ELECTRICAL ENGINEERING

DEPARTMENT

DEP30013

SYSTEM FUNDAMENTALS

1
 CHAPTER 2:

DIGITAL
MODULATION TECHNIQUES

2
Digital Communication
• Advantage of digital communication
- Ease of signal regeneration,
- data transmission provides higher level of noise immunity through
error detection and correction,
- more flexible in the bandwidth,
- ease of implementation using large scale integration of circuits.
• Digital communications broadly refers to the transmission of
information using digital messages or bit streams.
• Before any signals (analog) can be transmitted digitally, signal must
be converted into digital form.
• Conversion from analog to digital is carried out with the goal of
minimizing the signal distortion introduced in the conversion process.
• There are three levels in converting an analog signals to digital, i.e.,
sampling, quantization, and encoding.
3
 Basic Elements of Digital Communication
System
Information
Source Encoder Channel Encoder Digital Modulator
Source
ASK,FSK,PSK Signal
Analog, digital signal Message Symbol Code Symbol
(Bits) (Bits) Channel

Information
Source Decoder Channel Decoder Digital Demodulator
Sink

Figure 3.1 : Basic Elements of Digital Communication System

Format (PCM steps)

Digital Information
Textual Information
Analog Information Sampler Quantizer Encoder Channel Encoder

Figure 3.2 : Formatting Process

4
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 by format the signals (refer Figure 3.2) and compressed
that signal. These digital signals is called as serial bits or bit streams.
Then, these bits are grouped to from message symbols. For example
PCM & Character Encoding(ASCII code).
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 or code word.
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.
5
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.
6
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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Continue

Figure 3.3: Digital Communication

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Continue

CHANNEL DIGITAL CHANNEL DIGITAL CHANNEL

ENCODER MODULATOR DEMODULATOR DECODER

SOURCE SOURCE
ENCODER DECODER

INFORMATION INFORMATION
SOURCE SINK

Figure 3.4: Digital Communication

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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
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Advantages of Digital communication compare
to Analog Communication
• Digital signals are less susceptible than analog signals to
interference caused by noise.
• Errors caused by noise and interference can be detected and
corrected systematically.
• Digital system more secure than analog systems 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.
• Digital circuits easier to be interfaced compare to analog circuits
(because there are two levels of digital signals only, '1 'and '0').
• Ease of signal regeneration
• more flexible in the bandwidth
• ease of implementation using large scale integration of circuits.

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M-ary Encoding

 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 voltage,
levels, frequencies, phases and so on) is an M-ary system where M = 4.
 The number of bits that necessary to produce a given number of
conditions (M) is expressed mathematically as;
n  log 2 M
Where; n = number of bits
M = number of conditions, levels or combinations possible
with n bits
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M-ary Encoding

 Equation above can be simplified and rearranged to

express the number of conditions possible, M with n
bits.
M  2n

 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.

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Pulse Modulation

• In Pulse Modulation, Information Signal is in

analog waveform. While Carrier signal is in digital
waveform.

• 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.

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Pulse Modulation
• Definition: Pulse Modulation (PM) is a process of SAMPLING analog
information signals into sampled signal before converting those into
digital signals.

• In Pulse Modulation the Information Signal is in analog waveform.

While Carrier signal(sampling pulses) is in digital waveform

• The properties of sampling pulses (carrier signal) such as width,

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

• 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

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Pulse Modulation
Techniques

Information signal

Sampling Pulses

PWM

PPM

PAM
(Single Polarity/Flat Top Sampling)

PCM
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Pulse Modulation Techniques
• 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.
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PULSE WIDTH MODULATION (PWM)

 Amplitude (A) and Position (P) of pulses are constant.

 Width (W) of pulses is varied proportional to the amplitude of
analog Information signal.
 The higher amplitude of Information signal, the wider of Pulses.
 Maximum analog amplitude gives widest pulse, and minimum
amplitude gives narrowest pulse.

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Continue

Analog signal

Sampled signal

PWM signal

Figure 3.5: Pulse Width Modulation Process

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PULSE POSITION MODULATION (PPM)

 Amplitude (A) and Width (W) of pulses are constant.

 Position (P) of pulses is varied proportional to the amplitude of
analog Information signal.
 The higher amplitude of Information signal, the farther to the
right the pulse is positioned.

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Continue

Analog signal

Sampled signal

PPM signal

Figure 3.6: Pulse Position Modulation Process

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PULSE AMPLITUDE MODULATION (PAM)

Analog Information Signal

Sample pulses

 Width (W) and Position (P) of pulses are constant.

 Amplitude (A) of pulses is varied proportional to the amplitude
of analog Information signal.
 The higher amplitude of Information signal, the higher
amplitude of Pulses.

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Continue

Sampled signal

Analog signal

PAM signal

Figure 3.7: Pulse Amplitude Modulation Process

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Digital Pulse Modulation

• In DPM, a code 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

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PULSE CODE MODULATION
(PCM)

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Pulse Code Modulation

• Definition: PCM is a digital pulse modulation technique to convert

the analog signal to digital signal.

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

i. Sampling
ii. Quantization
iii. Encoding

• Before we sample, we have to filter the signal to limit the maximum

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.
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Block Diagram of PCM

Information signal

Bandpass Filter

Filtered signal
e.g : 300 – 3400Hz

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PCM Process
Quantized
signal

PCM encoder

Sampling Quantizing Encoding 11…1100

Digital data

Sampling
signal

• The PCM signal is generated 1. Sampling

by carrying out three basic • Process where the information signal (analog signal) is
operations: sampled by sampling pulse signal which generated at
 Sampling certain sampling rate, fs.
 Quantizing 2. Quantization
 Encoding • Process of rounding off the amplitudes of sampled
(PAM) signal to a countable number of quantization
level.
3. Encode
• Process of translating the quantized signal into a
decimal code number. Then this decimal code number is
converted to its representative binary sequence.
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Pulse Code Modulation (PCM)

Figure 3.8: Block diagram of a single-channel, simplex PCM transmission

system. (Full)
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Pulse Code Modulation (PCM)

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Continue
• The analog signal is first band-limited to a maximum frequency by a
low pass filter (LPF) in accordance with the sampling theorem.
• It is then sampled and converted to a multilevel PAM signal. The
sampled pulses (PAM samples) are rounded off to the nearest pre-
determined level (quantized level), in the quantizer.
• Then the quantized samples are converted into groups according to
the binary code in the encoder,
• These coded pulses are then modulated digitally in the modulator
before being transmitted through the channel.

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Sample & Hold

• Sampling is to convert analog signals to digital ones.

• Sample & hold is the process which keeps sample height until the
next sample occurs.
• The constants height steps are necessary to provide time for
digitizing to take place.

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Type of PCM Techniques
• Delta Modulation
• Delta Sigma Modulation
• DPCM - Differential pulse-code modulation
• ADPCM - Adaptive Delta Pulse Code Modulation
• Adaptive Delta Modulation

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Application of PCM

• Widely use in telcom system, CD laser disks, digital audio recording,

digitized video special effects, voice mail, digital video.
• Also can be used in Radio Control Units.
• Radio transmitter and receiver for remote controlled cars, planes,
boats

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Sampling

• Definition: Sampling is a process where the information signal

(in analog signal) is sampled by pulse signal which is
generated at certain rate, fs.

• Sampling process will converts 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 sampling pulses is varied
proportional to the analog amplitude information signal.

• This sampling process is performed by an electronic switch

called Sampler as shown in Figure 3.9.
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Sampling

Figure 3.9

 According to Figure 3.10, Analog

signal is sampled every TS seconds.
 When the pulse is generated, the
switch will ‘ON’ and the amplitude
of information signal will be
produced. When there is no pulse,
the switch will ‘OFF’ and there is
no output will produce.
Figure 3.10 36
Figure 3.11 : Three different sampling methods for PCM

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SAMPLING THEOREM

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

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

• The higher the sampling rate, fs the closer sampling interval, Ts,
the closer the recovered signal approaches the original signal
(see Figure 3.12).

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

reproducing the original signal but this is not practical due to
the bandwidth limitation.

So, what minimum Sampling Rate, fs is required to

recover the original signal?

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Figure 3.12 : Three different Sampling Rate for PCM

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SAMPLING THEOREM

According to the Nyquist Sampling theorem; to

reconstruct the original signal, the sampling rate must be at
least 2 times the highest frequency contained in the signal.
fs ≥ 2fm(max)

EXAMPLE 3.1 :
Given the bandwidth of the telephone lines signal is 300 to 3400 Hz.
Determine the minimum sampling rate that suitable for sampling that
signal.

ANSWER :
fs = 2fm(max) Ts = 1/ fs = 1/6800Hz = 0.147ms
= 2 ( 3400Hz) = 6800Hz
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SAMPLING THEOREM
EXAMPLE 3.2 :
• 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 3.13 below;
• For part (a) It can be seen that 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.
41
Figure 3.13 : Recovery of a sampled sine wave for different sampling rates

Original Signal Reconstruct Signal 42

QUANTIZATION
• Definition: Quantization is a process of converting the infinite
number of possibilities to a finite number of levels.

• Analog signal has an infinite number of amplitude possibilities. By

using the quantization process, the amplitude of sampled signal
(PAM signal) is divided to a finite set of levels, L.

• In other words, Quantization is a process of rounding off the

amplitudes of sampled signal to a manageable number of levels.

• The number of levels, L for the sampled signal depend on the

number of bits, n used to code the signal.

• It used M-ary formula to determine the number of levels, L.

n = number of bits of
L  2n L = number of finite quantization level
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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

• Quantization process will convert the sampled signal (PAM

signal) into quantized signal by doing rounding-off process.

• The more levels used means that an analogue signal can be

described more accurate.

• However, a PCM code could have only 8 bits which equates to

only L= 28 or 256 levels.

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QUANTIZATION
Code Quantized signal = 3.5V Sampled signal (PAM) = 3.1V
number

7 ∆ = Step Size @
Quantization Interval @
Resolution @ Quantum
6

Figure 3.14 45
QUANTIZATION

• According to Figure 3.14 above, we assume that the amplitude of

sampled signal (PAM) is confined between two limits:
a min(-mp), Vmin = -4V and a max(mp), Vmax =+4V.

• Since we want to use 3 bit PCM code (n=3), so the Quantization Level,
L = 2n is;
L = 23 = 8 level

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

• Each sampled signal’s amplitude is quantized (rounding-off) to the

midpoint of the interval (quantization level) in which it lies. For
example, the second sample has sample amplitude value of 3.1V. After
quantization, the sample amplitude value is quantized to 3.5V level.
46
ENCODING

• Definition : Encoding is a process of translating the quantized

signal into a decimal code number. Then this 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 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).

• The essential features of binary PCM are shown in Figure 3.15.

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ENCODING
Binary Quantized signal Sampled signal
number
111

110

101

100

011

010

001

000

Figure 3.15

48
EXAMPLE 1

• From figure 3.15 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.

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EXAMPLE 2
Figure 3.16

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

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EXERCISE
Q. Encode the following Quantized signal

Note:
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 3.17
51
Quantization Error (Qe) & SQR
• When a signal is quantized, we introduce an error - the coded signal
is an approximation of the actual amplitude value.

• The difference between sampled and quantized value is referred as

the quantization error (Qe).
Qe  Quantized value - Sampled value (V)

• For example, for first sample in Figure 3.16, the sampled value is -
1.22V, while the quantize value is -1.5V. So;
Qe = quantized value – sampled value
= -1.5V – (-1.22V) = -0.28V

• Quantization error (Qe) is also called Quantization noise (Qn) where

the maximum Qe = ± ∆/2 52
Quantization Error (Qe) & 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: What is the SQR in the Figure 3.16?

Solution:
From the Figure 3.13, the number of quantization level, L = 8. Then
the number of bit is; log 8
n  log 2 L  log 2 8   3bits
log 2
Since, number of bits is 3, so the SQR is
SQR(dB)  6.02(3)  1.76  19.82dB
53
Quantization Error (Qe) & SQR

• It can be proven that the contribution of the Qe and SQR of

the signals depends on number of bits, n or number of
quantization level, L.

• The more number of bits, n, the more Quantization level, L

the smaller step size , which results in smaller quantization
errors, Qe.

• 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.

54
Exercise

The PCM samples are encoded into 4-bit system. If the minimum
sampling rate used is 8 kHz, find the frequency of the information
signal and the quantization level.

fs  2 fm
L  2n
8k
fm   4kHz L  2 4  16
2

Answer: fm= 4 kHz , L=16 levels

55
Exercise

A 4 kHz audio signal is transmitted by using PCM system where

quantization level, L is 256. calculate the number of bits for every
sample and its transmission bit rate.
Correction:
L  2n Bit rate  n  f m = n X fs
log 256  8 4 = 8 x (2 x 4k)
n  8bits
log 2  32kHz = 64 kHz.

Answer: n=8 bits, bit rate = 32kHz

56
Exercise

A digital signal has two levels. Calculate how many bits are needed
per level if 10110001 to send and sketch those digital signals.

n  log 2 M Number of bits per level, n = 1 bit

log 2
n  1bit
log 2

57
Digital Modulation Techniques

• 3 basic of digital modulation techniques:

i. Amplitude Shift Keying (ASK)
ii. Frequency Shift Keying (FSK)
iii. Phase Shift Keying (PSK)

58
Digital Modulation Techniques

In Digital Modulation, Information Signal is in digital

waveform. While Carrier signal is in analog waveform.

• Amplitude Shift Keying (ASK) - the amplitude (V) of the

carrier is varied proportional to the information signal.

• Frequency Shift Keying (FSK) - the frequency (f) of the

carrier is varied proportional to the information signal.

• Phase Shift Keying (PSK) - the phase (Ɵ) of the carrier is

varied proportional to the information signal.

59
 Digital Modulation Techniques

Carrier signal
fC

Data

ASK

FSK

PSK

Figure 3.18 60
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.
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ASK

• Use different amplitude to represent 0 and 1.

• Advantages:
 Simple, low bandwidth requirement
• Disadvantages
 Multi-path propagation, noise or path loss heavily influence the amplitude.
 A constant amplitude in wireless environment can not be guaranteed.
 It requires excessive bandwidth & is therefore a waste of power.
 Very susceptible to interference / Sensitive to interference.
• Used in wired optical communication.
 A light pulse =1, no light =0.

62
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 3.18.

• 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).

63
Frequency Shift Keying (FSK)
• Used varied frequency in response to information.
• When binary input signal changes from 0 to logic 1 and vice versa,
the output frequency shift between two frequencies: logic 1 – f1 and
logic 0 – f0
0 1 0 1
Amplitude

Time

1 baud 1 baud 1 baud 1 baud

1 second

• Advantage:
 Not susceptible to noise as much as ASK
• Disadvantage:
 Need large bandwidth
64
Continue
• Binary FSK (BFSK)
– One frequency for 0 and one frequency for 1.
– needs larger bandwidth
• Avoid discontinuity
– Discontinuity creates high frequencies as side effects.
– Continuous phase modulation (CPM) can be used.
• Demodulation:
– Use two bandpass filters for 2 frequencies.

65
Continue

S1
DATA F1

S2 FSK SIGNAL

Block diagram of FSK generator.

• FSK: Two different carrier frequencies are used and they are
switched ON and OFF by the binary signals.
 Binary ‘1’ or ‘mark’ switches one carrier ON while the other
carrier is OFF and
 Binary ‘0’ or ‘space’ switches the second carrier ON while the
first carrier is OFF.

66
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.

67
Phase Shift Keying (PSK)

• 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.

68
Phase Shift Keying (PSK)
• Use different phased signals to represent binary values
• A 180 degree phase shift can be used as the bit changes from 1 to
0 or from 0 to 1. This scheme is also called binary shift keying.
 0 = in phase with reference
Reference
 1 = out of phase with reference

0 1 0 1
Amplitude

Time

1 baud 1 baud 1 baud 1 baud

1 second

69
Phase Shift Keying (PSK)
• Binary PSK (BPSK)
– Shift the phase by 180.
• Synchronization is important
• Advantage:
 More resistant to interference / robust against interference
• Disadvantage:
 More complex transmitters and receivers.

70