulation)

• This modulation is necessary to convert the analog

signal to digital signal for digital transmission.

• Usually used metallic cable & fiber optic cable.

Cannot use free space as channel.

54
PULSE MODULATION TECHNIQUE

• Definition: Pulse Modulation(PM) is a process of

sampling the analog information signals into sampled
signal before converting those into digital signals.

• In Pulse Modulation the Information Signal is in

analog waveform. While Sampling signal is in digital
pulses waveform.

• The properties of sampling pulses signal such as

width, position and amplitude will be varied in
proportion with amplitude of information signal.

55
PULSE MODULATION TECHNIQUE

• There are 4 predominant techniques for

Pulse Modulation :
a) Pulse Width/Duration Modulation
(PWM @ PDM)
Analog PM
b) Pulse Position Modulation (PPM) (sampling only)
c) Pulse Amplitude Modulation (PAM)
d) Pulse Code Modulation (PCM) Digital PM

56
Figure 3.3 : Pulse Modulation Pulse Modulation
Technique
Information signal

Sampling Pulses

PWM

PPM

PAM
(Single Polarity/Flat Top Sampling)

PCM
57
PULSE MODULATION TECHNIQUE
• PWM - the width of the pulses is varied proportional to the analog
amplitude information signal.
(The higher amplitude of Information signal, the wider of pulse.)
• PPM – the position of the pulses is varied proportional to the analog
amplitude information signal.
(The higher amplitude of Information signal, the farther to the right the
pulse is positioned).
• PAM - the amplitude of the pulses is varied proportional to the analog
amplitude information signal.
(The higher amplitude of Information signal, the higher amplitude of
pulse).

• PCM – With PCM, the analog information signal is sampled into PAM
signal and then converted to a serial n-bit binary code for transmission.
58
PULSE WIDTH MODULATION (PWM)

 Width (W) of pulses is varied proportional to the amplitude of

analog Information signal.
 Amplitude (A) and Position (P) of pulses are constant.
 The higher amplitude of Information signal, the wider of
Pulses.
59
PULSE POSITION MODULATION (PPM)

 Position (P) of pulses is varied proportional to the amplitude of

analog Information signal.
 Amplitude (A) and Width (W) of pulses are constant.
 The higher amplitude of Information signal, the farther to the
right the pulse is positioned.

60
PULSE AMPLITUDE MODULATION (PAM)

Analog Information Signal

PAM

Sample pulses

 Amplitude (A) of pulses is varied proportional to the amplitude

of analog Information signal.
 Width (W) and Position (P) of pulses are constant.
 The higher amplitude of Information signal, the higher
amplitude of Pulses.

61
PULSE MODULATION TECHNIQUE

• PWM and PPM are used in special-purpose

communication systems, seldom used for commercial
digital transmission.

• PAM is used as an intermediate form of modulation with

PCM, PSK and QAM; seldom it is used by itself.

• Among all pulse modulation techniques, there are two

important digital pulse modulation techniques which are
Pulse Code Modulation (PCM) and Delta Modulation
(DM).

62
DIGITAL PULSE MODULATION (DPM)

• In DPM, a code is used to represent the amplitude

of the samples that has been divided into various
levels.

• The types of DPM:

1. Pulse Code Modulation (PCM)
2. Delta Modulation (DM)
3. Delta-Sigma Modulation

• The further sub-topic will discuss only the PCM

techniques in details.
63
2.8 Remember Pulse Code Modulation

At the end of this learning session, student

should be able to :
- Define Pulse Code Modulation
- Know other types of PCM
- State the application of PCM

64
PULSE CODE MODULATION

65
Pulse Code Modulation (PCM)

• In digital transmission, any analog information data should be

changed into digital signal for the digital transmission.
• PCM is the only digitally encoded modulation technique that
is commonly used for digital transmission .

• The term Pulse Code Modulation is somewhat of a misnomer,

as it is not real a type of modulation but rather a form of
digitally coding analog signals.
• PWM, PPM, and PAM signals are digital signal (discrete-time
signal), but those signals does not represent in a single
binary digit (bits).
• Therefore, PCM technique is needed to convert the discrete
sampled signal(usually PAM) to serial bits. 66
Pulse Code Modulation (PCM)

• Definition: PCM is a digital pulse modulation

technique to convert the analog signal to digital signal.

• The PCM technique is the conventional/basic digital

pulse modulation technique. However, there are few
others type of PCM technique such as;
1. Delta Modulation
2. Delta Sigma Modulation

67
Pulse Code Modulation (PCM)

APPLICATION:
• In electronic communication circuit, the PCM technique is
applied at
– Analog to Digital Converter (ADC) device - in Coder
– Digital to Analog Converter (DAC) device - in Decoder
– In digital telephony Multiplexing (TDM-PCM)
– Digital PABX
– Digital Audio recording
– CD laser disks, etc.

• Figure 2.6 shows the PCM transmission block diagram.

Figure 2.7 shows the PCM Encoder(transmitter); while
Figure 2.8 shows the PCM Decoder(receiver). 68
2.9 Understand Pulse Code Modulation

At the end of this learning session, student

should be able to explain and solve the problem:
- Pulse Code Modulation steps
- Sampling process
- Quantizing process
- Encoding process

69
Figure 2.6 Block Diagram of PCM - Full

(a) PCM Transmitter PAM

Analog Signal Analog-to-Digital converter

Bandpass Sampler &

Quantizer Encoder
Filter Hold

Sampling Pulses
Serial PCM code (bits)

(b) Transmission medium

Regenerative
Repeater

Analog Signal
Decoder & Lowpass
Hold Filter
(c) PCM Receiver

Digital-to-Analog converter PAM

70
PCM Block Diagram

Figure 2.6 shows a simplified block diagram of a simplex PCM

system:

1) Bandpass Filter – the bandpass filter limits the analog input

signal to a certain bandwidth (fmin to fmax) to enter the Sampler.
For example: the filter allows only human voice bandwidth,
300Hz - 3.4kHz to enter the Sampler circuit.
2) Sampler & Hold – Periodically samples the analog input
signal and convert the Sampling Pulses signal to a multilevel
sampled PAM signal.
3) Quantizer - Convert the sampled PAM signal to quantized
PAM signal by rounding off the amplitude of sampled signal to
quantization levels, L.

71
PCM Block Diagram

Figure 2.6 shows a simplified block diagram of a simplex PCM

system:
4) Encoder – convert the quantized PAM signal to a parallel code
number. Then, convert the code numbers to a serial binary
pulses (encoded words).
5) Repeater – Amplify and regenerate the weaken digital pulses
during transmission process on transmission line.
6) Decoder & Hold – Convert back the digital pulses signal to
multilevel PAM signals.
7) Low pass Filter - to smooth the staircase amplitude of PAM
signals into an analog signal.

72
Pulse Code Modulation (PCM)

• PCM consists of three steps to digitize an analog signal:

i. Sampling
ii. Quantizing
iii. Encoding
• Before sampling process, the information signal should be
filtered to limit the maximum information frequency (fmmax)
that can enter into the sampler as it affects the sampling rate
(fs).
• Filtering should ensure that we do not distort the signal, i.e
remove high frequency components that affect the signal
shape.
73
Figure 2.7 Block Diagram of PCM Encoder (Transmitter)

Information signal

Bandpass Filter

Sampler Quantizer Encoder

& Hold

Filtered signal
e.g : 300 – 3400Hz

fmmin fmmax

74
2.10 Understand sampling process in PCM
2.11 Apply Nyquist Sampling Theorem

At the end of this learning session, student

should be able to explain and solve the problem:
- Sampling process
- Nyquist Sampling Theorem

75
1. SAMPLING
• Definition: Sampling is a process where the information
signal (in analog signal) is sampled by sampling pulse signal
which is generated at certain sampling rate,fs.
• Sampling process will convert an analog signal (in
continuous-time signal) to a sampled signal (in discrete-
time signal) either in PAM, PWM or PPM.
• For PCM, the sampled signal is PAM signal.
• By this process, the amplitude of pulses signal is varied
proportional to the analog amplitude of information signal.

76
1. Sampling
Figure 2.9
• According to Figure 2.9, Analog
information signal is sampled every
TS seconds.
Info signal • Ts is referred to as the Sampling
Interval or Sampling Period.
• The amplitude of sampling pulse
signal is varied according to
amplitude of information signal
-TS 0 TS 2TS 3TS
which produce a PAM signal.
Sampling Pulse Signal
• There are 3 methods of sampling
which are
i. Ideal Sampling
ii. Natural Sampling
-TS 0 TS 2TS PAM Signal iii. Flat-Top Sampling
77
Figure 2.10 : Three different sampling methods for PCM

Ts

Ts

78
1. SAMPLING
• Ideal Sampling – the analog information signal is sampled
instantaneously by pulses. This sampling is not practical
and cannot be easily implemented.

• Natural Sampling – The more practical sampling which is

performed by high-speed switching circuits as shown in
Figure 2.11.

• Flat-Top Sampling – The simplest and the most popular

sampling method which is performed by Sample-and-
Hold circuit as shown in Figure 2.12. However, this circuit
creates a flat-top (staircase) sampled signal.

79
1. SAMPLING
Sample Switch

fm fs

Figure 2.11: Natural Sampling

• When the pulse is generated, the

switch will CLOSED, and the
amplitude of information signal
will be produced.

• When there is no pulse, the switch

will OPEN and there is no output Figure 2.12: Flat-top Sampling
will produce. 80
1. SAMPLING

• Ts is referred to as the Sampling Interval or Sampling Period.

• fs = 1/Ts is called the Sampling Rate or Sampling Frequency.

• From Figure 2.13, it could be seen that the higher the sampling
rate, fs the smaller sampling interval, Ts, the closer the
recovered signal approaches the original signal.

• Ideally, an infinite sampling rate would be desirable in terms of

reproducing the original signal but it is not practical due to the
bandwidth limitation.
So, what minimum Sampling Rate, fs is required
to recover the original signal?
81
Figure 2.11 : Three different Sampling Rate for PCM

The recovered Signal

could be known by
connecting all the
dotted-amplitudes of
PAM signal.

Original
Information
Signal is
recovered.

Original
Information
Signal is NOT
recovered.

82
SAMPLING THEOREM
According to the Nyquist Sampling theorem; to reconstruct the
original signal, the sampling rate must be at least (minimum)
two times the highest frequency contained in the info signal.

fs ≥ 2fm(max)

EXAMPLE 2.4 :
Given the bandwidth of the telephone lines signal is 300 to 3400 Hz.
Determine the minimum sampling rate that suitable for sampling that signal
and sampling interval required.
ANSWER :
fs = 2fm(max) = 2 ( 3400Hz) = 6800 Hz @ 6800 samples/s
Ts = 1/ fs = 1/6800Hz = 0.147ms
83
SAMPLING THEOREM
EXAMPLE 2.5 :
• For an intuitive example of the Nyquist Theorem, let us sample a
simple sine wave at three sampling rates:
i) fs = 2fm (Nyquist rate),
ii) fs = 2(2fm) (2 times the Nyquist rate), and
iii) fs = ½(2fm) (one-half the Nyquist rate).

ANSWER:
From Figure 2.12 below;
• For part (a), it can be seen that the sampling at the Nyquist rate can
create a good approximation of the original sine wave
• Oversampling in part (b) can also create the same approximation, but it is
redundant and unnecessary.
• Sampling below the Nyquist rate (part c) does not produce a signal that
looks like the original sine wave. 84
Figure 2.12 : Recovery of a sampled sine wave for different sampling rates

Original Signal Reconstruct Signal

85
Nyquist-Shannon Sampling Theorem
Claude Shannon – father of information theory

Harry Nyquist
86
2.12 Understand Quantization process in PCM

At the end of this learning session, student

should be able to explain :
- Quantization process
- Quantization Error (Qe)
- Signal-to-Quantization Noise Ratio (SQR)

87
2. QUANTIZATION
• Definition: Quantization is a process rounding off the amplitudes
of sampled (PAM) signal to a countable number of quantization
levels.
• Analog signal has an infinite (uncountable) number of amplitude
possibilities. By using the quantization process, the amplitudes of
sampled PAM signal is rounding off to a finite(countable) set of
quantization levels, L.
• The number of amplitude levels, L for the quantization depend on
the number of bits, n used to code the signal.
• It use M-ary formula to determine the number of quantization
levels, L. n n = number of bits per level
L 2 L = number of finite quantization level
88
2. QUANTIZATION
• For example, if 3 bits is used to code the signal, therefore the
number of quantization levels, L are;
L 2 n 23
L 8 levels

• The more levels, L used means that an analog signal can be

described more accurate during signal recovery at receiver.
• This is because the more number of bits (n) and quantization
level (L) are used, the more quantization error (Qe) could be
reduced and the more accurate the recovery original signal.
• However, a PCM code could have only 8 bits maximum which
equates to only L= 28 or 256 levels.
89
2. QUANTIZATION

Code Quantized signal = 3.5V Sampled signal = 3.1V

number

7 ∆ = Step Size / Quantization

Interval / Resolution /
Quantum
6
∆ = 2 Vmax/L
5

1
Original
0 analog
Mid-point signal
Figure 2.13: Quantization 90
2. QUANTIZATION

• According to Figure 2.13 above, we assumed that the amplitude of

sampled PAM signal is confined between two limits: Vmax = 4V and Vmin
= -4V.
• The amplitude values of sampled PAM signal are infinite between
these two limits. So, we need to map the infinite amplitude values
onto a finite set of known values (L zone).

• This is achieved by dividing the distance between Vmax and Vmin into L
zones, each of height of step size, ∆.
• Since we want to use 3 bit PCM code (n=3), so the Quantization Level
is; L = 2n
L = 23 = 8 level

• Step size, ∆ = 2 Vmax/L = 2(4)/8 = 1V

91
2. QUANTIZATION

• The 8 zones are: -4V to -3V, -3V to -2V, -2V to -1V, -1V to 0V,
0V to 1V, 1V to 2V, 2V to 3V, and 3V to 4V.
• The midpoints are: -3.5V, -2.5V, -1.5V, -0.5V, 0.5V, 1.5V, 2.5V, and
3.5V.
• This midpoint is called quantization level, L. The midpoint of
each zone is assigned a value from 0 to L-1 quantization levels.
• Each sampled signal’s amplitude is quantized (rounding-off) to
the midpoint of the interval (quantization level) in which it lies.
• For example in Figure 2.13 above, the second sample has
sampled amplitude value of 3.1V. After quantization, the
sampled amplitude value is quantized to 3.5V level.
92
2. QUANTIZATION
• There are two types of Quantizing method which are;
i. Uniform Quantization – uniform step size, ∆
ii. Non-uniform Quantization – non-uniform step size, ∆
• The previous example is Uniform quantization where the step size is uniform
for each zone.
• However, a non-uniform quantization is commonly used because the
uniform quantization is not efficient for a signal that has smaller amplitude.
• For example, in speech communication (voice signal), the signal is found have
more smaller amplitudes rather than larger amplitudes
• Thus, uniform quantization scheme is wasteful for speech signals because
many of quantization levels, L are rarely used. The non-uniform quantizing
method is more suitable because the step size could be adjusted depends on
the amplitude of signal (smaller step size for lower amplitude and larger for
higher amplitude).
93
2.13 Apply the M-ary, Quantization error and SQR
formulas

At the end of this learning session, student

should be able to solve the problem:
- Quantization process
- Quantization Error (Qe)
- Signal-to-Quantization Noise Ratio (SQR)
- Bit rate

94
Quantization Error (Qe)
• When a signal is quantized, we introduce an error because
Quantization is an approximation process.
• The difference between sampled and quantized value is referred
as the quantization error (Qe).
Qe Quantized value - Sampled value (V)

• For example, for second sample in Figure 2.13, the sampled

value is 3.1V, while the quantize value is 3.5V. So;
Qe = quantized value – sampled value
= 3.5V – 3.1 = 0.4V
• Quantization error (Qe) is also called Quantization noise (Qn)
where the maximum error is Qe = ± ∆/2 for Uniform
Quantization.
95
Signal to Quantization Noise Power Ratio
(SQR)
• The Signal-to-Quantization Noise Power Ratio (SQR) is defined by the
equation:
SQR(dB) 6.02n  1.76 dB n = number of bits

Example 2.6: What is the SQR in the Figure 2.13?

Solution:
From the Figure 2.13, the number of quantization level, L = 8. Then the
log 8
number of bit is; n log 2 L log 2 8  3bits
log 2

Since, numberSQR(dB)
of bits is 3,
so6.02(3)
the SQRis1.76 19.82dB
96
Qe, SQR and BIT RATE
• From the SQR equation, it could be seen that the values of
SQR is depends on the number of bits, n.
• The higher number of bits, the higher value of SQR could be
achieved, the more quantization error could be reducing,
and the more accurate recovery signal could be achieved.
• This is because when the number of bit, n is increased, the
number of quantization level, L also increase and the step
size, ∆ will become smaller.
• When the step size become smaller, the amplitude difference
(gap) between sampled signal and quantized signal could be
minimized (or maybe no gap) which results in smaller errors.

97
Qe, SQR and BIT RATE
• Thus, the recovery original signal is more accurate when
the quantization errors are reduced.
• In conclusion, the quality of sampled PAM signal can be
improved by using a PCM code with more bits, n. BUT, the
more bits will introduce higher bit rate.
• Bit Rate is the number of bits transmitted during one
second and is expressed in bits per second (bps).
• The Bit Rate for PCM could be found from the formula;
Bit Rate = fS x n fS = sampling rate
n = number of bits per sample

98
BIT RATE
EXAMPLE 2.7
We want to digitize the human voice. What is the bit rate,
assuming 8 bits per sample?

SOLUTION Example 2.7

The human voice normally contains frequencies from 300 to
3.4kHz. So, the sampling rate and bit rate are calculated as
follows;
Sampling Rate, fS = 2fmmax = 2 x 3.4kHz = 6800 samples/s
Bit Rate = fS x n = 6800 x 8 bits = 54400 bps = 54.4kbps

99
EXERCISE 1
A signal has a bandwidth of 10 Mhz. The signal is sampled, quantize
and binary coded to obtain a pulse-code modulated (PCM) signal. The
signal is sampled at the Nyquist sampling rate.

(a) Calculate the Nyquist sampling rate?

(b) If the samples are to be encoded into 128 levels, what is the number of
binary pulses(bits) required to encode each sample?
(c) Based on your answer to part (a) and (b) what is the minimum binary
pulse rate (bits per second) of the binary coded signal?

100
101
3. ENCODING
• Definition : Encoding is a process of translating the quantized
signal into a decimal code number. Then this decimal code
number is converted to its representative binary sequence.

• The number of bits, n for each level of code number depends

on the number of quantization level, L used to quantize the
samples which can be determined using M-ary formula.
n log 2 L

• The quantizing and encoding operations are usually performed

in the same circuit which is called analog-to-digital converter
(ADC).

102
3. ENCODING
• The Decimal Code Number for each quantization level is
converted to its representative binary sequence by using
Binary Code or Grey Code or Folded Binary Code.
Code Number Binary Code Gray Code
7 111 110
6 110 111
5 101 101
4 100 100
3 011 000
2 010 001
1 001 011
0 000 010
• The essential features of binary PCM are shown in Figure 2.14

103
3. ENCODING
Binary Quantized signal Sampled signal
code
111

110

101

100

011

010

001

000

Figure 2.14

104
3. ENCODING
• From figure 2.14 above, we assign the decimal code
number 0 to the level -3.5V, the code number 1 to level -
2.5V, and so on until level +3.5V.
• Each decimal code number (0 - 7) has its own 3 bits
binary code representation, ranging from 000 for code
number 0 to 111 for code number 7.

• Therefore, the binary sequences (digital signal) that

produce from PCM are
011 111 101 001 000.

105
Example 2.8
Encode the following quantized signal and find the
quantization error for each samples.

4V
3.94
3.24
3V
2.2
2V
1.5
1V

0V
-1.2
-1V
-1.22 -1.1 -1.88
-1.88
-2V
-2.26
-3V

-4V

Figure 2.15
106
Solution Example 2.8

3.5V 3.94
3.24
2.5V 2.2

1.5V 1.5

0.5V
-0.5V
-1.2
-1.5V -1.22 -1.1
-1.88
-2.5V -2.26

-0.5V

-0.28

107
EXERCISE 1
Q. Encode the following Quantized signal

step size is
distributed evenly
Solution:
No. of level, L = 16
Therefore, no. of bits
for each level is
n = log2 L
n = log2 (16)
n = 4 bits

Figure 2.16
108
EXERCISE 2
A signal in the frequency range 300-3400Hz is limited to a peak swing of 8V. The
signal is sampled using a minimum sampling rate for digital transmission and the
samples are quantize to 8 evenly space level. Calculate the frequency sampling,
the step size value, transmission bit rate, and the signal to quantization noise
power ratio (SQR). Transfer each of the quantize signal in figure below into a code
word and serial bit.
PCM DECODER
• According to Figure 2.8, to recover an analog signal from
a digitized signal we follow the following steps:
1. We use a decoder and hold circuit that holds the
amplitude value of a pulse till the next pulse arrives.
This will produce a staircase PAM signal.

2. We pass this PAM signal through a low pass filter

which has the same cutoff frequency as the original
information signal at sender. The filter will smooth
the staircase amplitude of PAM signals into an
analog signal.

4.110
Figure 2.8 Block Diagram of PCM Decoder (Receiver)

PAM signal

Decoder
& Hold

111
PCM DECODER
• If the original info signal is sampled at (or greater than)
Nyquist Sampling Rate AND if there are enough
Quantization levels, the original signal would be
recovered back with less distortion.

• The higher the value of quantization level L, the less

distorted a signal is recovered.

Did you know CODEC

is stand for
Coder- Decoder?

112
Differential PCM (DPCM) &
Adaptive Differential PCM (ADPCM)

113
DIFFERENTIAL PCM
• In a typical PCM-encoded speech (human voice)
waveform, the successive samples signal has a little
difference of amplitude between two sampled signal.
• This necessitates transmitting several identical(same) PCM
codes, which is redundant.

• Therefore, Differential Pulse Code Modulation (DPCM) is

designed specifically to remove this redundancy in PCM
technique.
• With DPCM, the Prediction Error is quantized, encoded and
transmitted rather than actual samples themselves.
114
DIFFERENTIAL PCM
• Definition: Differential pulse code modulation (DPCM) is a
technique of converting an analog into a digital signal in
which an analog signal is sampled and then the difference
between the actual sample value and its predicted sample
value is quantized and then encoded forming a digital value.
• Predicted Sample = is a value where the current sample is
predicted based on previous sample.
• The difference between the actual sample value and its
predicted sample value = Prediction Error
• Figure 2.17 shows the DPCM Transmitter; while Figure 2.18
shows the DPCM Receiver.
115
DIFFERENTIAL PCM
Sampled
PAM
signal

Prediction
Error Figure 2.17: DPCM Transmitter

Figure 2.18: DPCM Receiver

116
2.14 Know and understand Digital Modulation
2.15 Apply the digital modulation techniques

At the end of this learning session, student

should be able to explain :
- Digital Modulation techniques
- Application of Digital Modulation

117
DIGITAL MODULATION

118
DIGITAL MODULATION

• Digital Signal cannot be transmitted through free

space (wireless) medium but Analog signal does.
• Therefore, digital data needs to be converted into
analog signal by doing the Digital Modulation
techniques.

• Digital Modulation is the process of changing one of

the characteristics of an analog carrier signal
based on the information in digital data.
• A carrier signal(fc) performs the function of
transporting the digital data in an analog waveform.

119
Figure 2.19 Digital-to-analog conversion

120
Figure 2.20 Types of Digital Modulation

Digital Modulation

121
DIGITAL MODULATION

In Digital Modulation, Information Signal is in digital

waveform; while Carrier signal is in analog waveform.
• Amplitude Shift Keying (ASK) - the amplitude (Vp)
of the analog carrier signal is varied proportional to
the digital information signal.
• Frequency Shift Keying (FSK) - the frequency (f) of
the analog carrier signal is varied proportional to
the digital information signal.
• Phase Shift Keying (PSK) - the phase (Ɵ) of the
analog carrier is varied proportional to the digital
information signal.

122
 Digital Modulation Techniques

Carrier signal
fC

Data

ASK

f0 f1

FSK

PSK

Figure 3.20 123

Amplitude Shift Keying (ASK)

• ASK is the simplest digital modulation techniques.

• Also called Digital Amplitude Modulation (DAM) or On-Off Keying (OOK).
• ASK is a process where the binary information signal directly modulates
the amplitude of an analog carrier.
• ASK is similar to standard amplitude modulation except there are only
two output amplitudes possible. Both frequency and phase remain
constant.
• When the binary data is logic ‘1’, the carrier signal has the constant
amplitude (Vp = A cosωct). When the data is logic ‘0’, the carrier signal
has no amplitude (Vp=0V).
• Whenever the binary input is ‘high’ (logic 1), the output of carrier is a
constant-amplitude, constant-frequency signal. While, when the binary
input is ‘low’ (logic 0), the carrier is off.
124
Figure 2.21: Binary Amplitude Shift Keying

125
Figure 2.22: Implementation of binary ASK

126
Frequency Shift Keying (FSK)

• FSK is another relatively simple, low-performance type of digital

modulation.
• FSK is a form of angle modulated, constant-amplitude similar to
standard FM except the information signal is a binary signal that
varies between two discrete voltage levels.
• Sometimes called binary FSK (BFSK).

• With FSK, the carrier centre frequency (fc) is shifted (deviated) up

and down in the frequency domain by the binary input as shown in
above figure 3.20.

• As the binary input signal changes from a logic 0 to a logic 1 and vice
versa, the output frequency shifts between two frequencies: logic 1 -
frequency (f1) and logic 0 - frequency (f0).
127
Figure 2.23: Binary Frequency Shift Keying

128
Figure 2.24: Implementation of binary FSK

129
Phase Shift Keying (PSK)

• PSK is another form of angle-modulated, constant-

amplitude digital modulation.
• PSK is an M-ary digital modulation scheme
• Similar to conventional PM except with PSK the input is a
binary digital signal and there are a limited number of
output phase possible.
• The input binary information is encoded into groups of bits
before modulates the carrier.
• A group has n bits (n = 1….12).
• The number of output phases is defined by M = 2n.

130
Phase Shift Keying (PSK)….cont

• The simplest form of PSK is binary phase shift keying (BPSK)

where n=1 and M=2.
• Therefore two phases are possible (21 = 2) for the carrier
which are logic ‘1’ and logic ‘0’.

• One phase represents a logic ‘1’ and other phase represents

logic ‘0’.

• As the input digital signal changes (i.e. from a 1 to a 0 or

from a 0 to a 1), the phase of the output carrier shifts
between two angles that are separated by 180º as shown on
above figure.
131
Figure 2.25: Binary Phase Shift Keying

132
Figure 2.26 Implementation of BPSK

133
 REFERENCES
• Wayne T. (2004). Electronic Communication Systems:
Fundamentals Through Advance (6th ed.). Prentice Hall. ISBN-
10: 0130453501 or ISBN-13: 9780130453501
• M. Forouzan, B.A. (2012). Data Communications and
Networking (5th ed). Mc Graw Hill. (ISBN: 978-0-07-131586-
9)
• Hwei Hsu (2002). Schaum’s Outline of Theory and Problems
of Analog and Digital Communications (2nd ed). McGraw-Hills.
ISBN-10: 0071402284. ISBN-13: 978-0071402286
• Miller, Gary M. (2008). Modern Electronic Communication
(9th ed.). Prentice Hall. ISBN: 0-13-225113-2.
134