CHAPTER 3

DIGITAL
COMMUNICATION
PROGRAMME OUTCOMES (POs)

PO1 Apply knowledge of mathematics, science and engineering

fundamentals to the solution of engineering problems.

PO2 Identify, formulate, research literature and analyze complex

electrical/electronic engineering problems reaching substantiated
conclusions.

COURSE OUTCOMES (COs)

CO1 Explain the different concepts of electronic communication in basic

communication engineering (C2).
CO2 Analyse the performance of analogue and digital modulation transmission
process in basic communication engineering (C4).
 DIGITAL TRANSMISSION

• Digital transmission : the transmission of digital signals

between two or more points in a communications
system.

• The signals can be binary or any other form of discrete-

level digital pulses.
• The original source may be in digital form, or it
could be analog signals that have been converted to
digital pulses before the transmission and
converted back to analog signals in the receiver.
• With digital transmission systems, a physical facility
(pair of wires, coaxial cable, optical fiber cable) is
required to interconnect the various points within
the system. The pulses will propagate through the
cable.
• Digital pulses cannot be propagated through a
wireless transmission system (Earth’s atmosphere
or free space (vacuum).
ADVANTAGES OF DIGITAL TRANSMISSION
Noise immunity
• Less susceptible to interference caused by noise compared to analog.

Suitable for signal processing

• Digital signal processing (DSP) is the processing of signals using digital
methods and includes band limiting the signal with filters, amplitude
equalization and phase shifting.

Suitable for multiplexing

• Multiplexing is a signal combining technique. It is the transmission of
information from one or more source or one or more destination over the
same transmission medium.

Simple storage
• Digital signals are much simpler to be stored than analog signals. The
transmission rate of digital signals can be easily changed to adapt to
different environments and to interface with different types of equipment.
Resistant to additive noise
• More resistant to additive noise compared to analog signals because they
use signal regeneration rather than signal amplification.

Further transmission distance

• Digital signals can be transported for longer distance than analog signals.

Simple to measure and evaluate

• Digital signals are simpler and easier to measure and evaluate than analog
signals. Therefore, it is easier to compare the error performance of one
digital system to another digital system

Easy error detection and correction

• Digitals signals allow transmission errors to be detected and corrected
more easily and accurately than is possible with analog signals.
DISADVANTAGES OF DIGITAL TRANSMISSION
Higher bandwidth requirement
• More bandwidth is required compared to the transmission of the original
analog signal. Bandwidth is one of the most important aspects of any
communications system because it is costly and limited.

Additional coding circuits’ requirement

• Analog signals must be converted to digital pulses prior to transmission and
converted back to their original analog form at the receiver, thus necessitating
additional encoding and decoding circuitry.

Precise time synchronization requirement

• Digital transmission requires precise time synchronization between clocks in
the transmitters and receivers.

System incompatibility
• Digital transmission systems are incompatible with older transmission systems.
Pulse Modulation
• Pulse modulation : Methods of converting information into
discrete pulse form for transferring pulses from a source to
destination over a physical transmission medium.
• Four (4) predominant methods for Pulse Modulation :

Pulse Width Modulation (PWM).

Pulse width (active portion of the duty cycle) is proportional to
the amplitude of the analog signal.
The maximum analog signal amplitude produces the widest
pulse, the minimum analog signal amplitude produces the
narrowest pulse.
Pulse Position Modulation (PPM)
 The position of a constant-width pulse is varied according to
the amplitude of the analog signal.
 The higher the amplitude of the sample, the farther to the
right the pulse is positioned within the prescribed time slot.
 The highest amplitude sample produces a pulse to the far
right, and the lowest amplitude sample produces a pulse to
the far left.
Pulse Amplitude Modulation (PAM)
 The amplitude of a constant-width, constant-position pulse is
varied according to the amplitude of the analog signal.
 The amplitude of a pulse coincides with the amplitude of the
analog signal.
 PAM waveforms resemble the original analog signal more
than the waveforms for PWM or PPM.
Pulse Code Modulation (PCM)
The analog signal is sampled and converted to a serial n-bit
binary code for transmission.
The binary number varies according to the amplitude of the
analog signal.
Each code has the same number of bits and requires the same
length of time for transmission.
most prevalent method of pulse modulation
Vm(t)
Vmax

(a) analog signal

Vmin

(b) sample pulse

PWM

(c) PWM
t

PPM

(d) PPM
t

PAM

(e) PAM
8-bit PCM code t

(f) PCM
t
 PULSE CODE MODULAT ION (PCM)
• PCM are commonly used for digital transmission
• PCM is a binary system where a pulse or lack of pulse within a
prescribed time slot represents either a logic 1 or a logic 0
condition.
• With PCM, the pulses are of fixed length and fixed amplitude.
• PWM, PPM, and PAM are digital but seldom binary, as a pulse
does not represent a single binary digit.
3.3.1) PCM BLOCK DIAGRAM
• Figure 3.2 shows a simplified block diagram of a single-channel, simplex
(one-way only) PCM transmitter and receiver system.

Figure 3.2: Simplified block diagram of a single-channel, simplex PCM transmission system
PCM BLOCK DIAGRAM
PCM Transmitter
• The bandpass filter limits the frequency of the analog input
signal to the standard voice-band frequency range of 300 Hz to
3000 Hz.
• The sample-and-hold circuit periodically samples the analog
input signal and converts those samples to a multilevel PAM
signal.
• The analog-to-digital converter (ADC) converts the PAM
samples to parallel PCM codes.
• The parallel-to-serial converter convert parallel PCM codes to
serial binary data in and then outputted onto the transmission
line as serial digital pulses.
• The transmission line repeaters are placed at prescribed
distances to regenerate the digital pulses.
PCM BLOCK DIAGRAM
PCM Receiver
• The serial-to-parallel converter converts serial pulses received
form the transmission line to parallel PCM codes.
• The digital-to-analog converter (DAC) converts the parallel PCM
codes to multilevel PAM signals.
• The hold circuit is basically a lowpass filter that converts the
PAM signals to its original analog form.
• PCM is one of the most commonly used method to convert analog data into
digital form. It involves three steps:
• Sampling
• Quantization
• Encoding
SAMPLING

The analog signal is sampled every T interval. Most

important factor in sampling is the rate at which
analog signal is sampled (sampling rate).
QUANTIZATION

Every discrete pattern from sampling shows the amplitude of the

analog signal at that instance. The quantization is done between the
maximum amplitude value and the minimum amplitude value.
Quantization is approximation of the instantaneous analog value.
ENCODING

Each approximated value is then converted

into binary format.
3.3.2) Sampling and Sampling Rate
Sampling
• Sampling is the process of taking samples of the analog input
signal at a rate of Nyquist sampling frequency.
• The function is to periodically sample the continually changing
analog input voltage and convert those samples to a series of
constant-amplitude pulses that can more easily be converted to
binary PCM code
Sampling Rate
• Nyquist Sampling Theorem state that :
“For a sample to be reproduced accurately at the receiver,
the sampling rate (fs) must be sampled at least twice of the
highest audio input frequency, fm (max) ”
• Consequently, the minimum sampling rate is equal to twice the
highest audio input frequency.
• Mathematically, the Nyquist sampling rate is :
fs ≥ 2fm(max) (3.1)

or the minimum Nyquist sample rate is :

fs(min) =2fm(max)

where: fs = Nyquist sample rate (Hz or sample/second)

fs(min) = Minimum Nyquist sample rate (Hz)
fm(max) = Maximum audio frequency to be sampled (Hz)

• Consequently, the minimum sampling rate is equal to twice the highest

audio input frequency.
• If fs is less than two times fm(max), distortion will result. This distortion is
called aliasing or foldover distortion.
3 basic condition of sampling process :

1) Sampling at fs = 2fm(max)
 When the modulating signal is sampled at a minimum sampling rate, the
frequency spectrum is shown below.
 The output includes the original inputs (audio signal), the fundamental
frequency of the sampling pulse with their sum and difference (fs+- fm ) and
all the harmonics of fm and fs (2 fs +- fm , 3 fs, +- fm , 4 fs +- fm , )and so on.
2) Sampling at fs > 2fm(max)

 The sampling rate creates a guard band between fm(max) and the lowest
frequency component (fs - 2fm(max)) of the sampling harmonics.
3) Sampling at fs < 2fm(max)

 The side frequencies from one harmonic fold over into the sideband
of another harmonic.
 The frequency that folds over is an alias of the input signal (hence the
name “aliasing” or “foldover distortion”). If an alias side frequency
from the first harmonic folds over into the input audio spectrum, it
cannot be removed through filtering or any other technique.
Example 3.1

For a PCM system with a maximum audio input

frequency 4 kHz, determine the minimum sample
rate and the alias frequency produced if a 5 kHz
audio signal were allowed to enter the sample-
and-hold circuit.
Solution
Using Nyquist’s sampling theorem (Equation 3.1), we have

fs ≥ 2fm therefore, fs ≥ 8 kHz

If a 5 kHz audio frequency entered the sample-and-hold

circuit, the output spectrum shown in Figure 3.4 below is
produced. It can be seen that the 5 kHz signal produces an
alias frequency of 3 kHz that has been introduced into the
original audio spectrum.
3.3.2) Quantization and Quantization Error
Quantization
• Quantization is the process of converting an infinite number of amplitude
possibilities (analog signal samples) to a finite number of conditions (pre-
determined discrete levels).
• The number of quantization levels, L, depends on the number of bits per
sample, n, used to code the analog signal, where :

L  2n (3.2)

2-bit resolution with four levels of quantization 3-bit resolution with eight levels of quantization
• Resolution, ΔV : the magnitude of the step-size of the quantization
levels. The resolution depends on the maximum voltage, Vmax, and the
minimum voltage, Vmin, of the information signal, where :

Vmax  Vmin
V  (3.3)
L 1

• The smaller the magnitude of the minimum step size, the better
(smaller) the resolution and the more accurately the quantization
interval will resemble the actual analog sample.
• Figure 3.5 shows an analog input signal, the sampling pulse, the
corresponding quantized signal (PAM), and the PCM code for each
sample.
• Each sample voltage is rounded off (quantized) to the closest available
level and then converted to its corresponding PCM code.
Quantization Error
• Quantization error (Qe) is the round-off errors in the transmitted signal
produced when the code is converted back to analog in the receiver.
• The maximum quantization error is given by :
V resolution
Qemax   
2 2
• Quantization error is also called quantization noise (Qn) and its
maximum magnitude is one half the voltage of the minimum step size
(Vlsb/2). For this example, Qe = 1 V/2 or 0.5 V.
 Example 3.2

A PCM system has a quantization level of 256 and

the allowable quantization error is 0.01 V. Determine
the:
i) resolution and
ii) peak to peak amplitude of the information signal.
Solution
i) ,
3.3.4) Encoding

Encoding is a process where each quantized

sample is digitally encoded into n-bits codeword,
where
n  log 2 L

where n = number of bits/sample

L = number of quantization levels
Example 3.3
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 number of quantization
levels.
Solution :

Form Nyquist sampling theorem, the minimum sampling rate,

fs(min) = 2fm.
Therefore, the frequency of the information signal is :

The number of quantization levels, L :

3.3.5) Transmission bit rate

• Transmission bit rate, R is the rate of information transmission

(bits/sec).
• It depends on the sampling frequency, fs and the number of bits
per sample used, n to encode the signal.
• The transmission bit rate is given by :

R  n fs

where n = number of bits/sample

fs = sampling frequency
Example 3.4
A PCM system transmits a 15 kHz audio signal.
The sampling frequency used is 15% higher than
the minimum sampling frequency of Nyquist
rate. The quantization level is 256 levels.
Determine the:
i) number of bits for every sample,
ii) sampling frequency, and
iii) transmission bit rate.
Solution :
ii)

iii)
Introduction to Coded Modulation
• Digital modulation : Transmission of digitally modulated analog signals
(carriers).
v(t )  V sin( 2  ft   )

ASK FSK PSK

QAM

• If the information signal is digital & the amplitude (V) of the carrier is varied
proportional to the information signal, an Amplitude Shift Keying (ASK) is
produced.
• If the frequency (f) is varied proportional to the information signal,
Frequency Shift Keying (FSK) is produced.
• if the phase of the carrier ( ) is varied proportional to the information
signal, Phase Shift Keying (PSK) is produced.
• If both the amplitude and the phase are varied proportional to the
information signal, Quadrature Amplitude Modulation (QAM) results.
Amplitude Shift Keying (ASK)
• Amplitude-shift keying (ASK) is a process where a 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.

• Amplitude shift keying is sometimes called digital amplitude modulation

(DAM) or on-off keying (OOK).

vc
vask (t )  [1  vm (t )][ cos(c t )]
2
 If binary ‘1’, carrier wave is transmitted.

Vc
vask (t )  [1  1][ cos(c t )]  Vc cos(c t )
2
 If binary ‘0’, carrier wave is suppressed.

Vc
vask (t )  [1  1][ cos(c t )]  0
2

Figure 3.7: Digital amplitude modulation: (a) input binary, (b) output DAM/ASK waveform.
Frequency Shift Keying (FSK)
• Similar to the standard frequency modulation (FM) except the modulating
signal is binary signal that varies between two discrete voltage levels.

v fsk (t )  Vc cos[ 2 ( f c  vm (t )f )t ]

 For a logic‘1’ input, vm (t )  1

v fsk (t )  Vc cos[ 2 ( f c  f )t ]

 For a logic ‘0’ input, vm (t )  1

v fsk (t )  Vc cos[ 2 ( f c  f )t ]
Frequency Shift Keying (FSK)
 When the binary input (fb) changes from a logic 1 to a logic 0
and vice versa, the FSK output frequency shifts from a mark
(fm) to a space (fs) frequency and vice versa.
 In Figure 3.8(a), the mark frequency is the higher frequency
(fc + Δf) and the space frequency is the lower frequency
(fc - Δf).

Figure 3.8: FSK in the time domain: (a) waveform, (b) truth table.
Question
Sketch sample analog output waveforms for the
following digital modulation techniques for a
binary input of 1011011.

i) Amplitude Shift Keying (ASK)

ii) Frequency Shift Keying (FSK)
3.4.3 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 phase modulation except that in PSK the
input signal is a binary digital signal and a limited
number of output phases are possible.
• The input binary information is encoded into groups of
bits before modulating the carrier.
• The number of output phases is defined by M.
3.4.3.1 M -ary Encoding
• M-ary is derived from M binary, where M represents a digit
corresponds to the number of conditions, level or combinations
possible for a given number of binary variables.

• The number of conditions possible with n bits can be expressed

as:

2n  M

• For example, with two bits, 2  4

2
conditions are possible.
3.4.3.2 Binary Phase Shift Keying (BPSK)
• The simplest form of PSK is Binary Phase Shift Keying
(BPSK), where “binary” meaning “2”.

• In BPSK, two outputs are possible for a single carrier

where n(input bits) = 1 and M(output phase) = 2.

• Therefore, with BPSK, two phases (21 = 2) are possible

for the carrier. One output phase represents a logic
“1” and the other a logic “0”.

• As the input digital signal changes state (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 180° out of
phase.
3.4.3.2 Quaternary Phase Shift Keying(QPSK)
• QPSK is an M-ary encoding technique where M = 4, four output phases are
possible for a single carrier frequency.

• Because there are four different output phases, there must be four different
input condtions.

• Using the same equation as before, 2 n  M , the number of input bits, n

for M = 4 is 2 bits (which is also called dibits).

• With 2 bits, there are four possible input conditions: 00, 01, 10, 11

• In the modulator, each dibit code generates one of the four possible output
phases (+45º, +135º, -45º and -135º).
 (a) (c)

(b)

Figure 3.10 : QPSK Modulator (a) Phasor diagram, (b) Constellation diagram (c) Truth table
3.4.3.3 8-Phase Shift Keying(8-PSK)
• With 8-Phase Shift Keying (8-PSK), three bits are encoded forming “tribits”
and producing eight different output phases.
• In 8-PSK, n = 3, M = 8, and there are eight possible output phases.

23  8

(a) (b) (c)

Figure 3.11 : 8-PSK Modulator (a) Truth table (b) Phasor diagram, (b) Constellation diagram
3.4.3.6 Quadrature-Amplitude Modulation (QAM)
• Quadrature-amplitude modulation (QAM) is a form of digital modulation
similar to PSK except the digital information is contained in both the
amplitude and the phase of the transmitted carrier.

• Unlike 8-PSK, the output signal from an 8-QAM modulator is not a constant-
amplitude signal.

• Figure below shows the output phase-versus-time relationship for an 8-

QAM modulator. Note that there are two output amplitudes, and only four
phases are possible.

8-QAM modulator: output phase-versus-time relationship.

Figure 3.12 : QAM Modulator (a) Truth table (b) Phasor diagram, (b) Constellation diagram
Exercise

Given the input binary to a digital modulator

is 1100110. Sketch the output waveform of:
i. Amplitude Shift Keying (ASK)
ii. Frequency Shift Keying (FSK)
iii. Binary Phase Shift Keying (BPSK)